Elastix :: The Open Source Unified Communications Server - Foros Sindicación del Foro Kunena http://elastix.org/ Fri, 24 May 2013 09:30:31 -0500 Kunena 1.5.11 http://elastix.org/components/com_kunena/template/default/images/english/emoticons/rss.gif Potenciado por Kunena - Kunena Spanish! Web http://elastix.org/ Sindicación del Foro Kunena Subject: probleme trunk iax2 - by: danardf http://elastix.org/index.php/en/component/kunena/84-le-coin-du-debutant/122414-probleme-trunk-iax2.html#122415 Bonjour et bienvenue sur notre forum Elastix. Il y a ICI (http://www.elastix.org/index.php/en/product-information/manuals-books.html) sur le site d'Elastix dans la rubrique Manuals Books, deux doc toutes faites sur le sujet et bien expliqué: TRUNKING - Trunking between two Elastix PBX Systems Via Internet - Trunking between two Elastix PBX Systems Via VPN Author: Bob Fryer Autre solution, utiliser l'addon Distributed DialPlan . Voilà. Fri, 24 May 2013 09:30:00 -0500 Subject: How to: Video calls between two Elastix 2.3.0 - by: Shaboha http://elastix.org/index.php/en/component/kunena/26-tips-and-tricks/122413-how-to-video-calls-between-two-elastix-230.html#122413 Hello, There are two server. Elastix 2.3.0 - IAX2 Trunk - Elastix 2.3.0. within each server video calls is working. but the video call between two servers is not working. videosupport = yes and allow videocodecs. but when you call phones show bitrate... but the audio channel of about 2 Mbit a video channel about 64 Kbit. how set up correctly? Fri, 24 May 2013 08:21:48 -0500 Subject: Problema grave de seguridad con llamadas SIP - by: virusbcn http://elastix.org/index.php/en/component/kunena/11-ayuda/122412-problema-grave-de-seguridad-con-llamadas-sip.html#122412 Hola amigos, hace tiempo que no me paso por aquí y HOY me he encontrado con un problema grave que me tiene mosqueado, a ver si me podéis ayudar. Me llama un cliente diciendo que en las dos últimas facturas de las líneas RDSI tiene llamadas a paises internacionales a horas intempestivas por importes altos, rápidamente me conecto a la centralita y empiezo a revisar, y observo llamadas sip desde la IP 80.28.196.21, cuyas trazas en el asterisk son: -- Executing [800972592727626@from-sip-external:1] NoOp("SIP/80.28.196.21-00001ff0", "Received incoming SIP connection from unknown peer to 800972592727626") in new stack -- Executing [800972592727626@from-sip-external:2] Set("SIP/80.28.196.21-00001ff0", "DID=800972592727626") in new stack -- Executing [800972592727626@from-sip-external:3] Goto("SIP/80.28.196.21-00001ff0", "s,1") in new stack -- Goto (from-sip-external,s,1) -- Executing [s@from-sip-external:1] GotoIf("SIP/80.28.196.21-00001ff0", "0?checklang:noanonymous") in new stack -- Goto (from-sip-external,s,5) -- Executing [s@from-sip-external:5] Set("SIP/80.28.196.21-00001ff0", "TIMEOUT(absolute)=15") in new stack Channel will hangup at 2013-05-24 12:11:42.385 CEST. -- Executing [s@from-sip-external:6] Answer("SIP/80.28.196.21-00001ff0", "") in new stack == Spawn extension (from-sip-external, s, 6) exited non-zero on 'SIP/80.28.196.21-00001ff0' -- Executing [h@from-sip-external:1] Hangup("SIP/80.28.196.21-00001ff0", "") Automáticamente me he puesto a revisar el Elastix por si faltara alguna medida de seguridad y he comprobado que las extensiones SIP sólo se puedan conectar desde la red local (192.168.x.x), que en el apartado "General settings" el "Allow incoming sip anonymous" está en OFF, y de paso he puesto un código a marcar en las llamadas internacionales, esto último SÍ que ha conseguido parar las llamadas salientes, pero sigo recibiendo intentos y peticiones de llamada que provienen de esta IP. ¿Qué puedo hacer? Ya me imagino que la mejor manera sería hacer ip_tables para sólo permitir conexiones SIP hacia mi proveedor SIP, pero antes de liarme con esto ... ¿puedo mejorar la seguridad de Elastix? porque aún así sigue conectado y la segunda sería, tengo la IP que siempre es la misma, en España esto se puede denunciar ¿?? ¿Cómo ???? Gracias adelantadas Fri, 24 May 2013 05:19:08 -0500 Subject: Identificar y desviar llamada - by: tmh http://elastix.org/index.php/en/component/kunena/50-novatos/122178-identificar-y-desviar-llamada.html#122411 Muchas gracias por tu respuesta jgutierrez, pero creo que para mi caso no serviría, ya que son bastantes números los que tengo que comprobar y tendría que crear una ruta entrante para cada Caller ID que tengo que comprobar. Además, aun haciendo rutas entrantes en base a los Caller ID, no funcionaría correctamente ya que a mi me llegan llamadas desde cualquier Caller ID a un DID concreto y tengo que responder a todos, entonces si tengo mis rutas entrantes para unos ciertos Caller ID, que pasa si llama alguien a mi DID y no coincide su Caller ID con los que yo defino en las rutas entrantes?? Tendría que crear también una ruta entrante en la que no especificase el Caller ID para esas llamadas, pero entonces por ahí podría entrar cualquier número (incluidos esos Caller ID que yo quiero comprobar...) No se si me he explicado bien o si me lié un poco... :P De todas formas, estoy viendo como hacer un script donde comparo los Caller ID con los números que yo introduzco en un archivo .csv (los números que quiero chequear), y luego en un "macro" poner un GoToIf para escoger la accion que deseo hacer según el resultado de la comparación. ¿Voy muy desencaminado? Gracias de nuevo, un saludo! Fri, 24 May 2013 04:04:17 -0500 Subject: Elastix CallCenter for outbounds campaigns - by: laurajones12 http://elastix.org/index.php/en/component/kunena/19-call-center/105950-elastix-callcenter-for-outbounds-campaigns.html#122410 Hi, for such type of problems you can try on hold messages . By using this you can forward the messages to your callers. Fri, 24 May 2013 03:37:30 -0500 Subject: Cannot reach Elastix from another network - by: Richardf http://elastix.org/index.php/en/component/kunena/3-help/122408-cannot-reach-elastix-from-another-network.html#122408 Hello, anybody please help. Here is situaltion. I have working Elastix of company network network 192.168.1.0/24 Second network (192.168.7.0/24) is on diferent place with IPSec tunel to first network, everything work fine but Im not able to reach PBX, no web interface, no ping, no connection. I tryed to turn off firewall on elastix, but its not working. Is there anything else I have to turn on or configure to be able to reach PXB from diferent network ? thank you Fri, 24 May 2013 02:25:45 -0500 Subject: Voicemail to email - by: Puttster http://elastix.org/index.php/en/component/kunena/25-newbies-corner-/122407-voicemail-to-email.html#122407 Well, Im lost with emailing VM's. The email account that I have set up works in thunderbird for sending and recieving, but i seem to be missing the point somewhere. The remote SMTP settings seem OK, but i don't know if the domain list or any roundcube settings need to be played with. The email address is with our account on IX web hosting I don't think that should be a problem. The local network does not have a domain controller or anything like that. The modem is a Billion 7402NX. I guess what would be most helpful would be knowing what part of elastix "talks" to what and what order this happens. As I said, I AM LOST ON THIS ISSUE!! Please Help. Thu, 23 May 2013 23:46:24 -0500 Subject: Have FXS port call Ring Group? - by: apextechservices http://elastix.org/index.php/en/component/kunena/3-help/122244-have-fxs-port-call-ring-group.html#122402 Actually, I determined yesterday that I need to use the FXO port and not FXS. The DoorKing gate system sits "inline" between the CO and the PBX and passes the dialtone and ring voltage through. The DoorKing acts as a relay mechanism. It also generates it's own ring voltage to ring the FXO port on Elastix PBX. I have a new problen however. I'm trying to route inbound calls ONLY on FXO Port 1 to ring group x600. I want FXO Ports 2-4 simply go to the IVR/auto-attendant and dedicate FXO Port 1 to the door gate system. Seems like such a simple thing to do but the web interface only shows how to route via DID or CallerID number. Matching by CallerID would have made my life easier and the more expensive DoorKing does generate its own CallerID number when it sends ring voltage. Alas, this DoorKing model doesn't generate CallerID when it calls FXO Port 1. So to sum up, I can't figure out how to send inbound calls to Sangoma FXO Port 1 to ring group 600 while FXO Ports 2-4 go to IVR. Thoughts? Thu, 23 May 2013 20:05:10 -0500 Subject: logs too big - by: c8aj http://elastix.org/index.php/en/component/kunena/25-newbies-corner-/122246-logs-too-big.html#122400 man.. thanks a lot!!! you are the best!!!... at last... after a long search.. you helped me figure out where that Gbs come from!!!! , now sorry to make a new question after this one, but this one came to my mind right now... too big logs.. can make elastix operation unstable?? is that true.. or it is just a urban legend?? Thu, 23 May 2013 18:00:31 -0500 Subject: Dialing a external conference bridge - by: gbrower http://elastix.org/index.php/en/component/kunena/3-help/122399-dialing-a-external-conference-bridge.html#122399 Hello, We are running Elastix 2.3 and asterisk 1.8.18.0 on a centos server. Our office has been dialing a 866 conference call number and it will just continue to ring but never pick up. You can dial the number from a cell phone and it will connect immediately. We are able to call other conference call numbers like go to meeting but this particular number doesn't connect. Do you think this is an elastix issue or a voip provider issue? Thank you for your help. Thu, 23 May 2013 16:53:59 -0500 Subject: Cisco SPA5XX Endpoint support - by: Bob http://elastix.org/index.php/en/component/kunena/29-announcements/120802-cisco-spa5xx-endpoint-support.html#122398 Climato Thanks for the addresses for the SPA508G. I believe the changes will be built into Elastix 2.5, but if you cant wait, the instructions are available (although I will add the address you have provided in the next day or so). Regards Bob Thu, 23 May 2013 16:53:16 -0500 Subject: No actualiza monitoring - by: jgutierrez http://elastix.org/index.php/en/component/kunena/55-call-center/122255-no-actualiza-monitoring.html#122397 Deberías postearlo en: http://bugs.elastix.org Thu, 23 May 2013 16:51:19 -0500 Subject: Outbound Caller ID not working on trunk ? - by: jgutierrez http://elastix.org/index.php/en/component/kunena/3-help/122254-outbound-caller-id-not-working-on-trunk-.html#122396 If you set any other outbound caller id, and it is not authorized by your provider, they will drop the call, that is an anti-spoofing mechanism. Thu, 23 May 2013 16:49:59 -0500 Subject: usuario de elastix. - by: jgutierrez http://elastix.org/index.php/en/component/kunena/11-ayuda/122228-usuario-de-elastix.html#122393 Elastix utiliza sqlite (/var/www/db/) Mientras que freepbx utiliza mysql Thu, 23 May 2013 16:35:50 -0500 Subject: Ayuda de documentacion - by: jgutierrez http://elastix.org/index.php/en/component/kunena/50-novatos/122189-ayuda-de-documentacion.html#122392 Sí claro... Revisa la sección de libros & manuales dentro de este sitio. Thu, 23 May 2013 16:31:07 -0500 Subject: Llamadas Salientes Que Inician por '5' - by: jgutierrez http://elastix.org/index.php/en/component/kunena/11-ayuda/122183-llamadas-salientes-que-inician-por-5.html#122391 Pega la salida del CLI (asterisk -r) de una llamada con problemas y otra sin problemas. Thu, 23 May 2013 16:28:41 -0500 Subject: SIP TLS y SRTP en Elastix - by: diego478 http://elastix.org/index.php/en/component/kunena/53-trucos/105875-sip-tls-y-srtp-en-elastix.html#122389 Hola a todos!! Estoy tratando de implementar esta solución! Alguien pudo descargar el archivo. Muy bueno el post Juan, podras enrchivo o subirlo para que lo descarguemos- gracias! Thu, 23 May 2013 16:27:08 -0500 Subject: Mensaje de extension a transferir y de ocupado - by: ndaniellq http://elastix.org/index.php/en/component/kunena/11-ayuda/122388-mensaje-de-extension-a-transferir-y-de-ocupado.html#122388 Buen dia me gustaria como puedo hacer para que cuando hago una transferencia a otra extension le suene un mensaje con el numero de la extension a la persona que estoy transfiriendo. Tambien me gustaria saber como puedo hacer para que cuando una extension no esta disponible o esta ocupada le suene un mensaje a la persona diciendole que la extension a la que esta llamando no esta disponible o si esta ocupada que esta ocupada pero que si desea esperar a que se desocupe. Thu, 23 May 2013 15:49:47 -0500 Subject: Configuración PBX - by: hgeorge123 http://elastix.org/index.php/en/component/kunena/11-ayuda/122384-configuracion-pbx.html#122387 Si la llamada proviene de la pstn podrias hacer un time condition busca en google un poco y veras como se hace es muy facil hacerlo Thu, 23 May 2013 15:12:33 -0500 Subject: Voicemail box size limit. - by: Oleg B http://elastix.org/index.php/en/component/kunena/3-help/122386-voicemail-box-size-limit.html#122386 Hi all! I've got strange situation with voicemail in Elastix 2.4. No new messages could be written to extension's voicemail box. Caller hear warning "...voicemail box size limit is reached" after. At the same time there are much free space at the harddisk. Where in configuration files are marked follow parameters: maximum amount of voicemail messages in voicemail box, maximum size of voicenail message (in seconds) and so one? Thu, 23 May 2013 15:08:47 -0500 Subject: Route to ring group depending on analog trunk? - by: apextechservices http://elastix.org/index.php/en/component/kunena/3-help/122385-route-to-ring-group-depending-on-analog-trunk.html#122385 I'm trying to route inbound calls to ring group depending on the analog trunk dialed. Can't seem to figure out the physical port mapping to inbound routes from the Elastix web admin. Hardware used for analog trunks and extensions: Sangoma A200 4-port FXO and 4 Port FXS Essentially, I'm having difficulty figuring out how to separate the analog trunk group (ZAP Channel g0 - Zap Identifier = g0) into individual trunk lines that I can then set specific inbound and outbound rules for. In addition to the default 'ZAP Channel g0' (Zap Identifier = g0), I also added 4 ZAP Trunks Trunk Name = trunk1 Zap Identifier = 1 Trunk Name = trunk2 Zap Identifier = 2 Trunk Name = trunk3 Zap Identifier = 3 Trunk Name = trunk4 Zap Identifier = 4 I'm not sure if the Zap Identifier maps to the physical port or not. I left the default 'ZAP Channel g0', so maybe that overrides these new trunks? Am I supposed to delete this default ZAP group? Anyway, here's what I'm trying to do: Scenario 1: Door Buzzer pressed - generates ring-voltage to Sangoma FXO trunk port #1 (Zap Identifier 1) - Rings specific trunk port (trunk1) on Elastix - Elastix detects this as a call on trunk1 and routes to ring group x600 to ring all IP phones (The other trunks 2, 3, and 4 go to IVR) - All IP Phones ring. First person to answer talks then presses 8 or 9, which opens DoorKing I can't seem to detect which trunk line the call comes in on, so I temporarily modified the default Incoming Route with no CallerID to route to x600 just as a test. This test works - was able to open door gate by pressing '9' after answering call. But now I need to restrict it so it ONLY routes to x600 on trunk1 Scenario 2: Employee dials the DoorKing connected to trunk1 from an IP phone extension Employee presses a certain line (line 2) on IP phone which picks specific trunkport1, goes off-hook to get 2nd dialtone, and then they can type touch-tone/DTMF digits (i.e. *79) to communicate with DoorKing, which is connected "inline" between the central office dialtone signal and the Elastix PBX. (it acts as a relay mechanism). Lastly, I need to be able to set number of rings on a specific trunk port (NOT GLOBALLY on all ports) to say 3 rings. How do I do this? Please help. Thu, 23 May 2013 14:35:47 -0500 Subject: cannot make out going call - by: deeeeeeeeeen http://elastix.org/index.php/en/component/kunena/25-newbies-corner-/122155-cannot-make-out-going-call.html#122383 Hi Bob Thanks so much for your valuable time, here is to answer your questions 1/. 119.136.221.140 it is my public ip at home 2/. 202.82.229.137 it is a solid hardware server which the elastix is installed 3/. one phone is zopier (838) as you can see and other is a Huawei sipphone (833) 4/. I am making testing at home(119.136.221.140) and the elastix is at external(202.82.229.137) therefore you see the public ip 5/. in the elastix I have not set anything yet only two extensions and one trunk, and My softphone and the Huawei phone can work on other sip accounts, so there is no problems on the firewall here is the debug to the trunk ip Connected to Asterisk 1.8.21.0 currently running on elastix (pid = 6736) Verbosity is at least 6 Really destroying SIP dialog '3a0c18100094e487652fa8507cc27bcc@127.0.0.1' Method: REGISTER REGISTER 11 headers, 0 lines Reliably Transmitting (NAT) to 58.254.250.26:5060: REGISTER sip:58.254.250.26 SIP/2.0 Via: SIP/2.0/UDP 202.82.229.137:5060;branch=z9hG4bK7b191458;rport Max-Forwards: 70 From: ;tag=as0d399bec To: Call-ID: 3a0c18100094e487652fa8507cc27bcc@127.0.0.1 CSeq: 6244 REGISTER User-Agent: FPBX-2.8.1(1.8.21.0) Authorization: Digest username="890777", realm="VoipSwitch", algorithm=MD5, uri="sip:58.254.250.26", nonce="136905636120210914005260120120", response="417fcfabdd1deb93c2d77e801cf2f793" Expires: 120 Contact: Content-Length: 0 --- SIP/2.0 200 OK CSeq: 6244 REGISTER Via: SIP/2.0/UDP 202.82.229.137:5060;branch=z9hG4bK7b191458;rport From: ;tag=as0d399bec Call-ID: 3a0c18100094e487652fa8507cc27bcc@127.0.0.1 To: ;tag=240517130216 Contact: ;expires=60 Expires: 60 Content-Length: 0 --- (9 headers 0 lines) --- Scheduling destruction of SIP dialog '3a0c18100094e487652fa8507cc27bcc@127.0.0.1' in 32000 ms (Method: REGISTER) Really destroying SIP dialog '3a0c18100094e487652fa8507cc27bcc@127.0.0.1' Method: REGISTER REGISTER 11 headers, 0 lines Reliably Transmitting (NAT) to 58.254.250.26:5060: REGISTER sip:58.254.250.26 SIP/2.0 Via: SIP/2.0/UDP 202.82.229.137:5060;branch=z9hG4bK53f71acc;rport Max-Forwards: 70 From: ;tag=as7f27c2a0 To: Call-ID: 3a0c18100094e487652fa8507cc27bcc@127.0.0.1 CSeq: 6245 REGISTER User-Agent: FPBX-2.8.1(1.8.21.0) Authorization: Digest username="890777", realm="VoipSwitch", algorithm=MD5, uri="sip:58.254.250.26", nonce="136905636120210914005260120120", response="417fcfabdd1deb93c2d77e801cf2f793" Expires: 120 Contact: Content-Length: 0 --- SIP/2.0 200 OK CSeq: 6245 REGISTER Via: SIP/2.0/UDP 202.82.229.137:5060;branch=z9hG4bK53f71acc;rport From: ;tag=as7f27c2a0 Call-ID: 3a0c18100094e487652fa8507cc27bcc@127.0.0.1 To: ;tag=240518130201 Contact: ;expires=60 Expires: 60 Content-Length: 0 --- (9 headers 0 lines) --- Scheduling destruction of SIP dialog '3a0c18100094e487652fa8507cc27bcc@127.0.0.1' in 32000 ms (Method: REGISTER) elastix*CLI> sip set debug 58.254.250.26 No such command 'sip set debug 58.254.250.26' (type 'core show help sip set debug' for other possible commands) Really destroying SIP dialog '3a0c18100094e487652fa8507cc27bcc@127.0.0.1' Method: REGISTER elastix*CLI> sip set debug ip 58.254.250.26 SIP Debugging Enabled for IP: 58.254.250.26 REGISTER 11 headers, 0 lines Reliably Transmitting (NAT) to 58.254.250.26:5060: REGISTER sip:58.254.250.26 SIP/2.0 Via: SIP/2.0/UDP 202.82.229.137:5060;branch=z9hG4bK76963f9a;rport Max-Forwards: 70 From: ;tag=as2b2b9dc5 To: Call-ID: 3a0c18100094e487652fa8507cc27bcc@127.0.0.1 CSeq: 6246 REGISTER User-Agent: FPBX-2.8.1(1.8.21.0) Authorization: Digest username="890777", realm="VoipSwitch", algorithm=MD5, uri="sip:58.254.250.26", nonce="136905636120210914005260120120", response="417fcfabdd1deb93c2d77e801cf2f793" Expires: 120 Contact: Content-Length: 0 --- SIP/2.0 200 OK CSeq: 6246 REGISTER Via: SIP/2.0/UDP 202.82.229.137:5060;branch=z9hG4bK76963f9a;rport From: ;tag=as2b2b9dc5 Call-ID: 3a0c18100094e487652fa8507cc27bcc@127.0.0.1 To: ;tag=240518130246 Contact: ;expires=60 Expires: 60 Content-Length: 0 --- (9 headers 0 lines) --- Scheduling destruction of SIP dialog '3a0c18100094e487652fa8507cc27bcc@127.0.0.1' in 32000 ms (Method: REGISTER) -- Registered SIP '833' at 14.155.30.67:5060 Really destroying SIP dialog '3a0c18100094e487652fa8507cc27bcc@127.0.0.1' Method: REGISTER REGISTER 11 headers, 0 lines Reliably Transmitting (NAT) to 58.254.250.26:5060: REGISTER sip:58.254.250.26 SIP/2.0 Via: SIP/2.0/UDP 202.82.229.137:5060;branch=z9hG4bK0cb253a3;rport Max-Forwards: 70 From: ;tag=as4edaee89 To: Call-ID: 3a0c18100094e487652fa8507cc27bcc@127.0.0.1 CSeq: 6247 REGISTER User-Agent: FPBX-2.8.1(1.8.21.0) Authorization: Digest username="890777", realm="VoipSwitch", algorithm=MD5, uri="sip:58.254.250.26", nonce="136905636120210914005260120120", response="417fcfabdd1deb93c2d77e801cf2f793" Expires: 120 Contact: Content-Length: 0 --- SIP/2.0 200 OK CSeq: 6247 REGISTER Via: SIP/2.0/UDP 202.82.229.137:5060;branch=z9hG4bK0cb253a3;rport From: ;tag=as4edaee89 Call-ID: 3a0c18100094e487652fa8507cc27bcc@127.0.0.1 To: ;tag=240519130231 Contact: ;expires=60 Expires: 60 Content-Length: 0 --- (9 headers 0 lines) --- Scheduling destruction of SIP dialog '3a0c18100094e487652fa8507cc27bcc@127.0.0.1' in 32000 ms (Method: REGISTER) == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 elastix*CLI> sip set debug off SIP Debugging Disabled -- Remote UNIX connection -- Remote UNIX connection disconnected elastix*CLI> Thu, 23 May 2013 13:25:04 -0500 Subject: Portech MV-372 - BUSY Permanente - by: tiziolucas http://elastix.org/index.php/en/component/kunena/116-security/122382-portech-mv-372-busy-permanente.html#122382 Buenas tardes, queria saber si alguien puso solucionar este problema el cual no encontre la solucion. Tengo un portech MV-372 el cual se registra con distintos puestos 5060-5062. Chip 1 type=peer qualify=yes port=5060 insecure=port host=192.168.0.XX context=from-trunk call-limit=1 ----------------------- chip 2 type=peer qualify=yes port=5062 insecure=port host=192.168.0.XX context=from-trunk call-limit=1 hasta aca todo perfecto, en la ruta saliente pongo que salga primero por chip 1luego por el chip 2, cada uno de ellos tiene el call-limit=1 para que solo saquen una llamada cada uno y luego salga por el segundo. El problema es cuando me llaman al chip 1... en teoria si el operador A esta hablando con la persona que llamo al chip 1 yo voy a un telefono B y quiero llamar a un celular pasa que quiere salir por el chip 1 el cual esta ocupado por la persona que llamo (llamada entrante del portech), saliendo un mensaje de BUSY y ya no puedo sacar mas llamadas a celulares hasta que se corte el llamado entrante porque no sigue la secuencia CHIP1, CHIP2 Le paso a alguien esto? se puede poner que si el chip uno estan entrando llamadas al sacar asterisk detecte que esta ocupado y salga por el chip 2 Espero me puedan ayudar, Lucas, Mar Del Plata, ARG Thu, 23 May 2013 13:13:13 -0500 Subject: DTMF issue - by: akmayuga http://elastix.org/index.php/en/component/kunena/27-miscellaneous/122365-dtmf-issue.html#122381 Hi Redfone, thank you so much for that information, I just notice this scenario: in my gsm gateway(goip4), there is a certain sim card that dosent recognized the DTMF tone, lets say its "sim-A", but when i inserted different sim card with a diferent provider (sim-B,sim-C and Sim-D) it works perfectly and it has the same configuration, do you think it has something to do with our provider on "sim card-A"? best regards Ian Thu, 23 May 2013 13:00:17 -0500 Subject: Elastix got HACKED!! - by: ukez http://elastix.org/index.php/en/component/kunena/116-security/120206-elastix-got-hacked.html?limit=10&start=10#122380 Bob wrote: Close off all access to your Elastix box. If you need SIP /RTP ports open, then try to secure them - e.g. use a router that only allows particular ports open to particular IP addresses (E.g. your voip provider)... Regards Bob Hi Bob, regarding your comment: "Close off all access to your Elastix box. If you need SIP /RTP ports open, then try to secure them - e.g. use a router that only allows particular ports open to particular IP addresses (E.g. your voip provider)" I've got a bit of a query; I'm wondering what your views are on using local elastix server with a web hosted version of Vtiger 5.4.0 Via port 5038? My router on my current elastix box only allows external access to my VOIP providers IP addresses and nothing else. If we ever do use it remotely for extensions its done via an encrypted VPN network. Providing the security, updates or security patches were up to date on a web hosted version of vTiger 5.4.0, do you personally think it would be okay to utilise the Asterisk Manager Interface (AMI) on port 5038, If I only allowed inbound access from my vtiger web hosting company on that port via my router would that be okay.. The vtiger host would also be over https.. Thanks! Thu, 23 May 2013 12:49:31 -0500 Subject: Phone AAStra 6731i SIP registration - by: rgsalaverria http://elastix.org/index.php/en/component/kunena/45-aastra/122379-phone-aastra-6731i-sip-registration.html#122379 as recorded an Aastra 6731i, worked fine but one day it desconfiguraron 10 phone and have the message NO SERVICE. I have rebooted, I changed its extension, but on the phone have the code 403 and generates no SIP registration. Elastix if recognized, but not the service is enabled. I'm new to Elastix and I have the version 2.0.0-57. Thanks Thu, 23 May 2013 11:49:17 -0500 Subject: Automating HA (heartbeat + DRBD) Install - by: Redfone http://elastix.org/index.php/en/component/kunena/26-tips-and-tricks/113919-automating-ha-heartbeat--drbd-install.html?limit=10&start=10#122378 Any luck? thanks Thu, 23 May 2013 11:32:04 -0500 Subject: Auto password generate - by: samv http://elastix.org/index.php/en/component/kunena/26-tips-and-tricks/122377-auto-password-generate.html#122377 Check this http://www.elastix.org/index.php/en/component/kunena/10-success-stories/113779-the-best-way-to-protect-your-elastix-from-anonymus.html Thu, 23 May 2013 11:20:13 -0500 Subject: Multiple google voice - by: samv http://elastix.org/index.php/en/component/kunena/10-success-stories/98971-multiple-google-voice.html?limit=10&start=120#122375 Thu, 23 May 2013 10:29:35 -0500 Subject: problemas con comando dial - by: Redfone http://elastix.org/index.php/en/component/kunena/11-ayuda/122372-problemas-con-comando-dial.html#122373 Copia y pega tu dahdi-channels.conf y chan_dahdi.conf para revisarlos de cerca. Y /etc/dahdi/system.conf por si los canales no estan definidos bien en esa config. Thu, 23 May 2013 10:19:36 -0500 Subject: Caller ID not showing when using Call pickup *8 - by: 3d3dman http://elastix.org/index.php/en/component/kunena/3-help/122371-caller-id-not-showing-when-using-call-pickup-8.html#122371 Currently using elastix 2.2.0 - we are having trouble trying to see the incoming caller ID when using call pickup (*8). When calls are picked up in a ring group the caller ID is working fine but when using call pickup the incoming caller ID does not show on the phone - all it shows on the phone is *8. Is there a way of displaying the caller ID on a call pickup? We are using a mixture of Polycomm 331 handsets and Cisco SPA525 handsets. Thu, 23 May 2013 10:00:53 -0500 Subject: Elastix with IP public adress - by: Ngouti http://elastix.org/index.php/en/component/kunena/25-newbies-corner-/122359-elastix-with-ip-public-adress.html#122370 Thank you for your response. I have seen several documents, but I don't know the document I need to download? Thu, 23 May 2013 09:50:39 -0500 Subject: Xen paravirtualization vs. Elastix 2.4.0 Stable - by: ricofranco http://elastix.org/index.php/en/component/kunena/1-installation-issues/118574-xen-paravirtualization-vs-elastix-240-stable.html#122369 Dear Sirs, If we get an Elastix Server running the Version 2.0.0 and make "yum update" the server will be upgrade to the version 2.4.0 stable ? Thu, 23 May 2013 09:47:29 -0500 Subject: interval workouts and require you to head to a lap pool to complete a set of laps. - by: joe46h60 http://elastix.org/index.php/en/component/kunena/124-simmrate/122368-interval-workouts-and-require-you-to-head-to-a-lap-pool-to-complete-a-set-of-laps.html#122368 Lyons Island,polo ralph lauren shirts (http://tourchautauqua.com/ralph.aspx), Ontario $339,or "good" cholesterol (http://www.winsonworld.net/home/space.php?uid=23127&do=blog&id=266483),357 CADLyons Island represents a rare opportunity to own a private island in picturesque Prince Edward County. Ideally located on sheltered East Lake mere minutes from Sandbanks National Park,ralph lauren outlet (http://tourchautauqua.com/ralph.aspx), this 1.5 acre island is a dream retreat,polo ralph lauren shirts (http://tourchautauqua.com/ralph.aspx), but only a little more than 2 hours from Toronto. It is developed with a 3bedroom cottage.. 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Could that glossy melamine tray transmogrify into a mirror when filled with reflective glass? Could those faux florals twist into chic arrangements worthy of Elton John dressing room? In creative hands, yes. Lap Swimming WorkoutThe book "Total Immersion" says: "One of the most important reasons for an adult to swim is to increase flexibility, because this sport promotes joint flexibility better than any other aerobic exercise." The book mentions swimmers have been known to have more powerful and lower heart beats, lower blood pressure and much better tolerance to exercise than nonswimmers. Lap workouts are simply designed to build cardio endurance and tone muscles. Lap workouts are generally longer and more paced than short,toms shoes outlet (http://senecanation.com/Media/index.aspx), interval workouts and require you to head to a lap pool to complete a set of laps. Thu, 23 May 2013 09:43:40 -0500 Subject: Asterisk Phonebook for Elastix - by: encryptedangel http://elastix.org/index.php/en/component/kunena/3-help/122297-asterisk-phonebook-for-elastix.html#122366 Bob, Thanks for your detailed reply. While I was searching about it, I came around Aptus (http://aptus.com) who claims to do the same (and other stuff too). It seems a new application (not yet launched) and the features seem good to me. Now here is my question, do you think it will work for me and for the rest of Asterisk community? and last but not least, if you or any other person is thinking to signup for the demo, please use my referral link http://lnc.hr/f1SKJ as I am looking to grab the early demo (little favor please :P ). Thu, 23 May 2013 08:23:51 -0500 Subject: add-on not working - by: discriptive http://elastix.org/index.php/en/component/kunena/3-help/122358-add-on-not-working.html#122364 yes sir i checked above two but how i can check third one . Thu, 23 May 2013 07:50:34 -0500 Subject: Documentation Communication unifiée - by: danardf http://elastix.org/index.php/en/component/kunena/82-communautee-elastix/122338-documentation-communication-unifiee.html#122357 Salut et bienvenue sur notre site Elastix. :) Toutes les doc sont sur le site Elastix Product Info - Manuals/Books (Plus haut dans le menu). Thu, 23 May 2013 05:18:48 -0500 Subject: Multiples SDA - by: danardf http://elastix.org/index.php/en/component/kunena/85-aide/122336-multiples-sda.html#122356 Salut et bienvenue sur le forum Elastix. :) Deux solutions: 1 - Ton opérateur te fournit le n° SDA et dans ce cas il faudra renseigner le n° dans le champ DID. 2 - Ton opérateur ne te fournit pas le n° SDA et dans ce cas, il faudra utiliser le context (did-from-sip "voir le forum pour plus d'info"), ce context sera à inséré dans extensions_custom.conf. Il te suffira alors de remplacer from-trunk par did-from-sip dans les paramètres de ton trunk. Juste renseigner le n° dans le champ DID soit de la route entrante, soit par le champ DID de l'extension ( qui renseignera la route de toute manière ). La solution 2, prend en compte le n° composé par l'appelant et simule un DID. Il est possible que tu ais à cocher la case priority route (un truc comme çà). Thu, 23 May 2013 05:16:23 -0500 Subject: Communicating with Skype and Twitter & Facebook - by: bghayad http://elastix.org/index.php/en/component/kunena/18-modules/122355-communicating-with-skype-and-twitter-a-facebook.html#122355 Hello; Is there a channel or module to be able to communicate with Skype? Is it free? What about twitter and facebook, any integration with it? Regards Bilal Thu, 23 May 2013 05:09:22 -0500 Subject: SMS sending and receiving and integration - by: bghayad http://elastix.org/index.php/en/component/kunena/27-miscellaneous/122354-sms-sending-and-receiving-and-integration.html#122354 Hello; I need to do the following: 1) From the vtigerCRM or OpenERP, to send SMS. How I can use Elastix in this? What is the required to achieve this integration? 2) If someone send for us an SMS message (to elastix), I need to use information in this message to do a query in the database (the database of the CRM or ERP) and then getting information from this database and sending it as SMS for the original sender. What are the main keys to achieve this? 3) If I need to send the google map for a customer as SMS, what are the method? Regards Bilal Thu, 23 May 2013 05:06:19 -0500 Subject: IAX trunk betwen two servers not working properly - by: marky http://elastix.org/index.php/en/component/kunena/3-help/122353-iax-trunk-betwen-two-servers-not-working-properly.html#122353 Hi all. I’ve 2 x Elastix servers connected remotely using an IAX2 trunk. Both are running 2.8.1. Both servers use SIP extensions and SIP trunks (the remote to a GXW4004 interfacing to PSTNs, the local to a SIP provider), and desiring only using the IAX2 trunk for inter branch connectivity. Individually, each server can make/receive calls as designed to external numbers. We've also successfully implemented time of day, IVR, etc. The problem I have is when a remote extension, say 306, calls a local extension, say 202, via the IAX2 trunk. The call is routed correctly and can be answered however the sound is not present. Actually there is a low volume static sound, in time with speaking. Testing has been carried out softphone > snom320 and softphone > softphone. This also happens in the reverse direction. Peer Details ---------------- username=dubai type=peer secret=xxxxxx requirecalltoken=no qualify=yes host=80.xxx.xxx.xxx context=from-internal trunk=yes disallow=all allow=alaw&ulaw&gsm User Details --------------- type=user secret=xxxxxxx host=80.xxx.xxx.xxx context=from-internal disallow=all allow=alaw&ulaw&gsm We have port forwarding rules in each location for: 4569 UDP – IAX2 5036 UDP – IAX1 (this is not is use) 5004:5082 UDP and SIP 10000 – 20000 UDP (There are others for SSH, Webmin etc I’ve not listed) An output of the local server: -- Executing [s@macro-record-enable:8] GotoIf("IAX2/australia-14053", "1?continue") in new stack -- Goto (macro-record-enable,s,13) -- Executing [s@macro-record-enable:13] Set("IAX2/australia-14053", "ITER=2") in new stack -- Executing [s@macro-record-enable:14] GotoIf("IAX2/australia-14053", "1?begin") in new stack -- Goto (macro-record-enable,s,8) -- Executing [s@macro-record-enable:8] GotoIf("IAX2/australia-14053", "1?continue") in new stack -- Goto (macro-record-enable,s,13) -- Executing [s@macro-record-enable:13] Set("IAX2/australia-14053", "ITER=3") in new stack -- Executing [s@macro-record-enable:14] GotoIf("IAX2/australia-14053", "0?begin") in new stack -- Executing [s@macro-record-enable:15] GotoIf("IAX2/australia-14053", "0?IN") in new stack -- Executing [s@macro-record-enable:16] ExecIf("IAX2/australia-14053", "1?MacroExit()") in new stack -- Executing [202@from-internal:23] Set("IAX2/australia-14053", "RingGroupMethod=ringallv2-prim") in new stack -- Executing [202@from-internal:24] Set("IAX2/australia-14053", "_FMGRP=202") in new stack -- Executing [202@from-internal:25] GotoIf("IAX2/australia-14053", "1?doconfirm") in new stack -- Goto (from-internal,202,28) -- Executing [202@from-internal:28] Macro("IAX2/australia-14053", "dial-confirm,25,tr,0404xxxxxx#,202") in new stack -- Executing [s@macro-dial-confirm:1] Set("IAX2/australia-14053", "DB(RG/202/IAX2/australia-14053)=RINGING") in new stack -- Executing [s@macro-dial-confirm:2] Set("IAX2/australia-14053", "__UNIQCHAN=IAX2/australia-14053") in new stack -- Executing [s@macro-dial-confirm:3] Set("IAX2/australia-14053", "USE_CONFIRMATION=TRUE") in new stack -- Executing [s@macro-dial-confirm:4] Set("IAX2/australia-14053", "RINGGROUP_INDEX=202") in new stack -- Executing [s@macro-dial-confirm:5] Set("IAX2/australia-14053", "ARG4=") in new stack -- Executing [s@macro-dial-confirm:6] Macro("IAX2/australia-14053", "dial,25,tr,0404xxxxxx#") in new stack -- Executing [s@macro-dial:1] GotoIf("IAX2/australia-14053", "1?dial") in new stack -- Goto (macro-dial,s,3) -- Executing [s@macro-dial:3] AGI("IAX2/australia-14053", "dialparties.agi") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi dialparties.agi: Starting New Dialparties.agi dialparties.agi: Caller ID name is 'salesLeslie (Support)' number is '306' > dialparties.agi: USE_CONFIRMATION: 'TRUE' > dialparties.agi: RINGGROUP_INDEX: '202' dialparties.agi: Methodology of ring is 'ringallv2-prim' -- dialparties.agi: Added extension 0404xxxxxx# to extension map > dialparties.agi: got fmgrp_prering: 5, fmgrp_grptime: 20 > dialparties.agi: fmgrp_totalprering: 25 > dialparties.agi: extension not in group list, ringing only during prering time > dialparties.agi: ringallv2 ring times: REALPRERING: 5, PRERING: 5 -- dialparties.agi: Extension 202 cf is disabled -- dialparties.agi: Extension 0404xxxxxx# cf is disabled -- dialparties.agi: Extension 202 do not disturb is disabled > dialparties.agi: extnum 202 has: cw: 1; hascfb: 0 [] hascfu: 0 [] dialparties.agi: EXTENSION_STATE: 0 (NOT_INUSE) dialparties.agi: Extension 202 has ExtensionState: 0 -- dialparties.agi: dbset CALLTRACE/202 to 306 > dialparties.agi: extnum 0404xxxxxx# has: cw: 0; hascfb: 0 [] hascfu: 0 [] > dialparties.agi: Built External dialstring component for 0404xxxxxx: Local/RG-202-0404xxxxxx#@from-internal -- dialparties.agi: Filtered ARG3: 202-0404xxxxxx > dialparties.agi: NODEST: 202 adding M(auto-blkvm) to dialopts: trM(auto-blkvm) -- AGI Script dialparties.agi completed, returning 0 -- Executing [s@macro-dial:7] Dial("IAX2/australia-14053", "Local/FMPR-202@from-internal&Local/FMGL-0404xxxxxx#@from-internal,25,trM(auto-blkvm)") in new stack -- Called Local/FMPR-202@from-internal -- Called Local/FMGL-0404xxxxxx#@from-internal -- Executing [FMPR-202@from-internal:1] NoOp("Local/FMPR-202@from-internal-00000002;2", "In FMPR 202 with 202") in new stack -- Executing [FMPR-202@from-internal:2] Set("Local/FMPR-202@from-internal-00000002;2", "RingGroupMethod=") in new stack -- Executing [FMPR-202@from-internal:3] Set("Local/FMPR-202@from-internal-00000002;2", "USE_CONFIRMATION=") in new stack -- Executing [FMPR-202@from-internal:4] Set("Local/FMPR-202@from-internal-00000002;2", "RINGGROUP_INDEX=") in new stack -- Executing [FMPR-202@from-internal:5] Macro("Local/FMPR-202@from-internal-00000002;2", "simple-dial,202,5") in new stack -- Executing [s@macro-simple-dial:1] Set("Local/FMPR-202@from-internal-00000002;2", "__EXTTOCALL=202") in new stack -- Executing [s@macro-simple-dial:2] Set("Local/FMPR-202@from-internal-00000002;2", "RT=5") in new stack -- Executing [s@macro-simple-dial:3] Set("Local/FMPR-202@from-internal-00000002;2", "CFUEXT=") in new stack -- Executing [s@macro-simple-dial:4] Set("Local/FMPR-202@from-internal-00000002;2", "CFBEXT=") in new stack -- Executing [s@macro-simple-dial:5] Set("Local/FMPR-202@from-internal-00000002;2", "CWI_TMP=") in new stack -- Executing [s@macro-simple-dial:6] Macro("Local/FMPR-202@from-internal-00000002;2", "dial-one,5,tr,202") in new stack -- Executing [s@macro-dial-one:1] Set("Local/FMPR-202@from-internal-00000002;2", "DEXTEN=202") in new stack -- Executing [s@macro-dial-one:2] Set("Local/FMPR-202@from-internal-00000002;2", "DIALSTATUS_CW=") in new stack -- Executing [s@macro-dial-one:3] GosubIf("Local/FMPR-202@from-internal-00000002;2", "0?screen,1") in new stack -- Executing [s@macro-dial-one:4] GosubIf("Local/FMPR-202@from-internal-00000002;2", "0?cf,1") in new stack -- Executing [s@macro-dial-one:5] GotoIf("Local/FMPR-202@from-internal-00000002;2", "1?skip1") in new stack -- Goto (macro-dial-one,s,8) -- Executing [s@macro-dial-one:8] GotoIf("Local/FMPR-202@from-internal-00000002;2", "0?nodial") in new stack -- Executing [s@macro-dial-one:9] GotoIf("Local/FMPR-202@from-internal-00000002;2", "0?continue") in new stack -- Executing [s@macro-dial-one:10] Set("Local/FMPR-202@from-internal-00000002;2", "EXTHASCW=ENABLED") in new stack -- Executing [s@macro-dial-one:11] GotoIf("Local/FMPR-202@from-internal-00000002;2", "0?next1:cwinusebusy") in new stack -- Goto (macro-dial-one,s,23) -- Executing [s@macro-dial-one:23] GotoIf("Local/FMPR-202@from-internal-00000002;2", "1?next3:continue") in new stack -- Goto (macro-dial-one,s,24) -- Executing [s@macro-dial-one:24] ExecIf("Local/FMPR-202@from-internal-00000002;2", "0?Set(DIALSTATUS_CW=BUSY)") in new stack -- Executing [s@macro-dial-one:25] GotoIf("Local/FMPR-202@from-internal-00000002;2", "0?nodial") in new stack -- Executing [s@macro-dial-one:26] GosubIf("Local/FMPR-202@from-internal-00000002;2", "1?dstring,1:dlocal,1") in new stack -- Executing [dstring@macro-dial-one:1] Set("Local/FMPR-202@from-internal-00000002;2", "DSTRING=") in new stack -- Executing [dstring@macro-dial-one:2] Set("Local/FMPR-202@from-internal-00000002;2", "DEVICES=202") in new stack -- Executing [dstring@macro-dial-one:3] ExecIf("Local/FMPR-202@from-internal-00000002;2", "0?Return()") in new stack -- Executing [dstring@macro-dial-one:4] ExecIf("Local/FMPR-202@from-internal-00000002;2", "0?Set(DEVICES=02)") in new stack -- Executing [dstring@macro-dial-one:5] Set("Local/FMPR-202@from-internal-00000002;2", "LOOPCNT=1") in new stack -- Executing [dstring@macro-dial-one:6] Set("Local/FMPR-202@from-internal-00000002;2", "ITER=1") in new stack -- Executing [dstring@macro-dial-one:7] Set("Local/FMPR-202@from-internal-00000002;2", "THISDIAL=SIP/202") in new stack -- Executing [dstring@macro-dial-one:8] GosubIf("Local/FMPR-202@from-internal-00000002;2", "1?zap2dahdi,1") in new stack -- Executing [zap2dahdi@macro-dial-one:1] ExecIf("Local/FMPR-202@from-internal-00000002;2", "0?Return()") in new stack -- Executing [zap2dahdi@macro-dial-one:2] Set("Local/FMPR-202@from-internal-00000002;2", "NEWDIAL=") in new stack -- Executing [zap2dahdi@macro-dial-one:3] Set("Local/FMPR-202@from-internal-00000002;2", "LOOPCNT2=1") in new stack -- Executing [zap2dahdi@macro-dial-one:4] Set("Local/FMPR-202@from-internal-00000002;2", "ITER2=1") in new stack -- Executing [zap2dahdi@macro-dial-one:5] Set("Local/FMPR-202@from-internal-00000002;2", "THISPART2=SIP/202") in new stack -- Executing [zap2dahdi@macro-dial-one:6] ExecIf("Local/FMPR-202@from-internal-00000002;2", "0?Set(THISPART2=DAHDI/202)") in new stack -- Executing [zap2dahdi@macro-dial-one:7] Set("Local/FMPR-202@from-internal-00000002;2", "NEWDIAL=SIP/202&") in new stack -- Executing [zap2dahdi@macro-dial-one:8] Set("Local/FMPR-202@from-internal-00000002;2", "ITER2=2") in new stack -- Executing [zap2dahdi@macro-dial-one:9] GotoIf("Local/FMPR-202@from-internal-00000002;2", "0?begin2") in new stack -- Executing [zap2dahdi@macro-dial-one:10] Set("Local/FMPR-202@from-internal-00000002;2", "THISDIAL=SIP/202") in new stack -- Executing [zap2dahdi@macro-dial-one:11] Return("Local/FMPR-202@from-internal-00000002;2", "") in new stack -- Executing [dstring@macro-dial-one:9] Set("Local/FMPR-202@from-internal-00000002;2", "DSTRING=SIP/202&") in new stack -- Executing [dstring@macro-dial-one:10] Set("Local/FMPR-202@from-internal-00000002;2", "ITER=2") in new stack -- Executing [dstring@macro-dial-one:11] GotoIf("Local/FMPR-202@from-internal-00000002;2", "0?begin") in new stack -- Executing [dstring@macro-dial-one:12] Set("Local/FMPR-202@from-internal-00000002;2", "DSTRING=SIP/202") in new stack -- Executing [dstring@macro-dial-one:13] Return("Local/FMPR-202@from-internal-00000002;2", "") in new stack -- Executing [s@macro-dial-one:27] GotoIf("Local/FMPR-202@from-internal-00000002;2", "0?nodial") in new stack -- Executing [s@macro-dial-one:28] GotoIf("Local/FMPR-202@from-internal-00000002;2", "0?skiptrace") in new stack -- Executing [s@macro-dial-one:29] GosubIf("Local/FMPR-202@from-internal-00000002;2", "1?ctset,1:ctclear,1") in new stack -- Executing [ctset@macro-dial-one:1] Set("Local/FMPR-202@from-internal-00000002;2", "DB(CALLTRACE/202)=306") in new stack -- Executing [ctset@macro-dial-one:2] Return("Local/FMPR-202@from-internal-00000002;2", "") in new stack -- Executing [s@macro-dial-one:30] Set("Local/FMPR-202@from-internal-00000002;2", "D_OPTIONS=trM(auto-blkvm)") in new stack -- Executing [s@macro-dial-one:31] ExecIf("Local/FMPR-202@from-internal-00000002;2", "0?SIPAddHeader(Alert-Info: )") in new stack -- Executing [s@macro-dial-one:32] ExecIf("Local/FMPR-202@from-internal-00000002;2", "0?SIPAddHeader()") in new stack -- Executing [s@macro-dial-one:33] ExecIf("Local/FMPR-202@from-internal-00000002;2", "0?Set(CHANNEL(musicclass)=)") in new stack -- Executing [s@macro-dial-one:34] GosubIf("Local/FMPR-202@from-internal-00000002;2", "0?qwait,1") in new stack -- Executing [s@macro-dial-one:35] Set("Local/FMPR-202@from-internal-00000002;2", "__CWIGNORE=") in new stack -- Executing [s@macro-dial-one:36] Set("Local/FMPR-202@from-internal-00000002;2", "__KEEPCID=TRUE") in new stack -- Executing [s@macro-dial-one:37] Dial("Local/FMPR-202@from-internal-00000002;2", "SIP/202,5,trM(auto-blkvm)") in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [FMGL-0404xxxxxx#@from-internal:1] NoOp("Local/FMGL-0404xxxxxx#@from-internal-00000003;2", "In FMGL 202 with 0404xxxxxx#") in new stack -- Called SIP/202 -- Local/FMPR-202@from-internal-00000002;1 is ringing -- Local/FMPR-202@from-internal-00000002;1 connected line has changed. Saving it until answer for IAX2/australia-14053 -- Executing [FMGL-0404xxxxxx#@from-internal:2] GotoIf("Local/FMGL-0404xxxxxx#@from-internal-00000003;2", "0?dodnd") in new stack -- Executing [FMGL-0404xxxxxx#@from-internal:3] Wait("Local/FMGL-0404xxxxxx#@from-internal-00000003;2", "1") in new stack -- Local/FMPR-202@from-internal-00000002;1 connected line has changed. Saving it until answer for IAX2/australia-14053 -- SIP/202-0000014c is ringing -- Local/FMPR-202@from-internal-00000002;1 is ringing -- Executing [FMGL-0404xxxxxx#@from-internal:4] GotoIf("Local/FMGL-0404xxxxxx#@from-internal-00000003;2", "0?dodnd") in new stack -- Executing [FMGL-0404xxxxxx#@from-internal:5] Wait("Local/FMGL-0404xxxxxx#@from-internal-00000003;2", "1") in new stack -- Executing [FMGL-0404xxxxxx#@from-internal:6] GotoIf("Local/FMGL-0404xxxxxx#@from-internal-00000003;2", "0?dodnd") in new stack -- Executing [FMGL-0404xxxxxx#@from-internal:7] Wait("Local/FMGL-0404xxxxxx#@from-internal-00000003;2", "3") in new stack -- Local/FMPR-202@from-internal-00000002;1 connected line has changed. Saving it until answer for IAX2/australia-14053 -- SIP/202-0000014c answered Local/FMPR-202@from-internal-00000002;2 -- Executing [s@macro-auto-blkvm:1] Set("SIP/202-0000014c", "__MACRO_RESULT=") in new stack -- Executing [s@macro-auto-blkvm:2] NoOp("SIP/202-0000014c", "Deleting: BLKVM/202/IAX2/australia-14053 TRUE") in new stack -- Local/FMPR-202@from-internal-00000002;1 answered IAX2/australia-14053 -- Executing [s@macro-auto-blkvm:1] Set("Local/FMPR-202@from-internal-00000002;1", "__MACRO_RESULT=") in new stack -- Executing [s@macro-auto-blkvm:2] NoOp("Local/FMPR-202@from-internal-00000002;1", "Deleting: BLKVM/202/IAX2/australia-14053 ") in new stack == Spawn extension (from-internal, FMGL-0404xxxxxx#, 7) exited non-zero on 'Local/FMGL-0404xxxxxx#@from-internal-00000003;2' -- Executing [h@from-internal:1] Macro("Local/FMGL-0404xxxxxx#@from-internal-00000003;2", "hangupcall") in new stack -- Executing [s@macro-hangupcall:1] GotoIf("Local/FMGL-0404xxxxxx#@from-internal-00000003;2", "1?endmixmoncheck") in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] NoOp("Local/FMGL-0404xxxxxx#@from-internal-00000003;2", "End of MIXMON check") in new stack -- Executing [s@macro-hangupcall:10] GotoIf("Local/FMGL-0404xxxxxx#@from-internal-00000003;2", "1?nomeetmemon") in new stack -- Goto (macro-hangupcall,s,28) -- Executing [s@macro-hangupcall:28] NoOp("Local/FMGL-0404xxxxxx#@from-internal-00000003;2", "End of MEETME check") in new stack -- Executing [s@macro-hangupcall:29] GotoIf("Local/FMGL-0404xxxxxx#@from-internal-00000003;2", "1?noautomon") in new stack -- Goto (macro-hangupcall,s,34) -- Executing [s@macro-hangupcall:34] NoOp("Local/FMGL-0404xxxxxx#@from-internal-00000003;2", "TOUCH_MONITOR_OUTPUT=") in new stack -- Executing [s@macro-hangupcall:35] GotoIf("Local/FMGL-0404xxxxxx#@from-internal-00000003;2", "1?noautomon2") in new stack -- Goto (macro-hangupcall,s,41) -- Executing [s@macro-hangupcall:41] NoOp("Local/FMGL-0404xxxxxx#@from-internal-00000003;2", "MONITOR_FILENAME=") in new stack -- Executing [s@macro-hangupcall:42] GotoIf("Local/FMGL-0404xxxxxx#@from-internal-00000003;2", "1?skiprg") in new stack -- Goto (macro-hangupcall,s,45) -- Executing [s@macro-hangupcall:45] GotoIf("Local/FMGL-0404xxxxxx#@from-internal-00000003;2", "1?skipblkvm") in new stack -- Goto (macro-hangupcall,s,48) -- Executing [s@macro-hangupcall:48] GotoIf("Local/FMGL-0404xxxxxx#@from-internal-00000003;2", "1?theend") in new stack -- Goto (macro-hangupcall,s,50) -- Executing [s@macro-hangupcall:50] AGI("Local/FMGL-0404xxxxxx#@from-internal-00000003;2", "hangup.agi") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/hangup.agi -- AGI Script hangup.agi completed, returning 0 -- Executing [s@macro-hangupcall:51] Hangup("Local/FMGL-0404xxxxxx#@from-internal-00000003;2", "") in new stack == Spawn extension (macro-hangupcall, s, 51) exited non-zero on 'Local/FMGL-0404xxxxxx#@from-internal-00000003;2' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'Local/FMGL-0404xxxxxx#@from-internal-00000003;2' -- Executing [h@macro-dial-one:1] Macro("Local/FMPR-202@from-internal-00000002;2", "hangupcall,") in new stack -- Executing [s@macro-hangupcall:1] GotoIf("Local/FMPR-202@from-internal-00000002;2", "1?endmixmoncheck") in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] NoOp("Local/FMPR-202@from-internal-00000002;2", "End of MIXMON check") in new stack -- Executing [s@macro-hangupcall:10] GotoIf("Local/FMPR-202@from-internal-00000002;2", "1?nomeetmemon") in new stack -- Goto (macro-hangupcall,s,28) -- Executing [s@macro-hangupcall:28] NoOp("Local/FMPR-202@from-internal-00000002;2", "End of MEETME check") in new stack -- Executing [s@macro-hangupcall:29] GotoIf("Local/FMPR-202@from-internal-00000002;2", "1?noautomon") in new stack -- Goto (macro-hangupcall,s,34) -- Executing [s@macro-hangupcall:34] NoOp("Local/FMPR-202@from-internal-00000002;2", "TOUCH_MONITOR_OUTPUT=") in new stack -- Executing [s@macro-hangupcall:35] GotoIf("Local/FMPR-202@from-internal-00000002;2", "1?noautomon2") in new stack -- Goto (macro-hangupcall,s,41) -- Executing [s@macro-hangupcall:41] NoOp("Local/FMPR-202@from-internal-00000002;2", "MONITOR_FILENAME=") in new stack -- Executing [s@macro-hangupcall:42] GotoIf("Local/FMPR-202@from-internal-00000002;2", "1?skiprg") in new stack -- Goto (macro-hangupcall,s,45) -- Executing [s@macro-hangupcall:45] GotoIf("Local/FMPR-202@from-internal-00000002;2", "1?skipblkvm") in new stack -- Goto (macro-hangupcall,s,48) -- Executing [s@macro-hangupcall:48] GotoIf("Local/FMPR-202@from-internal-00000002;2", "1?theend") in new stack -- Goto (macro-hangupcall,s,50) -- Executing [s@macro-hangupcall:50] AGI("Local/FMPR-202@from-internal-00000002;2", "hangup.agi") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/hangup.agi -- AGI Script hangup.agi completed, returning 0 -- Executing [s@macro-hangupcall:51] Hangup("Local/FMPR-202@from-internal-00000002;2", "") in new stack == Spawn extension (macro-hangupcall, s, 51) exited non-zero on 'Local/FMPR-202@from-internal-00000002;2' in macro 'hangupcall' == Spawn extension (macro-dial-one, h, 1) exited non-zero on 'Local/FMPR-202@from-internal-00000002;2' == Spawn extension (macro-dial-one, s, 37) exited non-zero on 'Local/FMPR-202@from-internal-00000002;2' in macro 'dial-one' == Spawn extension (macro-simple-dial, s, 6) exited non-zero on 'Local/FMPR-202@from-internal-00000002;2' in macro 'simple-dial' == Spawn extension (from-internal, FMPR-202, 5) exited non-zero on 'Local/FMPR-202@from-internal-00000002;2' -- Executing [h@macro-dial:1] Macro("IAX2/australia-14053", "hangupcall") in new stack -- Executing [s@macro-hangupcall:1] GotoIf("IAX2/australia-14053", "1?endmixmoncheck") in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] NoOp("IAX2/australia-14053", "End of MIXMON check") in new stack -- Executing [s@macro-hangupcall:10] GotoIf("IAX2/australia-14053", "1?nomeetmemon") in new stack -- Goto (macro-hangupcall,s,28) -- Executing [s@macro-hangupcall:28] NoOp("IAX2/australia-14053", "End of MEETME check") in new stack -- Executing [s@macro-hangupcall:29] GotoIf("IAX2/australia-14053", "1?noautomon") in new stack -- Goto (macro-hangupcall,s,34) -- Executing [s@macro-hangupcall:34] NoOp("IAX2/australia-14053", "TOUCH_MONITOR_OUTPUT=") in new stack -- Executing [s@macro-hangupcall:35] GotoIf("IAX2/australia-14053", "1?noautomon2") in new stack -- Goto (macro-hangupcall,s,41) -- Executing [s@macro-hangupcall:41] NoOp("IAX2/australia-14053", "MONITOR_FILENAME=") in new stack -- Executing [s@macro-hangupcall:42] GotoIf("IAX2/australia-14053", "0?skiprg") in new stack -- Executing [s@macro-hangupcall:43] NoOp("IAX2/australia-14053", "Cleaning Up Confirmation Flag: RG/202/IAX2/australia-14053") in new stack -- Executing [s@macro-hangupcall:44] NoOp("IAX2/australia-14053", "Deleting: RG/202/IAX2/australia-14053 RINGING") in new stack -- Executing [s@macro-hangupcall:45] GotoIf("IAX2/australia-14053", "0?skipblkvm") in new stack -- Executing [s@macro-hangupcall:46] NoOp("IAX2/australia-14053", "Cleaning Up Block VM Flag: BLKVM/202/IAX2/australia-14053") in new stack -- Executing [s@macro-hangupcall:47] NoOp("IAX2/australia-14053", "Deleting: BLKVM/202/IAX2/australia-14053 ") in new stack -- Executing [s@macro-hangupcall:48] GotoIf("IAX2/australia-14053", "0?theend") in new stack -- Executing [s@macro-hangupcall:49] NoOp("IAX2/australia-14053", "Deleting: FM/DND/202/IAX2/australia-14053 ") in new stack -- Executing [s@macro-hangupcall:50] AGI("IAX2/australia-14053", "hangup.agi") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/hangup.agi -- AGI Script hangup.agi completed, returning 0 -- Executing [s@macro-hangupcall:51] Hangup("IAX2/australia-14053", "") in new stack == Spawn extension (macro-hangupcall, s, 51) exited non-zero on 'IAX2/australia-14053' in macro 'hangupcall' == Spawn extension (macro-dial, h, 1) exited non-zero on 'IAX2/australia-14053' == Spawn extension (macro-dial, s, 7) exited non-zero on 'IAX2/australia-14053' in macro 'dial' == Spawn extension (macro-dial-confirm, s, 6) exited non-zero on 'IAX2/australia-14053' in macro 'dial-confirm' == Spawn extension (from-internal, 202, 28) exited non-zero on 'IAX2/australia-14053' -- Hungup 'IAX2/australia-14053' atom-au*CLI> Any pointers would be greatly appreciated. Thanks Thu, 23 May 2013 04:29:43 -0500 Subject: uElastix, Web-interface problem. - by: Aiiar http://elastix.org/index.php/en/component/kunena/1-installation-issues/122142-uelastix-web-interface-problem.html#122352 bajji wrote: hi there the problème is a problème with mysql data base if you can login in with putty try to start mysql data base elese the is no solution use the line : /etc/init.d/mysqld start Thank you Bajji for your reply. I got this after writing your line in putty -bash: /etc/init.d/mysqld: No existe el fichero o el directorio any help !? Thu, 23 May 2013 03:51:11 -0500 Subject: MOH - by: Itsm http://elastix.org/index.php/en/component/kunena/3-help/122150-moh.html#122351 Oh, there it is: [code] Kernel Linux(x86_64)-2.6.18-348.4.1.el5.centos.plus Elastix elastix-2.4.0-1 elastix-portknock-0.0.1-0 elastix-agenda-2.4.0-1 elastix-email_admin-2.4.0-1 elastix-addons-2.4.0-1 elastix-fax-2.4.0-1 elastix-voipprovider_addon-2.3.0-1 elastix-firstboot-2.4.0-1 elastix-framework-2.4.0-1 elastix-reports-2.4.0-1 elastix-callcenter-2.2.0-0 elastix-my_extension-2.4.0-1 elastix-extras-2.4.0-1 elastix-vtigercrm-5.2.1-7 elastix-system-2.4.0-1 elastix-security-2.4.0-1 elastix-im-2.4.0-1 elastix-a2billing-1.9.4-5 elastix-asterisk-sounds-1.2.3-1 elastix-pbx-2.4.0-1 RounCubeMail RoundCubeMail-0.3.1-12 Mail postfix-2.3.3-6.0.1.el5_9 cyrus-imapd-2.3.7-12.el5_7.2 IM openfire-3.7.1-2 FreePBX freePBX-2.8.1-16 Asterisk asterisk-11.3.0-1 asterisk-perl-0.10-2 asterisk-addons-11.3.0-1 FAX hylafax-4.3.10-2rhel5 iaxmodem-1.2.0-2 DRIVERS dahdi-2.6.1-5 rhino-0.99.6-0.b2 wanpipe-util-7.0.0-0 [/code] Thu, 23 May 2013 03:14:46 -0500 Subject: Recording name /nom des enregistrement - by: bajji http://elastix.org/index.php/en/component/kunena/1-installation-issues/122350-recording-name-nom-des-enregistrement.html#122350 HI how can i edit the recording name default name : OUT9000-20130522-194203-1369244523.6258 and i want to edit edit and make the caller number Bonjour : SVP je veux modifier les nom des neregistrements par defaut ils senregistre sous cet forme :OUT9000-20130522-194203-1369244523.6258 et je désire metre le numéro appelé CDT Thu, 23 May 2013 02:56:41 -0500 Subject: Please help - by: raj2013 http://elastix.org/index.php/en/component/kunena/3-help/122197-please-help.html?limit=10&start=10#122347 Please Explain why this error occurs Executing [s@macro-hangupcall:6] NoOp("SIP/482-00002e1b", "SYSTEMSTATUS = APPERROR") in new stack. In Elastix --->Security-->Advanced Settings -->Enable anonymous SIP calls:ON/OFF In this which need to enable - ON or OFF Can We Connect 3 extension(account) on Granstream Phone.Will if create any traffic? Can we use same outgoing caller ID for all extension? Thank you Thu, 23 May 2013 00:59:16 -0500 Subject: Elastix 3FXO 1FXS Set-up - by: michiXile http://elastix.org/index.php/en/component/kunena/25-newbies-corner-/122345-elastix-3fxo-1fxs-set-up.html#122345 I have a current set-up for my Elastix set-up I have an AEX 410 with noise cancelling hardware. 3 telephone line trunk i have configured 1 telephone trunk that goes to IVR. my question is that how do i add the remaining two? I am confused with Channel, Trunk Zap and DID/CID work around. Thanks in advance. Thu, 23 May 2013 00:41:31 -0500 Subject: Custom Context. to disble outbound route - by: fuyun1965 http://elastix.org/index.php/en/component/kunena/18-modules/122344-custom-context-to-disble-outbound-route.html#122344 Hi, I am using Elastix 2.0 with custom context module installed. I can set up different context, for contolling ext.'s outbound route. Now I have problem. I have an inbound route, forward to DISA. How can I set up a diffent DISA with a pre-defined Context controlling outbound route (like disable all calls with prefix 4)? So that I can control the inbound call to continue dail outbound route. Thanks! Thu, 23 May 2013 00:39:22 -0500 Subject: [AYUDA] Varios FOP. - by: Curioso http://elastix.org/index.php/en/component/kunena/60-flash-operator-panel/83837-ayuda-varios-fop.html#122343 Yo quisiera hacer lo mismo que dice el señor afos0110 para que se vean todas la extensiones en el FOP yo uso elastix 2.2 Wed, 22 May 2013 23:33:29 -0500 Subject: Release FreePBX Ebook in Chinese! - by: james.zhu http://elastix.org/index.php/en/component/kunena/27-miscellaneous/122342-release-freepbx-ebook-in-chinese.html#122342 hello, all members: i released FreePBX Ebook in chinese. please refer this: http://www.hiastar.com/index.php/2011-04-19-18-57-12/2011-04-19-18-57-49 Wed, 22 May 2013 22:54:26 -0500 Subject: Problem in remote extension with SIP trunk - by: voip_starter0211 http://elastix.org/index.php/en/component/kunena/3-help/122241-problem-in-remote-extension-with-sip-trunk.html#122341 can any one advise on this case ? Wed, 22 May 2013 22:08:47 -0500 Subject: connecting wcdma gateway - by: Artush http://elastix.org/index.php/en/component/kunena/25-newbies-corner-/122339-connecting-wcdma-gateway.html#122339 Hi friends, I have got an wcdma (3G) gateway from china and want to connect it with elastix for call termination and billing! Here is Ejoin GOIP User Manual! Any ideas? https://docs.google.com/file/d/0B07ySZ54CQneaDZQbHk4b2hERDg/edit?usp=sharing Wed, 22 May 2013 19:49:57 -0500 Subject: No ingresan llamadas por PSTN SOLUCIONADO - by: jfernandez84 http://elastix.org/index.php/en/component/kunena/50-novatos/122112-no-ingresan-llamadas-por-pstn-solucionado.html#122337 Gracias amigo. No se que pasó, pero el problema se solucionó reiniciando la máquina :) Sigo leyendo y aprendiendo mas y cada día mas a gusto con el sistema. Wed, 22 May 2013 19:27:49 -0500 Subject: IP address locations in Elastix... - by: Bob http://elastix.org/index.php/en/component/kunena/1-installation-issues/122332-ip-address-locations-in-elastix.html#122335 mcavin, There is no special place as far as I know that Elastix keeps the IP address for the GUI. The IP address, gateways, DNS is all still standard CENTOS. So whatever CENTOS 5 guides there are for changing details is still valid. I have changed addresses hundreds of times (especially when doing builds) with no issues no loss of GUI. You may have to reboot the box after the change but that is about it. If your DNS is not set right, it can cause your browser to hang at times, but you should still see the GUI. Regards Bob Wed, 22 May 2013 17:08:58 -0500 Subject: Extensions unregister randomly - by: Bob http://elastix.org/index.php/en/component/kunena/3-help/122172-extensions-unregister-randomly.html#122333 diaznoel, Great to hear you found the cause.... Thanks for posting back as it will help others.. Regards Bob Wed, 22 May 2013 16:49:36 -0500 Subject: Problema con Voicemail - by: afos0110 http://elastix.org/index.php/en/component/kunena/11-ayuda/27384-problema-con-voicemail.html?limit=10&start=20#122331 Por favor necesito que me colaboren, no se porque cuando configure una extensión y acitve el voice mail, cuando realizo una llamada y está esta disponible solo timbra una ves y de una entra al buzon de mensaje he buscado en los foros de elastix para este problema y no he encontrado información. Este es el CLIC -- Executing [104@from-internal:1] Macro("SIP/1002-00000017", "exten-vm,104,104") in new stack -- Executing [s@macro-exten-vm:1] Macro("SIP/1002-00000017", "user-callerid,") in new stack -- Executing [s@macro-user-callerid:1] Set("SIP/1002-00000017", "AMPUSER=1002") in new stack -- Executing [s@macro-user-callerid:2] GotoIf("SIP/1002-00000017", "0?report") in new stack -- Executing [s@macro-user-callerid:3] ExecIf("SIP/1002-00000017", "1?Set(REALCALLERIDNUM=1002)") in new stack -- Executing [s@macro-user-callerid:4] Set("SIP/1002-00000017", "AMPUSER=1002") in new stack -- Executing [s@macro-user-callerid:5] Set("SIP/1002-00000017", "AMPUSERCIDNAME=1002") in new stack -- Executing [s@macro-user-callerid:6] GotoIf("SIP/1002-00000017", "0?report") in new stack -- Executing [s@macro-user-callerid:7] Set("SIP/1002-00000017", "AMPUSERCID=1002") in new stack -- Executing [s@macro-user-callerid:8] Set("SIP/1002-00000017", "CALLERID(all)="1002" ") in new stack -- Executing [s@macro-user-callerid:9] ExecIf("SIP/1002-00000017", "1?Set(CHANNEL(language)=es)") in new stack -- Executing [s@macro-user-callerid:10] GotoIf("SIP/1002-00000017", "0?continue") in new stack -- Executing [s@macro-user-callerid:11] Set("SIP/1002-00000017", "__TTL=64") in new stack -- Executing [s@macro-user-callerid:12] GotoIf("SIP/1002-00000017", "1?continue") in new stack -- Goto (macro-user-callerid,s,19) -- Executing [s@macro-user-callerid:19] Set("SIP/1002-00000017", "CALLERID(number)=1002") in new stack -- Executing [s@macro-user-callerid:20] Set("SIP/1002-00000017", "CALLERID(name)=1002") in new stack -- Executing [s@macro-user-callerid:21] NoOp("SIP/1002-00000017", "Using CallerID "1002" ") in new stack -- Executing [s@macro-exten-vm:2] Set("SIP/1002-00000017", "RingGroupMethod=none") in new stack -- Executing [s@macro-exten-vm:3] Set("SIP/1002-00000017", "VMBOX=104") in new stack -- Executing [s@macro-exten-vm:4] Set("SIP/1002-00000017", "__EXTTOCALL=104") in new stack -- Executing [s@macro-exten-vm:5] Set("SIP/1002-00000017", "CFUEXT=") in new stack -- Executing [s@macro-exten-vm:6] Set("SIP/1002-00000017", "CFBEXT=") in new stack -- Executing [s@macro-exten-vm:7] Set("SIP/1002-00000017", "RT=15") in new stack -- Executing [s@macro-exten-vm:8] Macro("SIP/1002-00000017", "record-enable,104,IN") in new stack -- Executing [s@macro-record-enable:1] GotoIf("SIP/1002-00000017", "1?check") in new stack -- Goto (macro-record-enable,s,4) -- Executing [s@macro-record-enable:4] ExecIf("SIP/1002-00000017", "0?MacroExit()") in new stack -- Executing [s@macro-record-enable:5] GotoIf("SIP/1002-00000017", "0?Group:OUT") in new stack -- Goto (macro-record-enable,s,15) -- Executing [s@macro-record-enable:15] GotoIf("SIP/1002-00000017", "1?IN") in new stack -- Goto (macro-record-enable,s,20) -- Executing [s@macro-record-enable:20] ExecIf("SIP/1002-00000017", "1?MacroExit()") in new stack -- Executing [s@macro-exten-vm:9] Macro("SIP/1002-00000017", "dial-one,15,tr,104") in new stack -- Executing [s@macro-dial-one:1] Set("SIP/1002-00000017", "DEXTEN=104") in new stack -- Executing [s@macro-dial-one:2] Set("SIP/1002-00000017", "DIALSTATUS_CW=") in new stack -- Executing [s@macro-dial-one:3] GosubIf("SIP/1002-00000017", "0?screen,1") in new stack -- Executing [s@macro-dial-one:4] GosubIf("SIP/1002-00000017", "0?cf,1") in new stack -- Executing [s@macro-dial-one:5] GotoIf("SIP/1002-00000017", "1?skip1") in new stack -- Goto (macro-dial-one,s,8) -- Executing [s@macro-dial-one:8] GotoIf("SIP/1002-00000017", "0?nodial") in new stack -- Executing [s@macro-dial-one:9] GotoIf("SIP/1002-00000017", "0?continue") in new stack -- Executing [s@macro-dial-one:10] Set("SIP/1002-00000017", "EXTHASCW=") in new stack -- Executing [s@macro-dial-one:11] GotoIf("SIP/1002-00000017", "1?next1:cwinusebusy") in new stack -- Goto (macro-dial-one,s,12) -- Executing [s@macro-dial-one:12] GotoIf("SIP/1002-00000017", "0?docfu:skip3") in new stack -- Goto (macro-dial-one,s,16) -- Executing [s@macro-dial-one:16] GotoIf("SIP/1002-00000017", "1?next2:continue") in new stack -- Goto (macro-dial-one,s,17) -- Executing [s@macro-dial-one:17] GotoIf("SIP/1002-00000017", "1?continue") in new stack -- Goto (macro-dial-one,s,25) -- Executing [s@macro-dial-one:25] GotoIf("SIP/1002-00000017", "0?nodial") in new stack -- Executing [s@macro-dial-one:26] GosubIf("SIP/1002-00000017", "1?dstring,1:dlocal,1") in new stack -- Executing [dstring@macro-dial-one:1] Set("SIP/1002-00000017", "DSTRING=") in new stack -- Executing [dstring@macro-dial-one:2] Set("SIP/1002-00000017", "DEVICES=104") in new stack -- Executing [dstring@macro-dial-one:3] ExecIf("SIP/1002-00000017", "0?Return()") in new stack -- Executing [dstring@macro-dial-one:4] ExecIf("SIP/1002-00000017", "0?Set(DEVICES=04)") in new stack -- Executing [dstring@macro-dial-one:5] Set("SIP/1002-00000017", "LOOPCNT=1") in new stack -- Executing [dstring@macro-dial-one:6] Set("SIP/1002-00000017", "ITER=1") in new stack -- Executing [dstring@macro-dial-one:7] Set("SIP/1002-00000017", "THISDIAL=SIP/104") in new stack -- Executing [dstring@macro-dial-one:8] GosubIf("SIP/1002-00000017", "1?zap2dahdi,1") in new stack -- Executing [zap2dahdi@macro-dial-one:1] ExecIf("SIP/1002-00000017", "0?Return()") in new stack -- Executing [zap2dahdi@macro-dial-one:2] Set("SIP/1002-00000017", "NEWDIAL=") in new stack -- Executing [zap2dahdi@macro-dial-one:3] Set("SIP/1002-00000017", "LOOPCNT2=1") in new stack -- Executing [zap2dahdi@macro-dial-one:4] Set("SIP/1002-00000017", "ITER2=1") in new stack -- Executing [zap2dahdi@macro-dial-one:5] Set("SIP/1002-00000017", "THISPART2=SIP/104") in new stack -- Executing [zap2dahdi@macro-dial-one:6] ExecIf("SIP/1002-00000017", "0?Set(THISPART2=DAHDI/104)") in new stack -- Executing [zap2dahdi@macro-dial-one:7] Set("SIP/1002-00000017", "NEWDIAL=SIP/104&") in new stack -- Executing [zap2dahdi@macro-dial-one:8] Set("SIP/1002-00000017", "ITER2=2") in new stack -- Executing [zap2dahdi@macro-dial-one:9] GotoIf("SIP/1002-00000017", "0?begin2") in new stack -- Executing [zap2dahdi@macro-dial-one:10] Set("SIP/1002-00000017", "THISDIAL=SIP/104") in new stack -- Executing [zap2dahdi@macro-dial-one:11] Return("SIP/1002-00000017", "") in new stack -- Executing [dstring@macro-dial-one:9] Set("SIP/1002-00000017", "DSTRING=SIP/104&") in new stack -- Executing [dstring@macro-dial-one:10] Set("SIP/1002-00000017", "ITER=2") in new stack -- Executing [dstring@macro-dial-one:11] GotoIf("SIP/1002-00000017", "0?begin") in new stack -- Executing [dstring@macro-dial-one:12] Set("SIP/1002-00000017", "DSTRING=SIP/104") in new stack -- Executing [dstring@macro-dial-one:13] Return("SIP/1002-00000017", "") in new stack -- Executing [s@macro-dial-one:27] GotoIf("SIP/1002-00000017", "0?nodial") in new stack -- Executing [s@macro-dial-one:28] GotoIf("SIP/1002-00000017", "1?skiptrace") in new stack -- Goto (macro-dial-one,s,30) -- Executing [s@macro-dial-one:30] Set("SIP/1002-00000017", "D_OPTIONS=tr") in new stack -- Executing [s@macro-dial-one:31] ExecIf("SIP/1002-00000017", "0?SIPAddHeader(Alert-Info: )") in new stack -- Executing [s@macro-dial-one:32] ExecIf("SIP/1002-00000017", "0?SIPAddHeader()") in new stack -- Executing [s@macro-dial-one:33] ExecIf("SIP/1002-00000017", "0?Set(CHANNEL(musicclass)=)") in new stack -- Executing [s@macro-dial-one:34] GosubIf("SIP/1002-00000017", "0?qwait,1") in new stack -- Executing [s@macro-dial-one:35] Set("SIP/1002-00000017", "__CWIGNORE=") in new stack -- Executing [s@macro-dial-one:36] Set("SIP/1002-00000017", "__KEEPCID=TRUE") in new stack -- Executing [s@macro-dial-one:37] Dial("SIP/1002-00000017", "SIP/104,15,tr") in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called SIP/104 -- SIP/104-00000018 is ringing -- Nobody picked up in 15000 ms -- Executing [s@macro-dial-one:38] ExecIf("SIP/1002-00000017", "0?Set(DIALSTATUS=)") in new stack -- Executing [s@macro-dial-one:39] GosubIf("SIP/1002-00000017", "0?s-NOANSWER,1") in new stack -- Executing [s@macro-dial-one:40] MacroExit("SIP/1002-00000017", "") in new stack -- Executing [s@macro-exten-vm:10] GotoIf("SIP/1002-00000017", "0?exit") in new stack -- Executing [s@macro-exten-vm:11] Set("SIP/1002-00000017", "SV_DIALSTATUS=NOANSWER") in new stack -- Executing [s@macro-exten-vm:12] GosubIf("SIP/1002-00000017", "0?docfu,1") in new stack -- Executing [s@macro-exten-vm:13] GosubIf("SIP/1002-00000017", "0?docfb,1") in new stack -- Executing [s@macro-exten-vm:14] Set("SIP/1002-00000017", "DIALSTATUS=NOANSWER") in new stack -- Executing [s@macro-exten-vm:15] NoOp("SIP/1002-00000017", "Voicemail is '104'") in new stack -- Executing [s@macro-exten-vm:16] GotoIf("SIP/1002-00000017", "0?s-NOANSWER,1") in new stack -- Executing [s@macro-exten-vm:17] NoOp("SIP/1002-00000017", "Sending to Voicemail box 104") in new stack -- Executing [s@macro-exten-vm:18] Macro("SIP/1002-00000017", "vm,104,NOANSWER,") in new stack -- Executing [s@macro-vm:1] Macro("SIP/1002-00000017", "user-callerid,SKIPTTL") in new stack -- Executing [s@macro-user-callerid:1] Set("SIP/1002-00000017", "AMPUSER=1002") in new stack -- Executing [s@macro-user-callerid:2] GotoIf("SIP/1002-00000017", "0?report") in new stack -- Executing [s@macro-user-callerid:3] ExecIf("SIP/1002-00000017", "0?Set(REALCALLERIDNUM=1002)") in new stack -- Executing [s@macro-user-callerid:4] Set("SIP/1002-00000017", "AMPUSER=1002") in new stack -- Executing [s@macro-user-callerid:5] Set("SIP/1002-00000017", "AMPUSERCIDNAME=1002") in new stack -- Executing [s@macro-user-callerid:6] GotoIf("SIP/1002-00000017", "0?report") in new stack -- Executing [s@macro-user-callerid:7] Set("SIP/1002-00000017", "AMPUSERCID=1002") in new stack -- Executing [s@macro-user-callerid:8] Set("SIP/1002-00000017", "CALLERID(all)="1002" ") in new stack -- Executing [s@macro-user-callerid:9] ExecIf("SIP/1002-00000017", "1?Set(CHANNEL(language)=es)") in new stack -- Executing [s@macro-user-callerid:10] GotoIf("SIP/1002-00000017", "1?continue") in new stack -- Goto (macro-user-callerid,s,19) -- Executing [s@macro-user-callerid:19] Set("SIP/1002-00000017", "CALLERID(number)=1002") in new stack -- Executing [s@macro-user-callerid:20] Set("SIP/1002-00000017", "CALLERID(name)=1002") in new stack -- Executing [s@macro-user-callerid:21] NoOp("SIP/1002-00000017", "Using CallerID "1002" ") in new stack -- Executing [s@macro-vm:2] Set("SIP/1002-00000017", "VMGAIN=""") in new stack -- Executing [s@macro-vm:3] GotoIf("SIP/1002-00000017", "1?vmx,1") in new stack -- Goto (macro-vm,vmx,1) -- Executing [vmx@macro-vm:1] Set("SIP/1002-00000017", "MEXTEN=104") in new stack -- Executing [vmx@macro-vm:2] Set("SIP/1002-00000017", "MMODE=NOANSWER") in new stack -- Executing [vmx@macro-vm:3] Set("SIP/1002-00000017", "RETVM=") in new stack -- Executing [vmx@macro-vm:4] Set("SIP/1002-00000017", "MODE=unavail") in new stack -- Executing [vmx@macro-vm:5] GotoIf("SIP/1002-00000017", "1?chknomsg") in new stack -- Goto (macro-vm,vmx,7) -- Executing [vmx@macro-vm:7] GotoIf("SIP/1002-00000017", "0?s-NOANSWER,1") in new stack -- Executing [vmx@macro-vm:8] GotoIf("SIP/1002-00000017", "1?notdirect") in new stack -- Goto (macro-vm,vmx,10) -- Executing [vmx@macro-vm:10] NoOp("SIP/1002-00000017", "Checking if ext 104 is enabled: ") in new stack -- Executing [vmx@macro-vm:11] GotoIf("SIP/1002-00000017", "1?s-NOANSWER,1") in new stack -- Goto (macro-vm,s-NOANSWER,1) -- Executing [s-NOANSWER@macro-vm:1] Macro("SIP/1002-00000017", "get-vmcontext,104") in new stack -- Executing [s@macro-get-vmcontext:1] Set("SIP/1002-00000017", "VMCONTEXT=default") in new stack -- Executing [s@macro-get-vmcontext:2] GotoIf("SIP/1002-00000017", "0?200:300") in new stack -- Goto (macro-get-vmcontext,s,300) -- Executing [s@macro-get-vmcontext:300] NoOp("SIP/1002-00000017", "") in new stack -- Executing [s-NOANSWER@macro-vm:2] VoiceMail("SIP/1002-00000017", "104@default,u""") in new stack -- Playing 'vm-theperson.gsm' (language 'es') -- Playing 'digits/1.gsm' (language 'es') -- Playing 'digits/0.gsm' (language 'es') -- Playing 'digits/4.gsm' (language 'es') -- Playing 'vm-isunavail.gsm' (language 'es') -- Playing 'vm-intro.gsm' (language 'es') -- Playing 'beep.gsm' (language 'es') -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/default/104/tmp/DBIihc format: wav49, 0xb7d86570 -- x=1, open writing: /var/spool/asterisk/voicemail/default/104/tmp/DBIihc format: wav, 0xb7d8ddd0 -- User hung up == Parsing '/var/spool/asterisk/voicemail/default/104/INBOX/msg0001.txt': == Found == Parsing '/var/spool/asterisk/voicemail/default/104/INBOX/msg0001.txt': == Found == Spawn extension (macro-vm, s-NOANSWER, 2) exited non-zero on 'SIP/1002-00000017' in macro 'vm' == Spawn extension (macro-exten-vm, s, 18) exited non-zero on 'SIP/1002-00000017' in macro 'exten-vm' == Spawn extension (from-internal, 104, 1) exited non-zero on 'SIP/1002-00000017' -- Executing [h@from-internal:1] Macro("SIP/1002-00000017", "hangupcall") in new stack -- Executing [s@macro-hangupcall:1] GotoIf("SIP/1002-00000017", "1?endmixmoncheck") in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] NoOp("SIP/1002-00000017", "End of MIXMON check") in new stack -- Executing [s@macro-hangupcall:10] GotoIf("SIP/1002-00000017", "1?nomeetmemon") in new stack -- Goto (macro-hangupcall,s,15) -- Executing [s@macro-hangupcall:15] NoOp("SIP/1002-00000017", "MEETME_RECORDINGFILE=") in new stack -- Executing [s@macro-hangupcall:16] GotoIf("SIP/1002-00000017", "1?noautomon") in new stack -- Goto (macro-hangupcall,s,18) -- Executing [s@macro-hangupcall:18] NoOp("SIP/1002-00000017", "TOUCH_MONITOR_OUTPUT=") in new stack -- Executing [s@macro-hangupcall:19] GotoIf("SIP/1002-00000017", "1?noautomon2") in new stack -- Goto (macro-hangupcall,s,25) -- Executing [s@macro-hangupcall:25] NoOp("SIP/1002-00000017", "MONITOR_FILENAME=") in new stack -- Executing [s@macro-hangupcall:26] GotoIf("SIP/1002-00000017", "1?skiprg") in new stack -- Goto (macro-hangupcall,s,29) -- Executing [s@macro-hangupcall:29] GotoIf("SIP/1002-00000017", "1?skipblkvm") in new stack -- Goto (macro-hangupcall,s,32) -- Executing [s@macro-hangupcall:32] GotoIf("SIP/1002-00000017", "1?theend") in new stack -- Goto (macro-hangupcall,s,34) -- Executing [s@macro-hangupcall:34] Hangup("SIP/1002-00000017", "") in new stack == Spawn extension (macro-hangupcall, s, 34) exited non-zero on 'SIP/1002-00000017' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/1002-00000017' Gracias, JF Wed, 22 May 2013 16:20:15 -0500 Subject: Audio Voicemail - by: afos0110 http://elastix.org/index.php/en/component/kunena/11-ayuda/76606-audio-voicemail.html#122330 Tengo el siguiente problema con el voicemail cuando marco a unas extension, solo timbra una ves y de una pasa al voicemail, aun estado la extension disponible. -- Executing [104@from-internal:1] Macro("SIP/1002-00000017", "exten-vm,104,104") in new stack -- Executing [s@macro-exten-vm:1] Macro("SIP/1002-00000017", "user-callerid,") in new stack -- Executing [s@macro-user-callerid:1] Set("SIP/1002-00000017", "AMPUSER=1002") in new stack -- Executing [s@macro-user-callerid:2] GotoIf("SIP/1002-00000017", "0?report") in new stack -- Executing [s@macro-user-callerid:3] ExecIf("SIP/1002-00000017", "1?Set(REALCALLERIDNUM=1002)") in new stack -- Executing [s@macro-user-callerid:4] Set("SIP/1002-00000017", "AMPUSER=1002") in new stack -- Executing [s@macro-user-callerid:5] Set("SIP/1002-00000017", "AMPUSERCIDNAME=1002") in new stack -- Executing [s@macro-user-callerid:6] GotoIf("SIP/1002-00000017", "0?report") in new stack -- Executing [s@macro-user-callerid:7] Set("SIP/1002-00000017", "AMPUSERCID=1002") in new stack -- Executing [s@macro-user-callerid:8] Set("SIP/1002-00000017", "CALLERID(all)="1002" ") in new stack -- Executing [s@macro-user-callerid:9] ExecIf("SIP/1002-00000017", "1?Set(CHANNEL(language)=es)") in new stack -- Executing [s@macro-user-callerid:10] GotoIf("SIP/1002-00000017", "0?continue") in new stack -- Executing [s@macro-user-callerid:11] Set("SIP/1002-00000017", "__TTL=64") in new stack -- Executing [s@macro-user-callerid:12] GotoIf("SIP/1002-00000017", "1?continue") in new stack -- Goto (macro-user-callerid,s,19) -- Executing [s@macro-user-callerid:19] Set("SIP/1002-00000017", "CALLERID(number)=1002") in new stack -- Executing [s@macro-user-callerid:20] Set("SIP/1002-00000017", "CALLERID(name)=1002") in new stack -- Executing [s@macro-user-callerid:21] NoOp("SIP/1002-00000017", "Using CallerID "1002" ") in new stack -- Executing [s@macro-exten-vm:2] Set("SIP/1002-00000017", "RingGroupMethod=none") in new stack -- Executing [s@macro-exten-vm:3] Set("SIP/1002-00000017", "VMBOX=104") in new stack -- Executing [s@macro-exten-vm:4] Set("SIP/1002-00000017", "__EXTTOCALL=104") in new stack -- Executing [s@macro-exten-vm:5] Set("SIP/1002-00000017", "CFUEXT=") in new stack -- Executing [s@macro-exten-vm:6] Set("SIP/1002-00000017", "CFBEXT=") in new stack -- Executing [s@macro-exten-vm:7] Set("SIP/1002-00000017", "RT=15") in new stack -- Executing [s@macro-exten-vm:8] Macro("SIP/1002-00000017", "record-enable,104,IN") in new stack -- Executing [s@macro-record-enable:1] GotoIf("SIP/1002-00000017", "1?check") in new stack -- Goto (macro-record-enable,s,4) -- Executing [s@macro-record-enable:4] ExecIf("SIP/1002-00000017", "0?MacroExit()") in new stack -- Executing [s@macro-record-enable:5] GotoIf("SIP/1002-00000017", "0?Group:OUT") in new stack -- Goto (macro-record-enable,s,15) -- Executing [s@macro-record-enable:15] GotoIf("SIP/1002-00000017", "1?IN") in new stack -- Goto (macro-record-enable,s,20) -- Executing [s@macro-record-enable:20] ExecIf("SIP/1002-00000017", "1?MacroExit()") in new stack -- Executing [s@macro-exten-vm:9] Macro("SIP/1002-00000017", "dial-one,15,tr,104") in new stack -- Executing [s@macro-dial-one:1] Set("SIP/1002-00000017", "DEXTEN=104") in new stack -- Executing [s@macro-dial-one:2] Set("SIP/1002-00000017", "DIALSTATUS_CW=") in new stack -- Executing [s@macro-dial-one:3] GosubIf("SIP/1002-00000017", "0?screen,1") in new stack -- Executing [s@macro-dial-one:4] GosubIf("SIP/1002-00000017", "0?cf,1") in new stack -- Executing [s@macro-dial-one:5] GotoIf("SIP/1002-00000017", "1?skip1") in new stack -- Goto (macro-dial-one,s,8) -- Executing [s@macro-dial-one:8] GotoIf("SIP/1002-00000017", "0?nodial") in new stack -- Executing [s@macro-dial-one:9] GotoIf("SIP/1002-00000017", "0?continue") in new stack -- Executing [s@macro-dial-one:10] Set("SIP/1002-00000017", "EXTHASCW=") in new stack -- Executing [s@macro-dial-one:11] GotoIf("SIP/1002-00000017", "1?next1:cwinusebusy") in new stack -- Goto (macro-dial-one,s,12) -- Executing [s@macro-dial-one:12] GotoIf("SIP/1002-00000017", "0?docfu:skip3") in new stack -- Goto (macro-dial-one,s,16) -- Executing [s@macro-dial-one:16] GotoIf("SIP/1002-00000017", "1?next2:continue") in new stack -- Goto (macro-dial-one,s,17) -- Executing [s@macro-dial-one:17] GotoIf("SIP/1002-00000017", "1?continue") in new stack -- Goto (macro-dial-one,s,25) -- Executing [s@macro-dial-one:25] GotoIf("SIP/1002-00000017", "0?nodial") in new stack -- Executing [s@macro-dial-one:26] GosubIf("SIP/1002-00000017", "1?dstring,1:dlocal,1") in new stack -- Executing [dstring@macro-dial-one:1] Set("SIP/1002-00000017", "DSTRING=") in new stack -- Executing [dstring@macro-dial-one:2] Set("SIP/1002-00000017", "DEVICES=104") in new stack -- Executing [dstring@macro-dial-one:3] ExecIf("SIP/1002-00000017", "0?Return()") in new stack -- Executing [dstring@macro-dial-one:4] ExecIf("SIP/1002-00000017", "0?Set(DEVICES=04)") in new stack -- Executing [dstring@macro-dial-one:5] Set("SIP/1002-00000017", "LOOPCNT=1") in new stack -- Executing [dstring@macro-dial-one:6] Set("SIP/1002-00000017", "ITER=1") in new stack -- Executing [dstring@macro-dial-one:7] Set("SIP/1002-00000017", "THISDIAL=SIP/104") in new stack -- Executing [dstring@macro-dial-one:8] GosubIf("SIP/1002-00000017", "1?zap2dahdi,1") in new stack -- Executing [zap2dahdi@macro-dial-one:1] ExecIf("SIP/1002-00000017", "0?Return()") in new stack -- Executing [zap2dahdi@macro-dial-one:2] Set("SIP/1002-00000017", "NEWDIAL=") in new stack -- Executing [zap2dahdi@macro-dial-one:3] Set("SIP/1002-00000017", "LOOPCNT2=1") in new stack -- Executing [zap2dahdi@macro-dial-one:4] Set("SIP/1002-00000017", "ITER2=1") in new stack -- Executing [zap2dahdi@macro-dial-one:5] Set("SIP/1002-00000017", "THISPART2=SIP/104") in new stack -- Executing [zap2dahdi@macro-dial-one:6] ExecIf("SIP/1002-00000017", "0?Set(THISPART2=DAHDI/104)") in new stack -- Executing [zap2dahdi@macro-dial-one:7] Set("SIP/1002-00000017", "NEWDIAL=SIP/104&") in new stack -- Executing [zap2dahdi@macro-dial-one:8] Set("SIP/1002-00000017", "ITER2=2") in new stack -- Executing [zap2dahdi@macro-dial-one:9] GotoIf("SIP/1002-00000017", "0?begin2") in new stack -- Executing [zap2dahdi@macro-dial-one:10] Set("SIP/1002-00000017", "THISDIAL=SIP/104") in new stack -- Executing [zap2dahdi@macro-dial-one:11] Return("SIP/1002-00000017", "") in new stack -- Executing [dstring@macro-dial-one:9] Set("SIP/1002-00000017", "DSTRING=SIP/104&") in new stack -- Executing [dstring@macro-dial-one:10] Set("SIP/1002-00000017", "ITER=2") in new stack -- Executing [dstring@macro-dial-one:11] GotoIf("SIP/1002-00000017", "0?begin") in new stack -- Executing [dstring@macro-dial-one:12] Set("SIP/1002-00000017", "DSTRING=SIP/104") in new stack -- Executing [dstring@macro-dial-one:13] Return("SIP/1002-00000017", "") in new stack -- Executing [s@macro-dial-one:27] GotoIf("SIP/1002-00000017", "0?nodial") in new stack -- Executing [s@macro-dial-one:28] GotoIf("SIP/1002-00000017", "1?skiptrace") in new stack -- Goto (macro-dial-one,s,30) -- Executing [s@macro-dial-one:30] Set("SIP/1002-00000017", "D_OPTIONS=tr") in new stack -- Executing [s@macro-dial-one:31] ExecIf("SIP/1002-00000017", "0?SIPAddHeader(Alert-Info: )") in new stack -- Executing [s@macro-dial-one:32] ExecIf("SIP/1002-00000017", "0?SIPAddHeader()") in new stack -- Executing [s@macro-dial-one:33] ExecIf("SIP/1002-00000017", "0?Set(CHANNEL(musicclass)=)") in new stack -- Executing [s@macro-dial-one:34] GosubIf("SIP/1002-00000017", "0?qwait,1") in new stack -- Executing [s@macro-dial-one:35] Set("SIP/1002-00000017", "__CWIGNORE=") in new stack -- Executing [s@macro-dial-one:36] Set("SIP/1002-00000017", "__KEEPCID=TRUE") in new stack -- Executing [s@macro-dial-one:37] Dial("SIP/1002-00000017", "SIP/104,15,tr") in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called SIP/104 -- SIP/104-00000018 is ringing -- Nobody picked up in 15000 ms -- Executing [s@macro-dial-one:38] ExecIf("SIP/1002-00000017", "0?Set(DIALSTATUS=)") in new stack -- Executing [s@macro-dial-one:39] GosubIf("SIP/1002-00000017", "0?s-NOANSWER,1") in new stack -- Executing [s@macro-dial-one:40] MacroExit("SIP/1002-00000017", "") in new stack -- Executing [s@macro-exten-vm:10] GotoIf("SIP/1002-00000017", "0?exit") in new stack -- Executing [s@macro-exten-vm:11] Set("SIP/1002-00000017", "SV_DIALSTATUS=NOANSWER") in new stack -- Executing [s@macro-exten-vm:12] GosubIf("SIP/1002-00000017", "0?docfu,1") in new stack -- Executing [s@macro-exten-vm:13] GosubIf("SIP/1002-00000017", "0?docfb,1") in new stack -- Executing [s@macro-exten-vm:14] Set("SIP/1002-00000017", "DIALSTATUS=NOANSWER") in new stack -- Executing [s@macro-exten-vm:15] NoOp("SIP/1002-00000017", "Voicemail is '104'") in new stack -- Executing [s@macro-exten-vm:16] GotoIf("SIP/1002-00000017", "0?s-NOANSWER,1") in new stack -- Executing [s@macro-exten-vm:17] NoOp("SIP/1002-00000017", "Sending to Voicemail box 104") in new stack -- Executing [s@macro-exten-vm:18] Macro("SIP/1002-00000017", "vm,104,NOANSWER,") in new stack -- Executing [s@macro-vm:1] Macro("SIP/1002-00000017", "user-callerid,SKIPTTL") in new stack -- Executing [s@macro-user-callerid:1] Set("SIP/1002-00000017", "AMPUSER=1002") in new stack -- Executing [s@macro-user-callerid:2] GotoIf("SIP/1002-00000017", "0?report") in new stack -- Executing [s@macro-user-callerid:3] ExecIf("SIP/1002-00000017", "0?Set(REALCALLERIDNUM=1002)") in new stack -- Executing [s@macro-user-callerid:4] Set("SIP/1002-00000017", "AMPUSER=1002") in new stack -- Executing [s@macro-user-callerid:5] Set("SIP/1002-00000017", "AMPUSERCIDNAME=1002") in new stack -- Executing [s@macro-user-callerid:6] GotoIf("SIP/1002-00000017", "0?report") in new stack -- Executing [s@macro-user-callerid:7] Set("SIP/1002-00000017", "AMPUSERCID=1002") in new stack -- Executing [s@macro-user-callerid:8] Set("SIP/1002-00000017", "CALLERID(all)="1002" ") in new stack -- Executing [s@macro-user-callerid:9] ExecIf("SIP/1002-00000017", "1?Set(CHANNEL(language)=es)") in new stack -- Executing [s@macro-user-callerid:10] GotoIf("SIP/1002-00000017", "1?continue") in new stack -- Goto (macro-user-callerid,s,19) -- Executing [s@macro-user-callerid:19] Set("SIP/1002-00000017", "CALLERID(number)=1002") in new stack -- Executing [s@macro-user-callerid:20] Set("SIP/1002-00000017", "CALLERID(name)=1002") in new stack -- Executing [s@macro-user-callerid:21] NoOp("SIP/1002-00000017", "Using CallerID "1002" ") in new stack -- Executing [s@macro-vm:2] Set("SIP/1002-00000017", "VMGAIN=""") in new stack -- Executing [s@macro-vm:3] GotoIf("SIP/1002-00000017", "1?vmx,1") in new stack -- Goto (macro-vm,vmx,1) -- Executing [vmx@macro-vm:1] Set("SIP/1002-00000017", "MEXTEN=104") in new stack -- Executing [vmx@macro-vm:2] Set("SIP/1002-00000017", "MMODE=NOANSWER") in new stack -- Executing [vmx@macro-vm:3] Set("SIP/1002-00000017", "RETVM=") in new stack -- Executing [vmx@macro-vm:4] Set("SIP/1002-00000017", "MODE=unavail") in new stack -- Executing [vmx@macro-vm:5] GotoIf("SIP/1002-00000017", "1?chknomsg") in new stack -- Goto (macro-vm,vmx,7) -- Executing [vmx@macro-vm:7] GotoIf("SIP/1002-00000017", "0?s-NOANSWER,1") in new stack -- Executing [vmx@macro-vm:8] GotoIf("SIP/1002-00000017", "1?notdirect") in new stack -- Goto (macro-vm,vmx,10) -- Executing [vmx@macro-vm:10] NoOp("SIP/1002-00000017", "Checking if ext 104 is enabled: ") in new stack -- Executing [vmx@macro-vm:11] GotoIf("SIP/1002-00000017", "1?s-NOANSWER,1") in new stack -- Goto (macro-vm,s-NOANSWER,1) -- Executing [s-NOANSWER@macro-vm:1] Macro("SIP/1002-00000017", "get-vmcontext,104") in new stack -- Executing [s@macro-get-vmcontext:1] Set("SIP/1002-00000017", "VMCONTEXT=default") in new stack -- Executing [s@macro-get-vmcontext:2] GotoIf("SIP/1002-00000017", "0?200:300") in new stack -- Goto (macro-get-vmcontext,s,300) -- Executing [s@macro-get-vmcontext:300] NoOp("SIP/1002-00000017", "") in new stack -- Executing [s-NOANSWER@macro-vm:2] VoiceMail("SIP/1002-00000017", "104@default,u""") in new stack -- Playing 'vm-theperson.gsm' (language 'es') -- Playing 'digits/1.gsm' (language 'es') -- Playing 'digits/0.gsm' (language 'es') -- Playing 'digits/4.gsm' (language 'es') -- Playing 'vm-isunavail.gsm' (language 'es') -- Playing 'vm-intro.gsm' (language 'es') -- Playing 'beep.gsm' (language 'es') -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/default/104/tmp/DBIihc format: wav49, 0xb7d86570 -- x=1, open writing: /var/spool/asterisk/voicemail/default/104/tmp/DBIihc format: wav, 0xb7d8ddd0 -- User hung up == Parsing '/var/spool/asterisk/voicemail/default/104/INBOX/msg0001.txt': == Found == Parsing '/var/spool/asterisk/voicemail/default/104/INBOX/msg0001.txt': == Found == Spawn extension (macro-vm, s-NOANSWER, 2) exited non-zero on 'SIP/1002-00000017' in macro 'vm' == Spawn extension (macro-exten-vm, s, 18) exited non-zero on 'SIP/1002-00000017' in macro 'exten-vm' == Spawn extension (from-internal, 104, 1) exited non-zero on 'SIP/1002-00000017' -- Executing [h@from-internal:1] Macro("SIP/1002-00000017", "hangupcall") in new stack -- Executing [s@macro-hangupcall:1] GotoIf("SIP/1002-00000017", "1?endmixmoncheck") in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] NoOp("SIP/1002-00000017", "End of MIXMON check") in new stack -- Executing [s@macro-hangupcall:10] GotoIf("SIP/1002-00000017", "1?nomeetmemon") in new stack -- Goto (macro-hangupcall,s,15) -- Executing [s@macro-hangupcall:15] NoOp("SIP/1002-00000017", "MEETME_RECORDINGFILE=") in new stack -- Executing [s@macro-hangupcall:16] GotoIf("SIP/1002-00000017", "1?noautomon") in new stack -- Goto (macro-hangupcall,s,18) -- Executing [s@macro-hangupcall:18] NoOp("SIP/1002-00000017", "TOUCH_MONITOR_OUTPUT=") in new stack -- Executing [s@macro-hangupcall:19] GotoIf("SIP/1002-00000017", "1?noautomon2") in new stack -- Goto (macro-hangupcall,s,25) -- Executing [s@macro-hangupcall:25] NoOp("SIP/1002-00000017", "MONITOR_FILENAME=") in new stack -- Executing [s@macro-hangupcall:26] GotoIf("SIP/1002-00000017", "1?skiprg") in new stack -- Goto (macro-hangupcall,s,29) -- Executing [s@macro-hangupcall:29] GotoIf("SIP/1002-00000017", "1?skipblkvm") in new stack -- Goto (macro-hangupcall,s,32) -- Executing [s@macro-hangupcall:32] GotoIf("SIP/1002-00000017", "1?theend") in new stack -- Goto (macro-hangupcall,s,34) -- Executing [s@macro-hangupcall:34] Hangup("SIP/1002-00000017", "") in new stack == Spawn extension (macro-hangupcall, s, 34) exited non-zero on 'SIP/1002-00000017' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/1002-00000017' Gracias por su ayuda, Saludos, JF Wed, 22 May 2013 15:48:35 -0500 Subject: How to handle two different companies on handsets? - by: hinzinho http://elastix.org/index.php/en/component/kunena/3-help/121601-how-to-handle-two-different-companies-on-handsets.html#122329 sharq, not sure what you are looking for. Inbound route is handled by the rules set in Inbound Routes. Wed, 22 May 2013 15:15:47 -0500 Subject: Event Notification API - by: rolandli http://elastix.org/index.php/en/component/kunena/3-help/122328-event-notification-api.html#122328 We are building a custom CRM and we want to integrate the phone system with it. Here is what we want to do: 1. A call comes in 2. We look up the caller ID in our CRM and find out who the account manager is 3. Transfer the call to the account manager This involves setting up Elastix to talk to our CRM and I am at lost how to achieve this. Digium has great API for even notifications: http://developers.digium.com/switchvox/wiki/index.php/UrlManagerAPI_overview I am wondering if Elastix has something similar to this. Than kyou Wed, 22 May 2013 15:10:44 -0500 Subject: ivr no permite marcación directa - by: jmontoya http://elastix.org/index.php/en/component/kunena/50-novatos/122326-ivr-no-permite-marcacion-directa.html#122326 buenos días bueno, soy nuevo por este foro tengo varios meses de estar utilizando Elastix y quisiera comentar un problema que me está afectando. tengo un IVR configurado en mi central Elastix versión Elastix 2.0.0 versión de Asterisk 1.6.2.13 la cual tiene integración con una central Panasonic por medio de un primario a través de esta Conexión tengo otro escenario donde las panasonic se interconectan entre ellas con los prefijos de marcación interna para extensiones son los siguientes: 1XX 2XX 3XX tengo un Elastix2 en una región remota y su prefijo es 4XX Ok hasta ahí vamos bien, en mi Elastix1 configuré un IVR y marque la opción de activar la marcación directa pero como las opciones del contestador son iguales a los prefijos de mis otras centrales, el Elastix solo escucha el primero dígito y me cumple las condiciones del IVR pero no la extensión requerida agregue la opcion siguiente: En /etc/asterisk/chan_dahdi.conf: escribí, relaxdtmf=yes aún así no he podido encontrar la respuesta a mi problema les paso una copia de mi Cli para que la vean. de antemando agradezco la atención y la ayuda que me puedan brindar. Nota: escribí en este foro porque se habla de errores de IVR's mas no sé si estoy en lo correcto si cometí ese error disculpen muchas gracias saludos a todos. n new stack -- Executing [s@macro-user-callerid:6] GotoIf("DAHDI/9-1", "1?report") in new stack -- Goto (macro-user-callerid,s,10) -- Executing [s@macro-user-callerid:10] GotoIf("DAHDI/9-1", "0?continue") in new stack -- Executing [s@macro-user-callerid:11] Set("DAHDI/9-1", "__TTL=63") in new stack -- Executing [s@macro-user-callerid:12] GotoIf("DAHDI/9-1", "1?continue") in new stack -- Goto (macro-user-callerid,s,19) -- Executing [s@macro-user-callerid:19] NoOp("DAHDI/9-1", "Using CallerID "89536964" ") in new stack -- Executing [s@macro-exten-vm:2] Set("DAHDI/9-1", "RingGroupMethod=none") in new stack -- Executing [s@macro-exten-vm:3] Set("DAHDI/9-1", "VMBOX=novm") in new stack -- Executing [s@macro-exten-vm:4] Set("DAHDI/9-1", "EXTTOCALL=841") in new stack -- Executing [s@macro-exten-vm:5] Set("DAHDI/9-1", "CFUEXT=") in new stack -- Executing [s@macro-exten-vm:6] Set("DAHDI/9-1", "CFBEXT=") in new stack -- Executing [s@macro-exten-vm:7] Set("DAHDI/9-1", "RT=""") in new stack -- Executing [s@macro-exten-vm:8] Macro("DAHDI/9-1", "record-enable,841,IN") in new stack -- Executing [s@macro-record-enable:1] GotoIf("DAHDI/9-1", "1?check") in new stack -- Goto (macro-record-enable,s,4) -- Executing [s@macro-record-enable:4] ExecIf("DAHDI/9-1", "0?MacroExit()") in new stack -- Executing [s@macro-record-enable:5] GotoIf("DAHDI/9-1", "0?Group:OUT") in new stack -- Goto (macro-record-enable,s,15) -- Executing [s@macro-record-enable:15] GotoIf("DAHDI/9-1", "1?IN") in new stack -- Goto (macro-record-enable,s,20) -- Executing [s@macro-record-enable:20] ExecIf("DAHDI/9-1", "1?MacroExit()") in new stack -- Executing [s@macro-exten-vm:9] Macro("DAHDI/9-1", "dial,,tr,841") in new stack -- Executing [s@macro-dial:1] GotoIf("DAHDI/9-1", "1?dial") in new stack -- Goto (macro-dial,s,3) -- Executing [s@macro-dial:3] AGI("DAHDI/9-1", "dialparties.agi") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi dialparties.agi: Starting New Dialparties.agi dialparties.agi: Caller ID name is '89536964' number is '89536964' dialparties.agi: Methodology of ring is 'none' -- dialparties.agi: Added extension 841 to extension map -- dialparties.agi: Extension 841 cf is disabled -- dialparties.agi: Extension 841 do not disturb is disabled dialparties.agi: EXTENSION_STATE: 0 (NOT_INUSE) dialparties.agi: Extension 841 has ExtensionState: 0 -- dialparties.agi: Checking CW and CFB status for extension 841 -- dialparties.agi: dbset CALLTRACE/841 to 89536964 -- dialparties.agi: Filtered ARG3: 841 -- AGI Script dialparties.agi completed, returning 0 -- Executing [s@macro-dial:7] Dial("DAHDI/9-1", "SIP/841,,tr") in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called 841 -- SIP/841-00000a71 is ringing -- Executing [1@ivr-5:1] NoOp("DAHDI/4-1", "Deleting: ") in new stack -- Executing [1@ivr-5:2] Set("DAHDI/4-1", "__NODEST=") in new stack -- Executing [1@ivr-5:3] Goto("DAHDI/4-1", "ext-queues,2,1") in new stack -- Goto (ext-queues,2,1) -- Executing [2@ext-queues:1] Macro("DAHDI/4-1", "user-callerid,") in new stack -- Executing [s@macro-user-callerid:1] Set("DAHDI/4-1", "AMPUSER=20107600") in new stack -- Executing [s@macro-user-callerid:2] GotoIf("DAHDI/4-1", "0?report") in new stack -- Executing [s@macro-user-callerid:3] ExecIf("DAHDI/4-1", "1?Set(REALCALLERIDNUM=20107600)") in new stack -- Executing [s@macro-user-callerid:4] Set("DAHDI/4-1", "AMPUSER=") in new stack -- Executing [s@macro-user-callerid:5] Set("DAHDI/4-1", "AMPUSERCIDNAME=") in new stack -- Executing [s@macro-user-callerid:6] GotoIf("DAHDI/4-1", "1?report") in new stack -- Goto (macro-user-callerid,s,10) -- Executing [s@macro-user-callerid:10] GotoIf("DAHDI/4-1", "0?continue") in new stack -- Executing [s@macro-user-callerid:11] Set("DAHDI/4-1", "__TTL=64") in new stack -- Executing [s@macro-user-callerid:12] GotoIf("DAHDI/4-1", "1?continue") in new stack -- Goto (macro-user-callerid,s,19) -- Executing [s@macro-user-callerid:19] NoOp("DAHDI/4-1", "Using CallerID "20107600" ") in new stack -- Executing [2@ext-queues:2] Answer("DAHDI/4-1", "") in new stack -- Executing [2@ext-queues:3] Set("DAHDI/4-1", "__BLKVM_OVERRIDE=BLKVM/2/DAHDI/4-1") in new stack -- Executing [2@ext-queues:4] Set("DAHDI/4-1", "__BLKVM_BASE=2") in new stack -- Executing [2@ext-queues:5] Set("DAHDI/4-1", "DB(BLKVM/2/DAHDI/4-1)=TRUE") in new stack -- Executing [2@ext-queues:6] ExecIf("DAHDI/4-1", "1?Set(_DIAL_OPTIONS=trM(auto-blkvm))") in new stack -- Executing [2@ext-queues:7] Set("DAHDI/4-1", "__NODEST=2") in new stack -- Executing [2@ext-queues:8] Set("DAHDI/4-1", "MONITOR_FILENAME=/var/spool/asterisk/monitor/q2-20130520-163528-1369089316.8510") in new stack -- Executing [2@ext-queues:9] Queue("DAHDI/4-1", "2,t,,") in new stack -- Started music on hold, class 'default', on DAHDI/4-1 -- Executing [152@from-queue:1] Set("Local/152@from-queue-f095;2", "QAGENT=152") in new stack -- Executing [152@from-queue:2] Goto("Local/152@from-queue-f095;2", "2,1") in new stack -- Goto (from-queue,2,1) -- Executing [2@from-queue:1] Goto("Local/152@from-queue-f095;2", "from-internal,152,1") in new stack -- Goto (from-internal,152,1) -- Executing [152@from-internal:1] Macro("Local/152@from-queue-f095;2", "user-callerid,SKIPTTL,") in new stack -- Executing [s@macro-user-callerid:1] Set("Local/152@from-queue-f095;2", "AMPUSER=20107600") in new stack -- Executing [s@macro-user-callerid:2] GotoIf("Local/152@from-queue-f095;2", "1?report") in new stack -- Goto (macro-user-callerid,s,10) -- Executing [s@macro-user-callerid:10] GotoIf("Local/152@from-queue-f095;2", "1?continue") in new stack -- Goto (macro-user-callerid,s,19) -- Executing [s@macro-user-callerid:19] NoOp("Local/152@from-queue-f095;2", "Using CallerID "20107600" ") in new stack -- Executing [152@from-internal:2] Set("Local/152@from-queue-f095;2", "INTRACOMPANYROUTE=YES") in new stack -- Executing [152@from-internal:3] Set("Local/152@from-queue-f095;2", "_NODEST=") in new stack -- Executing [152@from-internal:4] Macro("Local/152@from-queue-f095;2", "record-enable,20107600,OUT,") in new stack -- Executing [s@macro-record-enable:1] GotoIf("Local/152@from-queue-f095;2", "1?check") in new stack -- Goto (macro-record-enable,s,4) -- Executing [s@macro-record-enable:4] ExecIf("Local/152@from-queue-f095;2", "0?MacroExit()") in new stack -- Executing [s@macro-record-enable:5] GotoIf("Local/152@from-queue-f095;2", "0?Group:OUT") in new stack -- Goto (macro-record-enable,s,15) -- Executing [s@macro-record-enable:15] GotoIf("Local/152@from-queue-f095;2", "0?IN") in new stack -- Executing [s@macro-record-enable:16] ExecIf("Local/152@from-queue-f095;2", "1?MacroExit()") in new stack -- Executing [152@from-internal:5] Macro("Local/152@from-queue-f095;2", "dialout-trunk,8,152,,") in new stack -- Executing [s@macro-dialout-trunk:1] Set("Local/152@from-queue-f095;2", "DIAL_TRUNK=8") in new stack -- Executing [s@macro-dialout-trunk:2] GosubIf("Local/152@from-queue-f095;2", "0?sub-pincheck,s,1") in new stack -- Executing [s@macro-dialout-trunk:3] GotoIf("Local/152@from-queue-f095;2", "0?disabletrunk,1") in new stack -- Executing [s@macro-dialout-trunk:4] Set("Local/152@from-queue-f095;2", "DIAL_NUMBER=152") in new stack -- Executing [s@macro-dialout-trunk:5] Set("Local/152@from-queue-f095;2", "DIAL_TRUNK_OPTIONS=trM(auto-blkvm)") in new stack -- Executing [s@macro-dialout-trunk:6] Set("Local/152@from-queue-f095;2", "OUTBOUND_GROUP=OUT_8") in new stack -- Executing [s@macro-dialout-trunk:7] GotoIf("Local/152@from-queue-f095;2", "1?nomax") in new stack -- Goto (macro-dialout-trunk,s,9) -- Executing [s@macro-dialout-trunk:9] GotoIf("Local/152@from-queue-f095;2", "1?skipoutcid") in new stack -- Goto (macro-dialout-trunk,s,12) -- Executing [s@macro-dialout-trunk:12] ExecIf("Local/152@from-queue-f095;2", "0?AGI(fixlocalprefix)") in new stack -- Executing [s@macro-dialout-trunk:13] Set("Local/152@from-queue-f095;2", "OUTNUM=152") in new stack -- Executing [s@macro-dialout-trunk:14] Set("Local/152@from-queue-f095;2", "custom=DAHDI/g2") in new stack -- Executing [s@macro-dialout-trunk:15] ExecIf("Local/152@from-queue-f095;2", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^)trM(auto-blkvm))") in new stack -- Executing [s@macro-dialout-trunk:16] Macro("Local/152@from-queue-f095;2", "dialout-trunk-predial-hook,") in new stack -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("Local/152@from-queue-f095;2", "") in new stack -- Executing [s@macro-dialout-trunk:17] GotoIf("Local/152@from-queue-f095;2", "0?bypass,1") in new stack -- Executing [s@macro-dialout-trunk:18] GotoIf("Local/152@from-queue-f095;2", "0?customtrunk") in new stack -- Executing [s@macro-dialout-trunk:19] Dial("Local/152@from-queue-f095;2", "DAHDI/g2/152,300,trM(auto-blkvm)") in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g2/152 -- Local/152@from-queue-f095;1 is ringing -- DAHDI/32-1 is proceeding passing it to Local/152@from-queue-f095;2 -- DAHDI/32-1 is ringing -- Local/152@from-queue-f095;1 is ringing -- Channel 0/4, span 1 got hangup request, cause 16 -- Stopped music on hold on DAHDI/4-1 == Spawn extension (ext-queues, 2, 9) exited non-zero on 'DAHDI/4-1' -- Hungup 'DAHDI/32-1' == Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on 'Local/152@from-queue-f095;2' in macro 'dialout-trunk' -- Hungup 'DAHDI/4-1' == Spawn extension (from-internal, 152, 5) exited non-zero on 'Local/152@from-queue-f095;2' -- Executing [h@from-internal:1] Macro("Local/152@from-queue-f095;2", "hangupcall") in new stack -- Executing [s@macro-hangupcall:1] GotoIf("Local/152@from-queue-f095;2", "1?noautomon") in new stack -- Goto (macro-hangupcall,s,3) -- Executing [s@macro-hangupcall:3] NoOp("Local/152@from-queue-f095;2", "TOUCH_MONITOR_OUTPUT=") in new stack -- Executing [s@macro-hangupcall:4] GotoIf("Local/152@from-queue-f095;2", "1?skiprg") in new stack -- Goto (macro-hangupcall,s,7) -- Executing [s@macro-hangupcall:7] GotoIf("Local/152@from-queue-f095;2", "1?skipblkvm") in new stack -- Goto (macro-hangupcall,s,10) -- Executing [s@macro-hangupcall:10] GotoIf("Local/152@from-queue-f095;2", "1?theend") in new stack -- Goto (macro-hangupcall,s,12) -- Executing [s@macro-hangupcall:12] Hangup("Local/152@from-queue-f095;2", "") in new stack == Spawn extension (macro-hangupcall, s, 12) exited non-zero on 'Local/152@from-queue-f095;2' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'Local/152@from-queue-f095;2' -- Channel 0/9, span 1 got hangup request, cause 16 == Spawn extension (macro-dial, s, 7) exited non-zero on 'DAHDI/9-1' in macro 'dial' == Spawn extension (macro-exten-vm, s, 9) exited non-zero on 'DAHDI/9-1' in macro 'exten-vm' == Spawn extension (from-did-direct, 841, 1) exited non-zero on 'DAHDI/9-1' -- Hungup 'DAHDI/9-1' Wed, 22 May 2013 12:08:27 -0500 Subject: SIP Trunk with multiple numbers question - by: israr75 http://elastix.org/index.php/en/component/kunena/25-newbies-corner-/122325-sip-trunk-with-multiple-numbers-question.html#122325 Hi All, I have a question hoping someone could help me with. I have 2 Sip Trunks/Extentions as listed below and I have 4 numbers. What I want to know is when I setup SIP Extention in Elastics how do I setup for just for one number so the other numbers are not effected. Below are the SIP settings I have at hand Extension Number Username Password Registrar 100 EXT100 xxxxxxxx sip2.abc.co.uk 101 EXT101 xxxxxxxx sip2.abc.co.uk SIP Address 0123111@sip2.abc.co.uk SIP Address 0123222@sip2.abc.co.uk SIP Address 0123333@sip2.abc.co.uk SIP Address 0124444@sip2.abc.co.uk I am newbie and would really like someone. Its quite possible I've misunderstood some part of the process in my reading. Kind Regards Israr Wed, 22 May 2013 11:43:59 -0500 Subject: la consola WEB monitoring no muestra las llamadas - by: miguelaga http://elastix.org/index.php/en/component/kunena/11-ayuda/122324-la-consola-web-monitoring-no-muestra-las-llamadas.html#122324 Hola!! espero me puedan ayudar tengo el problema de que en Elastix 2.4.0 en la consola WEB en la pestaña MONITORING no me muestra la tabla de llamadas.. cabe mencionar que hice un yum -y update.. no se si al momento de realizar cualquiera de estos pasos se modifico a alguna llamada a la base de datos de asterisk.. y simplemente no se muestra la lista de llamadas de que se hicieron de las grabaciones en elastix... pero las grabaciones si estan en el directorio /var/spool/asterisk/monitor espero me puedan ayudar... Wed, 22 May 2013 11:32:58 -0500 Subject: Problemas con addon Call Center - by: santiagomr http://elastix.org/index.php/en/component/kunena/55-call-center/122322-problemas-con-addon-call-center.html#122322 Buen día, instalé elastix 2.4.0 en una máquina virtual siguiendo el manual, todo sin problema. Pero a la hora de dar click en la pestaña del addons me manda el siguiente error: "The system can not connect to the Web Service resource. Please check your Internet connection. SOAP-ERROR: Parsing WSDL: Couldn't load from 'http://webservice.elastix.org/modules/addons_availables/webservice/addons.wsdl' " ¿A qué se debe? ¿Cómo puedo solucionarlo? De antemano gracias Les mando lo que tengo con el ifconfig: root@PbxElastix www]# ifconfig eth0 Link encap:Ethernet HWaddr E8:00:27:5E:3F:8C inet addr:192.168.0.240 Bcast:192.168.0.255 Mask:255.255.255.0 UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1 RX packets:1307 errors:0 dropped:0 overruns:0 frame:0 TX packets:622 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:1000 RX bytes:148844 (145.3 MiB) TX bytes:437324 (427.0 KiB) Interrupt:177 Base address:0xd020 lo Link encap:Local Loopback inet addr:127.0.0.1 Mask:255.0.0.0 UP LOOPBACK RUNNING MTU:16436 Metric:1 RX packets:526 errors:0 dropped:0 overruns:0 frame:0 TX packets:526 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:0 RX bytes:43275 (42.2 KiB) TX bytes:43275 (42.2 KiB) [root@centralita ~]# hostname PbxElastix Wed, 22 May 2013 11:14:43 -0500 Subject: Phone mapping - by: gatemehdi http://elastix.org/index.php/en/component/kunena/35-openfire/122320-phone-mapping.html#122320 Hello evryone, I need your help, cause when I want to call a person in spark using openfire mixed with elastix, the account call itself, I know that the is with the phone mapping but where I don't know, the tutoriel I've followed is Elastix whitout tears. Thanks in advance. Wed, 22 May 2013 10:59:47 -0500 Subject: Llamadas entre servidores - by: edarius http://elastix.org/index.php/en/component/kunena/50-novatos/122319-llamadas-entre-servidores.html#122319 Cordial saludo, he instalado y configurado dos servidores con asterisk y los tengo conectados mediante troncales SIP, ya de un servidor se puede llamar al otro, pero quiero que cuando un persona llame al servidor A y le conteste el IVR pueda marcar directamente una extensión del servidor B, es esto posible? Wed, 22 May 2013 10:50:35 -0500 Subject: code before calling - by: c8aj http://elastix.org/index.php/en/component/kunena/25-newbies-corner-/122318-code-before-calling.html#122318 hellow Elastix family, I ask for your help on how can I make like a predial... below I describe what Id like to know: 1) get my phone, unhook it. We have here 10 pstn 2) I need to get a specific line (its the only one with international call feature available) and before making a call, I need to unlock the international call feature through a code that the local pstn office provided me to secure that line. Thanks in advance for your help and advice. Wed, 22 May 2013 10:48:38 -0500 Subject: The best way to protect your elastix from anonymus - by: samv http://elastix.org/index.php/en/component/kunena/10-success-stories/113779-the-best-way-to-protect-your-elastix-from-anonymus.html#122317 A little bit modified to this script. I add on date and time to it. #!/bin/sh newpassword=$(apg -a 1 -n 1 -x 8 -M CLn) date=$(date +"%A %B %d, %Y at %r") /usr/bin/sqlite3 /var/www/db/acl.db "UPDATE acl_user SET md5_password = '`echo -n $newpassword|md5sum|cut -d ' ' -f 1`' WHERE name = 'admin'" echo "$date New Password is: $newpassword" | mail -s "Password has changed on my Ipbx box1" yourmail@gmail.com exit 0 Sam Wed, 22 May 2013 10:31:00 -0500 Subject: Problema de timbrado con telefonos digitales - by: victoralan http://elastix.org/index.php/en/component/kunena/11-ayuda/122315-problema-de-timbrado-con-telefonos-digitales.html#122315 Buenos dias a todos, tengo el siguiente escenario con problema, espero puedan ayudarme, tengo instalado elastix 2.3 para comunicacion solamente VOIP no tengo ninguna tarjeta FXO/FXS, para la comunicacion entre los telefonos uno adaptadores VoIP Linksys, el problema es que al realizar una llamada y tener un telefono digital inalambrico marca Vtech conectado al adaptador VoIP entra la llamada pero este no timbra, se levanta el telefono y se establece sin problemas la llamada pero nunca timbra al entrar. Pero al conectar un telefono analogico marca Panasonic al mismo adaptador Voip si timbra en las llamadas entrantes. Determiné que el problema no es el telefono Vtech ya que lo conecte a una linea telefonica convencional se realizo una llamada entrante y timbro sin problemas. Por favor ayudenme a ver en que puedo estar fallando. De antemano gracias. Wed, 22 May 2013 10:12:57 -0500 Subject: "Follow me Settings" different internal | external - by: como http://elastix.org/index.php/en/component/kunena/25-newbies-corner-/122313-qfollow-me-settingsq-different-internal-nexternal.html#122313 Hi, i have a question, how can i use different follow me settings for internal and external calls without create a lot of useless extensions? For incoming external calls i would like to have some follow me settings. For example after the initial ring time of 20 seconds the call should go to the front desk. If an college is calling, the call should not go to the front desk. Is there a way to handle this :) ? Kind regards como Wed, 22 May 2013 09:07:09 -0500 Subject: "toutes les lignes sont occupées" - by: shm74 http://elastix.org/index.php/en/component/kunena/79-general/121931-qtoutes-les-lignes-sont-occupeesq.html?limit=10&start=20#122312 Pas de soucis. Wed, 22 May 2013 08:58:14 -0500 Subject: Configurar elastix con dos tarjetas de red - by: thawizz http://elastix.org/index.php/en/component/kunena/11-ayuda/89543-configurar-elastix-con-dos-tarjetas-de-red.html#122309 pedrovalencia escribió: Gracias por su participación, el problema ya lo pude resolver. Por favor podrias explicar como lo resolviste? Wed, 22 May 2013 08:44:18 -0500 Subject: Elastix et la base de données - by: danardf http://elastix.org/index.php/en/component/kunena/79-general/121830-elastix-et-la-base-de-donnees.html?limit=10&start=10#122307 Nabilpan04 wrote: Oui j'ai bien lu sauf que il n'y a pas de base de donnnées coté application Ha ben j'en sais rien mon garçon.. ce n'est pas moi qui est fait l'appli. :) Nabilpan04 wrote: Je voudrais que directement , lorsque le client saisie ses données ,ces derniers vont vers la base de données d'elastix Je ne comprend pas! A quoi elles te servent ces données saisies si elles sont stockés nul part? Nabilpan04 wrote: esque il n'y a pas une solution pour résoudre ce probleme ?? ou est ce que je devrais créer une DB coté client et je fais ce que vous avez dis càd une requete http dans le CID lookup ?? Soit tu créés une base de données MySQL sur le serveur Elastix avec un droit d'accès pour une connexion externe afin que ton appli puisse renseigner tes données dans la base. Ou alors... Le plus simple c'est que tu créés une base de données sur ton appli et qu'Elastix vienne chercher les informations via http get ou autre. Wed, 22 May 2013 08:25:21 -0500 Subject: Recording - by: bajji http://elastix.org/index.php/en/component/kunena/3-help/122289-recording.html#122303 sorry im bad in english but i use elastix 2.4 and how can i find the folder wher record are ? Wed, 22 May 2013 07:56:15 -0500 Subject: Route calls from 1 SIP trunk to another SIP trunk - by: sharq http://elastix.org/index.php/en/component/kunena/3-help/120562-route-calls-from-1-sip-trunk-to-another-sip-trunk.html#122300 what happen if you modify a trunks on server A to from-internal context? Wed, 22 May 2013 07:23:14 -0500 Subject: incomplete Fax - by: matter97 http://elastix.org/index.php/en/component/kunena/38-hylafax/122287-incomplete-fax.html#122298 Bob Firs thank you for your fast replay I use version 2.4 and I already enabled T.38 pass-through and Elastix running on dedicated hardware I don’t know what is the problem that I received good Faxs from some and incomplete from other and according to Wiredshark log that it traced by Grandstream the problem coming from high speed and I configured also the speed from freePBX to Max 9600 but also nothing change below you will find some log of incomplete faxs May 20 12:33:11.78: [26203]: [2:OK] May 20 12:33:11.86: [26203]: RECV send ERR (confirm end of retransmisison) May 20 12:33:11.86: [26203]: [7:CONNECT] May 20 12:33:11.88: [26203]: [2:OK] May 20 12:33:14.70: [26203]: RECV recv DCN (disconnect) May 20 12:33:14.70: [26203]: MODEM input buffering enabled May 20 12:33:14.70: [26203]: RECV FAX (000000568): from , page 1 in 0:01:48, INF, 3.85 line/mm, 2-D MMR, 4800 bit/s May 20 12:33:14.70: [26203]: RECV FAX (000000568): recvq/fax000000315.tif from , route to , 1 pages in 0:01:51 May 20 12:33:14.70: [26203]: RECV FAX: bin/faxrcvd.php "recvq/fax000000315.tif" "ttyIAX1" "000000568" "" "0233370705" "0233370705" "" "s" May 20 12:33:14.70: [26203]: RECV FAX: end May 20 12:33:14.70: [26203]: SESSION END May 19 11:56:24.46: [ 3023]: --> [2:OK] May 19 11:56:24.46: [ 3023]: RECV keeping unconfirmed page May 19 11:56:24.48: [ 3023]: RECV/CQ: Invalid WhiteTable code word, row 13, x 1198 May 19 11:56:24.48: [ 3023]: RECV/CQ: Adjusting for EOFB at row 13 May 19 11:56:24.48: [ 3023]: RECV: 17408 bytes of data, 13 total lines May 19 11:56:24.48: [ 3023]: MODEM set XON/XOFF/DRAIN: input ignored, output disabled May 19 11:56:24.48: [ 3023]: RECV FAX (000000535): from , page 1 in 5:45, INF, 3.85 line/mm, 2-D MMR, 4800 bit/s May 19 11:56:24.48: [ 3023]: RECV FAX (000000535): recvq/fax000000284.tif from , route to , 1 pages in 5:45 May 19 11:56:24.49: [ 3023]: RECV FAX: Failed to properly detect high-speed data carrier. May 19 11:56:24.49: [ 3023]: [7:CONNECT] May 19 11:56:24.50: [ 3023]: Wed, 22 May 2013 07:17:38 -0500 Subject: incoming fax not working elastix 2.4 - by: Bob http://elastix.org/index.php/en/component/kunena/38-hylafax/122041-incoming-fax-not-working-elastix-24.html?limit=10&start=10#122295 velez, Sorry to hear and sorry we could not help.....but i think in the long run, it is the best move..something definitely did not work correctly. Just to make sure (and make sure no bugs exist), I have taken a Elastix 2.4 fresh build Added an IAX2 extension of 1100 Setup a virtual FAX pointing to 1100 Setup a DID pointing to 1100 Called the DID - two rings and then I get the fax signal.... Good luck with it... Regards Bob Wed, 22 May 2013 06:58:08 -0500 Subject: CallerID to CDR Report, Summary by.. and Monitorig - by: mostafa33 http://elastix.org/index.php/en/component/kunena/26-tips-and-tricks/85614-callerid-to-cdr-report-summary-by-and-monitorig.html?limit=10&start=10#122275 Good modifications :) how can i add new column in reports summary to display the average call duration for incoming and outgoing calls per extension? Wed, 22 May 2013 04:29:49 -0500 Subject: Tono colgado con router OneAccess 100, de Vodafone - by: albertofcc http://elastix.org/index.php/en/component/kunena/31-asterisk/122272-tono-colgado-con-router-oneaccess-100-de-vodafone.html#122272 Hola a toda la comunidad, y felicidades por este sistema tan potente de unificación de comunicaciones. Acabo de terminar la instalación de una PBX con 4 fxo, (Nicherons TDM1600p), conectadas a un router OneAccess 100 de Vodafone España, todo va bien, pero tengo el típico problemazo con las detecciones de tono de colgado. La línea no tiene cambio de polaridad, o por lo menos, Elastix 2.4.0 no la detecta callprogress=yes hace que las llamadas salientes no lleven audio (la parte receptora no escucha nada) loaddefault=es y busydetect=yes son inoperantes el patrón de colgado es idéntico a congestion Cambiar de fxs_ks a fxs_ls hizo que la centralita dejara de funcionar, no se recibía tono de marcado en ninguna extensión y no se tramitaban las llamadas Y ya no sé por donde seguir... Ahora mismo chan_dahdi.conf tiene la configuración por omisión Gracias a tod@s de antemano. Un saludo. Wed, 22 May 2013 03:08:08 -0500 Subject: Eco-friendly side of VoIP - by: Voicebuy http://elastix.org/index.php/en/component/kunena/115-voip-providers/122271-eco-friendly-side-of-voip.html#122271 With the ongoing technological advances and eco-friendly practices that happen nowadays, more and more companies become motivated for giving their share of contribution to better and greener environment. To make their workplace more efficient and pleasing for employees, a great many businesses give their preference to VoIP. The latter serves as a unity of various converged services at the same time combining a lot of features that may help business become more environmentally friendly. Here are some of the benefits companies may reap from using VoIP for greener business: 1.Choosing VoIP as a tool to handle the overall internal and external communication first of all implies less power consumption since VoIP services need only internet connection without using an abundance of additional equipment or lines. 2.VoIP makes it possible for employees to do the same type of full-time work from home with the opportunity to handle business phone calls directly on their own mobile or landline phones. This option will obviously result in traffic pollution reduction as well as will save more on water, electricity and gas at the office. 3.In addition, VoIP allows employees to set up communication and videoconferencing with their colleagues, partners and business unities to any corner in the world without any tiring plane travels. This comes in handy not only for employees who don’t feel like being on constant business trips, but also for the environment, since it cuts down the number of people who travel by plane, which in its turn minimizes the amount of toxins airplanes release to the environment. 4.Another green side of VoIP is also the reduction of the amount of paper and ink used, since IP faxing allows employees to have all the documents arranged through digital units 5.Overall, VoIP supports long distance calls that are safe, eco-friendly and come without any wires set in substances that are potential carriers of dangerous chemicals. So it’s time for you as well to step in hand with green industry, switch to VoIP services Voicebuy offers and let your business flourish in a more eco-friendly environment! Wed, 22 May 2013 02:56:21 -0500 Subject: agent console , no active call and blank - by: cucup_wawaw http://elastix.org/index.php/en/component/kunena/19-call-center/37997-agent-console--no-active-call-and-blank.html?limit=10&start=10#122269 i'm facing same problem.. anyone has solved it yer??? thanks before Wed, 22 May 2013 00:22:05 -0500 Subject: callcenter callback extensions - by: dariohimo http://elastix.org/index.php/en/component/kunena/65-otros/122267-callcenter-callback-extensions.html#122267 Hola, estoy creado opciones de agente callback extensions y me sale este error con algunas extensiones a crear. En el modulo no sale nada creado. Los agentes estan todo creado en el modulo agente. y callback a dicionar siempre sale esto: Error al intentar insertar un agente. Agent already exists he instalado y desistalado el modulo de callcenter y nada que se arregle. don puedo ver si esta creado el callback y no lodeja crear. revise agent.conf y esta creado el agente. gracias, por su colaboracion. Wed, 22 May 2013 00:06:11 -0500 Subject: Babytel SIP 401 Unauthorized incoming call problem - by: kokimbochis http://elastix.org/index.php/en/component/kunena/3-help/100492-babytel-sip-401-unauthorized-incoming-call-problem.html#122266 I am having the same problem, but I do not want to enable anonymous incoming SIP calls because I read it's opens a vulnerability to hackers: http://www.elastix.org/index.php/en/component/kunena/116-security/49972-how-to-lock-your-system.html Let me know if you find a workaround. Tue, 21 May 2013 23:27:44 -0500 Subject: Cron Error Info - by: Amphibian http://elastix.org/index.php/en/component/kunena/3-help/122260-cron-error-info.html#122260 Hey Team, I just noticed within my cron log file I keep seeing this message: May 21 21:56:17 pbx crond[3082]: (CRON) STARTUP (V5.0) May 21 21:56:17 pbx crond[3082]: (*system*) BAD FILE MODE (/etc/cron.d/postfix_stats.cron) May 21 21:56:17 pbx crond[3082]: (*system*) WRONG FILE OWNER (/etc/cron.d/a2billing.cron) May 21 21:58:01 pbx crond[3544]: (asterisk) CMD (/var/lib/asterisk/bin/freepbx-cron-scheduler.php) As one can see that there is a "bad file mode" and a "wrong file owner"..... Guess I should have caught this sooner but I usually don't worry much with cron log files as I should. Could someone advise what the file mode should be and whom should the file owner be? I'm not sure about the bad file mode but shouldn't the file owner for a2billing be 755? It be showing 644 now. Thanks amphibian Tue, 21 May 2013 22:19:47 -0500 Subject: IVR NO ENTRA ELASTIX AXTEL CONMIGO FIBRA OPTICA - by: milocheri http://elastix.org/index.php/en/component/kunena/50-novatos/122194-ivr-no-entra-elastix-axtel-conmigo-fibra-optica.html#122253 Pega tu CLI para saber que es lo que esta pasando en tu caja Tue, 21 May 2013 17:30:02 -0500 Subject: usuarios, grupos y privilegios - by: voliva http://elastix.org/index.php/en/component/kunena/50-novatos/121827-usuarios-grupos-y-privilegios.html#122251 si si Ya he probado 3 nevagadores diferentes Tue, 21 May 2013 16:39:24 -0500 Subject: Complejidad contraseñas exntesiones SIP - by: Curioso http://elastix.org/index.php/en/component/kunena/11-ayuda/103370-complejidad-contrasenas-exntesiones-sip.html#122249 En el free pbx ya con la version 2.4 desactive el modulo Weak Password Detection pero sin embargo la contraseña sigue muy compleja alguna sugerencia? Tue, 21 May 2013 16:26:07 -0500 Subject: Call center gui - by: jgutierrez http://elastix.org/index.php/en/component/kunena/19-call-center/122057-call-center-gui.html#122243 Before changing any logos and credits for the developers, read about GNU License v2, Elastix uses it. Tue, 21 May 2013 13:53:23 -0500 Subject: mi asterisk se cuelga - by: jgutierrez http://elastix.org/index.php/en/component/kunena/57-asterisk/122039-mi-asterisk-se-cuelga.html#122242 Parecería que se trata de un problema de nat. Revisa algunos posts referentes a ese tema. Tue, 21 May 2013 13:46:45 -0500 Subject: elastix 3.0.0 alpha, elastix 2.4.0 beta, melastix - by: mustardman http://elastix.org/index.php/en/component/kunena/20-elastix-community-/110181-elastix-300-alpha-elastix-240-beta-melastix.html?limit=10&start=20#122240 Bottom line is that Elastix is still on Asterisk 1.8 and FreePBX v2.8 with nothing else coming along any time soon. Meanwhile everyone else has moved on to v1.11 and v2.11 with too many improvements and enhancements to go into. The FreePBX ecosystem of 3rd party add ons is quite large. As is community involvement documenting things. Working on fixes etc. Elastix has fallen very very very far behind and people will judge for themselves. I don't see what fantasy land they are in thinking they can go it alone and somehow catch up or improve things or keep up with features and fixes of the FreePBX community. I just don't see it happening. I don't know who determines the direction of Elastix but I think they are smart enough to know this too. I don't think we are getting the whole story and there must be some other motivation or perhaps they want the Elastix project to die. Who knows. Tue, 21 May 2013 11:54:52 -0500 Subject: ELASTIX 2.2 GRABACIONES SE CORTAN AL TRANSFERIR - by: Curioso http://elastix.org/index.php/en/component/kunena/50-novatos/116490-elastix-22-grabaciones-se-cortan-al-transferir.html#122239 Efectivamente con la version elastix 2.4 el problema ya quedo resuelto, lo estoy usando ahora. Tue, 21 May 2013 10:15:53 -0500 Subject: fail2ban elastix2.4.0 - by: rminier http://elastix.org/index.php/en/component/kunena/116-security/122238-fail2ban-elastix240.html#122238 hi everyone!! i ' m trying to install fail2ban on elastix 2.4.0 i can setup for ssh but for asterisk for blocking registration i can't. the fail2ban version is "fail2ban-0.8.4" jail.conf to configure [asterisk-iptables] like this [asterisk-iptables] enabled = true filter = asterisk action = iptables-allports[name=ASTERISK, protocol=all] sendmail-whois[name=ASTERISK, dest=root@localhost, sender=fail2ban@pbx.dyndns.org] logpath = /var/log/asterisk/messages maxretry = 5 bantime = 1800 the [asterisk.conf] like this # Fail2Ban configuration file # # # $Revision: 250 $ # [INCLUDES] # Read common prefixes. If any customizations available -- read them from # common.local #before = common.conf [Definition] #_daemon = asterisk # Option: failregex # Notes.: regex to match the password failures messages in the logfile. The # host must be matched by a group named "host". The tag "" can # be used for standard IP/hostname matching and is only an alias for # (?:::f:)?(?P\S+) # Values: TEXT # failregex = NOTICE.* .*: Registration from '.*' failed for '' - Wrong password NOTICE.* .*: Registration from '.*' failed for '' - No matching peer found NOTICE.* .*: Registration from '.*' failed for '' - Username/auth name mismatch NOTICE.* failed to authenticate as '.*'$ NOTICE.* .*: No registration for peer '.*' (from ) NOTICE.* .*: Host failed MD5 authentication for '.*' (.*) # Option: ignoreregex # Notes.: regex to ignore. If this regex matches, the line is ignored. # Values: TEXT # ignoreregex = [logger.conf] I have included this "messages => notice, debug" i think my problem is the logpath = /var/log/asterisk/messages I'm not sure if I'm directing well log thank a lot. Tue, 21 May 2013 10:07:12 -0500 Subject: Root Password - by: Bob http://elastix.org/index.php/en/component/kunena/3-help/122229-root-password.html#122235 Michaeltagui, Can anybody help to break root password of elastix. Do you mean Linux root or Elastix admin password.....?? Regards Bob Tue, 21 May 2013 08:55:24 -0500 Subject: Recorded files by gsm - 33 and 66 bytes only - by: soborno http://elastix.org/index.php/en/component/kunena/3-help/102381-recorded-files-by-gsm-33-and-66-bytes-only.html#122226 I´m again with this matter, the only new thing I can point out it´s that the recordings are made by extensions in a simple queue (that have always/always set in each extension) that is the only destination of an Inbound Route of an E1. If someone, has a clue, It will really help me. Regards, Claudio Tue, 21 May 2013 07:03:17 -0500 Subject: Agent Console login error - by: pulak http://elastix.org/index.php/en/component/kunena/19-call-center/121808-agent-console-login-error.html#122223 Hi, Did you check the table (/var/www/db/acl.db) using SQlite manager. is there user exist or not? Thanks Tue, 21 May 2013 06:49:00 -0500 Subject: Affichage du numéro SDA de l'appellant - by: danardf http://elastix.org/index.php/en/component/kunena/84-le-coin-du-debutant/119458-affichage-du-numero-sda-de-lappellant.html?limit=10&start=10#122221 Cliques ICI (http://www.elastix.org/index.php/en/component/kunena/3-help/118735-missing-0-from-isdn-caller-id.htm) Tue, 21 May 2013 06:35:38 -0500 Subject: How to show unicode call information during callig - by: pulak http://elastix.org/index.php/en/component/kunena/19-call-center/122218-how-to-show-unicode-call-information-during-callig.html#122218 Hi, I have uploaded the csv file of call information which contain phone number,first name,last name in the "outgoing calls" Tab. after uploading the csv file it will store successfully in to MySql Database[field set as CHARACTER SET utf8 COLLATE utf8_unicode_ci). But when I am trying to campaigning from "Agent Console" Tab call information comes one after another but content not showing properly. (showing the First Name: ??????? and Last Name:??????). [ when uploading the csv file before that I have run the following commands for encoding as follows: $this->_DB->genQuery('SET character_set_results=utf8'); $this->_DB->genQuery('SET names=utf8'); $this->_DB->genQuery('SET character_set_client=utf8'); $this->_DB->genQuery('SET character_set_connection=utf8'); $this->_DB->genQuery('SET character_set_results=utf8'); $this->_DB->genQuery('SET collation_connection=utf8_general_ci'); ] But In case of fetching the data from the call_attributes table in call_center database on that time they are using ECCP connection instead of mysql query. That's why I am not able to set character encoding before collect the attributes through getcallinfo() function. Do you have any Idea how can I resolved the problem? Regards, Tue, 21 May 2013 06:02:09 -0500 Subject: call center - by: tresor http://elastix.org/index.php/en/component/kunena/79-general/122217-call-center.html#122217 bonjour svp quel type de trunk doit-je configurer en local entre deux machines pour tester les appels vers un centre d'appel précédemment mis sur pied merci pour vos suggestion et reponses Tue, 21 May 2013 06:00:27 -0500 Subject: Hacking through ISDN line - by: Bob http://elastix.org/index.php/en/component/kunena/3-help/122171-hacking-through-isdn-line.html#122215 ipwn, To the best of my knowledge, unless your ISDN is providing IP connectivity to the Internet( 1 channel voice/1 channel data), then you are reasonably safe. Happy to be proven wrong on this one, but in all my experience, other than someone phreaking a pbx or utilising a system with DISA enabled (recipe for disaster), I have never heard of hacking via the ISDN line. Further on your question, if you do not open your router ports to the Elastix box, then you are 95% safe.... I leave the 5% for the following factors 1) Poor password complexity 2) Possible trojan attacks from inside the LAN (none known but did a proof of concept and is definitely possible) 3) other devices on the Local area network hacked (including workstations) that then provide access to the Elastix box. I talk about that 5%, but most of this can be mitigated by 1) Elastix firewall (only allowing SSH for instance from your desktop only) 2) Fail2ban - alerting you to the fact that someone is trying to connect to your box Don't panic with the hacking issues, but read everything that you can on IP PBX security, think of your own setup, and don't lock it so tight that it does nothing for you and makes everything to hard, but the risks that you do see, mitigate them. Regards Bob Tue, 21 May 2013 05:27:00 -0500 Subject: Elastix database ? - by: Bob http://elastix.org/index.php/en/component/kunena/25-newbies-corner-/121712-elastix-database-.html#122210 makadarel, Great to hear that you found what you need. On your question, I cannot provide a definitive answer as I have not done any in-depth work with the AsteriskCDRDB database. However, I believe other than being a repository for all call records it has no bearing on other functionality of the pure Asterisk system to the best of my knowledge. This is further backed up my experience with a few systems that fail to populate the CDR database (usually after a poor yum update - a long while back now), and Asterisk and other systems continue to run as normal. That said, other systems look at the database for reports, both internally in Elastix, and also add-on programs such as A2billing etc... Regards Bob Tue, 21 May 2013 05:03:50 -0500 Subject: Missing 0 from ISDN Caller ID - by: dnlvry http://elastix.org/index.php/en/component/kunena/3-help/118735-missing-0-from-isdn-caller-id.html#122205 Thanks Bob, that has got the leading 0 working. How do I get a 9 to appear before the number on incoming calls so it's easy to redial? 9 needs to be dialled to get an external line. Tue, 21 May 2013 04:30:20 -0500 Subject: Centralita no detecta que han descolgado remotamen - by: juande007 http://elastix.org/index.php/en/component/kunena/50-novatos/122199-centralita-no-detecta-que-han-descolgado-remotamen.html#122199 Buenos dias a todos. Desde hace unos dias mi elastix no detecta que han descolgado la llamada en el destino. Yo escucho al cliente pero ellos a mi no y pasado un par de minutos elastix tira la llamada. Se le ocurre a alguien lo que puede pasar? Gracias... Tue, 21 May 2013 03:37:54 -0500 Subject: freepbx and elastix - by: Itsm http://elastix.org/index.php/en/component/kunena/3-help/122190-freepbx-and-elastix.html#122196 Hello, It happened to me once, it's being caused due to a mismatch of passwords/user between what's in your DB and the one's in your config files. Please make sure that it matches in the files "amportal.conf & manager.conf" and in the SQL DB (look at for the SQL DB http://www.debian-administration.org/articles/442) for future reference, don't update from the freepbx, learn how to yum update, and only the elastix. Good Luck. Tue, 21 May 2013 02:19:36 -0500 Subject: SCCP - No audio for calls to VM - by: danardf http://elastix.org/index.php/en/component/kunena/3-help/122075-sccp-no-audio-for-calls-to-vm.html#122193 From my side, i just enabled TCP/UDP port #2000 from iptables and all works fine. Maybe an old issue exist yet under cyrus config which could be conflictual. (using the same port 2000) Mon, 20 May 2013 23:15:47 -0500 Subject: One Way connection - hearing voice at one side - by: raj2013 http://elastix.org/index.php/en/component/kunena/3-help/122191-one-way-connection-hearing-voice-at-one-side.html#122191 Hi All, We are Using the Elastix Easy with Xorcom 2000 Server and implemented the voip with internet. Configured the extension users and setup ready for use now,but when i make calls one extn to another extn, can able to hear my voice in another end,but not able to hear the voice in my side. Sometimes when i make calls it is just ringing and automatically close the call Please help me to fix the issue. let me know if any dubugging tools available for troubleshooting in elastix easy Thank you Mon, 20 May 2013 22:22:13 -0500 Subject: no ingresan llamadas a la troncal sip - by: jogs87 http://elastix.org/index.php/en/component/kunena/50-novatos/122186-no-ingresan-llamadas-a-la-troncal-sip.html#122186 Buen día He montado una troncal Sip, mi proveedor de telefonía me asigno la ip de la planta, gateway,ip de la troncal y numero de cabecera he realizado la rutas necesarias para el enrutamiento necesario Destination Gateway Genmask Flags Metric Ref Use Iface ip troncal gateway 255.255.255.255 UGH 0 0 0 eth1 red de la troncal 0.0.0.0 255.255.255.252 U 0 0 0 eth1 192.168.10.0 0.0.0.0 255.255.255.0 U 0 0 0 eth0 169.254.0.0 0.0.0.0 255.255.0.0 U 0 0 0 eth0 la configuración de mi troncal sip PEER DETAIL type=peer qualify=yes host=ip troncal contex=from-trunk careinvite=no relaxdtmf=yes dtmfmde=rfc2833 t38pt_udptl=yes nat=yes User Details type=friend qualify=yes host=ip troncal context=from-trunk canreinvite=no insecure=yes relaxdtmf=yes dtmfmode=rfc2833 t38pt_udptl=yes nat=yes se realizan las rutas de salida con marcado 6|NXXXXXX al momento de realizar las pruebas, puedo realizar salida de llamadas sin ningún problema. pero al momento de entrada de llamadas nos da un mensaje que el numero marcador (numero de cabecera dado por el proveedor) se encuentra fuera de servicio y lo deletrea con 571+numero de cabecera dado por el proveedor. reporte esta falla con el proveedor pero ellos informan que el error se encuentra en la configuración de la planta, en la cual no he definido ningún prefijo con 571.. si alguno de ustedes presento el mismo error, les agradecería informar ya que en estos momentos solo tengo salida de llamadas por la troncal SIP y recepción de llamadas por linea una linea análoga que tengo conectada a la planta. Mon, 20 May 2013 16:24:12 -0500 Subject: Elastix & Fail2ban - by: gtec http://elastix.org/index.php/en/component/kunena/3-help/78742-elastix-a-fail2ban.html#122184 Hi, I have installed fail2ban as mentioned in most of the sites and everything fine until restart. I am getting a mail like : ************************************ Hi, The jail ASTERISK has been started successfully. Regards, Fail2Ban **************************************** Then I tried to register the extension using wrong password 5 times(In my jail.conf it is 5) but still not blocking my IP. Following are my configurations : vi /etc/fail2ban/filter.d/asterisk.conf ********************************************* # Fail2Ban configuration file # # # $Revision: 250 $ # [INCLUDES] # Read common prefixes. If any customizations available -- read them from # common.local #before = common.conf [Definition] #_daemon = asterisk # Option: failregex # Notes.: regex to match the password failures messages in the logfile. The # host must be matched by a group named "host". The tag "" can # be used for standard IP/hostname matching and is only an alias for # (?:::f:)?(?P\S+) # Values: TEXT # failregex = NOTICE.* .*: Registration from '.*' failed for ':.*' - Wrong password NOTICE.* .*: Registration from '.*' failed for ':.*' - No matching peer found NOTICE.* .*: Registration from '.*' failed for ':.*' - No matching peer found NOTICE.* .*: Registration from '.*' failed for ':.*' - Username/auth name mismatch NOTICE.* .*: Registration from '.*' failed for ':.*' - Device does not match ACL NOTICE.* .*: Registration from '.*' failed for ':.*' - Peer is not supposed to register NOTICE.* .*: Registration from '.*' failed for ':.*' - ACL error (permit/deny) NOTICE.* .*: Registration from '.*' failed for ':.*' - Device does not match ACL NOTICE.* .*: Registration from '\".*\".*' failed for ':.*' - No matching peer found NOTICE.* .*: Registration from '\".*\".*' failed for ':.*' - Wrong password NOTICE.* failed to authenticate as '.*'$ NOTICE.* .*: No registration for peer '.*' \(from \) NOTICE.* .*: Host failed MD5 authentication for '.*' (.*) NOTICE.* .*: Failed to authenticate user .*@.* NOTICE.* .*: failed to authenticate as '.*' NOTICE.* .*: tried to authenticate with nonexistent user '.*' VERBOSE.*SIP/-.*Received incoming SIP connection from unknown peer # Option: ignoreregex # Notes.: regex to ignore. If this regex matches, the line is ignored. # Values: TEXT # ignoreregex = ********************************************* vi /etc/fail2ban/jail.conf ******************************************** DEFAULT] ignoreip = 127.0.0.1 ******my ips********** [asterisk-iptables] enabled = true filter = asterisk action = iptables-allports[name=ASTERISK, protocol=all] sendmail-whois[name=ASTERISK, dest=*****, sender=fail2ban@******] logpath = /var/log/asterisk/full maxretry = 5 bantime = 259200 ******************************************** vi /etc/asterisk/logger.conf ******************************************** ; ; Logging Configuration ; ; In this file, you configure logging to files or to ; the syslog system. ; ; For each file, specify what to log. ; ; For console logging, you set options at start of ; Asterisk with -v for verbose and -d for debug ; See 'asterisk -h' for more information. ; ; Directory for log files is configures in asterisk.conf ; option astlogdir ; [general] dateformat=%F %T [logfiles] ; ; Format is "filename" and then "levels" of debugging to be included: ; debug ; notice ; warning ; error ; verbose ; ; Special filename "console" represents the system console ; ;debug => debug ; The DTMF log is very handy if you have issues with IVR's ;dtmf => dtmf ;console => notice,warning,error ;console => notice,warning,error,debug ;messages => notice,warning,error full => notice,warning,error,debug,verbose ;syslog keyword : This special keyword logs to syslog facility ; syslog.local0 => notice,warning,error ; ******************************************** I can see the wrong password log in my /var/log/asteris/full : [2013-05-20 16:10:26] NOTICE[3575] chan_sip.c: Registration from '"won"' failed for 'IP:23603' - Wrong password But the IP is not blocking by fail2ban !!!! Please help :( Thanks, G Mon, 20 May 2013 15:45:49 -0500 Subject: Comunicacion entre troncales - by: parisito http://elastix.org/index.php/en/component/kunena/67-tarjetas-de-telefonia/108143-comunicacion-entre-troncales.html#122181 Tengo el mismo problema, He conectado dos elastix en diferentes localidades, Cuando le doy iax show peers aparece Name/Username Host Mask Port Status name ame XXX.XXX.XXX.XXX (S) 255.255.255.255 4569 (T) OK (172 ms) 1 iax2 peers [1 online, 0 offline, 0 unmonitored] En los dos aparecen sus respectivos ips pero cuando trato realizar la llamada de una a otra localidad me dice: "Todas las lineas están ocupadas por favor intentelo más tarde". He leido de todo en el foro y no logro solucionarlo. Si alguien pudiera ayudarme lo agradeceré. Mon, 20 May 2013 15:07:39 -0500 Subject: Polycom vvx1500 support - by: macmann http://elastix.org/index.php/en/component/kunena/44-polycom/122055-polycom-vvx1500-support.html#122180 I would be interested to know which features you have working currently with the version you have. Intercom? Park? Etc... Mon, 20 May 2013 14:53:16 -0500 Subject: Configuracion Troncal Sip autenticacion por ip - by: milocheri http://elastix.org/index.php/en/component/kunena/11-ayuda/122176-configuracion-troncal-sip-autenticacion-por-ip.html#122179 Tienes IP Publica fija ? Si es asi, solo tienes que darle la direccion IP a tu proveedor, te conviene que la autenticacion sea por IP, eso lo hacen los providers generalmente para que la cuenta no pueda ser usada con otra direccion IP mas que con la que tiene tu central, el registro de la cuenta es similar a cuando usa usuario y contraseña, salvo que en este caso en el "Register String:" se deja vacio, en mi caso la troncal queda asi PEER Details [code]type= username= fromuser= secret= host= context=from-trunk disallow=all allow=ulaw;g729;gsm qualify=yes nat=never insecure=invite[/code] USER Details [code]type=peer context=from-trunk host= [/code] Espero te ayude, saludos Mon, 20 May 2013 14:45:31 -0500 Subject: SIP trunk addition issues *** HELP *** - by: Amphibian http://elastix.org/index.php/en/component/kunena/25-newbies-corner-/121948-sip-trunk-addition-issues--help-.html#122177 donovanpl123, After you install Elastix did you do an update? I know I had some issues with Elastix 2.3.05 and once I installed 2.3.08 problems were solved. I would prob do the update and then see if prob still exsist. You can search several post on here on how to do the update. amphibian Mon, 20 May 2013 14:12:27 -0500 Subject: Les appels entrants sont repondu automatiquement - by: danardf http://elastix.org/index.php/en/component/kunena/103-openvox/120672-les-appels-entrants-sont-repondu-automatiquement.html#122173 De rien. ;) Mon, 20 May 2013 11:04:15 -0500 Subject: Elastix with Avantfax Virtual Host Issue - by: yaseen659 http://elastix.org/index.php/en/component/kunena/39-others/122169-elastix-with-avantfax-virtual-host-issue.html#122169 Dear all, Please help me out I have installed elastix and avantfax-3.3.3 I have a issue with virtual host. 192.168.1.X== elastix is opening and 192.168.1.x:4443 avantfax is not opening ... I tried opening X4443/admin and /index.php But noting is working Please help in this matter Mon, 20 May 2013 09:49:10 -0500 Subject: Elastix-2.4 stable in chinese released! - by: james.zhu http://elastix.org/index.php/en/component/kunena/6-translations/122168-elastix-24-stable-in-chinese-released.html#122168 hello: Elastix-2.4 stable in chinese released! please refer this: http://www.hiastar.com/index.php/2011-04-19-18-57-12/2011-04-19-18-57-49/22-wendang/113-elastix-cn Mon, 20 May 2013 09:22:17 -0500 Subject: Deleting from monitoring list - by: soborno http://elastix.org/index.php/en/component/kunena/3-help/30205-deleting-from-monitoring-list.html?limit=10&start=10#122166 What´s your specific problem? Regards, Claudio Mon, 20 May 2013 08:56:15 -0500 Subject: Ayuda en configuracion de Elastix - by: soborno http://elastix.org/index.php/en/component/kunena/50-novatos/122115-ayuda-en-configuracion-de-elastix.html#122165 El problema puntualmente se da ya que en el campo 'ID' se debe colocar el número de la extension. Saludos, Claudio Mon, 20 May 2013 08:52:45 -0500 Subject: LSI MegaRaid 9220 issue - by: Bob http://elastix.org/index.php/en/component/kunena/1-installation-issues/121594-lsi-megaraid-9220-issue.html#122164 johnatan, I cannot believe it a new Cisco rack server just simple not supported... There is no such thing as simple when it comes to RAID controllers and their drivers. It is not so much that Elastix does not support it, but the underlying O/S. As far as I know Redhat 5.7, through to 6.1 does not have support for the raid controller you mention. I have heard of users installing Centos 5.8 adding their special drivers and then installing Elastix after this.....Someone on this forum may have tried this alread and can give you the answer.... Basically all the iso does is : 1) Install the O/S including partitioning 2) runs a script which loads the rest.... You might want to try and look at the ISO and how it runs. I personally have not tried it, but in theory it is a possibility, just make sure that Linux OS is the same as the one expected by the script Regards Bob Mon, 20 May 2013 08:22:13 -0500 Subject: H264 via Elastix - by: Bob http://elastix.org/index.php/en/component/kunena/25-newbies-corner-/122078-h264-via-elastix.html#122159 bswash, hopefully I will get a chance to get some testing done as well during this week look forward to your results Regards Bob Mon, 20 May 2013 07:43:32 -0500 Subject: Not able to hear the Voice - One Way Voice - by: raj2013 http://elastix.org/index.php/en/component/kunena/31-asterisk/122145-not-able-to-hear-the-voice-one-way-voice.html#122145 Hi All, We are Using the Elastix Easy with Xorcom 2000 Server and implemented the voip with internet. Configured the extension users and setup ready for use now,but when i make calls one extn to another extn, can able to hear my voice in another end,but not able to hear the voice in my side. Sometimes when i make calls it is just ringing and automatically close the call Please help me to fix the issue. let me know if any tools available for troubleshooting purpose in elastix easy Thank you Mon, 20 May 2013 03:31:39 -0500 Subject: Video Conferencing through Elastix - by: danardf http://elastix.org/index.php/en/component/kunena/3-help/122138-video-conferencing-through-elastix.html#122143 Sorry. I haven't any experiences about that. So, i could not help you I know a little bit Asterisk 1.8, but not the version 10 and 11. However, about WEBRTC, it seems possible (i think). Try to find some thread into this forum. Otherwise, if it's doable on Asterisk 1.8, then it's doable under Elastix. Mon, 20 May 2013 02:43:15 -0500 Subject: Outgoing campaign, simultaneous calls - by: nana_MeDBeD http://elastix.org/index.php/en/component/kunena/19-call-center/122135-outgoing-campaign-simultaneous-calls.html#122135 Hello everybody ! Have installed and working Elastix 2.3 with Call Center 2.2. Agent use call back login. I use "outgoing campaign" and issue is how limit number of simultaneous calls. When i create campaigh i specify "Max. used channels" = 1, but when campaign started i see in log that "call center" started next call when 1st is in queue yet(or talk with agent). Check\uncheck box "Enable overcommit of outgoing calls" and "Enable predictive dialer behavior" have no result. Sorry for my bad English. Mon, 20 May 2013 00:15:04 -0500 Subject: Incoming call: Got SIP response 400 "Bad Request" - by: phqr58 http://elastix.org/index.php/en/component/kunena/31-asterisk/92863-incoming-call-got-sip-response-400-qbad-requestq.html#122134 I have the same problem with a fresh installation, elastix 2.4 This shows whether or not you call in progress asterisk -r - Got SIP response 400 "Bad Request" back from 10.18.15.150:5060 - Remote UNIX connection - Remote UNIX connection disconnected - Got SIP response 400 "Bad Request" back from 10.18.15.150:5060 pbx1*CLI> someone found the solution? pbx1*CLI> sip show registry Host dnsmgr Username Refresh State Reg.time 10.18.15.150:5060 N 59343811744 120 Request Sent 1 SIP registrations. pbx1*CLI> in Reg.time is blank ??? Mon, 20 May 2013 00:13:45 -0500 Subject: Issue with Elastix Web Session - Not working! - by: Aiiar http://elastix.org/index.php/en/component/kunena/25-newbies-corner-/103277-issue-with-elastix-web-session-not-working.html#122133 Can u tell me how to delete the recorded voicemail to free some space using putty, while I cant login using web browser. Thanx Mon, 20 May 2013 00:09:30 -0500 Subject: Creating a new ssl certificate - by: D3VIATION http://elastix.org/index.php/en/component/kunena/3-help/1083-creating-a-new-ssl-certificate.html#122132 This works. Thanks Mon, 20 May 2013 00:00:51 -0500 Subject: Agent Callback Logon - by: nana_MeDBeD http://elastix.org/index.php/en/component/kunena/19-call-center/122131-agent-callback-logon.html#122131 Hello everyone ! how to register Agents from phone, without web interface, with full reports in call-center section of elastix? Mon, 20 May 2013 00:00:23 -0500 Subject: OpenVox A1200P/A800P No voice - Noise problem - by: upperhua http://elastix.org/index.php/en/component/kunena/13-openvox/119859-openvox-a1200pa800p-no-voice-noise-problem.html#122129 HI Do not know if the problem has been solved or not, If still exists, I think you need to reinstall the driver to have a try , You can download it from this link http://downloads.openvox.cn/pub/drivers/dahdi-linux-complete/releases/1.3.5/openvox_dahdi-linux-complete-2.4.1.2+2.4.1.tar.gz You can add my Skype upper.hua . I will check it for you. Best regards upper Sun, 19 May 2013 22:44:32 -0500 Subject: help - by: danardf http://elastix.org/index.php/en/component/kunena/3-help/122113-help.html#122125 Hi You can set externnotify : Want to run an external program whenever a caller leaves a voice mail message for a user? This is where the externnotify command comes in handy. Externnotify takes a string value which is the command line you want to execute when the caller finishes leaving a message. Note: see an example of an external notification script here. Note: This command will also run after a person who has logged into a mailbox exits the VoiceMailMain() application. (Remark: This seems not to be the case for Asterisk 1.2.x) Took on this link: http://www.voip-info.org/wiki/view/Asterisk+config+voicemail.conf And next with your script, you could use Asterisk auto dial for calling the extension. Regards Sun, 19 May 2013 21:54:28 -0500 Subject: endpoint - by: c3b4Xx http://elastix.org/index.php/en/component/kunena/11-ayuda/122121-endpoint.html#122122 La version 2.4 ya puesdes hacer la configuracion para este tipo de telefono yealink Suerte !! Sun, 19 May 2013 21:45:31 -0500 Subject: No corta la llamada en Grandst HT503 (Solucionado) - by: Guzmanweb http://elastix.org/index.php/en/component/kunena/48-historias-de-/122117-no-corta-la-llamada-en-grandst-ht503-solucionado.html#122117 Muchas idas y vueltas con el equipo decidí documentarlo lo primero que hice busque por todo los foros que pueda pero no encontré lo que buscaba, el problema que tenia era que cunado me llamaban de afuera a mi teléfono fijo y me encontré con un usuario en foro de Grandstream, pasaban los datos, desde ese momento cree un Ticket de soporte en el día tuve la respuesta, fueron muy amables los de soporte de Grandstream, me ayudaron con mi problema aqui les comparto los paso a seguir son: Primero: Capturar la llamada entrante y tomar el trace usando PCMU, PCMA. tcpdump -i “nombre-de-la-interfaz-de-red-que-usa” -s 1500 -w “Nombre-del-archivo.pcap” Inicie esta captura antes de hacer la llamada, y siga las instrucciones, La captura la puedes hacer usando tcpdump desde el cli del Asterisk, el proceso para tomarlo debe ser el siguiente, 1) hacer una llamada desde un numero de externo a la linea del HT503, 2) responder esta llamada por un interno del servidor. 3) colgar la llamada desde la PSTN 4) el interno no debe colgar el auricular, debe esperar al menos unos 20s antes de colgar 5) luego esto envíenos el archivo .pcap de la captura y le indicaremos los tonos correctos de colgado de su proveedor PSTN. Segundo: Una ves que tengas el archivo de captura.pcap enviar a soporte de Grandstream, y ellos te pasan los parámetros correctos para que corte la llamada entrante. Ajuste el valor PSTN Disconnect Tone: en la pestaña FXO port PSTN Disconnect Tone:f1=424@-19,f2=424@-19,c=505/505; estos parámetros son del proveedor Coperativa de COTEL La Paz – Bolivia y verifique el siguiente parametro Enable Current Disconnect: No Enable PSTN Disconnect Tone Detection: YES Para el caso del caller ID establezca el valor Number of Rings: en 4 luego de esto si no funciona por favor prueba valor a valor en Impedance-based DID, CID El DID en el ATA se configura en Unconditional Call Forward to VOIP: ahi debes colocar numeros distintos si deseas crear una ruta entrante para cada ata, Cuando agregues la ruta debes establecer los DID que configuraste en los atas. Para crear la ruta de llamadas entrantes desde colocar DID difeentes, cuando vas a crearla llena el campo DID, CID con el valor que configures en el ATA, cada ATA debe tener un valor diferente, En cuanto al caso del fax, tienes dos opciones, 1.- crear una extensión en el ata, crear un IVR para que la llamada sea pasada a la extensión del FAX, 2.- crear una ruta entrante solo para la extensión que tengas en el puerto FXS del ata donde tendrás e fax conectado espero que les sirva, no olvide de comentar. Saludos Ricardo Sun, 19 May 2013 17:05:19 -0500 Subject: llamadas entrantes por dinstar gsm se cortan - by: juancag http://elastix.org/index.php/en/component/kunena/11-ayuda/122111-llamadas-entrantes-por-dinstar-gsm-se-cortan.html#122111 Saludo cordial, permitanme hacerles una pregunta, yo llevé a cabo un tutorial posteado en este sitio relacionado con la configuracion de un DWG , las llamadas salen perfectamente , mi problema radica con las llamadas entrantes, ya que cuando entran las recibe el ivr , pero cuando este las direcciona a la extencion de destino se cuelga la llamada o cuando se hace la marcacion directa de la extencion tambien se cuelga.. Saben a que se debe??? version del gateway dinstar DWG2000 1G version de elastix 2.4 Sun, 19 May 2013 13:50:06 -0500 Subject: Cannot register Cisco 7942 to Asterisk - by: jwright20 http://elastix.org/index.php/en/component/kunena/31-asterisk/74453-cannot-register-cisco-7942-to-asterisk.html?limit=10&start=50#122110 Hello any news on this??? I'm trying to register a Cisco 7965 with firmware SIP45.9-3-1SR2-1S. The phone sees the TFTP and was able to download the firmware but somewhere in the SEPMACADDRESS.cnf.xml there may be an issue. When I look at the phone is give the following message No trusted list installed Error verifying config file I was trying to do a sip debug to see the issue by running sip set debug ip 192.168.20.xxx (ip address of phone) but nothing was listed in the CLI nor in the asterisk full log. What am I doing wrong? Can someone send me a simple configuration file that works? Desperately in need. J Sun, 19 May 2013 11:48:52 -0500 Subject: FaX using ATA in Elastix - by: navastele http://elastix.org/index.php/en/component/kunena/25-newbies-corner-/116203-fax-using-ata-in-elastix.html#122109 Hi All I have brand New AG198 its not working IAX2 registered with the Elastix PBX , what ever you change in the IAX config page its still shows unapplied , strange Sun, 19 May 2013 11:05:51 -0500 Subject: How about ZTE MF180 using with Elastix?! - by: Oleg B http://elastix.org/index.php/en/component/kunena/25-newbies-corner-/122106-how-about-zte-mf180-using-with-elastix.html#122106 Hi All! More then half of year ago i asked you about using ZTE MF180 USB 3G modem with Elastix. As we know Huawei E1550 works very well. But i couldn't find E1550 any more at my area. More then half of year ago i've got 4 ZTE MF180 modems by very small price. I hoped could use it in my future Elastix projects. Sorry, i still can't use ZTE MF180 USB 3G modems:-( There are no new versions of chan_dongle and modeswitch programms since november 2012:-( I mean no new devices support appears since that time:-( Maybe somebody have solve same problem already? Please help me! Sun, 19 May 2013 10:07:17 -0500 Subject: Elastix 2.4 no detecta red ni tarjeta fxs fxo - by: Bob http://elastix.org/index.php/en/component/kunena/76-digium/122031-elastix-24-no-detecta-red-ni-tarjeta-fxs-fxo.html#122103 Rensi, Muy bueno oír que funciona! Gracias por probar lo que me propuse y por sus comentarios. Esto ayudará a los demás Saludos Bob Sun, 19 May 2013 09:07:24 -0500 Subject: manual de a2billing - by: chimo http://elastix.org/index.php/en/component/kunena/59-a2billing/119097-manual-de-a2billing.html#122100 Te aconsejo este manual que es muy completo. www.voztovoice.org/?q=node/471 Sun, 19 May 2013 04:46:02 -0500 Subject: Integration with Zimbra 8 - by: Bob http://elastix.org/index.php/en/component/kunena/5-new-features/122092-integration-with-zimbra-8.html#122096 damirabal, As a community member, I have no involvement in what goes into Elastix. However, I believe the features you talk about, especially the UC features in Zimbra 8, only come in the commercial version....happy to proven otherwise... Regards Bob Sun, 19 May 2013 02:58:31 -0500 Subject: Add sub menu item - by: rosonery http://elastix.org/index.php/en/component/kunena/27-miscellaneous/99757-add-sub-menu-item.html#122093 it's very useful and i send much thanks to you bro ... :) Sun, 19 May 2013 01:20:05 -0500 Subject: sound card related Kernel panic - by: Bob http://elastix.org/index.php/en/component/kunena/3-help/121845-sound-card-related-kernel-panic.html#122085 farzad741, I understand what you mean.... However, just in case someone has some ideas and allow further input....could you post the following from the linux commands lsusb (so we have the device number of the USB device) lspci (so we have the USB Controller in your machine) uname -a Also expand your own search on google and particularly look for the following usb centos kernel panic particularly you are looking for posts 1-2 years old.... Regards Bob Sat, 18 May 2013 18:55:38 -0500 Subject: Elastix 2.3.0 & Sperate Vtiger 5.4.0 server - by: Bob http://elastix.org/index.php/en/component/kunena/3-help/121928-elastix-230-a-sperate-vtiger-540-server.html#122083 ukez, Lets go back to basics....lets confirm that your AMI can be seen 1) From your Elastix box Use telnet on port 5038 and connect to the local machine. You should see Asterisk Call Manager/1.1 in the terminal screen 2) From your Vtiger box use telnet on Port 5038 and connect to your Elastix box You should see Asterisk Call Manager/1.1 in the terminal screen If all the above are ok you have confirmed the following 1) AMI is running 2) AMI is not being blogged by firewalls 3) Networking is working correctly Next, just to make sure authentication is working, you want to login again try locally and then from your VTiger box I wont try and give you all the commands...but the following will give you the info you need to try authentication. http://www.scribd.com/doc/54892783/374/AMI-over-TCP Look for the section called AMI over TCP and If you get a little stuck, there are plenty of other articles on AMI over TCP (don't wander into AMI via HTTP - you'll end up on another tangent)....Also check things like manager.conf if you wish but don't edit them manually, this is all done via the GUI and works (otherwise half the rest of the tools would fail) If you still have issues post back here... Regards Bob Sat, 18 May 2013 18:37:33 -0500 Subject: experiencia Elastix con equipos Dinstar FXO/FXS - by: juancag http://elastix.org/index.php/en/component/kunena/68-endpoints/113791-experiencia-elastix-con-equipos-dinstar-fxofxs.html#122082 Amigo, saludo cordial, permitame hacerle una pregunta, yo llevé a cabo tu tutorial, las llamadas salen perfectamente , mi problema radica con las llamadas entrantes ya que cuando entran las recibe el ivr , pero cuando este las direcciona a la extencion de destino se cuelga la llamada o cuando se hace la marcacion directa de la extencion tambien se cuelga.. Sabes a que se debe??? version del gateway dinstar DWG2000 1G Sat, 18 May 2013 18:31:41 -0500 Subject: BT / SIP trunk / VOIP providers - by: bswash http://elastix.org/index.php/en/component/kunena/27-miscellaneous/122079-bt--sip-trunk--voip-providers.html#122079 Hi, Anyone else using BT Business Infinity noticing SIP trunk registration problems ??? In the last couple of weeks I've been having trouble registering and or placing calls via my SIP trunk. At work and other locations it all works fine, just not on my BT connection. Was trawling the net and found this: http://www.theregister.co.uk/2013/04/30/bt_trolling_sip_in_battle_with_google/ http://www.btplc.com/Innovation/Licensing/Patentlicensingprogrammes/SIPTrunking/Formlicenceagreement.htm Is BT clamping down on SIP traffic / VOIP providers ??? Am considering moving to Virgin (sic) or another provider - any suggestions ??? Thanks, J :-) Sat, 18 May 2013 17:58:27 -0500 Subject: Chanspy; Listening In To All But These Extensions - by: ukez http://elastix.org/index.php/en/component/kunena/3-help/119917-chanspy-listening-in-to-all-but-these-extensions.html#122077 Itsm wrote: I have never seen a Chanspy configuration that actually works in order to filter extensions.. You can put a password, and even select through which exten. will it work from, but not to select to whom it may listen, I find FOP2 the best solution for those kind of things. Yeah I purchased FOP2 in the end and its money well spent if you ask me... I highly recommend it. Sat, 18 May 2013 17:15:45 -0500 Subject: Como "resucitar" telefono CISCO 7911 tras reseteo? - by: e.cruz http://elastix.org/index.php/en/component/kunena/11-ayuda/89701-como-qresucitarq-telefono-cisco-7911-tras-reseteo.html#122067 Estimado, si ha encontrado alguna solución, favor de ayudarme que estamos en la misma situación...gracias Sat, 18 May 2013 08:50:46 -0500 Subject: Impossible de se connecter à l'interface Asterisk - by: danardf http://elastix.org/index.php/en/component/kunena/86-question-installations/120726-impossible-de-se-connecter-a-linterface-asterisk.html?limit=10&start=20#122062 Ok merci de l'info et de l'avoir fait. Pour le suivi, je vais poster un mesage sur tout ticket de manière à être dans la boucle. Ne t'inquiètes pas pour çà. Bon week Sat, 18 May 2013 04:24:02 -0500 Subject: Elastix para administrar linea analogica - by: erikajulissa http://elastix.org/index.php/en/component/kunena/72-otros/121840-elastix-para-administrar-linea-analogica.html#122045 Muchas gracias por su respuesta. De hecho comprare una tarjeta análoga Digium, la intención es aplicar a esa línea todas las opciones posibles del servidor elastix. Saludos Fri, 17 May 2013 20:06:00 -0500 Subject: Modulo monitoring no muestra llamadas de queues - by: atrejo http://elastix.org/index.php/en/component/kunena/54-modulos/110997-modulo-monitoring-no-muestra-llamadas-de-queues.html?limit=10&start=10#122037 Yo tenia ese mismo problema corrientdo la 2.3 y segui la recomendacion en este foro de la actualizacion de elastix con el "[code]yum update elastix elstix-* asterisk asterisk-* dahdi dahdi-*[/code] pero al hacerlo, se afecto la parte de dahdi y ya no pude usar la T1 que tenia conectada. Termine por reinstalar toda la vesion 2.4 desde cero y con eso solucione el poder visualizar las llamadas de las colas en el monitoring y logico, los drivers de dahdi y su modulo se cargo fresquesito desde cero y ya me funciono todo. Lo malo es que tuve que reconfigurar todo de nuevo. Si en tu caso tu no tienes ninguna tarjeta que use dahdi, podras seguir trabajando sin problemas y tendras las grabaciones de las colas disponibles. Mi recomendacion es lo que yo hice; reinstalar Elastix 2.4 desde cero, ya que de esa manera obtienes el beneficio de tener las grabaciones y dahdi funcionando correctamente. Saludos. Fri, 17 May 2013 17:10:32 -0500 Subject: Callcenter - by: hgeorge123 http://elastix.org/index.php/en/component/kunena/55-call-center/121964-callcenter.html#122035 No el problema no es de tiempo el problema es que cuando termina la llamada la opcion de agendar llamada no existe desaparece Fri, 17 May 2013 15:58:39 -0500 Subject: procedencia de logs - by: c8aj http://elastix.org/index.php/en/component/kunena/50-novatos/122018-procedencia-de-logs.html#122034 muy buen comando netsfk.. muchas gracias, y ahora.. como borro esos logs.. donde los encuentro... marcan 8.5 Gb me estan ocupando demasiado espacio en disco.. Fri, 17 May 2013 13:12:16 -0500 Subject: T1 not working anymore. - by: atrejo http://elastix.org/index.php/en/component/kunena/31-asterisk/121968-t1-not-working-anymore.html#122033 Hello. I ended up reinstalling Elastix 2.4 from scratch and it seems to have installed all the correct modules, as apparently it is working now. I still need to have an actual T1 connected to it but it is recognizing a loopback connector so far. Thank you for your response. Antonio. Fri, 17 May 2013 13:02:31 -0500 Subject: custom trank in elastix 3.0 - by: jgutierrez http://elastix.org/index.php/en/component/kunena/1-installation-issues/120951-custom-trank-in-elastix-30.html#122028 If you can't do it, post it on: http://bugs.elasix.org Fri, 17 May 2013 12:16:31 -0500 Subject: conf file where SIP trunk is stored? - by: jgutierrez http://elastix.org/index.php/en/component/kunena/3-help/121993-conf-file-where-sip-trunk-is-stored.html#122026 If you create them through the web interface, they are created on sip_additionals.conf Fri, 17 May 2013 12:11:58 -0500 Subject: Hardware óptimo para server Elastix - by: jgutierrez http://elastix.org/index.php/en/component/kunena/11-ayuda/11930-hardware-optimo-para-server-elastix.html#122024 Puedes utilizar el mismo hardware que ya tienes, o si es que quieres usar un nuevo servidor, puedes enviar un correo a: sales@elastix.com solicitando un catálogo de nuestra línea de appliances con sus respectivas capacidades. Fri, 17 May 2013 12:09:42 -0500 Subject: Elastix to Avaya T1 trunk - by: knib http://elastix.org/index.php/en/component/kunena/3-help/114520-elastix-to-avaya-t1-trunk.html?limit=10&start=20#122022 Use ISDN trunk between them! Sangoma on your Elastix box -> DS1 card on your Avaya. This is how our Elastix production boxes works! H.323 is for Avaya only. Do not use the SIP service of Avaya as you will just waste of money for licensing. You know why? You will experience dead calls and robot calls. Lastly, Avaya is not good in SIP, it is not stable. Fri, 17 May 2013 12:05:03 -0500