do you have an ngrep of the sip traffic? This can happen if the sip/rtp cannot connect (perhaps blocked by the dsl router)
Fred Posner The Palner Group, Inc. http://www.palner.com (web) +1-503-914-0999 (direct) +1-954-472-2896 (fax) On 07/01/2014 01:12 PM, Carlos Rangel wrote: > Hello List > > > > Hopefully someone can help. This is the problem when the call is hug up 20-30 > seconds after it initiates. The call is only hung on when the remote > extension initiates the call. If the remote extension receives the call there > is no problem the call is not hung on. I changed the remote cisco phone for a > yealink and it is the same behavior. It thought it was the phone. > > > > This is what I am using in kamailio.cfg > > > > #!define WITH_MYSQL > > #!define WITH_AUTH > > #!define WITH_ASTERISK > > #!define WITH_USRLOCDB > > #!define WITH_ANTIFLOOD > > > > Remote User Internet > Internal network > > Yealink IP TG28P ----DSL router ---|------Internet --------|-----Cisco ASA > 5500 FW--------------Kamailio/Freepbx (Same Box)------IAX > Trunk----------Freepbx Production Server --------|------ PSTN > > > > > > Thanks > > Carlos Rangel > > > > De: Carlos Rangel [mailto:cran...@globaltelesourcing.com] > Enviado el: jueves, 26 de junio de 2014 01:27 p.m. > Para: mico...@gmail.com; 'Kamailio (SER) - Users Mailing List' > Asunto: RE: [SR-Users] Kamailio Freepbx Integration Dropping Calls > > > > Hi Daniel > > > > Thank you so much for your response. Here is the SIP trace of one of the > calls, I am not sure where the call initiates but you can see at the end of > the file in bold X-Asterisk-HangupCause: No user responding. I am not sure > why is it sending this message though. > > > > The variables are > > > > Extension/Username=XXXXX > > Ext_IP= Public IP > > Internal_IP= Asterisk/Kamailio internal IP > > > > Sorry for the long file but again I am not sure where the call initiates > > > > This is the part where that call is hung on. > > > > U 2014/06/26 13:36:11.831965 Kamailio_IP:5080 -> Kamailio_IP:5060 > > BYE sip:XXXXX@65.190.71.203:5060;user=phone;transport=udp SIP/2.0. > > Via: SIP/2.0/UDP Kamailio_IP:5080;branch=z9hG4bK7ce1be47. > > Route: <sip:Kamailio_IP;lr=on;ftag=000653dc394000970f227678-1fafb4e2>. > > Max-Forwards: 70. > > From: <sip:919707249077@Kamailio_IP:5060>;tag=as2e670ea4. > > To: "User" <sip:XXXXX@Kamailio_IP:5060>;tag=000653dc394000970f227678-1fafb4e2. > > Call-ID: 000653dc-3940000b-33caf1b2-20ccd185@192.168.0.22. > > CSeq: 102 BYE. > > User-Agent: FPBX-2.11.0(11.10.2). > > X-Asterisk-HangupCause: No user responding. > > X-Asterisk-HangupCauseCode: 18. > > Content-Length: 0. > > . > > U 2014/06/26 13:36:11.832260 Kamailio_IP:5060 -> 65.190.71.203:5060 > > BYE sip:XXXXX@65.190.71.203:5060;user=phone;transport=udp SIP/2.0. > > Via: SIP/2.0/UDP > Kamailio_IP;branch=z9hG4bKcf68.d6ef5aa9cc5bd0fb0ab13a563b7cf284.0. > > Via: SIP/2.0/UDP Kamailio_IP:5080;branch=z9hG4bK7ce1be47. > > Max-Forwards: 69. > > From: <sip:919707249077@Kamailio_IP:5060>;tag=as2e670ea4. > > To: "User" <sip:XXXXX@Kamailio_IP:5060>;tag=000653dc394000970f227678-1fafb4e2. > > Call-ID: 000653dc-3940000b-33caf1b2-20ccd185@192.168.0.22. > > CSeq: 102 BYE. > > User-Agent: FPBX-2.11.0(11.10.2). > > X-Asterisk-HangupCause: No user responding. > > X-Asterisk-HangupCauseCode: 18. > > Content-Length: 0. > > > > > > > > > > Descripción: Description: Description: DLR-Logo-No-TextCARLOS RANGEL | > INFORMATION TECHNOLOGY DIRECTOR > > Global Telesourcing México, S. de R.L. de C.V. | Aarón Sáenz #1891-1 | > Monterrey, N.L., México > Direct 703 894 1667 | Mobile US 703 894 1667 | Mobile MX +52 1 812 000 7362 | > cran...@globaltelesourcing.com > > ________________________________________________________________________________________________________________ > > The information contained in this e-mail and any attached documents may > contain information that is confidential or otherwise protected from > disclosure. If you are not the intended recipient of this message, or if this > message has been sent to you in error, please immediately alert the sender by > reply e-mail and then delete this message, including any attachments. Any > dissemination, distribution or other use of the contents of this message by > anyone other than the intended recipient is strictly prohibited. > > > > > > De: sr-users-boun...@lists.sip-router.org > [mailto:sr-users-boun...@lists.sip-router.org] En nombre de Daniel-Constantin > Mierla > Enviado el: jueves, 26 de junio de 2014 03:12 a.m. > Para: Kamailio (SER) - Users Mailing List > Asunto: Re: [SR-Users] Kamailio Freepbx Integration Dropping Calls > > > > Hello, > > can you gran the SIP trace on kamailio server for such case? > > You can use ngrep, like: > > ngrep -d any -qt -W byline port 5060 > > and send the output to the mailing list. You can replace any sensitive > information (e.g., ip address) before sending to mailing list. > > The typical call drop after 30-40 secs is when ACK is not routed properly, > but we have to see that in the sip trace. > > Cheers, > Daniel > > On 25/06/14 18:50, Carlos Rangel wrote: > > Hello > > > > I have successfully (I believe) implemented Kamailio 4.1.4 integration with > Freepbx 5.2.11 taking as a guide Daniel’s tutorial > http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb. > > I just did not create the voicemail tables because voice mail is handled by > Freepbx. I installed the system in a separate box for testing and connected > to the Freepbx Production server via IAX trunk. > > > > The system is behind a Cisco Firewall and looks like this > > > > Remote User Internet > Internal network > > Cisco 7960 ----DSL router ---|------Internet --------|-----Cisco ASA 5500 > FW--------------Kamailio/Freepbx (Same Box)------IAX Trunk----------Freepbx > Production Server --------|------ PSTN > > > > I have configured the FW to allow UDP and TCP traffic from the corresponding > IP as well as tfpt that is needed for the Ciscos to pick up the configuration > from the server. I have a few remotes Cisco 7960 phones that can register > remotely in Kamailio as long as the user is added with kamctl add user > password and as long as the extension is created in Freepbx. > > > > The problem that I have is when try to make a call from the remote Ciscos the > call is dropped after 30 or 40 seconds. I can see from the logs that the > problem appears to be that the server is not receiving responses from the > phone > > > > > > 06-25 10:57:30] WARNING[1814] chan_sip.c: Retransmission timeout reached on > transmission 000653dc-39400006-2579bbcd-13d9adcb@192.168.0.22 for seqno 102 > (Critical Response) -- See > https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions > > Packet timed out after 32001ms with no response > > [2014-06-25 10:57:30] WARNING[1814] chan_sip.c: Hanging up call > 000653dc-39400006-2579bbcd-13d9adcb@192.168.0.22 - no reply to our critical > packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). > > > > Is this something that we can adjust in kamailio or could it be related to > the FW configuration?? Sorry but I am very new to kamailio and sip. > > > > Thanks > > Carlos > > > > > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
