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:crangel@globaltelesourcing.com]
Enviado el: jueves, 26 de junio de 2014 01:27 p.m.
Para: miconda@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.

CARLOS RANGEL | INFORMATION TECHNOLOGY DIRECTOR

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De: sr-users-bounces@lists.sip-router.org [mailto:sr-users-bounces@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
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