[SR-Users] Kamailio Freepbx Integration Dropping Calls

[Carlos Rangel]

Thank you for Your response Daniel.

 

I am not sure how to do that but I will figure it out.

 

 

Regards

Carlos

 

De: sr-users-bounces at lists.sip-router.org
[mailto:sr-users-bounces at lists.sip-router.org] En nombre de Daniel Grotti
Enviado el: jueves, 26 de junio de 2014 03:16 a.m.
Para: sr-users at lists.sip-router.org
Asunto: Re: [SR-Users] Kamailio Freepbx Integration Dropping Calls

 

Hi,
a call trace could help you to understand why your server is not receiving
response (which one actually ?) form the Cisco.
Is that because the Cisco didn't receive your SIP message? Or is it because
Cisco replied but the response didn't reach your server ?

Try to make a sip trace in order to understand that.

Daniel



On 06/25/2014 06:50 PM, 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 at 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 at 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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