[SR-Users] Kamailio Freepbx Integration Dropping Calls

Daniel-Constantin Mierla miconda at gmail.com
Thu Jun 26 10:12:24 CEST 2014


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 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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-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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