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