Getting the sip_call_id variable (Call-ID header) for each leg (eg from xml cdrs) will give you a field that makes it very easy to pull the call out of a pcap. -Steve On 18 December 2012 00:00, Anthony Minessale <anthony.minessale@gmail.com>wrote:
usually the 406 is done in an established call to refuse a codec change during re-invite. Its possible the other end thinks we want to change the codec when we do the session-timer re-invite but I'm sure we don't but the sip trace can help shed some light. You can run a pcap too at the same time so when we find the bad call in the logs we can filter it out of the pcap too. To avoid it getting too big you can just restart it every so often or use sippcapdump and delete calls that are not affected.
On Mon, Dec 17, 2012 at 5:51 PM, Sean Devoy <sdevoy@bizfocused.com> wrote:
Waiting for another failure with siptrace REALLY on this time.****
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If the user has clicked DND on these cisco phones, could that cause these messages?****
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Sean****
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*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Anthony Minessale *Sent:* Monday, December 17, 2012 12:32 PM
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]****
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The siptrace is not in here? Did you enable it on the console and then capture to the log file perhaps?****
you should do "sofia tracelevel debug" too to route traces to the log file.****
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Its hard to say for sure with no sip trace but it seems like the far end is rejecting the session timer re-invite causing the call to end.****
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On Mon, Dec 17, 2012 at 11:05 AM, Sean Devoy <sdevoy@bizfocused.com> wrote:****
http://pastebin.freeswitch.org/20342****
****
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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