[Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]

Sean Devoy sdevoy at bizfocused.com
Tue Dec 18 05:18:48 MSK 2012


Ah, this is interesting.  I have pointed out to the customer that this error
has ONLY happened when the call is from THIS SPECIFIC caller.  They said it
is likely just because this is a current customer site where they have many
calls.  That may point to a codec issue, but why only occasionally?

 

After hours today, I tired an identical enterprise bridge statement  from my
office something like 50 times and could not reproduce the error.

I am waiting for a recurrence.

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Monday, December 17, 2012 7:00 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]

 

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 at bizfocused.com> wrote:

Waiting for another failure with siptrace REALLY on this time.

 

If the user has clicked DND on these cisco phones, could that cause these
messages?

 

Sean

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at 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]

 

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.

 

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.

 

 

On Mon, Dec 17, 2012 at 11:05 AM, Sean Devoy <sdevoy at bizfocused.com> wrote:

http://pastebin.freeswitch.org/20342





 

-- 
Anthony Minessale II

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-- 
Anthony Minessale II

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