Hey Gents,
What is the Jira for this issue?
Dale,
Did you get any SIP traces. I am interested to have a look. You can
use NGREP if your system is Linux.
Regards,
Jim
On Thu, May 28, 2009 at 11:08 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
btw,
3 and 4 are not useful
btw,
3 and 4 are not useful without 1
we only debug issues with svn trunk
On Thu, May 28, 2009 at 8:07 AM, Anthony Minessale
anthony.miness...@gmail.com wrote:
Also you should be putting these details in a jira report.
http://jira.freeswitch.org
open an issue report and attach all
Also you should be putting these details in a jira report.
http://jira.freeswitch.org
open an issue report and attach all relevant logs, do not attach tarballs or
gzipped files and make sure text files have a .txt extension.
On Wed, May 27, 2009 at 6:58 PM, Dale Trub dalet...@gmail.com wrote:
Anthony,
Thank you for your suggestions! We are working on 1), but need to
re-integrate code we've changed, and do regression testing. That's in
progress, and we expect to be able to upgrade by the end of next week.
We did manage to do 3) and 4), and we now have SIP logs (attached). Are you
1) update to lastest trunk (you are at least 1000 revisions behind)
2) disable the presence debug in sofia.conf
3) enable sip trace instead sofia profile internal siptrace on
4) reproduce your problem.
Make sure you include more of the log from before the hangup happened.
The one you posted here
We're running FreeSwitch as part of a teleconferencing service, inside a
telcom (so no
internet latency/NAT issues) and using g.729
We are receiving some complaints of dropped calls,
including from landlines. This means they join the conference, and x
minutes in they simply drop.
I know that
I have experienced the same a while ago, I originated calls from my
freeswitch server to some landlines and calls would simply drop after
X minutes.
I tried to debug the thing but found nothing relevant, maybe I had the
same issue as you.
Let me know if you figure it out what it was.
Diego
On
On May 21, 2009, at 9:15 PM, Dale Trub wrote:
We're running FreeSwitch as part of a teleconferencing service,
inside a telcom (so no
internet latency/NAT issues) and using g.729
So you're using g729 with conferences?
We are receiving some complaints of dropped calls,
including from
Dale Trub dalet...@gmail.com wrote:
Does anyone have any recommendations about how to troubleshoot this?
sofia profile external siptrace on
and watch the SIP traces to see what happens.
Any known issues/patches in FS that could be biting us?
You didn't say which version you were running.
Thanks Brian! To answer your questions:
Freeswitch svn revision: 12148
Centos rev: 2.6.18-92.el5
And apologies, actually I guess we're using g711 not 729.
Jason: I agree it would seem to be on the switch/telco side. And, the
telco says many other people are in the same set-up as us and don't
If I had a few bucks for every time the telco has said this to me I
could just about retire! You using 100% SIP?
/b
On May 21, 2009, at 10:09 PM, Dale Trub wrote:
And, the telco says many other people are in the same set-up as us
and don't have any issues, so they're insisting it's on our
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