[Freeswitch-users] calls appear to be dropping ... from landlines

Tue Jun 2 05:25:38 PDT 2009

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.minessale at gmail.com> wrote:
> 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.minessale at 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 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 <daletrub at 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 able to see anything that's out of the ordinary that we should be paying
>>> attention to?
>>>
>>> Best,
>>> Dale
>>>
>>> On Fri, May 22, 2009 at 5:47 AM, Anthony Minessale
>>> <anthony.minessale at gmail.com> wrote:
>>>>
>>>> 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 is missing some of the info from the few seconds
>>>> prior but with the incomplete
>>>> info it looks like the other side sent a BYE ending the call.
>>>>
>>>>
>>>> On Thu, May 21, 2009 at 10:09 PM, Dale Trub <daletrub at gmail.com> wrote:
>>>>>
>>>>> 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 have any
>>>>> issues, so they're insisting it's on our end.
>>>>> On Thu, May 21, 2009 at 7:28 PM, Brian West <brian at freeswitch.org>
>>>>> wrote:
>>>>>>
>>>>>> 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 landlines.   This means they join the conference, and x
>>>>>> minutes in they simply drop.
>>>>>> I know that cellphones tend to drop calls frequently, but landlines
>>>>>> are pretty reliable, and we're hearing it a lot.  From the FreeSwitch
>>>>>> side of things, it just
>>>>>> looks like those callers hung up (but then dialed back in just a
>>>>>> moment later).
>>>>>> I'm attaching two different snippets of the FS log files where these
>>>>>> issues are occurring.
>>>>>>
>>>>>> Next time please call them .txt because you cause extra work to have
>>>>>> to open them otherwise.
>>>>>>
>>>>>> Does anyone have any recommendations about how to troubleshoot this?
>>>>>> Any known issues/patches in FS that could be biting us?
>>>>>>
>>>>>> Depends you failed to include some very valid info such as what
>>>>>> version or svn rev you're running and what linux distro.
>>>>>>
>>>>>> Is there some SIP logging we can do to debug?
>>>>>>
>>>>>> Yes covered on the wiki.
>>>>>>  http://wiki.freeswitch.org/wiki/Debugging_Freeswitch
>>>>>>
>>>>>> Are there any paid contractors avail who would have the expertise to
>>>>>> look into this?
>>>>>>
>>>>>> email consulting at freeswitch.org
>>>>>>
>>>>>> Any help appreciated ... this is a major issue for us!
>>>>>> Thanks much,
>>>>>> -Dale
>>>>>>
>>>>>> Brian West
>>>>>> brian at freeswitch.org
>>>>>> -- Meet us at ClueCon!  http://www.cluecon.com
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
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>>>>>>
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>>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
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>>>>>
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>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org
>>>> iax:guest at conference.freeswitch.org/888
>>>> googletalk:conf+888 at conference.freeswitch.org
>>>> pstn:213-799-1400
>>>>
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>>>>
>>>
>>>
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>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:213-799-1400
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
>
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>
>