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

Thu May 28 06:08:23 PDT 2009

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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>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
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>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 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 <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 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 <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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