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

Thu May 28 06:07:27 PDT 2009

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
>>
>> 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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>>
>
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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
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