[Freeswitch-users] Freeswitch not responding to sip messages from call out leg

[Zvi Agmon]

Hi,

Michael - thanks for your response.
I tried to follow the ""reporting bugs" instructions and open a Jira issue
about 10 days ago - http://jira.freeswitch.org/browse/FS-5592

I added to this issue the relevant logs (FS and WireShark) and tried to
clearly describe what is happening.
Followed Anthony comment I reproduce the issue with a different gateway
provider and added these logs also to Jira.
Now, I also put the latest freeswitch log to pastebin -
http://pastebin.freeswitch.org/21214

Please let me know if there's any information that's still missing or if
the issue is not clear enough - I'll be more than happy to provide anything
needed to track down the problem.



Best regards

Zvi Agmon
Lexifone
email: zvi at lexifone.com
Office: +972-4-6817711
Cell: +972-54-4505109


On Sun, Jul 21, 2013 at 7:37 PM, Michael Collins <msc at freeswitch.org> wrote:

> You'll need to get a full console debug log with sip trace and put in
> pastebin.freeswitch.org. The folks here will try to help and it's easier
> for them to do so if they have more information. See also the "reporting
> bugs" wiki page for more details on gathering data for troubleshooting.
>
> -MC
> On Jul 21, 2013 5:49 AM, "Zvi Agmon" <zvi at lexifone.com> wrote:
>
>> Hello,
>>
>> Can any one help with this issue? We have no idea what is wrong.
>> The problem reproduced with 2 different gateway providers and with the
>> basic configuration that FS comes with (gateways and the bridge dial plan
>> were added).
>> Is there anything in the configuration that can cause such a problem?
>>
>> Here are the configurations
>>
>> *In public.xml:*
>>
>>  <extension name="public_did">
>>  <condition field="destination_number"  "expression="^.*$">
>> <!--action application="info"/-->
>>  <action application="log" data="INFO in public_did" />
>> <action application="set" data="domain_name=$${domain}"/>
>>  <action application="transfer" data="972544505109 XML bridge_call"/>
>> </condition>
>>  </extension>
>>
>> *bridge_call dial plan:*
>>
>>   <include>
>> <context name="bridge_call">
>>  <extension name="bridge_call">
>>   <condition field="destination_number" expression="(.*)">
>>     <action application="log" data="INFO bridge_call
>> destination_number=${destination_number}" />
>> <action application="bridge"
>> data="sofia/gateway/Jajah/${destination_number}"/>
>>   </condition>
>> </extension>
>> </context>
>>  </include>
>>
>>
>> *The gateway:*
>>
>> <gateway name="Jajah">
>> <param name="realm" value="91.194.5.180"/>
>>  <param name="username" value=""/>
>> <param name="password" value=""/>
>>  <param name="register" value="false"/>
>> <param name="caller-id-in-from" value="true"/>
>>  <param name="sip_cid_type" value="none"/>
>> </gateway>
>>
>>
>> Please help...!
>> Thanks!
>>
>>
>> Best regards
>>
>> Zvi Agmon
>> Lexifone
>> email: zvi at lexifone.com
>> Office: +972-4-6817711
>> Cell: +972-54-4505109
>>
>>
>> On Mon, Jul 8, 2013 at 10:55 AM, Zvi Agmon <zvi at lexifone.com> wrote:
>>
>>> Hello,
>>>
>>> We are using FS to bridge incoming call to an outbound telephone through
>>> a gateway provider.
>>> The bridge is created OK, but we found that after 3 minutes FS is no
>>> longer responding to sip messages from the call out provider. Thus, in case
>>> the provider sends sip-Invite message, it gets no response from FS and
>>> disconnect the call out leg. Note that FS is also not aware of the Bye
>>> message so leg A is still on the line and when it hangs up FS send Bye
>>> message to leg B and get as response "Unknown Dialog".
>>>
>>> I set the siptrace on and see that no packets arriving sofia (when the
>>> calling leg hanged up I do see the packets).
>>> Also I set all sofia components log level to 7 - this is what was
>>> printed out after the call answered:
>>>
>>> 2013-07-08 10:28:49.661014 [NOTICE] sofia.c:6288 Channel
>>> [sofia/external/97249126793] has been answered
>>> 2013-07-08 10:28:49.661014 [DEBUG] switch_channel.c:3291
>>> sofia/external/97249126793 process sched_hangup +7199
>>> DJIWR4KKCZB3DGAS7LYLI76IPE at 81.201.82.45 alloted_timeout:
>>> sched_hangup(+7199 DJIWR4KKCZB3DGAS7LYLI76IPE at 81.201.82.45alloted_timeout)
>>>
>>> 2013-07-08 10:28:49.661014 [DEBUG] switch_scheduler.c:214 Added task 6
>>> switch_ivr_schedule_hangup (DJIWR4KKCZB3DGAS7LYLI76IPE at 81.201.82.45) to
>>> run at 1373275728
>>> 2013-07-08 10:28:49.681018 [DEBUG] switch_core_session.c:892 Send signal
>>> sofia/external/97249126793 [BREAK]
>>> 2013-07-08 10:28:49.681018 [DEBUG] switch_core_session.c:892 Send signal
>>> sofia/internal/972522977131 at voxbone.com [BREAK]
>>> nta.c:8950 outgoing_timer_dk() nta: timer D fired, terminate INVITE
>>> (46298196)
>>> nta.c:8778 _nta_outgoing_timer() nta_outgoing_timer: 0/0 resent, 0/1
>>> tout, 1/1 term, 1/2 free
>>> nta.c:8831 outgoing_timer_bf() nta: timer F fired, terminating ACK
>>> (46298196)
>>> nta.c:8778 _nta_outgoing_timer() nta_outgoing_timer: 0/0 resent, 1/1
>>> tout, 0/0 term, 1/1 free
>>>
>>>
>>>
>>> We are running FreeSWITCH Version 1.2.7+git~20130307T054051Z~0a2e713593
>>> (git 0a2e713 2013-03-07 05:40:51Z)
>>>
>>> On ubuntu release 12.04
>>>
>>> Please help.
>>>
>>> Thanks
>>> Zvi Agmon
>>> Best regards
>>>
>>> Zvi Agmon
>>> Lexifone
>>> email: zvi at lexifone.com
>>> Office: +972-4-6817711
>>> Cell: +972-54-4505109
>>>
>>
>>
>> _________________________________________________________________________
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>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
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>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
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