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

Zvi Agmon zvi at lexifone.com
Thu Jul 25 15:59:39 MSD 2013


Hi,

Don't want to nag... but did not get any response...
Any idea how to proceed?

Best regards

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


On Mon, Jul 22, 2013 at 9:46 AM, Zvi Agmon <zvi at lexifone.com> wrote:

> 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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>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>>
>>
>
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