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

Doron Kruh doron at lexifone.com
Thu Jul 25 16:14:14 MSD 2013 

Following this issue,

We tried to reproduce this issue with another SIP server (Voxeo Prophecy)
and we saw that the SIP server replies back an answer on the outgoing port
to INVITE requests.

Is there any configuration property which might be causing FreeSwitch to
stop listen the port of the outgoing leg?

This issue is very important and we would appreciate your response.

Best regards,

Doron.


On Thu, Jul 25, 2013 at 2:59 PM, Zvi Agmon <zvi at lexifone.com> wrote:

> 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
>>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
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>>>>
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>>>>
>>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
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>>>
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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://www.freeswitch.org
>
>


-- 
Doron Kruh,
Senior Developer
Lexifone , Hapalyam 7 Haifa.
Mobile: 972-52-3823361
Office:   972-4-9126792
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