[Freeswitch-users] Response status from client
Steven Ayre
steveayre at gmail.com
Thu Jun 9 14:52:59 MSD 2011
Great. :)
No problem.
-Steve
On 9 June 2011 11:48, Alessandro <a.luppi at seletech.com> wrote:
> hi,
>
> i found the problem, was in the default.xml dialplan.
>
> Thanks for help
>
> Regards
>
> Alessandro
>
> Il 09/06/2011 12:03, Alessandro ha scritto:
>
> Hi,
>
> This is de default diaplan:
> http://pastebin.freeswitch.org/16464
>
> but I think that sofia read public dial plan, ths is the public dial plan:
>
> http://pastebin.freeswitch.org/16465
>
>
> I don't see the point where the FS read operation reported at lines
> 342-343.
>
> Thanks
>
> Regards
>
> Alessandro
>
>
>
> Il 09/06/2011 10:16, Steven Ayre ha scritto:
>
> It's a mistake in your dialplan. See lines 342-343:
>
> 1. Dialplan: sofia/internal/1009 at 192.168.2.101 Action bridge(user/${
> dialed_extension}@${domain_name})
> 2. Dialplan: sofia/internal/1009 at 192.168.2.101 Action answer()
> 3. Dialplan: sofia/internal/1009 at 192.168.2.101 Action sleep(1000)
> 4. Dialplan: sofia/internal/1009 at 192.168.2.101 Action voicemail(default
> ${domain_name} ${dialed_extension})
>
>
> You're answering the call after the bridge before it goes to voicemail.
> That answer is what is generating the 200.
>
> See lines 507-508 to see it happening:
>
> 1. 2011-06-09 09:15:15.627871 [INFO] mod_dptools.c:2393 Originate
> Failed. Cause: CALL_REJECTED
> 2. EXECUTE sofia/internal/1009 at 192.168.2.101 answer()
>
> -Steve
>
>
>
> On 9 June 2011 09:11, Steven Ayre <steveayre at gmail.com> wrote:
>
>> I don't see the siptrace in that log?
>>
>> -Steve
>>
>>
>> On 9 June 2011 08:35, Alessandro <a.luppi at seletech.com> wrote:
>>
>>> Hi,
>>>
>>> the url is: http://pastebin.freeswitch.org/16463
>>> i made a call from phone1 to phone2, the called party refused the call
>>> with code 603. FS received the status 603 form the called (softphone 2)
>>> party. Than FS sent to the calling party (softphone 1) the message 200 and
>>> bye.
>>>
>>> This is the resume of the log:
>>>
>>>
>>> 1000 at localnet_ip FS(ip:localnet_ip)
>>> 1001 at localnet_ip
>>>
>>> INVITE ---------->
>>> INVITE --------------->
>>> <-------------- trying <--------------------trying
>>> <------------------ 603
>>>
>>> <-------------- 200
>>>
>>> ACK------------------> <--------------------BYE
>>>
>>>
>>>
>>> *You said you had voicemail before... you can't send 603 back to the
>>> client and continue to voicemail because the 603 terminates the call.*
>>>
>>> When the called party terminates the call before answering, the calling
>>> party receive e registered message like "The phone called is not available,
>>> leave a message ...". Than i found the registered message in freeswitch.
>>> (I'm using fusion-pbx)
>>>
>>>
>>> Thanks
>>>
>>> Regards
>>>
>>> Alessandro
>>>
>>>
>>> Il 08/06/2011 21:55, Steven Ayre ha scritto:
>>>
>>> Question 1:
>>>> i'm developing a custom client sip with pjsip. This client when receive
>>>> a call that can't be accepted respond with status 603. I think that
>>>> freeswitch filter this status.
>>>>
>>>
>>> 603 gets treated fine for me. I think we need to see more information -
>>> can you put a debug level log of the call with siptrace enabled (sofia
>>> global siptrace on) on pastebin (http://pastebin.freeswitch.org/) and
>>> then post the url here?
>>>
>>> Chances are you're doing something in the dialplan that's answering the
>>> call, either before or after the failed bleg.
>>>
>>> You said you had voicemail before... you can't send 603 back to the
>>> client and continue to voicemail because the 603 terminates the call.
>>>
>>>
>>>> Question:2
>>>>
>>> It's possible a custom Header pass trough in status response like trying
>>>> or session in progress? I'm able to use custom header only on invite adding
>>>> to invite a header with name like X-myheader. Any suggestion?
>>>>
>>>
>>> Yes, you can for 180/183, with the sip_ph_X- prefix. That puts the header
>>> on any provisional response.
>>> http://wiki.freeswitch.org/wiki/Sofia#Adding_Response_Headers
>>>
>>> For example:
>>> <action application="set" data="sip_p_X-myheader=value"/>
>>>
>>> AFAIK you won't be able to do the same for a 100 Trying since Sofia
>>> doesn't let FS do any handling at the required point. But have a try anyway
>>> just to be sure.
>>>
>>> -Steve
>>>
>>>
>>> On 8 June 2011 20:14, Alessandro <a.luppi at seletech.com> wrote:
>>>
>>>> Hi,
>>>>
>>>> I have two questions about FS:
>>>>
>>>> Question 1:
>>>> i'm developing a custom client sip with pjsip. This client when receive
>>>> a call that can't be accepted respond with status 603. I think that
>>>> freeswitch filter this status.
>>>> This is an example of desired behaviour:
>>>> 1000 at localnet_ip FS(ip:localnet_ip)
>>>> 1001 at localnet_ip
>>>>
>>>> INVITE ---------->
>>>> INVITE --------------->
>>>> <-------------- trying <--------------------trying
>>>> <-------------- 603 <------------------ 603
>>>> ACK ------------> ACK------------------>
>>>>
>>>>
>>>>
>>>>
>>>> The current behaviour of FS is:
>>>>
>>>> 1000 at localnet_ip FS(ip:localnet_ip)
>>>> 1001 at localnet_ip
>>>>
>>>> INVITE ---------->
>>>> INVITE --------------->
>>>> <-------------- trying <--------------------trying
>>>> <------------------ 603
>>>>
>>>> <-------------- 200
>>>>
>>>> ACK------------------> <--------------------BYE I'd like to avoid the
>>>> current behaviour. It's possible a kind of message status path trough? If
>>>> the called party terminate the call before answering, FS send always to the
>>>> calling partner 200 and BYE. First thought was related to the voice-mail.
>>>> Now voice-mail is disabled but the behaviour is the same. Question:2 It's
>>>> possible a custom Header pass trough in status response like trying or
>>>> session in progress? I'm able to use custom header only on invite adding to
>>>> invite a header with name like X-myheader. Any suggestion? Thanks Good
>>>> Evening
>>>>
>>>> Alessandro
>>>>
>>>> --
>>>> Ing. Alessandro Luppi
>>>> Software development
>>>> Seletech srl
>>>> Via Collodi 8, 20052 Monza (MI) - Italy
>>>> Tel: +39.039.5962000 - Fax: +39.039.9716905
>>>> email: a.luppi at seletech.com - Web: www.seletech.com or www.seletech.eu
>>>>
>>>>
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>>>>
>>>
>>> _______________________________________________
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>>>
>>>
>>>
>>> --
>>> Ing. Alessandro Luppi
>>> Software development
>>> Seletech srl
>>> Via Collodi 8, 20052 Monza (MI) - Italy
>>> Tel: +39.039.5962000 - Fax: +39.039.9716905
>>> email: a.luppi at seletech.com - Web: www.seletech.com or www.seletech.eu
>>>
>>>
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>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>>>
>>
>
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>
>
>
> --
> Ing. Alessandro Luppi
> Software development
> Seletech srl
> Via Collodi 8, 20052 Monza (MI) - Italy
> Tel: +39.039.5962000 - Fax: +39.039.9716905
> email: a.luppi at seletech.com - Web: www.seletech.com or www.seletech.eu
>
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> --
> Ing. Alessandro Luppi
> Software development
> Seletech srl
> Via Collodi 8, 20052 Monza (MI) - Italy
> Tel: +39.039.5962000 - Fax: +39.039.9716905
> email: a.luppi at seletech.com - Web: www.seletech.com or www.seletech.eu
>
>
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