[Freeswitch-users] Response status from client
Alessandro
a.luppi at seletech.com
Thu Jun 9 14:48:38 MSD 2011
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
>> <mailto: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
>> <mailto: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
>>> <mailto: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 <mailto:a.luppi at seletech.com> - Web:www.seletech.com <http://www.seletech.com> orwww.seletech.eu <http://www.seletech.eu>
>>>
>>>
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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 <mailto:a.luppi at seletech.com> - Web:www.seletech.com <http://www.seletech.com> orwww.seletech.eu <http://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 orwww.seletech.eu
>
>
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> FreeSWITCH-users mailing list
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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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