[Freeswitch-users] Response status from client
Steven Ayre
steveayre at gmail.com
Thu Jun 9 12:16:05 MSD 2011
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
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
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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 list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>>
>>
>
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