[Freeswitch-users] Response status from client
Alessandro
a.luppi at seletech.com
Thu Jun 9 14:03:13 MSD 2011
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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--
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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