[Freeswitch-users] Response status from client

Steven Ayre steveayre at gmail.com
Thu Jun 9 12:11:53 MSD 2011


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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