It's a mistake in your dialplan. See lines 342-343:
  1. Dialplan: sofia/internal/1009@192.168.2.101 Action bridge(user/${dialed_extension}@${domain_name})
  2. Dialplan: sofia/internal/1009@192.168.2.101 Action answer()
  3. Dialplan: sofia/internal/1009@192.168.2.101 Action sleep(1000)
  4. Dialplan: sofia/internal/1009@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@192.168.2.101 answer()
-Steve



On 9 June 2011 09:11, Steven Ayre <steveayre@gmail.com> wrote:
I don't see the siptrace in that log?

-Steve


On 9 June 2011 08:35, Alessandro <a.luppi@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@localnet_ip                      FS(ip:localnet_ip)                                      1001@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@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@localnet_ip                      FS(ip:localnet_ip)                                      1001@localnet_ip

INVITE ---------->
                                     INVITE --------------->                                
<-------------- trying                <--------------------trying
<-------------- 603                  <------------------  603
ACK ------------>                    ACK------------------>



The current behaviour of FS is:
1000@localnet_ip                      FS(ip:localnet_ip)                                      1001@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@seletech.com - Web: www.seletech.com   or   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@seletech.com - Web: www.seletech.com   or   www.seletech.eu

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