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.
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?
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.
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:
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