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