[Asterisk-Dev] Status of supervised transfer?

F.S.Salloum ssal at intracom.gr
Wed Sep 22 22:38:30 MST 2004


Hi all, 

I checked out the bug site, but it seems the problem with Supervised
transfer that I have is related with SIP RFC. Let's make it more clear..

During a supervised transfer asterisk should reply with 

202 Accepted
Invite 
And then a bye 

Instead in my case scenario it doesn't send at all the Invite message,
I guess that some parameter which has been set by the masquerade related
functions found in channel.c, controls if an invite message will be sent
Or not. Can somebody inform me which variable is this and where is being
set?


In case supervised transfer works well for other member's, I would
appreciate if you could send me tcpdum (or ethereal) data in order to check
the SIP messages (ssal at intracom.gr) in our case the REFER implementation on
INTRACOM phones are based on RFC standard.

I also checked some hard coding in the chan_sip.c with the nobye parameter
which I cannot understand why is put there, for sure the blind transfer
works is this the reason? 

The blind transfer works fine, with the specific INTRACOM phones, which work
fine also with non-Asterisk Solutions such us Cisco etc

Thanks

PS: Because I'm new I posted the question in this mailing list, if this is
not the appropriate place please inform me  where I should post the
questions (maybe in the bug site?)




-----Original Message-----
From: asterisk-dev-bounces at lists.digium.com
[mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Chris Shaw
Sent: Wednesday, September 22, 2004 11:56 PM
To: Asterisk Developers Mailing List
Subject: Re: [Asterisk-Dev] Status of supervised transfer?

No, it's a hack based on the 't' and 'T' dial options. It's basically for
stupid non-attended transfer phones like the GrandStreams. The reason it's a
hack is that it requires * to be in the media stream to work, this breaks
re-invite functionality in the phone.

Don't get me wrong though, this feature rules! I have a grandstream and have
been wanting to be able to use attended transfers, huge karma to anthm for
this one!

check out bug # 2460

    -Chris

_______________________________________________
Asterisk-Dev mailing list
Asterisk-Dev at lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-dev
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-dev




More information about the asterisk-dev mailing list