[Asterisk-Users] Call Transfer using SIP clients

Frank Schoep
Mon, 04 Jul 2005 07:11:13 -0000

Hello all,

First of all, let me apologize about the length of this message, but I suppose
it was necessary to include the details.

I've spent quite some time already trying to get the call transfer function
to
work on my Asterisk installation. Let me first describe the general situation
of the setup I am using, so you might be able to pinpoint the cause of the
problem.

I'm currently using Asterisk CVS as of July 4th 2005. The only means of
communication at the moment is the XTen X-Lite SIP Client, I already added
the following entries to my "sip.conf" configuration file:

[frank]
canreinvite=no
type=friend
secret=frank
username=frank
nat=yes
host=dynamic

[test]
canreinvite=no
type=friend
secret=test
username=test
nat=yes
host=dynamic

The SIP setup is working without a problem, the X-Lite application correctly
registers the users and I can set up calls between them. I've also tested
queues and they work without a problem, too. Next up is my extensions
configuration, of which the interesting section now looks like this:

[default]
include => general ; longshot, added out of desparation
include => parkedcalls ; longshot, added out of desparation
include => featuremap ; longshot, added out of desparation

exten => 800,1,Answer
exten => 800,2,Dial(SIP/frank,20,tT)
exten => 800,3 Hangup

exten => 802,1,Answer
exten => 802,2,Dial(SIP/test,20,tT)
exten => 802,3 Hangup

Notice the inclusion of several contexts that should or would have to be
defined in the features configuration. My features.conf looks something like
this, I trimmed the 'general' section for brevity:

[general]
; (trimmed) default options

[featuremap]
blindxfer => #1 ; Blind transfer
disconnect => *0 ; Disconnect
automon => *1 ; One Touch Record
atxfer => *2 ; Attended transfer

My testing scenario starts as follows:
- log in both X-Lite SIP clients
- from the 'test' phone, call extension 800
- on X-Lite client 'frank' accept the call
- talk to eachother

At this point I want to transfer to call to another extension, also defined in
"sip.conf" but unlisted here. The problem is that nothing happens when
I
press the "#1" or "*2" keys in the 'frank' X-Lite
client. I also tested these
key combinations on the 'test' X-Lite client during the call, but that
also
had not effect.

I searched the web and the mailing list archive for a solution, and if I
recall correctly, someone stated that call transfer is only available for
calls originating from the PSTN. Is this correct, also in regard of the
current version of Asterisk? Has anyone got an idea how to get call transfer
to work?

One thing I tried was to change the DTMF settings in the clients, so they are
sent in-band, but this also didn't help. Should I revert this option?

Thanks in advance for your time and patience.

Sincerely,

Frank Schoep