[Asterisk-Users] Call Transfer using SIP clients

Tulika Pradhan
Mon, 04 Jul 2005 20:57:03 -0000

call transfer works for me fine without any additions in features.conf
by simply using Dial(SIP/${EXTEN},20,tT)
and pressing #<number to be transfered to>
this works both from caller as well as callee.

tulika

>From: Frank Schoep <frank@tintel.nl>
>Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
><asterisk-users@lists.digium.com>
>To: Asterisk-Users@lists.digium.com
>Subject: [Asterisk-Users] Call Transfer using SIP clients
>Date: Mon, 4 Jul 2005 16:11:13 +0200
>
>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
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