[Asterisk-Users] Call Transfer using SIP clients

Elwin Andriol
Mon, 04 Jul 2005 07:47:03 -0000

Frank Schoep wrote:

>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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>
>
>
I don't know if this will be of any help to you, but at least I can
confirm problems with transfering calls with SIP agents. A little while
ago we were having big problems getting transfers using DTMF to work.

In that particular situation we were using a mix of only "hard" SIP
devices (BT101's and Elmeg/Snom190's) and ZAP channels. We tried both
the stable version of asterisk and the CVS HEAD, but without results
(but negative). In the end, we solved the problem by not using DTMF
transfers at all, but by using the transfer capabilities of the SIP
devices themselves (transfer for and hold buttons). These buttons did
not appear to work (correctly) with the stable asterisk version we
initially used (1.0.7), but with the CVS HEAD (> 29-MAY-2005) they
appear to work just fine.

I'm not familiar with "soft" SIP agents, so I don't know if
the ones you
use have such build-in transfer capabilities as their hardware
counterparts like the BT101's and Snom190's have. I they do, you might
wan't to give it a try. This is of course rather a workaround than a
solution to your problem.

E. Andriol

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