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
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 >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > >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 -- ----------------------------------------------------------------------- HeuvelTop ICT Diensten v.o.f. ----------------------------------------------------------------------- "There are management solutions to technical problems, but no technical solutions to management problems" -----------------------------------------------------------------------
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
>_______________________________________________
>Asterisk-Users mailing list
>Asterisk-Users@lists.digium.com
>http://lists.digium.com/mailman/listinfo/asterisk-users
>To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
_________________________________________________________________
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Frank Schoep
2005-Jul-05 00:29 UTC
[SPAM:******** SpamScore] [Asterisk-Users] Call Transfer using SIP clients
On Monday 04 July 2005 16:11, Frank Schoep wrote:> I've spent quite some time already trying to get the call transfer function > to work on my Asterisk installation. > ... > 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. > ... > 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.OK, I'm happy to report that I finally managed to get the call transfer to work with a different softphone application called LinPhone. I will investigate why X-Lite doesn't work with sending DTMFs, but at least I know now that my Asterisk setup isn't at fault. Sadly, I can not share a solution any more detailed than this, the softphone switch was enough to convince me that the Asterisk server is working perfectly, it was primarily my X-Lite configuration at fault, so my sincere apologies for posting this on the Asterisk mailing list. If I find out how to get it working, I will append that information to the thread so others can reuse that knowledge later on, I'm sure someone will appreciate it. Thanks to each and all contributors who've got me thinking about finding the problematic part in my setup. Sincerely, Frank