Re: [SPAM:***** SpamScore] RE: [Asterisk-Users] Call Transfer using SIP clients
On Tuesday 05 July 2005 05:57, Tulika Pradhan wrote:
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
Could you provide me with some more information so I can check where the
differences in our setups are? It would really help to see how you
implemented your extensions and SIP configuration. Could you describe the
following regarding your Asterisk installation:
- Asterisk version
- The SIP clients you use
- Excerpt of extensions.conf, which definitions and contexts do you include
- Excerpt of features.conf, which lines (if any) are in there
- (Maybe) an excerpt of sip.conf, how are the SIP peers configured
I hope you find the time to post these bits and pieces as it will make it
easier for me to debug the situation. I've already tried numerous settings
and combinations of options, but haven't had any luck yet. Thanks in advance
for your precious time.
If anyone else has some ideas regarding my question, feel free to jump in -
the more the merrier.
Sincerely,
Frank
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Re: [SPAM:***** SpamScore] Re: [Asterisk-Users] Call Transfer using SIP clients
On Monday 04 July 2005 16:47, Elwin Andriol wrote: 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 The X-Lite softphone does indeed have a Transfer and Hold in the interface, but the functionality of those buttons appears to have been disabled when the client is connected and registered on the Asterisk server. Pressing the on-screen buttons doesn't have any effect while either having a call or while idle. Related to client-side transferring: I set the canreinvite option in the SIP configuration to no because both clients are behind a NAT / firewall and I read in the documentation that you'd want to disable the canreinvite option in those situations. I haven't had any trouble because of this, as I stated earlier calling and talking is working without hitches. I haven't had the chance to try hardware phones yet, the testing I'm doing at the moment involves softphones only. Now that I think of it, I'll try to setup other applications again which might send DTMFs in a different form compared to X-Lite. In the meantime thanks in advance to everyone involved in this thread now and in the future. Sincerely, Frank ___ 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
Re: [Asterisk-Users] Call Transfer using SIP clients
Frank Schoep wrote: The X-Lite softphone does indeed have a Transfer and Hold in the interface, but the functionality of those buttons appears to have been disabled when the client is connected and registered on the Asterisk server. Pressing the on-screen buttons doesn't have any effect while either having a call or while idle. I'm pretty sure it's disabled on purpose on the free Linux version of the phone. I remember reading that somewhere on their site once upon a time. B. ___ 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
Re: [Asterisk-Users] Call Transfer using SIP clients
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 --- ___ 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
RE: [Asterisk-Users] Call Transfer using SIP clients
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 [EMAIL PROTECTED]
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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