hi All, We are working on Soft-PBX using Asterisk. This relates to CALL TRANSFERRING aspects of Asterisk. We were able to do one type of call transfering, ie, the called person can transfer the original call to another person. but we were unable to do the other, that is, call initiator him/her self couldn't transfer the call. Eventhough the documentation for Dial application intructs to use "T" to achieve that. and we learnt that it has not been implemented yet in Asterisk. Is this true? Is some one workin on this issue? if the answer is NO, we can give a try to implement it, with a help of u all , ofcourse :-) (cos, we are quite new to asterisk-only 1 month of experience, but amazed of its great performance) Thank you very much, Surajee -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030601/01aa5a49/attachment.htm
U get the following output when u execute the "show application Dial"
command in the Asterisk prompt,
-= Info about application 'Dial' =-
[Synopsis]:
Place an call and connect to the current channel
[Description]:
Dial(Technology/resource[&Technology2/resource2...][|timeout][|options][|URL]):
Requests one or more channels and places specified outgoing calls on them.
As soon as a channel answers, the Dial app will answer the originating
channel (if it needs to be answered) and will bridge a call with the channel
which first answered. All other calls placed by the Dial app will be hunp up
f a timeout is not specified, the Dial application will wait indefinitely
until either one of the called channels answers, the user hangs up, or all
channels return busy or error. In general, the dialler will return 0 if it
was unable to place the call, or the timeout expired. However, if all
channels were busy, and there exists an extension with priority n+101 (where
n is the priority of the dialler instance), then it will be the next
executed extension (this allows you to setup different behavior on busy from
no-answer).
This application returns -1 if the originating channel hangs up, or if the
call is bridged and either of the parties in the bridge terminate the call.
The option string may contain zero or more of the following characters:
't' -- allow the called user transfer the calling user
'T' -- to allow the calling user to transfer the call.
'r' -- indicate ringing to the calling party, pass no audio until
answered.
'm' -- provide hold music to the calling party until answered.
'd' -- data-quality (modem) call (minimum delay).
'c' -- clear-channel data call (PRI-PRI only).
'H' -- allow caller to hang up by hitting *.
'C' -- reset call detail record for this call.
'P[(x)]' -- privacy mode, using 'x' as database if
provided.
In addition to transferring the call, a call may be parked and then picked
up by another user.
The optionnal URL will be sent to the called party if the channel supports
it.
Surajee
----- Original Message -----
From: George Lin
To: surajee@infotechs.lk
Sent: Monday, June 02, 2003 1:11 PM
Subject: FW: [Asterisk-Users] Call Transfer Problem
Hi,
Which document describes the Dial with ?T? option ? Could you let me know or
email it to me.
Thanks,
George Lin
-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Surajee Ratnayake
Sent: Sunday, June 01, 2003 9:10 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Call Transfer Problem
hi All,
We are working on Soft-PBX using Asterisk. This relates to CALL TRANSFERRING
aspects of Asterisk.
We were able to do one type of call transfering, ie, the called person can
transfer the original call to another person.
but we were unable to do the other, that is, call initiator him/her self
couldn't transfer the call. Eventhough the documentation for Dial
application intructs to use "T" to achieve that.
and we learnt that it has not been implemented yet in Asterisk. Is this true?
Is some one workin on this issue? if the answer is NO, we can give a try to
implement it, with a help of u all , ofcourse :-)
(cos, we are quite new to asterisk-only 1 month of experience, but amazed of
its great performance)
Thank you very much,
Surajee
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yes, u are quite right, you can find this feature in almost every pbx now.
We are also wondering whether, presently some one is implementing this feature
or not, if no body is doing that, we can
start on that
Surajee
----- Original Message -----
From: George Lin
To: surajee@infotechs.lk
Sent: Wednesday, June 04, 2003 3:36 AM
Subject: RE: [Asterisk-Users] Call Transfer Problem
so, What should the call initiator do if s/he wants to transfer the call
initiated by himself/herself, by using flash keypad or what else ?
I can see such application can be used in some big office, where the BOSS
always asks the secretary to make the call, once the call is connected, then the
secretary can trasfer the call to the BOSS. in order to let the BOSS talk on the
phone. am I right ??
Please let me know once the feature is implemented.
George Lin
-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Surajee Ratnayake
Sent: Monday, June 02, 2003 1:05 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Call Transfer Problem
U get the following output when u execute the "show application
Dial" command in the Asterisk prompt,
-= Info about application 'Dial' =-
[Synopsis]:
Place an call and connect to the current channel
[Description]:
Dial(Technology/resource[&Technology2/resource2...][|timeout][|options][|URL]):
Requests one or more channels and places specified outgoing calls on them.
As soon as a channel answers, the Dial app will answer the originating
channel (if it needs to be answered) and will bridge a call with the channel
which first answered. All other calls placed by the Dial app will be hunp up
f a timeout is not specified, the Dial application will wait indefinitely
until either one of the called channels answers, the user hangs up, or all
channels return busy or error. In general, the dialler will return 0 if it
was unable to place the call, or the timeout expired. However, if all
channels were busy, and there exists an extension with priority n+101 (where
n is the priority of the dialler instance), then it will be the next
executed extension (this allows you to setup different behavior on busy from
no-answer).
This application returns -1 if the originating channel hangs up, or if the
call is bridged and either of the parties in the bridge terminate the call.
The option string may contain zero or more of the following characters:
't' -- allow the called user transfer the calling user
'T' -- to allow the calling user to transfer the call.
'r' -- indicate ringing to the calling party, pass no audio
until answered.
'm' -- provide hold music to the calling party until answered.
'd' -- data-quality (modem) call (minimum delay).
'c' -- clear-channel data call (PRI-PRI only).
'H' -- allow caller to hang up by hitting *.
'C' -- reset call detail record for this call.
'P[(x)]' -- privacy mode, using 'x' as database if
provided.
In addition to transferring the call, a call may be parked and then picked
up by another user.
The optionnal URL will be sent to the called party if the channel supports
it.
Surajee
----- Original Message -----
From: George Lin
To: surajee@infotechs.lk
Sent: Monday, June 02, 2003 1:11 PM
Subject: FW: [Asterisk-Users] Call Transfer Problem
Hi,
Which document describes the Dial with ?T? option ? Could you let me know
or email it to me.
Thanks,
George Lin
-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Surajee Ratnayake
Sent: Sunday, June 01, 2003 9:10 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Call Transfer Problem
hi All,
We are working on Soft-PBX using Asterisk. This relates to CALL
TRANSFERRING aspects of Asterisk.
We were able to do one type of call transfering, ie, the called person can
transfer the original call to another person.
but we were unable to do the other, that is, call initiator him/her self
couldn't transfer the call. Eventhough the documentation for Dial
application intructs to use "T" to achieve that.
and we learnt that it has not been implemented yet in Asterisk. Is this
true?
Is some one workin on this issue? if the answer is NO, we can give a try
to implement it, with a help of u all , ofcourse :-)
(cos, we are quite new to asterisk-only 1 month of experience, but amazed
of its great performance)
Thank you very much,
Surajee
-------------- next part --------------
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For testing purposes, my dial line is:
Dial(${ARG2},20,tT)
When I call from one machine through asterisk to another, I can press # from
either side and hear "Transfer."
However, from the caller side I can continue on and put people on hold by
dialing '700'.
>From the callee side, I can press # but if I try to dial an extension I hear
"I'm sorry. That is not a valid extension. Please try again."
Asterisk displays a message "Unable to find extension '7' in
context ' ' "
What this tells me is that if a VOIP client picks up a line that has been
Dial()ed from asterisk, that client is not given a context and, therefore,
cannot dial extensions. How can this be fixed? Have I messed up the setup
somehow? If so, can anyone give me a working example?
John.
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Hi Users, I am having a prblem using attended call transfer with asterisk. Actually my sip phone does not seem to support it. Can i use attended call transfer using the dial plan ... ??? means can somehow messing up with extesnions.conf I can get attended call transfer ? And yes also is there any way I can do it with AGI scripting ? Any AGI similar examples will be a lot of help. Thanks ! Usman.
This patch works a treat for us: http://bugs.digium.com/bug_view_page.php?bug_id=0002460 Makes all # transfers attended, but the transfer button on the phones can still be used for blind transfers from our SIP phones. Cheers, Michael On Fri, 8 Oct 2004 01:56:53 -0500 (CDT), usman@user.iphonica.net <usman@user.iphonica.net> wrote:> Hi Users, > > I am having a prblem using attended call transfer with asterisk. Actually > my sip phone does not seem to support it. Can i use attended call transfer > using the dial plan ... ??? means can somehow messing up with > extesnions.conf I can get attended call transfer ? And yes also is there > any way I can do it with AGI scripting ? Any AGI similar examples will be > a lot of help. Thanks ! > > Usman. > > _______________________________________________ > 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 >-- Michael Nolan michael.james.nolan@gmail.com http://www.michaelnolan.co.uk/
I do not want to use the default key of '#' for call transfer, because as we all know, it interferes with many IVRs that require # as a termination character. I modified features.conf and added: [featuremap] atxfer => ** The double-star now works great. If I press it while on a call, I go into transfer mode. The problem is that the # still works as well! Shouldn't the atzfer specification turn off the #? Any insight would be appreciated. Thanks, Adam The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any.