Hi All,
Thanks for your suggestions.
I am using DeadAgi application for origination of calls, i have set same
context to Transfer context.
I have also added Tt options in dial options.
When I am receiving calls to grandstream phone,
I am using transfer button to transfer the call, but it is not transfering
with AGI application,
Can anyone provides me suggestions for blind transfer with AGI application.
My Dialplan is given Below. I have used PHPAGI for the origination of calls.
[bt200]
exten => _X.,1,Set(__TRANSFER_CONTEXT=bt200)
exten => _X.,n,DeadAGI(testing_agi/testing.php)
exten=> h,1,NoOp(${DIALSTATUS})
Thanks,
Max Alex
Voip Developer
On Wed, Apr 8, 2009 at 9:47 PM, Klaus Darilion <[email protected]
> wrote:
> Haven't you read my email?
>
> 1. Wrong list
> 2. Missing log entries (set debug 4, set verbose 4)
>
> klaus
>
> Max Alex schrieb:
> > Hi All,
> > Thanks for your reply.
> > I got this refer message in asterisk.
> > but there is not any active channel of blind transfer.
> > ----------------------
> > REFER sip:[email protected] <sip%[email protected]> <mailto:
> sip%[email protected] <sip%[email protected]>> SIP/2.0
> > Via: SIP/2.0/UDP 192.168.1.30:5060;branch=z9hG4bK5880efa5cca586b0
> > From: <sip:[email protected]:5060;transport=udp>;tag=3699e1bcbed17687
> > To: "1101" <sip:[email protected] <sip%[email protected]>
> > <mailto:sip%[email protected] <sip%[email protected]>
> >>;tag=as32ed6c48
> > Contact: <sip:[email protected]:5060;transport=udp>
> > Supported: replaces, path
> > Refer-To: <sip:[email protected] <sip%[email protected]>
> > <mailto:sip%[email protected]<sip%[email protected]>
> >>
> > Referred-By: <sip:[email protected] <sip%[email protected]> <mailto:
> sip%[email protected] <sip%[email protected]>>>
> > Call-ID: [email protected]
> > <mailto:[email protected]>
> > CSeq: 34526 REFER
> > User-Agent: Grandstream BT200 1.1.6.46
> > Max-Forwards: 70
> > Allow:
> > INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
> > Content-Length: 0
> >
> > <------------->
> > --- (14 headers 0 lines) ---
> > Call [email protected]
> > <mailto:[email protected]> got a SIP call
> > transfer from caller: (REFER)!
> > SIP transfer to extension 1631xxxx...@outgoing by [email protected]
> > <mailto:[email protected]>
> > localhost*CLI>
> > <--- Transmitting (NAT) to 192.168.1.30:5060 <http://192.168.1.30:5060>
> --->
> > SIP/2.0 202 Accepted
> > Via: SIP/2.0/UDP
> > 192.168.1.30:5060;branch=z9hG4bK5880efa5cca586b0;received=192.168.1.30
> > From: <sip:[email protected]:5060;transport=udp>;tag=3699e1bcbed17687
> > To: "1101" <sip:[email protected] <sip%[email protected]>
> > <mailto:sip%[email protected] <sip%[email protected]>
> >>;tag=as32ed6c48
> > Call-ID: [email protected]
> > <mailto:[email protected]>
> > CSeq: 34526 REFER
> > User-Agent: Asterisk PBX
> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> > Supported: replaces
> > Contact: <sip:[email protected] <sip%[email protected]> <mailto:
> sip%[email protected] <sip%[email protected]>>>
> > Content-Length: 0
> >
> >
> > <------------>
> > ----------------------------------------
> > Is there any options we need to enable in asterisk or grandstream phone?
> > I have already user transfer option 'Tt' in dialplan of this.
> > Please provide me some help.
> > Thanks in advance!!
> >
> > Thanks,
> > Max Alex
> > Voip Developer
> >
> >
> >
> > On Wed, Apr 8, 2009 at 2:04 AM, Klaus Darilion
> > <[email protected] <mailto:[email protected]>>
> wrote:
> >
> > Max Alex wrote:
> > > Hi All,
> > > I have working asterisk version 1.4.24.
> > > I have a blind transfer issue with grandstream bt200.
> >
> > Does it work with other phones? That means is it a Grandstream isue
> or a
> > general issue?
> >
> > > I have updated the latest firmware to the phone.
> > > The phone is sending the *refer* to asterisk but asterisk is not
> > able to
> > > connect with the another call
> >
> > Why? some log messages would help us helping you.
> >
> > > that i have checked in sip debug.
> > > I am using transfer button of the grandstream phone.
> > > Can anybody provide help for this issue?
> >
> > Please ask again on the user mailing lists and provide some log
> messages
> >
> > > Thanks in advance!!
> > >
> > > Thanks,
> > > Max Alex
> > > Voip Developer
> > >
> > >
> > >
> >
> ------------------------------------------------------------------------
> > >
> > > _______________________________________________
> > > -- Bandwidth and Colocation Provided by
> http://www.api-digital.com --
> > >
> > > asterisk-users mailing list
> > > To UNSUBSCRIBE or update options visit:
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> > _______________________________________________
> > --Bandwidth and Colocation Provided by http://www.api-digital.com--
> >
> > asterisk-dev mailing list
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-dev
> >
> >
> >
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
