[asterisk-users] [asterisk-dev] Grandstream blind transfer issue
Max Alex
max.asterisk at gmail.com
Sat Apr 11 08:44:16 CDT 2009
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 <klaus.mailinglists at pernau.at
> 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:1101 at 192.168.1.25 <sip%3A1101 at 192.168.1.25> <mailto:
> sip%3A1101 at 192.168.1.25 <sip%253A1101 at 192.168.1.25>> SIP/2.0
> > Via: SIP/2.0/UDP 192.168.1.30:5060;branch=z9hG4bK5880efa5cca586b0
> > From: <sip:7500 at 192.168.1.30:5060;transport=udp>;tag=3699e1bcbed17687
> > To: "1101" <sip:1101 at 192.168.1.25 <sip%3A1101 at 192.168.1.25>
> > <mailto:sip%3A1101 at 192.168.1.25 <sip%253A1101 at 192.168.1.25>
> >>;tag=as32ed6c48
> > Contact: <sip:7500 at 192.168.1.30:5060;transport=udp>
> > Supported: replaces, path
> > Refer-To: <sip:1631XXXXXXX at 192.168.1.25 <sip%3A1631XXXXXXX at 192.168.1.25>
> > <mailto:sip%3A1631XXXXXXX at 192.168.1.25<sip%253A1631XXXXXXX at 192.168.1.25>
> >>
> > Referred-By: <sip:7500 at 192.168.1.25 <sip%3A7500 at 192.168.1.25> <mailto:
> sip%3A7500 at 192.168.1.25 <sip%253A7500 at 192.168.1.25>>>
> > Call-ID: 4d6a024a07f2b0f904a3cfe26360e58e at 192.168.1.25
> > <mailto:4d6a024a07f2b0f904a3cfe26360e58e at 192.168.1.25>
> > 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 4d6a024a07f2b0f904a3cfe26360e58e at 192.168.1.25
> > <mailto:4d6a024a07f2b0f904a3cfe26360e58e at 192.168.1.25> got a SIP call
> > transfer from caller: (REFER)!
> > SIP transfer to extension 1631XXXXXXX at outgoing by 7500 at 192.168.1.25
> > <mailto:7500 at 192.168.1.25>
> > 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:7500 at 192.168.1.30:5060;transport=udp>;tag=3699e1bcbed17687
> > To: "1101" <sip:1101 at 192.168.1.25 <sip%3A1101 at 192.168.1.25>
> > <mailto:sip%3A1101 at 192.168.1.25 <sip%253A1101 at 192.168.1.25>
> >>;tag=as32ed6c48
> > Call-ID: 4d6a024a07f2b0f904a3cfe26360e58e at 192.168.1.25
> > <mailto:4d6a024a07f2b0f904a3cfe26360e58e at 192.168.1.25>
> > CSeq: 34526 REFER
> > User-Agent: Asterisk PBX
> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> > Supported: replaces
> > Contact: <sip:1101 at 192.168.1.25 <sip%3A1101 at 192.168.1.25> <mailto:
> sip%3A1101 at 192.168.1.25 <sip%253A1101 at 192.168.1.25>>>
> > 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
> > <klaus.mailinglists at pernau.at <mailto:klaus.mailinglists at pernau.at>>
> 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
> > >
> > >
> > >
> >
> ------------------------------------------------------------------------
> > >
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