Hi All,<br>Thanks for your suggestions.<br>I am using DeadAgi application for origination of calls, i have set same context to Transfer context.<br>I have also added Tt options in dial options.<br>When I am receiving calls to grandstream phone, <br>
I am using transfer button to transfer the call, but it is not transfering with AGI application,<br>Can anyone provides me suggestions for blind transfer with AGI application.<br><br>My Dialplan is given Below. I have used PHPAGI for the origination of calls.<br>
[bt200]<br>exten => _X.,1,Set(__TRANSFER_CONTEXT=bt200)<br>exten => _X.,n,DeadAGI(testing_agi/testing.php)<br>exten=> h,1,NoOp(${DIALSTATUS})<br><br><br clear="all">Thanks,<br>Max Alex<br>Voip Developer<br><br>
<br><br><div class="gmail_quote">On Wed, Apr 8, 2009 at 9:47 PM, Klaus Darilion <span dir="ltr"><<a href="mailto:klaus.mailinglists@pernau.at">klaus.mailinglists@pernau.at</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Haven't you read my email?<br>
<br>
1. Wrong list<br>
2. Missing log entries (set debug 4, set verbose 4)<br>
<br>
klaus<br>
<br>
Max Alex schrieb:<br>
<div class="im">> Hi All,<br>
> Thanks for your reply.<br>
> I got this refer message in asterisk.<br>
> but there is not any active channel of blind transfer.<br>
> ----------------------<br>
</div>> REFER <a href="mailto:sip%3A1101@192.168.1.25">sip:1101@192.168.1.25</a> <mailto:<a href="mailto:sip%253A1101@192.168.1.25">sip%3A1101@192.168.1.25</a>> SIP/2.0<br>
<div class="im">> Via: SIP/2.0/UDP 192.168.1.30:5060;branch=z9hG4bK5880efa5cca586b0<br>
> From: <sip:7500@192.168.1.30:5060;transport=udp>;tag=3699e1bcbed17687<br>
> To: "1101" <<a href="mailto:sip%3A1101@192.168.1.25">sip:1101@192.168.1.25</a><br>
</div>> <mailto:<a href="mailto:sip%253A1101@192.168.1.25">sip%3A1101@192.168.1.25</a>>>;tag=as32ed6c48<br>
<div class="im">> Contact: <sip:7500@192.168.1.30:5060;transport=udp><br>
> Supported: replaces, path<br>
> Refer-To: <<a href="mailto:sip%3A1631XXXXXXX@192.168.1.25">sip:1631XXXXXXX@192.168.1.25</a><br>
</div>> <mailto:<a href="mailto:sip%253A1631XXXXXXX@192.168.1.25">sip%3A1631XXXXXXX@192.168.1.25</a>>><br>
> Referred-By: <<a href="mailto:sip%3A7500@192.168.1.25">sip:7500@192.168.1.25</a> <mailto:<a href="mailto:sip%253A7500@192.168.1.25">sip%3A7500@192.168.1.25</a>>><br>
<div class="im">> Call-ID: <a href="mailto:4d6a024a07f2b0f904a3cfe26360e58e@192.168.1.25">4d6a024a07f2b0f904a3cfe26360e58e@192.168.1.25</a><br>
</div>> <mailto:<a href="mailto:4d6a024a07f2b0f904a3cfe26360e58e@192.168.1.25">4d6a024a07f2b0f904a3cfe26360e58e@192.168.1.25</a>><br>
<div class="im">> CSeq: 34526 REFER<br>
> User-Agent: Grandstream BT200 1.1.6.46<br>
> Max-Forwards: 70<br>
> Allow:<br>
> INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK<br>
> Content-Length: 0<br>
><br>
> <-------------><br>
> --- (14 headers 0 lines) ---<br>
> Call <a href="mailto:4d6a024a07f2b0f904a3cfe26360e58e@192.168.1.25">4d6a024a07f2b0f904a3cfe26360e58e@192.168.1.25</a><br>
</div>> <mailto:<a href="mailto:4d6a024a07f2b0f904a3cfe26360e58e@192.168.1.25">4d6a024a07f2b0f904a3cfe26360e58e@192.168.1.25</a>> got a SIP call<br>
<div class="im">> transfer from caller: (REFER)!<br>
> SIP transfer to extension 1631XXXXXXX@outgoing by <a href="mailto:7500@192.168.1.25">7500@192.168.1.25</a><br>
</div>> <mailto:<a href="mailto:7500@192.168.1.25">7500@192.168.1.25</a>><br>
> localhost*CLI><br>
> <--- Transmitting (NAT) to <a href="http://192.168.1.30:5060" target="_blank">192.168.1.30:5060</a> <<a href="http://192.168.1.30:5060" target="_blank">http://192.168.1.30:5060</a>> ---><br>
<div class="im">> SIP/2.0 202 Accepted<br>
> Via: SIP/2.0/UDP<br>
> 192.168.1.30:5060;branch=z9hG4bK5880efa5cca586b0;received=192.168.1.30<br>
> From: <sip:7500@192.168.1.30:5060;transport=udp>;tag=3699e1bcbed17687<br>
> To: "1101" <<a href="mailto:sip%3A1101@192.168.1.25">sip:1101@192.168.1.25</a><br>
</div>> <mailto:<a href="mailto:sip%253A1101@192.168.1.25">sip%3A1101@192.168.1.25</a>>>;tag=as32ed6c48<br>
<div class="im">> Call-ID: <a href="mailto:4d6a024a07f2b0f904a3cfe26360e58e@192.168.1.25">4d6a024a07f2b0f904a3cfe26360e58e@192.168.1.25</a><br>
</div>> <mailto:<a href="mailto:4d6a024a07f2b0f904a3cfe26360e58e@192.168.1.25">4d6a024a07f2b0f904a3cfe26360e58e@192.168.1.25</a>><br>
<div class="im">> CSeq: 34526 REFER<br>
> User-Agent: Asterisk PBX<br>
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>
> Supported: replaces<br>
</div>> Contact: <<a href="mailto:sip%3A1101@192.168.1.25">sip:1101@192.168.1.25</a> <mailto:<a href="mailto:sip%253A1101@192.168.1.25">sip%3A1101@192.168.1.25</a>>><br>
<div class="im">> Content-Length: 0<br>
><br>
><br>
> <------------><br>
> ----------------------------------------<br>
> Is there any options we need to enable in asterisk or grandstream phone?<br>
> I have already user transfer option 'Tt' in dialplan of this.<br>
> Please provide me some help.<br>
> Thanks in advance!!<br>
><br>
> Thanks,<br>
> Max Alex<br>
> Voip Developer<br>
><br>
><br>
><br>
> On Wed, Apr 8, 2009 at 2:04 AM, Klaus Darilion<br>
</div><div><div></div><div class="h5">> <<a href="mailto:klaus.mailinglists@pernau.at">klaus.mailinglists@pernau.at</a> <mailto:<a href="mailto:klaus.mailinglists@pernau.at">klaus.mailinglists@pernau.at</a>>> wrote:<br>
><br>
> Max Alex wrote:<br>
> > Hi All,<br>
> > I have working asterisk version 1.4.24.<br>
> > I have a blind transfer issue with grandstream bt200.<br>
><br>
> Does it work with other phones? That means is it a Grandstream isue or a<br>
> general issue?<br>
><br>
> > I have updated the latest firmware to the phone.<br>
> > The phone is sending the *refer* to asterisk but asterisk is not<br>
> able to<br>
> > connect with the another call<br>
><br>
> Why? some log messages would help us helping you.<br>
><br>
> > that i have checked in sip debug.<br>
> > I am using transfer button of the grandstream phone.<br>
> > Can anybody provide help for this issue?<br>
><br>
> Please ask again on the user mailing lists and provide some log messages<br>
><br>
> > Thanks in advance!!<br>
> ><br>
> > Thanks,<br>
> > Max Alex<br>
> > Voip Developer<br>
> ><br>
> ><br>
> ><br>
> ------------------------------------------------------------------------<br>
> ><br>
> > _______________________________________________<br>
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</div></div></blockquote></div><br>