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>REFER <a href="mailto:sip%3A1101@192.168.1.25">sip:1101@192.168.1.25</a> SIP/2.0<br>
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>>;tag=as32ed6c48<br>
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>Referred-By: <<a href="mailto:sip%3A7500@192.168.1.25">sip:7500@192.168.1.25</a>><br>
Call-ID: <a href="mailto:4d6a024a07f2b0f904a3cfe26360e58e@192.168.1.25">4d6a024a07f2b0f904a3cfe26360e58e@192.168.1.25</a><br>CSeq: 34526 REFER<br>User-Agent: Grandstream BT200 1.1.6.46<br>Max-Forwards: 70<br>Allow: 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> got a SIP call 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>localhost*CLI><br><--- Transmitting (NAT) to <a href="http://192.168.1.30:5060">192.168.1.30:5060</a> ---><br>
SIP/2.0 202 Accepted<br>Via: SIP/2.0/UDP 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>>;tag=as32ed6c48<br>
Call-ID: <a href="mailto:4d6a024a07f2b0f904a3cfe26360e58e@192.168.1.25">4d6a024a07f2b0f904a3cfe26360e58e@192.168.1.25</a><br>CSeq: 34526 REFER<br>User-Agent: Asterisk PBX<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>
Supported: replaces<br>Contact: <<a href="mailto:sip%3A1101@192.168.1.25">sip:1101@192.168.1.25</a>><br>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 clear="all">Thanks,<br>Max Alex<br>Voip Developer<br><br>
<br><br><div class="gmail_quote">On Wed, Apr 8, 2009 at 2:04 AM, 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;">
<div class="im">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>
</div>Does it work with other phones? That means is it a Grandstream isue or a<br>
general issue?<br>
<div class="im"><br>
> I have updated the latest firmware to the phone.<br>
> The phone is sending the *refer* to asterisk but asterisk is not able to<br>
> connect with the another call<br>
<br>
</div>Why? some log messages would help us helping you.<br>
<div class="im"><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>
</div>Please ask again on the user mailing lists and provide some log messages<br>
<div class="im"><br>
> Thanks in advance!!<br>
><br>
> Thanks,<br>
> Max Alex<br>
> Voip Developer<br>
><br>
><br>
</div>> ------------------------------------------------------------------------<br>
<div class="im">><br>
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