<br><br><div class="gmail_quote">On Thu, Jan 7, 2010 at 11:15 AM, William Stillwell (Lists) <span dir="ltr"><<a href="mailto:william.stillwell-lists@ablebody.net">william.stillwell-lists@ablebody.net</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;">I have several sip stations that on a that are on a nat'd network behind a<br>
nice friend firewall.. no audio path issues, 2 way audio works, etc,etc,etc.<br>
<br>
<br>
However, I can't get any of my phones to Transfer or Blind Transfer..<br>
<br>
I search and search, and well, just about gone nuts on this one.<br>
<br>
Here is sip debug from pressing "transfer->blind->dial dest->Dial Key" (note<br>
both stations do have access tot eh dial-dst ext of 202010)<br>
<br>
<------------><br>
-- Started music on hold, class 'default', on channel<br>
'SIP/1050-0a6ffa70'<br>
<--- SIP read from XXX.XXX.232.66:8986 ---><br>
ACK sip:1050@XXX.XXX.232.175 SIP/2.0<br>
Via: SIP/2.0/UDP XXX.XXX.232.66:8986;branch=z9hG4bK3dc3ce44DF7EE83D<br>
From: "1051" <sip:1051@XXX.XXX.232.66:8986>;tag=D117C080-6FFBC539<br>
To: "1050" <sip:1050@XXX.XXX.232.175>;tag=as140f4415<br>
CSeq: 1 ACK<br>
Call-ID: 75235b51626f63440c264f6b70dc5718@XXX.XXX.232.175<br>
Contact: <sip:1051@XXX.XXX.232.66:8986><br>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY,<br>
PRACK, UPDATE, REFER<br>
User-Agent: PolycomSoundPointIP-SPIP_331-UA/3.2.2.0477<br>
Accept-Language: en<br>
Max-Forwards: 70<br>
Content-Length: 0<br>
<br>
<br>
<-------------><br>
--- (12 headers 0 lines) ---<br>
<--- SIP read from XXX.XXX.232.66:8986 ---><br>
REFER sip:1050@XXX.XXX.232.175 SIP/2.0<br>
Via: SIP/2.0/UDP XXX.XXX.232.66:8986;branch=z9hG4bKd2064841D6E30B4A<br>
From: "1051" <sip:1051@XXX.XXX.232.66:8986>;tag=D117C080-6FFBC539<br>
To: "1050" <sip:1050@XXX.XXX.232.175>;tag=as140f4415<br>
CSeq: 2 REFER<br>
Call-ID: 75235b51626f63440c264f6b70dc5718@XXX.XXX.232.175<br>
Contact: <sip:1051@XXX.XXX.232.66:8986><br>
User-Agent: PolycomSoundPointIP-SPIP_331-UA/3.2.2.0477<br>
Accept-Language: en<br>
Refer-To: sip:202010@XXX.XXX.232.175;user=phone<br>
Referred-By: <sip:1051@XXX.XXX.232.175><br>
Max-Forwards: 70<br>
Content-Length: 0<br>
<br>
<br>
<-------------><br>
--- (13 headers 0 lines) ---<br>
Call 75235b51626f63440c264f6b70dc5718@XXX.XXX.232.175 got a SIP call<br>
transfer from caller: (REFER)!<br>
<--- Transmitting (no NAT) to XXX.XXX.232.66:8986 ---><br>
SIP/2.0 603 Declined (policy)<br>
Via: SIP/2.0/UDP<br>
XXX.XXX.232.66:8986;branch=z9hG4bKd2064841D6E30B4A;received=XXX.XXX.232.66<br>
From: "1051" <sip:1051@XXX.XXX.232.66:8986>;tag=D117C080-6FFBC539<br>
To: "1050" <sip:1050@XXX.XXX.232.175>;tag=as140f4415<br>
Call-ID: 75235b51626f63440c264f6b70dc5718@XXX.XXX.232.175<br>
CSeq: 2 REFER<br>
User-Agent: Asterisk PBX<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO<br>
Supported: replaces<br>
Contact: <sip:1050@XXX.XXX.232.175><br>
Content-Length: 0<br>
<br>
<br>
<------------><br>
-- Stopped music on hold on SIP/1050-0a6ffa70<br>
<br></blockquote><div><br> Do you have notransfer=yes and canreinvite=no set anywhere? Just a shot in the dark.<br><br>Thanks,<br>Steve Totaro<br></div></div>