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