[asterisk-users] Sip REFER failes w/603 Decline (Policy), Polycom Phone

[Steve Totaro]

On Thu, Jan 7, 2010 at 11:15 AM, William Stillwell (Lists) <
william.stillwell-lists at ablebody.net> 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:1050 at XXX.XXX.232.175 SIP/2.0
> Via: SIP/2.0/UDP XXX.XXX.232.66:8986;branch=z9hG4bK3dc3ce44DF7EE83D
> From: "1051" <sip:1051 at XXX.XXX.232.66:8986>;tag=D117C080-6FFBC539
> To: "1050" <sip:1050 at XXX.XXX.232.175>;tag=as140f4415
> CSeq: 1 ACK
> Call-ID: 75235b51626f63440c264f6b70dc5718 at XXX.XXX.232.175
> Contact: <sip:1051 at XXX.XXX.232.66:8986>
> 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:1050 at XXX.XXX.232.175 SIP/2.0
> Via: SIP/2.0/UDP XXX.XXX.232.66:8986;branch=z9hG4bKd2064841D6E30B4A
> From: "1051" <sip:1051 at XXX.XXX.232.66:8986>;tag=D117C080-6FFBC539
> To: "1050" <sip:1050 at XXX.XXX.232.175>;tag=as140f4415
> CSeq: 2 REFER
> Call-ID: 75235b51626f63440c264f6b70dc5718 at XXX.XXX.232.175
> Contact: <sip:1051 at XXX.XXX.232.66:8986>
> User-Agent: PolycomSoundPointIP-SPIP_331-UA/3.2.2.0477
> Accept-Language: en
> Refer-To: sip:202010 at XXX.XXX.232.175;user=phone
> Referred-By: <sip:1051 at XXX.XXX.232.175>
> Max-Forwards: 70
> Content-Length: 0
>
>
> <------------->
> --- (13 headers 0 lines) ---
> Call 75235b51626f63440c264f6b70dc5718 at XXX.XXX.232.175 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:1051 at XXX.XXX.232.66:8986>;tag=D117C080-6FFBC539
> To: "1050" <sip:1050 at XXX.XXX.232.175>;tag=as140f4415
> Call-ID: 75235b51626f63440c264f6b70dc5718 at XXX.XXX.232.175
> CSeq: 2 REFER
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
> Supported: replaces
> Contact: <sip:1050 at XXX.XXX.232.175>
> 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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