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
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