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William Stillwell (Lists)

2010-Jan-07 16:15 UTC

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

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

Steve Totaro

2010-Jan-07 16:23 UTC

head link

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

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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Olle E. Johansson

2010-Jan-07 16:34 UTC

head link

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

7 jan 2010 kl. 17.15 skrev William Stillwell (Lists):
> 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.Check the "allowtransfer" setting in sip.conf.

/Olle
> 
> 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
> 
> 
> 
> 
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> asterisk-users mailing list
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---
* Olle E Johansson - oej at edvina.net
* Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden

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