Also I don't think it's a Polycom problem. This only happens when doing
attended transfer to FreeSWITCH services. Attended transfer to
everything else (including park) works fine.
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 7/16/2010 4:00 PM, Josh Patten wrote:
http://www.polycom.com/support/voice/soundpoint_ip/soundpoint_ip550.html
I don't see it
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 7/16/2010 3:49 PM, Tony Graziano wrote:
FWIW - Firmware 3.3.0 is now posted... though you may still have the
same problem.
On Fri, Jul 16, 2010 at 12:36 PM, Josh Patten
<[email protected] <mailto:[email protected]>> wrote:
I have now confirmed this is not a problem with the gateways. I
posted a ticket here:
http://track.sipfoundry.org/browse/XX-8652
Even with firmware 3.1.3revC this is still happening.
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 7/16/2010 10:29 AM, Tony Graziano wrote:
So the question still remains if it happens with firmware
3.13RevC.
Its the polycom complaining... 3.2 aint all that.
============================
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: [email protected]
<mailto:[email protected]>
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
Fax: 434.984.8427
Helpdesk Contract Customers:
http://www.myitdepartment.net/gethelp/
----- Original Message -----
From: [email protected]
<mailto:[email protected]>
<[email protected]
<mailto:[email protected]>>
To: [email protected]
<mailto:[email protected]><[email protected]
<mailto:[email protected]>>
Sent: Fri Jul 16 11:27:35 2010
Subject: Re: [sipx-users] Help with Patton gateway
EDIT - This also seems to be occuring with my Audiocodes
gateways as
well so apparently it's not isolated just to the Patton gateways.
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 7/16/2010 10:23 AM, Josh Patten wrote:
I'm forwarding the support request I sent to Patton
regarding a
problem with their gateways and sipX. Here is where the
engineer said
things are going wrong:
Line 1068, the smartnode sends BYE to polycom to ip
10.200.24.250 as
showed below:
23:39:24 SIP_TR> [STACK]> Stack: to 10.200.24.250
BYE sip:[email protected]
<mailto:sip%[email protected]>;x-sipX-nonat SIP/2.0
Via: SIP/2.0/UDP
10.200.50.11:5060;branch=z9hG4bKaba231ea2aa038eb1
Route:
<sip:10.200.24.250:5060;lr;sipXecs-CallDest=LOCL;sipXecs-rs=*auth~.*from~MTRCRjc0MkYtNzZERkIzMTY$60.900_ntap*id~Mjg4ODUtNTg4NA$60$60!df02a35b10ef9ba395dee26f5cb05618>
Max-Forwards: 70
From:<sip:[email protected]
<mailto:sip%[email protected]>;user=phone>;tag=1105402544
To: "Josh Patten"<sip:[email protected]
<mailto:sip%[email protected]>>;tag=14BF742F-76DFB316
Call-ID: [email protected]
<mailto:[email protected]>
<mailto:[email protected]
<mailto:[email protected]>>
CSeq: 12761 BYE
User-Agent: Patton SN4524 JO EUI 00a0ba05061C R5.T
2010-05-20 H323 SIP
FXS FXO M5T SIP Stack/4.0.29.29 <http://4.0.29.29>
Content-Length: 0
Line 1093, Polycom answered back with message error 481
as showed below:
23:39:24 SIP_TR> [STACK]< Stack: from 10.200.24.250
SIP/2.0 481 Call Leg/Transaction Does Not Exist
Via: SIP/2.0/UDP
10.200.50.11:5060;branch=z9hG4bKaba231ea2aa038eb1
From:<sip:[email protected]
<mailto:sip%[email protected]>;user=phone>;tag=1105402544
To: "Josh Patten"<sip:[email protected]
<mailto:sip%[email protected]>>;tag=14BF742F-76DFB316
Cseq: 12761 BYE
Call-Id: [email protected]
<mailto:[email protected]>
<mailto:[email protected]
<mailto:[email protected]>>
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.2.0477
Accept-Language: en
Content-Length: 0
Date: Tue, 13 Jul 2010 23:39:24 GMT
Do you know why the polycom is sending this message
instead to
terminate the call?
I honestly don't know why that's happening. Could someone
on this list
with a little more SIP knowledge point out where it's
going wrong?
I have attached the original debug and the original email
dialog. If I
need to get a snapshot let me know and I will.
-------- Original Message --------
Subject: Re: [Support #54038]: Consultative
(attended) transfer to
auto attendant in sipXecs causes incomplete transfer on phone
Date: Tue, 13 Jul 2010 18:47:06 -0500
From: Josh Patten<[email protected]
<mailto:[email protected]>>
To: [email protected] <mailto:[email protected]>
Debug is attached.
Here is the call scenario:
4676 calls 95745699
Patton strips the 9, dialing 5745699
Once connected, 4676 initiates a consultative (attended)
transfer to
4310 which is an auto attendant
After connected to the auto attendant, 4676 completes the
consultative
transfer. The call is transferred but appears to be on
hold on the
phone. The only way to clear this ghost call is to
un-hold then end
the call.
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 7/13/2010 2:44 PM, Patton Electronics Technical
Support wrote:
====== Please reply above this line ======
Hello Josh,
Thanks for contacting Patton Support.
Please run these debug commands via telnet and send
me the output as
a .txt file so we can see why the call is not being
disconnected:
enable
show running-config
show port fxo detail 5
debug fxo
debug ccfxo
debug call-router detail 5
debug call-control detail 5
debug context sip-gateway transport detail 5
debug context sip-gateway error detail 5
I have attached a debugging tutorial for reference.
Regards,
Daniel Lizaola
Technical Support Engineer
Patton Electronics Co
7622 Rickenbacker Drive
Gaithersburg MD 20879 USA
t: +1 301-975-1000
f: +1 301-869-9293
w: http://www.patton.com
Please consider your environmental responsibility
before printing
this e-mail.
Ticket History *Josh Patten* (Client) Posted On: 09
Jul 2010 09:54 PM
------------------------------------------------------------------------
Here is the dialing scenario laid out in the attached
debug:
3001 dials 95745699
Patton gateway strips 9 off and dials 5745699 on FXO
hunt group
Once connected, 3001 performs an attended transfer to
4310, an auto
attendant, by pressing transfer then dialing 4310
Once connected, 3001 presses transfer again to
complete the transfer.
5745699 is transfered to the auto attendant, but the
call on the
transferring phone is put on hold (even though it is
no longer an active
call). To end this "ghost call" the user has to
resume the ghost call
then hang up.
I have attached a debug and a copy of my
configuration. Please let me
know if you need anything else.
Attachments aa_debug.txt (496.75 KB)
SmartNode-4524.cfg (8.64 KB)
Ticket Details
Ticket ID: 54038
Department: Support for NA/LA/APAC
Priority: Standard
Status: Waiting for Response
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--
======================
Tony Graziano, Manager
Telephone: 434.984.8430
sip: [email protected]
<mailto:[email protected]>
Fax: 434.984.8431
Email: [email protected] <mailto:[email protected]>
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
sip: [email protected]
<mailto:[email protected]>
Fax: 434.984.8427
Helpdesk Contract Customers:
http://www.myitdepartment.net/gethelp/
Why do mathematicians always confuse Halloween and Christmas?
Because 31 Oct = 25 Dec.
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