Clarify the phone is using firmware 4.2 and firmware 3.1.3. Also, that you
did not apply any patches (sipxbridge) after applying 4.0.4. Lastly explain
how the caller is sending you the call (via itsp through sipxbridge).

I think the 30 minute issue was fixed as of 4.0.4, if possible reference the
jira issue you suspect.

Realize asterisk (pre 1.6) is problematic. You should consider a trace from
both sides to determine where the issue is. In using an itsp with a siptrunk
(no asterisk gateway), I don't see these issues on 4.0.4.
============================
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431

Email: tgrazi...@myitdepartment.net

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: sipx-users-boun...@list.sipfoundry.org
<sipx-users-boun...@list.sipfoundry.org>
To: sipx-users <sipx-users@list.sipfoundry.org>
Sent: Sat Dec 05 10:43:23 2009
Subject: [sipx-users] 30 min disconnects on calls originated from polycom

I have seen a couple of open tickets on the 30 min session reinvite issue.
This causes the media gateway to respond with a 488 and then a disconnect.
What are some possible configuration options here ? It happens only on
outbound calls

polycom 450 frimware 1.3.1
sipxecs 4.04
asterisk / sangoma (media gateway)
not nat
trunks built with sipXbridge as sip trunks


At 30 min sipxecs sends a sip/sdp invite to refresh the session ( i guess
... i thought this was handled with and update ??) in which the media gw
responds "not allowed here" 488

sipxecs says Bye
gw says ack
gw says bye

call is over .... caller is mad at me :(


Warm Regards





Gabrial Casey
Telecommunications
Franklin American Mortgage Company
Direct:615-468- 2945
Cell: 615-693-2833
Email:gca...@franklinamerican.com
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