Trace of a failed call would be most helpful. Also, can you indicate what
version of asterisk you are using?

On Sat, Dec 5, 2009 at 11:27 AM, Gabe Casey <gca...@franklinamerican.com>wrote:

> This actually does not happen pstn -> gw -> sipxecs -> phone
> but happens consistently polycom -> sipxecs -> gw -> pstn
>
>
> *Gabrial Casey*
> *Telecommunications*
> Franklin American Mortgage Company
> Direct:615-468-2945
> Cell: 615-693-2833
> Email:gca...@franklinamerican.com <email%3agca...@franklinamerican.com>
>
>
> ----- Original Message -----
> From: "Tony Graziano" <tgrazi...@myitdepartment.net>
> To: gca...@franklinamerican.com, sipx-users@list.sipfoundry.org
> Sent: Saturday, December 5, 2009 9:49:07 AM
> Subject: Re: [sipx-users] 30 min disconnects on calls originated from
> polycom
>
> 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 <email%3agca...@franklinamerican.com>
>



-- 
======================
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/
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