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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