Success with asterisk 1.4 calls proceed past 30 min.
Gabrial Casey Telecommunications Franklin American Mortgage Company Direct:615-468- 2945 Cell: 615-693-2833 Email:gca...@franklinamerican.com ----- Original Message ----- From: "mkitchin public" <mkitchin.pub...@gmail.com> To: "Gabe Casey" <gca...@franklinamerican.com> Sent: Saturday, December 5, 2009 12:59:09 PM Subject: Re: [sipx-users] 30 min disconnects on calls originated from polycom Nice to see someone else in Nashville on here. I'm starting a Sipx conversion. I work for DSI http://www.dsi-corp.com/ -Matthew Kitchin Gabe Casey wrote: > yeah its asterisk 1.6.2.0-rc4 which is something i am willing to > downgrade at anytime > > I have a call trace but its at work right now. I will supply it as > soon as i get there however the dialog was > normal until exactly 30min into the call when SBC sends another invite > and asterisk replies with a 488 and then the call ends. > > *Gabrial Casey* > /Telecommunications/ > Franklin American Mortgage Company > Direct:615-468-2945 > Cell: 615-693-2833 > Email:gca...@franklinamerican.com > > > ----- Original Message ----- > From: "Tony Graziano" <tgrazi...@myitdepartment.net> > To: "Gabe Casey" <gca...@franklinamerican.com> > Cc: sipx-users@list.sipfoundry.org > Sent: Saturday, December 5, 2009 10:32:09 AM > Subject: Re: [sipx-users] 30 min disconnects on calls originated from > polycom > > 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 <mailto: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 > <mailto:email%3agca...@franklinamerican.com> > > > ----- Original Message ----- > From: "Tony Graziano" <tgrazi...@myitdepartment.net > <mailto:tgrazi...@myitdepartment.net>> > To: gca...@franklinamerican.com > <mailto:gca...@franklinamerican.com>, > sipx-users@list.sipfoundry.org <mailto: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 > <mailto: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 > <mailto:sipx-users-boun...@list.sipfoundry.org> > <sipx-users-boun...@list.sipfoundry.org > <mailto:sipx-users-boun...@list.sipfoundry.org>> > To: sipx-users <sipx-users@list.sipfoundry.org > <mailto: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 > <mailto:email%3agca...@franklinamerican.com> > > > > > -- > ====================== > Tony Graziano, Manager > Telephone: 434.984.8430 > Fax: 434.984.8431 > > Email: tgrazi...@myitdepartment.net <mailto: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/ > > ------------------------------------------------------------------------ > > _______________________________________________ > sipx-users mailing list sipx-users@list.sipfoundry.org > List Archive: http://list.sipfoundry.org/archive/sipx-users > Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users > sipXecs IP PBX -- http://www.sipfoundry.org/
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