Yes, I have both and will do my part to help the asterisk community. However to anyone currently experiencing these issues please note my working configuration as: sipxecs 4.0.4 Using Sip Trunking on sipXbridge sangoma a108d firmware v39 configured TDM Voice Asterisk 1.4.26.3 as media gateway for PRI Trunks
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> Sent: Sunday, December 6, 2009 12:15:37 PM Subject: Re: [sipx-users] 30 min disconnects on calls originated from polycom So the question is, if you can get call traces for both versions, whether to follow up with it to discover if it is a bug at sipx or asterisk... On Sun, Dec 6, 2009 at 1:09 PM, Gabe Casey < gca...@franklinamerican.com > wrote: 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/ _______________________________________________ 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/ -- ====================== 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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