It's a local CLEC, Essex Telcom. The burden does lie with them, but I doubt they'll fix it since if you provision a grandstream, it works just fine.
I have a total of 5 numbers with them. Two are on the server that is experiencing issues. Another is on a different server with no issues. The remaining two aren't provisioned anywhere. I'm going to be adding another number shortly. ----- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -------------------------------------------------- From: "Steve Totaro" <[email protected]> Sent: Tuesday, February 10, 2009 9:50 AM To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[email protected]> Subject: Re: [asterisk-users] Asterisk - Trixbox > How many accounts do you have? If just one, then a single peer should > be fine but they should be sending the destination exten as a DID, > obviously they are not. > > I think the burden of fixing it lies with them? What carrier is this? > > > > On Tue, Feb 10, 2009 at 10:09 AM, Mike Hammett > <[email protected]> wrote: >> I disabled that last number's registration and moved to a new number (to >> test each number individually without the sip debugging from the others). >> I >> waited maybe 5 minutes and I restarted Asterisk to ensure the other side >> was >> done with whatever it was doing. I called the second number (8152641125) >> and the first number (8159911010) shows up as the peer. Not only that, >> but >> with this number, there's no compatible codecs. I ensured that both >> entries >> in sip.conf were the same other than things that needed to be different >> such >> as username. I even had that entry have allow=all. I still get the >> codec >> error. >> >> http://pastebin.com/f5b826d62 I highlighted the lines of interest. 34 >> is >> the peer issue whereas 42 is the codec issue. >> >> >> ----- >> Mike Hammett >> Intelligent Computing Solutions >> http://www.ics-il.com >> >> >> >> -------------------------------------------------- >> From: "Steve Totaro" <[email protected]> >> Sent: Tuesday, February 10, 2009 7:29 AM >> To: "Asterisk Users Mailing List - Non-Commercial Discussion" >> <[email protected]> >> Subject: Re: [asterisk-users] Asterisk - Trixbox >> >>> Mike, >>> >>> Please explain the problem more clearly and post a pastebin that shows >>> the problem and only the problem, not a huge SIP dump. >>> >>> If you could point out the line numbers where you suspect an issue. >>> >>> Thanks, >>> Steve >>> >>> On Mon, Feb 9, 2009 at 10:56 PM, Mike Hammett >>> <[email protected]> >>> wrote: >>>> Can anyone help me determine where the problem lies and how to fix it? >>>> >>>> >>>> ----- >>>> Mike Hammett >>>> Intelligent Computing Solutions >>>> http://www.ics-il.com >>>> >>>> >>>> From: Mike Hammett >>>> Sent: Thursday, January 15, 2009 1:00 PM >>>> To: [email protected] >>>> Subject: [asterisk-users] Asterisk - Trixbox >>>> My provider migrated from an old EOL softswitch to Trixbox. >>>> >>>> I have a number (8159093011) on a different server on a different >>>> network. >>>> It appears as though the incoming calls are trying to authenticate >>>> against >>>> that number, which isn't present on the box. Could someone help me >>>> decode >>>> this debugging output? I was calling 8159911010. My server is >>>> 208.100.1.33. Theirs is 208.1.87.235. I solved the s@ problem on the >>>> other >>>> server by adding insecure settings, but that didn't seem to solve it on >>>> this >>>> one. >>>> >>>> http://pastebin.com/f5151341f >>>> >>>> >>>> ----- >>>> Mike Hammett >>>> Intelligent Computing Solutions >>>> http://www.ics-il.com >>>> >>> >>> _______________________________________________ >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>> >>> asterisk-users mailing list >>> To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> >> >> _______________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > > -- > Thanks, > Steve Totaro > +18887771888 (Toll Free) > +12409381212 (Cell) > +12024369784 (Skype) > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
