First off kill the Glaw. It doesn't exist. Then try your call. But also why are you sending the line congestion when you first start to make a call. That's normally used as a closure.
But from what I can see about the only thing wrong is the GLAW. Kill that and you should be good to go. ..o-------------------------------------------------------o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of JP Russell Sent: Tuesday, July 12, 2005 5:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Unable to dial certain calls Of course. Note that I have no idea what "glaw" is but someone on some board shttps://paranoia.demon.nl/SkinFiles/jpruss.com/GoldenFleece/send.gif https://paranoia.demon.nl/SkinFiles/jpruss.com/GoldenFleece/send.gifugge sted it as a resolution to a similiar problem so I put it in. The entry from the iax.conf file is: [vbx] type=peer host= 213.61.187.150 secret=-my password- notransfer=yes context=def allow=glaw allow=ulaw allow=gsm and from extensions.conf I guess you need the [def] context entries. they are: ;NL exten => _00316.,1,Congestion exten => _00319.,1,Congestion exten => _0031X.,1,SetCallerID("Not Available" <7005551212>) exten => _0031X.,2,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN} exten => _0031X.,3,Playback(invalid) exten => _0031X.,4,Hangup ;US exten => _001X.,1,SetCallerID("Not Available" <7005551212>) exten => _001X.,2,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN} exten => _001X.,3,Playback(invalid) exten => _001X.,4,Hangup Finally sip.conf includes the below paramaters: [general] disallow=all allow=ulaw allow=glaw allow=gsm port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = from-sip ; Default for incoming calls callerid=No CallID [2203] port=5061 username=-thisusername- secret=-this password- host=dynamic type=friend nat=1 qualify=no ;reinvite=no canreinvite=yes context=intern On Mon, 11 Jul 2005 22:55:49 -0400 "Brian C. Fertig" <[EMAIL PROTECTED]> wrote: > Check your codecs.. Can you post a sniplet of your IAX, >SIP, and extensions.conf for dialing the US so we can see >were the problem may lie? > > Brian Fertig > > > ________________________________ > >From: [EMAIL PROTECTED] on behalf >of JP Russell > Sent: Mon 7/11/2005 9:12 PM > To: [email protected] > Subject: [Asterisk-Users] Unable to dial certain calls > > > > To begin with, I am new to both asterisk and VOIP and >although I've > gotten pretty far with my Asterisk setup and have two >different sip > accounts working fine for outgoing calls I am having >trouble with one > issue. > > My problem is that I have another provider who uses IAX2 >that I wish > to use for calling various countries, including local >(The > Netherlands) calls and calls to the US to name two. I >am able to > call local numbers without a problem through this >provider with > Asterisk, but calling US numbers is not working. > > I CAN call the same US numbers with the service by using >a direct > connection from a softphone for example. > > The entries that show up in the log after failed >attempts to call the > US are: > > Jul 11 20:04:04 WARNING[25225728]: File channel.c, Line >1851 > (ast_channel_make_compatible): No path to translate from >SIP/2203-2929 > (4) to IAX2[vbx]/1(16) > Jul 11 20:04:04 WARNING[25225728]: File app_dial.c, Line >672 > (dial_exec): Had to drop call because I couldn't make >SIP/2203-2929 > compatible with IAX2[vbx]/1 > > I don't see anything suspicious entries in the CLI >logging with IAX2 > debugging on. Searching the archives and google didn't >turn up a > solution to this or even point me in the right direction >I'm afraid. > > Anyone have any idea on what my problem is or I can go >for this issue? > > _______________________________________________ > Asterisk-Users mailing list > [email protected] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
