Hello, I am playing with my new * install, and there are a couple of things that I don't understand, if someone could point me in the right direction it will be appreciated. I am trying to configure a voipstunt.com account to place outgoing calls, and this is my config. sip.conf: [voipstunt] type=friend ; (or "peer" if we don't need incoming calls, or if there is a separate section with "type=user") host=sip.voipstunt.com disallow=all allow=ulaw allow=alaw allow=gsm allow=g726 username=voipstuntuser_replacement fromuser=voipstuntuser_replacement secret=hiddenpassword qualify=1000 ; optional canreinvite=no ; new SIP servers don't like reINVITEs dtmfmode=inband ; only inband currently works, and not that well extensions.conf: [internal] exten => 787793,1,Dial(SIP/john) exten => 700099,1,Dial(SIP/maribel) exten => 100000,1,Dial(SIP/dieguez) exten => 100001,1,Dial(SIP/chparson) exten => _NXXXXXXXXXXX,1,Dial(SIP/+{EXTEN}@voipstunt) As stated in asteriskTFOT _NXXXXXXXXXXX will match 541152184829 which is the phone number of my place, which I am trying to place a call. I am asuming that the sign + in (SIP/+{EXTEN}@voipstunt) will be appended to what I press in my softphone. All I get when I call to 541152184829 is: -- Executing Dial("SIP/john-0819a010", "SIP/+{EXTEN}@voipstunt") in new stack -- Called +{EXTEN}@voipstunt Jul 22 20:15:45 NOTICE[1396]: chan_sip.c:1997 auto_congest: Auto-congesting SIP/voipstunt-0819f520 -- SIP/voipstunt-0819f520 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'SIP/john-0819a010' status is 'CONGESTION' Any idea? suggestion? Thanks in advance for any comment/help. Pablo
hey pablo i havent messed aorund with stun that much except as a repeator but maybe this is your problem ;] exten => _NXXXXXXXXXXX,1,Dial(SIP/+{EXTEN}@voipstunt) should be exten => _NXXXXXXXXXXX,1,Dial(SIP/+${EXTEN}@voipstunt) notice the $ sign in front of {EXTEN} that declares it a variable(haha like my oxymoron?) tells asteirsk to lookup the extensions dialed by the user> >Hello, I am playing with my new * install, and there are a couple of things >that I don't understand, if someone could point me in the right direction >it >will be appreciated. > >I am trying to configure a voipstunt.com account to place outgoing calls, >and this is my config. > >sip.conf: > >[voipstunt] >type=friend ; (or "peer" if we don't need incoming calls, or if there is a >separate section with "type=user") >host=sip.voipstunt.com >disallow=all >allow=ulaw >allow=alaw >allow=gsm >allow=g726 >username=voipstuntuser_replacement >fromuser=voipstuntuser_replacement >secret=hiddenpassword >qualify=1000 ; optional >canreinvite=no ; new SIP servers don't like reINVITEs >dtmfmode=inband ; only inband currently works, and not that well > >extensions.conf: > >[internal] >exten => 787793,1,Dial(SIP/john) >exten => 700099,1,Dial(SIP/maribel) >exten => 100000,1,Dial(SIP/dieguez) >exten => 100001,1,Dial(SIP/chparson) >exten => _NXXXXXXXXXXX,1,Dial(SIP/+{EXTEN}@voipstunt) > > >As stated in asteriskTFOT _NXXXXXXXXXXX will match 541152184829 which is >the >phone number of my place, which I am trying to place a call. > >I am asuming that the sign + in (SIP/+{EXTEN}@voipstunt) will be appended >to >what I press in my softphone. > >All I get when I call to 541152184829 is: > > -- Executing Dial("SIP/john-0819a010", "SIP/+{EXTEN}@voipstunt") in >new >stack > -- Called +{EXTEN}@voipstunt >Jul 22 20:15:45 NOTICE[1396]: chan_sip.c:1997 auto_congest: Auto-congesting >SIP/voipstunt-0819f520 > -- SIP/voipstunt-0819f520 is circuit-busy > == Everyone is busy/congested at this time (1:0/1/0) > == Auto fallthrough, channel 'SIP/john-0819a010' status is 'CONGESTION' > > >Any idea? suggestion? > >Thanks in advance for any comment/help. > >Pablo > >_______________________________________________ >--Bandwidth and Colocation provided by Easynews.com -- > >asterisk-users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users_________________________________________________________________ Don’t just search. Find. Check out the new MSN Search! http://search.msn.click-url.com/go/onm00200636ave/direct/01/
> exten => _NXXXXXXXXXXX,1,Dial(SIP/+{EXTEN}@voipstunt) > should be > exten => _NXXXXXXXXXXX,1,Dial(SIP/+${EXTEN}@voipstunt) > notice the $ sign in front of {EXTEN} that declares it a variable(hahalike> my oxymoron?) > tells asteirsk to lookup the extensions dialed by the userBrandon, that whas exactly the problem. Thank you very much!!! now I am calling :)
no problem most of the time it is these annoying little problems stay in touch glad i could be of assistance :D>From: "Pablo L. Arturi" <parturi@bairesweb.com> >Reply-To: Asterisk Users Mailing List - Non-Commercial >Discussion<asterisk-users@lists.digium.com> >To: "Asterisk Users Mailing List - Non-Commercial >Discussion"<asterisk-users@lists.digium.com> >Subject: Re: [asterisk-users] newbbie question >Date: Sat, 22 Jul 2006 22:01:28 -0300 >MIME-Version: 1.0 >Received: from lists.digium.com ([69.16.138.164]) by >bay0-mc7-f13.bay0.hotmail.com with Microsoft SMTPSVC(6.0.3790.2444); Sat, >22 Jul 2006 18:19:33 -0700 >Received: from digium-69-16-138-164.phx1.puregig.net (localhost >[127.0.0.1])by lists.digium.com (Postfix) with ESMTP id C4BE22FC146;Sat, 22 >Jul 2006 18:01:30 -0700 (MST) >Received: from psmtp.com (exprod8mx39.postini.com [64.18.3.139])by >lists.digium.com (Postfix) with SMTP id 0252F2FC34Dfor ><asterisk-users@lists.digium.com>;Sat, 22 Jul 2006 18:01:18 -0700 (MST) >Received: from source ([200.59.45.4]) by >exprod8mx39.postini.com([64.18.7.10]) with SMTP; Sat, 22 Jul 2006 18:01:19 >PDT >Received: from bworg196ib52so (unknown [201.216.206.221])by dnsba.com >(Postfix) with ESMTP id DD73C46A041for ><asterisk-users@lists.digium.com>;Sat, 22 Jul 2006 22:06:26 -0300 (ART) >X-Message-Info: LsUYwwHHNt1+kBqzHf1+IeLEzExvV0V0QFuYoMTxBDY>X-Original-To: asterisk-users@lists.digium.com >Delivered-To: asterisk-users@lists.digium.com >References: <BAY113-F19C637A0461E15BB4CFC48CE640@phx.gbl> >X-MSMail-Priority: Normal >X-Mailer: Microsoft Outlook Express 6.00.2800.1807 >X-MIMEOLE: Produced By Microsoft MimeOLE V6.00.2800.1807 >X-pstn-levels: (S: 2.18573/99.89068 FC:95.5390 LC:95.5390 R:95.9108 >P:95.9108M:96.8350 C:98.4741 ) >X-pstn-settings: 3 (1.0000:1.0000) s fc lc gt3 gt2 gt1 r p m c >X-pstn-addresses: from <parturi@bairesweb.com> [db-null] X-BeenThere: >asterisk-users@lists.digium.com >X-Mailman-Version: 2.1.5 >Precedence: list >List-Id: Asterisk Users Mailing List - Non-Commercial >Discussion<asterisk-users.lists.digium.com> >List-Unsubscribe: ><http://lists.digium.com/mailman/listinfo/asterisk-users>,<mailto:asterisk-users-request@lists.digium.com?subject=unsubscribe> >List-Archive: <http://lists.digium.com/pipermail/asterisk-users> >List-Post: <mailto:asterisk-users@lists.digium.com> >List-Help: <mailto:asterisk-users-request@lists.digium.com?subject=help> >List-Subscribe: ><http://lists.digium.com/mailman/listinfo/asterisk-users>,<mailto:asterisk-users-request@lists.digium.com?subject=subscribe> >Errors-To: asterisk-users-bounces@lists.digium.com >Return-Path: asterisk-users-bounces@lists.digium.com >X-OriginalArrivalTime: 23 Jul 2006 01:19:34.0856 (UTC) >FILETIME=[0ECDB480:01C6ADF6] > > > exten => _NXXXXXXXXXXX,1,Dial(SIP/+{EXTEN}@voipstunt) > > should be > > exten => _NXXXXXXXXXXX,1,Dial(SIP/+${EXTEN}@voipstunt) > > notice the $ sign in front of {EXTEN} that declares it a variable(haha >like > > my oxymoron?) > > tells asteirsk to lookup the extensions dialed by the user > >Brandon, that whas exactly the problem. > >Thank you very much!!! now I am calling :) > > >_______________________________________________ >--Bandwidth and Colocation provided by Easynews.com -- > >asterisk-users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users_________________________________________________________________ Don’t just search. Find. Check out the new MSN Search! http://search.msn.click-url.com/go/onm00200636ave/direct/01/