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
>
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> 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/