One thing that jumps out at me is your Dial line:
exten => _9X.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
I believe this should be:
exten => _9X.,1,Dial(SIP/voicepulse-out/${EXTEN:1},30,r)
What I think is happening is the ${EXTEN:1}@ is being treated as the
username when contacting voicepulse, which is not what you want.
In addition to the verbosity options you are using, you can
get some really detailed logs by turning "sip debug" on from
the * console. It may give you a bit more information.
Randy
On Sat, Jan 15, 2005 at 03:18:01AM -0500, Chris Wallace wrote:
> I have researched my issue a little more and this is what I have come up
> with. Here a examples of my configurations so far and the error I get when
> I try to dial an external number. It seems like I am so close, thanks for
> the help so far!
>
> Chris
>
> ############################################################################
> ############################################################################
> ftmy-voip-01*CLI>
> -- Executing Dial("SIP/100-9c8f", "SIP/[EMAIL PROTECTED]|30|r") in
> new stack
> -- Called [EMAIL PROTECTED]
> -- SIP/voicepulse-out-a68a is making progress passing it to SIP/100-9c8f
> Jan 15 02:08:13 WARNING[17333]: chan_sip.c:6811 handle_response: Forbidden -
> wrong password on authentication for INVITE to '"Chris Wallace"
> <sip:[EMAIL PROTECTED]>;tag=as772f7e09'
> -- SIP/voicepulse-out-a68a is circuit-busy
> == Everyone is busy/congested at this time
> Jan 15 02:08:19 WARNING[17333]: chan_sip.c:694 retrans_pkt: Maximum retries
> exceeded on call [EMAIL PROTECTED] for seqno 103
> (Non-critical Request)
> Jan 15 02:08:23 WARNING[17333]: pbx.c:1934 ast_pbx_run: Timeout, but no rule
> 't' in context 'local'
> ftmy-voip-01*CLI>
> ############################################################################
> ############################################################################
>
> ############################################################################
> ############################################################################
> ;
> ; SIP Configuration for Asterisk
> ;
> [general]
> port=5060
> bindaddr=0.0.0.0
> context=default
> externip=69.138.121.16
>
> register => s00******:[EMAIL PROTECTED]
>
> [voicepulse-out]
> type=peer
> context=voicepulse-out
> username=s00******
> authuser=s00******
> secret=********
> host=access1.voicepulse.com
> nat=yes
>
> [voicepulse-in]
> type=friend
> context=vp-incoming
> username=s00******
> secret=********
> host=access1.voicepulse.com
> nat=yes
>
> [100]
> type=friend
> context=local
> username=100
> secret=1234
> callerid="Chris Wallace" <239-935-0299>
> host=dynamic
> nat=yes
> canreinvite=no
> ############################################################################
> ############################################################################
>
> ############################################################################
> ############################################################################
> ;
> ; Extension Configuration for Asterisk
> ;
> [general]
> static=yes
> writeprotect=no
>
> [globals]
>
> [vp-incoming]
> exten => 2399350299,1,Answer
> exten => 2399350299,2,Wait,1
> exten => 2399350299,3,Playback(vm-goodbye)
> exten => 2399350299,4,Hangup
>
> [local]
> exten => _9X.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
> include=internal
>
> [internal]
> exten => 100,1,Dial(SIP/100,20)
> exten => 100,2,Voicemail(u100)
> exten => 100,102,Voicemail(b100)
> ############################################################################
> ############################################################################
>
> -----Original Message-----
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Randy
> Sent: Friday, January 14, 2005 11:30 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Asterisk and Voice Pulse Open Access
>
> Chris,
>
> I do not have VoicePulse Open Access, but I do have an incoming number
> through
> VoicePulse Connect. You might want to give the following a try unless you
> get
> a repsonse back from someone who uses Open Access specifically. (I found
> the
> access1.voicepulse.com address from another posting.)
>
> Edit sip.conf and extensions.conf as follows, editing the 2165551212 to
> match your assigned phone number from VoicePulse, as well as filling in your
> userid and password.
>
> To have the extension go to another context than default, you must specify
> it
> as the context in the general section in sip.conf - I was unable to get the
> normal peer matching to work for voicepulse, at the moment I suspect its due
> to inconsistent rev mappings for their ip's. If you do not have an
> extension
> that matches your number, it will defer to 's'.
>
> sip.conf
>
> ; in your [general] section add:
> register => userid:[EMAIL PROTECTED]
>
> extensions.conf
>
> ; add an extension matching your phone number to your default context (or
> the
> ; context specified in sip.conf)
> exten => 2165551212,1,Answer
> exten => 2165551212,2,Wait,1
> exten => 2165551212,3,Playback(vm-goodbye)
> exten => 2165551212,4,Hangup
>
> Hope this works for you - it does for me with VoicePulse Connect.
>
> Randy
>
> On Fri, Jan 14, 2005 at 10:19:17PM -0500, Chris Wallace wrote:
> >
> > Has any messed with getting Asterisk to work using the Voice Pulse
> > Open Access plan? I currently have 2 numbers with Voice Pulse, 1 is a
> > number that is assigned to their hardware device (Sipura SPA-2000),
> > the other is a Open Access number that uses SIP from any device (you
> > must authenticate with them). I want to be able to use the Open
> > Access number on my Asterisk server here at home with no FXO cards. I
> > have having a hard time getting this to work; I have only been using
> > Asterisk for about a week now. I have managed to get Asterisk working
> > with a plain phone line going into an XP100. This list is an awesome
> > tool, any help would be appreciated!!!
> >
> >
> > Chris
>
> > _______________________________________________
> > Asterisk-Users mailing list
> > [email protected]
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