[Asterisk-Users] Asterisk and Voice Pulse Open Access

[Brian Dingman]

Chris,
Did you ever get this working?


On Sat, 15 Jan 2005 03:18:01 -0500, Chris Wallace
<list-subscriber at jesustech.net> 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/3330000 at voicepulse-out|30|r") in
> new stack
>     -- Called 3330000 at voicepulse-out
>     -- 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:2399350299 at 192.168.0.20>;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 68bae442390ef4bd7310e0f262c3e675 at 192.168.0.20 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******:********@access1.voicepulse.com
> 
> [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:1}@voicepulse-out,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: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] 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:password at access1.voicepulse.com
> 
> 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