[Asterisk-Users] Asterisk and Voice Pulse Open Access

Sat Jan 15 01:18:01 MST 2005

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

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