Has anyone been able to get * working with VoicePulse OpenAccess (SIP not IAX). I have found a ton of information about VoicePulse Connect but very little on the proper * settings for OpenAccess. Tried contacting VP with no response. If anyone has this working, can they share their extensions.conf and sip.conf files? Better yet, if it could be posted on the Wiki. Keith -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041219/d5fd9035/attachment.htm
They have an entire knowledge base with example scripts, etc on there web site.
You can also call them and reach someone in tech support during the day.
----- Original Message -----
From: Keith O'Brien
To: asterisk-users@lists.digium.com
Sent: Sunday, December 19, 2004 11:09 AM
Subject: [Asterisk-Users] VoicePulse OpenAccess
Has anyone been able to get * working with VoicePulse OpenAccess (SIP not
IAX). I have found a ton of information about VoicePulse Connect but very
little on the proper * settings for OpenAccess. Tried contacting VP with no
response. If anyone has this working, can they share their extensions.conf and
sip.conf files? Better yet, if it could be posted on the Wiki.
Keith
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Thanks but I already tried their knowledgebase.they don't have any configs for OpenAccess only VoicePulse Connect. Also, tried calling their support and they were unable to assist in getting * working with OpenAccess only VoicePulse Connect. Does anyone have * working using SIP and VoicePulse Open Access? If so could you share your configs? From: Matthew Marlowe <matthew.marlowe <at> gmail.com> Subject: Re: VoicePulse OpenAccess <http://news.gmane.org/find-root.php?message_id=%3c007a01c4e69b%24277847d0%2 46a1e140a%40EMARLOWE%3e> Newsgroups: gmane.comp.telephony.pbx.asterisk.user <http://news.gmane.org/gmane.comp.telephony.pbx.asterisk.user> Date: Mon, 20 Dec 2004 08:49:39 -0500 They have an entire knowledge base with example scripts, etc on there web site. You can also call them and reach someone in tech support during the day -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041221/e0d83a40/attachment.htm
I did finally get through to them. However, after troubleshooting for about
? hour they were at a loss as to why it wasn?t working. I sent them my
sip.conf file and extensions.conf file and they didn?t notice any problems.
What you are saying doesn?t make sense. VoicePulse Open Access uses SIP,
VoicePulse Connect Uses IAX. How can the setup be the same if they are
using entirely different protocols? I have working IAX configs
I am looking
for a working SIP config with VoicePulse.
Their tech asked around and none of their engineers had an example sip.conf
that works with their service. They also don?t have a sip.conf file in
their knowledgebase. If anyone has this working, can they share their
sip.conf file??
Message: 14
Date: Tue, 21 Dec 2004 16:23:00 -0500
From: "Matthew Marlowe" <matthew.marlowe@gmail.com>
Subject: Re: [Asterisk-Users] Re: VoicePulse OpenAccess
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@lists.digium.com>
Message-ID: <005701c4e7a3$4038b440$6a1e140a@EMARLOWE>
Content-Type: text/plain; charset="iso-8859-1"
OpenAccess, according to the tech support rep I just got off the phone with
who answered after only one ring said the setup would be the exact same as
VoicePulse Connect.
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