Displaying 20 results from an estimated 11000 matches similar to: "Re: failover (was Re: voicepulse)"
2004 Jan 14
2
Re: failover (was Re: voicepulse)
> But this is not to say _you_ can't built a reliable VOIP based
> system. Get _two_ providers and set up your dial plan in
> extensions.conf to "fail over" if one service fails to
> connect to dial via the next one and finally if both fail
> use pstn. your users will see a system the "just works".
Now there's an idea.
I'm playing with this now,
2004 May 31
1
Failover: iconnecthere to voicepulse
Hi all,
I'm working on a setup for a small office. I'd like to use SIP/iconnecthere
most of the time, because they're cheap. But they only allow a single call.
When the single iconnecthere line is in use, I'd like to use IAX2/voicepulse
instead:
exten => _1NXXNXXXXXX,1,Dial,SIP/BYEXTENSION@iconnect
exten => _1NXXNXXXXXX,2,Dial,IAX2/npI42VoD38@voicepulse/${EXTEN}
Well,
2003 Dec 08
2
Problems with voicepulse.com
Greetings,
I have been experimenting with Asterisk for a few weeks and finally
decided to
take the plunge and purchase a few DIDs for inbound calling. Our
attempts at
IAX/IAX2 connectivity with VoicePulse have been less than successful.
We get
"Registration Refused" errors from Asterisk whenever we launch the
server. The
front-line support folks at VoicePulse suggested that we are
2004 Apr 08
0
IAX2 Trunk to PSTN (voicepulse) questions...
All,
I've almost got my Asterisk PBX setup, but I've having some problems with
the VoicePulse IAX trunk.
On outbound calls, when dialing a PSTN number through the IAX2 trunk,
music on hold (moh, using the m option in the dial command) does not work.
The console states that "stop sound" on IAX2 channel. Ring works, but
only without the r option. MOH works when trying to dial a
2007 Aug 22
0
Users Conference - Friday@12:30 PM EDT: Founders of Voicepulse
For this week's conference, the two founders of Voicepulse, Ravi
Sakaria and Ketan Patel, will be joining us. For those of you who
are not aware, Voicepulse is an asterisk friendly VOIP provider that
has won awards for service and innovation.
We will also have Trixbox news, updates, as well as discount codes.
Lastly, we are working feverishly to bring you more information
regarding legal
2004 Sep 13
0
voicepulse problems since new configs
Voicepulse has ignored four emails over the course of two weeks.
Anyone have any ideas of whats wrong?
- Executing Dial("IAX2/voicepulse-in-01@66.234.228.170:4569/7", "IAX2/acctname:acctpass@gwiaxt01.voicepulse.com/14109649073") in new stack
-- Called acctname:acctpass@gwiaxt01.voicepulse.com/14109649073
Sep 13 22:48:25 WARNING[131080]: chan_iax2.c:5375
2004 Dec 17
0
Newbie setup question (Voicepulse, FWD & IAXTEL)
Okay, I can receive calls through Voicepulse fine. All the various
attempts (too many to list) to create a workable configuration to Dial
to Voicepulse has failed, from 403s to "No authority found" to nothing.
The Voicepulse folks told me that the open access was SIP and I shouldn't
have a reference in the iax.conf file, but then said that they were
refering my question to the
2004 Aug 30
2
VoicePulse Connect DTMF with IAX2
Is there anyone out there who has VoicePulse Connect working with DTMF?
I've been unable to get it to work from the start, and the recent
VoicePulse updates
did not help.
A caller to my DID's hears Asterisk, but pressing DTMF does nothing:
On call setup "iax2 debug" shows:
-----------------
Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass:
ACK
2005 May 28
0
newbie asterisk SIP config question (using VoicePulse Connect)
Greetings,
I am new to all this VoIP stuff and have been having a bit of a hard
time getting my soft phone working as a SIP client thru Asterisk. I
apologize to start off with such a simple question and hope it's ok to
post this and see what others have done.
THE GOOD NEWS:
I have successfully setup Asterisk 1.07 on an OSX machine. The build
is running and working successfully. I am able
2005 May 28
0
newbie asterisk SIP config question (using VoicePulse Connect!)
Greetings,
I am new to all this VoIP stuff and have been having a bit of a hard
time getting my soft phone working as a SIP client thru Asterisk. I
apologize to start off with such a simple question and hope it's ok to
post this and see what others have done.
THE GOOD NEWS:
I have successfully setup Asterisk 1.07 on an OSX machine. The build
is running and working successfully. I am able
2004 Jan 29
1
re: help with voicepulse connect IAX2
hello,
after playing with an asterisk configuration for voip for a few weeks i'm
trying to get outbound dialing with voicepulse going - i've cut down the
asterisk to a very minimal install (1 SIP client) to try to localize the
problem. The SIP client works fine (SIP and * on the same NAT) and could
access the demo from samples before i removed it, and can call itself - so
i am
2005 Mar 21
5
VoicePulse Issues
I recently switched from BroadVoice to VoicePulse Connect on my Asterisk
box. Too many Meetme quality complaints (whether real or perceived).
I had to make a choice to use IAX2 or SIP with VoicePulse. I first
tried to go with SIP because I already had it working and all of our
devices are SIP. Problem is that every time I turn my back, the
Asterisk registration with the VoicePulse SIP server
2003 Oct 14
1
Iaxtel and Voicepulse
I'm having trouble configuring these services the way I want. Basically I
prefer using iLBC before GSM, however Iaxtel only want to talk GSM. It
_seems_ that Voicepulse prefers GSM also, because even if I put ILBC before
GSM in the "allow=" part of iax.conf, if GSM is mentioned, Voicepulse will
use it. If I don't allow GSM Voicepulse will use ILBC.
Does anyone know how to
2005 Jan 05
2
Allowing "pooling" or "rollover" for inbound calls on VoicePulse
My goal is to have only 1 primary phone number that can seamlessly
"pool" multiple VoicePulse accounts. Let's say I have 3 accounts with
VoicePulse Connect
212-555-1000 (primary)
212-555-1001
212-555-1002
When I receive inbound calls on 212-555-1000, I want to "forward" or
"roll over" the connection to 212-555-1001 and 212-555-1002 so that the
212-555-1000
2005 Jul 16
1
Voicepulse connect - unable to dial out, asterisk says "9696"
Hi,
for some weeks now I have been unable to make calls via my voicepulse
connect IAX account?
When I attempt the console looks like this:-
rt*CLI>
-- Executing Dial("SIP/2008-cf55",
"IAX2/NBhXXXXXX:XXXXXXN82@gwiaxt01.voicepulse.com/12124565900") in new
stack
-- Called NBhXXXXX:XXXXN82@gwiaxt01.voicepulse.com/12124565900
-- Call accepted by 66.234.228.160
2004 Feb 02
0
VoicePulse IAX2 lag
Yes, and they are aware of the problem.
-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Jonathan Tew
Sent: Monday, February 02, 2004 1:59 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] VoicePulse IAX2 lag
Is anyone else noticing high lag on their voicepulse IAX2 connections?
We're seeing
2004 Jun 14
4
Number Portability and VoicePulse
I have two questions regarding number portability...
1) If I port a DID over to Voicepulse, can I then move that DID elsewhere
somewhere down the road. Or does voicepulse now OWN that DID?
2) Can I take a DID assigned by Voicepulse and transfer it to someone else?
If not, why?
-jwb
2005 Feb 16
2
Anyone having trouble with VoicePulse Connect?
I've been using my voicepulse connect number for over
a month now, but today it simply won't connect. My
partner and I each have a number, both are mapped in
my iax.conf and extensions.conf files. This has been
working fine.
Today, either number gives this message:
Feb 16 21:53:14 NOTICE[4330]: chan_iax2.c:5757
socket_read: Rejected connect attempt from
66.234.228.170, request
2005 May 31
2
ISO Suggestions for Multiple Inbound Voicepulse Lines
I'm looking to set up multiple inbound Voicepulse Connect lines and have Asterisk route them direct to different IVR or Voicemail based on the inbound number that is called. Unfortunately, I just can't see how one would go about identifying the number that is being called. Has anyone been able to do something like this with Voicepulse?
I appreciate any assistance.
Phil
2005 Feb 17
1
Voicepulse Open Access & Asterisk Problems
I can't seem to dial out with Voicepulse Open Access service using *.
Incoming works fine. Another user posted a few weeks back that they
were having problems and there are some threads at dslreports.com
about this as well. Maybe someone here can figure out what the issue
is from the sip debug info below. I am at a loss.
The audible error message from Allison is 0984 (from VP server)
Here is