[asterisk-users] VoicePulse Connect

[Wes Baehr]

I had a lot of problems with garbled IAX calls (even when calling into just
the IVR). The problem was compacted when I would bridge an incoming IAX call
to an outgoing SIP call, though that may be a fault of Asterisk. Since using
SIP everything has been working perfectly. I never had any real problems
with dropping calls (that weren't on my end). However, I don't use IAX
anymore, so I am not aware of any current issues.

 

 

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Christopher
Robinson
Sent: Wednesday, August 08, 2007 4:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] VoicePulse Connect

 

Wes, I'm working through some issues with IAX and Voicepulse right now.  It
was regarding dropped inbound calls.  I was able to put my server into a
different location though, and so far the issues have disappeared so
hopefully it was a network problem somewhere between us.    Just curious
what problems you encountered as I would prefer to use IAX if possible.

John, I've tried a few services, and Voicepulse was the clear winner for me.
I still have two other services in my dialplan for failover, but Voicepulse
will remain the primary for now.  The voice quality has been very good, and
their technical support has been absolutely fantastic for a no-charge
service.

Wes Baehr wrote: 

If you cannot afford any dropped calls or poor audio quality, you need a PRI
or POTS connection. It doesn't matter how great the carrier is, the Internet
is an unreliable medium.
 
2-3 times VoicePulse has had issues with incomings calls ringing busy. Once
incoming calls were all garbled on my end, although the customer could hear
me fine.
 
Generally, the outbound service is reliable. However, you should have a
backup carrier anyway.
 
 
 
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of John Meksavan
Sent: Wednesday, August 08, 2007 1:09 PM
To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] VoicePulse Connect
 
Wes,
 
  What kind of service outages did you experienced?
 
  This would use for my office and I cannot afford for any dropped calls or
poor audio quality, when talking to customers.
 
-John
 
  

From: "Wes Baehr"  <mailto:wbaehr at abilitybusiness.com>
<wbaehr at abilitybusiness.com>
Reply-To: Asterisk Users Mailing List - Non-Commercial 
Discussion <mailto:asterisk-users at lists.digium.com>
<asterisk-users at lists.digium.com>
To: "'Asterisk Users Mailing List - Non-Commercial 
Discussion'" <mailto:asterisk-users at lists.digium.com>
<asterisk-users at lists.digium.com>
Subject: Re: [asterisk-users] VoicePulse Connect
Date: Wed, 8 Aug 2007 12:55:29 -0400
 
John,
 
Voicepulse Connect has been great to me. I've been using it for over a 
year now, and do not have any major complaints, except that there are 
no printable receipts for credit card transactions. SIP is also the 
preferable protocol, as IAX seems to have some issues. Customer service 
is usually pretty good, and there have been very few (although a 
couple) problems with service outages.
 
 
 
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of John 
Meksavan
Sent: Wednesday, August 08, 2007 12:30 PM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] VoicePulse Connect
 
Asterisk Users,
 
  Has anybody use Voicepulse Connect for Asterisk?
 
  I am trying to cover all my bases because in the past, I got burned 
with poor quality of service, along with failed DTMF tones with 3 
different SIP Providers (Vitelity, Broadvoice, and Teliax).
 
  I am running Asterisk 1.2.13 on the Debian Etch system, using the 
SIP protocol.  Any insights would be great.  Thanks.
 
 
-John
 
_________________________________________________________________
Tease your brain--play Clink! Win cool prizes!
http://club.live.com/clink.aspx?icid=clink_hotmailtextlink2
 
 
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
 
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
 
 
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
 
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
    

 
_________________________________________________________________
Find a local pizza place, movie theater, and more.then map the best route! 
http://maps.live.com/default.aspx?v=2
<http://maps.live.com/default.aspx?v=2&ss=yp.bars~yp.pizza~yp.movie%20theate
r> &ss=yp.bars~yp.pizza~yp.movie%20theater
&cp=42.358996~-71.056691&style=r&lvl=13&tilt=-90&dir=0&alt=-1000&scene=95060
7&encType=1&FORM=MGAC01
 
 
 
 
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
 
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
 
 
  

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070808/45bf1a44/attachment.htm