asterisk only supports IAX2, SIP and TEL, it will only use IAX2 and SIP
entries however
so it is used to route via the Net if it cannot find a route via the
Net or the link isn't working it will go to the next priority in your
dialplan and do whatever you want, it doesn't re-configure
calls.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Chris A.
Icide
Sent: 22 May 2004 13:26
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] VoicePulse SIP
Lars,
I could be quite wrong, but I think you only need a 'timing'
source if you
want
- just normal calls.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Chris A.
Icide
Sent: 22 May 2004 13:26
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] VoicePulse SIP
Lars,
I could be quite wrong, but I think you only need a 'timing
: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Chris A.
Icide
Sent: 22 May 2004 13:26
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] VoicePulse SIP
Lars,
I could be quite wrong, but I think you only need a 'timing'
source if you
want to use trunking over IAX. You can still
I'm using Coloco now, which so far is working well.
Where companies like VoicePulse buy services from a patchwork of CLECs
in order to cover their markets, Coloco is a CLEC. The upside is that
you cut out the middleman. But if you need a number in an area they
don't serve you'll need to find a
Brian Cuthie wrote:
I'm using Coloco now, which so far is working well.
Where companies like VoicePulse buy services from a patchwork of CLECs
in order to cover their markets, Coloco is a CLEC. The upside is that
you cut out the middleman. But if you need a number in an area they
don't
Welcome to Voicepulse and their lack of jitter buffer. This is the
cause of your horrible sound. Will be just as bad with SIP.
Which providers give you a jitter buffer?
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
Welcome to Voicepulse and their lack of jitter buffer. This is the
cause of your horrible sound. Will be just as bad with SIP.
Which providers give you a jitter buffer?
In Europe: VoipTalk and Magrathea. In the US: Iconnecthere. I am sure
there are more.
Lars Boegild Thomsen wrote:
H - can anybody confirm this. I have generally had little luck with IAX
in any case so I must admit I assumed (due to info from www.voip-info.org)
that it was due to lack of timing device. I have actually not tried to do
any trunking - just normal calls.
That
Andres wrote:
[EMAIL PROTECTED] wrote:
Which providers give you a jitter buffer?
In Europe: VoipTalk and Magrathea. In the US: Iconnecthere. I am sure
there are more.
Clearpath gives jitter buffer as well. http://www.clearpath1.com/
John
___
Dear Sirs,
Anybody ever tried running SIP up against Voicepulse? On their
http://connect.voicepulse.com they claim they support both SIP and IAX, but
I can't seem to get SIP running. I have as mentioned before on this list -
huge problems getting any timing devices running on some of my
SIP used to work fine with VoicePulse. But the funny thing is I could
never detect any signs that they were doing call accounting. I could
make IAX calls and see them show up in the CDR and the $$ deducted from
my account balance. But when I made SIP calls they appeared, by all
measures, to be
: Re: [Asterisk-Users] VoicePulse SIP
SIP used to work fine with VoicePulse. But the funny thing is I could
never detect any signs that they were doing call accounting. I could
make IAX calls and see them show up in the CDR and the $$ deducted from
my account balance. But when I made SIP
Who do you use now?
x-tad-smallerDavid Hickman
TSG Computer Consulting - Auctions
314-865-4752 x2
/x-tad-smaller
On May 21, 2004, at 8:49 PM, Brian Cuthie wrote:
SIP used to work fine with VoicePulse. But the funny thing is I could never detect any signs that they were doing call accounting. I
Lars Boegild Thomsen wrote:
Dear Sirs,
Anybody ever tried running SIP up against Voicepulse? On their
http://connect.voicepulse.com they claim they support both SIP and IAX, but
I can't seem to get SIP running. I have as mentioned before on this list -
huge problems getting any timing devices
Lars,
I could be quite wrong, but I think you only need a 'timing' source if you
want to use trunking over IAX. You can still use IAX without trunking if
you don't have any sort of timing device.
-Chris
On 06:39 PM 5/21/2004, Lars Boegild Thomsen wrote:
Dear Sirs,
Anybody ever tried running
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