Is iaxtel still around?
I was not able to go to www.iaxtel.com .
did the address changed?
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Is IAXTEL still around? I needed to call Digium and figured I would set it
up to save some miinutes when talking to them but I can't get it to
register.
-Kerry
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To
: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kerry Garrison
Sent: Tuesday, January 03, 2006 5:55 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] IAXTEL??
Is IAXTEL still around? I needed to call Digium and figured I would set it
up to save some
-Users] IAXTEL??
Is IAXTEL still around? I needed to call Digium and figured I would set it
up to save some miinutes when talking to them but I can't get it to
register.
-Kerry
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Asterisk
]
[mailto:[EMAIL PROTECTED] On Behalf Of
Bogdan Moldovan
Sent: Tuesday, January 03, 2006 8:01 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] IAXTEL??
From:
http://www.iaxtel.com/
The IAXTel Server is currently under maintenance. Some
technical
I know, this is the sad part :(
b
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tom Vile
Sent: Tuesday, January 03, 2006 6:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] IAXTEL??
That message has been
Iaxtel has been down for some time now.
But to get in contact with digium via your asterisk box all you need is to
set this dialing rule up.
exten = 500,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]) ;Call Digium
exten = 500,2,Congestion
Kerry Garrison wrote:
Is IAXTEL still around? I
Is IAXTEL still around? I needed to call Digium and figured I would set it
up to save some miinutes when talking to them but I can't get it to
register.
That hasn't worked for many many months.
Much easier to reach digium by using the Demo that is/was installed in
all asterisk installs. When
Ariel Batista wrote:
Iaxtel has been down for some time now.
But to get in contact with digium via your asterisk box all you need is
to set this dialing rule up.
exten = 500,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]) ;Call Digium
exten = 500,2,Congestion
Cool, I didn't think of that.
Rich Adamson wrote:
Any chance that we could get someone to implement the milliwatt
generator and echo test number. Would be kind of handy for testing
various items (eg, jitterbuffer).
It's running CVS HEAD (which means it has the new jb since we didn't
disable it, but then again it's all
Hello Everyone!
Over this weekend, we have updated IAXtel. Before the update, it was
running at almost 100% cpu load at an idle state because of the massive
amount of database transactions.
We enabled realtime caching and the box immediately crashed. We were
able to expose a serious bug
Over this weekend, we have updated IAXtel. Before the update, it was
running at almost 100% cpu load at an idle state because of the massive
amount of database transactions.
We enabled realtime caching and the box immediately crashed. We were
able to expose a serious bug related to
Figures... So... Everybody went to FWD :) ?
It mostly works, does IAX, so, yeah.
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Is iaxtel down?
Ive been getting this:
May 21 19:23:42 NOTICE[29984]: chan_iax2.c:2782 auto_congest:
Auto-congesting call due to slow response
-- IAX2/Iaxtel-12 is circuit-busy
-- Hungup 'IAX2/Iaxtel-12'
is it down or am I doing something wrong?
Its been doing that for months.
Figures... So... Everybody went to FWD :) ?
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Rich Adamson
|Sent: Domingo, 22 de Mayo de 2005 08:23 p.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users
Is iaxtel down?
Ive been getting this:
May 21 19:23:42 NOTICE[29984]: chan_iax2.c:2782 auto_congest:
Auto-congesting call due to slow response
-- IAX2/Iaxtel-12 is circuit-busy
-- Hungup 'IAX2/Iaxtel-12'
is it down or am I doing something wrong?
Is iaxtel down? Im trying to dial Echo test: 1700613 and I get a busy
signal...
Also, is the gw to FWD users down too?
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To
Duane wrote:
Marco Supino wrote:
Hi,
I tried to add the IAXTel config to my asterisk, so i can dial free
numbers inside the US from my SIP softphone (X-lite), everything seems
to be working, but the sound quality is terrible, the other side sounds
like a digitized voice, and the voice is cut, i
Hi,
I should be missing something. The password that go with my IAXTEL
registration include an @.
It seem that I can't use it because it thing that the second part of the
password is the host name.
I just don't know how to solve this one.
regards,
JYL
Hi,
I tried to add the IAXTel config to my asterisk, so i can dial free
numbers inside the US from my SIP softphone (X-lite), everything seems
to be working, but the sound quality is terrible, the other side sounds
like a digitized voice, and the voice is cut, i cant hear a full word,
I tried
Marco Supino wrote:
Hi,
I tried to add the IAXTel config to my asterisk, so i can dial free
numbers inside the US from my SIP softphone (X-lite), everything seems
to be working, but the sound quality is terrible, the other side sounds
like a digitized voice, and the voice is cut, i cant
As someone that's just recently setup an * server I agree. I thought
about setting up an Iaxtel account as well but couldn't see the point
in it because I had setup FWD for testing. I continue to use FWD for
all my toll free calls and the quality is just awesome. I can't see how
Iaxtel would
---
From: ... On Behalf Of Christopher Dobbs
Sent: Saturday, January 19, 2002 8:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] IAXTEL errors !
Use FWDNET.NET.
It is far better on call quality
Mark had made a post recently (last week or so maybe) -- could have been in
IRC too... (it starts to blur together) that he was aware of the IAXTEL
problems and that they were working on the issues.
Details are hazy... But then I drink alot too, so everything is hazy...
(that's the point)
Tom
On Fri, 21 Jan 2005 11:26:12 -0700, Steve Murphy [EMAIL PROTECTED] wrote:
I didn't get any response at all to my last request for status on
IAXTEL.
So, when this happens, I attribute it to one of a number of things:
1. No-one knows.
2. No-one cares.
3. Everyone knows, but are too busy to
On Fri, 21 Jan 2005 13:28:46 -0600, Leif Madsen wrote:
On Fri, 21 Jan 2005 11:26:12 -0700, Steve Murphy [EMAIL PROTECTED] wrote:
I didn't get any response at all to my last request for status on
IAXTEL.
So, when this happens, I attribute it to one of a number of things:
1. No-one knows.
I am testing IAXTEL and routing 800 number to them.. Sometimes
the call goes through and the other times it get the following error.
WARNING[20502]: chan_iax2.c:1477 attempt_transmit:
Max retries exceeded to host 69.73.19.178 on IAX2/iaxtel/3 (type = 6, subclass
= 9, ts=631, seqno=1)
Use FWDNET.NET.
It is far better on call quality!!
--
Christopher Dobbs
Manjit Riat wrote:
I am testing IAXTEL and
routing 800 number to them.. Sometimes
the call goes through and the other times it get the following error.
WARNING[20502]:
chan_iax2.c:1477
: Re: [Asterisk-Users]
IAXTEL errors !
Use FWDNET.NET.
It is far better on call quality!!
--
Christopher Dobbs
Manjit Riat wrote:
I am testing IAXTEL and routing 800 number to them..
Sometimes the call goes through and the other times it get the following error.
WARNING[20502]: chan_iax2.c
What is the best codex for iaxtel?
When I set in iax.conf
bandwidth=high
disallow=all
allow=ulaw
The call will not go through, if I set allow=all
it sets the format to ADPCM and the first 15sec. or so the voice is
choppy, it is hard to understand anything.
Is it reliable/practical to terminate
Joseph wrote:
What is the best codex for iaxtel?
When I set in iax.conf
bandwidth=high
disallow=all
allow=ulaw
The call will not go through, if I set allow=all
it sets the format to ADPCM and the first 15sec. or so the voice is
choppy, it is hard to understand anything.
Is it reliable/practical to
On Mon, 2005-01-17 at 12:20 -0600, Eric Wieling aka ManxPower wrote:
Joseph wrote:
What is the best codex for iaxtel?
When I set in iax.conf
bandwidth=high
disallow=all
allow=ulaw
The call will not go through, if I set allow=all
it sets the format to ADPCM and the first 15sec.
Joseph wrote:
On Mon, 2005-01-17 at 12:20 -0600, Eric Wieling aka ManxPower wrote:
Joseph wrote:
What is the best codex for iaxtel?
When I set in iax.conf
bandwidth=high
disallow=all
allow=ulaw
The call will not go through, if I set allow=all
it sets the format to ADPCM and the first 15sec. or so
What is the best codex for iaxtel?
When I set in iax.conf
bandwidth=high
disallow=all
allow=ulaw
The call will not go through, if I set allow=all
it sets the format to ADPCM and the first 15sec. or so the voice is
choppy, it is hard to understand anything.
Is
I've tried gsm but the call doesn't go through.
bandwidth=high could be screwing it up.
Post the CLI output of the failed call.
Executing Dial(SIP/11-0b9e,
IAX2/joseph:[EMAIL PROTECTED]/[EMAIL PROTECTED]) in new stack
-- Called joseph:[EMAIL PROTECTED]/[EMAIL PROTECTED]
--
Iaxtel only supports gsm.
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Rich Adamson
Sent: Monday, January 17, 2005 2:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] iaxtel
When I try to call iaxtel it goes to codec ADPCM even though I have
define in iax.conf gsm
Call accepted by 69.73.19.178 (format ADPCM)
-- Format for call is ADPCM
My settings:
[general]
port=4569
register = :[EMAIL PROTECTED]
bandwidth=high
jitterbuffer=no
tos=lowdelay
[voipjet]
Why is it switching me to Codec: ADPCM?
PS. It seems to me iaxtel has a problem with connection today, can
anybody confirm it?
I just tried to place a call via iaxtel and watched the packets with
ethereal. The iaxtel server is very very slow to respond to _any_
packet, indicating its not
There was a bug with codecs for a very long time with Asterisk. In
[general] remove the bandwidth= line (all it does is allow specific
codecs) and disallow=all and allow= for eac codec you want.
Joseph wrote:
When I try to call iaxtel it goes to codec ADPCM even though I have
define in
I cannot find the directory of 1700 numbers (iaxtel), nor where I can
edit my own entry. Can anybody publish the link, please?
bye
Ronald
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I have registered to iaxtel.com!
I forgot my iaxtel.com number, and cannot find the white pages of it.
As I see, you should setup in extensions.conf all 1700*,1888*, 1877*,
1866* and 1800* for this connection.
please correct me:
1700* is only other iaxtel.com users
1888* are tollfree numbers in
1800,1866,1877,1888 are all toll free numbers in the us
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] On Behalf Of Anders F
Eriksson
Sent: Tuesday, December 21, 2004 7:26 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] IAXTEL Configuration
Hi,
I think you should remove the [iaxtel_out] from iax.conf
This is a snip from mine iax.conf:
[general
I signed up for an
IAXTEL account and have been trying, unsuccessfully, to get it working. In
IAX.CONF I have:
[iaxtel_out]type=peerhost=iaxtel.comusername=USERNAMEsecret=SECRETauth=rsainkeys=iaxtel
[iaxtel]type=friendcontext=incominghost=iaxtel.comauth=rsainkeys=iaxtel
However, when I
/setup.html (which is where I got my settings).
/Anders
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Adam Robins
Sent: den 21 december 2004 22:52
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] IAXTEL
Hello,
I'm having problems connecting to other * boxes through IAXTel. I've
seen this addressed in the list archives, and other places on the web,
but haven't seen that anyone has come up with a solution. I'm dialing
in to my Asterisk server using DISA, authenticating OK, then
attempting to dial
i had this problem last night. sometimes it would work find and then i
would get errors or timeouts???
- hcir
On Oct 22, 2004, at 9:07 AM, pixelFiend wrote:
Hello,
I'm having problems connecting to other * boxes through IAXTel. I've
seen this addressed in the list archives, and other places on
I'm trying to run some inbound test to my Asterisk box using Telesthetic's gateway in MI to my GNU/IAXtel account.
Am I missing something? I set up my user account on the GNUPhonne site, configured Asterisk to talk to IAXTel. * registers fine. In fact I can make calls to other test users. I
On Sun, 29 Aug 2004, Kris Boutilier wrote:
Is timestamp information calculated purely from the relative timestamps of
each frame of the current incoming stream or is there some degree of RTC
synchronization expected between the two endpoints?
No sync is needed; its all relative.
On Sat, 28 Aug 2004, Michael George wrote:
So even with X11 eliminated the sound is still bad to Digium. I tried
another's 1700 number, and it sounded the same, so it's not something unique
to digium and me.
Would IAX/GSM be so sensitive to half-duplex that I cannot expect it to work
On Sat, 28 Aug 2004, Andrew Kohlsmith wrote:
Please note that it seems impossible to disable jitter buffer between 20040806
CVS HEAD endpoints. The jitterbuffer numbers in iax2 show channels look
live. The numbers look right (jitbuf 0ms) between 20040806 and RC1
(Nufone). I haven't
On Sunday 29 August 2004 02:06, [EMAIL PROTECTED] wrote:
On Sat, 28 Aug 2004, Andrew Kohlsmith wrote:
Please note that it seems impossible to disable jitter buffer between
20040806 CVS HEAD endpoints. The jitterbuffer numbers in iax2 show
channels look live. The numbers look right (jitbuf
On Sunday 29 August 2004 01:59, [EMAIL PROTECTED] wrote:
If you think that the jitter buffer isn't working right and should fix
this, then please capture debug from the buffer and send over to me.
To do that, in /etc/asterisk/logger.conf edit the debug line to be:
debug =
On Sunday 29 August 2004 01:59, [EMAIL PROTECTED] wrote:
If you think that the jitter buffer isn't working right and should fix
this, then please capture debug from the buffer and send over to me.
I notice that the timing measurements are still showing wild values at
times - here is a
Those wild times especially occur before any audio is sent. (e.g. while
ringing or pre ringing).
At 17:10 29/08/2004, you wrote:
On Sunday 29 August 2004 01:59, [EMAIL PROTECTED] wrote:
If you think that the jitter buffer isn't working right and should fix
this, then please capture debug
On Sat, Aug 28, 2004 at 11:31:48PM -0400, Andrew Kohlsmith wrote:
On Saturday 28 August 2004 23:01, Michael George wrote:
It's a PII 266 (okay, not the fatest system) with 192MB RAM. X is not
running and the Framebuffer has been turned off in /boot/grum/menu.lst. I
have disabled all the
At 17:10 29/08/2004, you wrote:
I notice that the timing measurements are still showing wild values at
times - here is a partial grab of an iax2 show channels:
Lag Jitter JitBuf Format
00020ms 6291456ms ms ALAW
00012ms 6291440ms ms ALAW
00017ms 0004ms ms ALAW
On Sun, Aug 29, 2004 at 07:59:20AM +0200, [EMAIL PROTECTED] wrote:
On Sat, 28 Aug 2004, Michael George wrote:
So even with X11 eliminated the sound is still bad to Digium. I tried
another's 1700 number, and it sounded the same, so it's not something unique
to digium and me.
Would
On Sun, 29 Aug 2004, Andrew Kohlsmith wrote:
Also, is are logs of problem conversations already in progress any use to you?
You nailed down the dead audio after 65535ms problem but every now and
again (very very rare) we will have a conversation where the incoming audio
goes totally
On Sun, 29 Aug 2004, joachim wrote:
Those wild times especially occur before any audio is sent. (e.g. while
ringing or pre ringing).
Yeah - because the sender does weird things to the timestamps it
generates. This is the problem that needs to be resolved; the jitter
buffer just shows
On Sunday 29 August 2004 15:52, [EMAIL PROTECTED] wrote:
The jitter buffer makes all its decisions about dejittering based on the
timestamps of incoming frames. There a fundamental expectation that the
sending side is correctly stamping each frame - 20msec, 40msec etc etc.
Right, this makes
On Sun, 29 Aug 2004, Andrew Kohlsmith wrote:
Hmm... I think next CVS update I'm gonna add a bit of code in chan_iax2 that
tries to verify that timestamps aren't getting sent incorrectly. Fun fun
fun. :-)
Its not that the generation is broken. Its that various optimisations and
things
On Sat, Aug 28, 2004 at 11:31:48PM -0400, Andrew Kohlsmith wrote:
On Saturday 28 August 2004 23:01, Michael George wrote:
It has nothing to do with IAX or GSM. Stop blaming them. My upstream is half
duplex as well (pretty much anyone on DSL or cable is on a half duplex
connection whether
: [Asterisk-Users] iaxtel and jitterbuffer
{clip}
The jitter buffer makes all its decisions about dejittering based on the
timestamps of incoming frames. There a fundamental expectation that the
sending side is correctly stamping each frame - 20msec, 40msec etc etc.
The problem is that the sending
Coast Regional District
-Original Message-
From: Michael George [mailto:[EMAIL PROTECTED]
Sent: August 27, 2004 11:58 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] iaxtel and jitterbuffer
I am trying to work out IAX -- IAX communications with my * box. I have a
registration
Had this problem earlier this week - ensure 'trunk=no' in iax.conf if you're
using fairly current CVS code. There is something not right w/the trunking
that causes choppy sound. See the wiki for more info.
I am using current CVS code and I have trunk=no. Still sounds crappy. I need
to
On Aug 28, 2004, at 7:39 AM, Rich Adamson wrote:
I do a lot of work with companies throughout the US on network
performance
and we _frequently_ run into routers, switches, servers, etc, that are
allowed to auto-negotiate their half vs full duplex nic interfaces.
About
50% of the time, systems
On Sat, Aug 28, 2004 at 08:39:50AM -0600, Rich Adamson wrote:
I do a lot of work with companies throughout the US on network performance
and we _frequently_ run into routers, switches, servers, etc, that are
allowed to auto-negotiate their half vs full duplex nic interfaces. About
50% of
On Sat, Aug 28, 2004 at 03:00:26PM -0400, Michael George wrote:
On Sat, Aug 28, 2004 at 08:39:50AM -0600, Rich Adamson wrote:
I do a lot of work with companies throughout the US on network performance
and we _frequently_ run into routers, switches, servers, etc, that are
allowed to
On Sat, 28 Aug 2004 15:24:01 -0400, Michael George wrote:
On Sat, Aug 28, 2004 at 03:00:26PM -0400, Michael George wrote:
On Sat, Aug 28, 2004 at 08:39:50AM -0600, Rich Adamson wrote:
I do a lot of work with companies throughout the US on network performance
and we _frequently_ run into
On Saturday 28 August 2004 15:00, Michael George wrote:
The difference between that and what I'm getting from IAX/GSM is profound,
with GSM being intolerably poor quality.
That's odd; every single voice call coming in and out of the company I work
for is using the GSM codec with asterisk and
On Saturday 28 August 2004 15:24, Michael George wrote:
I just saw a page on the wiki that mentions that running X11 or a VESA
frame buffer can cause jittery sound. I only have this problem with IAX2,
but that might be cause when I use Zap -- Zap or Zap -- SIP there is no
en/decoding
On Sat, Aug 28, 2004 at 05:08:30PM -0400, Andrew Kohlsmith wrote:
On Saturday 28 August 2004 15:24, Michael George wrote:
I just saw a page on the wiki that mentions that running X11 or a VESA
frame buffer can cause jittery sound. I only have this problem with IAX2,
but that might be cause
On Saturday 28 August 2004 23:01, Michael George wrote:
It's a PII 266 (okay, not the fatest system) with 192MB RAM. X is not
running and the Framebuffer has been turned off in /boot/grum/menu.lst. I
have disabled all the servers except for sshd. I have the latest source
from CVS HEAD as of
I am trying to work out IAX -- IAX communications with my * box. I have a
registration with iaxtel and I thought I would start there for my learning.
I am able to call the number for Digium's support line (700-428-6000), but the
sound is horribly chopping. Some reading revealed the jitterbuffer
-Original Message-
From: Michael George [mailto:[EMAIL PROTECTED]
Sent: August 27, 2004 11:58 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] iaxtel and jitterbuffer
I am trying to work out IAX -- IAX communications with my * box. I have a
registration with iaxtel and I thought I would start
On Fri, Aug 27, 2004 at 12:47:05PM -0700, Kris Boutilier wrote:
Had this problem earlier this week - ensure 'trunk=no' in iax.conf if you're
using fairly current CVS code. There is something not right w/the trunking
that causes choppy sound. See the wiki for more info.
I am using current CVS
hello I am trying to set up iaxtel with asterisk
and am using a sipura 1000 when my friend calls me he is sounding like he is in
a metal tank that is the best way I can describe it, how ever when he calls me
on my grand stream budjet phone 101 it sounds fine.
is there a fix for this really
I have just get an account on Iaxtel.com, and i woud like to know what can i do to
receive my Iaxtel calls in my asterisk server?
Actually i just can make IAX calls.
Thanks
--
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Powered by Outblaze
I've searched WIKI and Archives but nothing.
Im getting:
-- Called username:[EMAIL PROTECTED]/[EMAIL PROTECTED]
Jun 21 17:04:12 WARNING[1158883520]: chan_iax2.c:5097 socket_read: Call
rejected by 69.73.19.178: Unable to negotiate codec
-- Hungup 'IAX2[Iaxtel]/8'
== No one is available to
Kyle Hagan wrote:
I've searched WIKI and Archives but nothing.
Im getting:
-- Called username:[EMAIL PROTECTED]/[EMAIL PROTECTED]
Jun 21 17:04:12 WARNING[1158883520]: chan_iax2.c:5097 socket_read:
Call rejected by 69.73.19.178: Unable to negotiate codec
-- Hungup 'IAX2[Iaxtel]/8'
== No one is
Just dialed (or attempted to) a 800 number, still down
At 17:20 6/8/2004, you wrote:
Heh..yea, I made sure I did a search through the archives before posting
it :) (not that I'm complaining)
The weird thing though is that I _am_ able to call digium's iaxtel
number..
-Mark
tmpm wrote:
Just dialed (or attempted to) a 800 number, still down
you could always enable enum lookups and use either the freenum.org zone
or e164.org zone as they both contain IAX2 and SIP URLs for north
american and other countries toll free numbers...
--
Best regards,
Duane
Thanks for the tip. will look into that...
At 05:47 6/9/2004, you wrote:
tmpm wrote:
Just dialed (or attempted to) a 800 number, still down
you could always enable enum lookups and use either the freenum.org zone
or e164.org zone as they both contain IAX2 and SIP URLs for north american
and
Does anyone know if the 1-800 iaxtel gateway is down?
I've been trying to use it all day today and asterisk says it's ringing:
Channel (ContextExtensionPri ) State Appl.
Data
IAX2[iaxtel]/1 ( s1 ) Ringing AppDial
(Outgoing Line)
SIP/2201-a253 (home
Do you have r on your Dial line? If so, then Asterisk will override
whatever should you SHOULD be hearing and provide you with a ringing
sound.
On Tue, 2004-06-08 at 10:24, Mark Musone wrote:
Does anyone know if the 1-800 iaxtel gateway is down?
I've been trying to use it all day today and
Does anyone know if the 1-800 iaxtel gateway is down?
I've been trying to use it all day today and asterisk says it's ringing:
Channel (ContextExtensionPri ) State Appl.
Data
IAX2[iaxtel]/1 ( s1 ) Ringing AppDial
(Outgoing Line)
SIP/2201-a253
It seems to be down, I even tried dialing for
example 1-800-555-TELL. I tried yesterday
and again today.. Just get dead air.
Stephen Rosebush
Mark Musone wrote:
Does anyone know if the 1-800 iaxtel gateway is down?
I've been trying to use it all day today and asterisk says it's ringing:
Down here.
It seems to be down, I even tried dialing for
example 1-800-555-TELL. I tried yesterday
and again today.. Just get dead air.
Stephen Rosebush
Mark Musone wrote:
Does anyone know if the 1-800 iaxtel gateway is down?
I've been trying to use it all day today and asterisk
same with their 700 network
w
- Original Message -
From: Mark Musone [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, June 08, 2004 11:24 AM
Subject: [Asterisk-Users] iaxtel 1-800 gateway down?
Does anyone know if the 1-800 iaxtel gateway is down?
I've been trying to use it all
Ive got similar probs Mark, and no one either here (unless I havent got
thru the pile yet) or on the IRC channel last nite answered. Ive simply got
no response when I try to use Iaxtel to call anywhere. My distant end is
experienceing the exact same thing. I also tried FWD to Iaxtel, and it
Thanks for verifying that...thats what I thought...took two days to verify
it...
At 13:21 6/8/2004, you wrote:
same with their 700 network
w
- Original Message -
From: Mark Musone [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, June 08, 2004 11:24 AM
Subject: [Asterisk-Users
: Tuesday, June 08, 2004 4:37 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] iaxtel 1-800 gateway down?
Ive got similar probs Mark, and no one either here (unless I havent got
thru the pile yet) or on the IRC channel last nite answered. Ive simply
got
no response when I try to use Iaxtel to call
I've been playing around with asterisk for the last few weeks and now I have
the system up and running but whenever I make a call using iaxtel all is
good for the first call. After I hang up the call the d-link router looses
it's mind and must be rebooted. Nothing IP will work through the router
check for a firmware update first. i had problems with a d-link until i did
a firmware update and that fixed it.
- Original Message -
From: Christopher C. Howard [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, April 20, 2004 2:21 PM
Subject: [Asterisk-Users] iaxtel and d-link
Title: IAXTel toll-free gateway
Is anyone else having trouble placing toll-free calls though IAXTel lately? Mine just stopped working yesterday, yet I seem to be able to make 1-700 calls.
-brian
1-700-676-3830
Is anyone else having trouble placing toll-free calls though IAXTel lately?
Mine just stopped working yesterday, yet I seem to be able to
make 1-700 calls.
It's up/down/etc rather frequently, so no surprise. Good thing it's not
a paid service or we'd all have an issue. Consider it as a
With entries in sip.conf, I can route incoming SIP calls with an
extension specified in the register command:
register = user:[EMAIL PROTECTED]/123
The register command in iax.conf does not support specifying the
extension.
If I want to register multiple IAXTel accounts, how can I make them
You do this with contexts attached to the [provider] section in the iax.conf
file and you provide coresponding contexts and extensions in your
extensions.conf file.
John
Barton Hodges wrote:
With entries in sip.conf, I can route incoming SIP calls with an
extension specified in the register
Both register commands register with the iaxtel provider. No matter
which number is dialed to reach Asterisk, it takes you to the same
[provider] section, and thus the same context. I need for 2 register
commands, registering to the same provider, to branch to different
contexts or extensions.
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