Thank you for your interest in my question and quick response. I am
relatively new to Asterisk, so I have a few specific questions regarding
your suggestions.
Then I will post to the list with a more meaningful subject and results.
On 4/6/2010 10:31 AM, Steve Edwards wrote:
> On Tue, 6 Apr 2010
On Tue, 6 Apr 2010, bob gailer wrote:
> I have2 Trixbox Servers. Each has an IAX trunks to the other. One works
> the other fails:
>
> -- Executing [...@macro-dialout-trunk:19] Dial("SIP/526-09eec7c8",
> "IAX2/InterOffice/210,300,tr") in new stack
> -- Called InterOffice/210
> -- Hungu
I have2 Trixbox Servers. Each has an IAX trunks to the other. One works
the other fails:
-- Executing [...@macro-dialout-trunk:19] Dial("SIP/526-09eec7c8",
"IAX2/InterOffice/210,300,tr") in new stack
-- Called InterOffice/210
-- Hungup 'IAX2/InterOffice-7578'
== Everyone is bus
On Thu, 26 Mar 2009, Andrew Hakman wrote:
> So no one else has a problem routing IAX traffic through an
> intermediate Asterisk server? Does anyone else use Asterisk in such a
> configuration?
I do. Not had a problem apart from when Digium break the protocol.
1.2 -> Interweb -> 1.2 -> Interweb -
I initially had no trunking anywhere, and had the same behavior. I
thought trunking would help, but I can't figure out why the /dev/dahdi
device doesn't get created on C. The dahdi tools / modules don't seem
to have much error / debugging info available, or if they do, I sure
can't find it anywhere
Here's my troubleshooting help -- since the problem sounds like a timing
issue and part of the call is being trunked, then fix your timing problem,
or remove the trunking from A and B then see if the problem goes away.
On Thu, Mar 26, 2009 at 10:50 PM, Andrew Hakman wrote:
> So no one else has a
I'll have to get some VPN's setup, but I will give it a try with SIP.
Thanks for the input - you saved me building 2 more asterisk servers
for testing this issue locally (rather than across 3 networks).
Andrew
On Thu, Mar 26, 2009 at 11:12 PM, Steve Totaro
wrote:
> On Fri, Mar 27, 2009 at 12:50
On Fri, Mar 27, 2009 at 12:50 AM, Andrew Hakman wrote:
> So no one else has a problem routing IAX traffic through an
> intermediate Asterisk server? Does anyone else use Asterisk in such a
> configuration?
>
> On Thu, Mar 26, 2009 at 2:45 AM, Andrew Hakman
> wrote:
>> I'm having a problem with I
So no one else has a problem routing IAX traffic through an
intermediate Asterisk server? Does anyone else use Asterisk in such a
configuration?
On Thu, Mar 26, 2009 at 2:45 AM, Andrew Hakman wrote:
> I'm having a problem with IAX running through an intermediate asterisk
> box. Perhaps a small di
I'm having a problem with IAX running through an intermediate asterisk
box. Perhaps a small diagram will explain the situation better:
*A --- [cloud (public internet)] --- *B [cloud
(private network)]--- *C
Asterisk server's A, B, and C, are all connected together with IAX
n the basis of the information.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim Panton
Sent: Thursday, November 02, 2006 04:10
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] IAX problem
On 2 Nov 2006,
On 2 Nov 2006, at 02:38, Itamar Lavender wrote:
Hi All,
I'm having problem with IAX, I'm trying to connect to speex.co.il
from asterisk using:
register => username:[EMAIL PROTECTED]
What does the rest of iax.conf look like ?
Auth is a 2 way thing - you have sucessfully registered with
Hi All,
I'm having problem with IAX, I'm trying to connect to speex.co.il
from asterisk using:
register => username:[EMAIL PROTECTED]
and I cant get it to work.
Maybe someone who already got this to work will help…
When dialing my speex extension I see the next output from
riday, December 30, 2005 6:38
PM
Subject: [Asterisk-Users] IAX problem -
Bug or Compatibility issue?
Hello All,
I am looking for more thorough debug than
the one provided by the command "iax2 debug". Could anybody point me a good
documentation about this?
I have a
Hello All,
I am looking for more thorough debug than
the one provided by the command "iax2 debug". Could anybody point me a good
documentation about this?
I have a issue with IAX connection.
Sometimes it stucked. If so, I have to restart my asterisk through CLI
command "restart now".
C
Hi all,
I have 2 servers and I'm trying to configure iax to call from
Server2 (fxo) to Server1 (sip extension)
Server1: 2 sip's extension (123 and 321)
Server2: TDM-400 1 fxo(extension 999, channel 3) and 1 fxs (channel 4)
from 123 to 321 it's all right in both ways
from 999 to 123 no au
> > > > For #2, incoming calls would be handled with:
> > > > exten => 6789,1,Dial(SIP/1235)
> > > >
> > > Besides that :
> > >
> > > *CLI> iax2 show registry
> > > Host UsernamePerceived Refresh State
> > > X.X.X.X:4569 Username1 [MYIP]:4569
On Mon, Sep 26, 2005 at 11:02:47AM -0600, Rich Adamson wrote:
>
> > > For #2, incoming calls would be handled with:
> > > exten => 6789,1,Dial(SIP/1235)
> > >
> > Besides that :
> >
> > *CLI> iax2 show registry
> > Host UsernamePerceived Refresh State
> > X.X.
> > Two approaches that have been rather common are:
> > 1. use the separate contexts for each did,
> > 2. in the register statement, add /1234 at the end; like
> > register => username:[EMAIL PROTECTED]/6789
> >
> I don't think it will work , iax statement don't have
> exten on end.
>
>
On Sun, Sep 25, 2005 at 07:26:12AM -0600, Rich Adamson wrote:
>
> Two approaches that have been rather common are:
> 1. use the separate contexts for each did,
> 2. in the register statement, add /1234 at the end; like
> register => username:[EMAIL PROTECTED]/6789
>
I don't think it will wo
> I've 3 iax connections to my provider , each of them have own DID ,
>
> PH1<|
> |
>\/
> PH2<-->|-| <---> ||<-- DID1
>| A1 | <---> |ISP |<-- DID2
> PH3<-->|-| <---> ||<-- DID3
>
>
Hi
I've 3 iax connections to my provider , each of them have own DID ,
PH1<|
|
\/
PH2<-->|-| <---> ||<-- DID1
| A1 | <---> |ISP |<-- DID2
PH3<-->|-| <---> ||<-- DID3
I had iax phone
Hi,
I have some issues with communication between to * servers. They are
connected over DSL (3Mbps). One is behind NAT and the other on routable
network. Almost every time caller will hear the other end like fast forward
while the other end will have perfect quality. It doesn't matter if we use
SIP
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