I administer a group of Asterisk servers running a mix of 10.3, 10.4, and
1.8.8.1 (mostly 10.4). One of those servers is a call concentrator/relay for
E911 service. All of the other servers make an IAX connection to the relay
server, which then hands off to a SIP trunk to my E911 provider.
- Original Message -
From: Noah Engelberth n...@directlinkcomputers.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, July 3, 2012 10:56:10 AM
Subject: [asterisk-users] IAX trunking stopped working
I administer a group
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Matthew Jordan
Sent: Tuesday, July 03, 2012 2:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] IAX trunking
i have 2 asterisk servers one on CentOS and one on Fedora , i configured IAX
trunking between the 2 servers so that i dial -say from a sip extension 2000 on
fedora server to a sip extension 3000 on CentOS server the call seems to be
established but hangup automatically after very short time
thx guys , i think i discovered the problem , it seems that i had to put the
host=192.168.0.x in iax.conf and not host=dynamic ,,otherwise had to register
the clients
From: [EMAIL PROTECTED]: [EMAIL PROTECTED]: Tue, 26 Feb 2008 15:03:25
+0200Subject: [asterisk-users] iax trunking problem
Hi all,
Is it possible having a trunk using, for exemple, UDP port 4570 and all
the other IAX (not trunk) connection using the standard UDP port 4569?
Thanks.
Ronaldo.
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Hi,
do I have a chance to use iax trunking on OpenBSD where there is no zaptel
driver or ztdummy available? Do I can use sth. else as timing source?
kind regards
Sebastian
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asterisk-users
you can use FreeBSD 6.1 its working fine for me with ztdummy and I'm able to
use IAX2 trunk.
On 6/7/07, Sebastian Reitenbach [EMAIL PROTECTED] wrote:
Hi,
do I have a chance to use iax trunking on OpenBSD where there is no zaptel
driver or ztdummy available? Do I can use sth. else as timing
Hi Arun,
Arun Kumar [EMAIL PROTECTED] wrote:
you can use FreeBSD 6.1 its working fine for me with ztdummy and I'm able
to
use IAX2 trunk.
thanks for the suggestion, but the OS is not really exchangeable. Before I
am going to do this, I'd exchange asterisk with sth. else.
kind regards
Hello, i have been reading the archives and cant find the answer to my
question. Here is the scenario,
SIP dallas gateway -- IAX2 g729 codec -- mex gw
So i send all my sip traffic on iax2 to mexico using g729 and trunk
mode, but the trunk debug shows only mexico doing trunking, in mexico
i get
Ok... it'd be great if someone could explain this to me...
User A on pbx1 wants to dial User B on pbx2. We do a local lookup and don't
find user B on pbx1, so we do a DUNDi lookup of user B, get a result, and place
the call to user B on pbx2 with IAX2.
When pbx2 calls the AGI script that
Ok... it'd be great if someone could explain this to me...
User A on pbx1 wants to dial User B on pbx2. We do a local lookup and don't
find user B on pbx1, so we do a DUNDi lookup of user B, get a result, and place
the call to user B on pbx2 with IAX2.
When pbx2 calls the AGI script that
Hi,
I am experimenting with trunkfreq parameter.
When it is 20ms I can see both parties in IAX session sending IAX frames
every 20ms.
When I modify this parameter to 40ms then I can see that only server that
initiated
IAX connection works properly (i.e. sends IAX frames every 40ms while other
I am having trouble getting trunking to work between a couple of my servers.
All servers are running 1.0.9 stable version, and are working perfectly. All
have a Zaptel card of some description, so timing is not a problem. Each server
has a definition for each other server, using RSA auth, and
For those who are interested, the problem is being caused by IAX using the wrong
outside IP address as it's source address.
(Multi-homed firewall with numerous virtual interfaces). The workaround was to
add additional definitiions to the other servers, so the connection will be
trunked from
hello,
iax trunking not working we actually testing dial 500(Digium) two or
three calls simultaneously but bandwidth graph shows 95 to 100kbps not
match the results shows on wiki iax bandwidth pages i enable trunk=yes
in iax.conf is there any tweaking or optimization because i
desperately need
What codec are you using?
-Original Message-
From: Adnan Ahmed [mailto:[EMAIL PROTECTED]
Sent: Friday, May 13, 2005 9:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] iax trunking not works!
hello,
iax trunking not working we actually
Ahmed [mailto:[EMAIL PROTECTED]
Sent: Friday, May 13, 2005 9:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] iax trunking not works!
hello,
iax trunking not working we actually testing dial 500(Digium)
two or three calls simultaneously
Hi Altus,
I believe the only reason you require a Zaptel card is for a timing
source for trunking.
You could probably get away with the ztdummy device that uses USB as a
timing source.
Go have a look on www.voip-info.org for more info on ztdummy.
Cheers,
-Shaun
Altus Snyman wrote:
Good day all
Good day all
Why do I need a Zaptel card to do trunking in IAX??
What if I only had a voice/iax router?
Is there a way around this?
Thanks
Altus
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Hello,
I am trying to come
up with a good way to enforce a limit on the number of simultaneous calls that
can occupy an IAX trunk at any given time. I have searched around and so
far can't locate a config option that would directly label a IAX trunk with a
specific number to obey (is there
Hi
I would think that it is possible, but never been implemented in other
protocols.
Quintum has something like trunking they call packet saver, and
they use h.323
goodluck with your presenation.
Clive
On 26 Oct 2004 at 10:23, Chris Bshaw wrote:
Hi
I am putting together a
Hi
I am putting together a presentation on VoIP technology, and I just wanted
to make sure I get my facts straight.
I have read that one of the features of the IAX protocol is that in can
trunk multiple calls in a single UDP stream.
Anywhere I have read this, it seems to be implied
Correct, you also greatly reduce overhead which is worth mentioning.
Thanks,
Steve Totaro
[EMAIL PROTECTED]
www.totarotechnologies.com
- Original Message -
From: Chris Bshaw [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, October 26, 2004 5:23 AM
Subject: [Asterisk-Users] IAX
The majority of sip to pstn gateway providers (vonage, voicepulse, and
others) appear to be setup for a one line only type of set up. Their web
sites seem to be heavily geared for these one line setups.
Anyone willing to comment on what type of pricing plans these providers
offer when using iax2
I use Saww.net and Nufone for IAX2 to PSTN at a per min basis. So if i
pushed 5 calls i'd be charge per min for each call. Granted both the
companies above cater to * quite heavily.
On Wed, 2004-02-04 at 01:40, Chris Clifton wrote:
The majority of sip to pstn gateway providers (vonage,
Hi,
Im conecting to * servers using IAX2 no NAT in my setup. I read Wikis
docs and lists archives but there is no recomendation about what to use.
If I understand I can use friend in both sides but its isnt
recomended; so I should define both a peer and a user on each box ?
cause AFAIR in
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