Hello All
Anybody had used ooH323 for asterisk
i using ooH323-0.7.2 and asterisk CVS may 2005. OpenH323 1.17.1 and pwlib 1.9.0 and GNUGK 2.0.2
audio is very good, better than SIP and IAX, but i have problem.
how to router call from openh323 to outside PSTN.
my h323.conf setting
; Objective
Hi
Please use proper quoting...
See below
On Sat, Oct 08, 2005 at 12:23:21AM -0400, [EMAIL PROTECTED] wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir
Cohen
Sent: Friday, October 07, 2005 8:17 AM
To: asterisk-users@lists.digium.com
I don't know, after looking at their roadmap I don't get it. It
must be the asterisk commit policies that are driving this. They
have some good ideas, but they are going about this the wrong way if
their goal is to create a successful fork of asterisk.
Chris
On 8 Oct 2005, at 09:49, snacktime wrote:
I don't know, after looking at their roadmap I don't get it. It
must be the asterisk commit policies that are driving this. They
have some good ideas, but they are going about this the wrong way
if their goal is to create a successful fork of
Dear All;
Hope you are fine. I am developing an application for IAX using C#, and I have a problem sending frames to the server, I dont know how exactly I can send the frames. I have saw this site http://splurge.peoples-wireless.com/iax/ but I couldnt understand how can combine the
Dear All;
Hope you are fine. I am developing an application for IAX using C#, and I have a problem sending frames to the server, I dont know how exactly I can send the frames. I have saw this site http://splurge.peoples-wireless.com/iax/ but I couldnt understand how can combine the
Morning all,
we just download the today CVS and face a problem: in a context we want
to use brackets for matching extensions like exten =
_48[1-478]X.,1,Goto(validate,1) for instance. When dialing a number
like 4832285 we receive
== Auto fallthrough, channel 'Local/[EMAIL PROTECTED],2'
Sorry for noise, problem is solved. It was an priority error.
Administrator TOOTAI a écrit :
Morning all,
we just download the today CVS and face a problem: in a context we
want to use brackets for matching extensions like exten =
_48[1-478]X.,1,Goto(validate,1) for instance. When dialing a
Mike M wrote:
On Fri, Oct 07, 2005 at 09:45:53PM -0400, Paul wrote:
Also consider that there are situations where 100% open source is never
allowed. Check out visa/mastercard processor certification for a good
example. Digium dual licensing availability means I could actually stand
a
On Sat, Oct 08, 2005 at 07:41:48AM -0400, Paul wrote:
Credit card processing would be a good example. You could design *-based
systems for both the client(merchant) and server(processor) functions
but last I knew visa/mc would not certify open source solutions.
Note that you can use
On Sat, Oct 08, 2005 at 11:56:40AM +0200, Youssef Sayed wrote:
Dear All;
Hope you are fine. I am developing an application for IAX
using C#,
Any reason for not using iaxclient?
--
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |
Hello.
This is d Asterisk users list.
~Madhawa
Youssef Sayed wrote:
*Dear All;*
* *
* Hope you are fine. I am developing an application for IAX using
C#, and I have a problem sending frames to the server, I don’t know how exactly
I can send the frames. I have saw this site
thanks for that, i knew already but it misses the actual version
Jesse Keating wrote:
On Fri, 2005-10-07 at 11:17 +0200, Kib Eki wrote:
Hello,
can anybody tell me where to get the latetest SIP Firmware 1.6.2 for the Polycom
phones?
http://www.freedomphones.net/polycom/files/
Tzafrir Cohen wrote:
On Sat, Oct 08, 2005 at 07:41:48AM -0400, Paul wrote:
Credit card processing would be a good example. You could design *-based
systems for both the client(merchant) and server(processor) functions
but last I knew visa/mc would not certify open source solutions.
On Sat, Oct 08, 2005 at 09:20:07AM -0400, Paul wrote:
Tzafrir Cohen wrote:
On Sat, Oct 08, 2005 at 07:41:48AM -0400, Paul wrote:
Credit card processing would be a good example. You could design *-based
systems for both the client(merchant) and server(processor) functions
but last I
If you have configured Asterisk to remote to a SIP provider, how do you
verify
that the registration has been successful?
sip show peers
Or, do 'sip show registry' depending on who is registering with who.
From your words, I'd guess this is more of what you're looking for.
Hello,
I'm trying to use the version 0.6 of chan_capi-cm for outgoing calls it
works perfectly but for incoming calls it doesn't work.
I tried to set an extension to dial my from-pstn context and it works.
So I think there's a problem with my capi.conf or something...
Here's a debug when
On Sat, 8 Oct 2005, Cedric Fontaine wrote:
Hello,
I'm trying to use the version 0.6 of chan_capi-cm for outgoing calls it
works perfectly but for incoming calls it doesn't work.
I tried to set an extension to dial my from-pstn context and it works.
So I think there's a problem with my
I have been able to configure my Asterisk BOX to receive calls from
Mediatrix 1204.
I'm having problems sending calls out via my Mediatrix unit.
The SIP Invite is sent to the Mediatrix but the Mediatrix unit sends
back a Status : 480 Temporarily Unavailable.
This is my configuration
Hello,
I 'll ask to my reseller
Harry
--- [EMAIL PROTECTED] a écrit :
thanks for that, i knew already but it misses the
actual version
Jesse Keating wrote:
On Fri, 2005-10-07 at 11:17 +0200, Kib Eki wrote:
Hello,
can anybody tell me where to get the latetest SIP
Firmware 1.6.2
Armin Schindler wrote:
Please increase verbosity (set verbose 5) and switch on debugging
(capi debug). Then you should see what's happening.
*CLI set verbose 6
Verbosity was 3 and is now 6
*CLI capi debug
CAPI Debugging Enabled
CONNECT_IND ID=001 #0x06ed LEN=0042
Controller/PLCI/NCCI
I am using Teliax to terminate my calls, and I have 3 licenses' for
g729 from Digium. show translations verifies that the registration
took place.
When I place a call, having allow=g729 as the only allow option in
iax.conf, I get the following error:
WARNING[361]: chan_iax2.c:6017
On Sat, 8 Oct 2005, Cedric Fontaine wrote:
Armin Schindler wrote:
Please increase verbosity (set verbose 5) and switch on debugging
(capi debug). Then you should see what's happening.
*CLI set verbose 6
Verbosity was 3 and is now 6
*CLI capi debug
CAPI Debugging Enabled
Dear all expert,
(i posted this question one time, but i couldn't reach the answer
-so allow me to post here)
1)I download asterisk realse version 1.2 beta1.
After that i compile it successfully and run it with:
asterisk -vvvc
2)I follow the instruction in
Make sure you have g729 turned on from the Teliax customer panel on their
website.
Chris Coulthurst
[EMAIL PROTECTED]
- Original Message -
From: Rich Adamson [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com; John Reynolds
On Sat, Oct 08, 2005 at 09:20:07AM -0400, Paul wrote:
I find that amusing. I have a lot of experience with disassembly. I have
even reverse-engineered machine language code that ran on custom
processors which means you have to reverse-engineer the instruction set
as part of the task.
I
one more thing is that:
in my fedora core 4 where Asterisk is running, i use netstat -ta
i can't see: asterisk is listening in port 5060 (which include in sip.conf)
On 10/8/05, julien bos [EMAIL PROTECTED] wrote:
Dear all expert,
(i posted this question one time, but i couldn't reach the
Hello everybody,
Is there anybody successfully have Asterisk
running particularly on Sun Fire V100 (64 bits) with Solaris 9?
Any hints and suggestions would be much
appreciated.
Cheers,
Anto
___
--Bandwidth and Colocation sponsored by
Wait for the next UTStarCom version... Called F3000, Im not sure, but
something like that.
It will have better battery performance and will have 802.11g
support, and many other improvements. It will be available soon.
Denis.
On 07 de out de 2005, at 00:54, Andy Hamilton wrote:
Anyone
On Friday 07 October 2005 19:10, Troy Settle wrote:
Nice smartass remark... of course anyone can register a domain name.
Is forking asterisk legal? Of course it is! Asterisk is under the GPL,
which means that anyone can fork it at any time for any reason.
Look at this in a positive
On Sat, Oct 08, 2005 at 11:59:04AM -0400, Mike M wrote:
On Sat, Oct 08, 2005 at 09:20:07AM -0400, Paul wrote:
Closed source might delay the cracker but it also delays pre-crack and
post-crack countermeasures.
What's the alternative? Open source? Cracking is unnecessary with open
if you do a netstat -an, you will see asterisk on udp port 5060.
one more thing is that:
in my fedora core 4 where Asterisk is running, i use netstat -ta
i can't see: asterisk is listening in port 5060 (which include in sip.conf)
On 10/8/05, julien bos [EMAIL
Nice smartass remark... of course anyone can register a domain name.
Is forking asterisk legal? Of course it is! Asterisk is under the GPL,
which means that anyone can fork it at any time for any reason.
Look at this in a positive light... many open source projects have
forked,
Nice smartass remark... of course anyone can register a domain name.
Is forking asterisk legal? Of course it is! Asterisk is under the
GPL,
which means that anyone can fork it at any time for any reason.
Look at this in a positive light... many open source projects have
Rich Adamson wrote:
I'm certainly not an expert on this topic, but if OpenPBX stays with
GPL, it would appear that asterisk could use any piece developed under
OpenPBX (unless someone there puts restrictions on individual pieces).
No, since Asterisk requires that copyright be assigned to
Rich Adamson wrote:
I'm certainly not an expert on this topic, but if OpenPBX stays with
GPL, it would appear that asterisk could use any piece developed under
OpenPBX (unless someone there puts restrictions on individual pieces).
Only if the copyright holder(s) of that code choose to
Tony Hoyle wrote:
No, since Asterisk requires that copyright be assigned to Digium for all
patches. Submitters to OpenPBX may be unwilling to do this, especially
since that's one of the main reasons for its existance...
Please stop spreading misinformation. We have addressed this at least
On Sat, Oct 08, 2005 at 08:43:07PM +0300, Tzafrir Cohen wrote:
On Sat, Oct 08, 2005 at 11:59:04AM -0400, Mike M wrote:
On Sat, Oct 08, 2005 at 09:20:07AM -0400, Paul wrote:
Closed source might delay the cracker but it also delays pre-crack and
post-crack countermeasures.
What's the
On 10/8/05, Paul [EMAIL PROTECTED] wrote:
Mike M wrote:On Fri, Oct 07, 2005 at 09:45:53PM -0400, Paul wrote:Also consider that there are situations where 100% open source is neverallowed. Check out visa/mastercard processor certification for a good
example. Digium dual licensing availability means
On 10/8/05, Kevin P. Fleming [EMAIL PROTECTED] wrote:
Tony Hoyle wrote: No, since Asterisk requires that copyright be assigned to Digium for all patches. Submitters to OpenPBX may be unwilling to do this, especially since that's one of the main reasons for its existance...
Please stop spreading
snacktime wrote:
Being that Digium wants to be able to sell a commercial version, I don't
see how they could have been more accomodating then this. Digium can
They could just use the GPL as is, since they chose the license in the
first place.. they clearly have no issues with it.
They
On 10/08/05 13:32 Kevin P. Fleming said the following:
Once I return from Astricon, we will use this new build system to
produce FreeBSD modules for the same processor architectures, and also
this is wonderful ! how long has it been since licensed g.729 codecs were
available from digium
On 10/07/05 23:28 Jon Pounder said the following:
There are people out there who wish to contribute, and not have their work
lost on an individual project website since they do not choose to accept
digium's terms to contribute to asterisk. This gives them an opportunity
to do so, and have
There are a lots of strange things in my server fedora core 4 for asterisk.
The first thing is that:
- after (make, make install, make sample), i try to run asterisk by asterisk -vc from
terminal. Then, i can enter CLI screen.
-now i try to use netstat -a or netstat --listening, i can't see
I am in the process of converging PSTN and internet. I am now
looking to migrate from pots to VoIP handsets that are IEEE
802.3af (POE) compliant.
My question is this. Does the 3Com 3101 basic phone (P/Ns 3C10401A or
3C10401SPKRA) work with Asterisk???
Dinesh Nair wrote:
this is wonderful ! how long has it been since licensed g.729 codecs
were available from digium for freebsd ?
They have been on the web/FTP sites for some time, in the 'unsupported'
directory.
___
--Bandwidth and Colocation
On 10/8/05, Tony Hoyle [EMAIL PROTECTED] wrote:
snacktime wrote: Being that Digium wants to be able to sell a commercial version, I don't see how they could have been more accomodating then this. Digium canThey could just use the GPL as is, since they chose the license in the
first place.. they
On Sat, Oct 08, 2005 at 08:41:00PM +0100, Tony Hoyle wrote:
TBH I'd rather digium had chosen something like BSD to start with and
avoided all the GPL politics but the situation we have is the one we have.
But then you wouldn't have to pay them if you wanted your own propritary
fork. Not to
On 10/09/05 03:58 Kevin P. Fleming said the following:
Dinesh Nair wrote:
this is wonderful ! how long has it been since licensed g.729 codecs
were available from digium for freebsd ?
They have been on the web/FTP sites for some time, in the 'unsupported'
directory.
and they still go
On 10/09/05 02:46 Rich Adamson said the following:
I'm certainly not an expert on this topic, but if OpenPBX stays with
GPL, it would appear that asterisk could use any piece developed under
OpenPBX (unless someone there puts restrictions on individual pieces).
asterisk could, but i doubt
On 10/09/05 02:46 Rich Adamson said the following:
I'm certainly not an expert on this topic, but if OpenPBX stays with
GPL, it would appear that asterisk could use any piece developed under
OpenPBX (unless someone there puts restrictions on individual pieces).
if it's a fork of asterisk, it
Dinesh Nair wrote:
and they still go for US$10 a pop ?
Patent indemnification licenses are completely separate from the codec
binary you choose to use. There is no price difference for CPU type, OS
platform or anything else.
___
--Bandwidth and
Hi,
When we make an
outgoing call on ISDN (zaphfc) with overlap dialing we get immidiate hangup
after answer. But when we place a full number before dialing everything is ok.
Any help appriciated!! Thanks
here is info with
debug:
== Primary
D-Channel on span 1 up -- Executing
On 10/09/05 04:45 Kevin P. Fleming said the following:
Dinesh Nair wrote:
and they still go for US$10 a pop ?
Patent indemnification licenses are completely separate from the codec
binary you choose to use. There is no price difference for CPU type, OS
platform or anything else.
Have you founy any real life performance benefit of x86_64 (particularly EM64T on Xeon) as apposed to plan old x86?
On 10/9/05, Kevin P. Fleming [EMAIL PROTECTED] wrote:
Dinesh Nair wrote: and they still go for US$10 a pop ?Patent indemnification licenses are completely separate from the
Eric Bishop wrote:
Have you founy any real life performance benefit of x86_64 (particularly
EM64T on Xeon) as apposed to plan old x86?
Yes. On my Athlon-64 2200+ system on my desk, the 'opteron' version
encodes a 6722 block sample file in 478ms; the 'i686' version does it in
514ms. The
The F3000 is not anticipated to be available for distribution until late
December/January, FYI.
Cory Andrews
Senior Partner
+++
VOIPSupply.com
454 Sonwil Drive
Buffalo, NY 14225
+++
voice - 716.630.1555 X22
email - [EMAIL PROTECTED]
fax - 716.630.1548
Denis Galvão - iSolve
How does one check what codec translations are in use in a call?
I am connecting to sip system which says 488 4XX Not Acceptable Here. I don't
know what is stopping the call from being accepted and I'd like to know if
there are codec issues involved.
/Obelix
On a dual processor Xeon (EM64T) would you reccomend turning
hypertreading on or off? I tend go for it off dual processor machines
just in case 2 processes end up on the one physical processor rather
than 2 processes on 2 different physical processors. What do you think?On 10/9/05, Kevin P.
Eric Bishop wrote:
On a dual processor Xeon (EM64T) would you reccomend turning hypertreading
on or off? I tend go for it off dual processor machines just in case 2
processes end up on the one physical processor rather than 2 processes on 2
different physical processors. What do you think?
I
On 10/09/05 06:01 Obelix said the following:
I am connecting to sip system which says 488 4XX Not Acceptable Here. I don't
know what is stopping the call from being accepted and I'd like to know if
there are codec issues involved.
it's possible. try connecting with 'sip debug' turned on, and
Hi,
Is there anyway to eliminate the color coding (for
example [1;36;40m) to be stored in asterisk log file?
Regards.
__
Yahoo! Music Unlimited
Access over 1 million songs. Try it free.
http://music.yahoo.com/unlimited/
On Sat, Oct 08, 2005 at 04:15:59PM -0700, Samy Antoun wrote:
Hi,
Is there anyway to eliminate the color coding (for
example [1;36;40m) to be stored in asterisk log file?
Not to use the color options?
It seems that exactly the same prints go to the log and to the CLI,
right?
--
Tzafrir
Tzafrir Cohen wrote:
It seems that exactly the same prints go to the log and to the CLI,
right?
Asterisk already strips the color codes before putting the output into
the log files.
___
--Bandwidth and Colocation sponsored by Easynews.com --
--- Kevin P. Fleming [EMAIL PROTECTED] wrote:
Asterisk already strips the color codes before
putting the output into
the log files.
Kevin,
This is NOT true, bellow is some of my asterisk log
file:
Oct 8 16:41:49 VERBOSE[4016]: -- Executing
When I try to dial through a pbx I receive this message
to 216.127.66.119:0
Oct 8 23:53:48 WARNING[22849]: chan_sip.c:611 __sip_xmit: sip_xmit of 0x81331cc
(len 670) to 216.127.66.119 returned -1: Invalid argument
Retransmitting #5 (no NAT):
The line is silent and nothing happens.
/Obelix
I have been trying to connect via sip and things don't seem to work. What do
messages like this mean?
Oct 9 00:33:57 WARNING[22849]: chan_sip.c:611 __sip_xmit: sip_xmit of 0x81ab834
(len 361) to 216.127.66.119 returned -1: Invalid argument
Oct 9 00:33:58 WARNING[22849]: chan_sip.c:694
Check your sip.conf settings and make sure you have nat=yes
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Obelix
Sent: Saturday, October 08, 2005 7:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Does
Steve Kennedy wrote:
On Sat, Oct 08, 2005 at 08:43:07PM +0300, Tzafrir Cohen wrote:
On Sat, Oct 08, 2005 at 11:59:04AM -0400, Mike M wrote:
On Sat, Oct 08, 2005 at 09:20:07AM -0400, Paul wrote:
Closed source might delay the cracker but it also delays pre-crack and
post-crack
I am not able to register an external ATA on my asterisk 2.0 Beta
This is the debug
Any idea?
server01*CLI
-- SIP read from CLIENTIP:5060:
REGISTER sip:SIPSERVERIP SIP/2.0
Via: SIP/2.0/UDP PRIVATEIP;rport;branch=z9hG4bK455E4AEA9A9954FB135D6D788DA2
From: sip:[EMAIL PROTECTED];tag=1564789518
To:
Steve Underwood wrote:
Steve Kennedy wrote:
On Sat, Oct 08, 2005 at 08:43:07PM +0300, Tzafrir Cohen wrote:
On Sat, Oct 08, 2005 at 11:59:04AM -0400, Mike M wrote:
On Sat, Oct 08, 2005 at 09:20:07AM -0400, Paul wrote:
Closed source might delay the cracker but it also delays
Recently I've noticed two bits of odd behavior with respect to regcontext/regexten in CVS HEAD 1.2 Beta1, and I was wondering if anyone could shed some light on this.
I've set up a regcontext in sip.conf. I've set up two users with regexten entries, one in sip.conf and one in a mysql realtime
Hi, all
I have installed TDM400 with 1 FXS and 1 FXP ports.
Now I am goig through documentation on how to configure it.
It mentions 3 protocols: Loopstart, Groundstart and Koolstart. Which one do
I use?
Can someone send me sample zaptel.conf file for Australia? This will save me
some time
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