Hi,
Can you tell me how you could get the RC1 to work because for me RC1 does
not work as well as I got NO audio on both sides. I have use the stable
version of asterisk and it works for me with H323 audio, but once I upgraded
to RC1 , calls were not completing at all with H323 and I had to
Dear All
Now that RC1 is buggy, should we go back to the Dev cvs Head? OR how do we
get the bug resolved in RC1 version and where could we get them?
Thanks
TC
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of hank
Sent: Saturday, August 07, 2004 1:02 PM
To:
in and H323 out and I tried sip in and H323 out but the same results I get.
Thanks
TC
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of T. Chan
Sent: Saturday, August 07, 2004 4:27 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] RC1 problem
Hi, Scott
Thanks for your information. I have worse luck in load testing with
asterisk.
I have tried both SIP and H323 inbound calls and terminating on PSTN PRIs. I
am using a single Xeon 2.8G chip and 512M Ram and in both cases, once it
gets more than a T1, call quality starts to degrade with
Dear All,
There is a question about the H323 channels (H323 driver, not OH323), it is
not passing CallerID. If a call comes in on ZAP and out H323 to another
gateway, the other gateway does not see the ANI, and if Asterisk is used as
a passthrough, it receives callerID from the other gateway, but
Hi,
I am the unlucky one, I have similar problem, but I am mostly using
safe_asterisk, and this stop now...restart now never works, with neither
0.6.3 nor 0.6.2
TC
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Anthony Law
Sent: Thursday, July 08, 2004
Dear All,
I don't know but I tried all 0.6.x version of OH323 and normally I use
safe_asterisk to start asterisk, and everytime when I use 'stop now' to
terminate asterisk, it does not do anything, and you are rite, I have to use
kill -9 to kill the PIDs and threads. However, if I use asterisk
Hi, Anthony, can you try issuing stop now on safe_asterisk and see if it
works please? I am used to using safe_asterisk and with this new version and
when I tried issuing stop now, it did not do it.
Thanks
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of
cvs checkout -D mm/dd/yy asterisk-addons
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Harold
Workman
Sent: Monday, June 28, 2004 1:03 AM
To: [EMAIL PROTECTED]; T. Chan
Cc: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] asterisk addon mysql
Tommy
Hi,
I got the same thing, so what I did was for the asterisk-addons, I used CVS
April instead of the most current CVS and it worked. Of course, I would have
liked to use the most current CVS of asterisk-addons as well, but since the
old version works with the most current version of asterisk
Hi, Scott. Are you telling me that this native h.323 has been hardcoded
with fast start? Can you tell me where in ast_h323.cpp it is that you
disabled this faststart? Have you tried using the Stable cvs of the
Asterisk.
Can you let me know which version of the OH323 are you using ? Is it the
Hi, Scott, I am very interested in knowing the result of your loading test,
please share after you have done it. Are you using Asterisk as a
pass-through (kinda softswitch) or do have have digium hardware and use it
as an endpoint, because I believe the maxiumum number of channels you can
run
]
[mailto:[EMAIL PROTECTED] Behalf Of Jeremy
McNamara
Sent: Saturday, June 26, 2004 4:28 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] NO AUDIO IN BOTH DIRECTIONS
T. Chan wrote:
Jeremy, any way to fix that? Thanks again.
I've spent many many days trying to duplicate any of these problems
Dear All,
I second that, I have spent months on this. Since a few months ago when the
recommended H323 libraries were changed to pwlib 1.5.2 and OpenH323 12.2.2,
I have not been able to get audio. I tried different configurations and no
luck. I have spent months and lots of hours on it, but I was
. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto California London England
www.evtmedia.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of T. Chan
Sent: Friday, June 25, 2004 8:35 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] NO AUDIO
Jeremy
I speak for myself, I have been testing with oh323 driver as well, because
in my case, your h323 driver is not working, it was working before, but then
when I started to upgrade to 0.7.0 version of asterisk and from that point
onwards (beginning of January), calls have had no audio. I
-Users] oh323
T. Chan wrote:
Jeremy
I speak for myself, I have been testing with oh323 driver as well, because
in my case, your h323 driver is not working, it was working before, but
then
when I started to upgrade to 0.7.0 version of asterisk and from that point
onwards (beginning of January
: Re: [Asterisk-Users] oh323
T. Chan wrote:
Jeremy,
Yes, I felt that it was important to report my trouble and I did it three
times, reporting to the asterisk community, but for some reasons, I was
not
being responded to at all. I thought my messages were embedded among the
hundreds of them
Hi Glen, I have had the same problem for quite awhile, since around
February, all cvs codes that I have tried, and with h323, I have been
getting no audio. I am forced to stay with mid-Jan version of the cvs
because of this. I tried using ulaw, g729, but same results, I have in a few
occasions
Hi, I am dealing with Vontec Communication in Vancouver, they are selling
DID and DID usages terminating on your equipment via VOIP. You can contact
[EMAIL PROTECTED]
TC
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Linus Surguy
Sent: Monday, June 14,
Hi, I had the same problem, this is when I tried to run the latest CVS head.
I had to redownload the asterisk version dated 8June to compile. I don't
know if this is a bug with asterisk or if this is a compatible issue.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
: Rechenberg, Andrew [mailto:[EMAIL PROTECTED]
Sent: Tuesday, June 01, 2004 5:45 PM
To: [EMAIL PROTECTED]
Cc: T. Chan
Subject: RE: [Asterisk-Users] RE: H323
I am having a similar problem with one-way audio from an Avaya hardphone
calling a SIP soft phone. Audio from the hardphone is heard
Dear Michael
I tried using the newest version of your H323 driver, but somehow it seems
that it is not hanging up the channels and for some reasons, it is NOT
writing my cdr to the mysql database, it was writing properly before. As you
can see , the call finished at 2:40:12 but refused to hang up
Dear All,
Thanks, but I was already using a pre May 25 CVS version. Does anyone else
have any idea please? Thanks
TC
-Original Message-
From: Dmitry Mishchenko [mailto:[EMAIL PROTECTED]
Sent: Tuesday, June 01, 2004 6:22 AM
To: [EMAIL PROTECTED]; T. Chan
Subject: Re: [Asterisk-Users] RE
Dear All,
I have used Asterisk for a few months and I have been using a January CVS
version, it has been working but has been regularly crashing. I use Asterisk
mostly as a softswitch application receiving H323 calls from customers and
send to H323 carriers. Since I have been using an older CVS
Dear All,
I would like to install Asterisk to support my VOIP business, intending to
use Asterisk as a VOIP softswitch and/or gateways endpoints. I am
considering using either FreeBSD or Linux Redhat.
Can someone share the experience as to which OS would provide a better
environment for running
-Users] asterisk-oh323, new version 0.5.10
T.38 FAX is in the short-term plans for asterisk-oh323.
Michael
T. Chan wrote:
Dear Michael
Do you foresee implementing these in the near future, one or the other or
both? Thanks
Tc
-Original Message-
From: [EMAIL PROTECTED]
[mailto
-Users] asterisk-oh323, new version 0.5.10
Hi TC,
T.38 FAX and native bridging are not supported by asterisk-oh323.
Michael.
T. Chan wrote:
Dear Michael,
Does your H323 driver run T38 Fax? Also, does your H323 driver have the
capability of just proxying signal, and NOT proxying signal
Dear Michael,
Does your H323 driver run T38 Fax? Also, does your H323 driver have the
capability of just proxying signal, and NOT proxying signal and media, just
like the canrevite=yes in the sip scenario? Thanks
TC
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
Miguel
Can you let me know where I can find the gphone information so that I can
give it a try, thanks
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Miguel
Cavazos
Sent: Thursday, March 04, 2004 5:00 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users]
appreciated.
- SamW
-Original Message-
From: T. Chan [mailto:[EMAIL PROTECTED]
Sent: Friday, February 20, 2004 12:48 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] RE: codec negotiation prob solved
I have the same problem, most carriers out there deal with both g723.1
or
g729
Hi, Todd
Did you notice that when you made the calls, were the calls indicated as
answered, in both cases? And if so, did the indication answered pop up
when the calls were actually picked up and answered or right after the call
setup was completed
-Original Message-
From: [EMAIL
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] H323 calls drop on connect
Right after the call setup was completed...
Todd
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of T. Chan
Sent: Tuesday, March 02, 2004 4:01 PM
To: [EMAIL PROTECTED]
Subject: RE
. There might be better ways. But if you're interested in
pursuing it this way and not sure how to do, please follow up with another
question...
Ron
On Fri, 27 Feb 2004, T. Chan wrote:
Hi, all
I wonder when passing calls through asterisk with H323, is there anyway to
find out what codec the calls
Hi, all
I wonder when passing calls through asterisk with H323, is there anyway to
find out what codec the calls are using, anyone can help please, thanks alot
!
TC
---
Outgoing mail is certified Virus Free.
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Version: 6.0.590 / Virus
I have the same problem, most carriers out there deal with both g723.1 or
g729. During passing through via Asterisk, carrier customers will send us
calls broadcasting both codecs with one having priority over the other, the
way it is supposed to work is that asterisk will try to negotiate the top
Dear All,
I need your advices, all you experts out there, I have spent alot of time
testing but just could not get it to work, so I need your assistance please.
I have been trying to passthrough calls with asterisk, that is, receiving
calls from customer via VOIP and then directly send the calls
Dear All,
I have a very simple question but could not find any information from the
internet.
Is there anyway to match code on extensions.conf without having to specify
the number of digits?
For example, if I want to send 01163 (Philippines to a certain IP address),
is there anyway simpler to
Sent: Friday, February 06, 2004 4:31 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Fast question on extension matching
T. Chan wrote:
Dear All,
I have a very simple question but could not find any information from the
internet.
Is there anyway to match code on extensions.conf without
Dear All,
Now, it seems that both IAX and SIP can have the two endpoints communicate
the media directly without the media stream passing through the asterisk,
can we do the same with H323 too?
TC
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Adam Hart
!
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de T. Chan
Enviado el: lunes, 02 de febrero de 2004 23:56
Para: [EMAIL PROTECTED]
Asunto: RE: [Asterisk-Users] Can audio streams go client to cleint with
IAX?
Dear All,
Now, it seems that both IAX
Dear all,
I have been testing with Asterisk for a bit of time and yesterday I tried to
upgrade my kernel to 2.4.20-28, but after I upgraded the kernel, I was not
able to compile Zaptel. The kernel runs good and everything intact, I was
trying to recompile Asterisk in order to make sure that
(or 9) since this is the most up-to-date
kernel. Did you install the kernel-sources and kernel-util rpms as well?
You'll need these in order to compile and install zaptel.
-sb
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of T. Chan
Sent: Friday, January 30
Hi, all !
I have been testing with Asterisk for a bit of time and yesterday I tried to
upgrade my kernel to 2.4.20-28, but after I upgraded the kernel, I was not
able to compile Zaptel. The kernel runs good and everything intact, I was
trying to recompile Asterisk in order to make sure that
Dear all,
I have been testing with Asterisk for a bit of time and yesterday I tried to
upgrade my kernel to 2.4.20-28, but after I upgraded the kernel, I was not
able to compile Zaptel. The kernel runs good and everything intact, I was
trying to recompile Asterisk in order to make sure that
I think what Todd was referring to was to JUST do the signaling proxy on the
Asterisk but not proxying the media. The Asterisk box would ONLY do the
signaling handling between the two endpoints and hang over the media stream
to go directly between the two endpoints. This is a question I was
Dear All,
Just an experience to run by all you experts out there. I have started to
put more VOIP calls into Asterisk, most are pass-through calls and some are
terminating on the Digium card to PSTN. Whenever I get to 10 calls or more,
I would start to get choppy sound. I tried to ping other IP
on each port. Once you've done that, you should be well on your way
to
a reliable, scalable solution.
Quoting T. Chan [EMAIL PROTECTED]:
Dear All,
Just an experience to run by all you experts out there. I have started to
put more VOIP calls into Asterisk, most are pass-through calls and some
senstive,
it
should really be the only primary process running on your machine.
AND, YOU SHOULD NOT BE RUNNING XWINDOWS. The os should have been installed
in
console mode, and as little as possible relating to X installed.
Quoting T. Chan [EMAIL PROTECTED]:
Thanks alot, Ray
Well, looking
Dear All,
I will send you a couple more emails, the story is something like for some
reasons, Nufone website was not working for a bit of time or something, and
this and that. So, there were some customers / potential customers who
posted in the asterisk discussion forum titled 'Has Nufone gone
Hi,
Deepak, how are you?
I
don't quite understand what you meant by username and password sending calls to
a H323 service provider, do you mean you have to register onto their gatekeeper?
Or otherwise, you should not need username and password.
Meantime, I am trying to setup up SIP
I appreciate all your feedbacks, but they seems to have diverted from my
original question which was
I have been using Asterisk 10 days ago version loaded onto my Redhat 7.3
with kernel 2.4.18-3 running Jeremy's h323 driver. It has been running okay
with a bit of problems, like system crashing
I agree that it should be able to do more than 15 to 20 calls when NOT
transcoding, however, I WAS doing pass-through without any transcoding and
it was crashing after around 15 to 20 calls, that was the problem, while I
was expecting at least hundreds of simultaneous calls ( not channels ) doing
if Asterisk/Digium performance/compatibility improves over the Intel
platform.
MATT---
-Original Message-
From: WipeOut [mailto:[EMAIL PROTECTED]
Sent: Monday, January 19, 2004 2:54 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] RE: Latest version of asterisk
T. Chan wrote:
Dear
Dear all,
I have had an experience which I would run by all of you to see if this is
normal.
I am running a few asterisk servers with 512M RAM memory, and as I have
mentioned in previous notes, I have experienced frequent crashes when faced
with more than 15-20 simultaneous calls. I have tried
Title: RE: [Asterisk-Users] capacity testing
Jesse,
Thanks
for your feedback.
1. I
am running kernel 2.4.18.3 with linux 7.3, please let me know which version of
Redhat are you running on and which kernel are you running, I wonder if that
could make a difference too. I am surprised that
Title: RE: [Asterisk-Users] capacity testing
Dear All,
I have been using Asterisk "10
days ago" version loaded onto my Redhat 7.3 with kernel 2.4.18-3 running
Jeremy's h323 driver. It has been running okay with a bit of problems, like
system crashing after certain period of time with 15
Dear All,
Based on your experience and knowledge, which Redhat (7.3, 8 or 9) and which
kernel is most stable and reliable running the 0.7.1 version of Asterisk?
Thanks
Tom
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Dear All
Should one enable HT in the chip when running Asterisk or if we don't, would
that offer alot less processing power?
Tom
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Dear All,
So are you saying that I should see 1 PID for safe_asterisk and many PIDs
for asterisk -vvvg -c or just 1 PID for asterisk -vvvg -c, the problem
is I am seeing alot of PIDs for asterisk -vvvg -c on a couple of my
systems and only 1 PID for asterisk -vvvg -c, which way is correct and how
Dear All,
So are you saying that I should see 1 PID for safe_asterisk and many PIDs
for asterisk -vvvg -c or just 1 PID for asterisk -vvvg -c, the problem
is I am seeing alot of PIDs for asterisk -vvvg -c on a couple of my
systems and only 1 PID for asterisk -vvvg -c on other couple, which way
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of
[EMAIL PROTECTED]
Sent: Friday, January 16, 2004 1:33 PM
To: [EMAIL PROTECTED]
Cc: Alan Chan
Subject: Re: [Asterisk-Users] RE: PID
On Thu, Jan 15, 2004 at 09:02:00PM -0500, T. Chan wrote:
Does anyone have any
Thanks alot for your explanation. Can you tell me if there is a way to
confirm if I have the nptl in the boxes ?
Thanks
Tom
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of James H.
Cloos Jr.
Sent: Friday, January 16, 2004 9:23 PM
To: T. Chan
Cc: [EMAIL
Hi,
I am a newbie in Asterisk as well, intending to use it in a similar way as
you are, communicating with AS5300 as well as other gateways including
MAXTNT.
I have had similar, but yet different experiences than yours.
1. Asterisk does crash with the number of calls, but in my case, about or
Hi, all !
I have a fast question, I am running a few Asterisk systems, but I just
noticed one thing quite peculiar. After I started safe_asterisk, and when
I ran PS or TOP, I could see 1 PID safe_asterisk and almost 10 PIDs
asterisk -vvvg -c even when there was no call. However, for the other
Title: RE: [Asterisk-Users] capacity testing
Hi
all, and Jesse
1. So,
you did get the experience of crashing all of a sudden with the "Disconnected
from Asterisk server" error message. I got both this and the segmentation error
when crashing. I am running the version of asterisk, libpri
Dear All,I have had
a problem that I have posted before, the asterisk kept crashingon me. I have
thought that may be before the problem is resolved, I couldtry to implement
a cronjob to run /usr/sbin/safe_asterisk, and if Asteriskis not running at
that time, it will start it
Dear All,
I have had a
problem that I have posted before, the asterisk kept crashing on me. I have
thought that may be before the problem is resolved, I could try to implement a
cronjob to run /usr/sbin/safe_asterisk, and if Asterisk is not running at that
time, it will start it
Dear All,
I am a new user of Asterisk interested in setting up a VOIP network based on
Asterisk. I have deployed a few Asterisk servers running on T400P and have
started a few weeks ago to run some LIVE traffic on one of the servers. Most
of my current traffic is via H323 to and from other
Dear All,
I am a new user of Asterisk
interested in setting up a VOIP network based on Asterisk. I have deployed a few
Asterisk servers running on T400P and have started a few weeks ago to run some
LIVE traffic on one of the servers. Most of my current traffic is via H323 to
and from other
Dear All,
I am going to deploy a VOIP
network here in Canada with nodes all over town. This is for long distance
services and hence would need a good reliable solution.
I have looked into * and am
very interested in it with all the value-added features as well as its
capability to do H323
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