Dear All,
Thank for your visit our site, I found some users can not read our home page
from some browser, I will move all the pgae to the top directory later.
I had some idea, do you agree?
I want to setup a voip provider group to share the local PSTN connection, every
member must provide at
Tommy,
Thanks, how do i get the older version of asterisk-addons?
--
Harold Workman
Quoting T. Chan [EMAIL PROTECTED]:
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
Have you ever looked into adding support for dialing directly from a browser?
i.e. a href=iax:[EMAIL PROTECTED]click here to call foo/a and IAX
Phone pops up and dials.
I think estara's SIP softphone supports this.
-Brian
On Sun, 27 Jun 2004 20:49:55 -0500, Steven M. Sokol
[EMAIL PROTECTED]
what site?
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, June 27, 2004 11:01 PM
Subject: [Asterisk-Users] New idea
Dear All,
Thank for your visit our site, I found some users can not read our home
page
from some browser, I will move all the pgae
Nicholas Bachmann [EMAIL PROTECTED] wrote:
Kevin Walsh wrote:
Dr. Rich Murphey [EMAIL PROTECTED] wrote:
How do you balance the number of active connections per server?
In theory, you could use a load balancer. That's as long as you can
share the SIP/IAX registrations between the
On Mon, 2004-06-28 at 02:02, Samantha (Femtech) wrote:
Is there a cron that I con do to replace this, as the fx0 card doesnt
hang up properly
I had the same problem here, and fixed within zapata.conf by adding
these lines:
busydetect=1
busycount=5
Try reading this also:
Scott Stingel a écrit :
Hi-
In answer to your questions:
Someone on Friday had said that disabling Fast Start corrected the audio
problem with H.323, so yesterday I tried to disable it in
~/asterisk/channels/h323/ast_h323.cpp. Today, I noticed that Jeremy
(NuFone) uploaded a new version of this
anyone has idea what problem can be here,
something with codec but i have today CVS version and grandstream phone
with 1.5.0 firmware.I try to change codec in phone and also in
asterisk-sip.conf but the same.
What can be problem ?
tnx,
Tomaz
*CLI -- Executing Dial(SIP/102-767c, CAPI/2:5)
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,
As you are in the UK I assume you are using the X101P like me. The best
you can do with this card is compile agressive echo cancelling on and not
have the tx gain too high. I hope that when the new FXO module is
available here the issue will go away.
Out of curiostity anychance you can list
Upgrade your firmware on the SPA-2000 and see if it fixes the one way
audio problem. I had this problem and worked with Sipura to get it
resolved. If you are running a firmware earlier then version 2.0.6(c)
then you will have this problem.
Matt
-Original Message-
From: [EMAIL PROTECTED]
Well, I have to confess that I am disappointed that in a fairly high volume
list like this, I haven't had one reply to the questions I've asked.
(I know I haven't got any right to expect a reply, but communities are usually
fairly helpful).
It might be really obvious to you guys, but if you
Samantha (Femtech) [EMAIL PROTECTED] wrote:
(Article auto-converted from unnecessary HTML to nice plain text.)
Is there a cron that I con do to replace this, as the fx0 card doesnt
hang up properly
phonegc:/home/samantha# asterisk -r
Asterisk CVS-05/30/03-17:17:07, Copyright (C)
Cheers Chris!
Any idea when the new FXO Module will be available?
My setup = Grandstream/ATA186 Asterisk FXO Chris Bond [EMAIL PROTECTED] wrote:
As you are in the UK I assume you are using the X101P like me. The bestyou can do with this card is compile agressive echo cancelling on and nothave
I have AGGRESSIVE_SUPPRESSOR uncommented in zconfig.h and txgain set to
4.0; Its a little quiet but usable.
I've stopped playing with the settings now cos I hope to get the new fxo
module very soon.
Chris
- Original Message -
From: Chris Bond [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Yes it is possible, with the chan_CAPI drivers from junghanns.net
i only used the 4BRI cards from Eicon but they are similar to the PRI cards
i didn't have any ISDN knowledge before. but first tried to install the card with CAPI
on a redhat 9 machine
with exactly the description from eicon
I believe its out if you call digium direct - im gonna
give them a call later see what the latest is.
From: taf taffey [mailto:[EMAIL PROTECTED]
Sent: 28 June 2004 10:31 AMTo:
[EMAIL PROTECTED]Subject: RE: [Asterisk-Users]
Re:Latest Echo changes
Cheers Chris!
Any idea when the new FXO
Yes but telappliant (the uk disti) have yet to get approval for it in the
UK. I've just fired of an e-mail to them as they said they should have it by
the end of the month. As you say though you can go direct ...
Chris
- Original Message -
From: Chris Bond [EMAIL PROTECTED]
To: [EMAIL
Hi again,
always latest CVS from 27/06/04. Calling to a SIP gateway I receive:
Unable to find a path from G723 to ALAW
Unable to find a path from ULAW to G723
Asked to transmit frame type 4, while native format is 1 (read/write = 8/4)
Unable to forward voice
[last messages repeated lot of times]
On Sun, 27 Jun 2004 17:25:56 +0100, Vassilis Konstantinou
[EMAIL PROTECTED] wrote:
Thanks for the reply Greg,
The definition for the console is
[globals]
;CONSOLE=Console/dsp; Console interface for demo
CONSOLE=Zap/1
so if I am mistaken I have commented
Does anyone know if SetGroup and CheckGroup apply to only current
context or is it per server based?
Ta
SJ
___
Asterisk-Users mailing list
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To UNSUBSCRIBE or update options
Use the 0.6.2a version.
Michael.
Brian Wilkins wrote:
Hi,
I having a problem compiling the wrapper for oh323. I am running Debian,
kernel version 2.4.18-bf2.4. The pwlib version I have is 1.6.6 and the
openh323 version I have is 1.13.5. I execute the following commands first
before
Hei,
please never try to dial out on a particular b channel, you have to dial
out on a zaptel group which includes both b channels of the BRI line.
In a p2mp setup YOU cannot know which b channel will be chosen!
exten = _X.,1,Dial(ZAP/g1/${EXTEN})
will do(note the 'g')
best regards
Basically, have an old IVR application running under Apex's Omnivox software on a box
with 4 old intel dialogic D240SCs, and would like to allow remote clients to gain
access to aforementioned IVR application via softphone, 7960, ata, etc via asterisk
with a TE410P.
Unfortunately, all I know
Tommy,
Still waiting from you whether the CDRs are recorded with cdr_csv.
This is working just fine for me.
Michael.
T. Chan wrote:
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?
I have (as I have mentioned before) 2 identical servers connected to to
same cisco gatekeeper.
Server 1 works fine with no audio problems, server 2 is using cvs head
and there is no audio when connected.
using same configs on both servers (RH9). Disabling faststart didn't
help me.
I have spent
Hi people,
I have a user that forgets to hangup his conference calls, so they go on
forever. Is there a way of limiting the duration of a conf call?
Thanks in advance,
Pablo
--
Pablo Endres [EMAIL PROTECTED]
ComVoz Communications
USA: +1 954 343-2085 Ext 199
Venezuela: +58 212 7713195
Pablo Endres wrote:
Hi people,
I have a user that forgets to hangup his conference calls, so they go
on forever. Is there a way of limiting the duration of a conf call?
Thanks in advance,
Pablo
Try using ABSOLUTETIMEOUT before starting the conference?
On Mon, 2004-06-28 at 05:11, administrator tootai wrote:
Unable to find a path from G723 to ALAW
Unable to find a path from ULAW to G723
Asked to transmit frame type 4, while native format is 1 (read/write = 8/4)
Unable to forward voice
Even if I force my sip.conf to use only g723.1 I have
When I call a PBX system and enter digits, Asterisk is eating
away some digits. For example when I call ATT and when the
system prompts me to enter my phone number, Asterisk eats
away some digits, so ATT does not get the number that I
entered. I am using the extensions.conf as it
Hi - And thanks for the answer!
Unfortunately i get the exact same result with the g1 instead on just 1.
Kind Regards
NRB
- Original Message -
From: Klaus-Peter Junghanns [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, June 28, 2004 1:56 PM
Subject: Re: [Asterisk-Users]
Hello,
I am having some trouble with the Asterisk::AGI perl library. It seems
that the AGI-Exec() command is causing me a problem.
Here's the line in my AGI code: $AGI-exec('Record',$vmfile:wav, 30);
I'm trying to record voicemail to the file name stored in $vmfile with
a silence timeout of 30.
Eric Wieling a écrit :
On Mon, 2004-06-28 at 05:11, administrator tootai wrote:
Unable to find a path from G723 to ALAW
Unable to find a path from ULAW to G723
Asked to transmit frame type 4, while native format is 1 (read/write = 8/4)
Unable to forward voice
Even if I force my sip.conf
Hello all,
Bugfix release 0.6.3 is now available. Basically, call indications
should work ok now. Also, the OH323 channel variables for incoming calls
are set properly (they can be used for special authentication purposes).
Download:
http://www.inaccessnetworks.com/projects/asterisk-oh323
Regards,
$AGI-exec('Record',$vmfile:wav 30);
- Original Message -
From: Tom Daly [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, June 28, 2004 8:05 PM
Subject: [Asterisk-Users] AGI-Exec Problem
Hello,
I am having some trouble with the Asterisk::AGI perl library. It seems
that the
On Mon, 2004-06-28 at 09:36, administrator tootai wrote:
Third party only accept g723 or g729. No solution (or buy a g729
license)? What's the reason to not convert to g723?
Because the patent holders of the G723.1 patents do not want to license
their technology for a reasonable fee.
Here
Ok so here's one i have already asked but i don't know if anyone saw it
Has anyone managed to get the 'i' extension to work.
I have included within each context the following
exten = i,1,Goto(wrong-number,s,1)
then in
[wrong-number]
exten = s,1,GotoIf($[${EXTEN:0:2} = 43}]?10:2)
exten =
Sorry this has nothing to do with your audio issue, but I noticed you were
able to build oh323 1.13.5 and pwlib 1.6.6 sucessfully with asterisk-oh323
0.6.2. I get the following errors when trying to compile the oh323 wrapper
for asterisk:
-- snippet of errors --
In file included from
On Mon, 2004-06-28 at 09:55, Simon wrote:
Ok so here's one i have already asked but i don't know if anyone saw it
Has anyone managed to get the 'i' extension to work.
I have included within each context the following
exten = i,1,Goto(wrong-number,s,1)
then in
[wrong-number]
exten =
Michael:
I tried that version also and got the following errors. I just upgraded
to 0.6.3 version and it gave me the exact same errors. Any clues? PWLib and
Openh323 build just fine, maybe path got b0rked ? Thanks.
This is just a snippet of the hundred of errors that I got:
-- snip --
Has anyone seen this problem before?
I have a server with a single X100P card. The audio level is a low, but if I
raise the gain to more than -2db (Rx + Tx) it starts to oscillate in an echo
test. Not at a high frequency but with a noise that is best described as a
steam engine starting up. It
Hi Brian-
I think you have to use 0.6.2a not 0.6.2. Also, you might try the new
version from today: 0.6.3.
And just checking, in your Makefile, that you set ASTERISKSRCDIR =
/usr/src/asterisk. (maybe this is a 0.6.2a thing)
Regards
Scott
Scott M. Stingel
President,
Emerging Voice
Did you apply to the OpenH323 the included patch BEFORE configuring the
library (openH323)?
Also, try to use the latest version (0.6.3) if you are running current
Asterisk CVS code.
Michael.
Brian Wilkins wrote:
Sorry this has nothing to do with your audio issue, but I noticed you were
able to
Sorry, Tom, I missed this message when it came through. It seems this
problem is a continuing issue among the asterisk folk.
Tell me, what versions of IOS have you tested with, do you have any of
the h323 options enable/disabled in the 5300?
-g
On Fri, 2004-06-18 at 21:09, T. Chan wrote:
Just spoke to someone at telappliant and there not willing to sell the cards
in the uk yet as there not ratified to the UK standard.
I've just spoke to someone at digium direct and there forfilling backorders
at the moment. I've just placed an order at
We use an IAX2 trunk to our remote office and would like for the
receptionist to be able to transfer incoming calls from this trunk. but
all calls come in as one user, Is there a way to get a breakout on the
flash GUI of the incoming calls?
Thanks,
Justin
Todd at Teledynamics (see wiki page mentioned above) has been very responsive to
email, and we did not need to sign up as a reseller to purchase the Uniden phones.
Great!! I'll give him a call today and see if I can order one...this
looks like a really nice phone for the price and given the
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steven
Critchfield
Sent: 28 June 2004 16:22
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] (no subject)
On Mon, 2004-06-28 at 09:55, Simon wrote:
Ok so here's one i have already asked but i don't know
have just updated the sources from cvs
when i compile zaptel i get following error
can help me ?
nicolas
snip
zaptel.c: In function `zt_ctl_ioctl':
zaptel.c:3042: warning: assignment from incompatible pointer type
zaptel.c:3044: warning: assignment from incompatible pointer type
Michael:
Yes I did.
On Yaum al-Ithnain 10 Jumaada al-Awal 1425 11:28 am, Michael Manousos wrote:
Did you apply to the OpenH323 the included patch BEFORE configuring the
library (openH323)?
Also, try to use the latest version (0.6.3) if you are running current
Asterisk CVS code.
On Monday 28 June 2004 15:56, Chris Hirsch wrote:
Todd at Teledynamics (see wiki page mentioned above) has been very
responsive to email, and we did not need to sign up as a reseller to
purchase the Uniden phones.
Great!! I'll give him a call today and see if I can order one...this
looks
Tommy,
I reverted asterisk-addons to 04/01/2004 and I was able to compile it with the
latest asterisk CVS. Your a lifesaver. Ive been pondering over this problem
for over a week now. Thanks!
--
Harold Workman
CCNA, CCNP
Cytel Communications
[EMAIL PROTECTED]
Ph. 281-449-4000 x3098
Ok,
I got it all to work finally. I removed everything and started from
scratch. I also got the latest version of asterisk from the CVS. I built
PWLib, then applied the patch to oh323 1.13.5 then built oh323, and finally
built and installed the wrapper (0.6.3). I just started up Asterisk
Hi all,
* was running ... I have a WT405P and an AVM C4 with chan_capi 0.3.4a
Today chan_capi stopped working, without any changings at the system.
It seems, that not * is the reason, because isdn-log also shows no calls.
If I try to call * from outside via capi, I only get a busy.
That is
All,
I have been thru the archives and all the relevant URLs
sent to me. I have sent e-mail to those who have gone before me and are
attempting to accomplish the same goal no one has it working?. Doesnt
anyone have a WORKING asterisk pbx that hooks into vonage?
Thanks,
Jerry Roy
I've been doing some debugging on this and I think it's a code problem.
I'm by no means an expert on Asterisk or how it is written or
implemented, but the following patch to app_voicemail.c fixes the
issue. With this code change, Asterisk correctly transfers to the 'a'
and 'o' extensions as
On Mon, 28 Jun 2004, Tom Daly wrote:
Hello,
I am having some trouble with the Asterisk::AGI perl library. It seems
that the AGI-Exec() command is causing me a problem.
Here's the line in my AGI code: $AGI-exec('Record',$vmfile:wav, 30);
The proper usage would be:
$AGI-exec('Record',
Jeff Workman wrote:
O
--On Wednesday, June 23, 2004 4:26 PM -0400 Roger Gulbranson
[EMAIL PROTECTED] wrote:
On Wed, 2004-06-23 at 15:39, Jeff Workman wrote:
We are developing an on-demand teleconferencing solution. We will
be billing per-minute/per-user.
I've successfully gotten
On Mon, 2004-06-28 at 12:57, Jeff Workman wrote:
O
--On Wednesday, June 23, 2004 4:26 PM -0400 Roger Gulbranson
[EMAIL PROTECTED] wrote:
On Wed, 2004-06-23 at 15:39, Jeff Workman wrote:
We are developing an on-demand teleconferencing solution. We will be
billing per-minute/per-user.
I'm going to modify the queue announcements to allow
for rounded seconds (e.g. we want to know to the tens of seconds. E.g.
Average wait 1 minute 20 seconds).
I'm going to add the optional announce of seconds to
the queue config and a rounding factor (e.g. 10 in our case).
The
Roger Gulbranson wrote:
On Mon, 2004-06-28 at 12:57, Jeff Workman wrote:
O
--On Wednesday, June 23, 2004 4:26 PM -0400 Roger Gulbranson
[EMAIL PROTECTED] wrote:
On Wed, 2004-06-23 at 15:39, Jeff Workman wrote:
We are developing an on-demand teleconferencing solution. We will
be
Title: Message
I am trying to
implement a rollover of extensions.
exten = 3000,1,GotoIf($[${line1} =
Congestion]?3:2)exten = 3000,2,Dial(${line1},15,rt)exten =
3000,3,GotoIf($[${line2} = Congestion]?5:4)exten =
3000,4,Dial(${line2},15,rt)exten = 3000,5,GotoIf($[${line3} =
Thank you for the prompt reply but when I add 7;8;9, in my button number
field the iax2 button goes away. i just got .10 today
.
On Mon, 2004-06-28 at 11:51, Nicolas Gudino wrote:
Hi Justin,
Justin Carlson wrote:
We use an IAX2 trunk to our remote office and would like for the
Michael Manousos wrote:
The performance of the oh323 channel driver is limited by OpenH323.
asterisk-oh323 uses the (more complete) RTP implementation offered by
the library, and not that of Asterisk. Of course there are pros
(adaptive jitter buffer, RTCP implementation) and cons (lower
On Mon, 2004-06-28 at 16:02, Justin Carlson wrote:
Thank you for the prompt reply but when I add 7;8;9, in my button number
field the iax2 button goes away. i just got .10 today
.
That feature will be available in 0.11, is not complete yet (I'm working
on it). Please subscribe to the
Hi all,
On my 7940 phone when I dial out I press 9, then the number. After I
press the second number (IE: 9,1) the dialtone stops playing just like
it should. This is normal and similar to a regular phone.
On two of my 7940s the phones continue the dialtone. No matter how many
numbers you dial
On Mon, 2004-06-28 at 07:45, Michael Manousos wrote:
Hello all,
Bugfix release 0.6.3 is now available. Basically, call indications
should work ok now. Also, the OH323 channel variables for incoming calls
are set properly (they can be used for special authentication purposes).
Download:
This was fixed in cvs HEAD and stable on 4/13/2004 and a new source
release was made at the time (version 0.9.0)
I'm not sure why it would be brought up on a recent newsletter, it was
discussed in here (or maybe on -dev) sometime around 4/15/2004
James
On Mon, 28 Jun 2004, Jim Rosenberg wrote:
Hi List!
I use Asterisk as a pure voicemailbox at a customers place. Right now,
a telephone uses up two msns, one for the telephone itself, and one for
the telephones mailbox. If the user is absent, a telephonecall is
redirected to the voicemail msn of that users telephone.
The Problem is: The
I ran a PRI DEBUG SPAN 1 on our office system.
I could not see any FACILITIES messages on outgoing calls over the PRI.
So I suppose * does not send the CNAME messages at all on outgoing calls.
CLID NAME is just a subset of the generic user to user messaging on ISDN
networks.
It should be
Hello,
I'm in the process of trying to get Festival to work with Asterisk. I
followed the install process at
http://www.voip-info.org/wiki-Asterisk+festival+installation. To get the
Festival to compile I had to add the patch described in the comments.
Once added, Festival and the Speech
Hi,
What is the best SIP softphone to use with Asterisk?
I have a hard time finding OpenSource SIP soft phone.
Regards
Arve5
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To UNSUBSCRIBE or
I wonder if your issue and mine are related somehow.
I have a asterisk server with 4 FXO cards in it, and when a call comes
in one ZAP channel, then dials out another, I hear what could be
described as a steam engine starting up. It starts off kinda slower/
quiet, then quickly (in about 2-4
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Arve,
I've been using kphone (http://www.wirlab.net/kphone/) with success.
It's simple and works fine :-)
[]'s
Arve Rasmussen wrote:
| Hi,
|
| What is the best SIP softphone to use with Asterisk?
|
| I have a hard time finding OpenSource SIP soft
Try recompiling your zaptel package without the aggressive echo
cancellation enabled. I have aggressive cancellation help before, I
but I have also seen it hurt things before.
Brian
On Mon, 28 Jun 2004 16:26:32 +0100, Whisker, Peter
[EMAIL PROTECTED] wrote:
I have tried the latest CVS Head
I am also having the same problem. Latest CVS Latest Capi
When it does work and you pick up the phone, CAPI disconnects the call.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of ePyron Felix
Deierlein
Sent: 28 June 2004 18:34
To: [EMAIL PROTECTED]
MessageIf I am understanding your dialplan snippet correctly, you simply
want * to call extensions in a linear (or even round robin) fashion, ringing
the first one that's not busy correct? This functionality is built directly
into * and needs no special dialplan to implement. Please check the Wiki
Same here :-(
asterisk show's this error in the same moment i'm trying to pick up an
incoming call:
Jun 23 13:14:03 ERROR[-1284076624]: chan_capi.c:881 capi_write: dont know
how to write subclass 64
This problem starts with cvs update -D 6/21/04 21:00:00 CET
If i revert back to cvs update -D
Although nat=yes/qualify=yes can handle some NAT routers, it does not handle
all situation in both directions. Our experience suggests that nothing short
of a full SIP Outbound Proxy is going to handle things properly.
We have tried out ABP International's NATpass and SNOM's NATfilter, both
with
Murray Hooper wrote:
I am trying to work with zap and libpri to do some ISDN circuit testing with
Digium T100P. I am trying pritest but can't figure out what dchannel
number should be be when I try 24 or 1, I get failed to open dchannel
'24'. D-channel is on time slot 24 on our circuit, but
Hi,
We are looking at deploying Asterisk for about 60 phones. Since we are
in a public building, and are a mixed university and federal unit, we
must have our phones up near 100% of the time. Currently we have ~60
POTs lines. I am working on moving us to DIDs with a single PRI feeding
us. The
Title: Modems behind Asterisk - how?
The configuration I'm building replaces an existing PBX with Asterisk. There are 8 existing modems that people use to call in from the outside to connect to PC(s) on the inside to transfer data, etc. Callers access these modems by calling the main
Thanks I will give that a try.
Looks like this may need a bug report? We are all getting the same
errors.
Outgoing is fine for me.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andreas
Anderson
Sent: 28 June 2004 23:26
To: [EMAIL PROTECTED]
Subject:
On Mon, 2004-06-28 at 16:16, Rodrigo P. Telles wrote:
I've been using kphone (http://www.wirlab.net/kphone/) with success.
It's simple and works fine :-)
kphone only supports inband DTMF and so will only support DTMF when
using ulaw or alaw.
--
Eric Wieling * BTEL Consulting *
I tested SpanDSP as an internal extension, and it worked like a charm.
Now I am trying to receive faxes from a toll-free nufone DID. I am
running g.711uLaw in on this line, so no to cause too many problems.
However I receive the following errors after the fax is finished
receiving:
so the fax
Webpage still doesn't work.
On Mon, 2004-06-28 at 22:22, [EMAIL PROTECTED] wrote:
Dear All,
The home page already move to the top, you can try again.
Cary LEUNG
Network Operator
Hong Kong VOIP exchange Network
Glynn Condez [EMAIL PROTECTED]:
What happened to your website. I am
I'm planning to buy Dialogic licenses for one of my dialogic boards to use
with *. I have already that in the drawer and it's boring me to keep it
there with no use.
Although, I have heard that it doesn't work for dialout and I would like
to confirm if it's true... my plan is the following:
On Mon, 2004-06-28 at 14:18, mattf wrote:
In my experience HT on with SMP kernel does help. Others have stated on this
Thanks. I have had good experiences with RH ES and Core 1 and HT.
james
signature.asc
Description: This is a digitally signed message part
--On Monday, June 28, 2004 7:21 PM +0200 Michael Sandee [EMAIL PROTECTED]
wrote:
Other than that... if these problems are not being published when
fixed... then other distro's do not have a chance to fix it... (think
about distro's that use stable code, but haven't updated to 0.9 because
of
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I've only been watching this list for the past 2 days.
And it seems to be an one way street:
- -Tell about your problems and what you would like to do.
Usually no answer.
I have to admit I'm rather disappointed with Asterisk, information is
probably
On Mon, 28 Jun 2004, Jim Rosenberg wrote:
I have to say -- with somewhat less vehemence -- that I'm another user who
sure never noticed that the stable release of Asterisk had moved from
0.7.2 to 0.9x. This should have been an important announcement on *SEVERAL*
security grounds. As of
Hello:
Does anyone know how to configure any of the Cisco 79XX phones to get
custom ringers when using chan_sccp with Asterisk?
I've currently got Asterisk's 05-24-04 CVS-HEAD and Zozo's 0.2 release
of chan_sccp.
I've tried using ringlist.dat, but that appears to only be for the SIP
phones...
On Mon, 2004-06-28 at 20:08, Jean-Yves Avenard wrote:
I've only been watching this list for the past 2 days.
And it seems to be an one way street:
- -Tell about your problems and what you would like to do.
Usually no answer.
You will find many of us will ignore messages if they require us
--On Monday, June 28, 2004 9:16 PM -0400 James Golovich [EMAIL PROTECTED]
wrote:
It was fixed in CVS head and stable and at the same time 0.9.0 was
released. The existance was noted in the ChangeLog as well that comes
with asterisk
Good. But the OpenH323 patches were not back-patched for
The phone will TFTP the file RINGLIST.XML which wants to look something
like:
CiscoIPPhoneRingList
Ring
DisplayNameRing ring/DisplayName
FileNameringring.raw/FileName
/Ring
...
the raw files being in a format described on the Cisco site in some
On Mon, 2004-06-28 at 20:08, Jean-Yves Avenard wrote:
I've only been watching this list for the past 2 days.
And it seems to be an one way street:
- -Tell about your problems and what you would like to do.
Usually no answer.
Personally I've gotten tired of answering questions over and
On Mon, 2004-06-28 at 11:18, Ryan Courtnage wrote:
FYI - recent changes in chan_sip (RFC3581 support) will cause the UIP200 to
stop functioning properly.
Uniden has no current plans to support this RFC. We are currently working
with them to determine if they will make the phones at least
On Tuesday 29 June 2004 01:57, Eric Wieling wrote:
On Mon, 2004-06-28 at 11:18, Ryan Courtnage wrote:
FYI - recent changes in chan_sip (RFC3581 support) will cause the UIP200
to stop functioning properly.
Uniden has no current plans to support this RFC. We are currently
working with
Hi,
I'm testing a Polycom IP600.
With firmware version 1.1 the phone reboots at any time.
With firmware version 1.2, the first reboot was an endless reboot. Then
I moved the phone to another lan port, then it worked fine. Then I
installed again in the initial lan port and the phone works well.
Kevin Walsh wrote:
Nicholas Bachmann [EMAIL PROTECTED] wrote:
Kevin Walsh wrote:
Dr. Rich Murphey [EMAIL PROTECTED] wrote:
How do you balance the number of active connections per server?
In theory, you could use a load balancer. That's as long as you can
share the SIP/IAX
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