No its stable just not as featureful as head. The tag is the same and you
can still check it out.
bkw
- Original Message -
From: Chris Foster [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, July 06, 2004 12:32 AM
Subject: Re: [Asterisk-Users] What happened to the CVS
ok. Thank U for a hint.
I have find out, the problem was with my ATA-186.
That box just use '#' not as sending key.
Does anyone know how to force ATA-186 to use '#'
as sending key.
Have tried *8 from softphone and that works fine.
On Mon, 2004-07-05 at 21:19, Brancaleoni Matteo wrote:
Hi
Il
Hello
I am currenly trying to setup Citel Link with
Asterisk. So far i don't have any luck in able to assign IP address to Citel
Link channel but still thought not a bad idea and get some one's view about
Citel channel bank.
Please let me know your experience while setting up
Citel and
On Tue, 6 Jul 2004 00:32:20 -0500, Chris Foster [EMAIL PROTECTED] wrote:
stable's gone because it wasn't too stable. The lastest CVS source is
alot more full featured and stable then the old stable branch.
I've found it the opposite.
I've tried CVS Head a few times because I wanted some of
On Mon, 05 Jul 2004 22:38:22 -0500, Daniel Jimenez [EMAIL PROTECTED] wrote:
Hall, Eric M. wrote:
I'm trying to see if this is even possible.
AFAIK Asterisk has no way of knowing if you do not answer. To Asterisk,
the call is complete and answered when it starts ringing. A PSTN/POTS
can this be accomplished?
Yes.
You should start reading documentation before asking. A good starting
place is http://www.voip-info.org
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On Tue, Jul 06, 2004 at 12:32:20AM -0500, Chris Foster wrote:
Date: Tue, 6 Jul 2004 00:32:20 -0500
From: Chris Foster [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] What happened to the CVS asterisk_stable branch?
On Mon, 5 Jul 2004 22:02:37 -0700 (PDT), every buddy
[EMAIL PROTECTED]
Brian K. West [EMAIL PROTECTED] wrote:
No its stable just not as featureful as head. The
What is important to me is the fact that I have the
same known release on multiple installations, whether
it's more stable or not.
I don't want to have a different release on every
machine I look after.
Hi,
I'd like to use Asterisk with ISDN interface and normal analog interface to
door phone (or any other low cost connection type to door phone).
What would be your recomendations for needed HW in Europe? Is it possible to
have this in one PCI card?
Are there any lower cost voip door phones?
Hi,
I have SE P800 cellular phone and I'm curious whether I could connect to it
over internet and not over analog interface and GSM network.
Are there any other cellulars that can do this ? Which ones ?
Thanks in advance,
Robert.
___
Asterisk-Users
On Tue, 6 Jul 2004 00:18:15 -0700 (PDT), every buddy
[EMAIL PROTECTED] wrote:
That's good news. Unfortunately I don't seem to have
any record anymore, Always looked it up on the web
site. Would you care to post here what the command was
again to get the stable branch from CVS? thanks.
It
http://www.voip-info.org/wiki-Asterisk+phone+door might be of some use.
-Shaun
On Tue, 6 Jul 2004 09:27:07 +0200, Robert Rozman [EMAIL PROTECTED] wrote:
Hi,
I'd like to use Asterisk with ISDN interface and normal analog interface to
door phone (or any other low cost connection type to door
It is. I did a cross upgrade with Pulver's firmware. I could not notice
any improvements, though... I still had that annoying hangup problem.
lenz wrote:
I have heard that the 2000W is the same exact harware as the
PulverInnovations WiSip phone - http://www.pulverinnovations.com/ - so
the
hi,
I have a problem with passing caller id information from telco (isdn) to
sip client (grandstream).
i see callerid in asterisk verbose console but on grandstream (sip)
phone is just internal (own-gs) 101 number.
Isdn line is connected with hfc card and p2p , asterisk is latest CVS in
I have all ready been there the only refference I saw was the tips and
tricks for asterisk and grandstream
is there some info I am missing?
thanks
hank
- Original Message -
From: Holger Schurig [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, July 06, 2004 12:06 AM
Subject: Re:
it looks (to me) like asterisk is not doing an SRV lookup when
REGISTERing with another sip proxy. is that correct?
what i am trying to achieve is to register [EMAIL PROTECTED] with a
proxy using
register = jasko:secret:[EMAIL PROTECTED]
my problem is that asterisk is doing a simple A RR lookup
Shaun Ewing [EMAIL PROTECTED] wrote:
It was:
cvs checkout -r v1-0_stable asterisk
thanks a lot.
__
Do you Yahoo!?
Yahoo! Mail - You care about security. So do we.
http://promotions.yahoo.com/new_mail
Jasminko Mulahusic wrote:
it looks (to me) like asterisk is not doing an SRV lookup when
REGISTERing with another sip proxy. is that correct?
what i am trying to achieve is to register [EMAIL PROTECTED] with a
proxy using
register = jasko:secret:[EMAIL PROTECTED]
my problem is that
I don't see anything on the Wiki or in the documentation about disabling
this feature.
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I have all ready been there the only refference I saw was the tips and
tricks for asterisk and grandstream
Do it the roman way: Divide and conquer. Divide your problems into 3
little problems:
a) connect a Grandstream to Asterisk
b) connect Asterisk to Grandstream
c) dialplan magic to connect
Easy, just don't include t or T in the dial string options.
-Shaun
On Tue, 06 Jul 2004 01:38:23 -0700, Dameon D. Welch-Abernathy
[EMAIL PROTECTED] wrote:
I don't see anything on the Wiki or in the documentation about disabling
this feature.
___
We're using the Quad-BRI card from Junghanns.NET with corresponding
drivers (bristuff 0.0.2).
The driver tries to patch asterisk libpri, which fails for current
version.
Anyone got an idea what'S the latest version of asterisk / libtri usable
with the Quad-BRI Card?
Thanks, Martin
I don't see anything on the Wiki or in the documentation about
disabling this feature.
What about the product documentation? Certainly your phone has some means
of configuration, e.g. by config files, built-in menus or a web-browser.
Use that and the documentation for it.
Maybe I'm wrong,
We're using a couple of h323 IP Phones (innovaphone ip200) w/ asterisk.
Basic call setup works, but we can't get call transfers to work:
On pressing the transfer button on the phone (getting a new dialtone)
the 2nd endpoint is disconnected. Any idea if we can get this to work?
Same reaction
(I posted this note on
http://www.voip-info.org/wiki-ZyXEL+P2000W+configuration too)
I tried to put together comments that were asked for on the P2000W.
These configs seem to work fine for a ZyXEL P2000W, thanks to Giles Scott
for getting me started with it.
DTMF keys work fine and are read
Hi Martin,
The bristuff distribution comes with a install.sh script (./install.sh)
which downloads, compiles the required software on your system.
If you want to do it manually, look into download.sh to see the exact
cvs checkout options which downloads the required asterisk and libpri
Martin Bene wrote:
Asterisk version is 0.7.2 release.
How about running a current (cvs -head) version of Asterisk?
Jeremy McNamara
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The bristuff distribution comes with a install.sh script
(./install.sh)
which downloads, compiles the required software on your system.
If you want to do it manually, look into download.sh to see the exact
cvs checkout options which downloads the required asterisk and libpri
versions.
This issued has been discussed few weeks ago into great depth.
Look into May/June 04 archive!
i have indeed looked into archives (wiki, googled using tabs, asked my
barber) and what have been discussed were SRV records for outgoing
calls.
the same seems not to work for REGISTER.
jasko
How about running a current (cvs -head) version of Asterisk?
Would love to and of course tried to: no go because of Junghans Quad-BRI
ISDN Card, no driver for cvs -head.
Bye, Martin
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Martin Bene wrote:
How about running a current (cvs -head) version of Asterisk?
Would love to and of course tried to: no go because of Junghans Quad-BRI
ISDN Card, no driver for cvs -head.
Then complain to Junghans.
Jeremy McNamara
___
Asterisk-Users
Jasminko Mulahusic wrote:
This issued has been discussed few weeks ago into great depth. Look
into May/June 04 archive!
i have indeed looked into archives (wiki, googled using tabs, asked my
barber) and what have been discussed were SRV records for outgoing
calls.
the same seems not to
Then wait for the next version, which will support both branches.
If you can't wait you can use the patch from someone who merged the
bristuff patch with a more recent version of cvs head...
This one:
http://capi4linux.thepenguin.de/download/asterisk/bri-stuff-0.0.2a-pp.tar.gz
Michael
Martin
Hello everybody,
my * box is connected to gnugk with H323 channel. If I call from an H323
EP to SIP EP (GS HandyTone or Xlite), when callee is picking up, audio
start but noisy (scratch) , then became ok for callee (SIP EP) but still
scratching on H323 EP. Now I stop/start asterisk, call from
Just wanted to say, that the problem was codec-related (on sipphone
connected to *)
Changed codec-settings and zaphfc is now running cool with cvs head.
- Still I don't know why I got the Protocol error (6) - but who cares?! :-)
NRB
- Original Message -
From: Klaus-Peter Junghanns
If you can't wait you can use the patch from someone who merged the
bristuff patch with a more recent version of cvs head...
This one:
http://capi4linux.thepenguin.de/download/asterisk/bri-stuff-0.
0.2a-pp.tar.gz
Thanks for that pointer, I'll give it a try.
Bye, martin
OH323 seems to work... Might be an alternative
Scott M. 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 administrator
tootai
Sent: Tuesday, July
I replaced my X100P cards with two TDM04B fully populated (8 FXO
modules). They are working fine, I can make and receive calls, but I
noticed all modules are always in GREEN state, even if I disconnect the
line. Both zttools and a cat /proc/zaptel/device shows no RED alarm.
Is
Yes, I know which libpri/asterisk versions bristuff downloads when
using the included scripts (03/24/04). Problem is, I'd like to get the
features / bugfixes from later versions.
In the wiki (and the mailing list archive) there's a document how I got
Asterisk CVS working with bri-stuff.
The
Thank you! That's what I was thinking but being new I wanted to ask .
Thanks again
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Daniel
Jimenez
Sent: Monday, July 05, 2004 11:38 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Calling an
On Tue, 2004-07-06 at 01:49, Wolfgang Pichler wrote:
Am Di, den 06.07.2004 schrieb Steven Critchfield um 0:54:
On Mon, 2004-07-05 at 06:51, Wolfgang Pichler wrote:
hi all,
Am Fr, den 02.07.2004 schrieb Steven Critchfield um 17:20:
Chill down a bit. We here to help.
sorry
Is it possible to have a speed dial on a cisco 7960 which dials the voice
mail number and then dials the extention and password so a user can
just push a single button to get their voicemail? This is a no brainer
on a regular analog phone where you can do something like 8500p100p1234
i have successfully updated my cvs pull of zaptel
but for asterisk when i type "make clean"i have the folowing error:
Makefile:73: *** missing separator.
Arrêt
( Arrêt means stop)
Lamine
Hi all,
Based on the website http://svn.asteriskdocs.org/ast_data, I am trying to
migrate my config file to PostgreSQL,but I am having problem calling the
other endpoint which I configured his account and extension on sql. I got an
error code 484. I would like to ask what should be the correct
if you 're not using linux you have to use gmake, not make
Jean-Yves
On 06/07/2004, at 9:22 PM, Mamadou Lamine KA wrote:
i have successfully updated my cvs pull of zaptel but for asterisk when i type make cleani have the folowing error:
Makefile:73: *** missing separator. Arrêt
( Arrêt
Hello
I've finally made the switch from our old PABX (NEC) to an Asterisk
based server.
I've configured zapatel to go into an incoming profile
it gets into:
extension = s,1,Dial(SIP/phone1SIP/phone2SIP/phone3... etc..
So when there's an incoming call, all phones rings then it goes into a
Hi Folks
I try the following within context:
exten = foo,foo,VoiceMailMain
After providung MailBoxNumber I get asked for PassWord.
If now the input fails I see on CLI
Playing 'vm-incorrect' followed by Playing 'vm-password'
and I can hear both messages.
Next try is:
exten =
Call comes in from remote SIP, authorised, does the following and dies
Any idea why..
I have ports 5060 and 16384 to 16482 open
Do I need any others?
What am I missing
Remote user is using X-lite for windows..
-- Executing Dial(SIP/2004-944c, SIP/2001|20) in new stack
On 06/07/2004, at 10:00 PM, Thomas Niesel wrote:
I try the following within context:
exten = foo,foo,VoiceMailMain
After providung MailBoxNumber I get asked for PassWord.
If now the input fails I see on CLI
Playing 'vm-incorrect' followed by Playing 'vm-password'
and I can hear both messages.
Next
Hi,
On 06/07/2004, at 9:52 PM, Jean-Yves Avenard wrote:
the grandstream 101 phone only shows: TR1 (or something like that it's
hard to tell with this display).
I had this problem for a while - the phone is actually displaying the
work asterisk in lower case - but it can't.
The GrandStream 101's
Ever since i updated from CVS head i get this error
when trying to play music on hold.
res_musiconhold.c:314 moh0_exec: Unable to
start music on hold (class '') on channel SIP/3001-f3b6
any ideas?
Best Regards
Stuart Baggs(Sales Manager)
Web: www.t-hosting.bizEmail: [EMAIL
Hello,
I think I have the same problem as Martin Bene mentioned in
http://lists.digium.com/pipermail/asterisk-users/2004-January/034521.html
Since I found no further information about this I'd like to ask wether
you know what the reason for this problem is and how one can get around
this.
* is
Hallo Jean-Yves Avenard
On Tue, 6 Jul 2004 22:08:39 +1000 you wrote:
On 06/07/2004, at 10:00 PM, Thomas Niesel wrote:
I try the following within context:
exten = foo,foo,VoiceMailMain
After providung MailBoxNumber I get asked for PassWord.
If now the input fails I see on CLI
Anyway to make hitting `0` during a voice mail dial an extension? The
bosses used to have that feature and love it.
Their VM prompt would say: Hello, My name is blah blah. I am currently
unavailable. If you would like to speak to an operator press 0 now,
otherwise leave me a message.
I have several machines operating nicely on P2P ISDN lines with
QuadBRI's, which uses the same layer2 code...
ZapHFC's seem to give a lot of trouble on certain hardware... try using
a different machine to host both cards in.
Kind regards,
Michael
Ernst Lehmann wrote:
Hello List,
is someone
Hi Torsten,
I think I have the same problem as Martin Bene mentioned in
http://lists.digium.com/pipermail/asterisk-users/2004-January/
034521.html
Since I found no further information about this I'd like to
ask wether you know what the reason for this problem is and how one
can
get around
Kevin Walsh noted that his SPA-2000 takes time from his local NTP server
in a post back on Fri June 25.
Q: Where do you tell it to use NTP?
I'm a bit confused as to where my SPA-2000 is currently getting its
time. I told it GMT-5 in the misc section but it doesn't really tell me
where its going
Holger Schurig wrote:
I don't see anything on the Wiki or in the documentation about
disabling this feature.
What about the product documentation? Certainly your phone has some
means
of configuration, e.g. by config files, built-in menus or a
web-browser.
Use that and the documentation
On Tue, 2004-07-06 at 00:59, Steven Critchfield wrote:
Well what is the trouble with moving that information up into variables
and using the new functions in IAX to pass that information from one
side to the other. Basically, you are going to pass it kind of out of
band, but it will get from
David Cook wrote:
Kevin Walsh noted that his SPA-2000 takes time from his local NTP
server in a post back on Fri June 25.
Interesting that it must have broadcast to the local net for a NTP server.
From a net admin perspective, I'd consider that a benefit.
Q: Where do you tell it to use NTP?
You will have to change signalling to something like a channelized T1 to
use a loopback, I think. The PRI has complementary protocols for CPE and
NET sides of the link. Not sure if a loopback would come up if it is
configured for CPE.
--
Steven Critchfield [EMAIL PROTECTED]
If I understand what
I have a Cisco ATA 186 working with h323, and G.723.1 codec, but when it
makes a connection to a PBX phone, connected to Asterisk by a Digium E100P,
don't use G.723.1 codec, the command oh323 show info indicates G.711 for
it.
Anyone got an idea if Asterisk translates G.723.1 to ISDN channel ?
Much thanks to those of you who are following this
thread. The information has been most helpful.
Here's an update for those who are interested. I
unplugged everything from every phone line, and tried
it all again, and it worked!
It worked for about 5 hours. Then, I started to get
phantom
On Tue, 2004-07-06 at 09:06, Scott Stingel wrote:
You will have to change signalling to something like a channelized T1 to
use a loopback, I think. The PRI has complementary protocols for CPE and
NET sides of the link. Not sure if a loopback would come up if it is
configured for CPE.
--
Hi Asha
Could you please setup a test account for me and mail me the details
thanks
Hari"Kanuri, Seshu" [EMAIL PROTECTED] wrote:
Folks!Netweb Group, Inc. fully supports connectivity to any Asterisk PBX systems you have and can provide A-Z termination with immediate effect.Any volume is good
On Tue, 2004-07-06 at 09:48, [EMAIL PROTECTED] wrote:
I have a Cisco ATA 186 working with h323, and G.723.1 codec, but when it
makes a connection to a PBX phone, connected to Asterisk by a Digium E100P,
don't use G.723.1 codec, the command oh323 show info indicates G.711 for
it.
Anyone got an
On Tue, 2004-07-06 at 11:29, Martin Bene wrote:
The bristuff distribution comes with a install.sh script
(./install.sh)
which downloads, compiles the required software on your system.
If you want to do it manually, look into download.sh to see the exact
cvs checkout options which
http://ip/admin/advanced, click on System tab, bottom two options
are primary/secondary NTP server. I'm running 2.0.9(d)
-Original Message-
From: David Cook [mailto:[EMAIL PROTECTED]
Sent: Tuesday, July 06, 2004 8:47 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] SPA-2000 and
David Cook wrote:
Kevin Walsh noted that his SPA-2000 takes time from his local NTP
server in a post back on Fri June 25.
Interesting that it must have broadcast to the local net for a NTP server.
From a net admin perspective, I'd consider that a benefit.
Q: Where do you tell it to
On Tue, 2004-07-06 at 01:51, Shaun Ewing wrote:
Easy, just don't include t or T in the dial string options.
I guess I was searching for the wrong question in the documentation:
disabling the transfer feature instead of enabling it. :)
I'm only interested in disabling the # when I *make* a call
On Tuesday 06 July 2004 17:19, Rich Adamson wrote:
It's not uncommon for vendors to embed the IP address of some known
time source in code. Use ethereal, reboot the box, and watch.
True , and unfortunately, this sometimes goes horrendously wrong...
http://www.cs.wisc.edu/~plonka/netgear-sntp/
Ok, here it goes:
I know CVS and I know how to program. I don't know much about linux
program installation. I have a WORKING asterisk based on CVS from
04/2004. It's running and, as of three days ago, it's in production as
well (production = my wife's using it without knowing it).
I want to
NTP is time-zone and season agnostic. It always transmits UTC.
Offsets from this are set in the client, including DST stuff. If they
can't be set, get a better NTP client. :)
Chris.
David Cook wrote (on Jul 06):
Kevin Walsh noted that his SPA-2000 takes time from his local NTP server
in a
Dameon D. Welch-Abernathy wrote:
On Tue, 2004-07-06 at 01:51, Shaun Ewing wrote:
Easy, just don't include t or T in the dial string options.
I guess I was searching for the wrong question in the documentation:
disabling the transfer feature instead of enabling it. :)
I'm only interested
Jay Milk wrote:
Of course I know that I should based my modification on the
latest-available code, but I'm a bit reluctant to upgrade my WORKING
asterisk to the latest CVS. Can I rename my asterisk-dir in /usr/src to
something different, then check out the latest CVS, make my changes, and
if it
On Tue, 2004-07-06 at 10:42, Jay Milk wrote:
I want to patch voicemail.c to allow for configurable pager-messages.
Looked at the code, and I know I can do that in 10 minutes. Once done,
I'm planning to make this patch available to the community, provided
the paperwork (release form etc) takes
Hi Jay,
Jay Milk wrote:
I want to patch voicemail.c to allow for configurable pager-messages.
Looked at the code, and I know I can do that in 10 minutes. Once done,
I'm planning to make this patch available to the community, provided
the paperwork (release form etc) takes less time than the
Bradley D. Thornton [EMAIL PROTECTED] wrote:
Your on notice as of now and you're being watched. Don't try to
destroy this community like the trail of destruction behind you!
Who died and made you king of the mail list?
From what I can see, Randy Bush asked a question about whether he
should
On Tue, 6 Jul 2004 07:51:53 -0700 (PDT), Shaun Dawson
[EMAIL PROTECTED] wrote:
On Monday morning, I got the same behaviour, with the
phantom phone calls. After much troubleshooting, I
finally, changed the card out of the machine (I have
two cards), made a call out with a regular phone to
It's correct that neither the SRV lookup is handled correctly or
completely, nore is there in standard distro a way to register with the
proxy for a domain, if those names differ.
It wasn't a difficult task to change this.
If there is interest I might release the patch for this as part of
On Tue, 6 Jul 2004 07:51:53 -0700 (PDT), Shaun Dawson
[EMAIL PROTECTED] wrote:
On Monday morning, I got the same behaviour, with the
phantom phone calls. After much troubleshooting, I
finally, changed the card out of the machine (I have
two cards), made a call out with a regular
Any thoughts on the following?
I am running asterisk from CVS (downloaded yesterday's
version, just to be sure) on a test system with no
digium cards in it, so I have installed ztdummy (see
logs and screenshots below) as a timing source.
When I call the music on hold extension from a Sipura
Sip
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Stuart Baggs wrote:
Can someone please tell me what sound files to record to get wakeup.agi
to work?
I'd recommend William Hung's version of She Bangs.
If that does not wake up up, nothing will.
--
Bob Knight
[-w] the work option
[EMAIL PROTECTED]
925-449-9163
Karl Brose [EMAIL PROTECTED] wrote:
There is also the option of expanding, or better redesigning, the [peer]
sections with proper and logical configuration options
and adding a register=yes flag.
I would prefer to see a register = yes directive in the type = peer
sections of both sip.conf and
On Tue, 6 Jul 2004, Kevin Walsh wrote:
People are entitled to ask questions; If no questions were asked then
this mail list would not have the volume of articles that it has.
Absolutly correct - except for Randy who has a tendancy of starting
arguments over irrelevant trivia.
My own concern
On Sat, 2004-07-03 at 19:12, Sam Tilders wrote:
On Sat, Jul 03, 2004 at 06:45:13PM -0600, Jared Mashburn wrote:
Is there any way for me to add myself to a call queue from outside of my
Asterisk Box?
For example,
I have a queue set up on my asterisk box, and I want to call it on my
Loopback should always make your status LEDs glow steady green. If that's not
working then you've got other problems.
It seems I may have those other problems you talked about. I made a
loopback cable and tested it on the channel bank. After about three
seconds all the status lights went
http://quantumvoice.com
Anybody using this
company. They have all you can eat toll free service. Don't see any reference to
asterisk, but can use your own Cisco or Sipura.
If there is any
known working config, appreciate if it could be posted here.
DH
Hi,
I found this site to import worldwide number ranges!
http://www.numberingplans.com/index.php?goto=isdnaction=analyses=44870
0688688
Does any one know other source(s), preferably free :)
Ta
Senad
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Kevin Walsh wrote:
Karl Brose [EMAIL PROTECTED] wrote:
There is also the option of expanding, or better redesigning, the
[peer] sections with proper and logical configuration options and
adding a register=yes flag.
I would prefer to see a register = yes directive in the type =
peer
I have no new information, just a note of encouragement to those
traversing the bowels of h323:
I've been trying to get h323 working with asterisk for several months
now, trying with chan_h323 chan_oh323 with all kinds of different
combinations.
As with several folk on the list, I've had no
Oh yeah, the -d option. That's what happens if you get pampered by CVS
shells all the time.
Is there a kind volunteer who'd like to take my updated voicemail.c and
perform the needed administrivia? Figuring out the patch-process and
disclosure forms is just something I'd rather not do with my
Make sure you answer the line first.
exten = 999,1,Answer
exten = 999,2,MusicOnHold(default)
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Jack Turer
Sent: Tuesday, July 06, 2004 11:43 AM
To: [EMAIL PROTECTED]
Subject:
And when can we expect a patch from you for this? :P
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Kevin Walsh
Sent: Tuesday, July 06, 2004 11:49 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] RE: is srv lookup being
Senad Jordanovic wrote:
Kevin Walsh wrote:
Karl Brose [EMAIL PROTECTED] wrote:
There is also the option of expanding, or better redesigning, the
[peer] sections with proper and logical configuration options and
adding a register=yes flag.
I would prefer to see a register = yes directive in the
Ive seen this question asked a few times but with no resolving answer. Im
running CVS-06/24/04-22:20:31, on RedHat Fedora 1, and cannot get chan_cssp
to compile. Im getting
Now compiling chan_sccp.c 695 lines
chan_sccp.c:50: warning: type defaults to `int' in declaration of
2.4 kernel? I have a RH 9 w/ 2.4 using ztdummy just fine a bit older though.
Message seems to show that the phones have trouble reaching each
other. Did Sip to Sip between the phones work fine?
On Tue, 6 Jul 2004 09:43:18 -0700 (PDT), Jack Turer
[EMAIL PROTECTED] wrote:
Any thoughts on the
I too had difficulty with chan_h323 driver. However, I used chan_oh323
driver and it worked in the second attempt. The trick is to use the
right version on pwlib and openh323 libs. The best way to ensure that is
to get them from the same site where you get the chan_oh323 driver.
Works like a charm
Hi guys, I am a newbie in asterisk system. And I wanna
to make some questions.
I already had a system to solve my VoIP solution, but
this system only accept the SIP protocol. Therefore I
thinking to using the asterisk like a middle to
redirect the H323 calls to my existing system!!!
I would like
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