Just some problems in the past with doubled digits in DTMF.
Tested only with firmware P0S3-04-4-00 in SIP mode.
Dan
P.S. When connected over a less reliable connection, the phone link to the
Asterisk server is lost, even in the same conditions an IAX soft phone works
without any problems. I think
Hi Gus,
I have all those problems too, but all gone when update to the latest
Asterisk CVS.
Now I can use unattended transfer on ATA with '#' or Flash.
Check the following settings in ATA (I presume that SIP is used):
CallFeatures: 0x0ff80ff8
ConnectMode:0x00460400
Tell me exactly how ha
CW_ASN wrote:
I use 3Party using flash key and dialing the extension. When the other ATA
answer the call, I press flash again.
I test Call Transfer using # key (#ext#). If you know another way to do
that, please let me know.
I'm tearing my hair out trying to exercise a variation on this theme.
I'm
Sorry off topic.
Does anyone try to include gnophone in knoppix cd-bootable dist.?
http://www.knopper.net/knoppix/index-en.html
Regards
---
Masakazu Nakano.
Dairiten.com - an open source VoIP and Ubiquitous Portal site in Japan.
http://www.dairiten.com/modules/news/
powered by xoops at http://w
First of all you should have callprogress=no and immediate=no for any kind
of a PRI. Also why is your d-channel going down ? Can you send a trace
"pri intense debug span 1" ?
regards
Martin
On Mon, 18 Aug 2003, Barry Porch wrote:
>
>
> I managed to get Asterisk working with my PBX using T1, now
Additionaly, the release location is:
Location: Private network serving the local user
But, the direction message is incoming... (see the < and >).
Regards,
Gus
- Original Message -
From: "CW_ASN" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, August 18, 2003 9:14 PM
Subject
Q931 DISCONNECT message maps the release cause in both directions. For
instance, the release cause is:
Cause: Temporary failure (41), class = Network Congestion (2)
Do you delete o replace part of SETUP message? Is very weird this part:
> ^M> Called Number (len=14) [ Ext: 1 TON: National Number
Make sure you've installed the latest kernel-source for your new kernel, and
that you reboot after updating the kernel and kernel-source.
You'll get weird error messages like this if the OS is running a different
kernel than you're trying to compile against.
Believe me, I've tried. :)
-wade
>
Hi,
I updated kernel for RH8.0 and updated * from cvs. After that zaptel compile
exits with error. Is it because of new kernel or zaptel source code change?
In any case could somebody help me to fix this problem?
Thanks,
Serge
/usr/src/linux-2.4/include/l
When trying to make outbound calls I am getting the Warning: File
app_dial.c line 313 (wait_for_answer) Unable to forward voice.
When making the call it attempts to dial (pounds are actually numbers
but replaced to not show numbers we are dialing):
Executing Dial("Sip/donas-bd7b", Zap/g1/1#
Hi guys,
About a month ago, I went from asterisk infancy to diving in to my first
production level install. I have a Verizon ISDN/PRI coming in to my
asterisk box, pretty standard configuration with AT&T analog phones. In
the almost 3-4 weeks of evaluating, asterisk seems to perform superbly
I will copy the configurations and let you know (may be some parameter is
wrong).
Thanks a lot!
Gus
- Original Message -
From: "Brian West" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, August 18, 2003 7:03 PM
Subject: Re: [Asterisk-Users] Call transfer ATA186
> Works for m
I use 3Party using flash key and dialing the extension. When the other ATA
answer the call, I press flash again.
I test Call Transfer using # key (#ext#). If you know another way to do
that, please let me know.
Regards,
Gus
- Original Message -
From: "Fredrik Hedberg" <[EMAIL PROTECTED]>
Well a feature like that would requrire some sort of auth so joe blow
employee doesn't go picking up the phone when the boss is talkin to his
mistress. :P But then again joe blow would be getting a raise shortly
there after!
bkw
On Mon, 18 Aug 2003, Grzegorz Nosek wrote:
> On 18 Aug 2003 15:07:
zapata.conf
signalling=pri_cpe ;90% if not, then pri_net
switchtype=euroisdn
channel => 1-15,17-31
Martin
On Mon, 18 Aug 2003, Nicolas Cartron wrote:
> Folks,
>
> everything's in the subject, i've got a Linux Box with a Digium E100P
> E1 Card, modules are loaded, but I don't know which signal
> Wouldn't that break everybody's dialplans where they would have to
> replace all occurrences of Voicemail2 with Voicemail and all
> occurrences of Voicemailmain2 with Voicemailmain?
No, we would register with both names.
Mark
___
Asterisk-Users maili
You did say 'punch down block' in your initial message and to a
telephony person that either means a 66 or 110 block.
Besides, if you're looking for a cheap PoE devise check this one out.
http://www.demarctech.com/products/reliawave-poe/poe-main.html
I use these on the wireless access points I bu
How exactly does you 3Party calling work? ;)
Fred
ASN wrote:
Hi all:
I'm testing a new installation of *, bringing up some ATA186. In *
environment, all stuff works greats. The only thing that don't work is
a Call Transfer, but the 3Party works ok. Some time ago I read that
somebody had pro
John Todd wrote:
On the Granstream 102 box that I have in front of me, there is a
"feature list" on the side. One of the features has grabbed my attention:
" - optional voice encryption (model 102D)"
Now, digging through Grandstream's site, I see that it's not offered
quite yet. However, send
On 18 Aug 2003 15:07:12 -0600, Jared Smith wrote
> On Mon, 2003-08-18 at 14:59, Brian West wrote:
> > Maybe we can pester kram to make that an option. monitor.conf anyone?
> >
> > bkw
>
> Well, while we're in the "let's pester Mark" mood... why not
> have him fix res_monitor so it writes to just
On Mon, 2003-08-18 at 15:45, John Todd wrote:
> Don't jump to that conclusion so quickly - there are reasons one
> might want multiple files.
>
> As an example, I have found it useful in at least one case to mix two
> call legs such that each leg is a different channel in a stereo final
> recor
On Mon, 2003-08-18 at 16:44, Nathan Littlepage wrote:
> Not only that. I'd hate to accidentally lay my had over that 66 block.
> DC is not very forgiving no matter what amps.
Who does network punchdowns on a 66 block. You do them on a patch panel
and they usually have nice plastic guides that keep
Works for me.. I can press # and dial the ext and press # to transfer a
call.
www.bkw.org/~brian/ata.html for the settings I used in my ATA
bkw
On Mon, 18 Aug 2003, ASN wrote:
> Hi all:
>
> I'm testing a new installation of *, bringing up some ATA186. In * environment, all
> stuff works greats
I agree with jtodd on that one it would make life simpler.. I don't
care if the files are seperate or not.. thats an easy solution to
overcome.
bkw
On Mon, 18 Aug 2003, John Todd wrote:
> >On Mon, 2003-08-18 at 14:59, Brian West wrote:
> >> Maybe we can pester kram to make that an option.
Hi all:
I'm testing a new installation of *, bringing up
some ATA186. In * environment, all stuff works greats. The only thing that
don't work is a Call Transfer, but the 3Party works ok. Some time ago I read
that somebody had proven this functionality successfully. If somebody knows what
It's up one directly. It just moved.
Run "make" in h323 then do "make install" on asterisk again.
Mark
On Mon, 18 Aug 2003, John Fortman wrote:
> What happened to chan_h323.c in the asterisk cvs? I got ast_h323.cpp, ast_h323.h
> and chan_h323.h but no chan_h323.c. Hence chan_h323.so was not
On Mon, 2003-08-18 at 14:59, Brian West wrote:
Maybe we can pester kram to make that an option. monitor.conf anyone?
bkw
Well, while we're in the "let's pester Mark" mood... why not have him
fix res_monitor so it writes to just one file! That would sure make me
a lot happier...
Jared Smith
Do
Not only that. I'd hate to accidentally lay my had over that 66 block.
DC is not very forgiving no matter what amps.
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Mike Ciholas
> Sent: Monday, August 18, 2003 4:36 PM
> To: [EMAIL PROTECTED]
> Su
On the Granstream 102 box that I have in front of me, there is a
"feature list" on the side. One of the features has grabbed my
attention:
" - optional voice encryption (model 102D)"
Now, digging through Grandstream's site, I see that it's not offered
quite yet. However, sending mail to thei
> From: Steven Critchfield <[EMAIL PROTECTED]>
> To: [EMAIL PROTECTED]
> Date: Mon, 18 Aug 2003 12:45:19 -0500
> Reply-To: [EMAIL PROTECTED]
>
> It is possible to make your own PoE adapter built into a punch
> down block. This is if you can find an appropriate 48volt power
> supply. I built an ad
Title: Message
Has anyone had any
major issues with the Cisco 7940 and or 7960
phones?
Just alias the commands.
On Mon, 18 Aug 2003, Tilghman Lesher wrote:
> On Monday 18 August 2003 04:06 pm, Mark Spencer wrote:
> > Does anybody have any reason why I should *not* permenantly replace
> > app_voicemail with app_voicemail2? If so, speak now or forever cvs
> > update -D "8/18/2003".
hahahaha while we are at it.. he has to fix a few issues. And since Mark
didn't write res_monitor i'm sure its going to be a task that will take a
little bit of time.
http://bugs.digium.com/bug_view_page.php?bug_id=120
bkw
On Mon, 18 Aug 2003, Jared Smith wrote:
> On Mon, 2003-08-18 at 1
Go for it! :)
On Mon, 18 Aug 2003, Mark Spencer wrote:
> Does anybody have any reason why I should *not* permenantly replace
> app_voicemail with app_voicemail2? If so, speak now or forever cvs update
> -D "8/18/2003".
>
> Mark
>
> ___
> Asterisk-Users
And break their voicemail.conf stuff as well.
On Mon, 2003-08-18 at 16:11, Tilghman Lesher wrote:
> On Monday 18 August 2003 04:06 pm, Mark Spencer wrote:
> > Does anybody have any reason why I should *not* permenantly replace
> > app_voicemail with app_voicemail2? If so, speak now or forever cvs
On Monday 18 August 2003 04:06 pm, Mark Spencer wrote:
> Does anybody have any reason why I should *not* permenantly replace
> app_voicemail with app_voicemail2? If so, speak now or forever cvs
> update -D "8/18/2003".
Wouldn't that break everybody's dialplans where they would have to
replace all
On Mon, 2003-08-18 at 14:59, Brian West wrote:
> Maybe we can pester kram to make that an option. monitor.conf anyone?
>
> bkw
Well, while we're in the "let's pester Mark" mood... why not have him
fix res_monitor so it writes to just one file! That would sure make me
a lot happier...
Jared Smi
Does anybody have any reason why I should *not* permenantly replace
app_voicemail with app_voicemail2? If so, speak now or forever cvs update
-D "8/18/2003".
Mark
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinf
Maybe we can pester kram to make that an option. monitor.conf anyone?
bkw
On Mon, 18 Aug 2003, John Todd wrote:
> >So how does one emit the legally required ( in some locales)
> >10 to 30 sec soft beep, letting people know they are being recorded ??
> >
> >very cool trick using the end point as
Has anyone used * and IAX in a gateway to a videoconferencing
application?
Best,
PauloHM
Hi all,
I noticed yesterday that MOH doesn't seem to work any more on my SIP
channels. It works fine on PSTN calls (chan_capi) but on SIP a just get a
tiny burst of sound followed by silence.
I know it was working a couple of weeks ago, and I haven't made any config
changes, but I have updated fro
So how does one emit the legally required ( in some locales)
10 to 30 sec soft beep, letting people know they are being recorded ??
very cool trick using the end point as the anchor for mixing
the sounds :)
:wq
[snip]
There is currently no way of which I am aware to insert audio on a
connected ch
Doesn't seem necessary at this time. Why not just record the caller
ID in Asterisk when the user dials the *57 "Customer Originated
Trace" CLASS feature? It's easy enough to store "last number that
called this number" stuff in the DB, and then act on it with a perl
script or something. If yo
Where is the 403 message coming from? Use tethereal or tcpdump to
find that out.Please include your sip.conf file (all of it,
except for passwords.)Use "sip debug" and include that output
during a call.
JT
Asterix PBX is loggin to Vocal and the extension number is also
loggin on th
John Todd wrote
Cisco has an 802.11 phone called the 7920, which is apparently
shipping now. It is very expensive (>$550 USD) and only runs SCCP at
the moment, which is Cisco's proprietary VoIP protocol. However, if
it falls in line with some of Cisco's other high-end VoIP equipment,
th
I managed to get Asterisk working with my
PBX using T1, now I am moving on to trying to make PRI work.
I have my zaptel.conf and zapata.conf
configured as follows:
Zaptel.conf:
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24
loadzone=us
defaultzone=us
Zapata.conf:
[cha
Hi George
You probably will need to run a local Gatekeeper which registers to the
outside gatekeepers. So Asterisk registers to your local Gatekeeper and
the Local Gatekeeper registers to Germany and UK Gatekeepers.
Not too many answers on this mailing list unless you have a non-related
qustion
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Hash: SHA1
On Monday 18 August 2003 19:47, Tilghman Lesher wrote:
> > That's what I found. I've attached a log of a call from init to
> > hangup. Note that I removed pri dchannel debug and hid phone and
> > ipnumbers.
> Looks like mysql_log() is not actually gett
What happened to chan_h323.c in the asterisk
cvs? I got ast_h323.cpp, ast_h323.h and chan_h323.h but no
chan_h323.c. Hence chan_h323.so was not created so no h323 support in
asterisk.
Just wondering when to expect it again because I
was stupid and didn't make a backup of the asterisk cod
Folks,
everything's in the subject, i've got a Linux Box with a Digium E100P
E1 Card, modules are loaded, but I don't know which signalling to put
in my zapata.conf...
Thanks for you help.
--
Nicolas Cartron
[EMAIL PROTECTED]
___
Asterisk-Users ma
Hi Eric,
> The ATA has two codecs that work with Asterisk. The G711 codec works,
> but each call tkes 64K of bandwidth + IP overhead. For most people this
> is only useful if the ATA and your Asterisk server is on the same LAN.
This is my case too.
Dan
___
On Monday 18 August 2003 12:33 pm, Tais M. Hansen wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> On Monday 18 August 2003 18:55, Tilghman Lesher wrote:
> > > There's nothing cdr_mysql related debug output.
> > > $ asterisk -vgcdf
> >
> > Turn on debugging in /etc/asterisk/logger.c
On Sun, 2003-08-17 at 12:44, Mike Ciholas wrote:
> Hi all,
>
> I'm looking for recommendations on ethernet switches for a new
> install. Ideally would want switches with at least 24 ports,
> ideally with a GE uplink, and that support PoE (power over
> ethernet) on every port. I've seen lots of s
On Monday 18 August 2003 12:24 pm, Tais M. Hansen wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> On Monday 18 August 2003 18:58, Thorsten Lockert wrote:
> > Did you actually turn on debug output in
> > /etc/asterisk/logger.conf? If not you won't see any debug output
> > anywhere.
>
>
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Hash: SHA1
On Monday 18 August 2003 18:55, Tilghman Lesher wrote:
> > There's nothing cdr_mysql related debug output.
> > $ asterisk -vgcdf
> Turn on debugging in /etc/asterisk/logger.conf
Yes. Forgot debug was put into a different file. Sorry.
Aug 18 19:1
Brian West wrote:
bkw
PS: The worse that can happen is it can't be used in court.
I though that tended to be the best that could happen :-)
Regards,
Steve
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/aster
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On Monday 18 August 2003 18:58, Thorsten Lockert wrote:
> Did you actually turn on debug output in /etc/asterisk/logger.conf? If not
> you won't see any debug output anywhere.
Ahh... Found it. I let Asterisk put the debug output in a seperate file a
Still looks like a context problem.
Can you post your extensions.conf and sip.conf files? (remove any passwords
of course!)
--
Some more details
When I am dailing an extension on Asterixsk PBX
Maybe it will help some how
Some more details
When I am dailing an extension on Asterixsk
PBX
Maybe it will help some how
66.178.36.15 -> 66.178.36.220
SIP/2.066.178.36.15 -> 66.178.36.220 SIP/2.0/UDP
66.178.36.15:5060;branch=2b0b0ed72ad04c5615dcab707e0fbe4a.466.178.36.15
-> 66.178.36.220 SIP/2.0
Did you actually turn on debug output in /etc/asterisk/logger.conf? If not
you won't see any debug output anywhere.
Also, what does your cdr_mysql.conf look like? Does it have a [global] just
before the configuration statements? Compare it to the (updated)
cdr_mysql.conf.sample that you got whe
On Mon, 2003-08-18 at 10:48, John Brown wrote:
> Can I use a x100p from digium to receive
> and originate calls to the PTSN ??
>
Yes.
Jared Smith
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http://lists.digium.com/mailman/listinfo/asterisk-users
On Monday 18 August 2003 11:34 am, Tais M. Hansen wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> On Monday 18 August 2003 18:19, Tilghman Lesher wrote:
> > On Monday 18 August 2003 11:08 am, Tais M. Hansen wrote:
> > > Is cdr_mysql broken in latest CVS? It builds and loads fine but
>
Hi
I am rephrasing my quastion.
If I have a Quicknet lineJack, can I use the hardware codecs provided by
lineJack. It would save a lot of CPU if I did not have to use its cycles
for RTP generation.
Regards
___
Asterisk-Users mailing list
[EMAIL PROT
ok, so this may show how dumb I am :)
Can I use a x100p from digium to receive
and originate calls to the PTSN ??
the data sheet talks about receiving calls
but does not mention that I can "MAKE" calls
using the card.
I want to take my * server and hook it up on
a single POTS line to test in an
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On Monday 18 August 2003 18:19, Low, Adam wrote:
> I'm not running the latest CVS release but found a couple of days ago that
> CDR's were not being inserted into my MySQL tables, I restarted Asterisk
> and it worked fine again ...
That's not an optio
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On Monday 18 August 2003 18:19, Tilghman Lesher wrote:
> On Monday 18 August 2003 11:08 am, Tais M. Hansen wrote:
> > Is cdr_mysql broken in latest CVS? It builds and loads fine but it
> > doesn't insert cdrs in the database and there's no debug output
I have tried that also.
Maybe there is something wrong with Vocal
??
I cannot also call from a softphone not registered
in vocal to [EMAIL PROTECTED].
But I can call with I register it with vocal, it
seems like vocal in not allowing anybody then registered users.
I used option "allow unregist
On Mon, 2003-08-18 at 04:13, Dan wrote:
> P.S. I think that for the moment, the cheaper option is to use ATA with some
> good and cheap DECT phones (in Europe) without any other feature than Caller
> ID (name and number). It can cost you less than 120EURO per port (about 75
> EURO for 1/2 ATA and
New Mexico is also a single party state :)
The financial service providers have a requirement for
beeps on the line.
Some states (don't remember which, but will research again)
require that you have a beep on the line. That may have
changed since the last time I really had to deal with this.
http://www.digium.com/index.php?menu=documentation
Try that, too.
- Devon
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Dayo Adeyeye
Sent: Sunday, August 17, 2003 5:26 PM
To: Asterisk
Subject: [Asterisk-Users] Asterix Newbie
Hello,
Just installed Aster
I didn't get any feedback on this, I guess its nobody else has come across the
requirement maybe ?
> -Original Message-
> From: Low, Adam [mailto:[EMAIL PROTECTED]
> Sent: 12 August 2003 12:29
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Malicious Call Trace
>
>
> All,
>
> Has
On Monday 18 August 2003 11:08 am, Tais M. Hansen wrote:
> Is cdr_mysql broken in latest CVS? It builds and loads fine but it
> doesn't insert cdrs in the database and there's no debug output at
> all.
No, it should be just fine (works for me!). Turn on debugging output
and post what you get.
-T
I'm not running the latest CVS release but found a couple of days ago that CDR's were
not being inserted into my MySQL tables, I restarted Asterisk and it worked fine again
...
> -Original Message-
> From: Tais M. Hansen [mailto:[EMAIL PROTECTED]
> Sent: 18 August 2003 18:09
> To: [EMA
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
Is cdr_mysql broken in latest CVS? It builds and loads fine but it doesn't
insert cdrs in the database and there's no debug output at all.
- --
Regards,
Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
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Depends. I don't recall any requirement for any beeps. Just a disclaimer
that their call may be recorded. In Oklahoma state you don't have to let
the other party know. Single party state. How great is that! :P
Or you can use a ghetto beep.. just press a button on the phone every now
and then!
On Monday 18 August 2003 02:55 am, Scott Stingel wrote:
> Yes, I have installed mysql and mysql-devel, and the mysql server
> is running on the system.
>
> My question is how to produce the cdr_mysql.so loadable file. I
> have followed the setup instructions in cdr_mysql.conf etc, and
> have rebuil
is the sip extension on the vocal sip server also
1234? if not, that could be why it's not working... when you're dialing
sip, you have to use the format:
exten => LOCEXT,1,Dial(SIP/[EMAIL PROTECTED]:port)
so it would be something like
exten => 1234,1,Dial(SIP/[EMAIL PROTECTED]:5060)
I too have seen this problem. I had MOH with SIP phones (Cisco 7960)
working, then some time later, when I tried, I would just get a burst of
music, then nothing. Then they started working again
Lee
- Original Message -
From: "Jamie Neil" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: M
So how does one emit the legally required ( in some locales)
10 to 30 sec soft beep, letting people know they are being recorded ??
very cool trick using the end point as the anchor for mixing
the sounds :)
:wq
On Sun, Aug 17, 2003 at 10:48:25PM -0700, John Todd wrote:
> [apologies for no line w
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On Monday 18 August 2003 16:29, Wade Weppler wrote:
> I had a very similar problem with chan_oh323. I suspect that it was my
> underpowered, overtaxed machine that was causing lost interrupts somewhere.
The system it's running on isn't doing anything
Asterix PBX is loggin to Vocal and the extension number is also loggin on
the same vocal server.
I cannot make it work :(
- Original Message -
From:
Josh Roberson
To: [EMAIL PROTECTED]
Sent: Monday, August 18, 2003 11:43
AM
Subject: Re: [Asterisk-Users] 40
I'm new too, but alot of my 403 forbidden messages
when adding extensions were due to context rules.. make sure that
the client dialing the extension is included in the same context your extension
is in.
just my thoughts on it, as it resolved a lot of 403
errors for me.
- Or
Most OSS drivers don't support full duplex. We've upgraded to ALSA, and
most problems disappear.
-wade
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of santiago
> Sent: Monday, August 18, 2003 10:36 AM
> To: [EMAIL PROTECTED]
> Subje
hi list,
when I run asterisk, appears the following:
WARNING[1074459808]: File chan_oss.c, Line 346 (setformat): Requested
8000 Hz, got 8178 Hz -- sound may be choppy
WARNING[1074459808]: File chan_oss.c, Line 974 (load_module): XXX I
don't work right with non-full duplex sound cards XXX
WAR
A couple of weeks ago I posted a message entitled 'Bridged trunks stuck off
hook' about a situation where 2 of my trunks (loopstart pots - but Centrex)
are occasionally bridged together. It has occurred to me that what may be
happening is that a line hung up by Asterisk might quickly be reused and
Hello,
I have a question.
I set up an extension to 1234
exten => 1234,1,Dial(SIP/[EMAIL PROTECTED]:5060)
And when I dial that extension I got in SIP message
"403 FORBIDDEN"
Can somebody tell me why I cannot call that
extension? When I am not using Asterisk I can call that extension witho
I had a very similar problem with chan_oh323. I suspect that it was my
underpowered, overtaxed machine that was causing lost interrupts somewhere.
At the risk of starting another flamewar, you might want to try chan_h323
instead. It fixed the audio problems for me.
Your mileage may vary.
-wade
On Mon, 18 Aug 2003, Adams, Gavin wrote:
> > "sudo su -" is kind of a stange thing to do. You would probably be
> > better of doing "sudo bash" as it also will give you a bash prompt
> > with root login.
>
> Good point on Linux/BSD boxen. My sudo 'training' days came from AIX and
> Solaris. :)
I
I use it without issues.
[agentlogin]
exten => 800,1,AddQueueMember(techsupport|SIP/${CALLERIDNUM})
exten => 800,2,Playback(agent-loginok)
exten => 800,3,Hangup
exten => 801,1,RemoveQueueMember(techsupport|SIP/${CALLERIDNUM})
exten => 801,2,Playback(agent-loggedoff)
exten => 801,3,Hangup
Our de
Quoting [EMAIL PROTECTED]:
> While you gugs are on the subject of MOH and SIP, exactly where in my
> configs do I turn on MOH for my SIP clients?
> Also while we're on the XLite would anyone like to help in getting my
> XLite client to work. I worked with several people on the irc
> channel the
>
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Hash: SHA1
Hi.
Using inAccess Networks chan_oh323, I'm experiencing some clicks or pops, how
can I fix that?
- --
Regards,
Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.2 (GNU/Linux)
iD8DBQ
While you gugs are on the subject of MOH and SIP, exactly where in my
configs do I turn on MOH for my SIP clients?
Also while we're on the XLite would anyone like to help in getting my
XLite client to work. I worked with several people on the irc channel the
other night and still can't seem to
> From: Steven Critchfield [mailto:[EMAIL PROTECTED]
> On Fri, 2003-08-15 at 12:42, Adams, Gavin wrote:
> > Another thing I'm doing while soak testing an application (pre
> > /etc/init.d startup script) is to run 'screen' as an unpriviledged
user,
> > then 'su -' to root (or even better, 'sudo su -
Quoting Stuart Hirst:
> Jamie,
>
> I have had this problem before when using X-Lite.
>
> If you are using X-Lite, there is a new version which seems to fix
> something's and break other ones but the current build as of yesterday
> is 1059,
Thanks Stuart, that seems to have done the trick. I'm sure
Hii .. i am trying to get quick net line jack and phone jack to communicate
with asterisk since last couple of days . i have searched the whole mailing
list as well as google . phone jack is working but i just cant figure out
how to communicate with Line jack and make PSTN calls . can anyone p
Jamie,
I have had this problem before when using X-Lite.
If you are using X-Lite, there is a new version which seems to fix
something's and break other ones but the current build as of yesterday
is 1059,
Rgds,
Stuart
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
--On Monday, August 18, 2003 10:31 pm +1200 Roger De Salis
<[EMAIL PROTECTED]> wrote:
Interesting menu options implying mechanisms to take the 11
channels of WiFI, and dedicate 1-3 for voice, and turn the
rest over to data. There were some rumours that they only
work on Cisco Aironet base stati
Hi,
chan_capi 0.2.4d fixes the problem.
regards
kapejod
--
Klaus-Peter Junghanns
CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705392
fax:+49 30 79705391
iaxtel: 1-700-157-8753
email: [EMAIL PROTECTED]
http://www.junghanns.net/asterisk
Am Son, 20
John Todd wrote
> Cisco has an 802.11 phone called the 7920, which is apparently
> shipping now. It is very expensive (>$550 USD) and only runs SCCP at
> the moment, which is Cisco's proprietary VoIP protocol. However, if
> it falls in line with some of Cisco's other high-end VoIP equipment,
Stuart Hirst wrote:
Does anyone know of a Java based SIP client and if so have has anyone
used it.
I found JAIN at https://sip-communicator.dev.java.net/ but have not
tried it yet.
The NIST JAIN implementation is quite mature, and the soft-phone demo
app that it used to ship with has now bee
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