Can anyone say for certain whether the polycom ip 500 / 600 work with * ?
>From what I've seen from googling, it appears that they would.
Network Computing seemed to think highly of them -
http://www.nwc.com/shared/printArticle.jhtml?article=/1416/1416f2full.html&pub=nwc
- Chris Clifton
I'm running Asterisk 0.5.0 and using Cisco 7960 phones in a "sip only"
configuration currently. Everything is working except that caller ID is hosed.
Say for example extension 100 calls extension 200. 200 sees "100" as the
name but "200" as the number. IE, it gets its own number as the suppos
I've posted this text before, but...
"SoundPoint® IP supports shared call appearances (SCA) using the
SUBSCRIBE-NOTIFY
method in the "SIP Specific Event Notification" framework (RFC 3265)."
This from the admin guide at
http://www.polycom.com/common/pw_item_show_doc/0,1276,2545,00.pdf
Would love
The next thing is getting the polycom to work.. but yes I think its the
publish and subscribe stuff that will do this. Asterisk doesn't support
that and neither does the 7960.
bkw
On Fri, 6 Feb 2004, Chris Clifton wrote:
> As a follow up, looks like the polycom ip phones support this via their
What happens when you use a service like voiceglo on * with the
unlimited plan? Can you make multiple calls at the same time?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Bittner
Sent: Friday, February 06, 2004 8:15 PM
To: [EMAIL PROTECTED]
Subject
I was able to resolve this problem, after removing and adding back the
port settings in the firewall. I changed hardware and IP's. So I can
only guess that arp table was messed up. I'm sure rebooting the
firewall would have given me the same result. But everything has been
working fine since th
As a follow up, looks like the polycom ip phones support this via their
'buddy watch' presence feature. Anyone else used this on recent polycom
soundpoint ip 500 or 600 phones with * ?
Chris Clifton
- Original Message -
From: "Brian West" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent:
Are there provisions in the sip protocol to allow for this functionality ?
I would think this would be a often-requested feature in a modern pbx.
Thanks,
Chris Clifton
- Original Message -
From: "Brian West" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, February 06, 2004 7:4
I believe it is a requirement. When I bought mine, I had the same
issue. After talking to Digium, I was informed that the card would not
be recognized in a non-PCI 2.2 slot. I put it in another (newer) box
and it came right up.
-Original Message-
From: Andrew Kohlsmith [mailto:[EMAIL
Steve Foy wrote:
Right... It just happened there now, this came up:
Feb 5 14:34:18 WARNING[1133742896]: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 3 (Response)
So did it drop a few seconds into the call...like 5 - 15 seconds? If so
then you are having a problem with call
Hi,
After allot of trial and error I found what I did wrong. I was missing the
port.
This config works if anyone needs it.
Voiceglo config
[general]
port=5060
bindaddr=0.0.0.0
disallow=all
allow=gsm
allow=ulaw
allow=alaw
maxexpirey=180
defaultexpirey=160
tos=reliability
register=973111:[EMAI
SIP.CONF
[general]
; Codecs – your choice
disallow=>all
;allow=>gsm
allow=>ulaw
allow=>alaw
;allow=>ilbc
;allow=>spx
allow=>g723
allow=>g729
register=>1234:[EMAIL PROTECTED]/5000
[fwd.pulver.com]
type=>friend
secret=>password
username=>1234
host=>fwd.pulver.com
> Now we're getting somewhere! The TDM400P is a PCI 2.2 card. So
> depending on what you mean by "an older motherboard", that might be your
> problem.
Um, the TDM400P is PCI 2.2 compliant. PCI 2.2 is not a requirement to my
knowledge.
Regards,
Andrew
__
On Fri, Feb 06, 2004 at 07:58:09PM -0500, Sean Cheesman wrote:
> Now we're getting somewhere! The TDM400P is a PCI 2.2 card. So
> depending on what you mean by "an older motherboard", that might be your
> problem.
It's a dual Celery (A-Bit, I think) board from around 1999-2000.
Tim
--
>
---
Attention: Non-Delivery Report
---
This report is generated by the email server at:
mantraent.com
The message with subject:
"RE: [Asterisk-Users] modprobe wcfxs"
an
Now we're getting somewhere! The TDM400P is a PCI 2.2 card. So
depending on what you mean by "an older motherboard", that might be your
problem.
-Original Message-
From: Tim Sailer [mailto:[EMAIL PROTECTED]
Sent: Friday, February 06, 2004 6:53 PM
To: [EMAIL PROTECTED]
Subject: Re: [As
On Fri, 6 Feb 2004, John Bittner wrote:
> I am trying to get voiceglo to work with asterisk. I have tried many sip
> configs and cant seem to get it to register. Please if someone can look at
> this softphone config and let me know what I am doing wrong I would
> appreciated it.
I noticed that the
Do you here the beeps on the phone or on the Console machine. For about the
last 2 weeks I have been hearing random beeps on either of my two sip
phones. I do not have a console running anywhere so I have no text
printing.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTE
Nope.
bkw
On Fri, 6 Feb 2004, Chris Clifton wrote:
> On the 7960's with *, when an internal sip line is dialed, is it possible
> for the 7960 to display a status on the lcd that 'this ext is busy', etc. if
> the line is in use ? Does this happen by default ?
>
> Thanks,
> Chris Clifton
>
> _
Isn't the demo codec 1 channel only? Then one side is g729 and the other
is what?
do a sip show channels
bkw
On Fri, 6 Feb 2004, Wes Marderness wrote:
> Hi,
>
> Running Version 0.7.2, I receive the following error when attempting to
> connect two SIP Devices.
>
> WARNING[16399]:rtp.c : 1204 as
> "T" == T Chan <[EMAIL PROTECTED]> writes:
T> if I configure that way, even 01163 calls will all go to the second
T> IP address as per 011.,1,Application(). If I take out the 011.,
T> then calls WILL go to 01163., if I put the two together it will
T> always go to 011. extension.
The list arc
> "Scott" == Scott Russ <[EMAIL PROTECTED]> writes:
Scott> Does anyone know if/how well Asterisk will run under User Mode
Scott> Linux? Will the ztdummy or zaprtc modules work with it?
Haven't tried the modules, but an all-voip setup works well, provided
there is enough ram set aside for the
Hi, i finally was able to get dialtone on my
fxs board. !! however, i think i am missing something in the fwd setting to make
work my account.
i am getting an error authenticating my
account
could someone send me the exact settings to put on
sip.conf ? to make it work ?
i have my own
On Fri, Feb 06, 2004 at 05:36:47PM -0600, Tilghman Lesher wrote:
> On Friday 06 February 2004 16:26, Tim Sailer wrote:
> > OK, folks... I'm having the same problem as a few people. "device
> > not found" when I do the modprobe wcfxs. I looked in the archives,
> > and I see 4 or 5 people have had t
Does anyone know if/how well Asterisk will run under User Mode Linux? Will the
ztdummy or zaprtc modules work with it?
Thanks,
Scott
-
This mail sent through IMP: http://horde.org/imp/
___
Asterisk-User
On Friday 06 February 2004 16:26, Tim Sailer wrote:
> OK, folks... I'm having the same problem as a few people. "device
> not found" when I do the modprobe wcfxs. I looked in the archives,
> and I see 4 or 5 people have had the same problem. I even foudn the
> reply to a post like mine that said "
g.729 show license usage will show you how many G.729 licenses are currently being
used.
Derek Samford
Net Phone Blue, Inc
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Fri 2/6/2004 6:19 PM
To: [EMAIL PROTECTED]
Cc:
Subject:[Asterisk-Use
I installed the codec, got confirmation from the istall process.
Is there show command or a log that I can use to confirm calls are using
G.729.
Do I need to restart asterisk or can I just reload the config?
Thanks!
Mark Vickers, RealNetworks Inc. Desk: (206) 674-2391 Fax: (206)674-3588
___
Dear Chris,
Thanks for your lesson, it sort of works but not perfect.
I tried
exten => _01163.,1,Application()
exten => _011.,1,Application()
because I want to send Philippines to a different IP address than the rest
of the world, but if I configure that way, even 01163 calls will all go to
the
Hi,
I am trying to get voiceglo to work with asterisk. I have tried many sip
configs and cant seem to get it to register. Please if someone can look at
this softphone config and let me know what I am doing wrong I would
appreciated it.
Thanks
John Bittner
Simlab.net
This is my config and the
George Ye wrote:> Hi,> > I am a new player of the Asterisk. I have a strage problem with > musiconhold feature. Can anyone give some clues what might be the > problem? A description of the problem is as follows:> > 1. Call from Cisco ATA 186 without silence suppression, when I push > the "hold" b
Hi,
Running Version 0.7.2, I receive the following error when attempting to
connect two SIP Devices.
WARNING[16399]:rtp.c : 1204 ast_rtp_bridge : codec0 = 524556 is not codec1 =
524558, cannot native bridge.
The bridge is made but the quality of the call is bad, a lot of disturbing
noises in bac
On Fri, 2004-02-06 at 14:08, Brian West wrote:
> Post your config and we can see whats up..
Incoming lines are via a FXO card in a CAC channel bank, although we had
same issue with the lines connected to X100P cards.
T1 Channels 1-12 are FXS to the handsets, 13-24 are FXO, only 13 - 16
are conne
OK, folks... I'm having the same problem as a few people. "device not
found" when I do the modprobe wcfxs. I looked in the archives, and I see
4 or 5 people have had the same problem. I even foudn the reply to a post
like mine that said "look in the archives, others have had the same problem".
Ver
They're not the only ones doing that. Check out the IP Telephony
Solutions section and open the PDF for their "SIP Media Gateway and PBX"
product.
http://www.hautespot.net/products/index.html
I'm going to find out how much they're charging for it shortly...
Thanks,
Francois
-Original Me
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 having to specify
the number of digits?
For example, if I want to send 01163 (Philippines to a certain IP address),
is there anyway
Hi Gus,
Thanks for your reply. I have tried below and still didn't work.
exten => _1905XXX,1,Dial,OH323/h323:[EMAIL PROTECTED]
or
exten => _1905XXX,1,Dial,OH323/[EMAIL PROTECTED]
and now asterisk gives out below error
Feb 6 16:12:41 WARNING[30740]: pbx.c:1773 ast_pbx_run: Channel 'H323
On the 7960's with *, when an internal sip line is dialed, is it possible
for the 7960 to display a status on the lcd that 'this ext is busy', etc. if
the line is in use ? Does this happen by default ?
Thanks,
Chris Clifton
___
Asterisk-Users mailing li
While looking around for some ISDN phones I found this auction and
thought some of you may get a kick out of this.
http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=3075387057&category=11908
Seems they are selling a 2u server a T100P card and 10 Budgetone phones
for $3995. What I find funny is t
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 do
Is it possible to turn auto answer for the console off and on in the dialplan? If so
would someone be so kind as to post a short example.
I'd like to use the same sound card for external ringing over the paging system that
I'm using for overhead paging. So my idea was to put the console in the
Tilghman Lesher wrote:
On Thursday 05 February 2004 05:50, Jeremy McNamara wrote:
A type=friend is simply both a type=user and type=peer using the same
set of config directives. While a type=friend makes things almost
trivial to get calls working in both directions, it will limit the
flexibilit
Steven Critchfield wrote:
On Fri, 2004-02-06 at 13:46, Stephen R. Besch wrote:
Every once and a while * throws a new wrinkle at me. It has started, all
on its own, to make these annoying little beeps evey time a message
prints at the CLI. If I bring down * and restart, they go away for a
time,
> When a call is in progress I'll be watching it at the console with "iax2
> show channels" Here are my stats from one particular call:
>
> 66.225.202.72benshaw 1/16413 00048/00035 00489ms 0221ms ILBC
> 66.225.202.72benshaw 1/16413 00137/00125 00487ms 0270ms ILBC
Yes I have tried immediate = yes.
I do get dialtone immediately when I go off-hook or dial in, but then
Asterisk won't accept any further input whether dialing from the Norstar or
dialing on the T1 side. Essentially, I can't break dialtone.
- Original Message -
From: "Robert Hajime Lanni
Post your config and we can see whats up..
bkw
On Fri, 6 Feb 2004, Mark Farver wrote:
> We're still have problems with the outgoing voice message interfering
> with the touch tone detection. Often the first touch tone pressed will
> be detected twice. If I configure asterisk to not play the me
I have been kind of tracking IAX2 calls and trying to measure performance
with a given "iax2 set jitter" command. My default is 250ms..
When a call is in progress I'll be watching it at the console with "iax2
show channels" Here are my stats from one particular call:
66.225.202.72benshaw
On Fri, 2004-02-06 at 13:46, Stephen R. Besch wrote:
> Every once and a while * throws a new wrinkle at me. It has started, all
> on its own, to make these annoying little beeps evey time a message
> prints at the CLI. If I bring down * and restart, they go away for a
> time, then seem to sponta
Every once and a while * throws a new wrinkle at me. It has started, all
on its own, to make these annoying little beeps evey time a message
prints at the CLI. If I bring down * and restart, they go away for a
time, then seem to spontaneously reappear sometime later. It's almost as
if * is star
What is your zapata.conf?
Have you tried "imediate = yes"?
> John and sundry others:
>
> First thanks for your help.
>
> You have succiently summed up the problem. I do not get dialtone fast
> enough.
>
> The following is a test dialplan that I set up this morning after recieveing
> the many kind
John and sundry others:
First thanks for your help.
You have succiently summed up the problem. I do not get dialtone fast
enough.
The following is a test dialplan that I set up this morning after recieveing
the many kind e-mails, It's very basic, but it does allow me to process a
call to my phon
We're still have problems with the outgoing voice message interfering
with the touch tone detection. Often the first touch tone pressed will
be detected twice. If I configure asterisk to not play the message, or
if people wait till the outgoing message stops, it works flawlessly.
I've noticed th
> I don't think that we've reached 1.0 stable, though, have we?
> branching is an essential precursor in order to allow stablisation of
> the current featureset to happen in a different space to the addition of
> new features.
> Personally I welcome this - both branch & HEAD should benefit :)
> How
No snap shot is needed! You are able to check out the 1.0 branch from
cvs. Only bug fixes will go in this branch so you can automate the
checkout/update and rpm build process and produce daily 1.0 rpms.
To check out code from our STABLE 1.0 Branch CVS repository for Asterisk
ONLY:
# cd /usr/src
Hi,
I recently received my development kit with 1 x100p and one tdm400p (1) fxs
port.
I installed everything from the digium disk that i received with my kit,
however, i dont; know what to do next.
I would like to be able to call through the internet using xten (pc2phone)
and terminate the call i
On Fri, 6 Feb 2004, Brian West wrote:
> Is someone going to do the v1-0-stable RPMS?
As soon as there is a .tgz available, (assuming it isn't already on the
FTP site) I will be happy to do it.
> Not sure if anyone knows that it was branched yet or not. Everyone was
> jumping up and down and c
I am having the same problem with a new CVS.
Patrick also has the problem here
http://lists.digium.com/pipermail/asterisk-users/2004-January/035114.html
Keven had a problem here
http://lists.digium.com/pipermail/asterisk-users/2004-January/035262.html
but was able to get it fixed. Can you post a
George Ye wrote:
Hi,
I am a new player of the Asterisk. I have a strage problem with
musiconhold feature. Can anyone give some clues what might be the
problem? A description of the problem is as follows:
1. Call from Cisco ATA 186 without silence suppression, when I push
the "hold" button a
> However, I think it's too early for RPMS of a snapshot of this branch of
> CVS ;)
If I'm going to do the RPM's I'm probably going to want to do a daily
snapshot build system that builds the RPMs.
Chris
___
Asterisk-Users mailing list
[EMAIL PROTECTE
I've acquired a Cisco 7905 with H323 s/w that I have connected to *. It can
make calls happily enough to H323 & SIP extensions and out to the PSTN,
however when ever I try to call it from any destination the call fails with
H323:0 Could not call 192.168.9.23
Hungup 'H323:0'
Everyone is busy at t
Brian West wrote:
Is someone going to do the v1-0-stable RPMS?
Not sure if anyone knows that it was branched yet or not. Everyone was
jumping up and down and chanting BRANCH BRANCH BRANCH
I don't think that we've reached 1.0 stable, though, have we?
branching is an essential precursor in order
On Fri, 2004-02-06 at 04:33, Marc Fargas wrote:
> It drives me to a new question... how can I concatenate three strings on
> extensions.org ?
>
> That is, the command, and the two args; The arguments are the source e164
> and destination e164 numbers of the current call.
>
> Something like "/bin
I was wondering if this would work
set a variable (varX) in macro-test
call another macro [macro-subroutine]
have varX available within [macro-subroutine]
[macro-test-1]
; ${ARG1} - extension
setvar(var1=foo)
macro(subroutine,${ARG1})
[macro-subroutine]
do something with varX
Or do you h
I don't need them but just asked because others might.
bkw
On Fri, 6 Feb 2004, Chris Tooley wrote:
> I'd be happy to help. I've got several boxes in various stages of
> RedHat and I've a SuSE 9 box but nothing older than that.
>
> Chris
>
> On Fri, 2004-02-06 at 09:59, Brian West wrote:
> > Is
I'd be happy to help. I've got several boxes in various stages of
RedHat and I've a SuSE 9 box but nothing older than that.
Chris
On Fri, 2004-02-06 at 09:59, Brian West wrote:
> Is someone going to do the v1-0-stable RPMS?
>
> Not sure if anyone knows that it was branched yet or not. Everyone
Is someone going to do the v1-0-stable RPMS?
Not sure if anyone knows that it was branched yet or not. Everyone was
jumping up and down and chanting BRANCH BRANCH BRANCH It fially
happened and nobody says a word haha.. :)
bkw
On Fri, 6 Feb 2004, Chris Tooley wrote:
> On Thu, 2004-02-05 at
Small error in the zone file caused this. Its fixed now.
bkw
On Fri, 6 Feb 2004, Isamar Maia wrote:
>
> I bought recently a G729 and didn't any response...
> maybe for the same reason? :-(
>
> Isamar
>
>
> On Fri, 6 Feb 2004, Vic Cross wrote:
>
> > Chris,
> >
> > On Fri, 6 Feb 2004, Christopher
Hi,
Running Version 0.7.2, I receive the following error when attempting to
connect two SIP Devices.
WARNING[16399]:rtp.c : 1204 ast_rtp_bridge : codec0 = 524556 is not codec1 =
524558, cannot native bridge.
The bridge is made but the quality of the call is bad, a lot of disturbing
noises in bac
This seems to only apply to non-zap channels participating in the
conference, incidently.
On Fri, 6 Feb 2004, mattf wrote:
> Currently Asterisk will cause a Kernel Panic if you are using the Linux SMP
> kernel and have about 30 channels in conference. Here's the bug listing:
>
> http://bugs.digiu
On Thu, 2004-02-05 at 19:42, Greg Boehnlein wrote:
> On Wed, 4 Feb 2004, Greg Boehnlein wrote:
>
> > On Wed, 4 Feb 2004, Chris Tooley wrote:
> >
> > > Well, I don't really know all that much about SuSE either. I just
> > > installed it about 19 hours ago for the first time.
> >
> > Well, depend
Hi,
- Original Message -
From: "Cees de Groot" <[EMAIL PROTECTED]>
> >Do you mean that it works with version 0.9.3, but not with 0.9.6?
> >
> yes.
>
> >Have you tried with both IAX(1) and IAX2?
>
> no - iax1 only.
IAX (1) will not be supported in the future anymore.
Version 0.9.7 which I
Currently Asterisk will cause a Kernel Panic if you are using the Linux SMP
kernel and have about 30 channels in conference. Here's the bug listing:
http://bugs.digium.com/bug_view_page.php?bug_id=963
MATT---
-Original Message-
From: Paulo Mannheimer [mailto:[EMAIL PROTECTED]
Sent
Hi, we are setting a 120-channel conference server and would like to
learn if someone already did this (hardware, problems, etc...)
Best regards,
PauloHM
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[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asteri
Dan <[EMAIL PROTECTED]> said:
>Do you mean that it works with version 0.9.3, but not with 0.9.6?
>
yes.
>Have you tried with both IAX(1) and IAX2?
no - iax1 only.
>Can you use debug mode in 0.9.6 and send me the log?
>
sure. I'll gather some more data and send it to you directly. I mainly
poste
It must be:
exten => _1905XXX,1,Dial,OH323/h323:[EMAIL PROTECTED]
or
exten => _1905XXX,1,Dial,OH323/[EMAIL PROTECTED]
Hope this helps,
Gus
- Original Message -
From: "Anthony Law" <[EMAIL PROTECTED]>
To: "Mailing List Asterisk" <[EMAIL PROTECTED]>
Sent: Friday, February 06, 2004
Hi,
Thanks for your reply. I am definite that my h323 is running on ciscoB
because the below scenario is working fine.
pstnciscoA-ciscoBpstn
I have also eliminated access-list problem because if my access-list is
applied I could see packets hiting my access-list
permit tcp host 192.
You do have to add some extensions, copy some files and have some scripts
run on your Asterisk server, as well as have a MySQL database set up on a
machine somewhere before you can install the client on a machine. All of
that is explained in the documentation included with the package.
As for the
> -Original Message-
> From: mattf [mailto:[EMAIL PROTECTED]
> Sent: Thursday, February 05, 2004 4:33 PM
> To: '[EMAIL PROTECTED]'
> Subject: [Asterisk-Users] Asterisk GUI Client - New verison 0.9
>
> Hello,
>
> I have made many changes/improvements/bug fixes to the Asterisk GUI
client
>
Vic Cross ha scritto:
If anyone is interested, please let me know. I'll lodge it in the Mantis
for chan_sccp if we like it.
I like it!
Where can we get it?
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[EMAIL PROTECTED]
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how many simultanius calls does voiceglo permit???
Miguel
On Fri, 2004-02-06 at 01:28, Cameron Palmer wrote:
> IAX is what they use with glophone. http://webphone.voiceglo.com. It is a
> seperate server from the myphone.voiceglo.com SIP gateway. The IAX gateway
> is msps01-nyc.voiceglo.com on po
Hi Thomas,
I think you'll probably need a dedicated fax board to do this. Commetrex
in Atlanta has a very nice solution. Check it out at www.commetrex.com. Also
Natural MicroSystems, BrookTrout and Eichon all have very capable fax
boards. I have been developing voice and fax applications with th
Sorry folks (I know it's annoying when people reply to their own posts)...
On Fri, 6 Feb 2004, Vic Cross wrote:
I just wanted to advise that I've done my patch to chan_sccp to provide
this behaviour -- when a call comes in on any line, not just the
'selected' line, taking the phone off-hook an
I bought recently a G729 and didn't any response...
maybe for the same reason? :-(
Isamar
On Fri, 6 Feb 2004, Vic Cross wrote:
> Chris,
>
> On Fri, 6 Feb 2004, Christopher Lee wrote:
>
> >- The following addresses had permanent fatal errors -
> > <[EMAIL PROTECTED]>
> > (reason
> "Marc" == Marc Fargas <[EMAIL PROTECTED]> writes:
Marc> It drives me to a new question... how can I concatenate three
Marc> strings on extensions.org ?
Marc> That is, the command, and the two args; The arguments are the
Marc> source e164 and destination e164 numbers of the current call.
Ma
Looks like you are shy a zero
Try exten => _50.,Prefix,001051
At 12:06 07/01/04, you wrote:
Hello,
I can not seem to be able to get StripMSD and Prefix to work for me in
extensions.conf. This is an example of what I have:
exten => _050.,1,StripMSD,1
exten => _50.,Prefix,01051
exten => _001051.,
Hi,
- Original Message -
From: "Cees de Groot" <[EMAIL PROTECTED]>
> I had some users complaining that DIAX only rung twice in the call
> sequence for incoming calls (ring user 1 20 secs, user 2 20 secs, ...).
> When testing, it turned out that with DIAX 0.9.3 I got the expected
> result
I had some users complaining that DIAX only rung twice in the call
sequence for incoming calls (ring user 1 20 secs, user 2 20 secs, ...).
When testing, it turned out that with DIAX 0.9.3 I got the expected
result:
Feb 6 11:24:36 -- Executing Dial("CAPI[contr1/313650768]/0",
"IAX/ha/s|20|rt"
Chris,
On Fri, 6 Feb 2004, Christopher Lee wrote:
>- The following addresses had permanent fatal errors -
> <[EMAIL PROTECTED]>
> (reason: 554 <[EMAIL PROTECTED]>: Recipient address rejected: Relay
> access denied)
>
>- Transcript of session follows - ... while talkin
It drives me to a new question... how can I concatenate three strings on
extensions.org ?
That is, the command, and the two args; The arguments are the source e164
and destination e164 numbers of the current call.
Something like "/bin/false " + $SOURCE164 + " " + $DEST164
-Mensaje original-
Klaus-Peter Junghanns ha scritto:
oh yes...
i added callgroup support for chan_capi. That's why you have to load
res_parking.so before chan_capi.so. So in modules.conf you need.
load => res_parking.so
load => chan_capi.so
[global]
chan_capi.so=yes
best regards
kapejod
--
Klaus-Peter Junghanns
CE
oh yes...
i added callgroup support for chan_capi. That's why you have to load
res_parking.so before chan_capi.so. So in modules.conf you need.
load => res_parking.so
load => chan_capi.so
[global]
chan_capi.so=yes
best regards
kapejod
--
Klaus-Peter Junghanns
CEO, CTO
Junghanns.NET GmbH
Breit
Klaus-Peter Junghanns ha scritto:
Hi BRI people,
chan_capi 0.3.1 is now released, including a fix for the pipe leak.
bristuff 0.0.2rc7 is available now too. Including a zaptel driver
for the HFC-S PCI A based ISDN cards (with echo cancelation, TE and
NT mode).
We'll also have a devkit for zaptel
Is it just me or is everyone having problems with emailing digium?
I've tried sending two emails, but they keep getting returned with the
following errors:-
- The following addresses had permanent fatal errors -
<[EMAIL PROTECTED]>
(reason: 554 <[EMAIL PROTECTED]>: Recipient addres
Hi to everybody,
Is it possible with the skinny module (skinny.conf) in asterisk
configuring the buttons on a CISCO 12 SP+ ?
Is someone working on this ?
Thank You.
--
Davide Yachaya
HyperGrid s.r.l.
V.le Golgi 63 - 27100 Pavia - ITALY http://www.hypergrid.it
Tel: +39,
Hi Adam,
Could you show us your configs on Asterisk and on
Vega so everyone on the list can have a guide to get Vega working with
Asterisk?
Thanks!
David
- Original Message -
From:
Low,
Adam
To: '[EMAIL PROTECTED]'
Sent: Friday, February 06, 2004 12:47
AM
Klaus-Peter Junghanns <[EMAIL PROTECTED]> said:
>bristuff 0.0.2rc7 is available now too. Including a zaptel driver
>for the HFC-S PCI A based ISDN cards (with echo cancelation, TE and
>NT mode).
>
If I followed this a bit and understood it correctly, with e.g. the quad
BRI card it will be possible
> "Marc" == Marc Fargas <[EMAIL PROTECTED]> writes:
Marc> I've seen its possible to use the System applications, but what
Marc> about passing arguments to the command ?
A quick look at app_system.c shows that it just passes the string
unaltered to system(3). So, running "man 3 system" will s
I have
a Vega 50 BRI working without any of the issues you mentioned, the dual SIP
registrations is normal for most multi-line boxes enabled split
users.
Rgds,
Adam
-Original Message-From: Glenn Dalgliesh
[mailto:[EMAIL PROTECTED]Sent: 05 February 2004
20:11To: [EMAIL PROT
Hello,
I`m using a bunch of ATA-186 with MGCP firmware, and users are
complaining that sometimes, an avarage about one in 17-20 calls when
they try to do a supervised transfer via FLASH, the calling party is
dropped, or that could also happen when the press FLASH and then dial an
extension to
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