Hi,
You need the PA-VFC-2TE1+ cards. It supports 60 calls for codecs such as
G723 and 120 calls for G729a and b(with the addition of a PA-MCX card).
Cheers,
Abdul
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael Kane
Sent: 12 September 2003 18:30
T
hi.
actualy the iax2 conf file is the same of iax .
iax2 port is hardcoded in channels/iax2.h, line 72 (more or less)
You can change it & recompile.
matteo.
Il sab, 2003-09-13 alle 08:49, Dan ha scritto:
> Hi Martin,
>
> - Original Message -
> From: "Martin Pycko" <[EMAIL PROTECTED]>
>
- Original Message -
From: "Robert Boardman" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, September 12, 2003 10:26 PM
Subject: [Asterisk-Users] Dect Phone
> Hi
>
> I have a problem with a new DECT phone I have bought
>
> The key pad works like a Mobile phone where you dial
Hello All,
I am still having some difficulty working to monitor an already active
channel. I did some experimenting with the Monitor application without
achieving my desired results.
Here are the relevant parts of my extensions.conf file:
[CustomerSide]
exten => 2,1,StopMonitor
exten => 3,1,M
FWIW, I just immplemented * on a RH9 box using the CVS without any problems
whatsoever. The RH9 box was built from CD's as a workstation (with everything
installed), up2date ran to bring it reasonably current, etc. I had installed
"ser" a few weeks ago and it worked properly as well. Ser was shutdo
I'm having some of the same issues and it seems to be related to transmission
levels. CallerID worked fine prior to me messing with rxgain/txgain, but I've
not gone back to verify what I did to muck it up as yet.
> There are two things I can think of..
>
> 1. You are not
Can anyone please direct me to UK based suppliers of equipment. Website
URL's would be appreciated. TIA
--
*
Not everyone is touched by an Angel
Those that are, never forget the experience
*
___
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[EMAIL PROTEC
I bought some Snom phones which work nicely with Asterisk from:
ProVu Communications Ltd
Bank House
Marsden
Huddersfield
HD7 6BR
01484-840048
[EMAIL PROTECTED]
www.provu.co.uk
-Original Message-
From: Angel Gabriel
Sent: 13 September 2003 13:02
To: * Users
Subject: [Asterisk-Users
I was wondering, can I test * using just a modem card? I was want to
check ome of the features, before I go and buy some cards. (Thanks for
th elink to the reseller page, you know who you are!)
--
*
Not everyone is touched by an Angel
Those that are, never forget the experience
*
At Sat, 13 Sep 2003 09:52:34 +0200 , [EMAIL PROTECTED] wrote:
>hi.
>actualy the iax2 conf file is the same of iax .
>iax2 port is hardcoded in channels/iax2.h, line 72 (more or less)
>You can change it & recompile.
>
>matteo.
>
Thanks a lot.
Dan
...
__
http://www.telappliant.co.uk
- Original Message -
From: "Lee Redmayne" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Saturday, September 13, 2003 1:11 PM
Subject: RE: [Asterisk-Users] UK Suppliers
I bought some Snom phones which work nicely with Asterisk from:
ProVu Communicatio
On Sat, 2003-09-13 at 13:42, Rich Adamson wrote:
> FWIW, I just immplemented * on a RH9 box using the CVS without any problems
> whatsoever. The RH9 box was built from CD's as a workstation (with everything
> installed), up2date ran to bring it reasonably current, etc. I had installed
> "ser" a few
> > FWIW, I just immplemented * on a RH9 box using the CVS without any problems
> > whatsoever. The RH9 box was built from CD's as a workstation (with everything
> > installed), up2date ran to bring it reasonably current, etc. I had installed
> > "ser" a few weeks ago and it worked properly as wel
Cerrajetto wrote:
Hello:
I am testing Asterisk with oh323.
My question is: can Asterisk route some calls thru a second h323 gateway (a
h323 <-> PSTN gw)?
- Asterisk ip: 192.168.1.10
- h323<->PSTN gw: 192.168.1.20
I've tried:
exten => _9,1,Dial(OH323/192.1.1.20)
or
exten => _9
I'm looking for a source for 50-pin amphenol
cables, the ones used to connect Adtran's to
punch down blocks. Preferably, one that's
mail order and takes orders over the internet.
Thanks.
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On Sat, 2003-09-13 at 09:18, Peter Pauly wrote:
> I'm looking for a source for 50-pin amphenol
> cables, the ones used to connect Adtran's to
> punch down blocks. Preferably, one that's
> mail order and takes orders over the internet.
> Thanks.
You should be able to get it at any decent size co
Hi Rich-
Yes, I've been reasonably happy with RedHat 9 too.
The only problem I had was the defunct (zombie) process issue. This only
happened when I called an external AGI program (mine are in Perl). Are you
saying that you don't have this problem after returning from AGI programs?
If so, maybe
>From what I see this *IS* a problem with the CVS code...
as a quick fix I suggest using the zaptel code from august 18th 2003 since that is
known to work (I'm using it after having the same problems as you)
It's kinda strange if this isn;t regarded as a bug, as Digium have then EOL'd some of
>
>Yep, it probably will not work with your motherboard. You might try
>setting -DNO_CALIBRATION in the Makefile, then running 'make clean
>all install' and trying again (this has worked for some people).
>Failing that, try it with a different motherboard.
>
>-Tilghman
>
This is a CODE issue not
Scott,
I'm relatively new to * and have not done anything with AGI specifically
(unless the default installation is doing something that I'm not aware
of). So far, everything that I've done with * is from an educational
perspective attempting to learn/understand it, etc,
Rich
> Yes, I've been re
Is there an asterisk cvs commit email list?
Any project I have ever worked on in the past, always had a cvs commit email list.
Anytime someone does a commit you receive the file name
and comments. You can then make the decision if want to update or not.
It can also help you narrow your focus wh
http://lists.digium.com/mailman/listinfo/asterisk-cvs
matteo.
Il sab, 2003-09-13 alle 18:49, Bob Knight ha scritto:
> Is there an asterisk cvs commit email list?
>
> Any project I have ever worked on in the past, always had a cvs commit email list.
> Anytime someone does a commit you receive t
http://lists.digium.com
Jeremy McNamara
Bob Knight wrote:
Is there an asterisk cvs commit email list?
Any project I have ever worked on in the past, always had a cvs commit
email list. Anytime someone does a commit you receive the file name
and comments. You can then make the decision if
Nice one!
Took all of about 30 seconds to install, including downloading from the net.
Just got the latest CVS and copied it into the web folder and opened up
konqueror.
Everything seems to be working fine. Off to do some testing of it now.
Cheers,
Darren.
On Friday 12 Sep 2003 11:34 am, Pet
Have you tried:
exten => _9,1,Dial(OH323/${EXTEN:[EMAIL PROTECTED])
bkw
On Sat, 13 Sep 2003, Michael Manousos wrote:
> Cerrajetto wrote:
> > Hello:
> >
> > I am testing Asterisk with oh323.
> >
> > My question is: can Asterisk route some calls thru a second h323 gateway (a
> > h323 <->
Martin,
Your statement below is somewhat confusing. Where do you find the choice
of 1 or 2?
This is the latest voicemail.conf:
;
; Voicemail Configuration
;
[general]
; Default formats for writing Voicemail
;format=g723sf|wav49|wav
format=gsm|wav49|wav
; Who the e-mail notification should appear
On Saturday 13 September 2003 07:22, Angel Gabriel wrote:
> I was wondering, can I test * using just a modem card? I was want to
> check ome of the features, before I go and buy some cards.
The quick answer is no. If you'd like to know why not, I encourage you
to search the mailing list archives.
On Saturday 13 September 2003 11:09, Andy Powell wrote:
> From what I see this *IS* a problem with the CVS code...
>
> as a quick fix I suggest using the zaptel code from august 18th 2003
> since that is known to work (I'm using it after having the same
> problems as you)
>
> It's kinda strange if
I have gotten a Howlink CL-100 working with chan_h323, but I forget the
exact details. I think I had to disable fast_start & a few other things
on both ends.
Nick
On Fri, Sep 12, 2003 at 10:17:39PM -0700, [EMAIL PROTECTED] wrote:
> I'm trying to use a Howlink CL-100 ip phone with *
>
> It
I'm attempting to install * for the first time and encountering some
problems... I'm running an old 166MMX processor with 50megs RAM... SuSE 8.1
distro...
when I make clean ; make install in the /usr/src/asterisk dir, I get a "error:
termcap support not found". the termcap libraries are i
On Sat, 2003-09-13 at 12:41, [EMAIL PROTECTED] wrote:
> I'm attempting to install * for the first time and encountering some
> problems... I'm running an old 166MMX processor with 50megs RAM... SuSE 8.1
> distro...
>
>
>
>
> when I make clean ; make install in the /usr/src/asterisk dir, I g
Hi. I have sjphone installed on windows and working
except for dtmf. I read the docs for sjphone and it
uses inband dtmf. I configired dtmfmode=inband but it
still does not recognize it. Someone on the lists
said that inband only works using alaw or ulaw but i
tried only allowing that too but st
>> It's kinda strange if this isn;t regarded as a bug, as Digium have
>> then EOL'd some of their cards and not told anyone, while continuing
>> to sell them...
>
>Compare revision E to revision C of the card. Revision C is no longer
>being sold by Digium.
>
This may be true, however, they were b
- Original Message -
From: "Kevin Bockman" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Sunday, September 14, 2003 4:08 AM
Subject: [Asterisk-Users] SJphone DTMF?
> Hi. I have sjphone installed on windows and working
> except for dtmf. I read the docs for sjphone and it
> uses in
--- Shaun Ewing <[EMAIL PROTECTED]> wrote:
>
>
> - Original Message -
> From: "Kevin Bockman" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Sunday, September 14, 2003 4:08 AM
> Subject: [Asterisk-Users] SJphone DTMF?
>
>
> > Hi. I have sjphone installed on windows and
> workin
Andy Powell wrote:
This may be true, however, they were being sold in May of this year and I don;t
expect a piece of hardware to have a lifespan of 3.5 months!
From what I hear revision C cards are green and revision E cards are blue. It certainly
also sounds like some people were getting the C
> Nope! Frustrating. Can you show me what you are
> allowing/denying for codecs?
>
> Thanks.
>
I don't explicitly allow or deny any codecs, it's all default.
-Shaun
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Hello.
I've seen some mentions of asterisk possibly being used as an inexpensive
voicemail attachment to a commercial PBX etc.
Does anyone here, have experience of using it in this fashion ?
What commercial systems have been successfully attached too ?
How is the attachment made ?
Analog, dig
Hello,
I have ISA card LineJack. I could not find any information
if this card can work as fxo with Asterisk. If it can work,
can somebody point me how to install it on my Asterisk box.
Or maybe there is some documentation about it how to install
LineJack.
I will be very thankful for any help.
R
--- Shaun Ewing <[EMAIL PROTECTED]> wrote:
> > Nope! Frustrating. Can you show me what you are
> > allowing/denying for codecs?
> >
> > Thanks.
> >
>
> I don't explicitly allow or deny any codecs, it's
> all default.
>
> -Shaun
> ___
Okay, I switche
Martin,
We currently are in dire need of a system to replace our Audix Voice Power
connected to our Merlin Legend switch. A little while ago, I knew
absolutely nothing about how to do this, and inquired on the list. The
response I got were extremely helpful to me, as I had no idea where to
start
On Sat, Sep 13, 2003 at 05:21:40PM -0400, Steve Creel wrote:
> The Legend currently has 8 analog lines in a hunt group to the
> voicemail/auto-attendant system. I went in after-hours and put a buttset
> inline to monitor the first line in the hunt group. When I call in from
> an outside line, the
On Sat, 13 Sep 2003, John Brown wrote:
>On Sat, Sep 13, 2003 at 05:21:40PM -0400, Steve Creel wrote:
>> The Legend currently has 8 analog lines in a hunt group to the
>> voicemail/auto-attendant system. I went in after-hours and put a buttset
>> inline to monitor the first line in the hunt group.
no matter which way I have my /etc/zaptel.cfg
this only started happening after a yesterdays CVS update
to my zaptel source...
Line 1: fxoks=1
Line 2: fxsks=2
Line 23: loadzone=us
Line 28: defaultzone=us
Zaptel Configuration
==
Channel map:
Channel 01: FXO Kewlstart (Defa
Hi Martin,
I have an installation where * is hooked to a Siemens Hicon 300, serving more
than 100 voicemail boxes.
Lines going in * are analog, with the PBX sending me some DTMF codes of who is
ringing, motive (busy, not available, etc...)
> Hello.
>
> I've seen some mentions of asterisk pos
it seems that the CVS process needs a bit of refinement.
there are no (from what I can see) clear "release points" or
tags in the system.
it would be nice to have a
ast-release
ast-stable
ast-current
style environment.
with
release == a dot release like 1.1. You can run a
"p
Can you call Digium support Monday or find someone on IRC who can help?
(#asterisk, irc.freenode.net, i'm kram)
Mark
On Sat, 13 Sep 2003, John Brown wrote:
> no matter which way I have my /etc/zaptel.cfg
> this only started happening after a yesterdays CVS update
> to my zaptel source...
>
>
> L
Does anyone know whether the linux machine running * have to have a
sound card on it in order for musiconhold to work for sip phones?
I've tried about everything (including tons of google searching) to get
it to work, and nothing.
When a call is placed on hold between two C7960's, the CLI indic
Any knows a link where to download:
Open H323 v1.11.7
and
PWLib v1.4.11
Thanks
Senad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jeremy
McNamara
Sent: 12 September 2003 20:04
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] h323 v oh323
Lan
Nope, you only need a sound board if you want to hear
things on the * server
On Sat, Sep 13, 2003 at 05:05:32PM -0600, Rich Adamson wrote:
>
> Does anyone know whether the linux machine running * have to have a
> sound card on it in order for musiconhold to work for sip phones?
>
> I've tried
http://www.nufone.net/downloads/
On Sat, 2003-09-13 at 17:31, Senad Jordanovic wrote:
> Any knows a link where to download:
>
> Open H323 v1.11.7
> and
> PWLib v1.4.11
>
>
> Thanks
>
> Senad
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf O
On Saturday 13 September 2003 17:31, Senad Jordanovic wrote:
> Any knows a link where to download:
>
> Open H323 v1.11.7
> and
> PWLib v1.4.11
http://www.openh323.org/bin/
-Tilghman
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Thanks John, but the question is not regarding the server. MOH is 'not'
working with the C7960's. If you wouldn't mind, read through the remainder
of the stuff below. Everything seems to suggest it should be working, but
the question was oriented around the error message "sound device: Resource
te
Hi all:
Somebody knows the mysql table structure for VoiceMail2 application?
Thanks in advance,
Gus
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*This message was transferred with a trial version of CommuniGate(tm) Pro*
Hey all,
Just noticed something that might be an issue. I have just made
asterisk crash consistently by doing the following.
I have a D-Link DG1102s running MGCP into asterisk and an extension *9
setup which dumps me int
Ok, I found it.
Sorry.
- Original Message -
From: "CW_ASN" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Saturday, September 13, 2003 8:06 PM
Subject: [Asterisk-Users] VoiceMail2 mysql table structure
> Hi all:
>
> Somebody knows the mysql table structure for VoiceMail2 application
On Saturday 13 September 2003 18:06, CW_ASN wrote:
> Somebody knows the mysql table structure for VoiceMail2 application?
CREATE TABLE users (
context char(79)DEFAULT '' NOT NULL,
mailbox char(79)DEFAULT '' NOT NULL,
passwordchar(79)DEFAULT '
Rich, what are you running for mpg123 ? is it the
real mpg123 or is it the fake mpg321 ??
also whats in your musiconhold.conf ??
is mpg123 running ??
ps -ax | grep mpg
On Sat, Sep 13, 2003 at 05:46:15PM -0600, Rich Adamson wrote:
> Thanks John, but the question is not regarding the server.
John,
As noted in the original post:
16243 pts/1S 0:00 /usr/bin/mpg123 -q -s --mono -r 8000 -b 2048 For-You.
So, yes it appears to be running, and the "For-You" is an mp3 that I've listened
to from the linux console (just to verify).
The musiconhold.conf file looks like:
;
Did you build the latest gnugk? I'm having trouble getting it to compile
properly. At the last step, I get several "duplicate symbol" errors.
Asterisk built fine, however.
-G
- Original Message -
From: "Scott Stingel" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Saturday, September
I'm using moh in the following way:
exten => 2000,1,Dial(SIP/2000|10|mr)
exten => 2000,2,VoiceMail(u2000)
exten => 2000,102,VoiceMail(b2000)
I think that is the same...
To all guys:
Is necesary have present ztdummy in order to work moh when zaptel device is
installed?
- Original Message --
Hello!
There is much info using SIP for this connection type. I have a TDM400P
with a regular phone connected to it. What string in extensions.conf would
need to be added so I can call a FWD number as well as receive a call from
FWD? I have seen some mention of using IAX. Is this necessary o
Hi all:
I was reading a time ago back the necessity to make * reloads from web
pages. Here is a small script in php to make charges of the system via Web.
Is dangerous, I know, but it must have some cases where its applicable. Was
taken from phpconfig, asterisk.reload perl script.
Regards,
Gus
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Bill
> Flood Sent: Saturday, September 13, 2003 10:02 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] * <--> FWD
>
>
> Hello!
>
> There is much i
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Darren Poulson
> Sent: Saturday, September 13, 2003 1:27 PM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] phpconfig is out in CVS
>
>
> Nice
On Sat, 13 Sep 2003, Nick wrote:
> I have gotten a Howlink CL-100 working with chan_h323, but I forget the
> exact details. I think I had to disable fast_start & a few other things
> on both ends.
there doesnt seem to be any option to disable fast_start on the cl-100.
i disabled it in * but it do
If you have been paying attention, you already know this, but this
weekend I have spent time ironing out the various details with my
chan_skinny code that has been out there, if you knew where to look. I
believe I now have all basic features operational and am going to be
working on getting t
I did some searching in the archive, but found only one message with
this same question and no answer. Hopefully it's a simple config problem.
When the Caller-ID is delivered, it is surrounded by double-quotes,
like this:
"ATA-57 1"
On long caller-id strings, the last character is cut off to ma
I have the MusicOnHold feature working great when called from ATA-186
extensions. It's pretty cool.
However, when I call from a BudgeTone-100 phone, no music is heard --
instead it continues the ringing feedback and acts like the call is
unanswered. At the same time, I can call from (multiple) A
It works fine for me, I created a 2nd "music on hold", tossed a bunch of
mp3 files into a directory and I can listen to music on the
speakerphone:
;radio @
exten => ,1,Answer
exten => ,2,MusicOnHold(default)
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-use
they bothered me too so i made a little mod to remove them.
here is a patch if you are interested.
Index: channels/chan_sip.c
===
RCS file: /usr/cvsroot/asterisk/channels/chan_sip.c,v
retrieving revision 1.178
diff -c -r1.178 cha
Hi Martin,
I?m currently in progress to replace our ITS System in company that is
connected to a BOSCH/Tenovis Integral 33xe. The how way it is planned is by
adding an additional line that behaves like a trunk and passes the
extensions dialed. We plan to use the chan_capi with 2 AVM B1 cards in P2P
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