Hey all,
I'm trying to add some logic to a dial-plan to allow the caller to
terminate a number with a #, but also accept it without this
terminator. While trying this, I noticed that, for example,
extension _[*0-9]XXX.# always seems to match, whether the last digit
dialled is a # or not. It's
Hey Steve,
On Mon, Mar 22, 2004 at 22:13:37 +0800, Steve Underwood wrote:
Hi all,
If you have had trouble with multiple concurrent channels running
app_rxfax or ap_txfax, where was a silly bug. Updated versions are
available at ftp://ftp.opencall.org/pub/spandsp
The latest
Yo all,
I have a problem with a Dutch (KPN) PSTN-line, connected to an X100P.
The call wil sound OK at first, but after 10-20 minutes, the audio will
start to crackle. Soon after that, this crackle turns into a continuous
noise and the parties won't be able to hear eachother anymore. It also
Yo Eric,
On Thu, Oct 02, 2003 at 11:56:44 -0500, Eric Wieling wrote:
Check /proc/interrupts to make sure the cards are not shareing IRQs with
anything.
Sorry, forgot to mention it. All Zaptel-cards in that machine
already have their own unique interrupts. I will try moving the
cards to
Hey Chris,
On Thu, Aug 07, 2003 at 04:12:57 +0200, Chris Wetemans wrote:
- Original Message -
From: The Traveller [EMAIL PROTECTED]
Hey all,
As there seem to be some problems with DTMF-signalling between chan_sip
and several clients, due to which many could not properly
Hey Jim,
On Wed, Aug 06, 2003 at 15:12:50 -0500, Jim Friedeck wrote:
I am having trouble with the AgenCallBackLogin app. I can't seem to
define a context for the queue.
Here is the relevant configs:
[...]
extensions.conf:
[c_in_1];internal lines (up to 48 agents and admins)
Hey Dan,
On Mon, Jul 28, 2003 at 22:50:21 -0300, Dan Fernandez wrote:
Is there a way in iax to have to endpoints talk to each other directly (after the
call is setup by *) without going through *. In sip, with * you can do it by
configuring sip.conf with canreinvite = yes.
AFAIK, IAX
Hey all,
As there seem to be some problems with DTMF-signalling between chan_sip
and several clients, due to which many could not properly dial a number
at the dial-tone of the XS4ALL-gateway at FWD-number 42442, I've now
arranged for a prefix on FWD for this gateway.
From FWD, you can now dial
Hey AJ,
On Fri, Jul 25, 2003 at 19:23:50 -0400, [EMAIL PROTECTED] wrote:
I can't seem to get musiconhold to work. I'm running asterisk on a RH9
box, I have the mpg123 package installed. In my zapata.conf file I have
the line MusicOnHold=default . In my musiconhold.conf file, in the
Hey all,
I'm experiencing a problem with chan_sip on a multi-homed machine.
The machine has 1 interface to the rest of the world and 1 interface
on a local network. The local network has public IP-addresses, though,
and the IP-addresses of both interfaces are reachable from the outside
world,
Hey Neel,
On Fri, Jul 25, 2003 at 10:40:55 -0500, Neel Datta wrote:
Thanks Roy, I this worked! Only one thing I can't seem to do- If I have
a password set in my sip.conf as in the 'secret' key, I can't get the
msn client to authenticate properly. (And yes, I'm typing the exact
same word
Yo,
I'm trying to get Asterisk working with Messenger 4.7. After skimming
through the list-archives, I've got it to register to my Asterisk-box
and can make calls. Unfortunately, there's no audio from the Messenger-
side of the call to the other caller. I can hear the caller in Messenger,
Yo all,
I just added FWD (http://fwd.pulver.com/) to the XS4ALL PSTN-gateway.
Here's a quick update on how it works:
VoIP:
From IAXTel, dial 31800rest of number for Dutch toll-free numbers.
FWD is not (yet) directly reachable from IAXTel. I'll talk to Mark to
see if he's intrested in setting
Hey Florian,
On Wed, Jul 16, 2003 at 13:56:45 +0200, Florian Overkamp wrote:
Hi,
I've been playing with Voicemail and Voicemail2 a bit for my users, and
there are a few things I'm wondering about:
- We can specify parameters to the mailbox (s, b or u) to select which
prompts to play.
Hey Jan,
On Wed, Jul 16, 2003 at 11:45:13 -0700, Jan Rychter wrote:
Hi,
I'm running asterisk in the following setup
Phone - WX100USB - * - Internet - * - WX100P - PSTN
The two Asterisks talk to each other via IAX2 and use GSM for voice.
This seems to work quite well except for
Hey Jay,
On Tue, Jul 15, 2003 at 18:41:12 +1000, Jay Tyndall wrote:
Hi,
When I use the analog phone connected to Zap/1 how do I transfer hold the
caller ?
When I hit the flash key, all that happens is the caller hears a beep
(sounds like DTMF).
But no stutter dial tone on the Zap/1
Yo BK,
On Sat, Jul 12, 2003 at 11:52:42 +0900, BK [address only for mailing lists] wrote:
Hi
since callerid= in sip.conf doesn't set the Caller ID, I suppose it
must be there for some other reason.
Is this a not-yet-working feature for future releases of Asterisk?
If not, what does
Hi bk,
On Wed, Jul 09, 2003 at 17:16:55 +0900, BK [address only for mailing lists] wrote:
Hi
in order to keep the dial tone after pressing 9 for 'outside line' I
have this in my extensions.conf
[localpstn]
ignorepat = 9
exten = _9[123456789]XXX,1,Dial,${PSTN}/${EXTEN:1}
exten =
Hey all,
I have an Asterisk-box with an E100P and a PRI (Euro-ISDN) coming
into it from a Meridian-switch. The incoming numbers on this PRI all start
with the same digit and the last part of the dialled number is signalled to
Asterisk digit by digit, until Asterisk signals that the number is
that analog channels have ?
With PRI you're going to always receive the full number that was dialed.
So since you have a limited number of DID's do the matching one for one
and it'll work.
regards
Martin
On Wed, 9 Jul 2003, The Traveller wrote:
Hey all,
I have an Asterisk-box with an E100P
Hey Adam,
On Tue, Jul 08, 2003 at 00:58:08 +1000, Adam Goryachev wrote:
Without quite just saying me too, see below...
[...]
My setup is a dual PIII-750 with 1 gig of RAM, with an X100P, connected
to an analog line to my telco, and a TDM40P with analog phones
I have an AMD XP 1800 with
Hey Dan,
On Mon, Jul 07, 2003 at 18:47:07 +0300, Dan wrote:
Hi,
There is any possibility to dial a specific extension and then enter in your
own mailbox (the one defined for that specific SIP phone) without asking for
the exxtension number but only for the password?
Sure. Pass the
Hey Dan,
On Mon, Jul 07, 2003 at 20:42:16 +0300, Dan wrote:
It is possible to have a time stamp in the recorded message? I want to
know
when the message has been recorded.
I think someone here was working on a patch for that, which was waiting
for prompts to be recorded. Not sure
Hi Jim,
You're probably not receiving disconnect-supervision on your analog
lines, or have Zaptel configured incorrectly to recognize it. Check
the list-archives (available from www.asterisk.org). You could try the
busydetect-statement in zapata.conf. Also check Asterisk's main Makefile
for
Yo all,
As there has been some intrest, here's my updated version:
I post it to -dev as well as -users, as it may be of intrest to
both.
Inspired by the example in the tips tricks-section of
http://www.junghanns.net/asterisk/;, I built a more elaborate
set of features. Currently, my
Hey Mark,
I wouldn't make MEC3 the default just yet. I was just testing with
MeetMe, which was one of the things where MEC3 went wrong for me in the
past. After around 8 channels from an E100P-connected PRI joined the
conference, everything became one big chaos of noise. When enough
channels
Hey Matt,
On Sun, Jun 29, 2003 at 14:14:18 -1000, Matt Darnell wrote:
Aloha Oliver,
That was it! Thanks, I am going to download the eStara softphone and try to
talk from one phone to another!
Nice that it worked. An alternative method might be putting the line
noload = chan_zap.so into
Yo Dan,
Try adding the s to the arguments you give to VoiceMail2, so, for
example, Voicemail2(sb1000) for the busy-message of ext. 1000.
Note that only Voicemail2 allows the s to be used together with
b and u.
Grtz,
Oliver
On Fri, Jun 27, 2003 at 15:04:44 +0300, Dan wrote:
are interested in, though. The boring parts are all
half-done, and the interesting parts are all fairly high-quality.
Matthias
The Traveller [EMAIL PROTECTED] writes:
Hi Alex,
The problem is most likely to occur with high volumes of call-setups and
disconnects. This could be reproduced
24 18:23:25 mspgate03 kernel: [c0109023]
Jun 24 18:23:25 mspgate03 kernel:
Thank you.
Alex Zarubin
-Original Message-
From: The Traveller [mailto:[EMAIL PROTECTED]
Sent: Tuesday, June 17, 2003 3:10 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Dual T400P, SMP
Yo all,
The PSTN access-number for the Dutch IAXTel - PSTN-gateway has changed.
The new number is: +31 20 3987 567. Calling from IAXTel to Dutch
toll-free PSTN-numbers is still done in the same way, by calling
31800rest of number.
Mark: Could you please update your web-sites to reflect this
number...
Thanks
Andy
*** REPLY SEPARATOR ***
On 26/06/2003 at 21:55 The Traveller wrote:
Yo all,
The PSTN access-number for the Dutch IAXTel - PSTN-gateway has changed.
The new number is: +31 20 3987 567. Calling from IAXTel to Dutch
toll-free PSTN-numbers
Heya Mark (and others),
Here's an update on my adventures while trying to debug the Zaptel-related
panics, as discussed on this list a while back.
While debugging the problem, I completely swapped the machine for an
entirely different model (Supermicro dual Xeon 2.4GHz with 2Gb of RAM),
put the
Yo Iain,
On Tue, Jun 17, 2003 at 21:48:34 +0100, Iain McWilliams wrote:
Hi,
I'm trying to get asterisk running on kernel 2.4.20 however trawling through
the archives I've found a few references to patches to remove i4l's dtmf
detection, but have been unable to find the patch itself (I
Yo Martin,
On Tue, Jun 17, 2003 at 17:03:15 -0500, Martin Pycko wrote:
Hello,
I've commited the new busydetect routine to CVS.
You need to cvs update asterisk of course and then choose it
in asterisk/Makefile and recompile asterisk.
[...]
It fails to compile here (Redhat 9, gcc version
Yo,
I've seen very similar Zaptel-related freezes on a wide variety of
mainboards (SMP as well as non-SMP), with X100P's as well as with an E100P.
At some point, almost always at the moment a call through one of those cards
connects or disconnects, the machine completely stops responding and
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