I was just looking through the SER archives and someone mentioned using
SER with *, his comments about * where very complementary, is there
anyone using SER on this list. I could do with a bit of a HOWTO.
--
Dave Cotton [EMAIL PROTECTED]
___
The sort answer is no. The ztdummy code is written specifically for
usb-uhci and usb-ohci operates in an entirely different way. However,
there is an alternative to ztdummy that uses the real-time clock. Take a
look at zaprtc from here http://www.junghanns.net/asterisk/page1.html
Iain
I get a username/password prompt when I go to that page...
This is what Brian West at Thu, Jul 31, 2003 at 12:03:53AM -0500 wrote:
http://store.yahoo.com/grandstream-networks-inc/products.html
I think that will clear it up.
On Wed, 30 Jul 2003, Ricardo Villa wrote:
I was quoted $75 and
Everyone does now.. I don't get it.. they have a product we want.. but
they wont or can't sell it. Guess they can't keep up with demand right
now.
bkw
On Sat, 2 Aug 2003, Steven Honson wrote:
I get a username/password prompt when I go to that page...
This is what Brian West at Thu, Jul 31,
Here's a patch that changes the behaviour of # transfers in asterisk. A
single # is transferred to the remote phone/system. Two # in quick
succession will trigger a transfer. This is very useful for users who have
basic analogue phones connected to an ATA 186. For example, when calling a
This is a very useful patch. One of the biggest problems I've had with
another service (ComminuKate) that uses # as a control tone is that I
can't use any IVR system that needs # as a termination character.
On Sat, 2003-08-02 at 10:33, Iain Stevenson wrote:
Here's a patch that changes the
How would you go about doing something like this? Add yet another
variable to app_dial to hold the last called party number?
We could make app_dial return with 0 given a specific option.
Mark
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I'll be happy to merge the unified patch, but I'll need at least a
disclaimer from the original author.
http://www.digium.com/disclaim.changes
http://www.digium.com/disclaimer.txt
Either is fine. Fax to +1-256-971-6890 and then e-mail me the finalized
patch off-list.
Mark
On 31 Jul 2003,
Hi-
Asterisk (CVS 7/30/03) seems to leave my AGI processes (written in Perl) as
zombies, even though they exit normally with exit(0). I am running Red
Hat 9.
I tried the same AGI etc with an older CVS (7/1/03) and this does not
happen.
I think a zombie process is a process that doesn't
This is a known problem. I have the same situation with RH9 as you do.
I don't know if the problem has been added to the new bug tracking
system. We should check.
My workaround is to run the AGI scripts on a RH7 box and forward calls
using IAX.
Scott Stingel wrote:
Hi-
Asterisk (CVS
Thanks - I have added it to the new bug system. Its important to me as I
have 8 E1's to be installed within days - every call on each channel causes
my AGI routine to be called. Not really practical to drop back to an older
RedHat for me!
Scott
-Original Message-
From: [EMAIL
Unfortunately, I've found several problems with Asterisk running on
RedHat 9. (Most of my problems only happened under high call volume.)
For that reason, we've rolled back to RedHat 8 on all of our servers.
It's worked great for us.
Jared Smith
On Sat, 2003-08-02 at 12:55, Scott Stingel
On Sat, 2003-08-02 at 13:04, Daryl Jones wrote:
What other problems are you having with RH9?
When we had a lot of concurrent calls (the magic number seems to be
somewhere around 70), asterisk would stop accepting new calls. If I
remember correctly, Mark tried to debug the problem, but had
The calls are not getting passed: Agent 308 is logged in and idle.
Yet I have a customer that has been on hold for over 8 minutes.
Is this duplicatable? Perhaps I can ssh in and take a look. Agents and
queues have been in heavy development the last few weeks.
Mark
This *should* already be fixed.
Mark
On Fri, 1 Aug 2003, Adam Donnison wrote:
I actually found this same thing, and traced it down to
app_dial.c line 190. It doesn't explicitly check for
a valid chan before trying to use it and it segfaults when
it does a strlen on a chan entity. I simply
Can you make it die without the thread debugging turned on? If so, I can
ssh in and find the deadlock.
Mark
On Thu, 31 Jul 2003, Alex Zarubin wrote:
Hello,
CVS 07/31/03. Test with 130+ PSTN-to-SIP calls. Asterisk gets locked ...
grep -e Error -e eventually p-console
chan_sip.c line 1453
On Sat, 2003-08-02 at 14:20, Scott Stingel wrote:
Yes, this concerns me too, as I'm about to install 2 big systems in Europe.
How hard is it to roll-back to RH 8? I've only got a couple days to make
this all work in house before I have to debug in the field.
If you weren't under the gun for
I have noticed that there have been changes such as AddQueueMember.
Maybe this is why? Has the old AgentLogin app been deprecated?
John
On Saturday, August 2, 2003, at 03:30 PM, Mark Spencer wrote:
The calls are not getting passed: Agent 308 is logged in and idle.
Yet I have a customer that
I might do it anyway!
I do think that RH9's installation tools are quite convenient - I guess I'll
find out if Debian's are ok.
Looking for more feedback: *** Is anyone using RedHat in a high call volume
scenario?? ***
Thanks
Scott
Scott M. Stingel
Emerging Voice Technology Inc.
Palo Alto,
First off, there was a bug that caused this problem that has been fixed
about two days ago. If your CVS is older than that, update.
Generally, be sure to run asterisk with the -g flag so it dumps core,
then do (in /usr/src/asterisk):
gdb ./asterisk /path/to/core.pid
(gdb) bt
and send me the
You can turn on Asterisk's internal malloc debugging in the Makefile.
That can help track down the problem.
Mark
On Fri, 1 Aug 2003, Roy Sigurd Karlsbakk wrote:
hi all
seems there's a memory leak in asterisk somewhere. probably in chan_mgcp. It
seems like 2048kB is allocated but not
Need to be sure the ISDN device can send DTMF. Can't it be done by just
returning -1 in ast_digit or something?
Mrak
On Fri, 1 Aug 2003, Stefano Finetti wrote:
Hello,
I'm trying to understand why when I make a call from a SIP phone to an
external number who has an IVR system in which I've
On Fri, 2003-08-01 at 16:45, Dan wrote:
#include stdlib.h
#include stdio.h
int main() {
int ret;
ret = system(/bin/ls /dev/null);
printf(system(\/bin/ls /dev/null\) returned %d\n, ret);
return(ret);
}
gcc mysystem.c -o mysystem
I'm currently working on a perl script convert csv logs to a http log
equiv:
LogFormat %h %l %u %t \%r\ %s %b \%{Referer}i\ \%{User-agent}i\
Right now I have output something similar to this:
111 - - [02/Aug/2003:16:39:15 -0500] GET /300 HTTP/1.0 200 6144 sipext ANSWERED
INCOMING - -
I noticed a few issues with app_queue just wanted to know if its sip
related or ata186 related:
Ext 111 and Ext 112 are dynamically loged into the queue via
AddQueueMember.
Call hits queue with fewestcalls routing.
Rings ext 111 if 111 doesn't answer. It rings ext 112. If for some
reason ext
Typically you use AbsoluteTimeout app.
Martin
On Sat, 2 Aug 2003 [EMAIL PROTECTED] wrote:
hi everybody,
can anybody pls tell me a way to hangup an answered Zap/SIP/IAX call, after a
specified time period
expires, like after 10, 15 minutes.
Surajee
--This mail sent
I have a x100p card that has call waiting on the line comming into it and
then into *. is there any way i can use call waiting on that line?
Michael
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Somebody with redhat 9 and this problem, find me on irc (#asterisk,
irc.freenode.net) and let me login and try to debug it. That will make it
easier by far.
Mark
On Sat, 2 Aug 2003, Scott Stingel wrote:
Yes, this concerns me too, as I'm about to install 2 big systems in Europe.
How hard is
I was having problems getting my DG104S going after I updated the boot and
runtime firmware. The D-Link support folks told me to do a factory
reset. I did
and then set my IP settings and it worked fine.
My settings (right or wrong, but working):
mgcp.conf:
[dg104s]
host = 192.168.1.89
context
Well when you use your phone line and you hear the call waiting sound you
can press flash on your phone and then *0 and that will generate the flash
on your phone line. This switch to the incoming call.
regards
Martin
On Sat, 2 Aug 2003, lists wrote:
I have a x100p card that has call waiting
I have figured out that its a problem in app_queue, could be the
interaction between chan_sip and app_queue. Or the ATA is on crack.
in chan_sip if I change
case 501: /* Not Implemented */
if (owner)
ast_queue_control(p-owner, AST_CONTROL_CONGESTION, 0);
break;
to:
case 501: /* Not
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