On Sun, 11 Apr 2004 23:28:05 -0500
Andrew D Kirch [EMAIL PROTECTED] wrote:
Apologies if any of these have already been fixed in the working
version Page 6 third and fourth lines from the bottom Digium is the
solely capable should read Digium is solely capable
--
Andrew D Kirch |
If you're using MGCP, we need your help. There's a patch in bugs.digium.com that
needs testing by the community. Please spend some time testing and adding your
comments to the bug tracker.
The author writes:
--
I'm trying to make work Asterisk against a Cisco IAD2431 with
On Mon, 2004-04-12 at 03:39, Anon wrote:
On Sunday 11 April 2004 07:18 pm, Todd Lieberman wrote:
do a cat /proc/interupts
your should see your hardware showup.
OK...
cat /proc/interrupts
CPU0
0: 494600 XT-PIC timer
1: 5588 XT-PIC keyboard
Brian Cuthie wrote:
I've been using VoicePlus for a few days now, and overall I'm fairly
pleased. But one thing that truly scares me is that the drop-down
box on their site where you re-charge your account has values that go
all the way to $10,000.(!) I'm deathly afraid that one day in a
Paul Tyreman wrote:
du -sh /var/spool/asterisk/vm/*
At the command line, do
man du
You will have to know a bit about the operating system, this is not
point and click.
John Chapman
Yeah ok, I know I need to type it at the command prompt, I'm not
stupid.
I just wanted to know
Andrew Thompson wrote:
Brian Cuthie wrote:
I've been using VoicePlus for a few days now, and overall I'm fairly
pleased. But one thing that truly scares me is that the drop-down
box on their site where you re-charge your account has values that go
all the way to $10,000.(!) I'm deathly
Andreas,
The documentation you seek is on the Asterisk website and the Tiki.
If you jump on IRC, I'm sure there will be plenty of people around that can
answer questions, or for a small fee, perform your initial configuration for
you.
--
Troy Settle
Pulaski Networks
http://www.psknet.com
The latest recursive mutex additions in Asterisk will not compile on my
FreeBSD 4.9 systems. Anyone out there with FreeBSD or OpenBSD that got it working?
I guess I need an update to Gnu Pthreads, but can't find anything by looking
in ports. There's something called linuxthreads, but it's over my
On Sat, 2004-04-10 at 15:50, Thomas Gallaway wrote:
I run 4 X100P's in our asterisk box. Just make sure you give each card
it's own IRQ.
Paul,
Is the own IRQ per card a strict rule ? Becasue a I have a X100P +
TDM400P on a SMP PIII box, the X100P is sharing IRQ 11 with usb-uhci and
On Sat, 2004-04-10 at 15:50, Thomas Gallaway wrote:
I run 4 X100P's in our asterisk box. Just make sure you give each card
it's own IRQ.
Paul,
Is the own IRQ per card a strict rule ? Becasue a I have a X100P +
TDM400P on a SMP PIII box, the X100P is sharing IRQ 11 with usb-uhci and
no
Greg,
You're going through what I went through last year, and I feel your pain.
I started by mixing * with Lucent, but that turned into a nightmare. It was
a twisted and convoluted setup to transfer calls to more than a couple VOIP
extensions, or for callers to dial anyone's extension
My PRI is being reset at least once a day with the following errors in the
logs.
zaptel, zapata, libpri, and asterisk are from CVS this morning.. This has
been happening for weeks on all versions (including -stable).
the T100P card appears to NOT be sharing an IRQ.
xenon# cat /proc/interupts
On Mon, 2004-04-12 at 11:28 -0400, Troy Settle wrote:
At this point, I'm using straight Asterisk, with a a PSTN gateway at a data
POP passing calls via IAX to my PBX here in the office.
Who is the PSTN gateway provider?
The only CLEC around here that is seriously considering any sort of VoIP
This doesn't appear to be a load issue, since normally in that case I would
expect you would get a lot of (usually harmless) frame reject messages in
your /var/log/asterisk/messages file, and perhaps an occasional
missed/double interrupt message on the console.
I wonder if there have been new
I have been seeing this for over a month, and blaming it on our generally
incompetent telco, so it's definately not a new issue.
On Mon, 12 Apr 2004, Scott Stingel wrote:
This doesn't appear to be a load issue, since normally in that case I would
expect you would get a lot of (usually
How do you setup the timing in Meetme conference? I have a x100p and tdm4x card.
When I dialing to my conference I get a request to schedule in the past error message.
thanks
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent:
Mike-
Do you have access to any kind of PRI test set, like a T-Bird or something.
A red alarm would be easy to capture I imagine - it would be nice to confirm
that it's a problem particular to your site. The reason I mention software
is that I've noticed a lot of other messages regarding these
Re: Memory:
The cool thing is Linux can just discard the cached entries when a
application needs real RAM. Don't worry about your RAM usage until you
see swap climbing and/or the buffers and cache dropping down to near
zero.
Yes, yes, I knew about caches versus HD access, but I didn't realize
Is anyone selling asterisks systems??
Just wanting to know if it's profitable to try and start selling them.
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Is there an easy/cheap way to add g.723 to Asterisk? I have added g.729 and
need g.723.
Todd
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In article [EMAIL PROTECTED],
Hadar Pedhazur [EMAIL PROTECTED] wrote:
I have a system with no Digium hardware in it (two others with 2 X100P
cards in each of them as well). I'm interested in using MeetMe in the
one without the hardware (it works great in the ones with the
hardware). I can't
This was happening to me as well.
What finally fixed it was disabling echo cancelling on the X100P cards in
the zapata.conf files.
However this resulted in a horrid echo on my cisco phone when I was using a
line attached via one of my x100P cards.
So I went back and re-enabled echo cancelling
Hey all,
Quick Question. I have heard mention that Asterisk has
the capability to store voicemail inside a database, instead of storing each
voicemail in a separate file under a spool directory. Is this true?
If so, does it (or can it) use MySQL? Is there any
documentation available
try asterisk-biz
http://lists.digium.com/mailman/listinfo/asterisk-biz
On 12-Apr-04, at 10:19 AM, James Moran wrote:
Is anyone selling asterisks systems??
Just wanting to know if it's profitable to try and start selling them.
___
Asterisk-Users
I'm running Zaptel CVS from April 8, LibPRI CVS April 8, and v-1.0 CVS April 7. With
dual T400P cards with no PRI errors at all. Possibly something driver/config related?
Are you timing from your PRI? I remember getting some PRI errors when my timing
config was hosed. Could you post your
What about using NFS or AFS for this?
--Ernest
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Darren
NaySent: Monday, April 12, 2004 10:35 AMTo:
[EMAIL PROTECTED]Subject: [Asterisk-Users] Voicemail
storage in DB
Hey
all,
Quick Question. I
Is it at all possible?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling
Sent: Monday, April 12, 2004 12:31 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] G.723
Todd Wallace wrote:
Is there an easy/cheap way to add g.723 to Asterisk?
Hi,
I am trying to install chan_capi, with asterisk (cvs) on Suse 9.0, but I
get the following error:
==
linux:/usr/src/chan_capi-0.3.1 # make
gcc -pipe -Wall -Wmissing-prototypes -Wmissing-declarations -g
-I/usr/include/asterisk -D_REENTRANT
-D_GNU_SOURCE -O6 -march=i686 -DCAPI_ES
http://www.voip-info.org/wiki-Asterisk+codecs
G.723.1 can only be used in pass-thru mode.
-Original Message-
From: Todd Wallace [mailto:[EMAIL PROTECTED]
Sent: Monday, April 12, 2004 12:48 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] G.723
Is it at all possible?
/etc/zaptel.conf:
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24
loadzone = us
defaultzone=us
On Mon, 12 Apr 2004, Bisker, Scott (7805) wrote:
I'm running Zaptel CVS from April 8, LibPRI CVS April 8, and v-1.0 CVS April 7.
With dual T400P cards with no PRI errors at all. Possibly something
Todd Wallace wrote:
Is it at all possible?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling
Sent: Monday, April 12, 2004 12:31 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] G.723
Todd Wallace wrote:
Is there an easy/cheap way to
On Mon, 2004-04-12 at 12:47, Todd Wallace wrote:
Is it at all possible?
Technically it is possible, financially/legally it isn't. Search the
archives for exact problems with patent holders in the technology and
why they won't talk to you for less than a huge stack of bills.
-Original
WipeOut == WipeOut [EMAIL PROTECTED] writes:
WipeOut Have you thought of mounting the spool directolr on an NFS
WipeOut file server ... [I am] not sure if there would be any file
WipeOut locking issues..
Yes, there would be. This is the same issue as using nfs mail spools
with maildir style
Any help with this will be greatly appreciated. When re-compiling * to
include voicemail access from a MySQL database, I recieve the follwing
error. Anybody know how I can fix this? Am I missing packages somewhere?
app_voicemail.c:44:25: mysql/mysql.h: No such file or directory
In file included
Are there any known problems converting dtmf from oob over iax2 to
inband over rtp/ulaw?
Obviously it works when converting to inband over pri/ulaw et al,
but how about rtp?
I've got packet traces that confirm that 2833 packets are properly
generated when I have 2833 configured for the rtp link,
There's more than a little irony here, given that one of their products
is called Email Blaster.
-brian
Linus Surguy wrote:
[I'm sorry to trouble the list with this, but this is the only way I know to
contact the person concerned]
This message is for Stephen Karrington - it appears that you
Hi All,
In theory if I do this;
exten = s,1,Zapateller(nocallerid)
exten = s,2,Privacymanager
exten = s,3,Dial(a bunch of SIP extensions)
My callers should only hear the anti-telemarketing tones if they call from
a line that has no caller*ID and then get offered an opportunity to enter
it,
If I remember correctly (and I could be wrong) I think you have to
answer the line first...
exten = s,1,Answer
exten = s,2,Zapateller(nocallerid)
exten = s,3,Privacymanager
exten = s,4,Dial(a bunch of SIP extensions)
-Original Message-
From: Mark Phillips [mailto:[EMAIL PROTECTED]
Or you could specify the answer in the Zapateller line like:
...
exten = s,2,Zapateller(answer|nocallerid)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sean Cheesman
Sent: Monday, April 12, 2004 12:55 PM
To: [EMAIL PROTECTED]
Subject: RE:
WipeOut == WipeOut [EMAIL PROTECTED] writes:
WipeOut Have you thought of mounting the spool directolr on an NFS
WipeOut file server ... [I am] not sure if there would be any file
WipeOut locking issues..
Yes, there would be. This is the same issue as using nfs mail spools
with
Hello,
Email Blaster is spam proof and permission only. Its just the name
that gets people excited :)
Sincerely,
Stephen Karrington
Dreamtime.net Inc.
http://www.dreamtime.net
http://www.emailblaster.us
Corporate Office
101 California Street, 22nd Floor
San Francisco, CA 94111-5802
Voice -
I have set up a new SwissVoice phone and it can receive calls but I cannot
make calls out from it. The setup is simple for now, 2 phones: SwissVoice
is ext 7726 and Cisco 7960 (SIP) is ext 7999.
I can call from the Cisco phone and it rings on the SwissVoice phone but
when I dial from the
Much snipping along the way :-)
Tony Mountifield wrote:
Actually, I have used Zaprtc quite successfully. The only reason you
have to disable kernel RTC support is because Zaprtc is actually a
*replacement* for the standard RTC module. It provides the same
facilities, but includes extra parts for
Hi all,
Is it possible to dial anOUTSIDESIP address while inside AGI application? For example, within extension context, I could use
[from-sip]
exten = 7723,1,Dial(SIP/[EMAIL PROTECTED]) and this works
whereas when I'm inside agi app,
$AGI-exec('Dial',"SIP/[EMAIL PROTECTED]") and thisDOESN'T
On Thu, 2004-04-08 at 08:54, Martin Schenkelberg wrote:
Thank you problem solved.
I tried to use the (R) Button on my phone to place call on HOLD but Asterisk
says something of PRI Error : Dont know how to post-handle message of Tye
HOLD (36)
Is this feature not implemented in Bri-Stuff
Hello all,
Just a quick note, I have been putting together a TAPI
driver for Asterisk, this enables the user to perform things like click to dial
from any TAPI enabled app (such as outlook or ACT etc). At the moment it is
very basic and can only perform click to dial but further
On Mon, 2004-04-12 at 17:37, Tony Mountifield wrote:
Zaprtc is actually a *replacement* for the standard RTC module.
It provides the same
facilities, but includes extra parts for Zaptel use.
-SNIP-
All very interesting, thankyou :)
The zaprtc.c code is based on the rtc.c from 2.4.20. I am
Is it possible for asterisk to do an sql query in order to
get the member list of a call queue?
thanks
micko
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On Mon, 12 Apr 2004, Mark Phillips wrote:
I tried,
exten = s,1,Zapateller(answer|nocallerid)
exten = s,2,Privacymanager
exten = s,3,Dial(a bunch of SIP extensions)
But then every call was answered regardless of CID and the tones were heard.
I tried the sample I found at:
James == James H Thompson [EMAIL PROTECTED] writes:
James You could use NFS with the Maildir alogrithm or
James something similar to avoid the need for locking.
Here is an(other) idea if anyone is looking for a project:
When using a db for the meta data, there is no need for the filenames
to
Does anyone know of Phone that supports G.723 on H.323.
Todd
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I am looking to install a web interface for Asterisk to transfer calls and look who's
on the phone. If anybody has a working web interface please let me know. I installed
the www.asternic.com (operator)
But when I bring up my web browser it says transferring data and does not bring a
browser.
We have implement a IVR with CTD+CDR and queues, all on asterisk. Just for
statistics, we need to save the time which one client was waiting in queue.
Someone knows asterisk has a function than it can be load in a module
(programming by us) or if a module with this function was developtment.
On Monday 12 April 2004 13:29, AJ Grinnell wrote:
Any help with this will be greatly appreciated. When re-compiling *
to include voicemail access from a MySQL database, I recieve the
follwing error. Anybody know how I can fix this? Am I missing
packages somewhere?
app_voicemail.c:44:25:
[EMAIL PROTECTED] zaptel]# modprobe ztdummy
WARNING: Error inserting zaptel
(/lib/modules/2.6.5-1.315/misc/zaptel.ko): Invalid module format
WARNING: Error inserting zaptel
(/lib/modules/2.6.5-1.315/misc/zaptel.ko): Invalid module format FATAL:
Error inserting ztdummy
Would it make any sense to store the voice mail formatted as a email msg in a Maildir
directory
structure.
Then you could also retreive them with an email client.
Jim
James H. Thompson
[EMAIL PROTECTED]
- Original Message -
From: James H. Cloos Jr. [EMAIL PROTECTED]
To: [EMAIL
On Mon, 12 Apr 2004 18:19:51 -0500
Andrew D Kirch [EMAIL PROTECTED] wrote:
[EMAIL PROTECTED] zaptel]# modprobe ztdummy
WARNING: Error inserting zaptel
(/lib/modules/2.6.5-1.315/misc/zaptel.ko): Invalid module format
WARNING: Error inserting zaptel
(/lib/modules/2.6.5-1.315/misc/zaptel.ko):
Hi,
Anyone get the Immix C3-FXO Sip gateway to work with asterisk. I have it
working for outbound calls but cant get it to work for inbound calls. The
unit has an built-in greeting and it keeps picking up the call. Cant find
the command to turn it off and set it to forward the calls to asterisk.
On Mon, 12 Apr 2004, Ron McMillin wrote:
Is it possible to dial an OUTSIDE SIP address while inside AGI application? For
example, within extension context, I could use
[from-sip]
exten = 7723,1,Dial(SIP/[EMAIL PROTECTED]) and this works
whereas when I'm inside agi app,
ztdummy will not work on a 2.6 kernel.
The 2.6 series USB core is a complete re-write. I looked through
the source, and while not a kernel hacker, I doubt ztdummy can
be easily made to work with the new USB core.
Dan
-Original Message-
From: Andrew D Kirch [mailto:[EMAIL PROTECTED]
Hi Akk,
If this has been discussed/done then apologies be-4-hand. I
did not find it in the Wiki or the Archives. Here's the
question.
We have incoming PRI lines, all on the same main number. An
attendant is supposed to handle all incoming calls. Now,
let's say I have a multi-line SIP phone.
Are you using the more than one manager application? I.E. op_panel etc...
Reason I ask, is that I had two copies of op_panel running and lost a couple
of calls, but with just one running things have ben fine...
Matt Riddell
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL
Daniel Cubero Salas, Ing. wrote:
We have implement a IVR with CTD+CDR and queues, all on asterisk. Just for
statistics, we need to save the time which one client was waiting in queue.
Someone knows asterisk has a function than it can be load in a module
(programming by us) or if a module with
I was looking at the webmin module but does not seem complete, any ideas ?
kevin
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Matt Riddell wrote:
Are you using the more than one manager application? I.E. op_panel etc...
Reason I ask, is that I had two copies of op_panel running and lost a couple
of calls, but with just one running things have ben fine...
Matt Riddell
- Original Message -
From: [EMAIL
Eric Wieling wrote:
[EMAIL PROTECTED] wrote:
Yes turning off echo cancelling would be fatal. We have some serious
echo going on here that I can not seem to track done. I am assuming
it is just this old building. Maybe we can go ISDN or so but the
dropped calls are rather bad.
Rnadom
Firstly, let me just say I am new to asterisk and if anything I've said
is covered in an FAQ or in previous posts I apologise but I have tried
searching and I've attempted a few of the things I found but they didn't
help.
Has anybody got any experience using an X100P on an NTL phone line in
Another observation of something which doesn't work:
exten = 3200,1,Dial(SIP/3200,20,tTr)
exten = 3200,2,Playback(tt-weasels)
exten = 3200,3,Hangup
exten = 3200,102,Dial(SIP/3201,20,tTr)
exten = 3200,103,Playback(tt-weasels)
exten = 3200,104,Hangup
exten = 3200,203,Dial(SIP/3202,20,tTr)
exten =
On Apr 12, 2004, at 4:34 PM, Dan Austin wrote:
ztdummy will not work on a 2.6 kernel.
The 2.6 series USB core is a complete re-write. I looked through
the source, and while not a kernel hacker, I doubt ztdummy can
be easily made to work with the new USB core.
Since the system clock ticks at 1
X100P hangup detection works only sporadically in the US as
well.
Probably a bug in hardware and/or software. Not sure. It
just does not work for a lot of people.
WW
- Original Message Follows -
Firstly, let me just say I am new to asterisk and if
anything I've said is covered in an
- Original Message -
From: Thomas Gallaway
-big snip -
| I am just running 1 instance of the op_panel. But today I noticed that 2
| calls got ended after
| 2 minutes 34 seconds. I will disable the management thing just for testes.
| ___
I can't
Outside of Asterisk - is their anything a linux admin can do to optimize
or speed network traffic to/from the pbx to sip phones?
I'm looking for some options in /proc
Since Asterisks is a network bound/sensetive app I'd start their.
I'm running RH9. Optimizations for other linux and unix
Hello:
I'll Like to now how to insert a pause on a SIP string. I have a ATA 186 and
a FXS = FXO converter so I will like to program a extension that can be
dialed and it will dial the ATA extention, wait for dial tone and then dial
the phone number.
Thanks
Erick
On Apr 12, 2004, at 5:39 PM, Roger wrote:
Outside of Asterisk - is their anything a linux admin can do to
optimize or speed network traffic to/from the pbx to sip phones?
I'm looking for some options in /proc
Since Asterisks is a network bound/sensetive app I'd start their.
I'm running RH9.
Mike-
You didn't say what's at the other end of your PRI line, but you might try
having the other end be the timing sync source. Try: span=1,0,0,esf,b8zs
instead. Maybe that will help.
Regards
Scott Stingel
www.evtmedia.com
-Original Message-
From: [EMAIL PROTECTED]
Hello all,
I was wondering if anyone had used SendImage command?
And if so with what application, and what phone?
It sounds like an interesting command, but few phones support. (If any, as I
know of none.)
Thanks,
James Gardiner
___
Asterisk-Users
Holy crap people, trim your replies!
You didn't say what's at the other end of your PRI line, but you might try
having the other end be the timing sync source. Try: span=1,0,0,esf,b8zs
instead. Maybe that will help.
We need to get this documented *clearly* once and for all.
Zaptel T1/E1
Now you've got me utterly confused ...
So, in layman's terms, if I connect a T100P to a circuit
provided by the Telco, and the Telco says that they will
provide timing, I have to put WHAT?
span=1,0,0,esf,b8zs,yellow
this means '0' this span is not a sync source, i.e. the
Telco will provide my
Hi Steve,
You are a legend !!!It is properly disconnecting now.
Although I am having problem with the fax detection. Somehow fax will only
be detected if NO CLI was sent from the PSTN line but if the line received
the CLI, fax tone is ignored. Does anyone had the same experience? Is there
Matt Riddell wrote:
- Original Message -
From: Thomas Gallaway
-big snip -
| I am just running 1 instance of the op_panel. But today I noticed that 2
| calls got ended after
| 2 minutes 34 seconds. I will disable the management thing just for testes.
|
Yes, there could be a problem. The op_panel shows red and green balls!
Thomas Gallaway wrote:
Matt Riddell wrote:
- Original Message - From: Thomas Gallaway
-big snip -
| I am just running 1 instance of the op_panel. But today I noticed
that 2
| calls got ended after
| 2 minutes 34
James == James H Thompson [EMAIL PROTECTED] writes:
James Would it make any sense to store the voice mail formatted as a
James email msg in a Maildir directory structure. Then you could
James also retreive them with an email client.
That is not a bad idea. * would have to convert the mime
hi,
i write this looking for free conference room, i
checl code and don´t see any error but die at priority 7 if room 1001 have users
in
exten =
_1NXXNXX,1,RouteCall(${EXTEN})exten =
_1NXXNXX,2,GotoIf($[${DESTINATION1:0:3} = CONF]?3:13)exten =
_1NXXNXX,3,Setvar,var=0exten =
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