That should do it, it tells Asterisk to override the contact field which
includes the private IP, and use the public IP and port it received the
packet from instead.
Try a 'sip debug peer peer' and see what it is coming in as.
From: [EMAIL PROTECTED]
Hi list;
ASTCC supports IVR or there is a separate module for
IVR?
Can someone advise me a link to start download and
ready about ASTCC to do the configuration?
Regards,
-
ITS
IP Telephony and Contact Center Engineer
Bilal Ghayad
Mobile: 00865 9849460
Hi,
I am trying to SUBSCRIBE for message waiting indications to asterisk,
it sends 200 OK but contact header is missing(it is mandatory since
subscribe is dialog establishing method), due to which parsing fails, any
body knows about this issue...?
Regards,
Subramanya
Hi,
Where can I find some tools, such as ztmonitor for zaptel devices, to adjust
rxgain and txgain correctly on this card?
I've some troubles with finding the optimal configuration for the
echocancellator.
Thanks in advance,
Stefano Arata
That's a good note about MySQL replication. You can use it to remove
point-of-failure which currently is your DB server.
Regards/Pagarbiai,
Mindaugas Kezys
http://www.kolmisoft.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James Collier
Sent:
Dear all
I want to configure 2 port E1 card on my asterisk so which
version is best 1.2.x or 1.4.x can anyone suggest me which one is best right
now for asterisk and anyone have configuration file to configure E1 card and
zaptel.conf so i can configure it
Thanks for that Arnaud, saw it myself this morning, but the download
link takes me to a page not found cisco page :( I've reported it on
their broken links page...
Adrian Marsh
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Arnaud
Ligot
Sent: 22
Asterisk is an excellent PBX system, and makes a very good endpoint in
the SIP chain for all sorts of things -- IVR systems, voicemail
applications, automated messages, etc.
It has an extremely well-written CDR engine, so many people mesh it with
billing applications to produce accurate
Hi,
Hi,
I am trying to SUBSCRIBE for message waiting indications to asterisk,
it sends 200 OK but contact header is missing(it is mandatory since
subscribe is dialog establishing method), due to which parsing fails and
also expires is 0 in the 200 OK any body knows about these issue...?
Offers them? Yes. Offers them in a clean, friendly, usable package? Not
so much yet.
SEMS has raw capability, but if you want it to do many of the things
Asterisk can do, you need to know how to code that yourself, or you're
going to be digging about the code for documentation on features
Hi,
I am using debian 4.0 with version 2.6.18-4-686
I have downloaded the required files form site
asterisk-1.2.24.tar.gz
libmfcr2-0.0.3-1.4.tar.bz2
libsupertone-0.0.2.tar.gz
Hi,
I am using debian 4.0 with version 2.6.18-4-686
I have downloaded the required files form site
asterisk-1.2.24.tar.gz
libmfcr2-0.0.3-1.4.tar.bz2
libsupertone-0.0.2.tar.gz
Hi,
I am using debian 4.0 with version 2.6.18-4-686
I have downloaded the required files form site
asterisk-1.2.24.tar.gz
libmfcr2-0.0.3-1.4.tar.bz2
libsupertone-0.0.2.tar.gz
Hello,I have succeded in compiling and configuring My TDM Card and asterisk,
all works fine. But I have a problem using the PHP Agi.The CLI tells me that
when I call my number :-- Starting simple switch on 'Zap/4-1'-- Executing
[EMAIL PROTECTED]:1] Answer(Zap/4-1, ) in new stack--
Hello,
Is it possible to print the Asterisk message logs to a file, or is this already
done? By message logs I mean the display that shows up on the asterisk server
when a call is made from one user to another. I believe if the verbosity is
high, it can show what parts of the extension.conf
Hey there,
The file must be accessible by the process
calling upon it. Try a chown using the Asterisk process user name.
Cheers,
Louis-Eric
At 09:51 AM 8/23/2007, Karim H wrote:
Hello,
I have succeded in compiling and configuring My
TDM Card and asterisk, all works fine.
But I have a
Did you confirm that the file exists?
/var/lib/asterisk/agi-bin/rabot.agi
Also, in your script (wherever it actually is), put a space between php
and -q
#!/usr/bin/php -q
Karim H wrote:
Hello,
I have succeded in compiling and configuring My TDM Card and asterisk,
all works fine.
But I
Steve Kennedy wrote:
On Wed, Aug 22, 2007 at 01:19:14PM -0500, Asterisk Development Team wrote:
The Asterisk.org development team has announced the release of Zaptel
versions 1.2.20.1 and 1.4.5.1. These releases are to correct an error in
the install target in the Makefile of the 1.4.5 and
On Thu, 2007-08-23 at 20:16 +0530, [EMAIL PROTECTED]
wrote:
Hi,
I am using debian 4.0 with version 2.6.18-4-686
I have downloaded the required files form site
asterisk-1.2.24.tar.gz
libmfcr2-0.0.3-1.4.tar.bz2
Problem resolved :Two things :1 - php agi instead of php-cli (I have apache
running on my server too and it edits himself all the php.ini it finds)2 -
Error using php-q it is amistake : php -q (of course =) )Their is a real
problem with asterisk concerning errors in agi script.If there is an
On Thu, 2007-08-23 at 11:14 -0400, [EMAIL PROTECTED] wrote:
Is it possible to print the Asterisk message logs to a file, or is
this already done?
You want to look at the logger.conf configuration file, and see how your
Asterisk system is set to log the various types of information (such as
Hi,
im using asterisk-1.2.24 and zaptel-1.2.20, im having a problem running
meetme conference,
when i try to call meetme i get this from the asterisk console
Aug 24 00:14:12 WARNING[15466]: pbx.c:1720 pbx_extension_helper: No
application 'MeetMe' for extension (sample, 65000, 1)
i recompiled
Hi,
I have GCE4019VOIP IP phone with me. Can anybody tell me the steps
how to use it for communication in the LAN with other sip phones. I want
help
from the IP phone side as I have already done it with SIP soft
phone...
Thanks and Regards,
Sanchal
I'm finally migrating from the 6/7/05 CVS version to 1.4. Had quite a
run, I have to admit. Asterisk itself only segfaulted once or twice,
but the dns issues have been bothering me. And the box just needs to
go. Everything is going on a Ubuntu 6.06TLS server, that's been
perfectly stable.
Jay Milk wrote:
I'm finally migrating from the 6/7/05 CVS version to 1.4. Had quite a
run, I have to admit. Asterisk itself only segfaulted once or twice,
but the dns issues have been bothering me. And the box just needs to
go. Everything is going on a Ubuntu 6.06TLS server, that's been
Thanks for your reply. I have previously looked at the logger.conf file. I
see that the various types of information can be logged in different ways.
After setting the various information types with whatever I want logged, is it
possible to save the actual logs to a file (ie: As the
On Thursday 23 August 2007 12:46, [EMAIL PROTECTED] wrote:
Thanks for your reply. I have previously looked at the logger.conf file.
I see that the various types of information can be logged in different
ways. After setting the various information types with whatever I want
logged, is it
[EMAIL PROTECTED] wrote:
Hello,
Is it possible to print the Asterisk message logs to a file, or is this
already done? By message logs I mean the display that shows up on the
asterisk server when a call is made from one user to another. I believe if
the verbosity is high, it can show
[EMAIL PROTECTED] wrote:
Thanks for your reply. I have previously looked at the logger.conf file. I
see that the various types of information can be logged in different ways.
After setting the various information types with whatever I want logged, is
it possible to save the actual logs
Jason Parker wrote:
Jay Milk wrote:
I'm finally migrating from the 6/7/05 CVS version to 1.4. Had quite a
run, I have to admit. Asterisk itself only segfaulted once or twice,
but the dns issues have been bothering me. And the box just needs to
go. Everything is going on a Ubuntu
Jay Milk wrote:
I'm finally migrating from the 6/7/05 CVS version to 1.4. Had quite a
run, I have to admit. Asterisk itself only segfaulted once or twice,
but the dns issues have been bothering me. And the box just needs to
go. Everything is going on a Ubuntu 6.06TLS server, that's been
On Thu, Aug 23, 2007 at 01:07:36PM -0400, Ron Joffe wrote:
On Thursday 23 August 2007 12:46, [EMAIL PROTECTED] wrote:
Thanks for your reply. I have previously looked at the logger.conf file.
I see that the various types of information can be logged in different
ways. After setting the
Yes, any output from the console logs. I tried viewing the full file and it
looks like it's what I was looking for. Thanks for the help.
Denis
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Brian Jones
Sent: Thursday, August 23, 2007 1:11 PM
To:
On Fri, Aug 24, 2007 at 12:16:24AM +0800, Mark Quitoriano wrote:
Hi,
im using asterisk-1.2.24 and zaptel-1.2.20, im having a problem running
meetme conference,
when i try to call meetme i get this from the asterisk console
Aug 24 00:14:12 WARNING[15466]: pbx.c:1720 pbx_extension_helper:
im having a strange problem related to call-limit for peers. well im not
sure if its related to call-limmit or not. Bottom line is:
I call a user A, from user B. user B hears silence, untill it goes to
voicemail. when user B hangsup. user B's call limit is reset to 0 but user
A's call limit is
On Thursday 23 August 2007 14:00, Tzafrir Cohen wrote:
I utilize this command:
nohup script -f -c asterisk -vvvTn /tmp/asterisk.log
To start up my apps. This will log everything to a log file.
Why nohup? And if you have nohup, why script?
It will log everything until the
The problem with any of these choices is that they do not address
logfile rotation.
Try this:
http://cr.yp.to/daemontools.html
Ron Joffe wrote:
On Thursday 23 August 2007 14:00, Tzafrir Cohen wrote:
I utilize this command:
nohup script -f -c asterisk -vvvTn /tmp/asterisk.log
To
You can configure logger.conf so that it will log just about everything you
could want.
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+logger.conf
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de
[EMAIL PROTECTED]
Enviado el: jueves, 23 de
Hi List;
I saw this is written in that link:
http://www.voip-info.org/wiki/view/Asterisk+options
And really I was not able to understand for what is
that and where I can learn about it and how to write
such thing? Can some one advise me?
!/bin/bash
asterisk Startup script for the asterisk
Hi List;
I read the following sentence:
The CLI prompt is set with the ASTERISK_PROMPT UNIX
environment variable
In the following link:
http://www.voip-info.org/wiki/index.php
page=Asterisk+CLI+prompt
The question is: what is the ASTERISK_PROMPT UNIX
environment variable and where I can
On Thu, Aug 23, 2007 at 12:43:15PM -0600, Anthony Francis wrote:
The problem with any of these choices is that they do not address
logfile rotation.
Because this can be done with the standard system logrotate, or even by
asterisk (if you trust it to that). Decently-packaged Asterisk comes
with
On 8/23/07, Mark Quitoriano [EMAIL PROTECTED] wrote:
Hi,
im using asterisk-1.2.24 and zaptel-1.2.20, im having a problem running
meetme conference,
when i try to call meetme i get this from the asterisk console
Aug 24 00:14:12 WARNING[15466]: pbx.c:1720 pbx_extension_helper: No
I'm not sure what features/variables you can use, or where to find
information about that, but what this basically means is you can change
your CLI prompt by this:
export ASTERISK_PROMPT=new prompt
then, what you access the CLI, instead of:
hostname*CLI
you get
new prompt
Moj
bilal
Ben Dinnerville wrote:
The problem occurs when we have external (pstn) calls coming into / out
of the system (via an iax trunk), in which case we have no control over
frame size, as well as occurring with handsets directly connected to the
system.
Please contact Digium Support to work
From memory, your zaptel and zapata files look ok. signalling for an
FXO module would be FXS, and vice versa. As far as I can tell, you're
ok there.
Now, it's the FXO card that plugs into the phone line. The FXS card
gets a phone hooked up to it. Dialing the phone would be
Hi, folks.
I've been on the Asterisk Announce list for a while now, and it seems
to me that the release versions of Asterisk are a bit bleeding-edge.
They qualify as stable, but I wouldn't call them production stable
since half the time a new one comes out, a fix for it comes out the
next
I've had requests to processes incoming voicemails with voice recognition
routine and add the output text to the body of the email message from * with
the attached .wav file. Has anyone implemented this type of feature and
willing to share some notes?
Thanks!
Ryan M. Colbert
Director of
Show version from the CLI.
Ed Pastore wrote:
Hi, folks.
I've been on the Asterisk Announce list for a while now, and it seems
to me that the release versions of Asterisk are a bit bleeding-edge.
They qualify as stable, but I wouldn't call them production stable
since half the time a
Ed Pastore wrote:
since half the time a new one comes out, a fix for it comes out the
next day.
So... that said, what's a good version to linger on? I don't *need*
Until 1.4 improves, I'm staying with 1.2
I do know that I'm running some version of 1.2, and am also not sure
if I
On 8/23/07, Ed Pastore [EMAIL PROTECTED] wrote:
Hi, folks.
I've been on the Asterisk Announce list for a while now, and it seems
to me that the release versions of Asterisk are a bit bleeding-edge.
They qualify as stable, but I wouldn't call them production stable
since half the time a new
On 8/23/07, Ryan M. Colbert [EMAIL PROTECTED] wrote:
I've had requests to processes incoming voicemails with voice recognition
routine and add the output text to the body of the email message from * with
the attached .wav file. Has anyone implemented this type of feature and
willing to
If it is posible for a imcoming call to ring both the Polycom desk
phone and my cell phone at the same time, if I dont answer fall back
to my voice mail box.
I would like to hire someone to cofigure that for me.
Bob
--
We've Got Your Name at http://www.mail.com !
Get a FREE E-mail
1- I've tried running fxotune
2- I've tried turning off all un-necessary hardware in the BIOS
3- I've tried on a different PCI slot.
4- I've tried these suggestions:
http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting
5- How I check if it the clicking and popping correlates to
dial(SIP/polycom-on-my-deskLocal/5551212,15,tr)
as an example.
Bob Gibson wrote:
If it is posible for a imcoming call to ring both the Polycom desk
phone and my cell phone at the same time, if I dont answer fall
back to my voice mail box.
I would like to hire someone to
Dear Mojo;
Thanks for your help.
Why you said export ASTERISK_PROMPT=new prompt ?
Regards
Bilal
I'm not sure what features/variables you can use, or
where to find
information about that, but what this basically means
is you can change
your CLI prompt by this:
export ASTERISK_PROMPT=new
David Gomillion wrote:
On 8/23/07, *Ed Pastore* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Hi, folks.
I've been on the Asterisk Announce list for a while now, and it seems
to me that the release versions of Asterisk are a bit bleeding-edge.
They qualify as stable,
On 8/24/07, ram [EMAIL PROTECTED] wrote:
On 8/23/07, Mark Quitoriano [EMAIL PROTECTED] wrote:
Hi,
im using asterisk-1.2.24 and zaptel-1.2.20, im having a problem running
meetme conference,
when i try to call meetme i get this from the asterisk console
Aug 24 00:14:12
Can i use asterisk as a softswitch?
___
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
I have a PAP2 with 2 phone ports.
When I call them everything works fine until I hung up the call. There
is about 30-40 seconds until I can call to that extension again.
Before that it gives me busy messages.
Extension config:
exten = 199,1,Dial(SIP/199,30)
exten = 199,102,Hangup
Any
Steve Totaro wrote:
David Gomillion wrote:
On 8/23/07, *Ed Pastore* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Hi, folks.
I've been on the Asterisk Announce list for a while now, and it seems
to me that the release versions of Asterisk are a bit bleeding-edge.
They
I tried asking in another thread this week, but I'm not sure people saw
the actual subject of the question. Does anyone know where to find
documentation of xPL, the home automation interface? Specifically for
integrating it with Asterisk. xPL is part of Trixbox, so it's being
used, but
Probably.
PaulH
On Fri, 2007-08-24 at 06:55 +0800, Mark Quitoriano wrote:
Can i use asterisk as a softswitch?
___
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asterisk-users mailing list
To UNSUBSCRIBE or update options
Probably.
PaulH
On Fri, 2007-08-24 at 06:55 +0800, Mark Quitoriano wrote:
Can i use asterisk as a softswitch?
It is really a soft switch. Not a good one for high carrier class telco
usage I'd say but just fine for office PBX replacement, which is what it was
designed for. What it is
Mark Quitoriano wrote:
Can i use asterisk as a softswitch?
This thread has been discussed over and over. Search the archives,
there are more thoughts and opinions there than you probably have time
or desire to read.
Thanks,
Steve Totaro
___
Yes you could, but asterisk was designed to be a PBX. I would not use it as
soft switch due its limitations. It really depends on how much traffic you
are going to be passing.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: Friday, 24
Edit logger.conf and learn how to enable debugging,verbose and all
kind of messages. Enable all levels of messages, try again and tell us
what is the error message exactly.
Regards,
On 8/23/07, [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
Hi,
I am using debian 4.0 with version
Ryan M. Colbert wrote:
I’ve had requests to processes incoming voicemails with voice
recognition routine and add the output text to the body of the email
message from * with the attached .wav file. Has anyone implemented this
type of feature and willing to share some notes?
I seem to recall
Anthony Francis wrote:
dial(SIP/polycom-on-my-deskLocal/5551212,15,tr)
Will this work even if the Local is pointing to a Zap channel?
As far as I know, this only works with SIP or IAX outgoing.
-Stephen-
___
--Bandwidth and Colocation Provided by
Clayton Milos wrote:
It is really a soft switch. Not a good one for high carrier class telco
usage I'd say but just fine for office PBX replacement, which is what it was
designed for. What it is missing AFAIK that a carrier class switch has to
have is a SS7 stack. The world's TDM exchanges
On Thursday 23 August 2007 11:22:23 pm Stephen Bosch wrote:
dial(SIP/polycom-on-my-deskLocal/5551212,15,tr)
Will this work even if the Local is pointing to a Zap channel?
As far as I know, this only works with SIP or IAX outgoing.
I'm not sure where you are getting that assumption from, as I
Matthew Rubenstein wrote:
I tried asking in another thread this week, but I'm not sure people saw
the actual subject of the question. Does anyone know where to find
documentation of xPL, the home automation interface? Specifically for
integrating it with Asterisk. xPL is part of Trixbox,
On Tue, 21 Aug 2007 09:50:57 -0400
Dave Fullerton [EMAIL PROTECTED] wrote:
Zane C.B. wrote:
On Tue, 21 Aug 2007 07:33:23 +0530
Vidura Senadeera [EMAIL PROTECTED] wrote:
Dear All,
I would like to get community's feedback with regard to RAID1
( Software or Hardware) implementations
On Wed, 22 Aug 2007 12:37:26 -0600
Stephen Bosch [EMAIL PROTECTED] wrote:
Zane C.B. wrote:
1: Software RAID on Linux is way less than impressive. Plus last
a I checked Linux can't handle mirroring a entire disk. Last I
looked at it around a year ago you were limited to only mirroring
Nuance offers an SDK to do something similar, I think they say you can
only expect between 45-60% accuracy using it though. Total cost is
about $6K to $8K for one server license.
If there are enough people interested in pooling money I'd be willing
to help set up a system to process voicemails
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