On Wed, Jun 03, 2009 at 09:18:34AM -0500, Danny Nicholas wrote:
I will agree with most of what you say about the polycoms, but here is one
thing to help you out. The Polycom 501 has an HTTP configuration interface;
I set up mine thru http://127.0.0.1:8088/asterisk/static/phones where
Julien Chavanton schrieb:
I need to be able to play different files when calling different numbers
dynamicaly from the dial-plan, this solution will still require the creation
of multiple static entry in musiconhold.conf
For example, if it was possible to specify one file from the Dial
Hi,
Asterisk does not post CDR when dial status is CHANUNAVAIL.
Can someone tell me what are the conditions under which CDR is not posted?
Thanks
Jim
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asterisk-users mailing
Cary Fitch wrote:
We have a bunch of SNOM 360’s we are not using. I agree they are not
intuitive to the user. They work ok in general. I would part with 15
or so at an attractive price, one or more,
I like the Grandstream 2000 series. Easy to use, easy to set up, good
web page.
Hi All
we are using Asterisk 1.4.21 users having analog phone connected
with Audio codes g/w Mp124.
How we can put caller on hold when we receive call on Analog phone
(Panasonic). Any dial plan application or feature.conf need to use for this.
Audio code g/w having option which we
On Thu, 4 Jun 2009, Rob Hillis wrote:
Cary Fitch wrote:
We have a bunch of SNOM 360ÿÿs we are not using. I agree they are not
intuitive to the user. They work ok in general. I would part with 15
or so at an attractive price, one or more,
I like the Grandstream 2000 series. Easy to use,
Right! Whatever somebody likes more! I just say that the Snoms look
better at the side of my Mac. Wich is of course by far the superiour
system. ;-)
Chris
John Novack schrieb:
Hasn't this religious argument/discussion gone on long enough??
zoach...@securax.org wrote:
I personally find
Hello all,
I have a problem when I try to install FastAGI.
I try to do
#perl Makefile.PEL
And it return :
Can't locate inc/Module/Install.pm in @INC (@INC contains: /etc/perl
/usr/local/lib/perl/5.8.8 /usr/local/share/perl/5.8.8 /usr/lib/perl5
/usr/share/perl5 /usr/lib/perl/5.8
Manoj Panicker - FOES schrieb:
However I can always call any one pre-configured PSTN number using the
call forwarding feature, however I should be able to use my sogtphone
and dial a PSTN number using the integration which is not happening
today.
As far as I know the FritzBox only supports
On Thu, Jun 04, 2009 at 02:00:53PM +0200, BERGANZ François wrote:
I have a problem when I try to install FastAGI.
I try to do
#perl Makefile.PEL
And it return :
Can't locate inc/Module/Install.pm in @INC (@INC contains: /etc/perl
/usr/local/lib/perl/5.8.8
Hello!
I have a 64 bit Asterisk system and am wondering how to use Digium's 32 bit
fax driver. Is there some kind of emulation that can be used?
Thanks!
Elliot
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asterisk-users
Elliot Murdock wrote:
I have a 64 bit Asterisk system and am wondering how to use Digium's 32
bit fax driver. Is there some kind of emulation that can be used?
Unfortunately the only option for you is to build your entire Asterisk
system (core and all modules) as 32-bit modules to be able to
Do a blind transfer to a parking lot. #1700 would be the default for this.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of amit salunkhe
Sent: Thursday, June 04, 2009 4:27 AM
To: asterisk-users@lists.digium.com
Subject:
Joseph L. Casale schrieb:
I have a single server running asterisk 1.6.0.8 with a few sip voip providers
and a tdm card for redundancy. It has a caching name server and the sip
providers
are hard coded in the hosts file.
When the internet connection dies, it fails over to the dahdi channel
On Thu, 4 Jun 2009, Philipp Kempgen wrote:
Joseph L. Casale schrieb:
I have a single server running asterisk 1.6.0.8 with a few sip voip providers
and a tdm card for redundancy. It has a caching name server and the sip
providers
are hard coded in the hosts file.
When the internet
You could call an AGI to read a database based on the caller/extension and
play any number of MOH files.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Philipp
Kempgen
Sent: Thursday, June 04, 2009 1:54 AM
To:
That would be correct, since the TFTP/HTTP setup is a world-readable one.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen
Sent: Thursday, June 04, 2009 1:00 AM
To: asterisk-users@lists.digium.com
There was a nice post earlier this week about timing out a meetme
conference. You could combine that information with an AGI to monitor the
meetme room and kick out after the timeout if the user count did not change.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Elliot Murdock wrote:
Hello!
I have a 64 bit Asterisk system and am wondering how to use Digium's
32 bit fax driver. Is there some kind of emulation that can be used?
Thanks!
Elliot
Use the FAX support built into Asterisk 1.6 and you won't have that
limitation.
Steve
A persistent local DNS cache such as pdnsd[1] or djbdns[2] could help.
[1] http://en.wikipedia.org/wiki/Pdnsd
[2] http://en.wikipedia.org/wiki/Djbdns
Philipp Kempgen
I am guessing it fails to reverse lookup your internal addresses (which
would fail anyway, even with the DNS up). If your
Un-top-posting...
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kurian
Thayil Sent: Wednesday, June 03, 2009 11:22 PM
I am looking for an option in Meetme or similar which will enable to
skip to next priority (a voicemail) if the person in Meetme conference
is alone and
On Thu, 4 Jun 2009, Joseph L. Casale wrote:
A persistent local DNS cache such as pdnsd[1] or djbdns[2] could help.
[1] http://en.wikipedia.org/wiki/Pdnsd
[2] http://en.wikipedia.org/wiki/Djbdns
Philipp Kempgen
I am guessing it fails to reverse lookup your internal addresses (which
Hi all! Im not sure if it is the correct place but, Ive five boxes running
asterisk and each one with his own cdr mysql database. What Im looking for
is to get a core CDR system that holds information stored on each asterisk
server. Have you any suggestion/process to accomplish that?. Thanks!!!
Do you want a live repository or just a common gathering of the data? If
LR then you should set up a deamon on each box to transfer records as they
occur using something like the DBI functionality of PERL. If not, then just
do a mysql dump periodically and ssh the files to the common server.
On Thu, 4 Jun 2009, Danny Nicholas wrote:
Do you want a live repository or just a common gathering of the data? If
LR then you should set up a deamon on each box to transfer records as they
occur using something like the DBI functionality of PERL. If not, then just
do a mysql dump
Sure. I might name it something like dhcp127 though.
That makes sense :)
This must be your dialplan. Can you post it?
You are right, never trust users :) They had erased it or something,
it actually wasn't there. So it does go straight to vm as it should.
My bad...
Close. The packets won't
On Thu, Jun 4, 2009 at 6:42 PM, Jeff LaCoursiere j...@jeff.net wrote:
On Thu, 4 Jun 2009, Danny Nicholas wrote:
Do you want a live repository or just a common gathering of the data? If
LR then you should set up a deamon on each box to transfer records as they
occur using something like the
Gustavo A Gonzalez escribió:
Hi all! I’m not sure if it is the correct place but, I’ve five boxes running
asterisk and each one with his own cdr mysql database. What Im looking for
is to get a core CDR system that holds information stored on each asterisk
server. Have you any
Hi, we are experiencing a strange issue and I am hoping someone can point me
to the right direction or help out with some pointers.
We have asterisk 1.6.0.6 with a sangome a104DE card. We have basically 4
T1's for a total of DAHDI 96 channels.
We have an agi application (php) that acts as a kind
This all sounds very nice and do-able, but doesn't this sound like a
high-odds scenario for creating a single point-of-failure especially if the
5 machines are all creating a high volume of calls?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
You are keeping in mind that each call is actually two calls (SIP - CM --
CM - Asterisk/T1)?
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Deepak
Sent: Thursday, June 04, 2009 11:15 AM
To: Asterisk Users Mailing List -
On Thu, Jun 4, 2009 at 12:15 PM, Deepak dlal...@gmail.com wrote:
Hi, we are experiencing a strange issue and I am hoping someone can point me
to the right direction or help out with some pointers.
We have asterisk 1.6.0.6 with a sangome a104DE card. We have basically 4
T1's for a total of
On Thu, 4 Jun 2009, Deepak wrote:
Hi, we are experiencing a strange issue and I am hoping someone can point me
to the right direction or help out with some pointers.
We have asterisk 1.6.0.6 with a sangome a104DE card. We have basically 4
T1's for a total of DAHDI 96 channels.
We have an
On Thu, 4 Jun 2009, Deepak wrote:
Hi, we are experiencing a strange issue and I am hoping someone can
point me to the right direction or help out with some pointers.
We have asterisk 1.6.0.6 with a sangome a104DE card. We have basically 4
T1's for a total of DAHDI 96 channels.
We have an
Thanks. You are right in assumng that we query the database. I was not
aware that there is a limit to the number of DB connections to mysql. We
open/close db connections as needed. I will check if there is such a limit
and if yes, post the result.
Would you happen to know where to configure such
BTW,we are using an ODBC connection to Microsoft SQL Server.
We are not using MySQL.
Would that be a possible cause?
Thanks
On Thu, Jun 4, 2009 at 12:31 PM, David Backeberg dbackeb...@gmail.comwrote:
On Thu, Jun 4, 2009 at 12:15 PM, Deepak dlal...@gmail.com wrote:
Hi, we are experiencing
On Thu, 4 Jun 2009, Deepak wrote:
Thanks. You are right in assumng that we query the database. I was not
aware that there is a limit to the number of DB connections to mysql. We
open/close db connections as needed. I will check if there is such a
limit and if yes, post the result.
Would
On Thu, 4 Jun 2009, Deepak wrote:
BTW,we are using an ODBC connection to Microsoft SQL Server.
We are not using MySQL.
Would that be a possible cause?
Good lord who designed this mess? Start over.
j
___
-- Bandwidth and Colocation Provided by
Un-top-posting...
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gustavo A
Gonzalez Sent: Thursday, June 04, 2009 10:23 AM
Hi all! I?m not sure if it is the correct place but, I?ve five boxes
running asterisk and each one with his own cdr mysql database. What Im
looking for
AMEN!!! - How you got that far is nothing short of miraculous. Was Moses
your designer?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
LaCoursiere
Sent: Thursday, June 04, 2009 12:23 PM
To: Asterisk
Deepak schrieb:
Thanks. You are right in assumng that we query the database. I was not
aware that there is a limit to the number of DB connections to mysql.
We open/close db connections as needed. I will check if there is such a
limit and if yes, post the result.
Would you happen to know
Hi,
I have a serious problem with Asterisk 1.4.18.
Every so often, usually after Asterisk has been running for a few days
consistently, phones start dropping registrations.
However, when this happens, doing a sip show peer on those
extensions shows them as OK.
Therefore, I have no way to tell this
On Thu, Jun 4, 2009 at 1:18 PM, Deepak dlal...@gmail.com wrote:
BTW,we are using an ODBC connection to Microsoft SQL Server.
We are not using MySQL.
Would that be a possible cause?
No idea what your settings are for the SQL server, but you should
certainly take a look there.
I don't know how
Hi gang,
Since I'm getting no joy from device_Status or SIPPEER in
1.4.26-rc1, I thought I would do an AGI to read my hints and check for line
in use that way. The AGI works fine from a prompt, but returns the dreaded
utils.c:966 ast_carefulwrite: write() returned error: Broken
Not a real solution, but why don't you just set up a cron job to issue
asterisk -rx restart when convenient once a day? This will restart
asterisk on the first zero load opportunity.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Danny Nicholas schrieb:
The AGI works fine from a prompt, but returns the dreaded
utils.c:966 ast_carefulwrite: write() returned error: Broken pipe when I
try to run it from the dialplan.
Sounds like
$SIG{'PIPE'} = 'IGNORE';
is what you need.
http://perldoc.perl.org/perlipc.html#Signals
Hi!
I have a serious problem with Asterisk 1.4.18.
Every so often, usually after Asterisk has been running for a few days
consistently, phones start dropping registrations. However, when this
happens, doing a sip show peer on those extensions shows them as
OK.
Please check if this is
There were some serious issues with some of the earlier 1.4.x Asterisk
releases. You say it's a production server and can't upgrade because of
that. That is the one reason why you should upgrade. There are
security risks with certain versions and some serious bugs that were
fixed. While I
On Thu, 4 Jun 2009, Danny Nicholas wrote:
Hi gang,
It's not a gang -- it's a club :)
Since I'm getting no joy from device_Status or SIPPEER in
1.4.26-rc1, I thought I would do an AGI to read my hints and check for
line in use that way. The AGI works fine from a prompt, but
I agree when people say that your problem is at the database and not
Asterisk/AGI. Another thing came into my mind, thought: you're using G729 to
your SIP peers, and that means Asterisk is doing all the transcoding.
How much idle CPU % top reports at high load? If this is too high (like 90%)
Tzafrir Cohen escribió:
On Mon, Jun 01, 2009 at 09:23:48AM -0500, Miguel Molina wrote:
Hi all,
I just upgraded a production server to asterisk 1.4.25, compiling with
the following:
[*] 1. DONT_OPTIMIZE
[*] 2.
On Thu, 4 Jun 2009, Deepak wrote:
BTW,we are using an ODBC connection to Microsoft SQL Server. We are not
using MySQL.
Would that be a possible cause?
Probably not the cause, unless there is a bug in ODBC.
But, it does multiply your support costs.
Not because of the obvious you have to
Michael Graves bounced this to me this morning - it looks interesting
as a possible device for which an Asterisk channel driver could be
written:
http://www.redorbit.com/news/technology/1699391/rtx_releases_dectcatiq_20_usb_dongle/index.html?source=r_technology
DECT really is a nice
Here's what I got from agi debug:
agi debug
AGI Debugging Enabled
*CLI AGI Tx agi_request: hintcheck.agi
AGI Tx agi_channel: SIP/170-081c4ab8
AGI Tx agi_language: en
AGI Tx agi_type: SIP
AGI Tx agi_uniqueid: 1244140421.2
AGI Tx agi_callerid: 170
AGI Tx agi_calleridname: Danny Nicholas
AGI
It was not a conscious decision to use MS-SQL. We were forced to use
MS-SQL since the business rules were on MS-SQL and other apps are using
it.
If you think it is not ODBC, then what in your opinion is causing the issue?
I though you did mention that in your opinion database might be the
Are you using freetds? Is it a recent version or something that is a few years
old? Are you using a freetds connection pool?
http://www.freetds.org/userguide/tdspool.htm
Is unixODBC pooling turned on? See: http://www.unixodbc.org/doc/conn_pool.html
to understand why this doesn't work with
BTW,we are using an ODBC connection to Microsoft SQL Server.
We are not using MySQL.
Would that be a possible cause?
Good lord who designed this mess? Start over.
I feel the cold evil darkness spread through our lands..
___
-- Bandwidth
On Thu, 4 Jun 2009, Deepak wrote:
It was not a conscious decision to use MS-SQL. We were forced to use
MS-SQL since the business rules were on MS-SQL and other apps are using
it.
If you think it is not ODBC, then what in your opinion is causing the issue?
I though you did mention that in
Is there a limitation to the number of variables you can set from a PHP agi
script? I have a simple example and I can't get it to let me set more than
1. I am pretty sure I am just missing something, but I've searched all over
an can't find the answer. Here is the extensions.conf part:
exten =
On Thu, 4 Jun 2009, Peder wrote:
Is there a limitation to the number of variables you can set from a PHP agi
script? I have a simple example and I can't get it to let me set more than
1. I am pretty sure I am just missing something, but I've searched all over
an can't find the answer.
so who's writing the channel driver for it ?
Martin
On Thu, Jun 4, 2009 at 2:26 PM, John Todd jt...@digium.com wrote:
Michael Graves bounced this to me this morning - it looks interesting
as a possible device for which an Asterisk channel driver could be
written:
On Thu, 4 Jun 2009, Peder wrote:
Is there a limitation to the number of variables you can set from a PHP
agi script?
Not that I've found yet :)
One of my AGIs sets almost 600 channel variables.
It (written in C) takes all of 1/100 of a second to lookup the variables
from 2 tables in a
What are the steps to build in 32 bit in a 64 bit OS?
On Thu, Jun 4, 2009 at 10:03 AM, Steve Underwood ste...@coppice.org wrote:
Elliot Murdock wrote:
Hello!
I have a 64 bit Asterisk system and am wondering how to use Digium's
32 bit fax driver. Is there some kind of emulation that can
On Thu, 4 Jun 2009, Danny Nicholas wrote:
Here's what I got from agi debug:
agi debug
AGI Debugging Enabled
*CLI AGI Tx agi_request: hintcheck.agi
[snip]
AGI Rx SET VARIABLE LINESTAT=Idle
AGI Tx 200 result=1
[Jun 4 13:33:42] ERROR[28261]: utils.c:979 ast_carefulwrite: write()
Hi!
so who's writing the channel driver for it ?
What for? Unless you can exploit some cool new CAT-iq future features
(which exactly?) it is easier to buy a SIP-enabled DECT base station. No
need to worry about another channel driver.
Philipp
On Fri, 05 Jun 2009 02:11:19 +0200, Philipp von Klitzing wrote:
Hi!
so who's writing the channel driver for it ?
What for? Unless you can exploit some cool new CAT-iq future features
(which exactly?) it is easier to buy a SIP-enabled DECT base station. No
need to worry about another channel
First, the scenarios:
Call placed from Boston to locally configured Asterisk Meetme extension:
Cisco 7941 -SCCP- Cisco 2821(CME,Boston) -SIP- Asterisk(Boston)
Call placed from Boston to European Asterisk Meetme extension:
Cisco 7941 -SCCP- Cisco 2821(CME,Boston) -SIP- Cisco
2821(CME,Europe)
On Thu, Jun 4, 2009 at 9:34 PM, Phillip Heller phel...@me.com wrote:
Cisco 7941 -SCCP- Cisco 2821(CME,Boston) -SIP- Cisco
2821(CME,Europe) -SIP- Asterisk(Boston)
debugging enabled on Asterisk, I see that I often get duplicate DTMF
entries. So where I might have dialed 1234#, Asterisk sees
So the PBX in Europe has a local extension and DID configured for the
Asterisk MeetMe such that users in Europe have a local number to
dial
Placing the call from Boston to the European extension is only to
duplicate and hopefully solve the problem.
When I came aboard this company, it
On Thu, Jun 4, 2009 at 10:17 PM, Phillip Heller phel...@me.com wrote:
I have tooled around with the various dtmf-relay options, though to no
positive effect. I'll keep playing with it tomorrow. If you happen
to think of anything else, I certainly appreciate the input.
If you haven't already
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi Tilghman and Grygoriy.
Tilghman Lesher escribió:
I was testing both the recall key and uncomment the following lines
in the features.conf file:
blindxfer = #1
atxfer = *2
verifying previously that the extension uses the arguments tT with
- Original Message -
From: Trevor Hammonds
To: varun.rape...@spectross.com ; Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Sunday, May 31, 2009 8:57 AM
Subject: Re: [asterisk-users] Understanding Call Handling In Asterisk
On May 29, 2009, Varun Rapelly
- Original Message -
From: Trevor Hammonds
To: varun.rape...@spectross.com ; Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Sunday, May 31, 2009 8:57 AM
Subject: Re: [asterisk-users] Understanding Call Handling In Asterisk
On May 29, 2009, Varun Rapelly
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