Hi,
I have 2 trunks connected to my asterisk installation.
One is a inbound connection between ericsson pbx and the other is thru a
voip service.
I am using 4 digit numbers both in ericsson and asterisk..
And also i have full real prefix for that numbers..
As all 290 are real numbers and if
2009/6/17 Oguzhan Kayhan oguzh...@bilkent.edu.tr
Hi,
I have 2 trunks connected to my asterisk installation.
One is a inbound connection between ericsson pbx and the other is thru a
voip service.
I am using 4 digit numbers both in ericsson and asterisk..
And also i have full real prefix
2009/6/15 Tilghman Lesher tilgh...@mail.jeffandtilghman.com
On Monday 15 June 2009 04:03:48 am Olivier wrote:
I've just discovered IMPORT function existence.
It can be use to import values from channel's Variable section but
unfortunately, il can't be use to access to values from Info
2009/6/17 Oguzhan Kayhan oguzh...@bilkent.edu.tr
Hi,
I have 2 trunks connected to my asterisk installation.
One is a inbound connection between ericsson pbx and the other is thru
a
voip service.
I am using 4 digit numbers both in ericsson and asterisk..
And also i have full real prefix
dear All
how can play mp3 file in dialplan with READ application which also play
file and collecting digits from user
and also i know about MP3Player but it cannot take digits
regards
Dhaval
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also sprach Jeff Peeler jpee...@digium.com [2009.06.16.1757 +0200]:
Have you set the parkedcallreparking, parkedcalltransfers, and other
associated options?
Only parkext and parkpos and context. All others are left at their
defaults. But of course this seems to be what I am looking for.
also sprach Danny Nicholas da...@debsinc.com [2009.06.16.1656 +0200]:
The problem is the Asterisk Read function. It is set to accept as
many 0-9 and * as you want to throw at it, then stop on # or
timeout. Unless you disable the # stops, you can't use # in
features. I would strongly caution
Hello, all. The little bit of reading I've done on lua makes me eager
to give it a try. However, when I try to install it (Asterisk 1.6.1.1
on CentOS 5.3), it is not available in menuselect. I have
installed lua
and lua-devel. I've seen very little about it in my Internet
searches.
Grygoriy Dobrovolskyy wrote:
2009/6/17 Oguzhan Kayhan oguzh...@bilkent.edu.tr
mailto:oguzh...@bilkent.edu.tr
Hi,
I have 2 trunks connected to my asterisk installation.
One is a inbound connection between ericsson pbx and the other is
thru a
voip service.
I
John Novack jnov...@stromberg-carlson.org writes:
I have wondered for years now why someone thought there needed to be two
different transfer functions.
Transfer should be ONE function. If one wants to speak first to the
object of the transfer, then stay until they answer, otherwise hang up
Danny Nicholas da...@debsinc.com writes:
In the B leg, check for the variable value instead of Callerid(num).
What is needed is something that actually sends a SIP request to the
phone to update the caller-ID on the display. Changing a variable won't
do that.
/Benny
DHAVAL INDRODIYA wrote:
how can play mp3 file in dialplan with READ application which also play
file and collecting digits from user
and also i know about MP3Player but it cannot take digits
You should convert the MP3 to a native format (ulaw, gsm, wav, etc...) and let
Asterisk play that
Hi,
Quick question to the real world.
Approx what specs would I need on server to handle 95 ZAP or Dahdi - SIP
gateway using G729 on the SIP to carrier side (nothing else, just media
conversion)?
Does the latest Asterisk/DAHDI significantly improve these numbers over say,
Asterisk 1.2.X?
Sure,
Jared Smith jsm...@digium.com writes:
Yes... that bug number spawned a *lot* of additional work for connected
party information (transmission, reception, and updates) that recently
went into the trunk of Asterisk. Those features will be available in
the 1.6.3 branch of Asterisk, once it has
On Wed, 17 Jun 2009, Steve Totaro wrote:
Hi,
Quick question to the real world.
Approx what specs would I need on server to handle 95 ZAP or Dahdi - SIP
gateway using G729 on the SIP to carrier side (nothing else, just media
conversion)?
Does the latest Asterisk/DAHDI significantly
Make sure you are actually setting it as:
Set(CALLERID(num)=290)
The previous poster has the formatting incorrect. If your callerID is a 4
digit
number, and you want to modify it to have the prefix on it before you send
it
back out, you can do:
Hi all,
Is it possible to display or print variables in Asterisk (e.g. in the
CLI) for debugging purposes?
For example, I'm using two different types of SIP phones: the Snom M3
and the Siemens S675IP. However, when anonymous callers submit a
number to the PrivacyManager, only the Siemens
This is what I do
Exten = s,1,Verbose(var is ${var})
The is to tell me that the variable is empty or to let me determine the
length.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jaap Winius
Sent:
On Wed, 2009-06-17 at 07:49 -0400, Watkins, Bradley wrote:
Hello, all. The little bit of reading I've done on lua makes me eager
to give it a try. However, when I try to install it (Asterisk 1.6.1.1
on CentOS 5.3), it is not available in menuselect. I have
installed lua
and
On 6/17/09, Gordon Henderson gordon+aster...@drogon.net wrote:
On Wed, 17 Jun 2009, Steve Totaro wrote:
Hi,
Quick question to the real world.
Approx what specs would I need on server to handle 95 ZAP or Dahdi - SIP
gateway using G729 on the SIP to carrier side (nothing else,
On Wed, 17 Jun 2009, Matt Florell wrote:
The TC400B is up to 120 channels of G729a now:
http://www.digium.com/en/products/voice/tc400b.php
I guess the UK disty hasn't updated their website yet then :)
Gordon
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On Wednesday 17 June 2009 08:38:19 John A. Sullivan III wrote:
On Wed, 2009-06-17 at 07:49 -0400, Watkins, Bradley wrote:
Hello, all. The little bit of reading I've done on lua makes me eager
to give it a try. However, when I try to install it (Asterisk 1.6.1.1
on CentOS 5.3), it is
On Wed, 2009-06-17 at 09:01 -0500, Tilghman Lesher wrote:
On Wednesday 17 June 2009 08:38:19 John A. Sullivan III wrote:
On Wed, 2009-06-17 at 07:49 -0400, Watkins, Bradley wrote:
Hello, all. The little bit of reading I've done on lua makes me eager
to give it a try. However, when I
Benny Amorsen wrote:
Does the patch use the non-standard Remote-Party-ID or the proper
P-Asserted-Identity?
That depends on the value specified for 'sendrpid' in sip.conf.
sendrpid=pai ; This will use P-Asserted-Identity
sendrpid=rpid ; This will use Remote-Party-ID (the default)
--
Sean
In the makeopts file change
LUA_INCLUDE=
To
LUA_INCLUDE=-I/usr/include
And do make again.
This should fix it.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John A.
Sullivan III
Sent: Wednesday, June 17,
On Wed, 2009-06-17 at 09:57 -0500, Danny Nicholas wrote:
In the makeopts file change
LUA_INCLUDE=
To
LUA_INCLUDE=-I/usr/include
And do make again.
This should fix it.
snip
Argh! alas it did not fix it:
LUA_INCLUDE=-I/usr/include
configure:42697: checking for luaL_newstate in -llua5.1
Does anybody know of a way to tell the Polycom phones to stop trying to
download their config? We have some setup for tftp and some for ftp and if
they cannot reach the server, they just keep rebooting over and over and
over and never stop. I would think it should try once or twice and stop,
but
John A. Sullivan III wrote:
Argh! alas it did not fix it:
Looks like a problem with how the various distros are packaging and installing
it. I've created a patch to configure that should find it on your system, give
it a whirl:
$ cd path/to/asterisk-src/
$ wget -O -
Touch the syncinfo.xml file with a future time. This should tell the phone
to stop polling.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Peder
Sent: Wednesday, June 17, 2009 10:27 AM
To: 'Asterisk Users
Alright I've been having an issue when trying to dial out locally when
coming from SIP. This used to work no problem, now it doesn't. Now the
local PRI to Bell Is working fine I have calls coming in and out of it
constantly right now. BUT if I try and make a local call from SIP (from
X-Lite or one
On Wed, 2009-06-17 at 11:42 -0400, Sean Bright wrote:
John A. Sullivan III wrote:
Argh! alas it did not fix it:
Looks like a problem with how the various distros are packaging and installing
it. I've created a patch to configure that should find it on your system,
give
it a whirl:
$
Wow! Definitely a non-trivial patch. Alas, it does not work but the
errors are different:
[compu...@pbx01 asterisk-1.6.1.1]$ grep -i lua config.log
configure:42697: checking for luaL_newstate in -llua5.1
configure:42732: gcc -o conftest -g -O2 conftest.c -llua5.15
/usr/bin/ld:
My guess is that when running the compile test ( This line:
'configure:42995: gcc -o conftest -g -O2 conftest.c
-llua-5.15'
) it is necessary to add '-lm' in order to link in the standard math
library.
- Brad
One more bit of magic necessary here, as pbx/pbx_lua.c has includes
John A. Sullivan III wrote:
Wow! Definitely a non-trivial patch. Alas, it does not work but the
errors are different:
I've updated the patch to take into account your feedback as well as Bradley's.
You'll need to revert the previous patch (this will probably involve unrolling
a fresh 1.6.1.1
Watkins, Bradley wrote:
One more bit of magic necessary here, as pbx/pbx_lua.c has includes for:
#include lua5.1/lua.h
#include lua5.1/lauxlib.h
#include lua5.1/lualib.h
On Redhat-based systems, it needs to be:
#include lua.h
#include lauxlib.h
#include lualib.h
Gah. OK. So the patch
On Wed, 2009-06-17 at 12:37 -0400, Sean Bright wrote:
John A. Sullivan III wrote:
Wow! Definitely a non-trivial patch. Alas, it does not work but the
errors are different:
I've updated the patch to take into account your feedback as well as
Bradley's.
You'll need to revert the
On Wed, 2009-06-17 at 12:44 -0400, Sean Bright wrote:
Watkins, Bradley wrote:
One more bit of magic necessary here, as pbx/pbx_lua.c has includes for:
#include lua5.1/lua.h
#include lua5.1/lauxlib.h
#include lua5.1/lualib.h
On Redhat-based systems, it needs to be:
#include lua.h
On Wed, 2009-06-17 at 12:56 -0400, John A. Sullivan III wrote:
On Wed, 2009-06-17 at 12:44 -0400, Sean Bright wrote:
Watkins, Bradley wrote:
One more bit of magic necessary here, as pbx/pbx_lua.c has includes for:
#include lua5.1/lua.h
#include lua5.1/lauxlib.h
#include
On Wed, 2009-06-17 at 13:05 -0400, John A. Sullivan III wrote:
On Wed, 2009-06-17 at 12:56 -0400, John A. Sullivan III wrote:
On Wed, 2009-06-17 at 12:44 -0400, Sean Bright wrote:
Watkins, Bradley wrote:
One more bit of magic necessary here, as pbx/pbx_lua.c has includes for:
Hi,
I have a 2/3 second gap between the end of a welcome message played with
Playback and the start of the Queue music. Here is the dialplan:
exten = ${EXTEN},1,NoOp($EXTEN)
exten = ${EXTEN},n,SIPAddHeader(Alert-Info: Ring_CCC)
exten =
That worked. The system is still in enough of a test phase
that I can
destroy it again and rebuild it if you'd like to send me a
new version
of the patch. Thanks - John
ARGH Not so good. Asterisk now segfaults on start up :((( - John
Now that is a behavior I'm not seeing,
If this is a recorded sound, you might want to truncate it with lame or
audacity. It is quite common in my shop as we record using the phones.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Louis-David
Hi
I am trying to implement monitoring of asterisk (all 4 spans-i want to show
them line by line Up or down) using nagios using below script, but i always get
the status as down and red..can anyone let me know how to read an output from
nagios plugin ? nagios etc is configured already and is
On Thu, Jun 18, 2009 at 12:04:26AM +0530, Sriram wrote:
Hi
I am trying to implement monitoring of asterisk (all 4 spans-i want to show
them line by line Up or down) using nagios using below script, but i always
get the status as down and red..can anyone let me know how to read an output
*FAILS= UP*
should be
*FAILS=UP*
(without the space)
it is a syntax error and if you test the script in console you will notice
it immediately.
On Wed, Jun 17, 2009 at 2:34 PM, Sriram d_r_sri...@hotmail.com wrote:
Hi
I am trying to implement monitoring of asterisk (all 4 spans-i want to
On Jun 17, 2009, at 8:16 AM, Steve Totaro wrote:
Hi,
Quick question to the real world.
Approx what specs would I need on server to handle 95 ZAP or Dahdi -
SIP gateway using G729 on the SIP to carrier side (nothing else,
just media conversion)?
Does the latest Asterisk/DAHDI
Hi All,
I'm having a bit of trouble with my new *NOW setup.
I've downloaded and installed *NOW 1.5. We're using 1 SIP Trunk from
SimpleSignal.com. Outbound calling works great, but I'm having some trouble
with inbound calls.
First, we would get the the number you have dialed is not in
On Wednesday 17 June 2009 11:56:28 John A. Sullivan III wrote:
On Wed, 2009-06-17 at 12:44 -0400, Sean Bright wrote:
Watkins, Bradley wrote:
One more bit of magic necessary here, as pbx/pbx_lua.c has includes
for: #include lua5.1/lua.h
#include lua5.1/lauxlib.h
#include
On Wednesday 17 June 2009 03:45:54 Olivier wrote:
2009/6/15 Tilghman Lesher tilgh...@mail.jeffandtilghman.com
On Monday 15 June 2009 04:03:48 am Olivier wrote:
I've just discovered IMPORT function existence.
It can be use to import values from channel's Variable section but
But It still needs to hit the server to see that at some point. I just want
it to stop pulling config totally, unless I tell it to. It is web based, so
I would think there should be some way to only config it from the web
interface, but I can't get it to stop tftp/ftp.
-Original
Mine (501's) are set up to get config via HTTP. Have you tried setting
File TX Tries to 0?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Peder
Sent: Wednesday, June 17, 2009 2:43 PM
To: 'Asterisk Users
Never saw this appear on the list. So just resending it.
Alright I've been having an issue when trying to dial out locally when
coming from SIP. This used to work no problem, now it doesn't. Now the
local PRI to Bell Is working fine I have calls coming in and out of it
constantly right now.
On Wed, 2009-06-17 at 14:18 -0500, Tilghman Lesher wrote:
On Wednesday 17 June 2009 11:56:28 John A. Sullivan III wrote:
On Wed, 2009-06-17 at 12:44 -0400, Sean Bright wrote:
Watkins, Bradley wrote:
One more bit of magic necessary here, as pbx/pbx_lua.c has includes
for: #include
Is your SIP call-limit set to 1? That might explain the busy/congest
message.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James A.
Shigley
Sent: Wednesday, June 17, 2009 2:59 PM
To: Asterisk Users Mailing List -
Go to the phone keyboard [menu] [settings] [advanced] [4-5-6] [admin settings]
[network conf.] [sever menu]. Now (edit) change server type to http (or https)
using left/right arrows. Save.
Phone as all Polycom's will happily reboot and will not ask you again for
tftp/ftp.
You may have control
James A. Shigley wrote:
Never saw this appear on the list. So just resending it.
You might get more help if you include a PRI Debug that shows the call
being rejected.
Andres
http://www.neuroredes.com
Alright I’ve been having an issue when trying to dial out locally when
coming from SIP.
What happens if the http server is down? My point is that I don't want it
to try and pull any config from a server. I just want it to use its local
config.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
2009/6/17 Tilghman Lesher tilgh...@mail.jeffandtilghman.com
On Wednesday 17 June 2009 03:45:54 Olivier wrote:
2009/6/15 Tilghman Lesher tilgh...@mail.jeffandtilghman.com
On Monday 15 June 2009 04:03:48 am Olivier wrote:
I've just discovered IMPORT function existence.
It can be use
Then remove the FTP/HTTP server from the configuration.
You'll want to configure this in the boot loader by pressing the 'setup'
softkey immediately after it boots, while giving the 5 second count-down. Clear
the server name from the server options there.
Then additionally, make sure your DHCP
James A. Shigley wrote:
snip
The odd thing is that I can send the call down one of my other PRI ports
to our Amtelco Infinity system. (via exten=
9819213,1,Dial(${inf}/409${EXTEN}). I've tried everything I can think of
and googled for a good while trying to find an explanation for got
All Polycom's in our lab are configured that way to not waste time during
reboots.
Jacek
- Wiadomość oryginalna -
Od:: Darryl Dunkin ddun...@netos.net
Data:: środa, 17 Czerwiec 2009 22:33
Temat: Re: [asterisk-users] ODP: Re: Polycom Stop Downloading Config
Then remove the FTP/HTTP
On Wednesday 17 June 2009 14:55:55 John A. Sullivan III wrote:
On Wed, 2009-06-17 at 14:18 -0500, Tilghman Lesher wrote:
On Wednesday 17 June 2009 11:56:28 John A. Sullivan III wrote:
On Wed, 2009-06-17 at 12:44 -0400, Sean Bright wrote:
Watkins, Bradley wrote:
One more bit of magic
On Wed, Jun 17, 2009 at 09:34:55AM -0400, Matt Florell wrote:
On 6/17/09, Gordon Henderson gordon+aster...@drogon.net wrote:
On Wed, 17 Jun 2009, Steve Totaro wrote:
Hi,
[snip]
Gordon
The TC400B is up to 120 channels of G729a now:
It is possible to leave a single voicemail to multiple recipients using
the following syntaz for the VoiceMail command:
exten = s,1,VoiceMail(101102103)
This will leave voicemail in the INBOX for extension 101, 102 and 103.
The permissions for the voicemail audio file in 101 are correctly set to
On Wed, 2009-06-17 at 15:43 -0500, Tilghman Lesher wrote:
On Wednesday 17 June 2009 14:55:55 John A. Sullivan III wrote:
On Wed, 2009-06-17 at 14:18 -0500, Tilghman Lesher wrote:
On Wednesday 17 June 2009 11:56:28 John A. Sullivan III wrote:
On Wed, 2009-06-17 at 12:44 -0400, Sean Bright
Hi,
At the moment, I can't read Local channels variables using IMPORT function :
${IMPORT(Local/7...@pcdialer-5dff\;1,CALLERID(num))}
Is it normal behaviour ?
Any idea or suggestion ?
A DumpChan statement show usual variables.
Comparing a Local channel with a SIP channel, the only difference I
2009/6/17 Benny Amorsen benny+use...@amorsen.dk benny%2buse...@amorsen.dk
John Novack jnov...@stromberg-carlson.org writes:
I have wondered for years now why someone thought there needed to be two
different transfer functions.
Transfer should be ONE function. If one wants to speak first
Hi All;
asterisk-gui read/write from the conf files or database?
Any advise?
Regards
Bilal
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[2009-05-27 02:06:16.294] WARNING[6971] chan_dahdi.c: No D-channels
available! Using Primary channel 24 as D-channel anyway!
[2009-05-27 02:06:16.295] VERBOSE[6971] logger.c: [2009-05-27 02:06:16.295]
== Primary D-Channel on span 1 up
[2009-05-27 02:06:16.301] ERROR[6971] chan_dahdi.c: !! Got a
Hi Jim,
Thanks for your kind offer - I may well need to pick your knowledge at
some point.
I've not long got 2007 up and running and am trying to convert a few
people back at the office that this could be something useful to look at
(generally the NBX phone system we have currently doesn't
Do you have an example of your configuration?
I haven't converted my gateways to dahdi yet, but my configuration is,
in this order:
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On
/etc/dahdi/system.conf has this:
loadzone=us
defaultzone=us
#Sangoma A102 port 1 [slot:4 bus:7 span:1] wanpipe1
span=1,0,0,esf,b8zs
bchan=1-23
hardhdlc=24
/etc/wanpipe/wanpipe1.conf has this:
[devices]
wanpipe1 = WAN_AFT_TE1, Comment
[interfaces]
w1g1 = wanpipe1, , TDM_VOICE, Comment
[wanpipe1]
On Sunday 07 June 2009 09:35:50 Cary Fitch wrote:
When Asterisk sends a call to a phone company via a PRI/Dahdi, does it
actually send CALLERID(ANI), or only CALLERID(NUM)?
No, only CALLERID(num). I'm presently looking into the possibility of
supporting that in the future, though.
--
Hi,
I wanted to follow up on this thread about WB support on the MeetMe bridge that
is in 1.6.2. Does it only work for G722 or any WB codec ?
I am working with another 16k WB codec that I can transcode to 722 and vice
versa. I was curious if the 1.6.2 MeetMe bridge can also mix 722 with any
hardhdlc is for a BRI, use dchan=24 instead to set the d-channel.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jim
Dickenson
Sent: Wednesday, June 17, 2009 16:04
To: Asterisk User MailList
Subject: Re:
Greetings! I'm about to embark on a journey to implement an Asterisk
solution for providing VoIP telephony to a large number of locations
for a client. The idea is to have a central server(s) accepting IAX2
registrations from remote Asterisk boxes and then routing calls via an
upstream ITSP using
On Wednesday 17 June 2009 17:06:25 Olivier wrote:
At the moment, I can't read Local channels variables using IMPORT function
: ${IMPORT(Local/7...@pcdialer-5dff\;1,CALLERID(num))}
I'm confused as to why you're trying to escape the semicolon here. The name
of the channel cannot be hardcoded
On Wed, Jun 17, 2009 at 3:18 PM, John Todd jt...@digium.com wrote:
On Jun 17, 2009, at 8:16 AM, Steve Totaro wrote:
Hi,
Quick question to the real world.
Approx what specs would I need on server to handle 95 ZAP or Dahdi -
SIP gateway using G729 on the SIP to carrier side (nothing
On Wednesday 17 June 2009 17:48:37 Jim Dickenson wrote:
[2009-05-27 02:06:16.294] WARNING[6971] chan_dahdi.c: No D-channels
available! Using Primary channel 24 as D-channel anyway!
[2009-05-27 02:06:16.295] VERBOSE[6971] logger.c: [2009-05-27 02:06:16.295]
== Primary D-Channel on span 1 up
Changing to dchan=24 and rebooting the system caused the inability to make a
call and this error
WARNING[3978]: app_dial.c:1468 dial_exec_full: Unable to create channel of
type 'Dahdi' (cause 34 - Circuit/channel congestion)
Changing it back and rebooting allow me to place calls again.
--
Jim
Hi
I am trying to get transferring of calls working, I place a call from
ext 101 = 103 and then from 101 I try and transfer the call to 102
(such that it will be 102=103), I have tried flash and *2 and nothing
seems to work.
I have allowed transfers in sip.conf, I am expecting a dial tone when i
Hello, all. My apologies up front but I must be brain cramping on
something very simple. I've tried to pare down my configuration to the
absolute minimum for SIP traffic just to understand how it works. My
incoming calls are not finding the s extension in my dial-plan. I am
assuming SIP calls
I then fire up twinkle on my desktop and dial sip:3...@pbx.mycompany.com.
The Asterisk console shows:
[Jun 17 22:58:55] NOTICE[32060]: chan_sip.c:18160 handle_request_invite:
Call from '' to extension '36' rejected because extension not found.
If I use the same extensions.conf but change s to 36,
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