Un-top-posting...
Cutting re-posted digest -- again -- even after pointedly pointing it out
the first time...
On Sat, 9 Apr 2011, Steve Edwards wrote:
0) Don't re-post the entire digest back to the list it came from.
Posting 36k of cruft to ask 'How to change SIP port number?' seems
maill...@lightspeed.ca writes:
We've had several customers report since upgrading them to our new
Asterisk 1.6.2.17 server (from version 1.4), that their MWI no longer
works. No significant changes have been made to their SIP
configuration, nor to their ATA configuration.
My testing of
On Monday 11 April 2011 00:25:35 magnu...@inputinterior.se wrote:
Now i am lost.
exten = 0424449631,n,NoOp(${CALLERID(name)})
exten = 0424449631,n,NoOp(${CUT(CALLERID(name),\(,2):0:-1})
-- Executing [0424449...@fax.inputinterior.se:4] NoOp(OOH323/Avaya2-8,
Martela (fax)) in new stack
--
It was a 1.8 but then we started to do a lot of development (ooh323) so
today it is Asterisk SVN-may-ooh323_ipv6_direct_rtp-r311741MS-/trunk.
Can hardly se that we have done any changes that would cause my problem.
-Ursprungligt meddelande-
From: Tilghman Lesher
Sent: Monday, April
On Monday 11 April 2011 02:56:03 magnu...@inputinterior.se wrote:
It was a 1.8 but then we started to do a lot of development (ooh323) so
today it is Asterisk SVN-may-ooh323_ipv6_direct_rtp-r311741MS-/trunk.
Can hardly se that we have done any changes that would cause my
problem.
Are you sure
On Friday 08 Apr 2011, vip killa wrote:
Is there a way for asterisk's voicemail to send an email (including
voicemail attachment) to multiple email addresses?
It's probably easiest to set up a user on your mail server to receive the
voicemail messages that are meant for multiple recipients,
I dont know if my mail client will keep formatting as I se it, but for me it
sure looks like one space.
-- Executing [0424449...@fax.inputinterior.se:4] NoOp(OOH323/Avaya2-109,
Martela (fax)) in new stack
xyz
--
Dear Experts,
please send me the ways to change asterisk port from 5060 to 5061 i
need to configure it because we are already using 5060 port in router
then we cant use it again we have to configure other sip server so
please suggest me a way..
--
-- Forwarded message --
From: darin iv adari...@gmail.com
Date: Mon, 11 Apr 2011 14:33:24 +0530
Subject: changing port 5060 to 5061
To: asterisk-users@lists.digium.com
Dear Experts,
please send me the ways to change asterisk port from 5060 to 5061 i
need to configure it because
On 11 Apr 2011, at 10:03, darin iv wrote:
please send me the ways to change asterisk port from 5060 to 5061 i
need to configure it because we are already using 5060 port in router
then we cant use it again we have to configure other sip server so
please suggest me a
Hi Magnus,
exten = 0424449631,n,NoOp(${CUT(CALLERID(name),\(,1):0:-1})
But that gave me “Martela “ so my way of doing it is wrong.
Any that can tell me what I am doing wrong or have any better suggestion
howto do it?
I think you are not able to do it in one step. Can you try something like
Hi,
The reason I think Dial isn't appropriate is not to do with the database call.
Here's the wider context of the application I'm putting together:
Punter calls in, leaves a message, gets a reference number, hangs up. System
then initiates call to a queue of on-call staff and when one
Dears;
I have been faced with a problem that I am not sure about how can I solve
it...
I my scenario there is a variable which will be ready just after the callee
had hanged up and the caller, which coming throw a Queue.
But the CDR fields are logged into DB just after the Queue application. so
Not quite true. I use a PHP script to do my processing (called from
voicemail.conf [externnotify = /usr/local/bin/vmnotify.php]).
The main three lines are:
$vm_context = $argv[1];
$extension = $argv[2];
$number_of_messages = $argv[3];
Self explanatory really.
-Original Message-
Dear
there is some problem.
the true way for running php script, is using agi not system.
second after 5 sec, a lot of channel variables were removed, it makes your
program wrong.
with some little experience you can add your script to a2billing, try it.
best
On Sat, Apr 9, 2011 at 7:22 PM, Bruce
U were right, breaking it into two lines work.
exten = 0424449631,n,NoOp(${CALLERID(name)})
exten = 0424449631,n,Set(name=${CUT(CALLERID(name),\(,1)})
exten = 0424449631,n,NoOp(${name:0:-1})
-- Executing [0424449...@fax.inputinterior.se:4] NoOp(OOH323/Avaya2-150,
Martela (fax)) in new stack
--
I am using Elastix. Asterisk is used for PBX application in Elastix. I want
to make customize MOH. But not able to use MOH. I make simple extension in
asterisk conf file but no success :(
But when I used Vanilla Asterisk then All things are working
Below are the details of configuration
On Sun, Apr 10, 2011 at 9:12 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
Off-topic:
On Fri, Apr 08, 2011 at 03:30:58PM +, satish patel wrote:
[snip]
System: Linux/2.6.32-24-server built by root on
x86_64 2011-03-22 18:38:19 UTC
Ubuntu has a separate
On Mon, Apr 11, 2011 at 07:45:01AM -0400, Steve Totaro wrote:
On Sun, Apr 10, 2011 at 9:12 AM, Tzafrir Cohen
tzafrir.co...@xorcom.comwrote:
Off-topic:
On Fri, Apr 08, 2011 at 03:30:58PM +, satish patel wrote:
[snip]
System: Linux/2.6.32-24-server
We are talking about mailcmd not externnotify
I am aware of extennotify, problem is, it runs script when someone checks
their voicemail, i need a script to run only when a voicemail is left
On Mon, Apr 11, 2011 at 6:32 AM, Andrew Thomas a...@datavox.co.uk wrote:
Not quite true. I use a PHP
I don't understand what you guys talking about? You mean say there is
a issue in ubuntu kernel to use asterisk?
--
Sent from my iPhone
On Apr 11, 2011, at 8:05 AM, Tzafrir Cohen tzafrir.co...@xorcom.com
wrote:
On Mon, Apr 11, 2011 at 07:45:01AM -0400, Steve Totaro wrote:
On Sun, Apr 10,
Dear, I have two Asterisk PBXs with an E1 interface/RJ-45 port in both
boxes. I need to connect both PBXs with E1/R2 and UTP cable.
What are the requirements to deploy the UTP cable ??? Straight-through
or crossover ??? What are the pinouts in both peers ???
Thanks a lot,
Alejandro
--
Bruce B said:
We experience exact same thing on DAHDI with Sangoma USB FXO device on short
circuited lines. Phantom calls are actually due to a short in the lines that
happen occasionally.
-Bruce
Also, Warren Selby said:
I've seen this on cases where a phantom call comes in on a DAHDI
For PRI coross over cable following is pin layout
1 --- 4
2 --- 5
Date: Mon, 11 Apr 2011 10:43:51 -0300
From: aco1...@gmail.com
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Asterisk-Asterisk E1 connection
Dear, I have two Asterisk PBXs with an E1 interface/RJ-45 port in
You need a E1/T1 crossover cable, which isn't straight through or like a
network crossover cable. Search online for T1 crossover and you'll find the
pinout.
Remember one node needs to be the clock source (and only one node).
Technically UTP isn't the right cable for E1/T1s, but if your
@ Tzafrir
you mean say i shouldn't use -server kernel for asterisk ?
-Satish
Date: Mon, 11 Apr 2011 07:45:01 -0400
From: stot...@asteriskhelpdesk.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Ubuntu *-server kernels [was: Re:
IAX2/0.0.29.199]
On Sun, Apr
Hello,
I have the scenario: Link E1 with ISDN running with asterisk 1.8.2.3, voicerlib
4.2.3.0, libpri 1.4.11.4 and dgvchannel 1.0.8. Asterisk begins a call, when the
called party answer the call, it asks for the the called party to input some
numbers (via DTMF) but when the numbers are
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brian Henning
Sent: Monday, April 11, 2011 8:47 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Occasional call from asterisk
On Mon, Apr 11, 2011 at 01:49:59PM +, satish patel wrote:
@ Tzafrir
you mean say i shouldn't use -server kernel for asterisk ?
I meant that a different kernel flavour may work better. Though I asked
for feedback about that.
--
Tzafrir Cohen
icq#16849755
I wonder if you can test to see if this happens if you had an analogue phone
set connected. And if it doesn't then I am wondering why Asterisk or Sangoma
card is so sensitive and maybe the sensor can be set a bit higher so these
calls don't end-up ringing like they don't if an analogue phone set
Hi ,
In vicidial dialer
I need small Dialplan require. when i call from hardphone , in that has 1to9
no.s i want define the dipositions like when i press the 1 it will goes
NotIntrest, press 2 for NotAvailable.
How can i configure for this.
--
Best Regards,
Mahesh Katta
*BUZZ**WORKS* Business
Thanks for the input but I am not sure if that answer my question of if it's
normal behaviour for AGI scrip to terminate after the h extension rather
than end of x extension even if it was only run in x extension.
Regards,
On Mon, Apr 11, 2011 at 6:34 AM, Pezhman Lali l...@lopl.net wrote:
Dear
In my kernel i found following options. its configured for 100Hz do you think
if i set it to 1000Hz it will be more responsive ?
root@shirley:/boot# cat config-2.6.32-30-server | grep CONFIG_HZ_
CONFIG_HZ_100=y
# CONFIG_HZ_250 is not set
# CONFIG_HZ_300 is not set
# CONFIG_HZ_1000 is not set
Hi all,
I realise that asterisk's codec negotiation has been discussed in
the past multiple times. What I haven't been able to understand is
how asterisk decides which video codecs to advertise to the other
end when canreinvite=no in sip.conf and the initial caller
doesn't support video.
I'm using voicemail ODBC with Asterisk 1.6.2.17.2.
Why do I see Length is 186545 or something similar but a different number
in Asterisk CLI everytime someone leaves a message?
--
_
-- Bandwidth and Colocation Provided by
On 11 Apr 2011, at 15:28, vip killa wrote:
I'm using voicemail ODBC with Asterisk 1.6.2.17.2.
Why do I see Length is 186545 or something similar but a different number
in Asterisk CLI everytime someone leaves a message?
Because not all messages are the same length. I'd guess it's length in
indeed but why in console and the info is so limited, it doesn't say
which message or anything...strange
On Mon, Apr 11, 2011 at 10:31 AM, Steven Howes steve-li...@geekinter.netwrote:
On 11 Apr 2011, at 15:28, vip killa wrote:
I'm using voicemail ODBC with Asterisk 1.6.2.17.2.
Why
Hey Guys!
Just recently i come to know about this option CONFIG_HZ=1000 in kernel is this
important for asterisk application ? we have ubuntu with CONFIG_HZ=100 should
i think about this option ?
root@shirley:/boot# cat config-2.6.32-30-server | grep CONFIG_HZ_
CONFIG_HZ_100=y
#
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
Sent: Monday, April 11, 2011 9:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] voicemail odbc Length is .
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel
Sent: Monday, April 11, 2011 9:45 AM
To: asterisk-users
Subject: [asterisk-users] Asterisk kernel CONFIG_HZ=1000
Hey Guys!
Just recently i come to know about
Apologies you were correct, i had debug on... Sorry...
On Mon, Apr 11, 2011 at 10:45 AM, Danny Nicholas da...@debsinc.com wrote:
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa
Believe me i went through all google pages and read every possible post. But i
just wanted to know from asterisk point of view.
-S
From: da...@debsinc.com
To: asterisk-users@lists.digium.com
Date: Mon, 11 Apr 2011 09:50:36 -0500
Subject: Re: [asterisk-users] Asterisk kernel CONFIG_HZ=1000
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel
Sent: Monday, April 11, 2011 9:58 AM
To: asterisk-users
Subject: Re: [asterisk-users] Asterisk kernel CONFIG_HZ=1000
Believe me i went through all google pages
On Mon, Apr 11, 2011 at 02:44:54PM +, satish patel wrote:
Hey Guys!
Just recently i come to know about this option CONFIG_HZ=1000 in kernel is
this important for asterisk application ? we have ubuntu with CONFIG_HZ=100
should i think about this option ?
root@shirley:/boot# cat
Your last line in the dialplan should be StartMusicOnHold(), not just
MusicOnHold().
Thanks,
--Warren Selby, dCAP
On Apr 11, 2011, at 6:24 AM, virendra bhati virbh...@gmail.com wrote:
I am using Elastix. Asterisk is used for PBX application in Elastix. I want
to make customize MOH. But
h is hangup extension, and will be executed after hangup
On Mon, Apr 11, 2011 at 6:36 PM, Bruce B bruceb...@gmail.com wrote:
Thanks for the input but I am not sure if that answer my question of if
it's normal behaviour for AGI scrip to terminate after the h extension
rather than end of x
On 4/9/2011 11:56 PM, vip killa wrote:
I've already taken the steps you described...issue i ran into was
there is no variables passed to mailcmd only STDIN... as a result i
have to extract variables from STDIN...
On Fri, Apr 8, 2011 at 5:09 PM, Warren Selby wcse...@selbytech.com
Hi All,
I already try StartMusicOnHold() instead of MusicOnHold();
Even default asterisk MOH not playing.
On Mon, Apr 11, 2011 at 9:17 PM, Warren Selby wcse...@selbytech.com wrote:
Your last line in the dialplan should be StartMusicOnHold(), not just
MusicOnHold().
Thanks,
--Warren
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati
Sent: Monday, April 11, 2011 11:23 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk MOH not working with
On 4/11/2011 5:15 AM, Naomi Rosenberg wrote:
Hi,
The reason I think Dial isn't appropriate is not to do with the database
call. Here's the wider context of the application I'm putting together:
Punter calls in, leaves a message, gets a reference number, hangs up. System
then initiates
On Mon, Apr 11, 2011 at 8:47 AM, Brian Henning bhenn...@pineinst.comwrote:
H. I do see this in the /var/log/asterisk/messages log:
[Apr 5 00:05:36] NOTICE[9579] chan_dahdi.c: Got event 4 (Alarm)...
[Apr 5 00:05:38] NOTICE[9579] chan_dahdi.c: Got event 5 (No more alarm)...
[Apr 5
I have install ubuntu kernel 2.6.32-30-preempt to have 1000Hz timing.
Date: Mon, 11 Apr 2011 18:28:36 +0300
From: tzafrir.co...@xorcom.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk kernel CONFIG_HZ=1000
On Mon, Apr 11, 2011 at 02:44:54PM +, satish
Search the lists. Some hints:
Viking electronics makes a door box that connects to any analog line
(IIRC e-20).
They also make a DTMF keypad that integrates in series with any analog
line. They might also make a door box with a DTMF keypad on it.
Sandman makes a relay that will get energized when
On 11-04-11 05:44 AM, Arjan Kroon | Mobillion wrote:
We are using asterisk version 1.8.3.1 on the inbound server
For the outbound server I used version 1.8.2.2.
(If have tested with an inbound server with version 1.8.2.2 to the outbound
server and that works fine.)
Does anybody has an idea
Would be so much simpler if mailcmd acted just like externnotify or
externnotify only ran when a message was left but not when someone checks
their voicemail...
That's pretty much where you're at. What gets passed to STDIN is an
email, it's not set up for use by a script. Remember, what you're
You probably didn't read over my originally post carefully. In the dialplan
A2billing.php script is called in the X extension. Then there is
X,n,Hangup() so now X extension is dead.
After that in h extension I have ANOTHER script running. However, the CLI
output (which again I posted in my
On 4/11/2011 12:30 PM, vip killa wrote:
Would be so much simpler if mailcmd acted just like externnotify
or externnotify only ran when a message was left but not when
someone checks their voicemail...
That's pretty much where you're at. What gets passed to STDIN is an
email, it's
On 11-04-11 10:26 AM, Effie Mouzeli wrote:
This may lead to some buggy clients not to accept the call (with 488),
but I've noticed some cases where a callee was behind NAT,
an INVITE with one video codec would me forwarded properly
to the callee, but another INVITE with 3 video codecs, would
I'm not confused about this...
Everytime a voicemail is left, I need asterisk to run a script that will
query a database, and according to those results perform various actions.
These actions include calling a number and connecting it directly to
voicemailmain and/or sending out multiple emails...
Anyway, i figured out how to accomplish this using externnotify...
In app_voicemail.c, in the function vm_execmain i
commented out run_externnotify(vmu-context, vmu-mailbox, NULL);
Now externnotify is called by asterisk only when there is a new message
and not when someone checks their
On 4/11/2011 12:47 PM, vip killa wrote:
Anyway, i figured out how to accomplish this using externnotify...
In app_voicemail.c, in the function vm_execmain i
commented out run_externnotify(vmu-context, vmu-mailbox, NULL);
Now externnotify is called by asterisk only when there is a new
Yes ,
I show me the all configured MOH. But don't play the MOH.
After 12 sec of silence CLI give message
Stop music on hold
As I told you that default moh also not played :)
On Mon, Apr 11, 2011 at 9:55 PM, Danny Nicholas da...@debsinc.com wrote:
--
Hi ,
As we see the SIP shatus on CLI with *sip show status
How we get the status with phpagi function ?
*
--
-
Thanks and regards
Virendra Bhati
+91-9172341457
--
_
-- Bandwidth and Colocation Provided by
Hello all,
I was wondering if anybody found a solution for 0018418. It looks like
any 1.8.x version is affect by bug #0018418. The meetme application
allows crosstalk between participants when the w option (Wait for
Leader) is enabled.
Please let me know if anybody can help me with this. I have
On 4/11/2011 12:52 PM, virendra bhati wrote:
Hi ,
As we see the SIP shatus on CLI with *sip show status
How we get the status with phpagi function ?
*
--
-
Thanks and regards
Virendra Bhati
+91-9172341457
Googling is your friend:
Doesn't Elastix have it's own tool for MusicOnHold? Maybe check with that and
see if that makes a difference.
Thanks,
--Warren Selby, dCAP
On Apr 11, 2011, at 12:49 PM, virendra bhati virbh...@gmail.com wrote:
Yes ,
I show me the all configured MOH. But don't play the MOH.
After 12
Hi All,
I have asterisk 1.8.3.2 and having issue with not getting VoiceMail email. I
can send mail through command line using sendmail but not via asterisk. We have
centralized zimbra email server. why its trying to send email to local
127.0.0.1 address? is there any other configuration i am
suck sendmail
Solution:
sudo apt-get remove sendmail
sudo apt-get install ssmtp mailutils
From: satish...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Mon, 11 Apr 2011 20:11:27 +
Subject: [asterisk-users] Voicemail to email issue
Hi All,
I have asterisk 1.8.3.2
that is a sendmail issiue. Obviously asterisk is contacting 127.0.0.1 to try
and deliver e-mail.
Try help with sendmail folks, check that 127.0.0.1 is in the allowed to
relay list or so..
On 11 April 2011 21:11, satish patel satish...@hotmail.com wrote:
Hi All,
I have asterisk 1.8.3.2 and
The Asterisk Development Team announces the release of DAHDI-Linux 2.4.1.2.
DAHDI-Linux 2.4.1.2 and DAHDI-Linux-Complete 2.4.1.2+2.4.1 are
available for immediate download at:
http://downloads.asterisk.org/pub/telephony/dahdi-linux
http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete
Continuing top posting...
The same argument could be made for any commercial solution. Why use
Asterisk when we could throw $4,000 at our problem for a commercial
solution?
I'd like to have a solution that would have the features you suggest for
$400.
--Don
On Behalf Of C F
Sent: Monday,
Hi guys,
I'm trying to get blind transfer to work and automatically transfer call
to another number on key sequence press.
Extensions.conf_snippet
[from-pstn]
exten = _0399377744,1,Set(__DYNAMIC_FEATURES=blindxfer)
exten = _0399377744,n,Set(__GOTO_ON_BLINDXFR=to-pstn ^0388924326^1)
exten =
On Mon, 11 Apr 2011 12:58:39 +0200
magnu...@inputinterior.se wrote:
U were right, breaking it into two lines work.
exten = 0424449631,n,NoOp(${CALLERID(name)})
exten = 0424449631,n,Set(name=${CUT(CALLERID(name),\(,1)})
exten = 0424449631,n,NoOp(${name:0:-1})
-- Executing
One of our client facing this issue, we have try to solve it but we're lack
of asterisk knowledge. Anybody can help us? Isn't any problem with asterisk
configuration or the problem come from PRI E1 itself?
[Apr 11 15:32:48] VERBOSE[9231] chan_dahdi.c: -- Requested transfer
capability: 0x00 -
Hi,
Trying to create templates that allow higher compression of sip.conf, so for
example:
[internal-number](!)
type=friend
secret=bigsecret
host=dynamic
context=internal
disallow=all
allow=ulaw
[100](internal-extensions)
mailbox=100@internal-extensions
[101](internal-extensions)
Weired result:
exten = 0424449631,n,NoOp(${CALLERID(name)})
exten = 0424449631,n,NoOp(${${CUT(CALLERID(name),\(,1)}:0:-1})
-- Executing [0424449...@fax.inputinterior.se:4] NoOp(OOH323/Avaya2-248,
Martela (fax)) in new stack
-- Executing [0424449...@fax.inputinterior.se:5]
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