Once the call is completed you can use SOX to split the call. In my
opinion, you will have to get a larger ram disk or record the files to a
different format like WAV49, but maybe somebody has a better solution for
you.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Extensions do not register - peers do. A peer can register itself or be
registered by Asterisk. In most cases the extension is equivalent to the
peer (301 = 301) but it can be quite different (301 = sipuser1) or (301 =
d...@impalanetworks.com).
From: asterisk-users-boun...@lists.digium.com
Of Danny Nicholas
Sent: Monday, November 14, 2011 5:01 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] How do extensions stay registered
Extensions do not register - peers do. A peer can register itself or be
registered by Asterisk. In most cases
ChanisAvail?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jaap Winius
Sent: Wednesday, November 09, 2011 9:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] IAX2
What about this?
asterisk -rx core show function CONFBRIDGE_INFO
-= Info about function 'CONFBRIDGE_INFO' =-
[Synopsis]
Get information about a ConfBridge conference.
[Description]
This function returns a non-negative integer for valid conference
identifiers
(0 or 1 for 'locked')
by that name
registered.
Command 'core show function CONFBRIDGE_INFO' failed.
On Wed, Nov 9, 2011 at 12:24 PM, Danny Nicholas da...@debsinc.com wrote:
What about this?
asterisk -rx core show function CONFBRIDGE_INFO
-= Info about function 'CONFBRIDGE_INFO' =-
[Synopsis]
Get information about
Discussion
Subject: Re: [asterisk-users] ConfBridge 1.6.20 user count
confbridge(xxx,c) is a blocking call, so you can't get status back until
that command completes. Time to upgrade to 10.0.beta2 I guess...
On Wed, Nov 9, 2011 at 12:47 PM, Danny Nicholas da...@debsinc.com wrote:
10.0.beta2. Have you
If you have an ancient version of Asterisk you want to stick with, you can
do this with asterisk -rx sip set debug on and asterisk -rx agi set debug
on in your safe_asterisk script.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
It can have to do with either the telephones dial plan or the context in the
Asterisk dial plan combined with your features.conf settings.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ramiro Paz
Sent: Monday, November 07, 2011 8:46
Have you posted this to the forum Asterisk Support on asterisk.org? One
thing I see is that you are doing an attended transfer (*2) vs a blind
transfer (#1); that could be causing some sort of problem.
From: asterisk-users-boun...@lists.digium.com
transfer..
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: maandag 7 november 2011 16:16
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] bug in queuemanager?
Have you
Of Danny Nicholas
Sent: maandag 7 november 2011 16:46
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] bug in queuemanager?
Do you have an isolated environment where you can do a core show channels
verbose after the transfer, but before the end of the call
PBX, and not on
the asterisk which is handling the queues.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: maandag 7 november 2011 17:37
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re
My assumption is that SIP/213 is a multi-line phone like a Polycom. As for
the code
Line 1 - (the || is or) is the line coded for ringing while in use or ignore
when busy?
Line 2 is the line an unknown device or not in use
Line 3 set newstate using sub ast_parse_device_state
Line 4 is newstate =
:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] CallerID inconsistently presented through
ISDN/cellular networks
2011/11/3 Danny Nicholas da...@debsinc.com
snip
[callbob]
Exten = _XX.,1,answer
Exten = _XX.,n,Set(CALLERID(num)=${EXTEN
Please elaborate on your flavor of DAHDI and LIBPRI and what type of DAHDI
service you are using (PSTN, T1, etc). Speaking from a POTS line point of
view, there can easily be a 7-10 second delay in the processing of DAHDI
information (which would make your 1347 second call within tolerance).
What version of Asterisk? Is the forwarding done using Followme, attended
transfer or blind transfer?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Thursday, November 03, 2011 8:14 AM
To: Asterisk Users Mailing List -
, 2011 at 18:44, Danny Nicholas da...@debsinc.com wrote:
Please elaborate on your flavor of DAHDI and LIBPRI and what type of DAHDI
service you are using (PSTN, T1, etc). Speaking from a POTS line point of
view, there can easily be a 7-10 second delay in the processing of DAHDI
information (which
] CallerID inconsistently presented through
ISDN/cellular networks
2011/11/3 Danny Nicholas da...@debsinc.com
What version of Asterisk?
1.6.1.18
Is the forwarding done using Followme, attended transfer or blind
transfer?
a plain Answer plus Dial
From: asterisk-users-boun
-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, November 03, 2011 9:38 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] CallerID inconsistently presented through
ISDN/cellular networks
Something like this?
[callbob]
Exten = start,1
I use Polycom 501 phones. I have two networks - 192.168.23.0/24 and
192.168.33.0/24. My Asterisk server and most of my phones are on the 23
net. I have the one phone on the 33 net for cross-net testing (works fine on
1.4.41).
From: asterisk-users-boun...@lists.digium.com
Will this affect 10.X or is it just a 1.8 path?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard
Mudgett
Sent: Wednesday, November 02, 2011 9:27 AM
To: Asterisk Users Mailing List - Non-Commercial
You have set an insufficient range in rtp.conf. Asterisk uses 2 ports per
call, but allocates 4 for transferring, etc, so when you set up a range of
10001-10040 (for example) you are basically setting a range of 10 concurrent
calls. Check rtp.conf and make the end range larger by 8 or 12 or
: Re: [asterisk-users] Unable to build sip pvt data - Switching
equipment congestion
Hello,
thank you for your answer...
Current range (rtp.conf) : 11500 - 11650
Current calls : 20 à 25
Is this not sufficient ??
Jonas.
On 11/02/2011 04:00 PM, Danny Nicholas wrote:
You have set
, November 02, 2011 10:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Unable to build sip pvt data - Switching
equipment congestion
On 11/02/2011 04:13 PM, Danny Nicholas wrote:
150/4 = 37.5. maybe your sip peer has a conflicting range?
Where do I
: [asterisk-users] Nat Phone in Asterisk 10
Not info about networl settings. Please give output of
ip l
ip -4 a
ip ro
2011/11/1 Danny Nicholas da...@debsinc.com
Hello listers,
Another opportunity presents itself in my 1.4 to
10.0 conversion. My asterisk is set up
One way to do this (there are probably more and better ways). Incoming call
to 123456789 launches meetme(1234,b(connecta.agi))
Connecta.agi calls lines B and C and connects them to meetme(1234).
-Original Message-
From: asterisk-users-boun...@lists.digium.com
wrote:
on 11/01/2011 03:25 PM Danny Nicholas wrote the following:
One way to do this (there are probably more and better ways). Incoming
call
to 123456789 launches meetme(1234,b(connecta.agi))
Connecta.agi calls lines B and C and connects them to meetme(1234).
Thanks, but could you be more
If you are using the silent option of voicemail (b - busy, u -
unavailable, s - silent) you could set up a context to play the normal
silent message, then goodbye.
[normal-voicemail]
Exten = start,1,playback(unavail-msg)
Exten = start,n,voicemail(${ARG1}@default)
Exten =
(but not greetings)
On Mon, Oct 31, 2011 at 8:22 AM, Danny Nicholas da...@debsinc.com wrote:
If you are using the silent option of voicemail (b - busy, u -
unavailable, s - silent) you could set up a context to play the normal
silent message, then goodbye.
Completely irrelevant, but I always
and accepts
calls and calls out in 1.4.41 but shows unreachable in 10.0. What changed
that is killing my off-network phone?
Thanks in Advance
Danny Nicholas
--
_
-- Bandwidth and Colocation Provided by http://www.api
Please post output of CLI command dialplan show sipphones
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of motty.cruz
Sent: Friday, October 28, 2011 9:26 AM
To: 'Asterisk Users Mailing List - Non-Commercial
Now sip show peers and sip show peer 4783
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of motty.cruz
Sent: Friday, October 28, 2011 9:29 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject:
)D N
0Unmonitored
Thanks,
-motty
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Friday, October 28, 2011 8:21 AM
To: 'Asterisk Users Mailing List - Non
Do you have QOS Priority set to 7 on these phones (VOIP should get your
highest network priority unless you have critical data downloads)?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justin
Sherrill
Sent:
This information might be in /var/log/asterisk/messages or /v/l/a/full. If
not, you can change the logging and get it there (turn on debug in one of
them) (/etc/asterisk/logger.conf)
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
I don't work with 1.8, but wouldn't
Set(MEETME_RECORDINGFILE=/tmp/recording.wav) before starting meetme do it?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Thursday, October 27, 2011 10:54 AM
To: 'Asterisk Users Mailing
Here is a sneaky suggestion: start a local call that is recorded and have
it join the conference.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Thursday, October 27, 2011 12:04 PM
To: 'Asterisk Users Mailing List -
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Message: 2
Date: Thu, 27 Oct 2011 09:39:43 -0500
From: Danny Nicholas da...@debsinc.com
Subject: Re: [asterisk-users] Check which
You could temporarily change rtp.conf to use just 4 ports (say 10001-10004)
and monitor 10001 and 10002. On a production system you would have to do
something with a tool like netstat to try and predict which ports in the
range would be used.
From: asterisk-users-boun...@lists.digium.com
ID on the
transferred phone instead of the incoming callerID. My assumption is that
there is some new users.conf or sip.conf flag I'm not setting to make this
functionality work?
Thanks in Advance
Danny Nicholas
What does your context [from-internal-xfer] look like? (it should either
resemble or have an include for [default] context).
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ramiro Paz
Sent: Wednesday, October 19, 2011 10:33 AM
To:
Just a WAG - if you start the call in voice-mode, the video codecs aren't
loaded.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Karsten
Wemheuer
Sent: Wednesday, October 19, 2011 10:37 AM
To:
.
Ramiro PAZ
MASTERLINE LOGISTICS
On Wed, Oct 19, 2011 at 11:39 AM, Danny Nicholas da...@debsinc.com wrote:
What does your context [from-internal-xfer] look like? (it should either
resemble or have an include for [default] context).
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk
1. odbc has been successful for some posters
2. I would personally use System or AGI to handle my MYSQL stuff so you have
clean bash-like handling.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis
?
Thanks in Advance
Danny Nicholas
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
-Commercial Discussion
Subject: Re: [asterisk-users] nvfaxdetect in 10.0
On Tue, Oct 18, 2011 at 6:21 PM, Danny Nicholas da...@debsinc.com wrote:
Hi gang,
We are moving our 1.4 asterisk with DAHDI over to 10.0
with SIP. Everything is going nicely except that I can’t get
A Generic SIP device would be a SIP trunk, hardphone or softphone. A
Generic ZAP device would be a Zaptel/DAHDI device like a TDM400P or OBI110.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of michael k
Sent: Monday, October 17, 2011
You could possibly get the IP from sip show peers then curl back to that
address.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adam Robins
Sent: Friday, October 14, 2011 7:31 PM
To: Asterisk Users Mailing
Netstat -anp has been useful in finding this error for me in the past. A
normal Asterisk call will have 2 or 4 udp connections to carry traffic
to/from phone to pbx. On a one-way call, this will be an odd count. Then
you can check your rtp.conf and firewall and see how the channel got
blocked.
Not the answer you are looking for, but some controlling factors are
1. The available bandwidth. Since a call takes 30-90K depending on
the codec unless you are using a compression codec that can reduce this to
5K or so, your number of channels available will be limited by this.
2.
What happens if she keys in the number+# then presses dial?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adam Robins
Sent: Friday, October 14, 2011 10:29 AM
To: Asterisk Users Mailing List - Non-Commercial
I use 501's here and I can pull up the settings by typing
http://1.2.3.4/index.htm - where 1.2.3.4
http://1.2.3.4/index.htm%20-%20where%201.2.3.4 is the IP address of the
phone. If you can do that, perhaps something there will be of use to you.
From: asterisk-users-boun...@lists.digium.com
- wget won't work because the phone requires a user ID and password for
access, but lynx or curl might do the trick. I was able to use lynx -
couldn't figure out the voodoo for curl.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Can you post the call file with the pertinent info blacked out? I'm on
1.4.41 so I might be able to assist.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Thursday, October 13, 2011 3:31 PM
I would replace
SetVar: MEETME_PLAYFILE=/tmp/jerry.wav
With
SetVar: MEETME_PLAYFILE=/tmp/jerry
Also, you could replace your C AGI with a BASH AGI just to verify that the
data is getting there.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
It seems to me as though this is happening:
Asterisk A
Exten = 5,1,Dial(IAX/${EXTEN:1})
So the call goes to Asterisk B as
Exten = 5,1,Dial(IAX/1001)
So you need to change the IAX dial out command on Asterisk A to not truncate
the 5 and set up 5 on Asterisk B's inbound context so
According to this
http://www.voip-info.org/wiki/view/Asterisk+sip+permit-deny-mask you are
only allowing traffic In from 1.2.3.4.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of hussein
korbani
Sent: Tuesday, October 11, 2011 3:15 PM
I would change DAHDI/1 to DAHDI/G1 or DAHDI/R1 - DAHDI/1 is DAHDI line 1
only, R1 is Round Robin Group 1, G1 is sequential Group 1
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of michael k
Sent: Thursday, October 06, 2011 10:46 AM
To:
Depending on hardware and number of parking lots, could hints be used to let
everyone know that a parking lot was just put into use by blind transfer?
(I have a PERL web interface that does this kind of check for 1.4 but that
probably wouldn't help OP).
-Original Message-
From:
Usually this message is received because you did something like
playback(beep.gsm) or playback(beep.wav) instead of playback(beep). It is
(IMO) somewhat confusing because you have to do record(foo.gsm) but you have
to playback using playback(foo).
-Original Message-
From:
(${A_serviceline_file}.wav,0,60)
-Oorspronkelijk bericht-
Van: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Namens Danny Nicholas
Verzonden: 04-10-2011 16:30
Aan: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Onderwerp: Re: [asterisk
You have files in /var/lib/asterisk/moh1?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of salaheddine
elharit
Sent: Tuesday, October 04, 2011 12:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Assuming that you don't have some sort of reconnect protocol going on like
SIP headers, a native-bridge to a local channel might do the trick for you.
If you are using DAHDI, you might be out of luck.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
This belongs on the commercial list.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis
Sent: Thursday, September 29, 2011 9:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
29, 2011 at 10:47 AM Danny Nicholas da...@debsinc.com
wrote:
This belongs on the commercial list.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick
Khamis
Sent: Thursday, September 29, 2011 9:44 AM
I would either use a gotoif to determine which queues get recorded or put
the recordable queues into a separate context (probably the simpler
solution).
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lyle McKarns
Sent: Thursday,
As I read this, the following should be correct:
exten = 222,n,Dial(SIP/${EXTEN},,KkTtL(6))
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of salaheddine
elharit
Sent: Wednesday, September 28, 2011 1:23 PM
To: Asterisk Users
If there is an existing Asterisk install, the contents of /etc/dahdi/modules
should tell you this.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of salaheddine
elharit
Sent: Monday, September 26, 2011 1:16 PM
To: Asterisk Users Mailing
Or
/bin/llsmod|grep dahdi
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of salaheddine
elharit
Sent: Monday, September 26, 2011 1:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] model of
Matt - how dare you tell a man asking for a fish to learn how!
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Riddell
Sent: Monday, September 26, 2011 4:48 PM
To: asterisk-users@lists.digium.com
Subject:
@lists.digium.com
Subject: Re: [asterisk-users] Asterisk Realtime Time Dial App
On 27/09/11 10:51 AM, Danny Nicholas wrote:
Matt - how dare you tell a man asking for a fish to learn how!
:-) It would have taken way too much time to explain all the steps.
Although, having said that I am doing
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mehmet
Avcioglu
Sent: Friday, September 23, 2011 12:02 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] AGI Problem
Hello,
I have an AGI script
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Hiller
Sent: Friday, September 23, 2011 1:29 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Postgresql Reconnect on connection failure
Currently if asterisk loses
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Luke Hamburg
Sent: Thursday, September 22, 2011 3:24 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] bounty for ASTERISK-17474 streaming MusicOnHold
bug
Hi all-
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leif Madsen
Sent: Tuesday, September 20, 2011 7:39 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Fixing an old bug related to extension s -
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olle E.
Johansson
Sent: Tuesday, September 20, 2011 4:23 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Fixing an old bug
That's what the DISA function is for.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Malvin Rito
Sent: Tuesday, September 20, 2011 8:34 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Add PinCode on my dialplan
- Original Message -
From: Danny Nicholas mailto:da...@debsinc.com
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
mailto:asterisk-users@lists.digium.com
Sent: Tuesday, September 20, 2011 8:38 AM
Subject: Re: [asterisk-users] Add PinCode on my dialplan
That's what
Just do make menuselect
Then
Make make install
As long as you don't do any of the other steps after make install, no
configuration files should be updated.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of tran quoc tuan
Sent:
-Commercial Discussion
Subject: Re: [asterisk-users] How to add new Module in existed Asterisk
Thank Danny Nicholas for your reply ,
It means I may re-make menuselect in Asterisk version 1.4.36 to add 2 new
modules : jabber and chan_gtalk ? Can version Asterisk 1.4.36 support these
module : jabber
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Monday, September 19, 2011 9:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] /usr/sbin/asterisk -rx and AMI
Hello,
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Monday, September 19, 2011 9:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] question on DTMF
I am
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Monday, September 19, 2011 10:07 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] question on DTMF
pmap 14249
Could this be a problem with pbx_loopback?
Thanks
Danny Nicholas
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I would do this with ex-girlfriend logic
[mycontext]
Exten = s,1,playback(instructions)
Exten = s/5551212,n,goto(end)
Exten = s,n,read(var,prompt, .)
Exten = s,n,process..
Exten =s(end),n,hangup
From: asterisk-users-boun...@lists.digium.com
(rfc2833)
...
Now your x-girlfriend will never know if its her phone's fault or you've
done some trick :P
On Fri, Sep 16, 2011 at 11:39 PM, Danny Nicholas da...@debsinc.com wrote:
I would do this with ex-girlfriend logic
[mycontext]
Exten = s,1,playback(instructions)
Exten = s/5551212,n,goto
+1 (at least) Steve
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Thursday, September 15, 2011 3:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
+1 Dale - although it would be a good idea for OP to know the in's and out's
of both System and AGI, this is a simpler way for him to catch a fish today.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dale
Since the Read command takes in its input 1 digit at a time (I don't think
this changes in 1.8 or 10.X either), your best option here would be to
follow the read with a press 1 to accept or 2 to re-enter IVR
[get-number]
Exten = s,1,Read(number,prompt1,10,skip,1,10)
Exten =
If you use qualify=yes, you should only get OK when the line is functional.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of NaJIm
Sent: Wednesday, September 14, 2011 3:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
to make sure the trunk is still
alive.
In my case the trunk was completely down, and then it was showing status OK
as soon as the Internet came up.
Regards,
Najim
On Thu, Sep 15, 2011 at 2:02 AM, Danny Nicholas da...@debsinc.com wrote:
If you use qualify=yes, you should only get OK when
Did you read the “IAX/SIP registration” section (under Authentication) on
voip.ms?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of naren
Sent: Tuesday, September 13, 2011 2:22 PM
To: John Novack
Cc: Asterisk Users Mailing List -
, but it doesn't seem to be a link. I am not sure how I can find
the page that has the details about the IAX/SIP registration. I see in the wiki
there is a page that has the configuration info for iax.conf and
extensions.conf.
Thanks for your help.
naren
On Tue, Sep 13, 2011 at 2:25 PM, Danny
http://www.voip-info.org/wiki/view/Asterisk+cmd+SendDTMF
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ezequiel Lovelle
Sent: Tuesday, September 13, 2011 3:37 PM
To: Asterisk Users
Subject: [asterisk-users] Send DTMF
¿How can i
to be a link. I am not sure how I can
find the page that has the details about the IAX/SIP registration. I see in
the wiki there is a page that has the configuration info for iax.conf and
extensions.conf.
Thanks for your help.
naren
On Tue, Sep 13, 2011 at 2:25 PM, Danny Nicholas da
Sip show channels will give you the active codec. You can get the
information using an AGI or a system command.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tom Browning
Sent: Tuesday, September 13, 2011
If you change ${ocem} to ${ocem}@default, this will probably work as you
want it to.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kelly opal
Sent: Tuesday, September 13, 2011 4:36 PM
To: asterisk-users@lists.digium.com
Subject:
I personally would not bother with 1.6 unless you needed some feature in
that branch. 1.4 is the stable branch, but it seems that all of the
resources are being channeled into 1.8 and 10.0, so 1.6 is a rabbit hole
you really shouldn't be headed into.
-Original Message-
From:
I think that is your best bet. 1.8.6 unless somebody has a good reason not
to.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tarek Sawah
Sent: Monday, September 12, 2011 11:00 AM
To: Asterisk Users
Subject:
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