THIS IS INCORRECT! MOST changes to chan_dahdi.conf are applied on a reload.
There are a few items like switchtype and signaling and a few other items
which require chan_dahdi.so to be unloaded then loaded.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:
No, you don't need your script for this.
Prevent attacks by using fail2ban to block brute force attacks using iptables,
securing your server at the OS level, and NEVER EVER EVER let leave the web GUI
for FreePBX open to the internet. I'm sure others have more suggestions.
Over the years 100% o
users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: Monday, November 19, 2012 12:01 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Max app_voicemail line length
We are getting this message on an Asterisk 1.4.44 box.
[2012
We are getting this message on an Asterisk 1.4.44 box.
[2012-11-19 08:49:27] WARNING[11785] app_voicemail.c: List of extensions is too
long (>1323). Truncating.
I know Asterisk removed many of limitations in string lengths in in 1.6+. Does
anyone know if this also applies to app_voicemail?
-
Polycom phones after firmware 2.x register to BOTH the primary and backup
servers.
On Thu, Nov 15, 2012 at 8:59 AM, Chris Nighswonger
wrote:
Would the simplest approach to failover be to just configure my
primary asterisk server as the first SIP server and my backup
module unload res_musiconhold.so
and
module load res_musiconhold.so
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Markus
Sent: Tuesday, November 13, 2012 1:00 PM
To: Asterisk Users Mailing List - Non-Commerci
No, it isn't.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christopher
Harrington
Sent: Thursday, October 25, 2012 12:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk
I'm setting up a test server with a Digium TE122 and am getting the following
error on the console, spewing as fast as it can. Does anyone have any idea
what this error might be?
[Oct 19 11:24:53] NOTICE[2076]: chan_dahdi.c:3108 my_handle_dchan_exception:
PRI got event: Event 59 (59) on D-chan
I seem to recall seeing somewhere recently where there was a bugfix for
ulaw/alaw conversion which would cause poor audio.
Have you tried updating your Asterisk to the latest of whatever major version
you are running?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[m
Has anyone seen issues with recent Sangoma T-1 cards and Sangoma Analog cards
on multiple different servers?
On T-1: we get NO traffic, no interrupts, and no increase in number of packets
and the PRI does not come up.
On Analog: The ports do NOT go red when you unplug the phone line from FXO
p
Try: exten => _00336123412xx/_44XX.,1,Set(RINGTIME=90,g)
Notice the _ on your callerid pattern
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Geoffrey Yeoh
Sent: Thursday, October 11, 2012 1:15 PM
To: aste
Have you tried Dial instead of Transfer?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Deepesh D
Sent: Thursday, October 11, 2012 2:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: R
Once an option is set in the chan_dahdi.conf file it applies to every channel
=> line listed after the setting, until the option is changed. This is all you
really know.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On
A port is not a door if there is nothing listening on the port.
Open ports are not a security issue. Stuff running on open ports are.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Alvarez
Sent: Wedne
You do not have an exten => 700972595637212 in the context in extensions.conf
that SIP device is set to in sip.conf
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Smith
Sent: Monday, October 01, 2012 4:1
You are set up as a USA PRI, but not dialing a USA TN.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of motty.cruz
Sent: Wednesday, September 26, 2012 11:13 AM
To: 'Asterisk Users Mailing List - Non-Commercial D
Going to n+101 was deprecated in Asterisk 1.2 or 1.4. Don't use it.. Read the
docs for Authenticate and see what diaplan variables you can check.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Asterisk Newb
You are doing it wrong. I know 50 bazillion Asterisk dialplan examples on the
internet do it the same way. It is still wrong.
When you do a Dial on the dialplan you need check the value of DIALSTATUS or
HANGUPCAUSE before dialing again. Both variables will give you some indication
of why the
: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Friday, September 14, 2012 10:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Fax Detect on Demand
2012/9/13 Eric Wieling
Yes
t - Non-Commercial Discussion
Subject: Re: [asterisk-users] Fax Detect on Demand
2012/8/16 Eric Wieling
Using Asterisk 1.8.mumble. We would like to use fax detect on demand.
Both chan_dahdi and chan_sip support setting fax detetect on a static
basis,
For curiosity
Try adding qualify=yes
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Tuesday, September 11, 2012 3:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users]
Your best bet is a carrier class device from someone like Adtran and convert
the PRIs to SIP before passing the calls to Asterisk.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL INDRODIYA
Sent: Monday
pbx*CLI> core show function CHANNEL
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Stefan at WPF
Sent: Friday, August 24, 2012 6:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [aste
Adding the IPs of ALL local interfaces to /etc/hosts has helped solve this
issue for me in the past.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby
Sent: Thursday, August 23, 2012 4:25 PM
To: Ast
I find the UPGRADE*.txt files in the source tarball more useful for something
like this. It details only the significant operational changes rather than
every single code change.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digiu
Has anyone experimented with increasing the DAHDI chunk size in improve fax
reliability? If so, did it help, hurt, or not make any difference?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Ast
Using Asterisk 1.8.mumble. We would like to use fax detect on demand.
Both chan_dahdi and chan_sip support setting fax detetect on a static basis,
but no way I've been able to find to enable/disable it on demand in the
dialplan.
In 1.4 we used the NVFaxDetect 3rd party app, but that no longer
Using "n" with labels is what most people do. A dialplan isn't javascript, you
don't need two hundred 3 line functions.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leandro Dardini
Sent: Friday, August 03,
sk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: Wednesday, August 01, 2012 10:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk Dahdi 1.6.2.23 Iaxmodem
Sounds like DAHDI/4 is a FXO port. FXO ports are considered answered when
dialing
Sounds like DAHDI/4 is a FXO port. FXO ports are considered answered when
dialing is completed.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of motty.cruz
Sent: Wednesday, August 01, 2012 1:11 PM
To: 'Asteris
I'm getting the following warning with SOME calls on 1.8.14.1 Is it a cause
for concern? Is there a way to fix it? I can't tell for sure if it is
impacting calls or not.
WARNING[27796]: chan_sip.c:20497 handle_response_invite: just did sched_add
waitid(4077) for sip_reinvite_retry for dial
The Adtran NetVanta series has a number of good devices which support PRI/SIP
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Saturday, July 28, 2012 6:43 PM
To: Asterisk Users Mailing List - Non-Com
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming
Sent: Friday, July 27, 2012 10:45 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] still got ReceiveFax() problem, how to pr
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] T.38 (PRI) Fax Debugging
On 07/20/2012 09:48 AM, Eric Wieling wrote:
Neither of your questions relate to T.38 :-) Maybe you meant T.30 instead.
> 1) Does anyone know of any software to debug the g711cap audio files
> Asteris
1) Does anyone know of any software to debug the g711cap audio files Asterisk's
res_fax generates? Google has not been very helpful.
2) These files are in WAV format, but my Windows Media Player cannot play them.
The Linux "file" command reports "RIFF (little-endian) data, WAVE audio,
Microsof
Why would you NOT want the connectedline info sent immediately?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard Mudgett
Sent: Wednesday, July 18, 2012 12:24 PM
To: Asterisk Users Mailing List - Non-Comm
Remove the ",i" to start with. Do you have the various rpid related options in
sip.conf set?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Wednesday, July 18, 2012 12:08 PM
To: 'Asterisk Users Ma
eming
Sent: Wednesday, July 18, 2012 11:28 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk 1.8.13 / res_fax / res_fax_digium
On 07/18/2012 10:06 AM, Eric Wieling wrote:
> We are using res_fax_digium with a Sangoma PRI card on asterisk 1.8.13
>
> The docs at http://
We are using res_fax_digium with a Sangoma PRI card on asterisk 1.8.13
The docs at http://docs.digium.com/FAX/fax_for_asterisk_admin_manual.pdf
indicate v34 is supported, but when I enable it I get the message
"res_fax_digium.c:1624 dgm_fax_new: V.34 not supported, will be ignored." Is
v34 on
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Tuesday, July 10, 2012 5:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FreePBX: using conte
Recent Polycom firmware versions (4.x, I think) also have support for "user"
sort of stuff. See the 4.x Admin Guide.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming
Sent: Tuesday, July 10, 20
Channels can be in more than one group.
Make g0=1-15,17-31,32-46,48-62 and
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mahesh katta
Sent: Tuesday, July 10, 2012 10:04 AM
To: Asterisk Users Mailing List -
Asterisk, and by extension FreePBX, automatically end the voicemail recording
when the caller hangs up. You have some OTHER issue. Perhaps Asterisk is not
detecting the hangup?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.di
I've never seen this on incoming calls, only outgoing calls.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrew Colin
Sent: Friday, June 29, 2012 8:11 AM
To: Asterisk Users Mailing List - Non-Commercial Dis
We have been quite disappointed by the Adtran VQM. It often shows calls which
had audio issues as being close to perfect. It also often shows calls which
sound perfect as having significant quality issues.
We don't allow reinvites so this might be part of the issue. I don't have a
lot more
Is there anything specific in the plaintext SIP packets you want to secure?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce B
Sent: Friday, June 22, 2012 1:57 PM
To: Asterisk Users Mailing List - Non-Comm
This is a Polycom question, not an Asterisk question.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of motty.cruz
Sent: Tuesday, June 19, 2012 1:38 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion
o swapping.
-Vladimir
On 6/16/2012 7:40 PM, Eric Wieling wrote:
> I was assuming incoming DTMF detection. Try toneduration=250 in chan_dahdi
> to increase the duration of transmitted DTMF on your DAHDI channels. If that
> fixes it, try lowering it. I find 80 usually works with
I was assuming incoming DTMF detection. Try toneduration=250 in chan_dahdi to
increase the duration of transmitted DTMF on your DAHDI channels. If that
fixes it, try lowering it. I find 80 usually works with even the worst IVRs.
-Original Message-
From: asterisk-users-boun...@lists.d
In my experience when you have "intermittent problems with incoming caller ID,
FXS -- with DTMF detection" you have to adjust your rxgain and/or txgain. I am
NOT a fan of Digium cards, but these CallerID and DTMF issues are simple and
solvable and not related to the card itself.
-Original
For some reason 1.4.4.x was not reading chan_dahdi.conf. When I symlinked it
to zapata.conf it worked.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: Wednesday, May 30, 2012 2:35 PM
To
What does the output of "g729 show licenses" show? If it doesn't show licenses
then Asterisk is not licensed for G729 codec.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim King
Sent: Tuesday, June 05, 20
This is incorrect. The vast majority of settings in chan_dahdi.conf are
applied when you do a module reload chan_dahdi.so
You cannot change signaling, switchtype, or add or remove channels (I'm sure
there are a few others) on a module reload, but most settings will be applied
on a reload.
If
So there is no other setting except disable the caller id detection for the
system to pickup incoming call at the first ring?
Thanks a lot.
On 6/2/12, Eric Wieling wrote:
> Try usecallerid=no
>
> The immediate= option is mainly for FXS ports and is almost never used.
>
> -Original
Try usecallerid=no
The immediate= option is mainly for FXS ports and is almost never used.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Satria Anamarta
Sent: Saturday, June 02, 2012 12:06 AM
To: Asterisk Us
Last time I checked (a few years ago) Sangoma has half height brackets
available. Contact their support or sales.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ade Vickers
Sent: Friday, June 01, 2012 10:41
-Commercial Discussion
Subject: Re: [asterisk-users] Sangoma Card Issue
On Wed, May 30, 2012 at 02:34:55PM -0400, Eric Wieling wrote:
> Has anyone experienced an issue with Sangoma analog cards where the
> card suddenly stops working? Trying to dial out shows the channel as
> busy, even tho
Has anyone experienced an issue with Sangoma analog cards where the card
suddenly stops working? Trying to dial out shows the channel as busy, even
though there is no active call on that port?
This happened to us often when we used Digium cards (in fact this issue is why
we stopped using Digiu
I believe one of the patches involved in fixing for The Great Voicemail
Problem* about a year ago was to make voicemail automatically renumber the
mailbox files if it saw a gap.
* from memory: The Great Voicemail Problem is a bug where if you received a new
voicemail while listening to a messa
We use Adtran Total Access boxes to convert PSTN to SIP.Xorcom has some
PSTN/SIP USB boxes which people seem to love.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Chavez
Sent: Monday, May 21, 201
Do a "sip show peer PEERNAME" and check the codecs allowed for that specific
peer.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ricardo Carvalho
Sent: Wednesday, May 09, 2012 11:56 AM
To: Asterisk Users Mai
Now you have a totally different issue. 8-)
While the call is up do a "sip show channels" in the CLI. This will show you
the ACTUAL codec for the call. Likely the call was still using GSM. Did you
remember to put a disallow=all before the allow= lines?
I recommend dtmfmode=rfc2833 with what
Try using Dial(SIP/+44797XX@voipms,30,D(ww0788XX)t)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shahid H
Sent: Saturday, May 05, 2012 11:20 PM
To: asterisk-users@lists.digium.com
Subject: [aster
If you set the ctime (or maybe mtime) of your spool file to a date in the
future, then asterisk won't process the .call file until that future time.
I recommend creating your call files with a random ctime/mtime for 0 - 240
seconds in the future and make sure you have a random retry time in your
Or even Hangup(-${Z})
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Barry Miller
Sent: Wednesday, May 02, 2012 2:11 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] parsing issue
On Wed,
Have you tried the MATH() function?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of CDR
Sent: Wednesday, May 02, 2012 1:48 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] parsing issue
I get a
If you have quotes on one side of the = sign, then you need quotes on the other
side. In your dialplan line you are comparing + with "+". A plus sign is not
equal to quote plus sign quote
exten => _X., n, Set(CALLERID(num)=${IF($["${CALLERID(num):0:1}" =
"+"]?${CALLERID(num)}:0${CALLERID(nu
You should make sure to read the UPGRADE*.TXT files for 1.4.x on, they are
included in your Asterisk source code.
I re-read them every couple of months to keep the information fresh.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.di
If a value can EVER be empty, then you want to use quotes in your expressions.
exten => s,n,Set(torture.calls=${IF($["${torture.calls}" = "1"]?2:1)})
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytl
The only way you can do this is by enabling DND on the phone.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Thursday, April 26, 2012 4:27 AM
To: Asterisk Users Mailing List - Non-Commercia
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming
Sent: Wednesday, April 25, 2012 6:25 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Hangup Cause and SIP Response Code
On
Remove the Asterisk source dir, unpack the tarball again and run configure.
1.4 is weird about configure being built before DAHDI is installed.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad
Se
Which version of Asterisk are you using?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier CALVANO
Sent: Wednesday, April 18, 2012 2:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subje
No, if you are dialing to a TN which is in use you get a BUSY, except on FXO
signaled ports which are always considered ANSSWERED when the PBX finishes
dialing.
If you are trying to dial out via a LINE which is in use, you would likely get
a CONTESTION.
-Original Message-
From: aste
As long as Host does not contain the peer's IP address in iax2 show peers then
it is not going to work and is not registered.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph
Sent: Sunday, April 15, 2012
It was never valid.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Friday, April 13, 2012 3:36 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users
PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] priorityjumping - asterisk 1.8
On 04/13/12 13:00, Eric Wieling wrote:
>Priority jumping was deprecated in 1.2 I expect the feature was removed
>sometime before 1.8. Did you read all the UPGRAD
Priority jumping was deprecated in 1.2 I expect the feature was removed
sometime before 1.8. Did you read all the UPGRADE*.txt files included in 1.8?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph
Se
There is NO relation to kernel or anything else. They change the version
number whenever they feel like it. If you want the specific changes there is
change log included in the source code.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...
ril 05, 2012 2:24 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dial Plan - Routing via Caller ID
On Thu, Apr 5, 2012 at 11:00 AM, Eric Wieling wrote:
Priorities are not complicated. Each extension starts with priority 1,
all additional pr
Priorities are not complicated. Each extension starts with priority 1, all
additional priorities are "n" and you ALWAYS end your extension with a Goto or
a Hangup so the call doesn't fall off your intended extension and hump into the
middle of some other extension and screw everything up.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of A J Stiles
Sent: Tuesday, April 03, 2012 12:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Voicemail crashs asteri
http://www.itu.int/oth/T0202.aspx?parent=T0202
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lenz Emilitri
Sent: Thursday, March 29, 2012 5:13 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Su
"core show application saydigits"
"core show application SayUnixTime"
Or better yet "core show applications"
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad
Sent: Thursday, March 29, 2012 8:45 AM
You are welcome to an incomplete dataset I have. Data was gathered from
publically available sources, including the ITU and Wikipedia. Data does NOT
include information for country code 1.
http://rock.nyigc.net/e164.csv.gz
Enjoy.
-Original Message-
From: asterisk-users-boun...@lists.
Have you read the backtrace.txt included in the doc/ directory Asterisk source
code?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Thursday, March 22, 2012 12:06 PM
To: Asterisk Users Mai
Extension "i" only works for IVRs and other things like Background and
WaitExten, it does not work to match incoming calls to an invalid extension.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryant Zimmer
I'm a fan of Vitelity. They are no-frills, but they work well for my very low
usage. I think their web portal is ugly, not all that intuitive, but it does
work. I've been with them since early 2006 for my few low usage DIDs.
-Original Message-
From: asterisk-users-boun...@lists.digiu
Those messages someone or something is running "asterisk -r" or similar.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ahmed Munir
Sent: Wednesday, March 14, 2012 3:13 PM
To: asterisk-users@lists.digium.com
The faxdetect option is documented in the 1.8 sip.conf.sample.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl
Sent: Tuesday, March 13, 2012 6:17 PM
To: asterisk-users@lists.digium.com
Subject: Re:
This means the config file says 3 ports, but no card is detected.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of rama...@gmx.de
Sent: Tuesday, March 13, 2012 10:31 AM
To: asterisk-users@lists.digium.com
Subjec
Check the sip.conf.sample. 1.8 has several options related to the SIP UPDATE
support.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Arstan
Sent: Monday, March 12, 2012 10:10 PM
To: asterisk-users@lists.digi
There are no such statistics. Your usage patterns are unique to you and depend
on many factors. If you must look for the information then look in the mailing
list archives or on voip-info.org.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun
users@lists.digium.com
Subject: Re: [asterisk-users] Multi-record SRV records
Hi,
On 12/03/12 13:48, Eric Wieling wrote:
> Have you tried permit/deny on the peer?
>
No, I've not tried this, however, will those entries be checked if the inbound
call is not matched against the peer that those
Have you tried permit/deny on the peer?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Guy Gold
Sent: Monday, March 12, 2012 9:45 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Multi-reco
You would set a dialplan variable from inside your AGI.
In PHPAGI it might be something this:
$agi->set_variable("CUST_ID", $cust_id);
Then in your dialplan after your script runs you can use the variable like any
other.
You can also get dialplan variables AND functions:
$tmp = $agi->get_var(
1.4:
pbx> core show channels
[snip]
167 active channels
84 active calls
1.8:
pbx> core show channels
[snip]
23 active channels
12 active calls
9567 calls processed
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf
What you want to do is complicated with Asterisk. Your best solution may be to
write an application to monitor active calls via the Asterisk Manager interface.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
You should be able to configure the Polycom phones to failover/failback more
quickly. Check the Admin Guide.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov
Sent: Sunday, March 04, 2012 2:52 PM
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alejandro Imass
Sent: Tuesday, February 28, 2012 10:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Same provider
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