I believe that EVAL might be able to help you here:
https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Function_EVAL
Example:
Allphones=${EVAL(Kphones)}&${EVAL(Sphones)}
I'm not sure if in the globals it will let you, but in the dialplan for
sure it will.
On Tue, Jan 24, 2023 at
Have you tried adding “-n”?
Also, what version of asterisk are you using? newer versions only have
colorized output when your are connected to the console (-r) not for remote
commands (-rx)
On Wed, Aug 3, 2022 at 08:21 Carlos Chavez wrote:
> I usually like to have the colorized output
I would check if you don't have any channels in a hung/zombie state...
Have a look if "core show calls" matches "core show channels".
Either way, it seems wonky, so you might end up having to give that
asterisk a restart... :S
On Wed, Jul 27, 2022 at 6:21 PM David Cunningham
wrote:
> Hello,
>
Hi,
I wanted to add some comments to Sebastian's response:
1- When you have a lot of DIDs, you can't just "port" them over from
company1 to company2. Try to have 1M or so DIDs and ask if you can just
port them. No no, not that simple. There is a process that a lot of times
is not worth the
Well I skipped reading the part where you say “only for log files” :-(
On Thu, Jul 23, 2020 at 21:06 Joel Serrano wrote:
> Have you tried starting asterisk with the "-n" param?
>
>-n Disable console colorization
>
> On Thu, Jul 23, 2020 at 5:11 PM And
Have you tried starting asterisk with the "-n" param?
-n Disable console colorization
On Thu, Jul 23, 2020 at 5:11 PM Andrew Yager wrote:
> Hi,
>
> Is there a way to drop the ANSI colour strings from log files? In
> particular, I've got JSON logging throwing logs over to ES,
,Hangup()
>
> I added hints to see if that would make a difference and it hasn't.
>
> I also made a 'Anonymous' peer to see if that would help without any luck.
>
> On Thu, Jul 16, 2020 at 6:11 PM Joel Serrano wrote:
>
>> Hey John,
>>
>> In one installation I h
in the s,1 exten. Feel free to
remove this one.
Give it a try and let me know how it goes.
Alternatively, you may also be able to configure your SBC
(kamailio/opensips? if so check dispatcher docs for *_reply_codes modparam)
to accept a 404 reply to a SIP:OPTIONS as a valid response.
Hope i
or the uniqueid plus the sequence, I can grab the fields from the
row that has them.
On Tue, Feb 25, 2020 at 4:01 PM Joel Serrano wrote:
> Hello,
>
> I have a setup with asterisk 16.8.0, I'm facing a problem where calls that
> fail (CONGESTION) don't have filled in some extra f
ld empty in the db doesn't
make sense to me.
Any tips on where/how I can troubleshoot this?
Thanks,
Joel.
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Check out the new Asterisk communi
` for devices
registering via an intermediate proxy :)
Just wanted to update this in case it helps anyone else.
Cheers,
Joel.
My problem was caused by the *nat=* setting for the device.
I found
On Wed, Mar 20, 2019 at 11:37 AM Joel Serrano wrote:
> Hello,
>
> We have a couple asterisk1
Hello,
We have a couple asterisk11 servers behind a Kamailio4 proxy. We are in the
process of upgrading to asterisk16 and Kamailio5 and I'm testing out Path:
support with chan_sip (migration to PJSIP is not possible right now due to
integrations with other systems).
Functionality-wise things are
a=candidate:5 2 UDP 2111766782 192.168.56.1 62586 typ host
Thanks,
- Joel
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New to Asterisk? Join us for a live introductory webinar every Thurs
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
I use the mini-web browser built into the phone and have a custom
button (directory) that accesses the directory, which is hosted on a
web server.
It isn't perfect, but it's better than the XML files IMHO. That said,
there's an enterprise license for these phones which enables directory
My take on this is to not skimp on the phones. This is how people
relate to the phone system you install. Good phones will, to them,
imply a good system. And vise-versa.
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I have asterisk call out to a shell script which sends a jabber message to the
user (along with links to any open tickets in our ticketing system associated
with that CID). All free, but requires work to build.
On Jan 26, 2011, at 6:52 AM, Gilles codecompl...@free.fr wrote:
Hello
I'd
On Wed, Jan 26, 2011 at 7:55 AM, Tom Rymes try...@rymes.com wrote:
Ooh. I like this. Can you post a sample, or maybe a synopsis of what pieces
you are using to tie this all together?
I have a two processes - one to notify on an internal incoming call,
one to notify on tickets (both on incoming
On Tue, Jan 18, 2011 at 8:18 AM, Andrew Thomas a...@datavox.co.uk wrote:
Why do I top post? Simple. I read every message in the thread - and if
there are 10 messages (for example) in that thread - then why should I
have to read them all over again on the last one?
That's not the alternative
On Tue, Jan 18, 2011 at 1:21 PM, Steve Totaro
stot...@asteriskhelpdesk.com wrote:
I got it fixed with an all nighter, but I took a beating for the
problems for not fully testing and monitoring. After that, nobody had
faith in the fax solution.
So is FFA working for you now? What did you
On Thu, Jan 6, 2011 at 7:06 AM, Jim Dickenson dicken...@cfmc.com wrote:
Are there reasons to prefer the use of PRI over SIP or SIP over PRI?
Assuming you are talking to connect a PBX to the PSTN...
PRI advantages:
1. Relatively little equipment between the PTSN and the PBX. Less to
break or
Get rid of the spaces before and after the equal sign.
On Wed, Dec 29, 2010 at 4:15 PM, Joseph syscon...@gmail.com wrote:
I'm testing GotoIf($[${CALLERID(num) but I'm missing something as it is not
working:
[office-open]
exten = s,1,Wait(1)
exten = s,2,Answer()
; for Caller ID is
I'm going to guess you aren't going to get a lot of help on a list
hosted by Digium on how to use a potentially illegal codec...
That said, ast14 in the filename might signify what the problem is.
The APIs likely changed for modules between 1.4 and 1.8.
On Wed, Dec 22, 2010 at 7:58 AM, Giorgio
On Tue, Nov 30, 2010 at 2:28 AM, bilal ghayyad bilmar...@yahoo.com wrote:
If I ran IAX in TCP port, and in case my network was having a lot of users
doing browse on the internet and downloading, so in that case and if the IAX
used TCP port, so the voice will be better than using UDP (because
What format are the actual calls in? Are they in G.711u/a format or
are they in something else (perhaps gsm?) format? I'm asking to find
out if Asterisk would need to transcode them.
On Mon, Nov 22, 2010 at 6:47 AM, Vilius Adamkavicius
vilius.adamkavic...@invade.net wrote:
Hi All,
We have a
.
On Mon, Nov 22, 2010 at 7:58 AM, Vilius Adamkavicius
vilius.adamkavic...@invade.net wrote:
Hi Joel,
We have a meetme on which we are landing two G.711 alaw calls, one coming
from TDM another from SIP. Once we those parties are in the conference we
are adding one more leg using Local channel
NAT? Firewall?
On Thu, Nov 11, 2010 at 3:21 PM, Marek Soha ma...@snet.sk wrote:
Hi all.
I have an issue with T.38 and re-invites.
Topology:
provider - A (asterisk 1.6) - B (asterisk 1.6) - extension -
- (software fax, gateway whatever).
When between A and B trunk is canreinvite=no
On Mon, Nov 8, 2010 at 7:05 AM, Cary Fitch ca...@usawide.net wrote:
Does anyone know if ATT EELs delivered to a CLEC would be PRI, or Robbed
bit?
It won't be ISDN. It will be some form of RBS. You probably have several
choices as to which type of RBS (probably several ESF options, you'll
I believe this looks like a standard channel bank. Asterisk generates all
audio. Ring and hook status are sent out of band. Dial tones are in-band.
Ringback, busy, congestion are in-band audio. I would think a standard T1 card
would be fine.
That said, I would verify this with the LEC.
for the day. The server serves doesn't serve
international calls anywa...
Sat, Oct 30, 2010 at 9:33 PM, Joel Maslak jmas...@antelope.net wrote:
No. It seems that opening ...
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On Sun, Oct 31, 2010 at 2:40 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Sat, Oct 30, 2010 at 07:33:23PM -0600, Joel Maslak wrote:
The CPU usage is trivial to deny them. As is the bandwidth usage, if
you are not sitting on a slowish broadband connection.
s/slow/assymetric
On Oct 31, 2010, at 9:57 AM, Jeff LaCoursiere j...@sunfone.com wrote:
This only tells you after it is way too late that you now have upstream
bills to wrangle with your carriers about, or (like in my case) that your
balance is now depeleted, if it trips anything at all.
In my very recent
On Oct 31, 2010, at 9:39 AM, Mark Deneen mden...@gmail.com wrote:
On Sun, Oct 31, 2010 at 11:26 AM, Joel Maslak jmas...@antelope.net wrote:
If these are mobile users, I hope they never use any public networks
(hotels, starbucks) where other subscribers can do things like ARP attacks
to do
On Oct 31, 2010, at 9:40 AM, jon pounder j...@inline.net wrote:
what are you using that is tied to nagios ?
I'll package it up next week and make it available.
Basically, I use nrpe to call a shell script that looks at the last five
minutes, 60 minutes, and 1440 minutes of a asterisk -rx
passwords are breakable, can't wait for the day
that you'll wake up and smell the coffee.
On Sun, Oct 31, 2010 at 11:26 AM, Joel Maslak jmas...@antelope.net wrote:
On Sun, Oct 31, 2010 at 2:40 AM, Tzafrir Cohen tzafrir.co...@xorcom.com
wrote:
On Sat, Oct 30, 2010 at 07:33:23PM -0600, Joel Maslak
Is there really any benefit to blocking these, if you use good passwords?
On Sat, Oct 30, 2010 at 1:20 PM, Warren Selby wcse...@selbytech.com wrote:
I'm experiencing this on one of my clients servers. The attack is ongoing.
Thanks,
--Warren Selby
On Oct 30, 2010, at 2:28 PM, Zeeshan
Probably doing an ident lookup when you send mail to the list. Standard
sendmail behavior.
On Oct 30, 2010, at 5:37 PM, Hans Witvliet h...@a-domani.nl wrote:
While on the subject,
what is digium doing on my port 113?
just from my logfile:
Oct 31 01:11:07 fw2 kernel: EXT; INC,
).
On Oct 30, 2010, at 6:53 PM, C F shma...@gmail.com wrote:
You kidding?
On Sat, Oct 30, 2010 at 3:43 PM, Joel Maslak jmas...@antelope.net wrote:
Is there really any benefit to blocking these, if you use good passwords?
On Sat, Oct 30, 2010 at 1:20 PM, Warren Selby wcse...@selbytech.com
, Oct 30, 2010 at 01:43:49PM -0600, Joel Maslak wrote:
Is there really any benefit to blocking these, if you use good passwords?
Regardless of any threat from those attacks succeeding, they completely
saturated the uplink in our ADSL-connected office.
What are they after, anyway? Merely cheap
On Fri, Oct 15, 2010 at 8:53 AM, Michelle Dupuis mdup...@ocg.ca wrote:
When a single call is up, call quality is fine. When a second call is up,
outbound audio is immediately choppy. We're using ulaw, and confirmed that
traffic with 2 calls is 175kbps in/out. (IAX connection out)
Asterisk
On Fri, Oct 15, 2010 at 9:35 AM, Danny Nicholas da...@debsinc.com wrote:
Don't know if this will make acceptable GSM files, but should help with
the WAV ones.
Are you using GSM to talk to an ITSP (the idea of banking voice calls going
across the internet makes me cringe)? If not, what are
don't need to guess. :)
#!/usr/bin/perl -w
#
# Copyright (C) 2010 Local Matters, Inc.
# http://www.localmatters.com/
# Author: Joel C. Maslak
#
# Licensed under GPL version 3
#
use strict;
use Carp;
my %ignore;
MAIN: {
my @out = `/usr/sbin/dahdi_scan`;
for my $ig (@ARGV
I'm trying to use a couple of old Western Electric type 500 phones (desk
model, rotary dial). These phones work fine, as tested with telco lines
(they dial, receiver/transmitter works fine, etc).
I'm running Asterisk 1.6.2.11.
I can't get them to dial through Asterisk. They are connected to a
]
*On Behalf Of *Joel Maslak
*Sent:* Friday, September 17, 2010 12:29 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] Rotary phone on Asterisk
I'm trying to use a couple of old Western Electric type 500 phones (desk
model, rotary dial
On Mon, Sep 13, 2010 at 1:32 AM, Kevin Keane subscript...@kkeane.comwrote:
My numbers are from an ATT DSL line in California, suburban San Diego
county, and just around the corner from the central office. So it is not the
distance (with DSL, the distance does make quite a difference). On the
On Mon, Sep 13, 2010 at 1:12 PM, Hans Witvliet h...@a-domani.nl wrote:
No these are also geo-stationary (same altitude, so same delay),
commercial and military satelites,
Yes, exactly. Geostationary satellites have been used for telephone for
ages (and are still used for remote areas - they
Use lowercase for ftp:// . That might be the issue but it should be easy to
test. Do your FTP server logs shpw anything?
On Sep 10, 2010, at 5:35 PM, colin mcdermott colinjamesmcderm...@gmail.com
wrote:
Hi all
I have a few Polycom 331's but after following allot of advice I can't
get
I'm trying to install both a Sangnoma A102 (with echo cancellation) card and
a Digium 8 port analog card with echo cancellation (Digium AEX800E) in the
same server. I know I probably shouldn't have mixed vendors - lesson
learned for next time.
That said, I have everything working fine...except
In a moment of inspiration, I recompiled both DAHDI and Wanpipe - and this
seemed to have resolved my issues, all is working great now.
On Wed, Sep 8, 2010 at 10:52 AM, Joel Maslak jmas...@antelope.net wrote:
I'm trying to install both a Sangnoma A102 (with echo cancellation) card
and a Digium
g711 across a network without perfect jitter/delay characteristics will not
work.
You cannot do g711 faxing across the internet - at all.
It's not a perfect solution even in an office on a dedicated LAN environment
(you'll still get failed faxes).
On Fri, Sep 3, 2010 at 12:32 PM, dave george
On Sun, Aug 15, 2010 at 9:09 AM, Michelle Dupuis mdup...@ocg.ca wrote:
Are there any best practices for using a SAN with Asterisk? In the past
we've kept config files local, but voicemail on a SAN. Aree there any
issues with latency putting voice prompts, configs, etc. on a SAN?
Anyone
I've been replacing an old Toshiba DK switch with an Asterisk solution. I'm
needing a solution for fax machines that works as well as a POTS line from
my carrier. If the POTS line is the solution, I'll keep it, but I'd rather
move away from that.
Here's what I'm thinking...will it work?
I
On Tue, Aug 3, 2010 at 1:30 PM, Mark G. Thomas m...@misty.com wrote:
I didn't know there was a U option. I don't see any mention of it
on the voip-info.org wiki or other Dial() documentation, but didn't
check for new options in the built in documentation until just now.
I updated the dial
have
experience with this.
OT: The company I work for is hiring! We're hiring a Senior System
Administration and several software engineering positions in Denver.
Visit: http://www.localmatters.com/careers
--
Joel Maslak
!!
~Joel
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to be prefixed with 1 or something like that.
On Mon, Nov 3, 2008 at 11:16 AM, Joel Pearson
[EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi,
I have a really strange problem with a Polycom 430 phone and Asterisk
1.4.20.
Currently If I dial the Polycom from my mobile phone answer the call
.
I haven't tried switching the lines around to see if its just a problem with
it being on Line 2.
The Polycom is running the latest Bootrom and Sip version.
Does anyone have any idea what could be causing this?
Cheers,
-Joel
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Hi marek,
gr8. I am working on chan_ss7 now..
Regards,
Joel
- Original Message -
From: marek cervenka [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, March 03, 2008 3:55 AM
Subject: Re: [asterisk-users
using sangoma but I am
looking for open source ss7 implementation which is chan_ss7. so need to
know about stability and recommendation for using on production server.
Please provide your recommendation suggestions.
Regards,
Joel
- Original Message -
From: marek cervenka [EMAIL
checking wheather my mail goes to asterisk users mailling list or not
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Thanks,
Joel
On Tue, Feb 26, 2008 at 10:47 PM, Erik Anderson [EMAIL PROTECTED] wrote:
On Tue, Feb 26, 2008 at 10:59 AM, Joel Solanki [EMAIL PROTECTED]
wrote:
checking wheather my mail goes to asterisk users mailling list or not
ACK
using sangoma but I am
looking for open source ss7 implementation which is chan_ss7. so need to
know about stability and recommendation for using on production server.
Please provide your recommendation suggestions.
Regards,
Joel
- Original Message -
From: marek cervenka
[EMAIL
if i sent mail to
[EMAIL PROTECTED] even i should also recieve my email
right ?
may be i can be wrong. Any i am happy that message has reached the
asterisk-users :) Hope to recieve some feedback soon
Regards,
Joel
On Wed, Feb 27, 2008 at 3:24 AM, Rob Hillis [EMAIL PROTECTED] wrote:
Posting
signalling using sangoma but I am
looking for open source ss7 implementation which is chan_ss7. so need to know
about stability and recommendation for using on production server.
Please provide your recommendation suggestions.
Regards,
Joel
- Original Message -
From: marek cervenka [EMAIL
the Grandstream's but cheaper than the Polycoms around that Aastra price
range.
Cheers,
Joel.
On Thu, 2007-12-20 at 12:33 +0800, d tbsky wrote:
Hi:
i am surveying ip phones for our company. we will use them with asterisk.
we have office in taiwan, hong kong,singapore and china.
cisco
Hmm the shape looks like an Aastra but the buttons down the side look
like PlayStation buttons to me. Maybe it's a Sony Cisco joint effort.
Joel.
On Mon, 2007-10-29 at 16:35 -0500, Kyle Sexton wrote:
Does anyone know who really makes this phone:
http://www.hybsys.bg/Products/VoIP/IP/Phones
favorite flavor.
Cheers,
Joel.
On Thu, 2007-10-18 at 11:34 -0500, Brian West wrote:
I'm sorry I call bullshit on this one. CentOS has been 2.6 for some
time.
/b
On Oct 18, 2007, at 11:22 AM, [EMAIL PROTECTED] wrote:
Just 5 months ago CENTOS started to use Linux 2.6 one
Hi All,
I need to have the same file played from MoH every time someone gets to
MoH from a Dial. I want to play marketing messages from it and I want it
to start from file 1 every time.
Anyone know if/how this can be done?
Cheers,
Joel
Thanks for the suggestion, but I need it to play multiple messages.
Always starting with the same one.
Cheers,
Joel.
On Wed, 2007-09-26 at 09:36 -0400, Forrest Beck wrote:
Make the file the only one in the /var/lib/asterisk/moh directory.
Forrest Beck
[EMAIL PROTECTED]
www.shift8.biz
incompatibility. Other than that I've never had one skip a beat, so I
hope you have the same luck.
Cheers,
Joel Hill
Support Manager
Asterisk IT
On Mon, 2007-08-27 at 18:15 -0400, Steve Totaro wrote:
Steve Totaro wrote:
Arthur Miller wrote:
Hello list,
I have a customer who
for these customers.
I've got some core dump on 1.2.18 and the patch available on ticket 9602
have fix all issues, using 1.2.18 on lots of server without any issues
http://bugs.digium.com/view.php?id=9602
--
Joel Vandal
ScopServ Inc.
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investigation asterisk has renamed them starting again at
0. So running a CRON job to do the same thing should work fine.
Cheers,
Joel
On Tue, 2007-05-22 at 20:37 -0500, Eric ManxPower Wieling wrote:
David Florella wrote:
Thank you knox. Finally, I have chosen this solution : find
/var
Hi I'm looking for some help with Vicidial, If you have experience with
it and could help with some consulting please contact me off list.
Cheers,
Joel Hill
Asterisk IT
[EMAIL PROTECTED]
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is
driving me nuts!
I have an IAX connection from a provider coming in. Could this be the
cause? Has anyone experienced anything similar.
Thanks,
Joel.
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,Goto(incoming,_XX5,4)
exten = 8,4,Goto(STE1850,s,1)
exten = 9,1,Playback(STE-thankyou) ; hangs up after plays thank you
for calling msg
exten = 0,1,Goto(STE1850,s,1) ; sends to consultant
Cheers,
Joel
On Wed, 2007-05-02 at 22:49 -0400, Jonathan Barratt wrote:
Hi Joel,
6
/sbin/safe_asterisk
the rc.local script is loaded after all the others so it won't effect
anything else, and we had some trouble with some low heat VIA
motherboards so we did the modprobe twice for the PRI.
Hope this helps.
Cheers,
Joel Hill
Support Engineer
Asterisk IT
On Mon, 2007-04-30 at 19:14
I'm going to repeat this to you again, on the receiving side you need to
set:
You haven't set the a or p variable or whatever
Description *SMS(queuename|[a][s])*
*SMS(queuename|[s]|number|message)* *deprecated*
a answer, i.e. send initial FSK packet. s act as service centre talking to
a
You haven't set the a or p variable or whatever
Description *SMS(queuename|[a][s])*
*SMS(queuename|[s]|number|message)* *deprecated*
a answer, i.e. send initial FSK packet. s act as service centre talking to
a phone.
On 4/17/07, Per Jessen [EMAIL PROTECTED] wrote:
I've been googling and
Let me also add my interest, we've got a site using Nagios and haven't
had time to work anything out yet related to Asterisk.
Cheers,
Joel.
Joel Hill
Support Engineer
Asterisk IT
On Wed, 2007-04-11 at 18:42 -0400, Watkins, Bradley wrote:
Allow me to register my interest in any and all things
Give this a try it fixes a problem we have had with a couple of Via
boxes.
modprobe wcte11xp
modprobe wcte11xp
ztfcg -vv
zttool
We found that probing the card twice before running ztcfg helped alot.
Cheers,
Joel.
Joel Hill
Support Engineer
Asterisk IT
03 8320 8100
On Mon, 2007-04-02
switchtype = euroisdn
signalling = pri_cpe
group = 1
pridialplan=unknown
context = incoming
channel = 1-15
channel = 17-31
Cheers,
Joel
On Sun, 2007-03-04 at 23:41 +, younss azzayani wrote:
please can someone send to me his files like zaptel zapta if he si
using TE110P
thank you
for your help.
Joel.
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.
I`ve got similar problem and look like the patch #7860 is responsable of
this issue... like if this patch doesnt check if the line is an E1 or
T1. I have reverse the patch on 1.2.12 and all work perfectly now.
--
Joel Vandal, CTO
ScopServ Inc
#7860 is responsable of
this issue... like if this patch doesnt check if the line is an E1 or
T1. I have reverse the patch on 1.2.12 and all work perfectly now.
--
Joel Vandal, CTO
ScopServ Inc.
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One suggestion is to transfer the call to an on-hold extension that
plays music, then go pick up the call later... or get a new SIP phone.
: )
~Joel
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gregory
Duchatelet
Sent: Friday, December
experiences with the SC420 in relation with
Digium cards?
Thanks for your help.
Joel
Asterisk IT
www.asteriskit.com.au
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its supposed to.
Any suggestions?
Joel Lansden
Solutions Architect
[EMAIL PROTECTED]
tel 205.533.2039
fax 866.602.9130
digitalparadisesystems
http://www.digitalparadise.net
Could
That did it! Thank you very much!!
~Joel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Martin
Joseph
Sent: Sunday, October 22, 2006 11:56 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: G.729 operating on outgoing only
On 2006-10
Hi Noro,
Depending on what firmware you have this is the way to go.
Go to the Functions keys page, then look for the Record button, Change
the type to DTMF and in number put in *1 which is the default Asterisk
recording function.
Hope this helps
Cheers,
Joel
Asterisk
the message
This call may be monitored for training and coaching purposes. Etc..
Cheers,
Joel.
Remco Barendse wrote:
Thanks for this, I was looking for this too.
Will the DTMF tone be audible to the other side? (In other words will they
know something is happening)
On Thu, 5 Oct 2006
;
That's all there is to it, but it won't work.
Can anyone help?
Thanks!!!
~Joel
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I changed things so that the dialplan would answer, THEN launch the script, but
this made no difference. The script still won't wait for DTMF tones from the
caller.
~Joel
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pato Valarezo
Sent: Thursday
64, while native formats is 4 (read/write = 64/64)
I'm running Asterisk 1.2.11 and have tried a couple of different codecs
in SIP.conf.
Any ideas??
Cheers,
Joel Hill
Support Engineer
Asterisk IT
www.asteriskit.com.au
www.theasteriskshop.com.au
Hi Roger ,
Has anyone developed a web interface where users could setup their own
find-me/follow-me services?
Yes, this is available on the ScopServ Telephony GUI (Commercial).
--
Joel Vandal, CTO
ScopServ Inc.
http://www.scopserv.com
this now sends me to my voicemail exten = _1NXXNXX,1,dial(SIP/3109439
...@sip.broadvoice.com,60) exten = _1NXXNXX,2,congestion() exten = _1NXXNXX,102,busy() Incoming works great! but a bit choppy, so if any one can help me get calls going out thanks in advanced.
-- Joel BuenoContact Me
it on the xml file but doesnt work.
I have never used CallManager, I presume that we need this to generate a
template SEP cnf.xml files ? Not found any documentations on Cisco
site about SIP or SCCP parameters that must be in this file.
--
Joel Vandal, CTO
ScopServ Inc.
http://www.scopserv.com
directory=/var/lib/asterisk/mohmp3
random=yes
Change mode=mp3 to mode=files then do a moh reload on CLI
--
Joel Vandal
ScopServ Inc.
http://www.scopserv.com/
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in order to see if it can fix some echo problems. We will do major
testing on theses card because we work on an appliance and hope that
this will solve some echo problem that we get with non-sangoma cards.
Will be able to give you more information in 1-2 weeks.
--
Joel Vandal
ScopServ Inc
PROTECTED])
--
Joel Vandal, CTO
ScopServ Inc.
http://www.scopserv.com/
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and sysconfig file for zaptel.
You will be able to control zaptel using chkconfig and service (service
zaptel stop/start). Same thing for Asterisk. (make config).
--
Joel Vandal, CTO
ScopServ Inc.
http://www.scopserv.com/
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