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I was just looking to see if anyone knows about an open source app
using the xml interface. I just started dabbling with the xml
interface a little bit and it helps to look at what others are doing.
I am looking for a console type app for the operator. Very simple
operations like
Has anyone seen or know of a iphone/ipod sip client that may be in the
works?
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I have a client that is using the Sangoma A200DE with two phone lines
attached.
The problem is:
They use their phone (Grandstream GXP2020) to dial out of the system.
Instead of getting ringing, there is someone on the other end of the
line that happened to dial in at the exact same moment.
-users
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I used TelIAX for a while and was happy with the service. I used it
for testing before we connected to our PRI...
http://www.teliax.com
On Feb 23, 2008, at 7:22 AM, Zeeshan Zakaria wrote:
I posted the same question on asterisk-biz mailing list but didn't
have much response. So I am
Does anyone have a audio file they would be willing to share for on hold
music?
I am looking for something like the old norstar beep every few seconds.
I tried 3 seconds silence, beep.wav, beep.wav. But it just didn't sound
right. I need one that has a softer beep.
Thanks!
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.
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http://www.shift8.biz
On Jan 25, 2008, at 3:47 AM, George Pajari wrote:
Has anyone experience with (or an educated guess of) the largest
paging
group that can be supported by the Page() command?
We have an installation coming up with 110 phones -- any hope to page
I am looking to see if anyone has seen this problem before. I am
setting the MEETME_RECORDINGFILE variable in a macro, then using the r
option with the Page application to record the page. But the page is
only recorded to the file specified in MEETME_RECORDINGFILE
sometimes...
Have a look at
serveremail = [EMAIL PROTECTED]
and
fromstring = The Asterisk PBX
in voicemail.conf.
On Dec 18, 2007, at 2:28 PM, shadowym wrote:
Is there a way to change the return path sendmail uses when sending
out
voicemail to email?
Currently the voicemails my asterisk system
This will also happen if there is a zap card installed and
unconfigured in zaptel.conf zapata.conf.
Forrest Beck
[EMAIL PROTECTED]
www.shift8.biz
dCAP
On Nov 12, 2007, at 9:46 AM, Stefan Guenther wrote:
Hello,
I have a strange situation:
I can talk to other SIP phones and via ISDN
,1)
[blacklistednumber]
; This is where a call will land if the macro-checkblacklist decides
that
; the number should not be allowed to dial DA
exten = s,1,Wait(2)
exten = s,2,Playback(privacy-you-are-blacklisted)
exten = s,3,HangUp()
Forrest Beck
[EMAIL PROTECTED]
www.shift8.biz
On Oct 18
if you are using presence. Maybe I will
setup presence on a couple phones and see if they reboot.
Forrest Beck
[EMAIL PROTECTED]
http://www.shift8.biz/blog
On Oct 11, 2007, at 4:34 PM, Jim Canfield wrote:
Joseph Begumisa wrote:
I had the same problem with 45 polycom 601 phones in the same
evaluating it.
Forrest Beck
[EMAIL PROTECTED]
www.shift8.biz
On Oct 8, 2007, at 4:51 PM, Erik Anderson wrote:
I am currently using a T1 PRI from TWTelecom for DID and outgoing
calls, but I recently discovered that they're offering call
termination/origination over SIP trunks in my area now
I have tried using sysconfig/asterisk but never had luck. I always
just edited the safe_asterisk script.
vi /usr/sbin/safe_asterisk and look for a line with -vvvc then add as
many v's you want.
You can also set it on the console with
core set verbose 7
Forrest Beck
[EMAIL PROTECTED
on the FXS/FXO cards?
Thanks !!
Forrest Beck
[EMAIL PROTECTED]
www.shift8.biz
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OK, I found the answer to my echo question (32ms).
But, has anyone used it? Feelings?
Forrest Beck
[EMAIL PROTECTED]
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Make the file the only one in the /var/lib/asterisk/moh directory.
Forrest Beck
[EMAIL PROTECTED]
www.shift8.biz
On Sep 26, 2007, at 3:07 AM, Joel Hill wrote:
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
of the arguments be what event
triggered the script. Like if it was a message was left, or some
logged out of VoicemailAdmin
Forrest Beck
[EMAIL PROTECTED]
www.shift8.biz
On Sep 24, 2007, at 10:36 PM, Forrest Beck wrote:
I have googled and can seem to find the answer to this one
Upgrade your kernel.
Run:
# uname -r
if you do not see smp in the kernel version
Run:
# yum update kernel kernel-devel
If you do see smp
Run:
# yum update kernel-smp kernel-smp-devel
Forrest Beck
[EMAIL PROTECTED]
www.shift8.biz
On Sep 25, 2007, at 10:53 AM, Tzafrir Cohen wrote:
On Tue
I have googled and can seem to find the answer to this one Does
anyone here have experience with externnotify in voicemail.conf?
The sample states that it will run when a message is delivered and
retrieved.
Does asterisk pass any arguments to the script?
Thanks.
Forrest Beck
of MeetMe Conference 177928251d into file meetme-conf-
rec-177928251d-1190380716.710.wav.
Forrest Beck
[EMAIL PROTECTED]
www.shift8.biz
Begin forwarded message:
From: Forrest Beck [EMAIL PROTECTED]
Date: September 20, 2007 5:37:22 PM EDT
To: Asterisk Users Mailing List - Non-Commercial
channel = 1-23
;Norstar T1 (SPAN 2)
context=norstar
group=3
signalling = pri_net
channel = 25-47
Forrest Beck
[EMAIL PROTECTED]
www.shift8.biz
On Sep 21, 2007, at 9:31 AM, Brian Alexander wrote:
On 9/20/07, Jared Smith [EMAIL PROTECTED] wrote:
I'd look at your wiring, as an HDLC error
setting the variable
MEETME_RECORDINGFILE and start placing the recordings in the sounds
directory named meetme-conf-rec.##.wav. Which is the default is
MEETME_RECORDINGFILE is not set.
Anyone seen this issue before?
Thanks!
Forrest Beck
[EMAIL PROTECTED]
www.shift8.biz
#!/bin
.
Defaults to INBOX
exten = 99,n,VoiceMailMain([EMAIL PROTECTED],s)
Forrest Beck
[EMAIL PROTECTED]
www.shift8.biz
On Sep 19, 2007, at 12:03 PM, Mark Michelson wrote:
rrgv wrote:
Hi
in asterisk 1.4, I need to cancel the password check and allow users
enter in the mailbox without entering
You mean in sip.conf?
Look at adding to your voip providers peer/user config incominglimit,
outgoinglimit or call-limit:
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf
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On Sep 18, 2007, at 4:26 PM, Jim Boykin wrote:
Is there a way
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.VOIP (8647)
1.514.312.7030
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.
This message was recorded
January
14th
at
10
42
pm
Thanks for any ideas you may have.
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IAXTEL: 17002871718
[EMAIL PROTECTED]
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Has anyone found a way to enable the g722 codec as a prefered codec in
the Polycom provisioning files for the 550's? I couldn't find a pref
for voice.codecPref.IP_550.
What needs to be put into the allow field (sip.conf) for asterisk to
allow the codec?
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IAXTEL: 17002871718
.
Thanks all!!
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})
exten = s,16,Hangup()
On 5/8/07, Remco Post [EMAIL PROTECTED] wrote:
Forrest Beck wrote:
I have all my SIP users in a realtime database. I would like to use
MySQL command to query the database and use the results from the query
to page all the phones found in the query.
The results from
argumen
On 5/4/07, Sergio (Red) [EMAIL PROTECTED] wrote:
Hi,
Do you know how see the peers statuses like: sip show peers but when sip
peers are configured by Relatime method.
Thanks
0xception escribió:
yes you can use the type friend
On 5/3/07, *Forrest Beck* [EMAIL PROTECTED]
mailto:[EMAIL
Nevermind. Friday and my mind has gone home! :)
I forgot the ipaddr and port setting in the table.
On 5/4/07, Forrest Beck [EMAIL PROTECTED] wrote:
Let me check my table Voicemail and CDR in the MySQL database
works fine. sip show peers isn't giving me anything. Only the one
peer I
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shouldn't need the
zttransode module since I don't have a codec translation card. right?
To work around this I added zttranscode to RMODULES in the zaptel init script.
If I don't need the zttranscode module. I may try and rebuild zaptel
without it.
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IAXTEL: 17002871718
[EMAIL
-0400, Forrest Beck wrote:
I was just looking to see if anyone else has seen this problem as well.
When asterisk starts up it loads the zttranscode module. The problem
exist when I use the init scripts to stop asterisk and then use the
zaptel init script to unload modules. Since the zaptel init
sorry, I meant modprobe.conf
On 5/3/07, Forrest Beck [EMAIL PROTECTED] wrote:
So is anyone not using the zaptel init script to load modules? Anyone
using modules.conf? How an I load them at boot without using the init
script? Do I just remove --ignore-install from modprobe?
Thanks
On 5/3
I setup sip realtime. Is it possible to use a type of friend? User
and Peer seem to work fine.
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Of Forrest
Beck
Sent: Tuesday, April 24, 2007 5:28 PM
To: Asterisk Users List
Subject: [asterisk-users] Voicemail on Different Server
I have two seperate systems at two different locations. Each hosts
there own voicemail for their phones.
I have thought about just having all voicemail on one
way. What would be the best way to set the DID for when
a extension dials out on the PRI? In sip.conf I am using CallerID as
their internal number.
I thought of maybe adding a key for each extension to the astdb and
have a Macro query the astdb. Any other ideas?
Thanks.
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if I could save myself a step.
This may be where I will need to switch to MySQL.
On 4/19/07, Alex Balashov [EMAIL PROTECTED] wrote:
On Thu, 19 Apr 2007, Forrest Beck said something to this effect:
I thought of maybe adding a key for each extension to the astdb and
have a Macro query
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. The failover here is that the polycom phones
will register with the gateway if the primary server isn't available.
They won't have all the features and voicemail, but at least they can
dial out and get 911 if needed.
What do you think? Do you have a better solution?
Thanks!!
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We are looking at about 200 total phones with low usage. Probably
only 20 or so calls at once.
On 4/11/07, Andrew Joakimsen [EMAIL PROTECTED] wrote:
On 4/11/07, Forrest Beck [EMAIL PROTECTED] wrote:
2) Have two servers with the same dialplan. One in each location.
Each server has it's own
was just looking to see if there was anything
else out there.
Thanks!
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all the phones get paged
and the script finishes?
Then I thought maybe a Macro in the dialplan to dial a global var of
the group of phones, but that won't work. If phone isn't available,
none will get paged.
Has anyone done this before? I just don't know where to start.
Thanks
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Forgot to mention.
We are using Polycom phones on asterisk 1.4.2
I tried the allpage agi, but it checks for all SIP peers connected to
the server.
On 3/30/07, Forrest Beck [EMAIL PROTECTED] wrote:
First off, A lot of thanks to this list. I have learned ton from
reading through the posts
,VoicemailMain(${ARG1}) ; If they press *, send the user
into VoicemailMain
[phones]
exten = _2XXX,1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
DID example:
2001 = 5552871701
2002 = 5552871702
2003 = 5552871703
Thanks!
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IAXTEL: 17002871718
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so PCI won't be used. I will probably use
a small 14 2U server to handle the ZAP Cards.
Does anyone for see a problem with using the 1950? Good/Bad thoughts???
Thanks!
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Any idea when 1.4.1 will be available. There is a bug fix in the cvs
head that I need, and I don't want to run the cvs build on a
production machine.
Thanks...
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is the variable for BTN if so?
Many Thanks.
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= yes
enabled = yes
Thanks all.!
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OK. problem solved. It was something dumb on my part. /var/agentx
didn't have enough permissions to let asterisk access the socket.
On 2/25/07, Forrest Beck [EMAIL PROTECTED] wrote:
I followed Marks SNMP howto on Voip Magazine and ran into a small
problem... (http://www.voip-magazine.com
/07, Sergio R. D'Ippolito [EMAIL PROTECTED] wrote:
How can i see if snmp is running ok on mi * box ?
Thanks in advance
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Forrest Beck
Enviado el: Domingo, 25 de Febrero de 2007 06:14 p.m.
Para: Asterisk Users List
Does anyone have some method, or AGI scripts that will automatically
call a list of numbers from a database and play a pre-recorded
message?
For example, you have a database of
FirstName LastName PhoneNumber
Jon
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IAXTEL: 17002871718
[EMAIL PROTECTED
the number, when
answered play a pre-recorded message.
It could be used to notify parents at a school that a after school
game is canceled.
I appreciate any direction you can point me in.
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IAXTEL: 17002871718
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as my IMAP flags which works fine for a Courier IMAP backend.
Courier uses Maildir (not mbx) which works just fine for me?
Cheers,
Ray
Forrest Beck wrote:
OK. I needed to remove the flags from the string. So I modified
app_voicemail.c and recompiled.
It is working now by, not using
I know some of this doesn't belong on this list, but I am just
including it for problem history.
I am trying to setup IMAP Voicemail with our email server.
We are using a non-standards based groupware server called FirstClass.
The server has some built in support for IMAP.
My problem seems to
: {ares.school.da.org:143/imap//user=fbeck}INBOX
Works: {ares.school.da.org:143/imap/user=fbeck}INBOX
Thanks again!
On 1/11/07, Forrest Beck [EMAIL PROTECTED] wrote:
I know some of this doesn't belong on this list, but I am just
including it for problem history.
I am trying to setup IMAP Voicemail with our email
/imap.
Then I removed ,imapflags after imapport.
My next hurdle is the mailbox format. It's not mbx, and crashes
asterisk after creating the mail message.
Thanks.
On 1/11/07, Forrest Beck [EMAIL PROTECTED] wrote:
Thanks Ray. I also noticed in some a post reply
http://www.mail-archive.com
I am using a extension to dial the console which has autoanswer
enabled. I am getting a strange warning, has anyone seen this before?
Nothing on Google, or Voip-Info
[Jan 9 13:50:05] WARNING[5009]: chan_oss.c:1048 oss_request:
oss_request ty console data 0x0xb7851e00 dsp
Call to device 'dsp'
I was just wondering what you all are doing for music on hold files
for best quality. I am not much of an expert on sound rates, bits,
stereo, mono, tracks, and all that jazz. Currently I am taking music
from a CD (our campus jazz band has recorded a CD), converting to WAV,
using Audacity to
PRI is just a standard used on the T1 medium. If you have a solid T1
between the locations. Why not use PRI on the T1. If the T1 is
dedicated point to point between locations, then you can use PRI on
the line dedicating one channel to signaling (d channel). If you
can't give up the 24th
You can get switchtype from your carrier.
On 1/4/07, Forrest Beck [EMAIL PROTECTED] wrote:
PRI is just a standard used on the T1 medium. If you have a solid T1
between the locations. Why not use PRI on the T1. If the T1 is
dedicated point to point between locations, then you can use PRI
|saycid=yes|dialout=fromvm|callback=fromvm|review=yes|operator=yes|envelope=yes|sayduration=yes|saydurationm=1
;4073 = 1099,Bianca Paige,[EMAIL PROTECTED],,delete=1
;4110 = 3443,Rob Flynn,[EMAIL PROTECTED]
;4235 = 1234,Jim Holmes,[EMAIL PROTECTED],,Tz=european
2503 = 2503,Forrest Beck,[EMAIL PROTECTED
I am not sure if this is what you are looking for, but I will give it
a shot. There may be a better way to do this but... I would use
agent Queues for your users. Your users can log into the Queue, so
that if the dialed user is not available, then it will drop the caller
into a Queue for a
You should look at the asterisk-addons package. There is a addon
module in the package called format_mp3 that will play your mp3 files
instead of using mpg123 (which is a dead project).
I just use sox to convert my mp3's to GSM with something like this:
/usr/bin/sox musicfile.mp3 -r 8000 -c1
Is this what you are looking for
exten = _9.,1,Set(CALLERID(num)=3045551212)
exten = _9.,n,Dial(ZAP/g2/${EXTEN:1})
On 12/20/06, Bruce Reeves [EMAIL PROTECTED] wrote:
Look at the digit map in your Polycom configuration files. I had the same
problem and had to chage the digit map to support
www.asteriskguru.com
On 12/12/06, blackwater dev [EMAIL PROTECTED] wrote:
Does anyone know of any good step by step tutorials on getting sip set up?
I have asterisk installed but can't seem to figure out how to get an account
set up and connect from my xTen phone so I can try the demo. The
Have a look at TIMEOUT(digit)
http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+DigitTimeout
On 12/7/06, Stefan-Michael. Guenther (in-put GbR) [EMAIL PROTECTED] wrote:
Hi,
the german telco Colt Telekom has assigned the phone number block 56830-xxx to
one of our customers. In the
You can run dnsmasq on the machine for local caching of the dns names.
(http://thekelleys.org.uk/dnsmasq/doc.html) and then apply this patch
that will allow dnsmasq to set a minimum time to live
(http://lists.thekelleys.org.uk/pipermail/dnsmasq-discuss/2005q2/000253.html).
dnsmasq can be then
I use the GoToIf:
If the SIP phone is Extension 2501 and dials out (I am using the
norstar 9 to dial out convention). BTW. ${PSTNOUT} is a global
variable for ZAP/G2.
exten = _9X.,1,GoToIf($[${CALLERIDNUM} = 2501]?2:3)
exten = _9X.,2,Set(CALLERID(num)=9195551212)
exten =
30 Channels on Verizon? Is this in the US? T1 (24 channels) or E1(30
channels)? Are you dialing from the top (g1) of the group or bottom
(G1)?
On 12/5/06, Klaverstyn, David C [EMAIL PROTECTED] wrote:
I have just installed Asterisk wit a TE110P card. I have configured 30
channels which
That kernel-devel fix is just for ZAPTEL. The bug has been solved in 4.4
On 12/5/06, Vicky [EMAIL PROTECTED] wrote:
I am not sure but i think that fix is for compiling zaptel not asterisk .
Asterisk runs on centos with 0 problems :)
On 05/12/06, varun [EMAIL PROTECTED] wrote:
Thanks Karl.
I have a two port TE205P Digium card. I have set everything up to
create a native zap bridge between the two spans. Everything works
perfectly except one thing. Our telco has a password that has to be
entered as soon as a long distance call is made. So if I dial a long
distance call from my
Talk to the folks at Asteria. The have a product called Reign. It
looks just like your old interface, runs off .NET as a client on the
machine.
http://www.asteriasgi.com/pbx/reign
On 11/7/06, Stephen Wingfield [EMAIL PROTECTED] wrote:
Andres,
The Bicom Systems Operator Panel is probably
I am not sure if I am going to use SIP registration's or just specify
the host ip address in sip.conf. Are there any pros or cons to the
two? My phones will have a static IP address and won't be changed
unless a admin does it. So the logical path would be to just turn off
registration on the
What does zap show channels show? Are all the channels shown as in
use? What is set in zapata.conf for resetinterval= ? If anything. I
believe that resetinterval is used to reset unused channels for any
channels that are left open.
On 10/31/06, Asterisk [EMAIL PROTECTED] wrote:
Hi All,
When I look at TTY9 (using init.d and safe_asterisk to start the
asterisk process), I am getting some strange characters. When a
application is run the and the CLI shows the application executing the
languange almost looks russian...??
Anyone seen this before?
I need some help with AGI. I am unsure how it is written and works.
But I have a bash command that will spit out a two digit numerical
value (The temperature in the room). The bash command is:
#!/bin/bash
/usr/local/digitemp/digitemp-1.3/digitemp -a | tail -n1 | cut -d -f9 | cut
-d . -f1
What's in zapata.conf?
On 10/13/06, Remi Quezada [EMAIL PROTECTED] wrote:
When I reload the asterisk I get the following warnings:
Oct 13 08:29:17 WARNING[10170]: chan_zap.c:10874 setup_zap: Ignoring
switchtype
Oct 13 08:29:17 WARNING[10170]: chan_zap.c:10874 setup_zap: Ignoring
signalling
Oct
Has anyone used the Polycom HDvoice phone yet? I am curious if it
uses a different codec. Does it actually sound any better?
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You could use heartbeat http://www.linux-ha.org (or ultramonkey
http://ultramonkey.org). With this you set up a director that shares
the load to multiple servers. You can even set it to have consistent
connections so a originating IP will return the the same server. I
have hearbeat running on
@lists.digium.com
Sent: Tuesday, October 10, 2006 12:31 PM
Subject: Re: [asterisk-users] Echo Cancel Cards
Joseph wrote:
On Mon, 2006-10-09 at 20:41 -0400, Forrest Beck wrote:
Anyone using the echo cancelation cards from digium? We are using the
single span T1 card with out echo cancel and I was curious
Anyone using the echo cancelation cards from digium? We are using the
single span T1 card with out echo cancel and I was curious if it was
worth the money.
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I want to setup a asterisk server with two T1 spans (two TE110P
cards). The server will have one card connected to the PRI and the
other will connect to our Norstar Meridian ICS system. I want to have
a very simple dial plan for the context that the PRI card will be
assigned to something like
http://www.voipsupply.com/home.php
On 10/4/06, Devraj Mukherjee [EMAIL PROTECTED] wrote:
Nokia E series with proper firmware upgrade :)
On 10/5/06, Steve Glaus [EMAIL PROTECTED] wrote:
bilal ghayyad wrote:
Hi List;
I would like to know where I can find the IP Phones
that can be used
I am a little stumped on this one and it may be because my brain is
ready for the weekend. I am trying to set an extension for forwarding
all calls to voicemail. So if a user set's their phone to forward all
calls to extension 2000 it will drop the caller in the user's
voicemail box.
I
Nevermind. Just decided to use:
exten = _22XXX,1,Voicemail(u${EXTEN:[EMAIL PROTECTED])
On 10/6/06, Forrest Beck [EMAIL PROTECTED] wrote:
I am a little stumped on this one and it may be because my brain is
ready for the weekend. I am trying to set an extension for forwarding
all calls
build libpri.
On 10/3/06, Eugeniy Khvastunov [EMAIL PROTECTED] wrote:
yusuf пишет:
Eugeniy Khvastunov wrote:
Hello!
Why Asterisk tell: Unknown signalling method 'pri_cpe'
Why the asterisk does not know such signaling method?
[chan_zap.so] = (Zapata Telephony)
Oct 3 13:04:02
a cold/hot spare. I will post again if I have
luck.
On 9/26/06, Kevin Kiely [EMAIL PROTECTED] wrote:
Forrest,
I noticed your post on the mailing list and was curious if you had used that
server before with asterisk with any TDM cards in it?
Kevin
-Original Message-
From: Forrest Beck
can't remember the last time that I had to reboot on of them.
G.711 G.729 is built in.
James Taylor
1-903-691-0069
- Original Message -
From: Forrest Beck [EMAIL PROTECTED]
To: Asterisk Users List asterisk-users@lists.digium.com
Sent: Tuesday, September 26, 2006 12:50 PM
Subject
I am thinking of using a mini atx 1u server with a digium zaptel
(wcte11xp) installed to act as a SIP gateway. This way any of my
asterisk servers can forward calls to any gateway (seperated by about
3miles of fiber). Has anyone else tried this? I would just load a
basic asteisk config and
Setting the callerid should be passed from asterisk through the zap
channel. Have a look at CALLERID
http://www.voip-info.org/wiki-Asterisk+cmd+SetCallerID
I can speak for a 23channel PRI, not sure about analog.
On 9/22/06, Kevin Steil [EMAIL PROTECTED] wrote:
I would like to tie outbound
This suggestion may be sort of a hodge podge setup, but you could use
something like a airport express, which has wireless bridging built
in. Connected directly to a ATA
On 9/22/06, Brian Candler [EMAIL PROTECTED] wrote:
Sorry, one other equipment query: does anyone know of an ATA with
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