First of all I apologize for emailing everyone in one mass email like this,
but it is the only logical way to get this done. We have restarted the
Kickstarter campaign in hopes of raising the funds needed to get us into
the studio with a national producer.
PLEASE DONATE IF YOU CAN!
No Donation
I am trying to figure out the best way to deal with this. I want all of the
calls in the network to be G729 and this is working. I do have hardware
that provides me 30 g729 licenses. I am setting each extensions to
disallow=all and allow=g729. However when I have this setup, I get no voice
mail
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, 2012 2:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
I have installed and configures this card in asterisk 1.6. When trying to
load the module codec_sangoma.so I see the following in the asterisk log.
[2012-06-04 15:50:31] WARNING[18168] loader.c: Error loading module
'codec_sangoma.so': /usr/lib/asterisk/modules/codec_sangoma.so: undefined
symbol:
I have found numerous claims that 1.8 can do T.38 gateway with a patch,
however I am yet to find the patch or any instructions on implementing it.
Anyone have a link?
--
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I realize that faxing is not great with voip but here is my confusion. I
have been working on a web based fax system for 2 weeks. During this time I
have sent over 100 2 page faxes without any errors. Now today as things are
finally completed I can not seem to get any fax to go through unless it
*I have been testing this for a week now, and I am still struggling to make
it work. Here is the output from extension 11 just to show that permissions
are correct and asterisk can access faxnotify.php*
-- Executing [11@outb2:1] Verbose(SIP/616818-02f6,
Testing) in new stack
Jian,
I tried that and I get the results seem the same. It appears to run but does
not.
exten = 777,1,AGI(faxnotify.php,NOTIFY tim.compnetw...@gmail.com
616555 24/08/11 : 09:00:00 FAX_SUCCESS 1)
exten = 777,n,Playback(vm-goodbye)
exten = 777,n,Hangup()
-- Executing [777@outb2:1]
What does this mean? The suggestion from Jian did resolve the issue. Thank
You.
On Wed, Aug 24, 2011 at 3:07 PM, Steve Edwards asterisk@sedwards.comwrote:
Un-top-posting...
On 11-08-24 10:21 AM, Tim King wrote:
I have been testing this for a week now, and I am still struggling
When using the following my sytems commands to not seem to execute.
[outboundfax]
; exten = s,1,NoOp(send a fax)
exten = s,1,Set(FAXOPT(filename)=${FAXFILE})
exten = s,n,Set(FAXOPT(ecm)=yes)
exten = s,n,Set(FAXOPT(headerinfo)=${FAXHEADER})
exten = s,n,Set(FAXOPT(localstationid)=${LOCALID})
exten
My message with the configuration attached is awaiting moderator approval. I
will try to paste relevant data here.
*sip.conf*
[general]
context=inbound ;
allowoverlap=no
udpbindaddr=0.0.0.0
tcpenable=no
tcpbindaddr=0.0.0.0
srvlookup=yes
disallow=all
allow=ulaw
allow=alaw
dtmfmode
are the addresses of the devices in this conversation? so that you can
match the traffic to device
Cheers Duncan
On 10/03/2011, at 1:20 PM, Tim King wrote:
It looks like this:
19:18:34.782016 IP 199.173.66.22.53102 74.204.4.5.11732: UDP, length 172
19:18:34.789527 IP 74.204.4.5.11732
Still not working now that audio is restored jitter makes it inaudible? I
am ready to move this to commercial if someone can tell me how I need to pay
for support,
Thanks
Tim
On Thu, Mar 10, 2011 at 10:19 AM, Tim King t...@compnetwork.net wrote:
It looks like the issue was my provider
I am having trouble with no return audio on inbound calls. I have been
working on this for 18 hours and even built a fresh server and moved
everything over and I am getting the same results. I need someone that can
help get this resolved tonight. I can provide access to all machines
involved.
I have a very simple setup with two SIP routes to my carrier. I need to have
every other phone call placed to that carrier go to a different address.
This is what I need the call flow to look like. I have spent many hours
searching and have not found a working example.
Call1 exten =
.
On Thu, Oct 28, 2010 at 11:24 AM, Roger Burton West ro...@firedrake.orgwrote:
On Thu, Oct 28, 2010 at 11:08:12AM -0400, Tim King wrote:
I have a very simple setup with two SIP routes to my carrier. I need to
have
every other phone call placed to that carrier go to a different address.
I
It appears there is something wrong with my set command?
On Thu, Oct 28, 2010 at 2:15 PM, Tilghman Lesher tles...@digium.com wrote:
On Thursday 28 October 2010 13:06:00 Gordon Henderson wrote:
On Thu, 28 Oct 2010, Tim King wrote:
On Thu, Oct 28, 2010 at 11:24 AM, Roger Burton West
ro
, Tim King t...@compnetwork.net wrote:
I updated it as follows and I am still only getting the SayNumber(2)
[tim]
exten =
_X.,1,GotoIf(${SET(DB(avoics/route)=$[!0${DB(avoics/route)}])}?route1:route2)
exten = _X.,n(route1),SayNumber(1)
exten = _X.,n,Hangup()
exten = _X.,n(route2),SayNumber(2
These guys are pretty close. http://www.ss7box.com/
On Wed, Nov 4, 2009 at 4:16 AM, Khaled W Chehab kche...@xplorium.comwrote:
Dears,
Do Asterisk support SS7 SIGTRAN(SS7 over IP) protocol ?
And how to integrate
Regards
*Khaled Chehab*
* NGN Eng.*
When I make an outbound call I hear a half of a ring and than silence until
the call opens up.
It seems asterisk is sending a 183 after the 180 message. My CPE device does
not support multiple 18x messages in the same call setup. When we receive
the 180 we present ring back to the phone, but
Did you use ./Setup dahdi when installing the wanpipe drivers?
http://wiki.sangoma.com/wanpipe-linux-asterisk-dahdi
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asterisk-users mailing list
To UNSUBSCRIBE or update options
I thought that was it and tried each setting and did not see any change on
the line.
On Wed, Oct 28, 2009 at 3:58 PM, Kevin P. Fleming kpflem...@digium.comwrote:
Tim King wrote:
When I make an outbound call I hear a half of a ring and than silence
until the call opens up.
It seems
the project I will put a server on the
net with a card in it.
Let's make this happen.
Tim King
CEO
http://www.compnetwork.net/ CNS_LOGO_Beveled
7589 Cottonwood Drive Suite C
Jenison, MI 49428
Phone 616.301.3290Fax: 616.667.1104
Website: http://www.compnetwork.net/ http
and forums up soon.
Thanks for the support
Tim King
CEO
7589 Cottonwood Drive Suite C
Jenison, MI 49428
Phone 616.301.3290Fax: 616.667.1104
Website: http://www.compnetwork.net
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Baji
Panchumarti
I am looking for help from someone familiar with using asterisk and openser
to build a rather large VOIP network. I have 6 servers in place each with
their own purpose. I will give a brief summary and hopefully someone out
there is able to help be finalize this dialplan. I have six servers in
I have a fairly large system to configure. I was hoping to find someone
locally to employ for this project but remote configuration is considerable.
Pleas let me know if you are interested and have the time.
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up the local POTS line and call out. The other issue is the faxes, if I
could use asterisk to distribute the faxes and use voip from the stores to send
them out via asterisk I could save thousands there alone. Let me know if anyone
has had a similar setup. Thanks in advance!!
Regards
Tim King
Try lspci -vb
See if you can find you digium card and what interrupt it is running on.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of somesh s
Sent: Friday, September 23, 2005 5:49 AM
To: Asterisk Users
Cc: [EMAIL PROTECTED]
Subject: [Asterisk-Users]
What version of asterisk do I need to be running for this to
work? I have 1.0.9 running and when I try to install asterisk-addons from CVS I
get app_addon_sql_mysql.c23:19: mysql.h: No such file or directory.
So of course it fails to install that add-on. What am I
missing? I can find info
Well guys here comes the fun part. I have a Microsoft access
(VBA) application that interfaces with my SQL database. This app pulls of info from
the SQL record and than picks up the phone and dials that locations number. I
have purchased a few hundred NpaNxxs for my own use. I want get
I need to find someone to work with me in the Grand Rapids
Michigan Area. Someone good with Linux and Asterisk would be ideal. Please get
me contact info if you are interested.
Thanks
Tim
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Subject: Re: [Asterisk-Users] Outbound Extension problem
Have to answer your inbound call first - I suspect
On Aug 4, 2005, at 5:28 PM, Tim King wrote:
[macro-dialout-trunk]
exten = s,1,GotoIf($[foo${ARG3} = foo]?3:2)) ; arg3 is pattern
password
exten = s,2,Authenticate(${ARG3})
exten = s
New problem, I figured out how to get the extension working and
internally it works just fine. If I pick up a phone and hit 501 my cell starts
ringing. However if an inbound caller dials that extension Everything seems to
stop when it trys to bridge the two trunks together. Sound familiar
Of Tim King
Sent: Thursday, August 04, 2005
2:56 PM
To: 'Asterisk
Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Outbound
Extension problem
New problem, I figured out how to get the extension working
and internally it works just fine. If I pick up a phone and hit 501
the
call to a offsite phone number utilizing my Zap Trunk. Im sure someone has
done this already. Anyone want to point me in the right direction?
Tim King
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Ok everyone, I received help from Manny last night and it would seem that
asterisk at home failed to properly configure itself from the 1.3 ISO. All
of the configurations were correct. We downloaded the AAH package and
reinstalled it over the top of itself and than recompiled the kernel. Upon
to a offsite phone number utilizing my Zap Trunk. Im sure someone has
done this already. Anyone want to point me in the right direction?
Tim King
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- Original Message -
From: Tim King
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Sent: Wednesday, August
03, 2005 10:12 AM
Subject: [Asterisk-Users]
Transfer to outside line.
Finally got everything up and run with the help of Manny
Wise
How many times do you ask for help here before getting a respone? Every
single thing I do No matter what I get busy extensions. I am willing to pay
someone to help here. Anybody got a clue?
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Is this creating a problem because of the two FXO ports
being in the middle of the FXS ports?
Thanks
Tim King
Network Engineer
Computer Network Solutions LLC
1331 Plainfield Ave
Grand Rapids MI 49505
Phone: 800-669-3290
___
Asterisk-Users mailing
extension to the other, when you
dial-out? or all of the above?
Subject: [Asterisk-Users] WHat does it take
From: Tim King [EMAIL PROTECTED]
Date: Tue, 2 Aug 2005 11:28:37 -0400
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
.
Thanks
Tim King
Network Engineer
Computer Network Solutions LLC
1331 Plainfield Ave
Grand Rapids MI 49505
Phone: 800-669-3290
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I seem to have almost everything working now. The only
problem is all of my extensions seem to be busy. I can call out, but not in.
Can someone point me to the settings in the extensions file that could cause
this.
Thanks in advance guys.
Tim King
Network Engineer
Computer Network
Here is the output. These are Panasonic KX-TG2564s.
Does something need to be set for the phones? I can call out fine, but all of
the extensions seem to be busy.
Starting simple switch on 'Zap/5-1'
-- Executing Macro(Zap/5-1,
exten-vm|[EMAIL PROTECTED]|200) in new stack
-- Executing
Thanks for the response they are zap
extensions on Digium TDM40B and TDM22B
pbx*CLI zap show channel 3
Channel: 3
File Descriptor: 13
Span: 1
Extension:
Dialing: no
Context: from-internal
Caller ID string: Tim King
200
Destroy: 0
InAlarm: 0
Signalling Type: FXO Kewlstart
I was reading a thread where you were helping someone out
and noticed it ended without resolve. Was this issue ever taken care of?I seem
to be having the exact same problem.
Thanks
Tim King
Network Engineer
Computer Network Solutions LLC
1331 Plainfield Ave
Grand Rapids MI
.
Tim King
Network Engineer
Computer Network Solutions LLC
1331 Plainfield Ave
Grand Rapids MI 49505
Phone: 800-669-3290
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