Displaying 20 results from an estimated 11000 matches similar to: "TDM400P digium card"
2006 Jan 26
4
extension to extension dialing
Sorry for all the newbie questions. I really appreciate everyone's help
today.
Okay I've got outgoing and incoming calls working with no echo. yay! Now
I'm having an issue with SIP extension to extension calling. Any time I
dial another extension it goes right into voice mail. My
extensions.conf is pretty small and rough but, here's what I have right
now. Most of it was taken
2006 Feb 09
11
Asterisk vs. Traditional PBX
Hi everyone !
So here's my question of the day ! I need to make a decision on whether or not to go to a voip solution or configure an existing pbx (norstar) that my company has available. We are a small startup. I'm wanting a solution that will support up to about 200 people, with direct dial-in capability, up to about 30 concurrent phone calls and good voice quality. Right now I have
2006 Jan 26
3
snom 320 echo problems
Hi there -
I'm having some echo problems on my snom 320 phones. Anybody experience
this before ? I don't have any issues with the sipura 841s I have
though.
Any help is greatly appreciated.
Thanks !
Nora Lavelle
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2006 Feb 03
1
Calls fading in and out
Hi again everyone !
Was wondering if anyone had any pointers on how to debug voice quality
issues in asterisk. I've got a user who either can't be heard on her
phone calls (outgoing and incoming) and today someone that called her
said that her voice was coming in and out. Any pointers or suggestions
are appreciated!
Thanks so much again ! This list has been so helpful to me.
2011 Feb 18
2
no progress indication
I am integrating a new server (Asterisk 1.8.2.3, DAHDI 2.4.0) with VOIP
only trunks, and this server only has soft phones.
When I dial an extension and the phone is not registered, I don't get any
ring or progress indications, and eventually the Dial() times out and
drops into voicemail (as expected).
CLI output:
-- Executing [s at macro-StdExten:6] Dial("IAX2/barneveld-2036",
2007 Nov 29
2
Using existing extensions.conf macros, and co-habitation
I'm trying to set up my extensions.conf file using some of the existing
macros like stdexten, etc. while at the same time having two logically
separate virtual PBX's (with no "default" context) and two trunks coming
into separate contexts, i.e. one for residence and one for my at-home
business.
I noticed, however, that macro-stdexten depends on the "default" context:
2003 Jul 03
3
Using switch =>
hello,
I have a test setup with 2 asterisk servers, each having a one snom 100
via sip using it. I`m experimenting on how trunking between them would
work. I have them setup for RSA authentication which I plan to use in
the future.
So I`ve setup the keys and servers seem authenticate to each other. One
is named phila and other hurricane.
Here is what I see on phila:
-- Registered
2004 Sep 13
2
unavail and busy.
Hi guys,
What is different and the "context" to play unavail message and busy
message?
When a SIP connection is unregistered, voicemail will play unavail message,
right?
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2008 Mar 13
5
Newbie One-touch Recording: Does not work
I thought it was quite easy to implement but I cannot get one-touch
recording to work. Here are the changes what I did:
I restarted Asterisk after the change (because reload does not work for
changes in features.conf).
I press *1 on the Polycom IP600 phone to record a conversation but no
new wav file appear in /var/spool/asterisk/monitor or elsewhere.
Any suggestions?
Here is the console log:
2003 Dec 30
2
playback in [macro-stdexten] problem
I added the playback line to my [macro-stdexten] context but when I dail
an extension I don't get the "please hold while I try that extension"
message. It just dials the extexsion. Do I have a syntax problem
somewhere ?
exten => 8005,1,Macro(stdexten,8005,Zap/2)
exten => 8006,1,Macro(stdexten,8006,Sip/8006)
[macro-stdexten]
;
; Standard extension macro:
; ${ARG1} -
2008 Mar 13
3
Newbie One-touch Recording: Does not work (more info)
I thought it was quite easy to implement but I cannot get one-touch
recording to work. Here are the changes what I did:
I restarted Asterisk after the change (because reload does not work for
changes in features.conf).
I press *1 on the Polycom IP600 phone to record a conversation but no
new wav file appear in /var/spool/asterisk/monitor or elsewhere.
Test A: Outside line calling in
2009 Feb 12
4
Asterisk Queue and URL Calling
Dear All
I want to integrate sugarcrm and asterisk , so when customer call the call
center the agent or extension which answers the call , before pickup the
phone and talk to customer , view his/her information if it is available.
I do this as described below :
1-Setup login username for sugarcrm for each extension
2-Extension Users will login to the sugarcrm.
3-Develop php script to handle new
2004 Dec 11
2
help with detecting fax.
I have Spandsp working fine. Asterisk sees a fax on the zap port and redirects the call to the fax-in area.
This works if I have a simple dialing rules that goes answers first and waits 10 secs then goes to the next item. If it hears a fax it goes to the right place. Here is a sample that works.
[incoming]
exten => 2019,1,Goto(test,s,1)
[test]
exten => s,1,answer
exten => s,2,wait(5)
2004 May 04
1
Asterisk and windows h.323 gatekeeper calling problems...
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hi there, i have a working Microsoft ISA firewall with buildin H.323
Gatekeeper....
So Far, i got registerd the asterisk on the M$ Gatekeeper...
here is the h.323 configuration:
; Open H.323 driver configuration
;
[general]
port = 1720
bindaddr = 0.0.0.0
allow=all ; turns on all installed codecs
dtmfmode=rfc2833
gatekeeper =
2004 Dec 20
1
Example config for SPA-1001
Hi,
Has anyone managed to create a setup with a Sipura SPA-1001 as a client?
Right now I can connect to the device by dialing the extension number
but when I try to connect from the phone handset to make an outbound
call it gives an unavailable tone.
I'm using Line 2 on the SPA-1001 to connect to the local asterisk
server, line 1 is used to connect to my VOIP provider until I can get
the
2007 Dec 10
3
One server, multiple companies
Hello all,
Just starting to setup asterisk v 1.4.11 and need to run three distinct phone systems for three different companies.
So far, I have inbound lines going to the appropriate dial plan within the extensions.conf file. I'm using
exten => _X.,1,NoOp(FROM NUMBER: ${SIP_HEADER(TO):5:10})
to determine which number is being dialed by the caller and then using a gotoif to get to
2005 Jun 02
1
Newbie :Call Forwarding problem
Dear All,
I was trying to enable call forwarding, following the steps of the link
on voip.org regarding this issue it doesn't work and the phone I am
trying to implement on is still ringing. below is my conf in
extensions.conf and the CLI output during the process.
My configuration is :
exten => _*5X.,1,DBput(CF/${CALLERIDNUM}=${EXTEN:2})
exten => _*5X.,2,Hangup
exten =>
2005 Jan 18
1
Asterisk and IAX softphone (firefly) problem/question
Quick question from a newbie,
I have asterisk configured to dial IAX extensions (which works). When
dialing from one IAX extension (using Firefly) to another IAX extension
(also using Firefly), the Firefly client rings on the receiving end and
gives the option of accepting or denying the call. However, when I dial in
to Asterix using a VoicePulse number and dial the same extension Firefly
2004 Jun 15
3
anyone use mailboxexists?
I replied to a post of mine a few days ago asking of anyone uses
mailboxexists(). I haven't received any replies.
Perhaps few use it or perhaps the reply was overlooked. I thought I'd
post the question one last time before giving up on it for now...
Thanks!
-Michael
2004 Jun 26
2
Newbie needs help
I've been banging my head on a brick wall for about an hour now trying
to understand why the following doesn't work (which is even provided as
an example in the distribution!).
The goal is to create a voicemail-only extension not associated with a
phone. I'd rather not have an extension dedicated to VoicemailMain(),
so I would like the user to be able to hit '*' during