Displaying 20 results from an estimated 2000 matches similar to: "extension to extension dialing"
2006 Feb 27
7
TDM400P digium card
Okay everyone -
I'm moving away from using sipura 841 phones. I'm starting to test with
Polycom IP 501 phones. We plan to upgrade our server to a dual t1 but,
for now we have a digium tdm400P with 4 analog lines coming into it. So
my question is will upgrading the IP phones with my existing digium
tdm400 card be enough to satisfy my users ? or is it really a combo
deal needing to
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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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:
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:
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 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
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",
2012 Jul 12
1
Asterisk with OpenBTS and mobile phone
Hello mailinglist,
I want to connect Asterisk with OpenBTS and make a call with a mobile
phone.
I use:
Ubuntu 11.10 + Kernel 3.0.22
GnuRadio 3.3.0
Asterisk 1.8.13
OpenBTS 2.8
Nokia Mobile Phone
OpenBTS works and I can send sms from the OpenBTS server to the
mobile phone. What I also need is a call between Asterisk and OpenBTS.
I have also two soft phones which works with Asterisk. And also
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
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
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:
; ${ARG1} -
2004 Jun 26
2
Newbie needs help
I've been banging my head on a brick wall for about an hour now trying
to understand why the following doesn't work (which is even provided as
an example in the distribution!).
The goal is to create a voicemail-only extension not associated with a
phone. I'd rather not have an extension dedicated to VoicemailMain(),
so I would like the user to be able to hit '*' during
2005 May 16
1
Dial plan - does not stop after first match
My dial plan seems to work great - in that when I call extensions 1234
it connects to 1234. Strangely, after the call terminates (the other
side hangs up first), Asterisk continues in the same context and then
matches to extensions _. which causes an invalid extension error!
Why does asterisk not leave the context (called internalmenu) after the
remote hangup? Instead, it continues to the
2005 Aug 27
2
Problems with registration
My phone still says Not-Registered. I have a Polycom SoundPoint 600 SIP
phone.
Here is my sip.conf file:
;
; SIP Configuration
;
[general]
context=default ; Default context for incoming calls
port=5060 ;added
bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes ;
2006 Feb 03
1
Calls fading in and out
Hi again everyone !
Was wondering if anyone had any pointers on how to debug voice quality
issues in asterisk. I've got a user who either can't be heard on her
phone calls (outgoing and incoming) and today someone that called her
said that her voice was coming in and out. Any pointers or suggestions
are appreciated!
Thanks so much again ! This list has been so helpful to me.
2004 May 04
1
Asterisk and windows h.323 gatekeeper calling problems...
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hi there, i have a working Microsoft ISA firewall with buildin H.323
Gatekeeper....
So Far, i got registerd the asterisk on the M$ 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 =
2010 Apr 29
2
Code in extensions.conf to leave a voice mail in another PBX ?!
Hi Guys,
i spent some time to figure this out (since i love how dialplan is written)
but i decided to ask for your help guys.
i have two asterisk servers one is 1.2 the other is 1.4, from 1.4 (pbx1) to
1.2 (pbx2) i can leave a voice mail without any pb, but from pbx2 to pbx1 it
just hang up.
in pbx2 extensions.conf:
i am using: exten => 8029,1,Dial(IAX2/pbx1/${EXTEN},20,tTWwr)
in pbx1, i
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