Hello Bryant
Extension h is worked in any case of hangup.
It not important to that the call was answered or no.
It also be more flexible, if you use instead of ${DIALSTATUS}use
${HANGUPCAUSE}, while in most of cause ${DIALSTATUS} is returned the
same return code.
Hi,
Lots of VoIP, SIP and Asterisk-related discussion and some free phones
and Polycom software today, join us at the usual place:
http://www.voipusersconference.org
Call sip:200...@login.zipdx.com Skype:vuc.me (via Skype for Asterisk
and PhonefromHere.com)
Listen Live 16khz mp3 Stream:
On Fri, 2010-12-24 at 07:52 +1300, CB wrote:
Could anyone recommend some documentation regarding Asterisk 1.8 and the
realtime architecture? Specifically I want to know if it is possible to set
a priority label or to use n as a priority for realtime extensions in
Asterisk 1.8? My understanding
Dear listers,
I'm facing a puzzling situation with Digium TDM2400 card (12 FXO / 12
FXS). Once a day or so we detect 1 or 2 zombie FXO channels. These can be
either outbound or inbound calls. I thought this could be related to
obsolete DAHDI or Asterisk versions, so I upgraded to 2.4.0 and
Hi All;
A2Billing is working fine for Asterisk, but in case I need to use Asterisk and
Gnugk and I need to manage the accounts and the billing from one Database and
one billing system, so I need a prepaid billing that can work with both.
Which prepaid billing (open source) can be used to work
Friends,
Do we need to change any Asterisk configuration files (Or any file related
to Asterisk for that matter) when we put Asterisk box from one network to
another?
It is assumed that DB is on the same box.
Asterisk box has got Asterisk running in it with no issues.
Probably, it should not
Friends,
Do we need to change any Asterisk configuration files (Or any file related
to Asterisk for that matter) when we put Asterisk box from one network to
another?
It is assumed that DB is on the same box.
Asterisk box has got Asterisk running in it with no issues.
Probably, it should not
If you set bindaddr in any conf file you will need to change the IP address
there.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Dec 24, 2010, at 4:30 AM, Asterisk Man wrote:
Friends,
Do we need to change any Asterisk configuration files (Or any file related
Hi,
Is it possible to use a live audio stream in asterisk
I want to call a number and then hear an external audio stream.
For example http://www.radioveronica.nl/radioveronicaplayer/radioveronica.asx
I thought it was possible to use musiconhold, but I did not get it working.
This is my
If a call is hung up before an answer our h extension is not running in our
dial macro
Bryant
On Dec 24, 2010, at 3:47 AM, Vardan Harutyunyan hvarda...@gmail.com wrote:
Hello Bryant
Extension h is worked in any case of hangup.
It not important to that the call was answered or no.
It also
Hi,
We had 2 asterisk 1.4 connected together in iax, all was fine. One of
them was upgraded (server and Asterisk) in 1.6.2.15, the other end is in
1.4.38
When calling to 1.4 to 1.6.2 -remember, it's iax- all is good. But
calling from 1.6.2 to 1.4 give a bad audio to calling party (words are
If on the dial command you add option g, if the call is not answered, it will
fall through to the next statement which can be a hangup command and then it
will go to the h extension. If that does not then make the statement after the
dial command a goto h extension.
--
Jim Dickenson
I have my asterisk Server A registered as a client with another asterisk
Server B.
When I place a call from Server A to B I get the following:
WARNING[834]: chan_sip.c:12673 check_auth: username mismatch, have SS72,
digest has openbts1
NOTICE[834]: chan_sip.c:19961 handle_request_invite: Failed
On Wed, Dec 22, 2010 at 11:41:52PM -0800, Matt Darnell wrote:
Is there a way to forward a message to multiple people from within the
telephone user interface? Now there is only the ability to forward to
an individual.
I see there is a way to leave a message for multiple people using the
In AEL macro you must use catch h
for example
macro DialToSIPProv (tech,number,prov) {
Dial(${tech}/${numb...@${prov});
switch(${DIALSTATUS}) {
case BUSY:
Noop(BUSY);
[Do some one]
break;
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel Tryba
Sent: Friday, December 24, 2010 9:17 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Forward voicemail
On 24 December 2010 14:40, Administrator TOOTAI ad...@tootai.net wrote:
Hi,
We had 2 asterisk 1.4 connected together in iax, all was fine. One of them
was upgraded (server and Asterisk) in 1.6.2.15, the other end is in 1.4.38
When calling to 1.4 to 1.6.2 -remember, it's iax- all is good. But
On 23 December 2010 18:01, Steve Davies davies...@gmail.com wrote:
Hi Again,
I thought I had this sorted, but it appears that in a clean
environment I did not in fact fix it. There appears to be a bit of a
contradiction.
1) In 1.6.2.x, musiconhold requires DAHDI (which we have)
2) In
Doug Lytle wrote:
I'll let you know what I come up with, hopefully before the weekend ends.
Bruce,
I gave it a shot this weekend. It's very specific to whatever distro
they were using, most of the path information and program location
weren't found under Mandriva. The area where they
I am using the g option and it does not run the next statement or h extension
if the caller hangs up before an answers or time out event occurs during a
dial comand.
Bryant
On Dec 24, 2010, at 9:55 AM, Jim Dickenson dicken...@cfmc.com wrote:
If on the dial command you add option g, if the
Thanks for looking into it. Yes, it missed up and not worth looking at it.
Unfortuantly, so are a few products from the same company (probably trying
to make money of support which I understand)but it seems they released
an install script which is here for CentOS:
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