Hello, i have a problem with the hangup cause received from the AMI in the
Hangup events. All calls that arent answered after ringing are returning hangup
cause 16 (normal clearing) instead 19.
Im running asterisk 1.4.11, the calls are generated to a SIP peer using the AMI
originate command.
Dear all
I have recently install TE120P Digium E1 card now everything is fine
and working i have connect my asterisk with avaya but when anybody transfer
call from avaya i got this error on my asterisk consol
[Aug 27 14:46:50] WARNING[19527]: app_dial.c:741 wait_for_answer: Unable
Dear Philipp;
Kindly find the part of the configuration as below:
[general]
allow=all
disallow is comment by ( ; ).
[bilal_sip]
type=friend
context=internal
host=dynamic
canreinvite=no
dtmfmode=rfc2833
So where is the problem? The endpoint does not
register and nothing appear on trace level
Excellent! - posted on
http://oinko.net/astpligg/story.php?title=Asterisk_clusters_with_a_foneBRIDGE2
Thank you
l.
In data Sun, 26 Aug 2007 12:56:04 +0200, Vicente Aguilar
[EMAIL PROTECTED] ha scritto:
Hi
I've published my Asterisk/foneBRIDGE2/heartbeat setup: config files,
scripts...
Thank you David!
On 8/26/07, Klaverstyn, David C [EMAIL PROTECTED] wrote:
http://www.testforme.com/download/
I'll leave the files there for a few days.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stephen
Bosch
Sent: Monday, 27 August 2007
Nope, it only has chan_capi. I don't have any experience with AVM Fritz
cards so I'm afraid I can't help you with it. I think there is an
article on voip-info.org that explains howto use a Fritz card with
Asterisk.
Regards,
Patrick
Patrick thanks! I guess my question should have been
hi,
I am using debian 4.0 with version 2.6.18-4-686
I have downloaded the required files form site
asterisk-1.2.24.tar.gz
libmfcr2-0.0.3-1.4.tar.bz2
libsupertone-0.0.2.tar.gz
Seysan wrote:
Hi all,
I want to limit the outgoing trunk to certain extensions, so for example
6 extensions can call long distance, but 4 other extensions are not
allowed to do so.
How can I do it in FreePBX specially?
I don't know about Trixbox per say, but normally you would have all
On 26/08/07, Abhishek M S [EMAIL PROTECTED] wrote:
Dear Mr Galvin,
Gavin ;-)
Thank you for the links. Had gone through the bug tracker before though. I
was specifically referring to the schema for the driver 'Astirectory' and
not the one related to the real time LDAP driver for Open LDAP.
I see it is res_config_ldap. You'd be much better using the latest
version in the bug tracker.
On 27/08/07, Gavin Henry [EMAIL PROTECTED] wrote:
On 26/08/07, Abhishek M S [EMAIL PROTECTED] wrote:
Dear Mr Galvin,
Gavin ;-)
Thank you for the links. Had gone through the bug tracker before
Hi List;
I need to use an prepaid billing system with Asterisk,
and I do not know which one is more stable and
integrated with Asterisk?
A2Billing or AstBill or ASTCC?
Also, from where I can download it and ready about its
configuration?
Regards
ITS
IP Telephony and Contact Center Engineer
Hello,
Can someone outline what tutorials will be covered at this year's
AstiCon in AZ?
Are the tutorials going to be worthwhile for fellow Asterisk
users/admins that have been actively building, running, and
administering Asterisk boxes for years?
I have a suggestion for one demo that
bilal ghayyad wrote:
Hi List;
I need to use an prepaid billing system with Asterisk,
and I do not know which one is more stable and
integrated with Asterisk?
A2Billing or AstBill or ASTCC?
Also, from where I can download it and ready about its
configuration?
Regards
ITS
IP Telephony
On Mon, Aug 27, 2007 at 07:12:02AM -0400, Steve Totaro wrote:
Have you looked at ASTPP? Have not looked in a while but Darren had
plans to integrate it into OSCommerce and some other neat features. I
think he based it on the original ASTCC but has made some major
improvements.
Does it
This is what I did in Trixbox:
I added this to extensions_custom.conf
-
[restrict-local-only]
include = from-internal-additional-custom
include = app-recordings
include = app-callwaiting-cwoff
include = app-callwaiting-cwon
include = app-dialvm
include = app-vmmain
include =
What is happening ? Please email us the SIP Debug. Also with paging most phones
require a special SIP header for the phone to know that it has to pick up right
away.
- Original Message -
From: Stuart J. Newman
To: asterisk-users@lists.digium.com
Sent: Monday, August 13, 2007
Are there any speakerphones or other conferencing HW phones that play
the audio in stereo? Either their own speakers, or jacks for an amp with
room speakers? Is there any way for Asterisk to deliver call legs with
stereo channels in the RTP stream?
If not, is it possible for
although not stereo i believe its the closest you will get if the
codec is supported by asterisk. polycom has now HD codec
On 8/27/07, Matthew Rubenstein [EMAIL PROTECTED] wrote:
Are there any speakerphones or other conferencing HW phones that play
the audio in stereo? Either their own
Do any softphones run the HD codec? What exactly is the HD codec
technically called, and is there any info about its codec running inside
Asterisk?
On Mon, 2007-08-27 at 08:47 -0400, C F wrote:
although not stereo i believe its the closest you will get if the
codec is supported by
hi ppl, i'm using asterisk 1.2 because i'm making use of voiceone, but before i
was using asterisk
1.4 and had the same problem, it concerns an italian voip/sip provider called
eutelia/skypho, my
problem is the following one:
when i start my pbx my skypho account is working fine, meaning that
AFAIK the HD codec they use is the ITU-T G722.2 AKA GSM-AMR-WB, the
big improvement here is the sampling rate ( 16kHz ).
On 8/27/07, Matthew Rubenstein [EMAIL PROTECTED] wrote:
Do any softphones run the HD codec? What exactly is the HD codec
technically called, and is there any info
The codec is G722 I believe. and Polycom has a conference speaker
phone with a subwoofer option that has HD voice.
On 8/27/07, Matthew Rubenstein [EMAIL PROTECTED] wrote:
Do any softphones run the HD codec? What exactly is the HD codec
technically called, and is there any info about its
The HD Codec is just G.722
/b
On Aug 27, 2007, at 7:52 AM, Matthew Rubenstein wrote:
Do any softphones run the HD codec? What exactly is the HD codec
technically called, and is there any info about its codec running
inside
Asterisk?
On Mon, 2007-08-27 at 08:47 -0400, C F wrote:
The 601 has g722 (and its not g722.1 or .2)
/b
On Aug 27, 2007, at 8:14 AM, Bruce Reeves wrote:
The codec is G722 I believe. and Polycom has a conference speaker
phone with a subwoofer option that has HD voice.
On 8/27/07, Matthew Rubenstein [EMAIL PROTECTED] wrote:
Do any
FreeSWITCH supports 16k wideband conferences and supports G.722,
speex 16k and should work great with the phones that support it. I
have personally tested it with grandstream phones.
/b
On Aug 27, 2007, at 7:47 AM, C F wrote:
although not stereo i believe its the closest you will get if
On Thu, 2007-03-15 at 18:24 -0400, Steve Totaro wrote:
I am bringing up several Fedora Core 7 boxen into production now.
Besides a knee jerk reaction that Fedora Sucks, can someone give a
real argument as to why I should or should not use it for production?
(besides the several MB of yum
Am Montag, den 27.08.2007, 08:55 -0400 schrieb Jody Gugelhupf:
hi ppl, i'm using asterisk 1.2 because i'm making use of voiceone, but before
i was using asterisk
1.4 and had the same problem, it concerns an italian voip/sip provider called
eutelia/skypho, my
problem is the following one:
hi,
I am using debian 4.0 with version 2.6.18-4-686
I have downloaded the required files form site
asterisk-1.2.24.tar.gz
libmfcr2-0.0.3-1.4.tar.bz2
libsupertone-0.0.2.tar.gz
Hi,
In the early stages of deciding how to try and develop this environment, I
looked at all the protocols that could be used. SIP was chosen just because it
seemed to me that it was the most widely used protocol. I believe IAX is a new
protocol with a little less documentation and examples.
Anyone?
On Aug 24, 2007, at 3:35 PM, Ed Pastore wrote:
I understand this question is over-broad, but hopefully you can have
patience with a newbie and toss me a bone...
I am in the testing stage of deploying Asterisk. I have successfully
configured it to work behind the NAT of my ZyXEL
I have an entry for console/dsp in the dialplan.
When I call into that extension I get connected to the soundcard and I
hear myself etc...
everything is fine.
However, if I call in and get connected then a second call comes in they
also get connected.
I was expecting them to get a busy
Jerry Geis wrote:
However, if I call in and get connected then a second call comes in
they also get connected.
I was expecting them to get a busy signal or something...
Your dialplan needs to take this into account. I do the following:
;
On 8/26/07, Seysan [EMAIL PROTECTED] wrote:
Hello,
Let's say I have a Database of my clients about 50 clients, I want to
announce a new product or service to them, can asterisk do it for me? It is
something like a appointment reminder for doctors.
I want to know is there any software for
Steve Totaro wrote:
Can someone outline what tutorials will be covered at this year's
AstiCon in AZ?
Here is what is available so far:
http://www.astricon.net/files/2007-astricon-schedule.pdf
--
Russell Bryant
Software Engineer
Digium, Inc.
___
Thank you.
Now that the conexts are different can all the extension call to echother ?
Seysan
On 8/27/07, Steven [EMAIL PROTECTED] wrote:
This is what I did in Trixbox:
I added this to extensions_custom.conf
-
[restrict-local-only]
include =
Hi,
Then, If the number we called was busy, or he didn't pick up the phone, we
should call him again.
how we can keep track of those ?
On 8/27/07, Atis [EMAIL PROTECTED] wrote:
On 8/26/07, Seysan [EMAIL PROTECTED] wrote:
Hello,
Let's say I have a Database of my clients about 50
On Thu, 2007-03-15 at 18:24 -0400, Steve Totaro wrote:
[snip]
Besides a knee jerk reaction that Fedora Sucks, can someone give a
real argument as to why I should or should not use it for production?
(besides the several MB of yum updates daily, which to me is a good thing).
Steve,
Fedora 7
On 8/27/07, Seysan [EMAIL PROTECTED] wrote:
Hi,
Then, If the number we called was busy, or he didn't pick up the phone, we
should call him again.
how we can keep track of those ?
It's all described in link i gave.
There are MaxRetries and RetryTime parameters available, and you can
also
On 8/27/07, Atis [EMAIL PROTECTED] wrote:
On 8/27/07, Seysan [EMAIL PROTECTED] wrote:
Hi,
Then, If the number we called was busy, or he didn't pick up the phone, we
should call him again.
how we can keep track of those ?
It's all described in link i gave.
There are MaxRetries and
thank you
On 8/27/07, Atis [EMAIL PROTECTED] wrote:
On 8/27/07, Atis [EMAIL PROTECTED] wrote:
On 8/27/07, Seysan [EMAIL PROTECTED] wrote:
Hi,
Then, If the number we called was busy, or he didn't pick up the
phone, we
should call him again.
how we can keep track of those ?
On 8/27/07, Jared Smith [EMAIL PROTECTED] wrote:
Personally, I use CentOS (when I don't care about support) or RHEL (when
support is important to me) as my preferred server distribution, simply
because they guarantee to have *years* worth (at least five years!) of
security updates, even if I
Hi,
On 8/27/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
In the early stages of deciding how to try and develop this environment, I
looked at all the protocols that could be used. SIP was chosen just because
it seemed to me that it was the most widely used protocol. I believe IAX is
a new
They know what they are doing and do a lot of it. I don't have to give an
opinion myself. There is enough evidence all over for people to draw the
proper conclusions for themselves.
-Original Message-
From: C F [mailto:[EMAIL PROTECTED]
Sent: Sunday, August 26, 2007 4:39 PM
To:
And the FUD continues. Pain and misery eh? Google pain misery
insertmodel#here?
-Original Message-
From: Steve Underwood [mailto:[EMAIL PROTECTED]
Sent: Sunday, August 26, 2007 5:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PRI cards,
Thanks very much for the help, I appreciate it. Recently, one of my co-workers
and I have altered the code to just register with the Asterisk server and place
an audio call. This gets rid of the subscription part of the application, so I
do not get the 489 Bad Event error anymore. I believe
Steve Totaro wrote:
Andrew Joakimsen wrote:
On 8/21/07, Chris Mason (Lists) [EMAIL PROTECTED] wrote:
Steve Totaro wrote:
You should have no problems. Make sure you put surge protection and
ground your POTS lines. It is a small investment. I have had SEVERAL
FXO
Dear Mr Galvin,
As of today I am using the res_config_ldap of Astirectory in my test
Asterisk system to connect to a test LDAP database of my University. Things
seem to be working fine so far. Now I'm faced with the task of installing
this in the productive system. Before doing so, I'd sure like
Hello folks,
I'm interested in detecting tones on specific frequencies with
specific timing; for example, I'd like Asterisk to dial out and when
the channel starts/call connects, listen for a 1200Hz tone that plays
for 100ms.
Is this doable with Asterisk using something already extant? After
Dear all,
I'm faced with a similar situation of segregating users in 3 different
categories to be able to make: internal calls only (students); internal
local calls (staff); and internal, local international calls (profs). I do
understand that 3 different contexts would have to be defined in
On 27/08/07, Abhishek M S [EMAIL PROTECTED] wrote:
Dear Mr Galvin,
Gavin! ;-)
As of today I am using the res_config_ldap of Astirectory in my test
Asterisk system to connect to a test LDAP database of my University. Things
seem to be working fine so far. Now I'm faced with the task of
Hello list,
I have a customer who is interested in standardizing on dell servers for
asterisk deployments.
Has anyone had success with a particular configuration?
Anything specifically to watch out for?
Thank you for your time,
Art
Arthur Miller
Sr. Sales Associate
VoIP
I have used both the powedge line for large deployments and the
Optiplex N series for small offices. The only thing I have had to add
to the pc's is 12 power extensions at times and here lately I have
had a pc or 2 without the 4 pin molex connector so I had to find SATA
to molex adapters.
On
Hi,
On 8/27/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Thanks very much for the help, I appreciate it. Recently, one of my
co-workers and I have altered the code to just register with the Asterisk
server and place an audio call. This gets rid of the subscription part of
the
Arthur Miller wrote:
Hello list,
I have a customer who is interested in standardizing on dell servers
for asterisk deployments.
Has anyone had success with a particular configuration?
Anything specifically to watch out for?
Thank you for your time,
Art
Steve Totaro wrote:
Arthur Miller wrote:
Hello list,
I have a customer who is interested in standardizing on dell servers
for asterisk deployments.
Has anyone had success with a particular configuration?
Anything specifically to watch out for?
Thank you for your
Dear Mr Gavin,
Sorry for having miss pelt your name twice... Thank you once again for your
prompt reply. Is this the correct version of the driver for Asterisk 1.2.x :
res_config_ldap-v0.7.tar.gzhttp://bugs.digium.com/file_download.php?file_id=9565type=bug
from the link
Hi, About 2 years ago we made the decision to ship exclusively Dell
servers. Mostly we have shipped the 860 rackmount with a config of a
basic dual core proc couple gig of RAM and a pair of 75GB HDDs in RAID
1. And they are great but we put a limit of about 30 concurrent calls
through it.
That
Will this work on 1.2.x? I just installed it and did make samples.
The README references a file called html.conf which does not exist and
also abruptly ends with the word to on a blank line.
Besides that, what would the URL be for AsteriskNow? Is that
customizable in the elusive html.conf
On Mon, Aug 27, 2007 at 07:38:37PM -0400, Steve Totaro wrote:
Will this work on 1.2.x? I just installed it and did make samples.
Yeah. Just backport support for the manager over http, users.conf, and a
few other small things.
(read: no).
--
Tzafrir Cohen
icq#16849755
I've run up to 50 concurrent calls on the PE850 and PE860 using TE205p.
I also came across the te110p issue which manifests itself as popping and
crackling audio. It is rather insidious as zttest is fine, the problem does
not appear to be missed interrupts. In my case the Digium distributor
Hi:
I think app-conference is used where there isn't zaptel hardware,in the other
word when we use zaptel hardware we shouldn't use app-conference for conference
call sevice and we should use meetme application and load ztdummy.Is it true?
Best regards.
-
Hi:
I think app-conference is used where there isn't zaptel hardware,in the other
word when we use zaptel hardware we shouldn't use app-conference for conference
call sevice and we should use meetme application and load ztdummy.Is it true?
Best regards.
-
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