Hi,
Built a new server at the weekend and install Asterisk 1.6.0.9 and IAX and SIP
work great :) The one problem I am having is getting the OpenVox (TDM400 type
card) to work. It is successfully identified using WCTDM kernel module and
dahdi_scan picks it up just fine. The issue is when I
On Mon, Apr 27, 2009 at 07:37:05AM +0100, --[ UxBoD ]-- wrote:
Hi,
Built a new server at the weekend and install Asterisk 1.6.0.9 and
IAX and SIP work great :) The one problem I am having is getting the
OpenVox (TDM400 type card) to work. It is successfully identified
using WCTDM kernel
We use something like that in QueueMetrics to track outgoing calls for
call-centers:
http://forum.queuemetrics.com/index.php?topic=261.0
thanks
l.
2009/4/25 Sebastian s...@adinet.com.uy
Anyone thought about something like outgoing queues?
I mean, having same info that has for inbound queues
Hi,
I follow the
web,http://www.voip-info.org/wiki/view/Asterisk+config+musiconhold.conf
- mohstream.sh , to configure music on hold to play using mms but
failed. Anyone can play using mms?
ango
___
-- Bandwidth and Colocation Provided by
Shouldn’t the member has the statics per queue?
I mean, I have 2 queues test1 and test2, with member 1001 for example for
both queues, if I make a call to queue test1 and the member 1001 answers the
call, the statics for the member is up in both queues, (has taken 1 call….),
this should be
On 27/04/2009 4:22 p.m., Sam Hawkin wrote:
Hi,
Thanks for your reply.
I have tried as you suggested, I does not even come upto NoOp()
It hangups after AMD.
I have decreased the silence threshold from 256 to 100 and 50.
Try the NoOp in the h extension:
exten = h,1,NoOp(Status: ${AMDSTATUS}
Ok, and thats exactly what I mean monitoring outbound groups, so you can
have realtime info for monitoring.
And as with queues have the ability to reset the statics for monitoring
porpouses.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Hi,
If you are going to AMOOCON through Berlin Sunday evening and could
use a ride to Rostock, please feel free to email me. If you are are
going to be there I look forward to meeting you. I will be leaving
early Wednesday morning for Berlin as well. Reserve now and avoid the
rush :)
Hi,
I have some files in mp3 in my Asterisk but when I play it the volume is lo=
wer than wav files. Both the files (wav and mp3) are encoded with the same =
amplitude. In anothers players the audio volume of these files are equal.
Can I fix this diference between volume of mp3 and wav file?
On Sat, 25 Apr 2009 00:01:44 -0400
Michael van der Stoop mst...@dpia.ca wrote:
I call in once from a cell phone, which is
successful then I can call out with out issue.
It's a bug. Maybe this one?
http://bugs.digium.com/print_bug_page.php?bug_id=14577
Cheers,
--
|\ /|| |
Michael wrote:
Is it possible to force T38 for all invocations ReceiveFAX() ?
It already does that.
I can't with Digium fax, and it always fails at the point it decides to
switch
to T38.
You've posted two or three messages about this, but haven't included any
information we could use to
On Mon, 27 Apr 2009 00:33:44 +1200
Michael as...@nettrust.co.nz wrote:
I can't with Digium fax, and it always fails at the point it decides
to switch to T38.
Have you tried dedicating the line to fax only, no detection?
I tried using it, but for me it apparently fails the codec switch:
All;
I just came accross this problem, and I am a bit stumped. I am using Asterisk
1.4.23.1 and am using Asterisk Realtime Static for voicemail. I have not had a
problem before, but now when someone tries to leave a vm, I get the error No
format for saving voicemail? and Asterisk hangs up
cbbs...@hotmail.com schrieb:
All;
I just came accross this problem, and I am a bit stumped. I am using
Asterisk 1.4.23.1 and am using Asterisk Realtime Static for voicemail. I have
not had a problem before, but now when someone tries to leave a vm, I get the
error No format for saving
Greetings all,
This is a just-for-fun question. I was reading the
support forum and a fellow there wanted Read() to stop on * instead of #. I
thought that changing app_read.c would resolve this
current
if (tmp[x-1] == '#') {
tmp[x-1] = '\0';
break;
new
}if
On Apr 27, 2009, at 10:29 AM, Danny Nicholas wrote:
Greetings all,
This is a “just-for-fun” question. I was
reading the support forum and a fellow there wanted Read() to stop
on * instead of #. I thought that changing app_read.c would resolve
this
current
if
O boy. SIP infrastructure is so flexible that basically nobody gets
it right. :-)
You could easily have 20 different SIP network elements (/servers
/services). Even more.
And we get at least 5 new SIP-RFCs per day. They're all trying to
fix things which the previous specifications didn't address.
Didn't do mms but have implemented using Shoutcast. I have instructions
at the link below:
http://mark.hulber.com/voip/configuration/shoutcast-musiconhold-in-asterisk-16/
Rilawich Ango wrote:
Hi,
I follow the
web,http://www.voip-info.org/wiki/view/Asterisk+config+musiconhold.conf
-
Without having tried it I notice the output is x86-64 and not x86_64.
Could there be a typo somewhere?
sean darcy wrote:
1.6.1 svn 190575:
CC=cc CXX=g++ LD= AR= RANLIB= CFLAGS= make -C menuselect
CONFIGURE_SILENT=--silent menuselect
make[1]: Entering directory
I checked out the 190660 trunk and went all the way through make without
a problem.
Linux asterisk.hulber.com 2.6.18-128.1.6.el5 #1 SMP Tue Mar 24 12:05:57
EDT 2009 x86_64 x86_64 x86_64 GNU/Linux
--
Output through generating input for menuselect:
[r...@asterisk trunk]# ./configure
I've seen that message when then endpoint is not available.
Cary Fitch wrote:
[Apr 26 10:47:01] NOTICE[32151]: chan_sip.c:16223 sip_poke_noanswer: Peer
'3516533812' is now UNREACHABLE! Last qualify: 86
[Apr 26 10:47:11] NOTICE[32151]: chan_sip.c:12723 handle_response_peerpoke:
Peer
On Friday 24 April 2009 18:35:16 Atis Lezdins wrote:
Secondarily, MPEG audio compression takes a lot of CPU. Until the last
few years, desktop CPUs weren't even capable of doing realtime MPEG audio
compression, which is necessary if you're going to have the recording
ready by the time the
I followed the Ronald Lewis instructions and was able to get EC2 to run
Asterisk. I was able to use IAX2 so I'm not sure what you are saying.
You should also be able to build dahdi but of course you won't have any
physical devices in the machine. I think for meet-me dahdi provides a
On Mon, 27 Apr 2009, Danny Nicholas wrote:
This is a just-for-fun question. I was reading the
support forum and a fellow there wanted Read() to stop on * instead of
#. I thought that changing app_read.c would resolve this
Any chance features is getting in your way?
I have put canreinvite=no for all my internal SIP-clients in sip.conf
because I want Asterisk to be in the middle of the RTP-stream so he can
provide MusiconHold and so...
Now, what the Asterisk CLI tells me when I make a call from my one
internal SIP-phone to another internal SIP-phone is :
Daniel Hazelbaker wrote:
On Apr 27, 2009, at 10:29 AM, Danny Nicholas wrote:
Greetings all,
This is a “just-for-fun” question. I was reading
the support forum and a fellow there wanted Read() to stop on *
instead of #. I thought that changing app_read.c would resolve
jonas kellens wrote:
I have put canreinvite=no for all my internal SIP-clients in sip.conf
because I want Asterisk to be in the middle of the RTP-stream so he can
provide MusiconHold and so...
Now, what the Asterisk CLI tells me when I make a call from my one
internal SIP-phone to
Hey,
Just wondering if anyone can let me know what the status of IPv6 support
for Asterisk is currently. I see that the branch where development was
happening has gone away. I was trying:
http://svn.digium.com/svn/asterisk/team/blanchet/v6
Has this branched moved to somewhere else?
Cheers!
Hi list,
Anyone knows how to get free VoiP-in numbers from USA or Canada, I
have found some links for example sipnumber.com but it does not run.
Also I want to know how to configure it in my asterisk server.
Thanks in advance.
Regards
___
--
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I remember someone wrote a great document concerning Polycom server
provisioning that provided a way to ensure that updates to the firmware
did not overwrite customizations. I'll be damned if I can remember
where I saw it. It may have been
Barry L. Kline wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I remember someone wrote a great document concerning Polycom server
provisioning that provided a way to ensure that updates to the firmware
did not overwrite customizations. I'll be damned if I can remember
where I saw
Barry L. Kline wrote:
I remember someone wrote a great document concerning Polycom server
provisioning that provided a way to ensure that updates to the firmware
did not overwrite customizations. I'll be damned if I can remember
where I saw it. It may have been discussed during a VUC
On Tue, 2009-04-28 at 09:08 +1200, Andrew Ruthven wrote:
Hey,
Just wondering if anyone can let me know what the status of IPv6 support
for Asterisk is currently. I see that the branch where development was
happening has gone away. I was trying:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Kevin P. Fleming wrote:
It's easy; just don't edit the files that come with the firmware!
Hi Kevin.
That's the model I currently use. The one I'm interested in is linked
in Darrick's post below. It's an interesting approach.
Thanks for
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Darrick Hartman (lists) wrote:
That would be Karl Fife, of the famous Karl Fife experience.
http://kfife.com/voip/
That's what I'm looking for. Thanks Darrick!
Barry
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.5 (GNU/Linux)
On Apr 25, 2009, at 10:31 PM, Aryan Ameri wrote:
The second one, is built on a custom Fedora 8 image. The steps are not
repeatable on any other distro, not even a stock official Fedora 8
one. Fedora
8 itself is long EOLed and as such, not something I'd want to use on a
production server.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric
Chamberlain
Sent: April-27-09 7:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk EC2
On Apr 25,
Hi,
If anyone is interested in the low speed modems needed for POS
applications (V.22, V.22bis, V.22bisFC and V.29FC) please contact me. I
had some spare time while travelling, and finally got the V.22bis code I
started a long time ago into a start where its basically functional. I'm
now
Thanks. But I heard that mpg123 uses much resources (CPU memory) of
each connection. Is it true? How about using madplay?
On 4/28/09, M Hulber asterisk-ad...@hulber.com wrote:
Didn't do mms but have implemented using Shoutcast. I have instructions
at the link below:
On Tue Apr 28 2009 09:19:56 GMT+1000 (EST) Eric Chamberlain e...@rf.com wrote:
The original Feodra 8 image came from the Amazon EC2 team, they
optimized it to run in EC2. I chose the Amazon fc8 image, because I'm
not comfortable getting OS images from third-parties. When Amazon
Hi,
Thanks for your reply.
I have tried as you suggested.
In h extension it is giving Status as AMD_HANGUP.
Below is the log
-- Executing Answer(SIP/sip-874d, ) in new stack
-- Executing AMD(SIP/sip-874d, ) in new stack
-- AMD: SIP/sip-874d (null) (null) (Fmt: 4)
Apr 28
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