Hello,
1. is it possible that Asterisk does not translate between codecs H263
and H264 ?
2 If I set videosupport=yes in sip.conf [general], can I turn off the
video support on a peer ?
Kind regards,
Jonas.
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_
--
1. is it possible that Asterisk does not translate between codecs H263 and
H264 ?
Yes, I do not think that the plain (unpatched) Asterisk does video transcoding
at all - and do
note that video transcoding is a very heavy task.
Philipp
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Am 12.12.2010 20:49, schrieb dave george:
I am using Asterisk 1.6.2.5-0
running on ubuntu and I have a problem passing called ID
on calls to the PSTN
On Sun, 12 Dec 2010 23:49:50 +0100, Hans Witvliet h...@a-domani.nl
wrote:
I don't know what their price-range is, (just going through their site)
Other alternative i heard about, is the DSL-modems from AVM.
What i heard, is that you can use the 7170 and 7270 (perhaps their
latest models also) as
I want to use asterisk for testing PC-1.5 MTA.
I have few questions related to the capabilitied to asterisk-1.8
1. Does it provide the support for all Call-5 features- like call hold, call
transefer, call forwart, DND, SCB, Hotline, NBCS, Emergency caling etc.
2. Does it support IPSec, IKE and
sean darcy wrote:
But there is no /var/spool/asterisk/voicemail/default/100/unavail.gsm'
The voice mail application usually saves in both .gsm and .wav by
default. (At least under 1.4.x) I have several unavail.gsm files in
the mail folders of several phones.
Doug
--
Ben Franklin quote:
Le 13/12/2010 11:43, Gilles a écrit :
[...]
In case someone from France follows this thread, I'm interested in any
feedback about professional-grade ADSL that supports VoIP, as a
serious alternative to ISDN for telephony
We are selling our own xDSL but a France Telecom Pro can do the job.
Hi!
Thanks for the infos. I'll check out the AVM appliances.
www.avm.de/en/Produkte/FRITZBox/FRITZ_Box_Fon_WLAN_7270/
There is a German and an international (English) version, and there are three
hardware
versions with slightly different features. In Germany ISDN is typically Annex
B, but
On Thu, Dec 09, 2010 at 07:57:37AM -0600, Joe Greco wrote:
Specifically looking for examples of (or how to generate)
1) .*No registration for peer '.*' (from HOST)
2) .*Host HOST failed MD5 authentication for '.*' (.*)
3) .*Failed to authenticate user .*@HOST.*
If anyone who is
Does anyone have a super simple cookbook describing the steps to
integrate Mail into an Asterisk Dial Plan.
I have googled but have a lot of choppy results. I am running RH and
Asterisk 1.8
Cheers
Tom
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_
-- Bandwidth and
On Mon, 13 Dec 2010 12:06:56 +0100, Administrator TOOTAI
ad...@tootai.net wrote:
We are selling our own xDSL but a France Telecom Pro can do the job.
Always dedicate the ADSL line to VoIP, use the right codec and you will
have the quality you need. In big towns, some of our cutomers uses ADSL
On Mon, 13 Dec 2010 12:14:26 +0100,
klitz...@pool.informatik.rwth-aachen.de wrote:
The built-in SIP proxy is made for inside LAN usage, although there are ways
to make the
box also accept SIP UAs on the Internet as local phones. Do not expect too
many features
for these IP phones, for example
Hi,
Recently, I made issues after upgrading libpri from 1.4.11.2 to 1.4.11.5.
After coming back to 1.4.11.2, everything ran successfully.
Unfortunately, I didn't have much time to spend to investigate but I hope
I'll soon get an opportunity to try 1.4.11.5 again on another telco line.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thomas Perron
Sent: Monday, December 13, 2010 5:48 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Mail Integration
Does anyone have a super
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy
Sent: Sunday, December 12, 2010 8:04 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] 1.8.1: playing imaginary sound files
On 13 Dec 2010, at 14:25, Danny Nicholas wrote:
(god forbid) postal mail
Haha, I'm kind of tempted to write an app_cups module to print envelopes ;)
S
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Email integrated to voicemail.
Thanks for the nice humor
My bust.
How do I set up an Exchange or other Mail MX server to interoperate
with VoiceMail?
On Mon, Dec 13, 2010 at 9:25 AM, Danny Nicholas da...@debsinc.com wrote:
-Original Message-
From:
On Mon, Dec 13, 2010 at 11:49 AM, Thomas Perron thomas.per...@gmail.com wrote:
Email integrated to voicemail.
Thanks for the nice humor
My bust.
How do I set up an Exchange or other Mail MX server to interoperate
with VoiceMail?
You mean IMAP Storage...
Postal mail... heh... nice :)
On 12-13-2010 15:49, Thomas Perron wrote:
How do I set up an Exchange or other Mail MX server to interoperate
with VoiceMail?
Not sure if this is an asterisk issue at all. After setting the trivial
options in voicemail.conf, it's really just SMTP and relaying
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thomas Perron
Sent: Monday, December 13, 2010 8:49 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Mail Integration
Hello
I was wondering if someone knew of an application that could check
that the user has a firewall and a broadband connection that will work
OK with Asterisk and VoIP.
The app would first perform some bandwith + jitter tests, and will
then call a STUN server to check that the firewall isn't
Jonas,
Sorry for the late response. In fact I don't rely on Asterisk settings for
video codecs, I rather prefer controlling these in specific devices. The
reason being that I experienced video codecs priorities were not always
respected by client devices. What's even funnier, for GXV3140, if I
Hi!
BTW, is Asterisk now STUN-capable, or is it still to map ports
manually on the firewall to connect it to a VOSP trunk?
Until Asterisk 1.8 STUN support was faulty, and in 1.8 it has been corrected
(?) and strongly
limited. Search the asterisk-dev mailing list archive for STUN and do the
Hello,
I seem to be having an issue with voice mail on Asterisk 1.6.2.15 (file
storage). Whenever someone leaves a message that is distributed to another box
(like VoiceMail(100010011002,u)), but the VM never gets distributed to the
intended recipients. Instead, I get the following in the
Hello
This is a newbie question : With a simple Asterisk server on a private
LAN, an FXO port to handle the PSTN, and an ADSL connection to the
Net, ie. with no VOSP in the mix... how should I configure Asterisk so
that SIP clients can dial SIP numbers on the Net, such as those below
to perform
On Mon, 13 Dec 2010 18:35:03 +0100,
klitz...@pool.informatik.rwth-aachen.de wrote:
Until Asterisk 1.8 STUN support was faulty, and in 1.8 it has been corrected
(?) and strongly
limited. Search the asterisk-dev mailing list archive for STUN and do the same
in the Asterisk
bug tracker for more
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonathan C.
Bailey
Sent: Monday, December 13, 2010 12:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Voice mail
I assume you mean passing the context with the box number? I just tried that
and no dice..
-Jon
- Original Message -
From: Danny Nicholas da...@debsinc.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, December 13, 2010
On Mon, Dec 13, 2010 at 07:28:58PM +0100, Gilles wrote:
This is a newbie question : With a simple Asterisk server on a private
LAN, an FXO port to handle the PSTN, and an ADSL connection to the
Net, ie. with no VOSP in the mix... how should I configure Asterisk so
that SIP clients can dial SIP
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonathan C.
Bailey
Sent: Monday, December 13, 2010 12:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Voice mail
1001 and 1002 work individually. 1000 receives messages fine if you remove the
delete=1 (although the messages still don't get copied)..
This worked fine previously in the 1.4.x series (can't remember the exact
revision at the moment).
- Original Message -
From: Danny Nicholas
Hi Godson Gera,
thank you for your answer.
if i understand correctly, the EventMask filter all until i define all event
that i need.
This is not really helpful, because i must define the categories that i need.
mhh
is there another Solution for my Problem?
Thanks a lot
Daniel
Am 09.12.2010
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonathan C.
Bailey
Sent: Monday, December 13, 2010 2:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Voice mail
And that is more or less what I'm using.. Odd.. Maybe it's time for a bug
report, although I'd like to track down WHAT is causing the issue first...
-Jon
- Original Message -
From: Danny Nicholas da...@debsinc.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonathan C.
Bailey
Sent: Monday, December 13, 2010 2:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Voice mail
Le 13/12/10 19:00, asterisk-users-requ...@lists.digium.com a écrit :
Date: Mon, 13 Dec 2010 12:00:09 -0600 (CST)
From: Jonathan C. Bailey jbai...@co.marshall.ia.us
Subject: [asterisk-users] Voice mail distribution - missing messages
To: Asterisk Users Mailing List - Non-Commercial Discussion
Hi Everyone,
I ocassionally hear echo, static, and garbled voice when calling extension
to extension between two office (different geographic locations connected
using OpenVPN - 1 with DSL and other with T1 - 1500 KM apart). I am guessing
it's a bandwidth or jitter issue that is giving me faint
Hello Friends,
I am trying to Installl dahdi on amazon EC2 which have Open-SUSE-11.1 X86
version.
and here is snap of uname- a command
*Linux ip-10-160-86-41 2.6.32.19-0.3-ec2 #1 SMP 2010-09-17 20:28:21 +0200
x86_64 x86_64 x86_64 GNU/Linux*
when I try to run DAHDI distribution
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