Hi Karl,
that's funny you are asking this, am also currently looking at how to
solve the g722 codec negotiation riddle, in my particular case to play
nicely together with a KonfTel 300 IP conference phone.
> In other words, incoming calls are easy since codecs are negotiated
> from least-known
What is the current best practice to avoid transcoding on an outgoing call
to a
party whose codec preference is not known in advance?
In other words, incoming calls are easy since codecs are negotiated from
least-known (the remote party) to most-known (my endpoint) and my codecs can
simply be pre
Hi,
Does using a different codec affect the volume of the voice?
i was testing g711 and g729, voice seems to be softer on g729 compared
to g711. sorry not really familiar on how codecs work.
regards
Ron
___
-- Bandwidth and Colocation Provided by ht
Hello everyone,
I am trying Asterisk could manage codecs negotiations. I have some
telephones that supports g723.1 and G711, while others only support
G711. I would like, due to BW usage, that telephones supporting g723.1
used that codec in all calls between them but using g711 while
connecting to
Hi all,
Another simple question: does it make sense to use the append option in
MixMonitor (,a) when the codec is gsm? Or it works only when the codec
is an uncompressed one like ulaw, alaw or slin?
Thanks,
--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center
___
Sorry, I misread your message as "incoming" and "outgoing" calls.
Mr. Jones wrote:
Hi Eric,
I'm confused on where I would put this?
I'm also confused on how this would help with external calls (which we
want to be g729) vs internal calls to voicemail (which appear to need
to be g711)?
Thanks
Hi Eric,
I'm confused on where I would put this?
I'm also confused on how this would help with external calls (which we
want to be g729) vs internal calls to voicemail (which appear to need
to be g711)?
Thanks a ton!
Brian
On 9/19/06, Eric ManxPower Wieling <[EMAIL PROTECTED]> wrote:
Use typ
Use type=user for inbound and type=peer for outbound. Have different
codec settings for each of them.
Mr. Jones wrote:
Hi Folks,
We're trying to roll Asterisk out to production and are having a few
complications.
Most specifically we have G711 for our inbound origination, but would
prefer G7
Hi Folks,
We're trying to roll Asterisk out to production and are having a few
complications.
Most specifically we have G711 for our inbound origination, but would
prefer G729 for outbound termination, so far so good - it appears that
dtmfmode=auto works in both cases.
The area I'm having troub
I used codec_g729.so in stable realease so i set g729
with th highest priority .
With Asterisk SVN-trunk-r41990 i don't allow g729
Harry
--- Tzafrir Cohen <[EMAIL PROTECTED]> a écrit :
> On Fri, Sep 08, 2006 at 11:00:56AM +0200,
> [EMAIL PROTECTED] wrote:
> > Hello,
> >
> > I recorded some fil
On Fri, Sep 08, 2006 at 11:00:56AM +0200, [EMAIL PROTECTED] wrote:
> Hello,
>
> I recorded some files (gsm format) but i can not
> hear these files without g729
Any chance that you try to play them to a channel that uses a g729
codec?
I believe that this requires a separate g729 codec instance.
--- [EMAIL PROTECTED] a écrit :
> Hello,
>
> I recorded some files (gsm format) but i can not
> hear these files without g729
>
> -- Executing [EMAIL PROTECTED]:1]
> Answer("SIP/86-08218198",
> "") in new stack
> -- Executing [EMA
Hello,
I recorded some files (gsm format) but i can not
hear these files without g729
-- Executing [EMAIL PROTECTED]:1] Answer("SIP/86-08218198",
"") in new stack
-- Executing [EMAIL PROTECTED]:2] Dial("SIP/86-08218198",
"Sip/84|30|
On Mar 16, 2006, at 3:24 AM, Aisling wrote:
Hi everyone, I have an issue which is kind of a catch 22 situation. I had outgoing calls to my new PSTN provider working perfectly. Then I started focussing on incoming calls. It seems that I can solve an error which gets my incoming calls working but t
Hi everyone,
I have an issue which is kind
of a catch 22 situation. I had outgoing calls to my new PSTN provider working
perfectly. Then I started focussing on incoming calls. It seems that I can
solve an error which gets my incoming calls working but that in turns means my
outgoing cal
Hi all,I have an * box dual Xeon, 4Gb ram, 2 A104.Normally I use gsm codec, but to allow using faxes, I let some users to use g711 as default codec.My question is:Is it possible to detect what a certain call is?
So if is a phone call I'll use gsm, if is a fax I'll use g711.Thanks to all-- .:FaberK:
de Moises Silva
Envoyé : mardi 10 janvier 2006 22:51
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [Asterisk-Users] codecs order and so on
Doing in the console "show translation" i can see that it seems not be
possible to translate from any to g729 codec, or
Doing in the console "show translation" i can see that it seems not be
possible to translate from any to g729 codec, or from g729 to any. So,
let me try to find a reason for this.
When you have first allow=g729 (preferred codec)
all the calls to pstn providers work because the phones and asteris
Title: Message
The problem
:
an asterisk box
with 2 fxo
First fxo just
receive calls from pstn (ulaw)
Second fxo receive
and send call to mobile network thru a sipbox(ulaw)
Calls to pstn are
sent to a pstn provider accepting only g729
Internal calls
doesn't care of codec
> Hi all i have some problems with my pbx and asterisk codecs.
>
> if i use g711u or g711a codecs. the line never hangup. and the origin
> and destination are connected until i restart my pbx or asterisk
>
> But if i use GSM all work fine.
>
> is possible to solve this problem? or use only gsm
Hi all i have some problems with my pbx and asterisk codecs.
if i use g711u or g711a codecs. the line never hangup. and the origin
and destination are connected until i restart my pbx or asterisk
But if i use GSM all work fine.
is possible to solve this problem? or use only gsm codec?
--
.-
Unfortunately, we are on sip :(
Olivier
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de William Lloyd
Envoyé : mercredi 9 novembre 2005 18:12
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [Asterisk-Users] Codecs problem
If you want convert file audio, you using this on line
apllication:
http://www.asteriskguru.com/tools/audio_conversion.php
--- "Eric \"ManxPower\" Wieling" <[EMAIL PROTECTED]> ha
scritto:
> Olivier Taylor wrote:
> > User-agents have g729, g723.1 and gsm, isn't it
> possible to force user-agent
I've found that happens when one version of asterisk is 1.2 and the
other end is running 1.0.9 and you are connecting over IAX2.
If you bridge the two servers with SIP it will be fine.
-bill
On 9-Nov-05, at 11:52 AM, Olivier Taylor wrote:
That's a call to pstn
Callee and caller have 9729 b
That's a call to pstn
Callee and caller have 9729 but asterisk (astlinux and soekris) tell me that
there is no match and give me an error :(
Any idea?
Kind regards,
Olivier
9 headers, 11 lines
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 82.146.123.246:38098
Olivier Taylor wrote:
User-agents have g729, g723.1 and gsm, isn't it possible to force user-agent
to use gsm for voicemail and g729 for outbound calls?
No.
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Aster
: Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: RE : [Asterisk-Users] codecs
You simply need to have g729/g723 codecs. Asterisk comes with gsm by
default.
Regards,
Sahil Gupta
VoiceValley
On Wed, 9 Nov 2005, Olivier Taylor wrote:
> Right,
>
> I must suppose
ailto:[EMAIL PROTECTED] De la part de Angelito
Manansala
Envoyé : mercredi 9 novembre 2005 12:28
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [Asterisk-Users] codecs
i think gsm you mention is gsm sound files not gsm codecs.
On 11/9/05, Olivier Taylor <[EMAIL P
: Re: [Asterisk-Users] codecs
i think gsm you mention is gsm sound files not gsm codecs.
On 11/9/05, Olivier Taylor <[EMAIL PROTECTED]> wrote:
> Hi all,
>
> We use asterisk as a local pbx and we connect to a pstn/sip provider
> for calls to pstn.
>
> Since the messages
i think gsm you mention is gsm sound files not gsm codecs.
On 11/9/05, Olivier Taylor <[EMAIL PROTECTED]> wrote:
> Hi all,
>
> We use asterisk as a local pbx and we connect to a pstn/sip provider for
> calls to pstn.
>
> Since the messages on asterisk are on gsm format, we need gsm, but to call
>
Hi all,
We use asterisk as a local pbx and we connect to a pstn/sip provider for
calls to pstn.
Since the messages on asterisk are on gsm format, we need gsm, but to call
pstn, we need g729 or g723.
How can we enable both codecs to be able to call pstn and hearing voicemail
messages for example?
As someone that spent a week or more with anthm refactoring this code I can tell this is how it was when we were done and the code was accepted. So I do know a bit about this area of sip and iax./bOn Aug 16, 2005, at 3:03 PM, Pavel Jezek wrote:I remember many discussions about inteligent codecs ne
I remember many discussions about inteligent codecs negotiation in
asterisk, but seems, however, this isn't as simple to implement as it
looks... :-(
PJ
Erik Versaevel wrote:
That should be controllable by a weight, for example 2 peers:
A --> G729, G711
B --> G711, G729
What's currently hap
The way I said is the "gospel" of how it happens. /bOn Aug 16, 2005, at 1:42 AM, Erik Versaevel wrote:That should be controllable by a weight, for example 2 peers: ___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/ma
That should be controllable by a weight, for example 2 peers:
A --> G729, G711
B --> G711, G729
What's currently happening is that * starts transcoding between the two
(g729 for A and G711 for B), what i would like is to apply a weight to
peer A so that the codec of choise at both sides becomes t
Here is an example:
Call comes in via PSTN... ulaw is the native format of the channel.
On the sip side you have g729,ulaw as the codec order. That call
will end up being ulaw because we send the native format as our first
choice above all because we don't want to transcode.
/b
On Au
Pavel Jezek wrote:
Hi,
asterisk will negotiate codecs for both parties independently (use sip
show peer and look for "codec order" entry), so, if you have
prefered codec g729 for your sip phone/peer, asterisk will use them
(regardles of codec setting for other party - if codecs does not matc
Hi,
asterisk will negotiate codecs for both parties independently (use sip
show peer and look for "codec order" entry), so, if you have
prefered codec g729 for your sip phone/peer, asterisk will use them
(regardles of codec setting for other party - if codecs does not match,
asterisk will tr
hi,
i have this topology
pstn+(e1)asterisk1<->asterisk2<->sip client
asterisk1,asterisk2 allow (g729,alaw)
sip client prefer g729, then alaw
can you someone describe codec negotiation when call for sip client arrive
from pstn? (can i set g729 for calls from pstn? )
thanks
-
] On Behalf Of Tim Pushor
Sent: Monday, July 18, 2005 9:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Codecs and bandwidth
Hi Friends,
Something I'd like to shed some light on if possible - how is it that a
single ISDN conversation only uses
If you include down + up, yes, it's actually about 150-160 using uLaw +
IP/UDP/RTP/signaling overhead. But that's a little misleading, I think.
1/2 of that (~75-80) is down, 1/2 of that (~75-80) is up. So if you
have, say, a 1.5Mbps down/384 up DSL connection, you can do up to 4
calls simult
Of course - ISDN is bi-directional. I guess saying that ULAW takes 130K+
bandwidth depending on the framing type (local lan, w/1 hop, vlan, etc)
is not very clear. Thats total bandwidth. With lots of us at home and
small business using asynchronous connections - we need to keep that in
consider
Tim Pushor wrote:
Hi Friends,
Something I'd like to shed some light on if possible - how is it that a
single ISDN conversation only uses 64K for bidirectional communication
(using ulaw, correct?), but on several occasions now have seen
references to ulaw voip conversations using 64K per side
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim Pushor
Sent: Monday, July 18, 2005 9:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Codecs and bandwidth
Hi Friends,
Something I'd like to shed some lig
You are correct. Bandwidth is bidirectional. All those references
mentioned in the thread may be misleading. However, the bottom line
is that it does use 64Kbps up/down plus overhead. This does not mean
that to transport a single conversation you need ~150Kbps. You simply
need to make sure
Yeah that makes perfect sense, and was the way that I was initially
calculating bandwidth requirements and codec costs. I just found it odd
that bandwidth was reported both in simplex and duplex. It just confused
me (which doesn't seem to be too difficult, these days ;-)
Thanks,
Tim
Dan Per
> If you include down + up, yes, it's actually about 150-160 using uLaw +
> IP/UDP/RTP/signaling overhead. But that's a little misleading, I think.
>
> 1/2 of that (~75-80) is down, 1/2 of that (~75-80) is up. So if you
> have, say, a 1.5Mbps down/384 up DSL connection, you can do up to 4
>
Hi Friends,
Something I'd like to shed some light on if possible - how is it that a
single ISDN conversation only uses 64K for bidirectional communication
(using ulaw, correct?), but on several occasions now have seen
references to ulaw voip conversations using 64K per side of the
conversatio
Hi all,
I am trying to make a call from an X-Pro with only G.729 codec enabled to another with both G.711 and g.729. The Asterisk version is 1.0.3 and canreeinvite is set to Yes. What happens is that I get an 403 - Forbidden response and setting verbose 10 in Asterisk I can see the message:
May 2
Clive, cool - winter is getting quite near ova here...
Well, how would I find out what is happening - I mean how do I know
what * is connecting with to net2phone.
"...They have their own proprietry protocol..."
I thought it was because of the G723.1 codec and passthrough - but the
I must ta
Etienne, howzit
I am not 100% sure about this, but Net2phone do not always use
standard SIP as the protocol. They have their own proprietry
protocol as well, so perhaps your phone is trying to talk on the
proprietry protocol.
For G723.1 passthrough, you just allow it, and it should work fine,
Hello all,
I came a cross a problem yesterday that I don't quite know how to solve.
I am trying to use * to connect to net2phone, and have a net2phone MAX
IP-10 connect to net2phone. From the settings on
http://www.voip-info.org/ it was easy to get asterisk to connect to the
network - acting li
sterisk Users Mailing List - Non-Commercial Discussion"
<[EMAIL PROTECTED]>
Sent: Wednesday, December 15, 2004 1:02 PM
Subject: Re: [Asterisk-Users] Codecs and RealTime
> At 11:02 AM 12/15/04, you wrote:
> >Your sip_buddies table should have 2 columns, "allow" and
At 11:02 AM 12/15/04, you wrote:
Your sip_buddies table should have 2 columns, "allow" and "disallow". You
should be able to:
INSERT INTO sip_buddies (allow,disallow) VALUES ("g729;g726;gsm","g711");
to give the equiv of:
allow=g729
allow=g726
allow=gsm
disallow=g711
-Matthew
I have the sip in 2 ta
-Matthew
- Original Message -
From: "Damian Minkov" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[EMAIL PROTECTED]>
Sent: Wednesday, December 15, 2004 12:03 PM
Subject: Re: [Asterisk-Users] Codecs and RealTime
> The re
I have updated from latest CVS 2 days ago and I have run Realtime
SIPBuddies today i noticed
problem with codecs.
If there is nothing in the DB for allow and disallow
sip show peer ... :
Codecs : 0x10d (g723|ulaw|alaw|g729)
Codec Order : (g729|g723|ulaw|alaw)
But if I put in the
and "disallow". You
should be able to:
INSERT INTO sip_buddies (allow,disallow) VALUES ("g729;g726;gsm","g711");
to give the equiv of:
allow=g729
allow=g726
allow=gsm
disallow=g711
-Matthew
- Original Message -
From: "Damian Minkov" <[EMAIL PROTECTED]
-- Original Message -
From: "Damian Minkov" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, December 15, 2004 9:15 AM
Subject: [Asterisk-Users] Codecs and RealTime
> I have updated from latest CVS 2 days ago and I have run Realtime
> SIPBuddies today i noti
Hi all,
I am noticing echo/jitter problems when going sip -> asterisk
iax (ALAW)-> asterisk pstn depending on the codec I use. Both ULAW/ALAW
works fine on the budgetone and ata286 but g726 only works well on the
budgetone.
Ilbc just doesn't work well with broken speech and echo issues.
Hello,
I have a following setup:
IP phone (Cisco/Skinny) <-> * <-> NAT -- NAT <-> * <-> PSTN
Everything is perfect when i'm using it from right to left. From left to
right however, there is no voice, although the calls are being placed.
I played around with codeces but no change.
Does anybody k
Eric Jacksch wrote:
Are there any codecs that are particularly good for fax traffic? Any to avoid?
---
Eric Jacksch
[EMAIL PROTECTED]
See http://www.opencall.org/faq
Regards,
Steve
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.
On Mon, 2004-09-06 at 17:09, Eric Jacksch wrote:
> Are there any codecs that are particularly good for fax traffic? Any to avoid?
Google, google, google google.
http://www.google.com/search?hl=en&ie=UTF-8&q=fax+codec+site%3Alists.digium.com
please exert effort before sending a question to the li
Are there any codecs that are particularly good for fax traffic? Any to avoid?
---
Eric Jacksch
[EMAIL PROTECTED]
___
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[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options v
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Monday 19 July 2004 01:43 pm, [EMAIL PROTECTED] wrote:
> Hi,
> I'm planning to use a Asterisk with Digium E1 cards, I understand that
> using a codec such as G.729 can be very CPU demanding. What are the real
> advantages of using a codec such as G
: [Asterisk-Users] RE:RE: [Asterisk-Users] Codecs -
Advantages
Yes, I don't have any bandwidth issues, so I could use 2 Mbps for 30
calls. I do have issues with processing CPU capacity. Is g711 CPU
intensive as g729 ? I understand g729 is very CPU intensive.
>>>...
Forgive me, but what
especially for the end users of the system...
-Chris
- Original Message -
From: "brian" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, July 19, 2004 11:44 AM
Subject: RE: [Asterisk-Users] Codecs - Advantages
> If you have the bandwidth then use ulaw
Yes, I don't have any bandwidth issues, so I could use 2 Mbps for 30
calls. I do have issues with processing CPU capacity. Is g711 CPU
intensive as g729 ? I understand g729 is very CPU intensive.
>>>...
Forgive me, but what you just wrote tells you EXACTLY what you should
use!
__
/2004 18:29:27
Para: <[EMAIL PROTECTED]>
Título: RE: [Asterisk-Users] Codecs - Advantages
If you dont have bandwith issues, use g711, with 2 mb bandwith you can pass
30 calls, aprox.
G729 compress from g711 64 kbps to g729 8 kbps
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EM
If you have the bandwidth then use ulaw :)
bkw
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
> Sent: Monday, July 19, 2004 12:44 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Codecs -
.
Para: [EMAIL PROTECTED]
Asunto: [Asterisk-Users] Codecs - Advantages
Hi,
I'm planning to use a Asterisk with Digium E1 cards, I understand that
using a codec such as G.729 can be very CPU demanding. What are the real
advantages of using a codec such as G.729 ? Bandwidth only ? Usi
Hi,
I'm planning to use a Asterisk with Digium E1 cards, I understand that
using a codec such as G.729 can be very CPU demanding. What are the real
advantages of using a codec such as G.729 ? Bandwidth only ? Using no
compression wouldn't increase the scalability of my asterisk PBX ? This
is cons
Hi,
I'm planning to use a Asterisk with Digium E1 cards, I understand that using a codec
such as G.729 can be very CPU demanding. What are the real advantages of using a codec
such as G.729 ? Bandwidth only ? Using no compression wouldn't increase the
scalability of my asterisk PBX ? This is co
D]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Glover
Sent: 23 June 2004 12:34
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Codecs and pauses
On Wed, 23 Jun 2004, Matt wrote:
> >>I've been having similar problems to you. I found after reading an
> unrelated post, about the J
On Wed, 23 Jun 2004, Matt wrote:
> >>I've been having similar problems to you. I found after reading an
> unrelated post, about the Jitterbuffer option in
> >>iax.conf, setting this to yes has made things much better.
> Out of interest what have you set for your
> dropcount
> maxjitterbuffer
> Max
---Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Glover
Sent: 23 June 2004 12:17
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Codecs and pauses
Hi,
I've been having similar problems to you. I found after reading an unrelated
post, about the Jit
Hi,
I've been having similar problems to you. I found after reading an
unrelated post, about the Jitterbuffer option in iax.conf, setting this to
yes has made things much better.
Now the problem I have is that telappliant have lost one of there external
routers so I can't connect at the moment :-
Hi all
My * implementation is working brilliantly with only one small fault left to
kill.
I'm using IAXTalk from Telappliant for my incoming/outgoing calls to the
pstn network; if I set my codec to GSM everything works great - no pauses
but quality is a bit poor. If it set the codec to alaw (I th
greetings I'm running yellow dog 3.1 compiling Asterisk 0.7.1
during the make process it seems to die at the GSM build.
(summerized)
As build goes' through
must remake `src/add.o'.
entering dirctory `/usr/local/asterisk-0.7.1/codecs/gsm'
gcc -march= -fomit-frame-pointer -c -DneedFunctionprototyp
Hi!
Are the GIPS codecs now implemented with the Asterisk?
If I need more analog lines, say around 30, what's the
easiest way doing it? I checked the Mediatrix box with
24 connections, maybe that would be a good (and rel. cheap)
way to go? Any other suggestions? The ports has to
support fax ma
I know that this issue has been discussed a lot on this list in regard
to some of the recent CVS's. However, it has come up as an issue on an
older release (CVS Aug 05, 2003) as well. I thought that a heads up was
in keeping with the philosophy of the list. Here are the details:
Call from GS
On Friday 03 October 2003 08:06 am, listas iPfone wrote:
> I have that lines in sip.conf:
>
> disallow=all
> allow=gsm
> allow=ulaw
> allow=alaw
>
> when i use show codecs:
>
> localhost*CLI> show codecs
>1 (1 << 0) G.723.1
>2 (1 << 1) GSM
>4 (1 << 2) G.711 u-law
>
Hi!
I have some question about the use of codecs in sip.conf
I have that lines in sip.conf:
disallow=all
allow=gsm
allow=ulaw
allow=alaw
when i use show codecs:
localhost*CLI> show codecs
1 (1 << 0) G.723.1
2 (1 << 1) GSM
4 (1 << 2) G.711 u-law
8 (1 << 3) G.
John, Tx
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Hi,
What are real life bandwith stats for * supported codecs?
Is it true one can run 6-32 conversations over DSL, as stated in this list?
Senad
Google will also give you the results I just found.
http://lists.digium.com/pipermail/asterisk-users/2003-June/014788.html
JT
__
Thats all going to depend on the speed of your DSL...
bkw
On Wed, 17 Sep 2003, Senad Jordanovic wrote:
> Hi,
>
> What are real life bandwith stats for * supported codecs?
> Is it true one can run 6-32 conversations over DSL, as stated in this list?
>
>
> Senad
>
>
> _
Hi,
What are real life bandwith stats for * supported codecs?
Is it true one can run 6-32 conversations over DSL, as stated in this list?
Senad
___
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On Tuesday 29 July 2003 13:47, Tais M. Hansen wrote:
> I havn't used the h323 channel of Asterisk for a while, but today I needed
> to test a few things only I found out that Asterisk/H323 crashes my Siemens
> optipoint 400 phone. It seems to be the au
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
I havn't used the h323 channel of Asterisk for a while, but today I needed to
test a few things only I found out that Asterisk/H323 crashes my Siemens
optipoint 400 phone. It seems to be the audio codecs that's causing it. Is
something broken i
. Callis
Sent: Wednesday, July
23, 2003 2:56 AM
To:
[EMAIL PROTECTED]
Subject: [Asterisk-Users] Codecs
for use with Cisco 7960 and ATA-186
Are there any other codecs that can be used with the 7960
and the ATA-186? I have been using the default gsm codec and wanted to see if I
could make use of
should try to use a low bandwidth codec that is less than
64k.
K.
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Dan
> Sent: Wednesday, July 23, 2003 12:21 AM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] C
Hi,
For local connection to Asterisk (LAN), G.711 is the best option.
BR,
Dan
- Original Message -
From: "Kim C. Callis" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, July 23, 2003 10:13 AM
Subject: RE: [Asterisk-Users] Codecs for use with
:[EMAIL PROTECTED] On Behalf Of Kim C. Callis
Sent: Tuesday, July 22, 2003 11:56 PM
To:
[EMAIL PROTECTED]
Subject: [Asterisk-Users] Codecs
for use with Cisco 7960 and ATA-186
Are there any other codecs that can be used with the 7960
and the ATA-186? I have been using the default gsm codec and
Are there any other codecs that
can be used with the 7960 and the ATA-186? I have been using the default gsm codec and wanted to see if I could make use of
something a little less bandwidth intensive…
Kim Callis
You need G723 CODEC to be supportted on your asterisk server.
Best regards
Lubo
Dave Alan Caruana wrote:
My system is an asterisk machine,
with an E1 card (functioning) and
forwarding calls to a remote SIP
address ..
when a call connects I am getting the
following error :
NOTICE[1240577216]: File
My system is an asterisk machine,
with an E1 card (functioning) and
forwarding calls to a remote SIP
address ..
when a call connects I am getting the
following error :
NOTICE[1240577216]: File rtp.c, Line 330 (ast_rtp_read): Unknown RTP codec
19 received
can anybody tell me what this means
(& h
> Not trying to be difficult, but is that a "purchase" or a time-bound
> license?
>
> Just curious. I'm very interested in using it, but the $$ commitment is
> something of a question.
That's a purchase.
Mark
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Mark Spencer wrote:>>So it looks like the best codec is the GSM codec as
far and badwidth
vs voice quality, but I can't seem to find which hard phones support
the GSM codec or if * supports the G.729 codecs or others..
Which phones do the * user commumity find work the best?? and which
codecs do y
We've done 60 channels on a dual 1.8 Ghz Xeon. Trial channels are *not*
available because we have to purchase keys from Voiceage, and they are
unwilling to make any trial keys available.
Mark
On 27 Mar 2003, Jared Smith wrote:
> That's my question exactly... How many concurrent calls can I run
> Quick question what happens if you go over
> your channel licenses?
It cannot transcode.
Mark
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That's my question exactly... How many concurrent calls can I run over
G.729 before I have to go out and buy a bigger processor? Does anyone
have some data? I've heard rumors on IRC, but I'd rather have some
"real world" data...
(Maybe I'll have to try it myself! Mark, is it possible to get the
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