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
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
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 a
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
--- [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 [EMAIL
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.
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 files (gsm
On Mar 16, 2006, at 3:24 AM, Aisling wrote:
x-tad-smallerHi everyone,/x-tad-smallerx-tad-smaller /x-tad-smallerx-tad-smallerI 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
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 from g729 to any. So, let
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 asterisk
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?
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
pstn, we
: [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 on asterisk are on gsm format, we need
:[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 PROTECTED] wrote
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 I need gsm codec to hear
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.
___
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Asterisk-Users mailing list
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
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
to use
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
I've
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
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
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
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
Hi,
asterisk will negotiate codecs for both parties independently (use sip
show peer 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
Pavel Jezek wrote:
Hi,
asterisk will negotiate codecs for both parties independently (use sip
show peer 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
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
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
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
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
I assume ISDN accomplishes this since the PRI is set to use channel 24
for signaling. Your 64K channels is data and the control overhead is
sent on the signaling channel.
Actually, everything I have seen is around 80K full duplex for a uLaw
channel with overhead. That is point to point...
W
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
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
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
I've seen several references to 'simplex' ulaw using 64k + overhead and
'duplex' ulaw using 64k+overhead+64k+overhead (I wished that site was up).
Coming from this page:
http://voip-info.org/tiki-index.php?page=Bandwidth+consumption, the
following line strikes me:
(from near the top of the
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,
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
- 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 result after
INSERT INTO sip_buddies (allow,disallow
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
- Original Message -
From: Damian Minkov [EMAIL
The result after
INSERT INTO sip_buddies (allow,disallow) VALUES (g729;g726;gsm,ulaw);
is
Codecs : 0x11b (g723|gsm|alaw|g726|g729)
Codec Order : (g723|alaw|g729|g726|gsm)
But in sip.conf general i have
disallow=all
allow=g729
allow=g723.1
allow=ulaw
allow=alaw
But if do the following
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 tables,
- 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 disallow. You
should be able to:
INSERT INTO sip_buddies (allow
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=enie=UTF-8q=fax+codec+site%3Alists.digium.com
please exert effort before sending a question to the
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]
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 - Advantages
Hi,
I'm
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 :)
bkw
-Original Message-
From: [EMAIL PROTECTED
-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
]
[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 Jitterbuffer option in iax.conf, setting this to yes
] 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 Jitterbuffer option in
iax.conf, setting
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
8 (1 3) G.711
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
John, Tx
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
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
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
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
Actually, I found that both the 7960 and
ATA-186 support several codecs So the question should have been which is
the best codec to make use of? According to the literature, they support G729, G723.1 and G.711
u-law/a-law.
-Original Message-
From: [EMAIL PROTECTED]
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 Cisco 7960 and ATA-186
Actually, I
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] Codecs for use with Cisco
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
On Thu, 2003-03-27 at 06:22, WipeOut . wrote:
Hi,
From what I have been able to work out * supports G.711 a+u, GSM
and LPC-10 for VoIP calls.
So far it seems that the Hardphones out there support G.711, G.729
and some times a few other codecs..
So the common denominator seems to be
:[EMAIL PROTECTED]
Sent: Thursday, March 27, 2003 7:21 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] * + Codecs + Hardphones??
On Thu, 2003-03-27 at 06:22, WipeOut . wrote:
Hi,
From what I have been able to work out * supports G.711 a+u, GSM
and LPC-10 for VoIP calls.
So far
Quick question what happens if you go over
your channel licenses?
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
The same as you go over the number of PRI channels ?
regards
Martin
On Thu, 27 Mar 2003, James O. Sizemore III wrote:
Quick question what happens if you go over
your channel licenses?
Mark Spencer wrote:
So it looks like the best codec is the GSM codec as far and badwidth
vs voice
: Re: [Asterisk-Users] * + Codecs + Hardphones??
Quick question what happens if you go over
your channel licenses?
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
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
Quick question what happens if you go over
your channel licenses?
It cannot transcode.
Mark
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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
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
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