Re: [asterisk-users] CODECS: Best practice question: Avoid transcode when calling out?

2010-02-16 Thread Philipp von Klitzing
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

[asterisk-users] CODECS: Best practice question: Avoid transcode when calling out?

2010-02-16 Thread Karl Fife
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

[asterisk-users] codecs and volume

2009-12-29 Thread Ron
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

[asterisk-users] Codecs negotiation

2009-12-01 Thread Ignacio
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

[asterisk-users] Codecs with MixMonitor (,a) option

2009-10-27 Thread Miguel Molina
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 ___

Re: [asterisk-users] codecs/voicemail/DTMF

2006-09-20 Thread Eric \"ManxPower\" Wieling
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

Re: [asterisk-users] codecs/voicemail/DTMF

2006-09-20 Thread Mr. Jones
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

Re: [asterisk-users] codecs/voicemail/DTMF

2006-09-19 Thread Eric \"ManxPower\" Wieling
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

[asterisk-users] codecs/voicemail/DTMF

2006-09-19 Thread Mr. Jones
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

RE : Re: [asterisk-users] codecs translation in Asterisk SVN-trunk-r41990

2006-09-08 Thread harrygaillac-sip
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

Re: [asterisk-users] codecs translation in Asterisk SVN-trunk-r41990

2006-09-08 Thread Tzafrir Cohen
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.

RE : [asterisk-users] codecs translation in Asterisk SVN-trunk-r41990 [SOLVED]

2006-09-08 Thread harrygaillac-sip
--- [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

[asterisk-users] codecs translation in Asterisk SVN-trunk-r41990

2006-09-08 Thread harrygaillac-sip
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|

Re: [Asterisk-Users] Codecs? - Asked to transmit frame type 256, while native formats is 8 (read/write = 8/8)

2006-03-16 Thread Martin Joseph
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

[Asterisk-Users] Codecs? - Asked to transmit frame type 256, while native formats is 8 (read/write = 8/8)

2006-03-16 Thread Aisling
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

[Asterisk-Users] codecs choice

2006-02-06 Thread FaberK
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:

RE : [Asterisk-Users] codecs order and so on

2006-01-11 Thread Olivier Taylor
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

Re: [Asterisk-Users] codecs order and so on

2006-01-10 Thread Moises Silva
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

[Asterisk-Users] codecs order and so on

2006-01-10 Thread Olivier Taylor
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

Re: [Asterisk-Users] Codecs.

2005-12-16 Thread Rich Adamson
> 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

[Asterisk-Users] Codecs.

2005-12-16 Thread Pablo Allietti
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? -- .-

RE : [Asterisk-Users] Codecs problem

2005-11-09 Thread Olivier Taylor
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

Re: RE : RE : [Asterisk-Users] codecs

2005-11-09 Thread asterisk183
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

Re: [Asterisk-Users] Codecs problem

2005-11-09 Thread William Lloyd
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

[Asterisk-Users] Codecs problem

2005-11-09 Thread Olivier Taylor
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

Re: RE : RE : [Asterisk-Users] codecs

2005-11-09 Thread Eric \"ManxPower\" Wieling
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

RE : RE : [Asterisk-Users] codecs

2005-11-09 Thread Olivier Taylor
: 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

Re: RE : [Asterisk-Users] codecs

2005-11-09 Thread Sahil Gupta
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

2005-11-09 Thread Olivier Taylor
: 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

Re: [Asterisk-Users] codecs

2005-11-09 Thread Angelito Manansala
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 >

[Asterisk-Users] codecs

2005-11-09 Thread Olivier Taylor
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?

Re: [Asterisk-Users] codecs order

2005-08-16 Thread Brian West
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

Re: [Asterisk-Users] codecs order

2005-08-16 Thread Pavel Jezek
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

Re: [Asterisk-Users] codecs order

2005-08-16 Thread Brian West
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

Re: [Asterisk-Users] codecs order

2005-08-15 Thread Erik Versaevel
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

Re: [Asterisk-Users] codecs order

2005-08-15 Thread Brian West
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

Re: [Asterisk-Users] codecs order

2005-08-15 Thread Tony Hoyle
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

Re: [Asterisk-Users] codecs order

2005-08-15 Thread Pavel Jezek
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

[Asterisk-Users] codecs order

2005-08-15 Thread marek cervenka
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 -

Re: [Asterisk-Users] Codecs and bandwidth

2005-07-18 Thread Tim Pushor
] 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

Re: [Asterisk-Users] Codecs and bandwidth

2005-07-18 Thread Dan Perik
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

Re: [Asterisk-Users] Codecs and bandwidth

2005-07-18 Thread Tim Pushor
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

Re: [Asterisk-Users] Codecs and bandwidth

2005-07-18 Thread Brian Capouch
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

RE: [Asterisk-Users] Codecs and bandwidth

2005-07-18 Thread Wiley Siler
-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

Re: [Asterisk-Users] Codecs and bandwidth

2005-07-18 Thread Waldo Rubinstein
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

Re: [Asterisk-Users] Codecs and bandwidth

2005-07-18 Thread Tim Pushor
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

Re: [Asterisk-Users] Codecs and bandwidth

2005-07-18 Thread Rich Adamson
> 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 >

[Asterisk-Users] Codecs and bandwidth

2005-07-18 Thread Tim Pushor
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

[Asterisk-Users] codecs, asterisk, xpro

2005-05-02 Thread Dov Bigio
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

Re: [Asterisk-Users] Codecs and * pass through...

2005-04-13 Thread Etienne Pretorius
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

Re: [Asterisk-Users] Codecs and * pass through...

2005-04-13 Thread clive
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,

[Asterisk-Users] Codecs and * pass through...

2005-04-12 Thread Etienne Pretorius
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

Re: [Asterisk-Users] Codecs and RealTime

2004-12-15 Thread Matthew Boehm
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

Re: [Asterisk-Users] Codecs and RealTime

2004-12-15 Thread Greg - Cirelle Enterprises
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

Re: [Asterisk-Users] Codecs and RealTime

2004-12-15 Thread Matthew Boehm
-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

[Asterisk-Users] Codecs and RealTime

2004-12-15 Thread Damian Minkov
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

Re: [Asterisk-Users] Codecs and RealTime

2004-12-15 Thread Damian Minkov
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]

Re: [Asterisk-Users] Codecs and RealTime

2004-12-15 Thread Matthew Boehm
-- 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

[Asterisk-Users] Codecs and echo

2004-11-02 Thread Dee Lowndes
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.

[Asterisk-Users] Codecs Problem?

2004-09-25 Thread Christoph Kampka
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

Re: [Asterisk-Users] Codecs for fax traffic

2004-09-06 Thread Steve Underwood
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.

Re: [Asterisk-Users] Codecs for fax traffic

2004-09-06 Thread Steven Critchfield
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

[Asterisk-Users] Codecs for fax traffic

2004-09-06 Thread Eric Jacksch
Are there any codecs that are particularly good for fax traffic? Any to avoid? --- Eric Jacksch [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options v

Re: [Asterisk-Users] Codecs - Advantages

2004-07-19 Thread Steve
-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

RE: [Asterisk-Users] RE:RE: [Asterisk-Users] Codecs - Advantages

2004-07-19 Thread Wiley E. Siler
: [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

Re: [Asterisk-Users] Codecs - Advantages

2004-07-19 Thread Chris Shaw
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

RE: [Asterisk-Users] RE:RE: [Asterisk-Users] Codecs - Advantages

2004-07-19 Thread Senad Jordanovic
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! __

[Asterisk-Users] RE:RE: [Asterisk-Users] Codecs - Advantages

2004-07-19 Thread matiaspinedo
/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

RE: [Asterisk-Users] Codecs - Advantages

2004-07-19 Thread brian
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 -

RE: [Asterisk-Users] Codecs - Advantages

2004-07-19 Thread Sebastian Nocetti
. 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

RE: [Asterisk-Users] Codecs - Advantages

2004-07-19 Thread Senad Jordanovic
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

[Asterisk-Users] Codecs - Advantages

2004-07-19 Thread matiaspinedo
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

RE: [Asterisk-Users] Codecs and pauses

2004-06-23 Thread Matt
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

RE: [Asterisk-Users] Codecs and pauses

2004-06-23 Thread Chris Glover
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

RE: [Asterisk-Users] Codecs and pauses

2004-06-23 Thread Matt
---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

Re: [Asterisk-Users] Codecs and pauses

2004-06-23 Thread Chris Glover
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 :-

[Asterisk-Users] Codecs and pauses

2004-06-23 Thread Matt
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

[Asterisk-Users] Codecs compile error on yellowdog

2004-02-13 Thread Jeff Donovan
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

[Asterisk-Users] Codecs and more analog lines?

2004-01-22 Thread Kerker Staffan
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

[Asterisk-Users] Codecs and call failure with Grandstream

2003-11-11 Thread Stephen R. Besch
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

Re: [Asterisk-Users] codecs questions

2003-10-03 Thread Tilghman Lesher
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 >

[Asterisk-Users] codecs questions

2003-10-03 Thread listas iPfone
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.

RE: [Asterisk-Users] CODECS and thier practical usage stats

2003-09-18 Thread Senad Jordanovic
John, Tx ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] CODECS and thier practical usage stats

2003-09-18 Thread John Todd
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 __

Re: [Asterisk-Users] CODECS and thier practical usage stats

2003-09-17 Thread Brian West
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 > > > _

[Asterisk-Users] CODECS and thier practical usage stats

2003-09-17 Thread Senad Jordanovic
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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Codecs

2003-08-01 Thread Tais M. Hansen
-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 au

[Asterisk-Users] Codecs

2003-07-29 Thread Tais M. Hansen
-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

RE: [Asterisk-Users] Codecs for use with Cisco 7960 and ATA-186

2003-07-23 Thread Wade Weppler
. 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

RE: [Asterisk-Users] Codecs for use with Cisco 7960 and ATA-186

2003-07-23 Thread Kim C. Callis
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

Re: [Asterisk-Users] Codecs for use with Cisco 7960 and ATA-186

2003-07-23 Thread Dan
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

RE: [Asterisk-Users] Codecs for use with Cisco 7960 and ATA-186

2003-07-23 Thread Kim C. Callis
:[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

[Asterisk-Users] Codecs for use with Cisco 7960 and ATA-186

2003-07-22 Thread Kim C. Callis
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  

Re: [Asterisk-Users] codecs question ..

2003-06-23 Thread Lubomir Christov
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

[Asterisk-Users] codecs question ..

2003-06-23 Thread Dave Alan Caruana
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

Re: [Asterisk-Users] * + Codecs + Hardphones??

2003-03-27 Thread Mark Spencer
> 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 ___ Asterisk-Users mailing list [EMAIL PRO

Re: [Asterisk-Users] * + Codecs + Hardphones??

2003-03-27 Thread Brian Capouch
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

RE: [Asterisk-Users] * + Codecs + Hardphones??

2003-03-27 Thread Mark Spencer
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

Re: [Asterisk-Users] * + Codecs + Hardphones??

2003-03-27 Thread Mark Spencer
> Quick question what happens if you go over > your channel licenses? It cannot transcode. Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] * + Codecs + Hardphones??

2003-03-27 Thread Jared Smith
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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