[asterisk-users] CODECS: Best practice question: Avoid transcode when calling out?
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 preferred accordingly to match the remote. Outbound calls seem harder. Our endpoints always negotiate g.722 between themselves and Asterisk and then Asterisk must transcode to the preferred codec of the REMOTE party. Not ideal. It's easy if I'm calling to the PSTN--since I know in advance the preferences of my ITSP, but I'm stumped about how to avoid transcoding when calling a party whose codec preferences are not known in advance. Is there an elegant way to do this? -Karl -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CODECS: Best practice question: Avoid transcode when calling out?
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 (the remote party) to most-known (my endpoint) and my
codecs can simply be preferred accordingly to match the remote.
Look at setting the channel variable_SIP_CODEC - however it might or
might not work depending on your version of Asterisk, see for example:
https://issues.asterisk.org/view.php?id=13243
Here's a dialplan snippet that might give you another hint or two.
exten = 123,1,NoOp(-- Inbound read: ${CHANNEL(audioreadformat)} --)
exten = 123,n,NoOp(-- Inbound native: ${CHANNEL(audionativeformat)} --)
exten = 123,n,Set(WIDEBAND=0)
exten = 123,n,Set(WIDEBAND=${REGEX(g722
${SIPPEER(${SIPCHANINFO(peername)}:codecs)})})
exten = 123,n,ExecIf($[${WIDEBAND} = 1]|Set|_SIP_CODEC=g722)
exten = 123,n,Dial(SIP/abc123)
Please note the SPACE between ${REGEX(g722 and ${SIPPEER
Outbound calls seem harder. Our endpoints always negotiate g.722 between
themselves and Asterisk and then Asterisk must transcode to the preferred
codec of the REMOTE party. Not ideal.
Together with the g722 transcoding patch for Asterisk 1.4.17 it does not
work out, unfortunately. Currently I cannot make a statement on a more
recent 1.4 release.
g722 patch: http://users.netplex.net/~andrew/asterisk/#g722
Older patch that I use for 1.4.17:
http://users.netplex.net/~andrew/asterisk/g722-20080110.patch.gz
However I can successfully employ setting _SIP_CODEC if in the example
above instead of Dial() I do a MusicOnHold() - both with or without a
preceeding ANSWER; without means early audio playing of the native g722
encoded MOH file. My snom starts out with alaw, and then we switch to
g722.
Is there an elegant way to do this?
Consider the codec negotiation patch? I'd be interested to hear about
your results!
http://www.b2bua.org/wiki/AsteriskCodecNegotiationPatch
https://issues.asterisk.org/view.php?id=4825
Philipp
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[asterisk-users] codecs and volume
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 http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Codecs negotiation
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 old telephones. I have found a patch, but it seems to be for old versions of asterisk. http://www.rtpproxy.org/wiki/AsteriskCodecNegotiationPatch It is also suitable for me that asterisk took a look at sip.conf and decided which codec to use looking at supported codecs in every client (if it is easier). Do you have any idea? Thank you very much. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Codecs with MixMonitor (,a) option
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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] codecs/voicemail/DTMF
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 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 G729 for outbound termination, so far so good - it appears that dtmfmode=auto works in both cases. The area I'm having trouble with is, in order to have g729 on the outbound I have: disallow=all allow=g729 allow=ulaw allow=alaw In sip.conf at the [general] level. When we call voicemail, or the auto attendant internally touchtones don't work and we get: WARNING[8393]: dsp.c:1422 ast_dsp_process: Inband DTMF is not supported on codec g729. Use RFC2833 I'm just guessing, but I thought auto was supposed to negotiate the DTMF mode. Since it appears that the voicemail can't handle RFC2833, is there some way to force the codec to resort to G711? Thanks! Brian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] codecs/voicemail/DTMF
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 ton! Brian On 9/19/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: 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 G729 for outbound termination, so far so good - it appears that dtmfmode=auto works in both cases. The area I'm having trouble with is, in order to have g729 on the outbound I have: disallow=all allow=g729 allow=ulaw allow=alaw In sip.conf at the [general] level. When we call voicemail, or the auto attendant internally touchtones don't work and we get: WARNING[8393]: dsp.c:1422 ast_dsp_process: Inband DTMF is not supported on codec g729. Use RFC2833 I'm just guessing, but I thought auto was supposed to negotiate the DTMF mode. Since it appears that the voicemail can't handle RFC2833, is there some way to force the codec to resort to G711? Thanks! Brian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] codecs/voicemail/DTMF
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 trouble with is, in order to have g729 on the outbound I have: disallow=all allow=g729 allow=ulaw allow=alaw In sip.conf at the [general] level. When we call voicemail, or the auto attendant internally touchtones don't work and we get: WARNING[8393]: dsp.c:1422 ast_dsp_process: Inband DTMF is not supported on codec g729. Use RFC2833 I'm just guessing, but I thought auto was supposed to negotiate the DTMF mode. Since it appears that the voicemail can't handle RFC2833, is there some way to force the codec to resort to G711? Thanks! Brian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] codecs/voicemail/DTMF
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 G729 for outbound termination, so far so good - it appears that dtmfmode=auto works in both cases. The area I'm having trouble with is, in order to have g729 on the outbound I have: disallow=all allow=g729 allow=ulaw allow=alaw In sip.conf at the [general] level. When we call voicemail, or the auto attendant internally touchtones don't work and we get: WARNING[8393]: dsp.c:1422 ast_dsp_process: Inband DTMF is not supported on codec g729. Use RFC2833 I'm just guessing, but I thought auto was supposed to negotiate the DTMF mode. Since it appears that the voicemail can't handle RFC2833, is there some way to force the codec to resort to G711? Thanks! Brian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] codecs translation in Asterisk SVN-trunk-r41990
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|tTj) in new stack -- Called 84 -- Got SIP response 486 Busy Here back from 80.119.15.85 -- SIP/84-0821ddd0 is busy == Everyone is busy/congested at this time (1:1/0/0) -- Executing [EMAIL PROTECTED]:3] VoiceMail(SIP/86-08218198, u84) in new stack -- Playing 'vm-theperson' (language 'fr') [2006-09-08 10:38:43] WARNING[2382]: channel.c:2609 set_format: Unable to find a codec translation path from g729 to gsm [2006-09-08 10:38:43] WARNING[2382]: file.c:805 ast_streamfile: Unable to open digits/8 (format 0x100 (g729)): No such file or directory == Spawn extension (sip, 84, 3) exited non-zero on 'SIP/86-08218198' serveur1*CLI show v versionvideo voicemail serveur1*CLI show version Asterisk SVN-trunk-r41990 built by root @ serveur1.home.net on a i686 running Linux on 2006-09-04 17:07:12 UTC serveur1*CLI ___ Découvrez un nouveau moyen de poser toutes vos questions quelque soit le sujet ! Yahoo! Questions/Réponses pour partager vos connaissances, vos opinions et vos expériences. http://fr.answers.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE : [asterisk-users] codecs translation in Asterisk SVN-trunk-r41990 [SOLVED]
--- [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 PROTECTED]:2] Dial(SIP/86-08218198, Sip/84|30|tTj) in new stack -- Called 84 -- Got SIP response 486 Busy Here back from 80.119.15.85 -- SIP/84-0821ddd0 is busy == Everyone is busy/congested at this time (1:1/0/0) -- Executing [EMAIL PROTECTED]:3] VoiceMail(SIP/86-08218198, u84) in new stack -- Playing 'vm-theperson' (language 'fr') [2006-09-08 10:38:43] WARNING[2382]: channel.c:2609 set_format: Unable to find a codec translation path from g729 to gsm [2006-09-08 10:38:43] WARNING[2382]: file.c:805 ast_streamfile: Unable to open digits/8 (format 0x100 (g729)): No such file or directory == Spawn extension (sip, 84, 3) exited non-zero on 'SIP/86-08218198' serveur1*CLI show v versionvideo voicemail serveur1*CLI show version Asterisk SVN-trunk-r41990 built by root @ serveur1.home.net on a i686 running Linux on 2006-09-04 17:07:12 UTC serveur1*CLI ___ Découvrez un nouveau moyen de poser toutes vos questions quelque soit le sujet ! Yahoo! Questions/Réponses pour partager vos connaissances, vos opinions et vos expériences. http://fr.answers.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Découvrez un nouveau moyen de poser toutes vos questions quelque soit le sujet ! Yahoo! Questions/Réponses pour partager vos connaissances, vos opinions et vos expériences. http://fr.answers.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] codecs translation in Asterisk SVN-trunk-r41990
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. Alternatively, encode files as g729 off-line (e.g: using the convert application, or using ready-made prompts). -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE : Re: [asterisk-users] codecs translation in Asterisk SVN-trunk-r41990
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 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. Alternatively, encode files as g729 off-line (e.g: using the convert application, or using ready-made prompts). -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Découvrez un nouveau moyen de poser toutes vos questions quelque soit le sujet ! Yahoo! Questions/Réponses pour partager vos connaissances, vos opinions et vos expériences. http://fr.answers.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Codecs? - Asked to transmit frame type 256, while native formats is 8 (read/write = 8/8)
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 calls dont work. Strange. Anyhow I was getting an error: Process_sdp: No compatible codecs! And from the SIP debug I could see that the incoming SIP INVITE was getting a sip response of 488 Unacceptable here from my asterisk server. After doing a bit of searching I determined that this might be the fault of the codecs particularly the G729 codec. So in the peer block that I have for my PSTN provider in my sip conf I specified allow=g729. I called my PSTN geographic number again and was delighted when the incoming calls worked. However when I next went to make an outgoing call (after having added in the allow=g729 line), I got an infinite loop of warnings: WARNING: chan_sip.c: 2520 sip_write: Asked to transmit frame type 256, while native formats is 8 (read/write = 8/8) WARNING: codec_gsm.c165 gsmtolin_framein: Huh? A GSM frame that isnt a multiple of 33 or 65 bytes long from RTP After those warnings I thought there might be a problem with the gsm codec so I commented the lines containing allow=gsm and still kept the line allow=g729 because as Ive said already incoming calls wont work otherwise (but outgoing will). This however just gave another warning: WARNING: chan_sip.c: 2520 sip_write: Asked to transmit frame type 4 while native formats is 256 (read/write=64/64). When I comment this line out again I am back to my original situation where outgoing calls work and incoming dont. Has anyone any idea how I can work around this? Many thanks in advance, Aisling. ---Legal Disclaimer--- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codecs? - Asked to transmit frame type 256, while native formats is 8 (read/write = 8/8)
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 that I can solve an error which gets my incoming calls working but that in turns means my outgoing calls don’t work. – Strange./x-tad-smallerx-tad-smaller /x-tad-smallerx-tad-smallerAnyhow I was getting an error:/x-tad-smallerx-tad-smaller /x-tad-smallerx-tad-smallerProcess_sdp: No compatible codecs!/x-tad-smallerx-tad-smallerAnd from the SIP debug I could see that the incoming SIP INVITE was getting a sip response of 488 Unacceptable here from my asterisk server./x-tad-smallerx-tad-smaller /x-tad-smallerx-tad-smallerAfter doing a bit of searching I determined that this might be the fault of the codec’s particularly the G729 codec. So in the peer block that I have for my PSTN provider in my sip conf I specified allow=g729./x-tad-smallerx-tad-smallerI called my PSTN geographic number again and was delighted when the incoming calls worked. However when I next went to make an outgoing call (after having added in the “allow=g729” line), I got an infinite loop of warnings:/x-tad-smallerx-tad-smaller /x-tad-smallerx-tad-smallerWARNING: chan_sip.c: 2520 sip_write: Asked to transmit frame type 256, while native formats is 8 (read/write = 8/8)/x-tad-smallerx-tad-smallerWARNING: codec_gsm.c165 gsmtolin_framein: Huh? A GSM frame that isn’t a multiple of 33 or 65 bytes long from RTP/x-tad-smallerx-tad-smaller /x-tad-smallerx-tad-smallerAfter those warnings I thought there might be a problem with the gsm codec so I commented the lines containing “allow=gsm” and still kept the line “allow=g729” because as I’ve said already incoming calls won’t work otherwise (but outgoing will)./x-tad-smallerx-tad-smallerThis however just gave another warning:/x-tad-smallerx-tad-smaller /x-tad-smallerx-tad-smallerWARNING: chan_sip.c: 2520 sip_write: Asked to transmit frame type 4 while native formats is 256 (read/write=64/64)./x-tad-smallerx-tad-smallerWhen I comment this line out again I am back to my original situation where outgoing calls work and incoming don’t./x-tad-smallerx-tad-smaller /x-tad-smallerx-tad-smallerHas anyone any idea how I can work around this?/x-tad-smallerx-tad-smaller /x-tad-smallerI think telling us which type of gateway is between asterisk and the PSTN might be helpful in this case... Marty ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] codecs choice
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:. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE : [Asterisk-Users] codecs order and so on
Just have a lok at this config : [general] Disallow=all Allow=g729 Allow=ulaw [pstn] Disallow=all Allow=g729 [zap] Disallow=all Allow=ulaw In extensions.conf, I change the context for each call, Asterisk doesn't care of codecs in contexts, it uses the order of general... Could be good to have Ssterisk making a match between codecs in General and the context used to make a call But definitiely, Asterisk choose g729 even if I am in the zap context Any idea, help is welcome. Olivier -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part 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 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 agree to use g729, so no codec translation is done. all the calls to and from fxo fails because no translation can be made from ULAW to g729, and from g729 (phones) to ulaw. then asterisk is not smart enough to realize that can ask the phones to use ulaw (i assume the phones support ulaw) to not use translation to call the fxo??? When you have first allow=ulaw (prefered codec) all the calls to and from fxo works because the prefered codec is ulaw, then from fxo to phones using ulaw, no codec translation is made all the calls to pstn providers fails, again, because it seems asterisk gives preference to ulaw codec (the first list codec) so, the phones use ulaw, and is not possible to translate ulaw to g729 and viceversa?? im interested in knowing the reason too, any guidelines? regards On 1/10/06, Olivier Taylor [EMAIL PROTECTED] wrote: 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 codecs All Uas have g729 (g729 is then pass-thru when needed) All Uas have ulaw(of course) If I have in [general] disallow=all allow=g729 allow=ulaw In this case: all calls to pstn providers works all calls to and from fxo fails because of : No translator path exists for If I have in [general] disallow= all allow= ulaw allow= g729 In this case: all calls to and from fxo works all calls to pstn providers fails because of : No translator path exists for ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] codecs order and so on
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 codecs All Uas have g729 (g729 is then pass-thru when needed) All Uas have ulaw(of course) If I have in [general] disallow=all allow=g729 allow=ulaw In this case: all calls to pstn providers works all calls to and from fxo fails because of : No translator path exists for If I have in [general] disallow=all allow=ulaw allow=g729 In this case: all calls to and from fxo works all calls to pstn providers fails because of : No translator path exists for ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 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 agree to use g729, so no codec translation is done. all the calls to and from fxo fails because no translation can be made from ULAW to g729, and from g729 (phones) to ulaw. then asterisk is not smart enough to realize that can ask the phones to use ulaw (i assume the phones support ulaw) to not use translation to call the fxo??? When you have first allow=ulaw (prefered codec) all the calls to and from fxo works because the prefered codec is ulaw, then from fxo to phones using ulaw, no codec translation is made all the calls to pstn providers fails, again, because it seems asterisk gives preference to ulaw codec (the first list codec) so, the phones use ulaw, and is not possible to translate ulaw to g729 and viceversa?? im interested in knowing the reason too, any guidelines? regards On 1/10/06, Olivier Taylor [EMAIL PROTECTED] wrote: 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 codecs All Uas have g729 (g729 is then pass-thru when needed) All Uas have ulaw(of course) If I have in [general] disallow=all allow=g729 allow=ulaw In this case: all calls to pstn providers works all calls to and from fxo fails because of : No translator path exists for If I have in [general] disallow= all allow= ulaw allow= g729 In this case: all calls to and from fxo works all calls to pstn providers fails because of : No translator path exists for ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Codecs.
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? -- .- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codecs.
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? Yes, its possible to solve the problem. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] codecs
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? Any idea is welcome. Olivier ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 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? Any idea is welcome. Olivier ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards, Angelito Manansala www.voicefidelity.net Mobile: +639175425807 DID: (+63) 44 7906770 msn: [EMAIL PROTECTED] skype: bulcrack ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE : [Asterisk-Users] codecs
Right, I must suppose I need gsm codec to hear gsm files, I miss? olivier -Message d'origine- De : [EMAIL PROTECTED] [mailto:[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: 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? Any idea is welcome. Olivier ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards, Angelito Manansala www.voicefidelity.net Mobile: +639175425807 DID: (+63) 44 7906770 msn: [EMAIL PROTECTED] skype: bulcrack ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 gsm files, I miss? olivier -Message d'origine- De : [EMAIL PROTECTED] [mailto:[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: 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? Any idea is welcome. Olivier ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards, Angelito Manansala www.voicefidelity.net Mobile: +639175425807 DID: (+63) 44 7906770 msn: [EMAIL PROTECTED] skype: bulcrack ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE : RE : [Asterisk-Users] codecs
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? Olivier -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Sahil Gupta Envoyé : mercredi 9 novembre 2005 12:33 À : 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 I need gsm codec to hear gsm files, I miss? olivier -Message d'origine- De : [EMAIL PROTECTED] [mailto:[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: 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? Any idea is welcome. Olivier ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards, Angelito Manansala www.voicefidelity.net Mobile: +639175425807 DID: (+63) 44 7906770 msn: [EMAIL PROTECTED] skype: bulcrack ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE : RE : [Asterisk-Users] codecs
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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Codecs problem
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 Found description format G729 Found description format telephone-event Capabilities: us - 0x70f (g723|gsm|ulaw|alaw|g729|speex|ilbc), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Nov 9 16:46:13 NOTICE[402]: channel.c:1763 ast_set_read_format: Unable to find a path from g729 to gsm Nov 9 16:46:13 NOTICE[402]: channel.c:1730 ast_set_write_format: Unable to find a path from ilbc to g729 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codecs problem
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 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 Found description format G729 Found description format telephone-event Capabilities: us - 0x70f (g723|gsm|ulaw|alaw|g729|speex|ilbc), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Nov 9 16:46:13 NOTICE[402]: channel.c:1763 ast_set_read_format: Unable to find a path from g729 to gsm Nov 9 16:46:13 NOTICE[402]: channel.c:1730 ast_set_write_format: Unable to find a path from ilbc to g729 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE : RE : [Asterisk-Users] codecs
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 gsm for voicemail and g729 for outbound calls? No. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Yahoo! Mail: gratis 1GB per i messaggi e allegati da 10MB http://mail.yahoo.it ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE : [Asterisk-Users] Codecs problem
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 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 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 Found description format G729 Found description format telephone-event Capabilities: us - 0x70f (g723|gsm|ulaw|alaw|g729|speex|ilbc), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Nov 9 16:46:13 NOTICE[402]: channel.c:1763 ast_set_read_format: Unable to find a path from g729 to gsm Nov 9 16:46:13 NOTICE[402]: channel.c:1730 ast_set_write_format: Unable to find a path from ilbc to g729 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] codecs order
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 preffered choise of A (G729) on both sides so there won't be any transcoding. This would allow for some nice things as fax passtrough (A and B has to use G711 then, but if the weigted A says G711, B would use G711 to). Kind regards, Erik Brian West wrote: 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 Aug 15, 2005, at 1:10 PM, Tony Hoyle wrote: 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 match, asterisk will try to transcode between) imho ;-) It does seem to be a weakness of asterisk.. it's creating load on the server when it doesn't need to. Really it should look at the capabilities of both ends and see if there's a common set, and only start transcoding if there's no overlap. Tony ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- InfoPact Netwerkdiensten B.V. http://www.infopact.nl/ Emmastraat 11-13 3255 BD Oude Tonge tel. +31 (0)187 64 77 11 mob. +31 (0)645 18 69 67 fax. +31 (0)187 64 77 99 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] codecs order
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/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] codecs order
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 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 preffered choise of A (G729) on both sides so there won't be any transcoding. This would allow for some nice things as fax passtrough (A and B has to use G711 then, but if the weigted A says G711, B would use G711 to). Kind regards, Erik ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] codecs order
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 negotiation in asterisk, but seems, however, this isn't as simple to implement as it looks... :-( PJ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] codecs order
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 --- Marek Cervenka === ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] codecs order
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 try to transcode between) imho ;-) PJ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] codecs order
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 match, asterisk will try to transcode between) imho ;-) It does seem to be a weakness of asterisk.. it's creating load on the server when it doesn't need to. Really it should look at the capabilities of both ends and see if there's a common set, and only start transcoding if there's no overlap. Tony ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] codecs order
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 Aug 15, 2005, at 1:10 PM, Tony Hoyle wrote: 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 match, asterisk will try to transcode between) imho ;-) It does seem to be a weakness of asterisk.. it's creating load on the server when it doesn't need to. Really it should look at the capabilities of both ends and see if there's a common set, and only start transcoding if there's no overlap. Tony ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 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 conversation, plus packet overhead (http://www.zytrax.com/tech/protocols/voip_rates.htm - seems to be down now - plus other references) for a total of over 128K per ulaw 'full duplex' voice conversation? Thanks Tim ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codecs and bandwidth
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 simultaneously. 80*4 = 320. You'd be using 320kbps down and 320kbps up, which is within your 1500kbps down / 384 kbps up. Someone please correct me if I'm wrong. That analysis is right, however seldom will the math work out exactly like real world usage. In all likelihood, other dsl traffic will consume a part of that bandwidth and two or maybe three calls might be usable. Implementing Quality of Service (QoS) on the outbound data flow might increase the probability of having more usable simultanous calls though. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codecs and bandwidth
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 Perik wrote: 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 simultaneously. 80*4 = 320. You'd be using 320kbps down and 320kbps up, which is within your 1500kbps down / 384 kbps up. Someone please correct me if I'm wrong. - Dan Tim Pushor wrote: 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 consideration. Thanks for helping clear that up. Tim ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codecs and bandwidth
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 your upload and download capacity have at least ~75Kbps available. In the case of most standard DSL services (and even cable broadband) sold in the US, the service sold is Asymmetrical. For these scenarios, Dan's explanation should suffice. For most dedicated Internet services (such as T1, fractional T1s, Ethernet, etc), the capacity is Symmetrical. In the case of a full T1 of Internet service, you can assume you could transport ~20 simultaneous calls (~75Kbps x 20 = ~1500Kbps), which, unfortunately, turns out to give you less capacity than a channelized T1 or even a PRI using standard trunk services. I guess, at this point you need to evaluate codec use and a couple of other factors in order to maximize your capacity. - Waldo On Jul 18, 2005, at 3:58 PM, Dan Perik wrote: 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 simultaneously. 80*4 = 320. You'd be using 320kbps down and 320kbps up, which is within your 1500kbps down / 384 kbps up. Someone please correct me if I'm wrong. - Dan Tim Pushor wrote: 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 consideration. Thanks for helping clear that up. Tim ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Codecs and bandwidth
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 -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 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 conversation, plus packet overhead (http://www.zytrax.com/tech/protocols/voip_rates.htm - seems to be down now - plus other references) for a total of over 128K per ulaw 'full duplex' voice conversation? Thanks Tim ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codecs and bandwidth
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 the conversation, plus packet overhead ISDN, like T1, is full duplex. So your 64K channel is 64K in each direction, up and down. B. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codecs and bandwidth
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 consideration. Thanks for helping clear that up. Tim Steve Kennedy wrote: On Mon, Jul 18, 2005 at 11:28:57AM -0600, Tim Pushor wrote: ulaw is 64Kb/s over a p2p link (or circuit switched in the PSTN world). If you then convert to IP there's at least a 20% overhead, can be more depending on the situation. Might take 100Kb/s Thats what I thought as well, so I was wondering why ISDN is 64K, but VOIP RTP/ulaw has been documented to be 64K per direction + packet overhead (132K+) ISDN is 64K per direction too, on a synchronous link i.e. 64K in both directions. I guess if you packetise etc then you've got to worry about both directions and the bandwidth to supoort it ... Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codecs and bandwidth
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 simultaneously. 80*4 = 320. You'd be using 320kbps down and 320kbps up, which is within your 1500kbps down / 384 kbps up. Someone please correct me if I'm wrong. - Dan Tim Pushor wrote: 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 consideration. Thanks for helping clear that up. Tim ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codecs and bandwidth
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 page) ... So even if one voice call sets up two 64 Kbit RTP streams over UDP over IP over Ethernet... Also, other pages from the wiki seem to talk about this as well when computing bandwidth. Is it because there are 2 RTP streams that don't know anything about each other, hence the 2x64k? Why would ISDN be any different? Thanks, Tim BTW I am assuming that this has very little to do with the D channel in ISDN since the call has already been setup via SIP .. Wiley Siler wrote: 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 -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 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 conversation, plus packet overhead (http://www.zytrax.com/tech/protocols/voip_rates.htm - seems to be down now - plus other references) for a total of over 128K per ulaw 'full duplex' voice conversation? Thanks Tim ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] codecs, asterisk, xpro
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 15:47:36 WARNING[6690]: channel.c:2115 ast_channel_make_compatible: No path to translate from SIP/victor-a02d(4) to SIP/dediana-1fd9(256) -- SIP/victor-a02d is ringing -- SIP/victor-a02d answered SIP/dediana-1fd9May 2 15:47:38 WARNING[6690]: channel.c:2115 ast_channel_make_compatible: No path to translate from SIP/dediana-1fd9(256) to SIP/victor-a02d(4)May 2 15:47:38 WARNING[6690]: app_dial.c:1006 dial_exec: Had to drop call because I couldn't make SIP/dediana-1fd9 compatible with SIP/victor-a02d If I do the same when both softphones have only G.711 set, everything works fine. It seems that Asterisk tries to use the first codec in SDP and ignores others. Does this make sense? Thanks in advance. Dov ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Codecs and * pass through...
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 like a net2phone device/user. Anyway the problem arose when attempting to call the MAX IP-10 device through the net2phone network. They seem to be using the G732.1 codec. I have in my settings in sip.conf allow=G732.1 or what ever flavour of the like and still I can not talk to the two devices. I googled a bit and came across the fact of * being able to do a pass through - well I was not successful and this subject is either simple or not well documented. The devices are using SIP and there is a bridge initiated, but there is no audio and no voice being passed through... I have tried connecting as the receiving device a GrandStrem Budge Tone-100 and still no luck. So all that I am inquiring is has anyone successfully done a pass through and if so can someone please guide me through some of the settings. I have set the [net2phone] with a canreinvite=yes - that a post on a forum also suggested, and that also did not work. On a separate issue: When the Grandstream Budge Tone-100 is connected on the internal network then the audio and the voice in both directions work fine. But when the device is connected on a separate network - ie on an other ADSL line, then the device doesn't send voice packets although is receives packets. I have opened up IPTABLES, to allow udp 5060 and udp 1:2 in both directions on any interface and the problem still persists. (SIP phone: Grandstream Budge Tone 100 connects to * and the call is answered by a Softphone X-Lite with all the codecs enabled. As far as I can tell thy both are speaking with a G711 codec ULaw/ALaw). So can anyone please give me a guideline or some advise on where to look to solve the problem. -- Kind Regards Etienne ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codecs and * pass through...
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, as long as you do not try playing any voice prompts to the channel. good luck. regards Clive = Phone I.T. http://www.phonehome.co.za On 13 Apr 2005 at 8:52, Etienne Pretorius wrote: 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 like a net2phone device/user. Anyway the problem arose when attempting to call the MAX IP-10 device through the net2phone network. They seem to be using the G732.1 codec. I have in my settings in sip.conf allow=G732.1 or what ever flavour of the like and still I can not talk to the two devices. I googled a bit and came across the fact of * being able to do a pass through - well I was not successful and this subject is either simple or not well documented. The devices are using SIP and there is a bridge initiated, but there is no audio and no voice being passed through... I have tried connecting as the receiving device a GrandStrem Budge Tone-100 and still no luck. So all that I am inquiring is has anyone successfully done a pass through and if so can someone please guide me through some of the settings. I have set the [net2phone] with a canreinvite=yes - that a post on a forum also suggested, and that also did not work. On a separate issue: When the Grandstream Budge Tone-100 is connected on the internal network then the audio and the voice in both directions work fine. But when the device is connected on a separate network - ie on an other ADSL line, then the device doesn't send voice packets although is receives packets. I have opened up IPTABLES, to allow udp 5060 and udp 1:2 in both directions on any interface and the problem still persists. (SIP phone: Grandstream Budge Tone 100 connects to * and the call is answered by a Softphone X-Lite with all the codecs enabled. As far as I can tell thy both are speaking with a G711 codec ULaw/ALaw). So can anyone please give me a guideline or some advise on where to look to solve the problem. -- Kind Regards Etienne ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codecs and * pass through...
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 take the voice prompts way. :-) (Didn't thought that it'll cause a problem - just the warnings and notices but continue still...) Thank you for that tip. "...For G723.1 passthrough, you just allow it..." --- So that is in "sip.conf" [general] disallow=all; allow=G723; allow=ulaw; allow=alaw; allow=gsm; (some text later) [net2phone] (some text) canreinvite=yes; (some text) --- Sources for net2phone: http://www.voip-info.org/tiki-index.php?page=Asterisk%20settings%20Net2phone http://www.voip-info.org/tiki-index.php?page=Net2phone PS - I do get a frame error about expecting 4 getting 256 when * is trying to initiate to call through to net2phone device MAX IP-10 through the net2phone network - could be that protocall you were talking about or have I completely missed the plot? Kind Regards Etienne [EMAIL PROTECTED] wrote: 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, as long as you do not try playing any voice prompts to the channel. good luck. regards Clive = Phone I.T. http://www.phonehome.co.za On 13 Apr 2005 at 8:52, Etienne Pretorius wrote: 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 like a net2phone device/user. Anyway the problem arose when attempting to call the MAX IP-10 device through the net2phone network. They seem to be using the G732.1 codec. I have in my settings in sip.conf allow=G732.1 or what ever flavour of the like and still I can not talk to the two devices. I googled a bit and came across the fact of * being able to do a pass through - well I was not successful and this subject is either simple or not well documented. The devices are using SIP and there is a bridge initiated, but there is no audio and no voice being passed through... I have tried connecting as the receiving device a GrandStrem Budge Tone-100 and still no luck. So all that I am inquiring is has anyone successfully done a pass through and if so can someone please guide me through some of the settings. I have set the [net2phone] with a canreinvite=yes - that a post on a forum also suggested, and that also did not work. On a separate issue: When the Grandstream Budge Tone-100 is connected on the internal network then the audio and the voice in both directions work fine. But when the device is connected on a separate network - ie on an other ADSL line, then the device doesn't send voice packets although is receives packets. I have opened up IPTABLES, to allow udp 5060 and udp 1:2 in both directions on any interface and the problem still persists. (SIP phone: Grandstream Budge Tone 100 connects to * and the call is answered by a Softphone X-Lite with all the codecs enabled. As far as I can tell thy both are "speaking" with a G711 codec ULaw/ALaw). So can anyone please give me a guideline or some advise on where to look to solve the problem. -- Kind Regards Etienne ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codecs and RealTime
Ha. Fixed this. Drop the disallow column. Then readd that column 'before' allow. Then it will work. The problem was that the disallow column came after the allow column, so the allow stuff was being procssed first then the disallow=all, which caused no codecs to be available. -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 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 INSERT INTO sip_buddies (allow,disallow) VALUES (g729;g726;gsm,all); the result is : Codecs : 0x0 (nothing) Codec Order : (none) Is it normal all this ? Matthew Boehm 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 - 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 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 DB for example disallow : all allow : ulaw then sip show peer ... : Codecs : 0x0 (nothing) Codec Order : (none) Any ideas ? -- Best Regards, Damian Minkov COSMOS Software Enterprises, Ltd. Tel:(+359-2) 983-32-62 Mobile: (+359-88) 853-28-25 E-Mail: [EMAIL PROTECTED] http://www.space-comm.com Post address: P. O. Box 941, 1000 Sofia, Bulgaria Office address: ap. 9, fl. 4, 11 August str., No. 43, 1202 Sofia, Bulgaria ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codecs and RealTime
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 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 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 DB for example disallow : all allow : ulaw then sip show peer ... : Codecs : 0x0 (nothing) Codec Order : (none) Any ideas ? -- Best Regards, Damian Minkov COSMOS Software Enterprises, Ltd. Tel:(+359-2) 983-32-62 Mobile: (+359-88) 853-28-25 E-Mail: [EMAIL PROTECTED] http://www.space-comm.com Post address: P. O. Box 941, 1000 Sofia, Bulgaria Office address: ap. 9, fl. 4, 11 August str., No. 43, 1202 Sofia, Bulgaria ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codecs and RealTime
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 INSERT INTO sip_buddies (allow,disallow) VALUES (g729;g726;gsm,all); the result is : Codecs : 0x0 (nothing) Codec Order : (none) Is it normal all this ? Matthew Boehm 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 - 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 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 DB for example disallow : all allow : ulaw then sip show peer ... : Codecs : 0x0 (nothing) Codec Order : (none) Any ideas ? -- Best Regards, Damian Minkov COSMOS Software Enterprises, Ltd. Tel:(+359-2) 983-32-62 Mobile: (+359-88) 853-28-25 E-Mail: [EMAIL PROTECTED] http://www.space-comm.com Post address: P. O. Box 941, 1000 Sofia, Bulgaria Office address: ap. 9, fl. 4, 11 August str., No. 43, 1202 Sofia, Bulgaria ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Codecs and RealTime
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 DB for example disallow : all allow : ulaw then sip show peer ... : Codecs : 0x0 (nothing) Codec Order : (none) Any ideas ? -- Best Regards, Damian Minkov COSMOS Software Enterprises, Ltd. Tel:(+359-2) 983-32-62 Mobile: (+359-88) 853-28-25 E-Mail: [EMAIL PROTECTED] http://www.space-comm.com Post address: P. O. Box 941, 1000 Sofia, Bulgaria Office address: ap. 9, fl. 4, 11 August str., No. 43, 1202 Sofia, Bulgaria ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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,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, the general section is loaded in the ast_config table while each sip extension is defined in sip_buddies. The allow and disallow statements are in the ast_config table and not in the sip_buddies table. Is this wrong? Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codecs and RealTime
You can have them in both locations. If in the ast_config, that will apply the allow/disallows to all sip clients. Use them in sip_buddies to specifcy for specific clients. -Matthew - Original Message - From: Greg - Cirelle Enterprises [EMAIL PROTECTED] To: Asterisk 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 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, the general section is loaded in the ast_config table while each sip extension is defined in sip_buddies. The allow and disallow statements are in the ast_config table and not in the sip_buddies table. Is this wrong? Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Codecs and echo
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. SIP to sip works fine no matter what codec so I am thinking it's either IAX or transcoding causing the issue. Any idea's/ Dee ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Codecs Problem?
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 know, what I possibly am doing wrong? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Codecs for fax traffic
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 visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codecs for fax traffic
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 list. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codecs for fax traffic
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.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 ? Using no compression wouldn't increase the scalability of my asterisk PBX ? This is considering I have no bandwidth issues in my network. Thanks __ Todavía no tenés tu Ciudad Internet Mail? Obtenelo ahora! - http://webmail.ciudad.com.ar Descargá Gratis el nuevo Internet Explorer 6.0, el mejor software para actualizar tu PC. http://www.ciudad.com.ar/ar/servicios/ie/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Codecs - Advantages
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 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 considering I have no bandwidth issues in my network. Thanks __ Todavía no tenés tu Ciudad Internet Mail? Obtenelo ahora! - http://webmail.ciudad.com.ar Descargá Gratis el nuevo Internet Explorer 6.0, el mejor software para actualizar tu PC. http://www.ciudad.com.ar/ar/servicios/ie/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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. -- Mensaje Original -- Enviado por: Sebastian Nocetti [EMAIL PROTECTED] Fecha: 19/07/2004 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:[EMAIL PROTECTED] En nombre de [EMAIL PROTECTED] Enviado el: Lunes, 19 de Julio de 2004 02:44 p.m. 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 ? Using no compression wouldn't increase the scalability of my asterisk PBX ? This is considering I have no bandwidth issues in my network. Thanks __ Todavía no tenés tu Ciudad Internet Mail? Obtenelo ahora! - http://webmail.ciudad.com.ar Descargá Gratis el nuevo Internet Explorer 6.0, el mejor software para actualizar tu PC. http://www.ciudad.com.ar/ar/servicios/ie/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Todavía no tenés tu Ciudad Internet Mail? Obtenelo ahora! - http://webmail.ciudad.com.ar Descargá Gratis el nuevo Internet Explorer 6.0, el mejor software para actualizar tu PC. http://www.ciudad.com.ar/ar/servicios/ie/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [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 you just wrote tells you EXACTLY what you should use! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codecs - Advantages
Hmmm, remember though that G.711u/a sends 64Kbit Frames, however it is not actually 64Kbit/call... We're not doing Circuit-switching here, we're doing packet switching. If you figure on IP overhead as well as the RTP information and of course SIP messages, then you add the 64Kbit of G.711u/a Payload you get around 80Kbit/sec Per user... that means on a Full T1 you only get 19 simultaneous calls, not 24... and on an E1 you would get around 25 simultaneous calls, not 32... And that's assuming that you have a very good SLA (or european equivalent) and that your latency is VERY low... I hope to god you use QoS on your router or your calls will sound like absolute crap... It's totally possible, don't be scared off by what I said, I'm even using * on a cable connection at home and it works just fine, you just need to do some preparation, it's a big change 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 :) 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 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 considering I have no bandwidth issues in my network. Thanks __ Todavía no tenés tu Ciudad Internet Mail? Obtenelo ahora! - http://webmail.ciudad.com.ar Descargá Gratis el nuevo Internet Explorer 6.0, el mejor software para actualizar tu PC. http://www.ciudad.com.ar/ar/servicios/ie/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE:RE: [Asterisk-Users] Codecs - Advantages
Is this bascially setting your bandwith value = high inside of iax.conf? Or is there another place to designate the codec? Thanks, Wiley -Original Message- From: Senad Jordanovic [mailto:[EMAIL PROTECTED] Sent: Monday, July 19, 2004 2:11 PM To: [EMAIL PROTECTED] Subject: RE: [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 you just wrote tells you EXACTLY what you should use! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codecs - Advantages
-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.729 ? Bandwidth only ? Using no compression wouldn't increase the scalability of my asterisk PBX ? This is considering I have no bandwidth issues in my network. Thanks Actually besides from the best sound quality it's also not heavy on the CPU or bandwidth. It's actually better than any of the other. - -- Steve They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFA/GxjljK16xgETzkRAqf7AJ92M97CYwKYTYAOM843xafpl5pD4gCeJCPp Hwt5G6/tnw9Eq6T0+/2vCj0= =OcuB -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Codecs and pauses
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 think I'm using the correct one - I'm in the UK) I get intermittent pauses on the call. Initially I thought it was just a connectivity thing but I get a latency of less than 10ms to iax.voiptalk.org and I'm using a 2mb leased line. To further ensure it wasn't something on the line I've disconnected everything except the * box, a 7960 phone. My phones are all 7960's using SIP. There is a X100P card in the server moh timing etc but it isn't connected to the pstn. The * box itself is a PIII 833 with 256MB. Not at all stressed as this is all it does. Any hints or tips would be really useful as I'm stumped now. Thanks Matt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Codecs and pauses
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 Maxexcessbuffer Now the problem I have is that telappliant have lost one of there external routers so I can't connect at the moment :-) DOH! What is it with networks breaking today, first Dilbert and now Telappliant??? :-) Probably got something to do with the summer solstice, laylines etc :- -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 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 :-) What is it with networks breaking today, first Dilbert and now Telappliant??? :-) HTH Chris -- Chris -- E Mail: [EMAIL PROTECTED] SIP: [EMAIL PROTECTED] IAXTEL: 17003366726 On Wed, 23 Jun 2004, Matt wrote: 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 think I'm using the correct one - I'm in the UK) I get intermittent pauses on the call. Initially I thought it was just a connectivity thing but I get a latency of less than 10ms to iax.voiptalk.org and I'm using a 2mb leased line. To further ensure it wasn't something on the line I've disconnected everything except the * box, a 7960 phone. My phones are all 7960's using SIP. There is a X100P card in the server moh timing etc but it isn't connected to the pstn. The * box itself is a PIII 833 with 256MB. Not at all stressed as this is all it does. Any hints or tips would be really useful as I'm stumped now. Thanks Matt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Codecs and pauses
Chris, Just tried turning on the jitterbuffer, but I will have a play, as I do get occassional dropout when my machine checks for mail I'll also have a play once Telappliant is back up and post the results. Cheers Matt -Original Message- From: [EMAIL PROTECTED] [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 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 Maxexcessbuffer I haven't! Just tried turning on the jitterbuffer, but I will have a play, as I do get occassional dropout when my machine checks for mail, but as I'm only on ADSL I shouldn't be surprised really. Probably the longest sentance in the world! What is it with networks breaking today, first Dilbert and now Telappliant??? :-) Probably got something to do with the summer solstice, laylines etc :- If it's anything to do with that, it's two days late! At least Dilbert has come back now :-) Chris -- E Mail: [EMAIL PROTECTED] SIP: [EMAIL PROTECTED] IAXTEL: 17003366726 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Codecs compile error on yellowdog
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 -DneedFunctionprototypes=1 -funroll-loops -fPIC -DSASR -DNDEBUG -DWAV49 -I./inc src/add.c puting child (gsm/lib/libgsm.a) leave child put child (arc/add.o) leave child cc1: invalid option `arch=' Got SIGCHLD; 1 unreaped children reaping loosing child make[1]; *** [src/add.o] error 1 removing child from chain then it does the same for libgsm.a make: *** [gsm/lib/libgsm.a] Error 2 I'm thinking that the libgsm is not installed in my distribution can anyone guide me to on what I need to install to get passed this error? TIA --jeff --- jeff donovan basd network operations (610) 807 5571 x4 AIM xtdonovan fwd# 248217 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Codecs and more analog lines?
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 machines. rgds, /staffan -- -- Staffan Kerker / KIT Communications, AerotechTelub mail: [EMAIL PROTECTED] Don't get involved in politics man, just play the gig... /Sgt. Floyd, Electric Mayhem Band ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Codecs and call failure with Grandstream
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 via * to remote IAX to PSTN. Sound stream is established from PSTN to GS but no sound from GS to PSTN. By the way, calls from GS to the PSTN via * worked correctly. Only the IAX bridge failed. It turned out to be a codec problem. The fix is the same as well. Add to sip.conf [general] (or on a phone by phone basis): disallow=all allow=alaw allow=ulaw You may also need to enable additional codecs. Stephen R. Besch ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] codecs questions
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.711 A-law 16 (1 4) MPEG-2 layer 3 32 (1 5) ADPCM 64 (1 6) 16 bit Signed Linear PCM 128 (1 7) LPC10 256 (1 8) G.729A audio 512 (1 9) SpeeX 1024 (1 10) iLBC 65536 (1 16) JPEG image 131072 (1 17) PNG image 262144 (1 18) H.261 Video 524288 (1 19) H.263 Video My questions are: 1) What is the best configuration to use with fwd? 2) my sip.conf is correct? I can make calls to fwd but i have problems to listen to people that calls me. 3)Asterisk is using the G.723.1 as the first choice? 4) I have to configure my phones with the same codec that asterisk is using or the interoperable option in the snom phone is correct? Thanks Miklos ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] codecs questions
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 A-law 3)Asterisk is using the G.723.1 as the first choice? No. The only codecs that you will negotiate will be ulaw, alaw, and gsm, as specified in your config. 'show codecs' is informational only. It indicates nada, nothing, zip, zilch (get the idea?) about your configuration. Asterisk will use whatever codec is available on both sides, followed preferentially by the codec which has the least cost to translate. See the 'show translation' matrix. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CODECS and thier practical usage stats
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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CODECS and thier practical usage stats
John, Tx ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CODECS and thier practical usage stats
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 and thier practical usage stats
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 mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codecs
-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 audio codecs that's causing it. Is something broken in chan_h323? To follow up on this question, I flashed the Siemens ip phone with a really really old image and forced G711 alaw. This made it behave nicely. - -- Regards, Tais M. Hansen ComX -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.2 (GNU/Linux) iD8DBQE/KhWM2TEAILET3McRArR9AJ4hdDwaUtvEuLpKAwHH5m1joTfF2QCcDIM1 LaErjBZO5z1VD0Rti9L8khU= =C18H -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Codecs
-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 in chan_h323? - -- Regards, Tais M. Hansen ComX -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.2 (GNU/Linux) iD8DBQE/Jl7g2TEAILET3McRAmTzAJwNCGxGKJI5ZK92Vnri0CUJM+hFpQCfZo5o vf6vMWBmYjseyvtczNuM688= =8xpI -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[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 something a little less bandwidth intensive Kim Callis
RE: [Asterisk-Users] Codecs for use with Cisco 7960 and ATA-186
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] [mailto:[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 wanted to see if I could make use of something a little less bandwidth intensive Kim Callis
Re: [Asterisk-Users] Codecs for use with Cisco 7960 and ATA-186
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 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] [mailto:[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 wanted to see if I could make use of something a little less bandwidth intensive. Kim Callis ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Codecs for use with Cisco 7960 and ATA-186
I think that G.711 is the default codec that my system is using. At my home office I am running through a DSL connection with a Cisco ATA-186, and I notice that when I call to my office which are 7960, I get problems with the office hearing me, sound issues, and other things... I thought that 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 7960 and ATA-186 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 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] [mailto:[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 wanted to see if I could make use of something a little less bandwidth intensive. Kim Callis ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] codecs question ..
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 rtp.c, Line 330 (ast_rtp_read): Unknown RTP codec 19 received can anybody tell me what this means ( how I may fix it ?) cheers Dave ps. the service i'm connecting to uses G723 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] * + Codecs + Hardphones??
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 G.711, the problem with this codec is that it requires approx 64Kbps which is a little high for what I am trying to do.. 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 you use?? Thanks.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
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 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 G.711, the problem with this codec is that it requires approx 64Kbps which is a little high for what I am trying to do.. 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 you use?? G729 is currently not available due to patent and license issues. You will find that G711 is fine on a local network, and asterisk does real well using IAX protocol and GSM codec. So if you are deploying enough phones to make it worth while to drop a * machine on the LAN with the hardphones you are set. Otherwise you might want to look at the Snom phone since it supports GSM. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] * + Codecs + Hardphones??
What about G.723.1? Anybody have any experiece with this? I know I can order the documentation (including C source) from the ITU, what does this entitle me to do? Are there any licencing gotcha's to going with this approach? Lenny -Original Message- From: Steven Critchfield [mailto:[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 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 G.711, the problem with this codec is that it requires approx 64Kbps which is a little high for what I am trying to do.. 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 you use?? G729 is currently not available due to patent and license issues. You will find that G711 is fine on a local network, and asterisk does real well using IAX protocol and GSM codec. So if you are deploying enough phones to make it worth while to drop a * machine on the LAN with the hardphones you are set. Otherwise you might want to look at the Snom phone since it supports GSM. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
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 or if * supports the G.729 codecs or others.. Which phones do the * user commumity find work the best?? and which codecs do you use?? You can purchase G.729 from Digium at $10/channel. Contact Greg Vance (256-428-6262). The G.729 is currently considered beta. Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * + Codecs + Hardphones??
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 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 you use?? You can purchase G.729 from Digium at $10/channel. Contact Greg Vance (256-428-6262). The G.729 is currently considered beta. Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] * + Codecs + Hardphones??
For $10 a pop, I would buy 24 to cover upto a T1 (and I've already got my call into Greg). This is exactly what I'm looking for. I suspect nothing bad would happen if you went over, except for that pang of guilt you may feel until you ante up for it's use :-) I'm personally more interested in the performance of the codec (ie what kind of raw power will I need to run it, how many can I run at once on a decently powered box etc...) Lenny -Original Message- From: James O. Sizemore III [mailto:[EMAIL PROTECTED] Sent: Thursday, March 27, 2003 1:29 PM To: [EMAIL PROTECTED] Subject: 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 or if * supports the G.729 codecs or others.. Which phones do the * user commumity find work the best?? and which codecs do you use?? You can purchase G.729 from Digium at $10/channel. Contact Greg Vance (256-428-6262). The G.729 is currently considered beta. Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] * + Codecs + Hardphones??
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 G.729 code on a trial basis?) Jared Smith On Thu, 2003-03-27 at 14:01, Lenny Post wrote: I'm personally more interested in the performance of the codec (ie what kind of raw power will I need to run it, how many can I run at once on a decently powered box etc...) Lenny ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * + Codecs + Hardphones??
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??
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 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 G.729 code on a trial basis?) Jared Smith On Thu, 2003-03-27 at 14:01, Lenny Post wrote: I'm personally more interested in the performance of the codec (ie what kind of raw power will I need to run it, how many can I run at once on a decently powered box etc...) Lenny ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * + Codecs + Hardphones??
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 you use?? You can purchase G.729 from Digium at $10/channel. Contact Greg Vance (256-428-6262). The G.729 is currently considered beta. 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. Thanks. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
