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
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
