Hello, Does asterisk do transcoding when the call goes through the system, codecs are the same but signaling protocol is changed. example: SIP with GSM ---> IAX with GSM What quality destruction happen when I use transcoding? I know this is not a concrete/precise question, but I would like to know how is it in general. What CPU performance is needed for transcoding 30 channels e.g. from GSM to g723.1 in general? Thanks in advance, Thomas
Yes, asterisk does SIP to IAX/H323/PSTN/MGCP etc and any other combination. regards Martin On Sun, 2 Mar 2003, Thomas Jalsovsky wrote:> > Hello, > > Does asterisk do transcoding when the call goes > through the system, codecs are the same but signaling protocol is changed. > example: > SIP with GSM ---> IAX with GSM > > What quality destruction happen when I use transcoding? I know > this is not a concrete/precise question, but I would like to know how is > it in general. > > What CPU performance is needed for transcoding 30 channels e.g. > from GSM to g723.1 in general? > > Thanks in advance, > Thomas > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users at lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users >
Maybe it is not answer to your question but... Do you use g723.1 with asterisk? How did you get it? As far as I know it is not royalty free:-( Best regards, KRzysztof BUjak ----- Original Message ----- From: "Thomas Jalsovsky" <thomasj@pegasus.eworld.hu> To: <asterisk-users@lists.digium.com> Sent: Sunday, March 02, 2003 12:40 PM Subject: [Asterisk-Users] Transcoding> > Hello, > > Does asterisk do transcoding when the call goes > through the system, codecs are the same but signaling protocol is changed. > example: > SIP with GSM ---> IAX with GSM > > What quality destruction happen when I use transcoding? I know > this is not a concrete/precise question, but I would like to know how is > it in general. > > What CPU performance is needed for transcoding 30 channels e.g. > from GSM to g723.1 in general? > > Thanks in advance, > Thomas > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > >==========================================================================> Ta wiadomosc zostala sprawdzona na obecnosc wirusow komputerowych przez system antywirusowy na serwerze IT Form.>==========================================================================Ta wiadomosc zostala sprawdzona na obecnosc wirusow komputerowych przez system antywirusowy na serwerze IT Form.
I am tinkering with a D-Link DG-104S MGCP adapter and then X-Lite SIP softphone. The D-Link supports g711 and g729. X-Lite supports g711 gsm and g723.1. If I enable ulaw, then I can get them to talk to each other just fine. Meetme works fine, to. If I add disallow=all allow=gsm to /etc/asterisk/sip.conf, then meetme still works, but the phone attached to the DG-104S cannot hear the softphone. Is asterisk _supposed_ to transcode MGCP <-> SIP traffic?
Do you have allow=gsm in mgcp.conf? Do you have the X-Lite set to deal with GSM? What shows on the Asterisk CLI when you call SIP to MGCP using GSM? Jeremy McNamara Michael Van Donselaar wrote:>I am tinkering with a D-Link DG-104S MGCP adapter and then X-Lite SIP >softphone. > >The D-Link supports g711 and g729. > >X-Lite supports g711 gsm and g723.1. > >If I enable ulaw, then I can get them to talk to each other just fine. >Meetme works fine, to. > >If I add > >disallow=all >allow=gsm > >to /etc/asterisk/sip.conf, then meetme still works, but the phone >attached to the DG-104S cannot hear the softphone. > >Is asterisk _supposed_ to transcode MGCP <-> SIP traffic? > >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users > >
I have a Budgetone and an ATA but none of them support GSM. I?d like to place call to the PSTN with my X100P via a WAN (64kbps). g711 is out of the question. Can * transcode from g723.1 to GSM? How costly is it? I have tried different configurations on sip.conf and extensions.conf but have had no luck. Is this transcoding possible? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030627/f77dafac/attachment.htm
I though that Asterisk would transcode between codecs! All my SIP devices support G729a & 711a/u but my ISDN PRI/BRI lines use 711a/u. Asterisk is quite happy to accept a call from a SIP device using G729a and then complains that it can't translate into G711 to go onto the ISDN network. Does anybody know if there is some setting somewhere or if this is how it is supposed to work
AFAIK you need a license from Digium if you want to transcode to/from G729a... Hope this information is correct and it helps Regards Guido Hecken> I though that Asterisk would transcode between codecs! All my SIP devicessupport> G729a & 711a/u but my ISDN PRI/BRI lines use 711a/u. Asterisk is quitehappy to> accept a call from a SIP device using G729a and then complains that itcan't translate> into G711 to go onto the ISDN network. Does anybody know if there is somesetting> somewhere or if this is how it is supposed to work
Message is "no translator path exists for channel type CAPI (native 8) to 256">>> radamson@routers.com 19/07/05 13:44:29 >>>> I though that Asterisk would transcode between codecs! All my SIP devices support G729a &711a/u but my ISDN PRI/BRI lines use 711a/u. Asterisk is quite happy to accept a call from a SIP device using G729a and then complains that it can't translate into G711 to go onto the ISDN network. Does anybody know if there is some setting somewhere or if this is how it is supposed to work What does the sip debug show? Any CLI data to give us a clue? _______________________________________________ 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 Scanned by MailDefender - managed email security from intY - www.maildefender.net
Yes, I've purchased 20 G729a licenses and I know that * uses them OK>>> erik@infopact.nl 19/07/05 13:27:04 >>>Rich Adamson wrote:>>I though that Asterisk would transcode between codecs! All my SIP devices support G729a & >> >> >711a/u but my ISDN PRI/BRI lines use 711a/u. Asterisk is quite happy to accept a call from a SIP >device using G729a and then complains that it can't translate into G711 to go onto the ISDN >network. Does anybody know if there is some setting somewhere or if this is how it is supposed >to work > > >What does the sip debug show? > >Any CLI data to give us a clue? > > >_______________________________________________ >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 > > > >And you did buy some G729 codec licences? * won't do anything but passtrough without them _______________________________________________ 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 Scanned by MailDefender - managed email security from intY - www.maildefender.net
Why didn't I think of using that command... It shows all "-" for G729a which is presumably why I'm having a problem I have purchased 20 licenses from Digium, downloaded binary, registered the binary correctly, placed it in the correct directory and it is listed specifically in SIP.conf I'm sure that I have had some calls between SIP phones using G729a via Asterisk (not re-invited) How can I be sure that the G729a codec is working correctly? Very much appreciate your response>>> radamson@routers.com 19/07/05 14:31:57 >>>What does your 'show translation' look like? Can you copy/paste the specific *.conf entries for the sip devices and capi?> Message is "no translator path exists for channel type CAPI (native 8) to 256" > > >>> radamson@routers.com 19/07/05 13:44:29 >>> > > > I though that Asterisk would transcode between codecs! All my SIP devices support G729a & > 711a/u but my ISDN PRI/BRI lines use 711a/u. Asterisk is quite happy to accept a call from aSIP> device using G729a and then complains that it can't translate into G711 to go onto the ISDN > network. Does anybody know if there is some setting somewhere or if this is how it is supposed> to work > > > What does the sip debug show? > > Any CLI data to give us a clue?_______________________________________________ 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 Scanned by MailDefender - managed email security from intY - www.maildefender.net
Hi all! Is it possible to have a setup with a server only dedicated for transcoding from ulaw/alaw to G729. What is the capacity of a server like that in simultaneous calls? Regards Anders -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051001/83f4cbc5/attachment.htm
Quoting Anders Svensson <anders@bobascom.com>: this page might help. http://www.voip-info.org/tiki-index.php?page=Asterisk+dimensioning> > > Hi all! > > Is it possible to have a setup with a server only dedicated for transcoding > from ulaw/alaw to G729. What is the capacity of a server like that in > simultaneous calls? > > > > > > Regards > > Anders > > > > > >---------------------------------------------------------------- This message was sent using IMP, the Internet Messaging Program.