That's my point - SFA comes with a g729 licence, so why can't it transcode to the DAHDI channel ?
Thanks also for the info. Very useful. Julian 2009/12/6 Roeften <[email protected]>: > From what I understand your sip client can handle g729 whereas for DAHDI you > need transcoding to a|ulaw. > > I am using it with no problems (have g729 licenses as well though). > > A bit off topic, I have found some extra configuration that is not really in > the docs (or I could not find them): > > fullname=Your full name > country=gr > language=en > city=City > province=Province > phone_home=+fullinternationalnumber > phone_office=+fullinternationalnumber > [email protected] > homepage=http://www.example.com > avatar=/var/lib/asterisk/images/skype100x100.jpg > > Just a note the country code has to be lower case (i.e GR would not work). > > Panos > > On Sun, Dec 6, 2009 at 9:40 AM, Julian Lyndon-Smith <[email protected]> > wrote: >> >> Ok. So I bought 2x skpye channels. Doesn't that mean I have 2xg729 as well >> ? >> >> If so, why do I have the problem ? And would this affect local >> channels as well ? >> >> Julian >> >> 2009/12/6 Kevin P. Fleming <[email protected]>: >> > Julian Lyndon-Smith wrote: >> > >> >> external => ddi => dial(skype) >> >> >> >> and got a load of static with >> >> >> >> WARNING[15328]: channel.c:3098 set_format: Unable to find a codec >> >> translation path from 0x100 (g729) to 0x8 (alaw) >> >> >> >> on the console. >> >> >> >> Fired up a sip client, made the same call, and all was ok. >> >> >> >> Any clues ? >> > >> > The clues are in the documentation; SkypeIn and SkypeOut use G.729 for >> > nearly all calls, so handling calls via those paths requires a G.729 >> > transcoder on the system if the target of the call will not also be >> > using G.729. This is why the Skype For Asterisk license includes >> > licenses for Digium's G.729 software transcoder as well. >> > >> > -- >> > Kevin P. Fleming >> > Digium, Inc. | Director of Software Technologies >> > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA >> > skype: kpfleming | jabber: [email protected] >> > Check us out at www.digium.com & www.asterisk.org >> > >> > _______________________________________________ >> > -- 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 >> > >> >> _______________________________________________ >> -- 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 > > > _______________________________________________ > -- 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 > _______________________________________________ -- 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
