Inviato da iPhone
Il giorno 18/giu/2011, alle ore 06:40, Larry Moore <lmo...@starwon.com.au> ha scritto: > On 18/06/2011 5:36 AM, Matteo Campana wrote: >> >> Inviato da iPhone >> >> Il giorno 16/giu/2011, alle ore 16:37, Eric Wieling<ewiel...@nyigc.com> ha >> scritto: >> >>> We experience the same thing. The solution we use is to not change codecs >>> in the middle of a call. I assumed it was an issue with our upstream. >> >> Hi Eric, >> this behavior is an asterisk bug or asterisk can never change the codec "on >> the fly"? >> >> >> Thanks, >> Matteo >> > > The problem reported occurs after a fax tone is detected, one might expect > T.38 or G711 to be used to handle the fax, not G729! > > There is no configuration file information for each of the nodes/peers, no > debug of each peer involved nor a trace of the packets hence no one will > have ideas! > > Larry. > Hi, I'm out of the office this week, next Monday I will send the debug to the list. However I think It's strange asterisk behavior: it says 200 OK after a re-invite by the provider, but stops to send rtp. Regards, Matteo -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users