On Sat, Jun 18, 2011 at 6:40 AM, Larry Moore <[email protected]> wrote:
> On 18/06/2011 5:36 AM, Matteo Campana wrote: > >> >> Inviato da iPhone >> >> Il giorno 16/giu/2011, alle ore 16:37, Eric Wieling<[email protected]> >> 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 Larry, I have the SIP debug taken from asterisk. In this debug: 1.2.3.4 ---> IP SIP PROXY 5.6.7.8 ---> IP UAC (Linksys SPA 962) 9.10.11.12 ---> IP ASTERISK to connect to the provider 13.14.15.16 --> IP PROVIDER 17.18.19.20 --> IP ASTERISK The SIP debug is available at this link: http://pastebin.com/9DrFDmeC Thanks in advance, 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<http://lists.digium.com/mailman/listinfo/asterisk-users> >
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