I have been testing MixMonitor and Monitor to record some calls in Asterisk and I have noticed that MixMonitor works fine whereas in the Monitor files of the 2 separate channels, we can find little cuts of the audio. We are using U law codec and ..
> > I have a strange audio delay behaviour when placing a call between two > SIP devices using the same codec. > In my example, I have two devices forced to use GSM codec. > When placing a call, the first ~9sec have no audio, then the audio > sta..
Im relatively certain this is a silly question, but is anyone willing
to share their experience with asterisk in the amazon cloud?
Does it work?Or do timing issues mess with audio?..
we have 4 asterisk, versions are 1.4.35 1.4.36 18.104.22.168 and 1.4.42 One GrandStream GXV3000 is used for the tests. He is registered to asterisk 22.214.171.124 asa well as 1.4.35. Calling echo test is OK on both servers, get audio and video. Calling echo t..
On Fri, Jul 1, 2011 at 12:05 PM, Larry Moorewrote: > ** > On 28/06/2011 6:59 PM, Matteo Campana wrote: > > > >Larry, > I have the SIP debug taken from asterisk. > In this debug: 126.96.36.199 —> IP SIP PROXY >188.8.131.52 —> IP UAC (Linksys SPA 962) >9.10.11..
all, we have a problem with a reinvite sent by our SIP provider to change audio codec due to the recognition of a fax tone. After that the SIP call session has been established (INVITE and 200 OK) we have the following codec situation: UACASTERISK ..