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 wav files for the recording.
Anyone have suffered the same problems. It is that Monitor does not work well. It is another way to record the 2 legs of the call separately by using MixMonitor?
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> 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
> starts trasmitting.
> If I force one phone to use GSM and the other ULAW/ALAW, everything
> works fine. If I had to guess, I'd say that you don't have canreinvite/directmedia=no in sip.conf and there is possibly a NAT between the phones…
we have 4 asterisk, versions are 1.4.35 1.4.36 188.8.131.52 and 1.4.42 One
GrandStream GXV3000 is used for the tests. He is registered to asterisk
184.108.40.206 asa well as 1.4.35. Calling echo test is OK on both servers,
get audio and video. Calling echo test from asterisk 1.4.36 bye a SIP
trunk from both others servers is also working well. What fail, is video on echo test from asterisk 1.4.42 using SIP trunks:
we have audio but no videobeside the fact that video codec are
negociated as shown below. All servers are on public…
On Fri, Jul 1, 2011 at 12:05 PM, Larry Moore
> On 28/06/2011 6:59 PM, Matteo Campana wrote:
> Hi Larry,
> I have the SIP debug taken from asterisk.
> In this debug: 220.127.116.11 ---> IP SIP PROXY
> 18.104.22.168 ---> IP UAC (Linksys SPA 962)
> 22.214.171.124 ---> IP ASTERISK to connect to the
> 126.96.36.199 --> IP PROVIDER
> 188.8.131.52 --> IP ASTERISK
> The SIP debug is available at…