Hello all * usrs,
Does anyone know how to change your *'s primary codec and still keep the g723.1
pass through capability? Or altenativly make the phones renegociate the codec
to use when a channel is about to be bridged?
Thank you,
Etienne
___
Asteris
Hello all,
For the g723.1 pass-through the incoming call works fine, I have been playing
around a bit and was wandering if you can dynamically change the channel and
the associated devices using the channel to change their codecs for the
outbound call.
I have the following setup in extensions.con
Hello all,
For the g723.1 pass-through the incoming call works fine, I have been playing
around a bit and was wandering if you can dynamically change the channel and
the associated devices using the channel to change their codecs for the
outbound call.
I have the following setup in extensions.con
Sorry-> Solved my own problem. I was playing around with the GS BudgeTone 100
and had set up call forwarding on...
<-- SIP read from 192.168.10.24:5060:
SIP/2.0 302 Moved Temporarily
Via: SIP/2.0/UDP 192.168.10.1:5060;branch=z9hG4bK0bf1b64b
From: "asterisk" ;tag=as4a953271
To: ;tag=6fe736daf422320
Got some debug info... please see attachement.
Quoting [EMAIL PROTECTED]:
> Hello everyone.
> How was your weekend?
>
> Anyway...
> 'Got SIP response 302 "Moved Temporarily" back from 192.168.10.24'
>
> Lately I've been getting this error... well i am at a loss as to why I am
> getting this when
Hello everyone.
How was your weekend?
Anyway...
'Got SIP response 302 "Moved Temporarily" back from 192.168.10.24'
Lately I've been getting this error... well i am at a loss as to why I am
getting this when on Friday I was able to make a pass-through call with no
problems.
+--+ +--
