app_Originate codec translation?

app_Originate works from sip2sip with no problems but on sip2iax2 gives me
asterisk[27104]: WARNING[4118][C-00000019]: chan_iax2.c:12533 in iax2_request: Unable to create translator path for (g722|g729|gsm|alaw) to (slin192) on IAX2/Escape-31565

Where this bloody dawn slin192 codec comes from?
iax.conf allows the same codecs as sip.conf (g722|g729|gsm|alaw)

You probably have a local channel there as well.

Have you the the codec option

  • Codecs - Comma-separated list of codecs to use for this call.

I do not understand your sugestions.
Where is codec option on app_Originate?
And the error message seems clear, it is trying a translations path fom
g722|g729|gsm|alaw to slin192 wich does not exists. But why is it choosing slin192?

If you have a local channel, the channel will be setup without reference to the codecs on the physical channels. The situation used to be that the default, in that case, was slin, but is is possible that,with the demand for enhanced audio, the default policy is now to use the highest quality code supported. Remember, for local channels, at that stage it doesn’t know what codec it will eventually have to translate to.

I hadn’t come across anything more than slin16, until recently, and nothing as high as slin192

If using local channels, with G.729, you need to originate with g729, if you want to avoid using licences.

That option is only available if you are using AMI Originate action

I still do not undertand where is slin192 comes from, sure is the highest but even is not compiled in.
In true this Originite is used to allow another phone to join conference on ConfBridge.
For a lack o a better idea I tried changing values on this but honestly I think have to do
how the channels themselves

I was trying in confbridge.conf
internal_sample_rate = 16000
mixing_interval = 20

sip2sip works fine but not bridging sip2iax2
Anyway app_Dial sip2iax2 works, no problem and the codecs get right.

Maybe some relation to this on page 466 pdf Admin reference ver 14
Mixing Interval
The mixing interval for a conference is defined in its Bridge Profile. The allowable options are 10, 20, 40, and 80, all in milliseconds. Usage of 80ms
mixing intervals is only supported for conferences that are sampled at 8, 12, 16, 24, 32, and 48kHz. Usage of 40ms intervals includes all of the
aforementioned sampling rates as well as 96kHz. 192kHz sampled conferences are only supported at 10 and 20ms mixing intervals. These limitations
are imposed because higher mixing intervals at the higher sampling rates causes large increases in memory consumption. Adventurous users may,
through changing of the MAX_DATALEN define in bridge_softmix.c allow 96kHz and 192kHz sampled conferences to operate at longer intervals - set
to 16192 for 96kHz at 80ms or 32384 for 192kHz at 80ms, recompile, and restart.

But I did not change softmix.c anyway