[asterisk-users] SIREN14 call setup and record/playback

[Tom Browning]

> What are you reaching out to exactly? It would need to be a Siren14
> capable. Also, do you have the Siren codec binary installed? It's not part
> of the Asterisk distribution.
>

Inbound calls to Asterisk work (from a platform that supports both Siren14
and G.711).  Leaving ulaw out of the allow list forces the inbound call to
negotiate Siren14.  In terms of debugging the outbound call, what I'm
calling hasn't yet come into play as the error "No audio format found to
offer. Cancelling call to blahblah" happens before any INVITE is transmitted
from Asterisk.

Adding ulaw back into the allow list causes the outbound call to actually
transmit the INVITE, but then they negotiate ulaw and not Siren14.
Replacing ulaw with alaw in the allow list (a codec that is NOT supported by
the platform I am calling) will also allow the outbound INVITE to be sent
but no suitable codec is negotiated.  Again, inbound calls do negotiate
successfully.

The difference seems to be related to 'jointcapabilities' vs. 'capabilities'
in the chan_sip.c

Are there other configuration settings I can adjust to negotiate Siren14 on
outbound calls without hacking chan_sip.c?

I'm hoping I've completely missed something simple.

Also, you should know that all Siren14 calls are presently downsampled to 16
> KHz, so are effectively Siren7.Asterisk doesn't presently support sample
> rates beyond 16 KHz.
>

format_siren14.c clearly seems to state support for the 48Kbps 32Khz Siren14
flavor and Kevin Flemming's earlier reply to this thread implied support for
same.

Tom
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091104/2f5aa0d5/attachment.htm