<HTML>
<FONT SIZE="2" POINTSIZE="11" DEFAULT="SIZE">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.<BR>
<BR>
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.<BR>
<BR>
Michael<BR>
<BR>
--Original Message Text---<BR>
<B>From:</B> Tom Browning<BR>
<B>Date:</B> Wed, 4 Nov 2009 17:24:32 -0500<BR>
<BR>
Continuing the siren14 usage thread:<BR>
<BR>
sip.conf has:<BR>
<BR>
disallow=all ; First disallow all codecs<BR>
allow=siren14 ; <BR>
<BR>
<BR>
Should I be able to originate an outbound call with siren14 as my only codec?<BR>
<BR>
When I try originate using either the spool file or a CLI originate command I get:<BR>
<BR>
[Nov 4 17:21:49] WARNING[28427]: chan_sip.c:5722 sip_call: No audio format found to offer. Cancelling call to blahblah<BR>
<BR>
Inbound calls, record and playback work just great. Now I want to reach out with SIREN14<BR>
<BR>
Thanks in advance,<BR>
<BR>
Tom<BR>
<BR>
<BR>
</HTML>
<HTML>
<LEFT>
<FONT FACE="Arial" COLOR="#000001" SIZE="2" POINTSIZE="11" DEFAULT="ALL">
--<br>
Michael Graves<br>
mgraves<at>mstvp.com<br>
http://www.mgraves.org<br>
o713-861-4005<br>
c713-201-1262<br>
sip:mgraves@mstvp.onsip.com<br>
skype mjgraves<br>
Twitter mjgraves<br>
<br>
</HTML>