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

[Kevin P. Fleming]

Tom Browning wrote:

> vi call.txt
>     Channel: SIP/foo at bar.com <mailto:foo at bar.com>
>     CallerID: testcall
>     Context: default
>     Extension: demo
>     Codecs: siren14
> 
> cp call.txt /var/spool/asterisk/outgoing/
> 
>     -- Attempting call on SIP/foo at bar.com <mailto:foo at bar.com> for
> demo at default:1 (Retry 1)
> [Nov 10 07:43:13] WARNING[27630]: chan_sip.c:5735 sip_call: No audio
> format found to offer. Cancelling call to foo

Please run this test with the 'debug' level enabled for the 'console'
channel in logger.conf, and then ensure that you have 'core set verbose
10' and 'core set debug 10' before attempting the outbound call. This
should give us some information about why chan_sip did not allow the
channel to be created. I suspect it may be because your defined peer for
bar.com was not actually used, since your spool file has
"<mailto:foo at bar.com>" in the Channel header, since that is not valid
syntax.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpfleming at digium.com
Check us out at www.digium.com & www.asterisk.org