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

[Kevin P. Fleming]

Tom Browning wrote:
> On Tue, Nov 10, 2009 at 3:39 PM, Kevin P. Fleming <kpfleming at digium.com> wrote:
>> They are, but we won't be able to know what is happening unless you post
>> a detailed console log like I suggested in my previous reply.
> 
>     -- Attempting call on SIP/foo at bar.com for demo at default:1 (Retry 1)
> [Nov 10 17:32:37] DEBUG[28977]: chan_sip.c:24104 sip_request_call:
> Asked to create a SIP channel with formats: 0x40 (slin)

If you've specified the codecs properly in the spool file, then this is
clearly a bug, because the channel was requested from chan_sip in signed
linear (slin) format, not siren14. Since there is no siren14 codec
module available, chan_sip can't provide slin to the Asterisk core and
siren14 to the SIP peer, so it fails.

-- 
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