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

[Tom Browning]

On Tue, Nov 10, 2009 at 1:20 PM, Kevin P. Fleming <kpfleming at digium.com> wrote:
>
>
> 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.



Sorry, there is no mailto: header in the spool file, that must be
gmail parsing my paste as html and adding that format.

setting gmail to plain text:

Channel: SIP/foo at bar.com
CallerID: testcall
Context: default
Extension: demo
Codecs: siren14


The only difference between the call attempt that actually sends the
INVITE and the call attempt that complains is 'ulaw' vs 'siren14' in
the sip.conf allow= and spol file Codecs: header.

Clearly those codec choices are not treated the same to build an
outbound INVITE.

Tom