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

[Tom Browning]

On Thu, Nov 5, 2009 at 8:23 AM, Kevin P. Fleming <kpfleming at digium.com>wrote:

>
> We need to see how you are originating the calls; it's up to the
> originator to specify the formats that will be allowed for that call. In
> spool files, for example, there is a header that can be included to
> specify which audio (and video) codecs should be offered on the outgoing
> channel.
>
>
Thanks Kevin, I was unaware of the Codecs header for the spool file.

However Asterisk still appears to be less than satisfied when asked to
initiate a call with 'siren14' as the *only* "codec".  (Obviously it isn't
yet a full codec for Asterisk and is only a supported format.  I suspect
that is the key to this observation)

As a clean test, I did the following on a fresh install of CentOS:

svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk
cd asterisk
./configure
make menuselect
make install
make samples

cp /usr/local/src/asterisk/contrib/init.d/rc.redhat.asterisk
/etc/init.d/asterisk

asterisk
-vvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvv | grep
siren
  == Registered file format siren7, extension(s) siren7
 format_siren7.so => (ITU G.722.1 (Siren7, licensed from Polycom))
  == Registered file format siren14, extension(s) siren14
 format_siren14.so => (ITU G.722.1 Annex C (Siren14, licensed from Polycom))

(first make sure basic spool call works)

vi /etc/asterisk/sip.conf
        disallow=all
        allow=ulaw

service asterisk restart

vi call.txt
    Channel: SIP/foo at bar.com
    CallerID: testcall
    Context: default
    Extension: demo
    Codecs: ulaw

 cp call.txt /var/spool/asterisk/outgoing/

!!!! Outgoing INVITE sent to the folks at bar.com !!!!

(now let's try just siren14)

vi /etc/asterisk/sip.conf
        disallow=all
        allow=siren14

service asterisk restart

vi call.txt
    Channel: SIP/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 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

So while inbound calls work fine with siren14 as the only allow=, Asterisk
won't initiate an outbound call with siren14 as the only choice.

Tom
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091110/b8f508a4/attachment.htm