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

[Tom Browning]

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)
[Nov 10 17:32:37] DEBUG[28977]: chan_sip.c:7546 sip_alloc: Allocating
new SIP dialog for 5034f492225f4eef5db2149b20ad5d08 at 10.1.1.148 -
INVITE (No RTP)
[Nov 10 17:32:37] DEBUG[28977]: rtp_engine.c:328 ast_rtp_instance_new:
Using engine 'asterisk' for RTP instance '0x969b960'
[Nov 10 17:32:37] DEBUG[28977]: res_rtp_asterisk.c:423 ast_rtp_new:
Allocated port 12038 for RTP instance '0x969b960'
[Nov 10 17:32:37] DEBUG[28977]: rtp_engine.c:337 ast_rtp_instance_new:
RTP instance '0x969b960' is setup and ready to go
[Nov 10 17:32:37] DEBUG[28977]: res_rtp_asterisk.c:2197
ast_rtp_prop_set: Setup RTCP on RTP instance '0x969b960'
[Nov 10 17:32:37] DEBUG[28977]: chan_sip.c:5251 do_setnat: Setting NAT
on RTP to Off
[Nov 10 17:32:37] DEBUG[28977]: acl.c:499 ast_ouraddrfor: Found IP
address for this socket
[Nov 10 17:32:37] DEBUG[28977]: chan_sip.c:3851 ast_sip_ouraddrfor:
Setting SIP_TRANSPORT_UDP with address 10.1.1.148:5060
[Nov 10 17:32:37] DEBUG[28977]: frame.c:1235 ast_codec_choose: Could
not find preferred codec - Going for the best codec
[Nov 10 17:32:37] DEBUG[28977]: chan_sip.c:6977 sip_new: *** Our
native formats are 0x40 (slin)
[Nov 10 17:32:37] DEBUG[28977]: chan_sip.c:6978 sip_new: *** Joint
capabilities are 0x40 (slin)
[Nov 10 17:32:37] DEBUG[28977]: chan_sip.c:6979 sip_new: *** Our
capabilities are 0x4000 (siren14)
[Nov 10 17:32:37] DEBUG[28977]: frame.c:1235 ast_codec_choose: Could
not find preferred codec - Going for the best codec
[Nov 10 17:32:37] DEBUG[28977]: chan_sip.c:6980 sip_new: ***
AST_CODEC_CHOOSE formats are 0x40 (slin)
[Nov 10 17:32:37] DEBUG[28977]: chan_sip.c:6982 sip_new: *** Our
preferred formats from the incoming channel are 0x40 (slin)
[Nov 10 17:32:37] DEBUG[28977]: chan_sip.c:7010 sip_new: This channel
will not be able to handle video.
[Nov 10 17:32:37] DEBUG[28977]: chan_sip.c:5721 sip_call: Outgoing Call for foo
[Nov 10 17:32:37] DEBUG[28977]: chan_sip.c:5932 update_call_counter:
Updating call counter for outgoing call
[Nov 10 17:32:37] WARNING[28977]: chan_sip.c:5735 sip_call: No audio
format found to offer. Cancelling call to foo


Note that there are no peer definitions used.  I'm only setting codec
preference in sip.conf and the spool file.