[Asterisk-Users] Asterisk Jitters

[Zak]

Hi,

Every time I dial into my asterisk box i hear nothing but asterisk 
jittering.
The following is an example of what I get on the asterisk CLI

Thanks

*CLI> DEBUG[81926]: File chan_sip.c, Line 3826 (check_user): Setting NAT 
on RTP
to 0
DEBUG[81926]: File chan_sip.c, Line 4807 (handle_request): Check for res
DEBUG[81926]: File chan_sip.c, Line 952 (find_user): Call from user 
'xirak' is 1
 out of 0
DEBUG[81926]: File chan_sip.c, Line 3249 (build_route): build_route: 
Contact hop
: <sip:192.168.7.3>
    -- Executing VoiceMailMain2("SIP/xirak-259d", "") in new stack
DEBUG[294927]: File rtp.c, Line 1007 (ast_rtp_write): Ooh, format 
changed from U
NKN to ULAW
DEBUG[294927]: File channel.c, Line 947 (ast_settimeout): Scheduling 
timer at 16
0 sample intervals
    -- Playing 'vm-login'
DEBUG[81926]: File chan_sip.c, Line 540 (__sip_ack): Stopping 
retransmission on
'6E5D898E-492D-400B-A42B-8B25FE25F2EE at 192.168.7.3' of Response 1: Found
DEBUG[294927]: File channel.c, Line 947 (ast_settimeout): Scheduling 
timer at 0
sample intervals
DEBUG[294927]: File channel.c, Line 947 (ast_settimeout): Scheduling 
timer at 0
sample intervals
WARNING[294927]: File app_voicemail2.c, Line 2567 (vm_execmain): 
Couldn't read u
sername
  == Spawn extension (extensions, 1001, 1) exited non-zero on 
'SIP/xirak-259d'
DEBUG[294927]: File chan_sip.c, Line 980 (sip_hangup): find_user(xirak)