[Asterisk-Users] Asterisk Jitters

Steven Critchfield critch at basesys.com
Wed Sep 3 09:17:15 MST 2003


Do you have a zap device for timing?

On Wed, 2003-09-03 at 17:48, Zak wrote:
> 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)
> 
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-- 
Steven Critchfield  <critch at basesys.com>




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