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
> '[EMAIL PROTECTED]' 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 <[EMAIL PROTECTED]>
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