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CS1000E 7.5 to Asterisk 1.8.11.0 - SIP Trunk problem

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jmpp2011

IS-IT--Management
Jan 14, 2011
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we are connect a CS1000E 7.5 to Asterisk 1.8.11.0 using SIP Trunk and we are getting the following message:


-- Executing [8998;phone-context=cdp.udp@from-sip-external:1] NoOp("SIP/humano.local-00000036", "Received incoming SIP connection from unknown peer to 8998;phone-context=cdp.udp") in new stack
-- Executing [8998;phone-context=cdp.udp@from-sip-external:2] Set("SIP/humano.local-00000036", "DID=8998;phone-context=cdp.udp") in new stack
-- Executing [8998;phone-context=cdp.udp@from-sip-external:3] Goto("SIP/humano.local-00000036", "s,1") in new stack
-- Goto (from-sip-external,s,1)
-- Executing [s@from-sip-external:1] GotoIf("SIP/humano.local-00000036", "0?checklang:noanonymous") in new stack
-- Goto (from-sip-external,s,5)
-- Executing [s@from-sip-external:5] Set("SIP/humano.local-00000036", "TIMEOUT(absolute)=15") in new stack
Channel will hangup at 2012-10-18 14:47:51.544 AST.
-- Executing [s@from-sip-external:6] Answer("SIP/humano.local-00000036", "") in new stack
-- Executing [s@from-sip-external:7] Wait("SIP/humano.local-00000036", "2") in new stack
-- Executing [s@from-sip-external:8] Playback("SIP/humano.local-00000036", "ss-noservice") in new stack
-- <SIP/humano.local-00000036> Playing 'ss-noservice.gsm' (language 'en')
-- Executing [s@from-sip-external:9] PlayTones("SIP/humano.local-00000036", "congestion") in new stack
-- Executing [s@from-sip-external:10] Congestion("SIP/humano.local-00000036", "5") in new stack
== Spawn extension (from-sip-external, s, 10) exited non-zero on 'SIP/humano.local-00000036'
-- Executing [h@from-sip-external:1] Hangup("SIP/humano.local-00000036", "") in new stack
== Spawn extension (from-sip-external, h, 1) exited non-zero on 'SIP/humano.local-00000036'




some help ??
 
Can you post a SIP trace from the CS1000?

Note: I would remove the cdp.udp in the phone context.

Bryan J Miles
Implementation Engineer
Xeta Technologies Inc
 
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