NO JOY SIP URI Dialing to Asterisk No Working

sstasterisk

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== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
> 0x7f659c02b9a0 -- Strict RTP learning after remote address set to: xxx.xxx.xxx.xxx:2890
-- Executing [[email protected]:1] Set("SIP/8378-00000001", "MyDomain=") in new stack
-- Executing [[email protected]:2] NoOp("SIP/8378-00000001", "SIPDOMAIN: demo.nerdvittles.com") in new stack
-- Executing [[email protected]:3] GotoIf("SIP/8378-00000001", "0?FoundDomain") in new stack
-- Executing [[email protected]:4] Set("SIP/8378-00000001", "MyDomain=xxx.xxx.xxx.xxx") in new stack
-- Executing [[email protected]:5] Set("SIP/8378-00000001", "SIPDOMAIN=xxx.xxx.xxx.xxx") in new stack
-- Executing [[email protected]:6] GotoIf("SIP/8378-00000001", "1?OutAllRoutes") in new stack
-- Goto (from-internal,weather,8)
-- Executing [[email protected]:8] Set("SIP/8378-00000001", "DSTRING=7") in new stack
-- Executing [[email protected]:9] ExecIf("SIP/8378-00000001", "1?Goto(outbound-allroutes,weather,1") in new stack
-- Goto (outbound-allroutes,weather,1)
[2019-08-18 19:46:45] WARNING[18194][C-00000002]: pbx.c:4418 __ast_pbx_run: Channel 'SIP/8378-00000001' sent to invalid extension but no invalid handler: context,exten,priority=outbound-allroutes,weather,1
 

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