Everyone is busy/congested at this time in asterisk

Asterisk Asterisk WebRTC
1

Sometimes the call is getting connected but most of the time I am getting the error in the subject with the following dialplan:

Executing [6238900031@Internal:1] NoOp(“SIP/201004-00000138”, “New Outgoing call from 201004”) in new stack
– Executing [6238900031@Internal:2] Set(“SIP/201004-00000138”, “CALLERLEN=6”) in new stack
– Executing [6238900031@Internal:3] GotoIf(“SIP/201004-00000138”, “0?rotate”) in new stack
– Executing [6238900031@Internal:4] Set(“SIP/201004-00000138”, “CRMSTATE=NOT_INUSE”) in new stack
– Executing [6238900031@Internal:5] GotoIf(“SIP/201004-00000138”, “0?Kill,crmoffline,1”) in new stack
– Executing [6238900031@Internal:6] Set(“SIP/201004-00000138”, “GLOBAL(OUTGW)=2”) in new stack
== Setting global variable ‘OUTGW’ to ‘2’
– Executing [6238900031@Internal:7] NoOp(“SIP/201004-00000138”, “2”) in new stack
– Executing [6238900031@Internal:8] GotoIf(“SIP/201004-00000138”, “0?call”) in new stack
– Executing [6238900031@Internal:9] Set(“SIP/201004-00000138”, “GLOBAL(OUTGW)=1”) in new stack
== Setting global variable ‘OUTGW’ to ‘1’
– Executing [6238900031@Internal:10] NoOp(“SIP/201004-00000138”, “INVALID”) in new stack
– Executing [6238900031@Internal:11] Dial(“SIP/201004-00000138”, “SIP/6238900031,20,tT”) in new stack
== Everyone is busy/congested at this time (1:0/0/1)
– Executing [6238900031@Internal:12] Hangup(“SIP/201004-00000138”, “”) in new stack

This happens in WebRTC calls.

2

Then they’re probably not connected, or the Websocket has dropped, or some other thing. If you do “sip set debug on” and see no INVITE to them - then one of the first two.

3

The numbers, in parentheses, are important here. 1 in the final position means unavailable, rather than busy or congested, which would, I think, include invalid number, but might also include rejections for other reasons. The debug option will show why the remote system rejected the call.

A ten digit end point name, especially without any dialled digits, is an unusual, but not impossible,configuration. If this is to access an ITSP, you need to provide digits to dial.

chan_sip is not supported, and has been completely removed from Asterisk 21, and 22 (and onwards).

4

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