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How to find out why a call is dropped in Asterisk when everything is OK in the logs?
Hello. There are two PBXs. One for routing, the second is connected to the city telephone network via dahdi.
The problem is, when dialing short numbers in the 1XX format, everything is ok. But in the 7XXX format, the call is dropped.
You tell me what the problem is, "go to the console, logs, and look." And the problem is that in the console and logs, everything is just ok.
This is a broken call.
== Using SIP RTP CoS mark 5
-- Executing [[email protected]:1] Dial("SIP/pbx-00001410", "DAHDI/g1/7773") in new stack
-- Called DAHDI/g1/7773
-- DAHDI/6-1 answered SIP/pbx-00001410
-- Executing [[email protected]:1] Congestion("SIP/pbx-00001410", "10") in new stack
== Spawn extension (fromgorod, h, 1) exited non-zero on 'SIP/pbx-00001410'
-- Hanging up on 'DAHDI/6-1'
-- Hungup 'DAHDI/6-1'
== Spawn extension (fromgorod, 7773, 1) exited non-zero on 'SIP/pbx-00001410'
== Using SIP RTP CoS mark 5
-- Executing [[email protected]:1] Dial("SIP/pbx-00001411", "DAHDI/g1/135") in new stack
-- Called DAHDI/g1/135
-- DAHDI/6-1 answered SIP/pbx-00001411
-- Executing [[email protected]:1] Congestion("SIP/pbx-00001411", "10") in new stack
== Spawn extension (fromgorod, h, 1) exited non-zero on 'SIP/pbx-00001411'
-- Hanging up on 'DAHDI/6-1'
-- Hungup 'DAHDI/6-1'
== Spawn extension (fromgorod, 135, 1) exited non-zero on 'SIP/pbx-00001411'
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