When I made calls, I got the following logs. Can anybody tell me the reason why the server responded me with 503?
– Executing [6031@default:1] Dial(“SIP/6029-007996b0”, “SIP/6031”) in new stack
Really destroying SIP dialog ‘214563606065597965c4803f55cbe3e9@127.0.0.1’ Method: INVITE
[Sep 30 08:03:10] WARNING[5388]: app_dial.c:1196 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
== Auto fallthrough, channel ‘SIP/6029-007996b0’ status is 'CHANUNAVAIL’
localhost*CLI>
<— Transmitting (no NAT) to 192.168.7.210:54914 —>
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.7.210:54914;branch=z9hG4bK3887614428;received=192.168.7.210
From: sip:6029@192.168.7.56;tag=2202616676
To: sip:6031@192.168.7.56;tag=as51933c43
Call-ID: 3155334003@192.168.7.210
CSeq: 458332550 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:6031@192.168.7.56
Content-Length: 0
X-Asterisk-HangupCause: No route to destination
X-Asterisk-HangupCauseCode: 3
Thanks,
Jing