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Home » Kerio User Forums » Kerio Operator » KO is not choosing an unused sip-trunk (if the line is busy while making an outgoign call)
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Riddick

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So the situation. I have a GSM-gateway connected as a sip-trunk to KO. In Call Routing i created an ext and added the gsm-trunk and one more i had before (1-st is gsm). So the problem is if the gsm-trunk is busy KO doesn't switch to the next trunk to make a call and just gets sip response 400 "Busy here" and disconnects me. Here is the log of the call. Maybe you'll see smth special.

-- Called SIP/89374902425@i-GSM-Mega-outbound
[23/Aug/2013 18:24:26] {sip} asterisk[1163]: VERBOSE[1224]: chan_sip.c:25669 in handle_request_do: #012<--- SIP read from UDP:192.168.200.200:5062 --->#012SIP/2.0 400 Busy here#012Via: SIP/2.0/UDP 192.168.200.80:5060;received=192.168.200.80;branch=z9hG4bK02dee565#012Call-ID: 39f714830dd350b560f12c5d26c0d150@192.168.200.80:5060#012From: "Evgenij Klimov" <sip:9374778151@192.168.200.80>;tag=as66dd1e83#012To: <sip:89374902425<_at_>192.168.200.200:5061>;tag=z9hG4bK02dee565#012CSeq: 102 INVITE#012Content-Length: 0#012#012<------------->
[23/Aug/2013 18:24:26] {sip} asterisk[1163]: VERBOSE[1224]: chan_sip.c:25679 in handle_request_do: --- (7 headers 0 lines) ---
[23/Aug/2013 18:24:26] {sip} asterisk[1163]: VERBOSE[1224]: chan_sip.c:21511 in handle_response:     -- Got SIP response 400 "Busy here" back from 192.168.200.200:5061
[23/Aug/2013 18:24:26] {sip} asterisk[1163]: VERBOSE[1224]: chan_sip.c:4286 in send_request: Transmitting (no NAT) to 192.168.200.200:5061:#012ACK sip:89374902425@192.168.200.200:5061 SIP/2.0#015#012Via: SIP/2.0/UDP 192.168.200.80:5060;branch=z9hG4bK02dee565#015#012Max-Forwards: 70#015#012From: "ME" <sip:9374778151@192.168.200.80>;tag=as66dd1e83#015#012To: <sip:89374902425@192.168.200.200:5061>;tag=z9hG4bK02dee565#015#012Contact: <sip:9374778151@192.168.200.80:5060>#015#012Call-ID: 39f714830dd350b560f12c5d26c0d150<_at_>192.168.200.80:5060#015#012CSeq: 102 ACK#015#012User-Agent: Asterisk PBX#015#012Content-Length: 0#015#012#015#012#012---
[23/Aug/2013 18:24:26] {sip} asterisk[1163]: VERBOSE[6142]: app_dial.c:1296 in wait_for_answer:     -- SIP/i-GSM-Mega-outbound-00002ad7 is circuit-busy
[23/Aug/2013 18:24:26] {sip} asterisk[1163]: VERBOSE[6142]: app_dial.c:1114 in wait_for_answer:   == Everyone is busy/congested at this time (1:0/1/0)
[23/Aug/2013 18:24:26] {sip} asterisk[1163]: VERBOSE[6142]: pbx.c:4255 in pbx_extension_helper:     -- Executing [s@macro-dial-i-GSM-Mega:8] Goto("SIP/111-00002ad6", "dialEnd") in new stack
[23/Aug/2013 18:24:26] {sip} asterisk[1163]: VERBOSE[6142]: pbx.c:9871 in pbx_builtin_goto:     -- Goto (macro-dial-i-GSM-Mega,s,10)
[23/Aug/2013 18:24:26] {sip} asterisk[1163]: VERBOSE[6142]: pbx.c:4255 in pbx_extension_helper:     -- Executing [s@macro-dial-i-GSM-Mega:10] GotoIf("SIP/111-00002ad6", "1?hangup") in new stack
[23/Aug/2013 18:24:26] {sip} asterisk[1163]: VERBOSE[6142]: pbx.c:9871 in pbx_builtin_goto:     -- Goto (macro-dial-i-GSM-Mega,s,13)
[23/Aug/2013 18:24:26] {sip} asterisk[1163]: VERBOSE[6142]: pbx.c:4255 in pbx_extension_helper:     -- Executing [s@macro-dial-i-GSM-Mega:13] Hangup("SIP/111-00002ad6", "") in new stack
[23/Aug/2013 18:24:26] {sip} asterisk[1163]: VERBOSE[6142]: app_macro.c:430 in _macro_exec:   == Spawn extension (macro-dial-i-GSM-Mega, s, 13) exited non-zero on 'SIP/111-00002ad6' in macro 'dial-i-GSM-Mega'
[23/Aug/2013 18:24:26] {sip} asterisk[1163]: VERBOSE[6142]: pbx.c:5071 in __ast_pbx_run:   == Spawn extension (sip-locals, 89374902425, 17) exited non-zero on 'SIP/111-00002ad6'
[23/Aug/2013 18:24:26] {sip} asterisk[1163]: VERBOSE[6142]: chan_sip.c:3999 in sip_scheddestroy: Scheduling destruction of SIP dialog '4934067a-2e325b41<_at_>192.168.200.23' in 32000 ms (Method: INVITE)
[23/Aug/2013 18:24:26] {sip} asterisk[1163]: VERBOSE[6142]: chan_sip.c:4239 in send_response: #012<--- Reliably Transmitting (no NAT) to 192.168.200.23:5061 --->#012SIP/2.0 603 Declined#015#012Via: SIP/2.0/UDP 192.168.200.23:5061;branch=z9hG4bK-416b17f9;received=192.168.200.23#015#012From: 111 <sip:111@192.168.200.80>;tag=58b3526925f1c85o1#015#012To: <sip:89374902425@192.168.200.80>;tag=as57c62a84#015#012Call-ID: 4934067a-2e325b41<_at_>192.168.200.23#015#012CSeq: 102 INVITE#015#012Server: Asterisk PBX#015#012Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH#015#012Supported: replaces, timer#015#012Content-Length: 0#015#012#015#012#012<------------>
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silars

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Check to see how many outgoing calls you allow on that trunk. You will need to make sure it is set to 1 (or whatever is allowed).

You can find it under the Advanced tab -> Concurrent Calls.
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Riddick

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Thanks a lot, it realy helped!
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