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Created February 6, 2013 09:38
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Дщв
-- Executing [continue@macro-dialout-trunk:1] GotoIf("SIP/122-000002bc", "1?noreport") in new stack
-- Goto (macro-dialout-trunk,continue,3)
-- Executing [continue@macro-dialout-trunk:3] NoOp("SIP/122-000002bc", "TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 21 - failing through to other trunks") in new stack
-- Executing [continue@macro-dialout-trunk:4] Set("SIP/122-000002bc", "CALLERID(number)=122") in new stack
-- Executing [73432189230@from-internal:6] Macro("SIP/122-000002bc", "dialout-trunk,3,73432189230,") in new stack
-- Executing [s@macro-dialout-trunk:1] Set("SIP/122-000002bc", "DIAL_TRUNK=3") in new stack
-- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/122-000002bc", "0?sub-pincheck,s,1()") in new stack
-- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/122-000002bc", "0?disabletrunk,1") in new stack
-- Executing [s@macro-dialout-trunk:4] Set("SIP/122-000002bc", "DIAL_NUMBER=73432189230") in new stack
-- Executing [s@macro-dialout-trunk:5] Set("SIP/122-000002bc", "DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing [s@macro-dialout-trunk:6] Set("SIP/122-000002bc", "OUTBOUND_GROUP=OUT_3") in new stack
-- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/122-000002bc", "1?nomax") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/122-000002bc", "0?skipoutcid") in new stack
-- Executing [s@macro-dialout-trunk:10] Set("SIP/122-000002bc", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing [s@macro-dialout-trunk:11] Macro("SIP/122-000002bc", "outbound-callerid,3") in new stack
-- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/122-000002bc", "0?Set(CALLERPRES()=)") in new stack
-- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/122-000002bc", "0?Set(REALCALLERIDNUM=122)") in new stack
-- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/122-000002bc", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,6)
-- Executing [s@macro-outbound-callerid:6] Set("SIP/122-000002bc", "USEROUTCID=") in new stack
-- Executing [s@macro-outbound-callerid:7] Set("SIP/122-000002bc", "EMERGENCYCID=") in new stack
-- Executing [s@macro-outbound-callerid:8] Set("SIP/122-000002bc", "TRUNKOUTCID=") in new stack
-- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/122-000002bc", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,12)
-- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/122-000002bc", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/122-000002bc", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/122-000002bc", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/122-000002bc", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
-- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/122-000002bc", "0?sub-flp-3,s,1()") in new stack
-- Executing [s@macro-dialout-trunk:13] Set("SIP/122-000002bc", "OUTNUM=73432189230") in new stack
-- Executing [s@macro-dialout-trunk:14] Set("SIP/122-000002bc", "custom=SIP/mtt") in new stack
-- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/122-000002bc", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default))") in new stack
-- Executing [s@macro-dialout-trunk:16] ExecIf("SIP/122-000002bc", "0?Set(DIAL_TRUNK_OPTIONS=M(confirm))") in new stack
-- Executing [s@macro-dialout-trunk:17] Macro("SIP/122-000002bc", "dialout-trunk-predial-hook,") in new stack
-- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/122-000002bc", "") in new stack
-- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/122-000002bc", "0?bypass,1") in new stack
-- Executing [s@macro-dialout-trunk:19] ExecIf("SIP/122-000002bc", "1?Set(CONNECTEDLINE(num,i)=73432189230)") in new stack
-- Executing [s@macro-dialout-trunk:20] ExecIf("SIP/122-000002bc", "1?Set(CONNECTEDLINE(name,i)=CID:122)") in new stack
-- Executing [s@macro-dialout-trunk:21] GotoIf("SIP/122-000002bc", "0?customtrunk") in new stack
-- Executing [s@macro-dialout-trunk:22] Dial("SIP/122-000002bc", "SIP/mtt/73432189230,300,") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Audio is at 14068
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100008 (g729) to SDP
Adding codec 100009 (speex) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 80.75.130.136:5060:
INVITE sip:73432189230@80.75.130.136 SIP/2.0
Via: SIP/2.0/UDP 10.24.153.12:5060;branch=z9hG4bK7ade0676;rport
Max-Forwards: 70
From: "sale-2" <sip:73433619971@10.24.153.12>;tag=as09661d3d
To: <sip:73432189230@80.75.130.136>
Contact: <sip:73433619971@10.24.153.12:5060>
Call-ID: 1d029136752274ba29473f4874439192@10.24.153.12:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.10.1(11.1.0)
Date: Tue, 29 Jan 2013 20:05:56 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 333
v=0
o=root 707943639 707943639 IN IP4 10.24.153.12
s=Asterisk PBX 11.1.0
c=IN IP4 10.24.153.12
t=0 0
m=audio 14068 RTP/AVP 0 8 18 110 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
-- Called SIP/mtt/73432189230
<--- SIP read from UDP:80.75.130.136:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.24.153.12:5060;received=10.24.153.12;branch=z9hG4bK7ade0676;rport=5060
From: "sale-2" <sip:73433619971@10.24.153.12:5060>;tag=as09661d3d
To: <sip:73432189230@80.75.130.136>
Call-ID: 1d029136752274ba29473f4874439192@10.24.153.12:5060
CSeq: 102 INVITE
<------------->
--- (6 headers 0 lines) ---
<--- SIP read from UDP:80.75.130.136:5060 --->
SIP/2.0 183 Progress
Via: SIP/2.0/UDP 10.24.153.12:5060;received=10.24.153.12;branch=z9hG4bK7ade0676;rport=5060
From: "sale-2" <sip:73433619971@10.24.153.12:5060>;tag=as09661d3d
To: <sip:73432189230@80.75.130.136>;tag=2899311355-3792780656-380390839-4943405
Call-ID: 1d029136752274ba29473f4874439192@10.24.153.12:5060
CSeq: 102 INVITE
Contact: <sip:73432189230@80.75.130.136:5060;transport=udp>
Content-Type: application/sdp
Server: TS-v4.5.1-06f
Content-Length: 243
v=0
o=- 1360142231 1360142231 IN IP4 80.75.130.136
s=-
c=IN IP4 80.75.130.136
t=0 0
m=audio 33872 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=silenceSupp:off - - - -
<------------->
--- (10 headers 12 lines) ---
list_route: hop: <sip:73432189230@80.75.130.136:5060;transport=udp>
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|g729|speex), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 80.75.130.136:33872
-- SIP/mtt-000002be is making progress passing it to SIP/122-000002bc
Audio is at 18594
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Transmitting (NAT) to 10.24.154.10:3604 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.24.154.10:3604;branch=z9hG4bK-d8754z-e1188c5f6827d164-1---d8754z-;received=10.24.154.10;rport=3604
From: "122"<sip:122@10.24.153.12>;tag=8802805a
To: "73432189230"<sip:73432189230@10.24.153.12>;tag=as08985e73
Call-ID: NjM0M2FhNjQ2NjUxNjlmOWRhZWIxOGI4ZWYzNDg5YzA.
CSeq: 2 INVITE
Server: FPBX-2.10.1(11.1.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:73432189230@10.24.153.12:5060>
Content-Type: application/sdp
Content-Length: 257
v=0
o=root 752474745 752474745 IN IP4 10.24.153.12
s=Asterisk PBX 11.1.0
c=IN IP4 10.24.153.12
t=0 0
m=audio 18594 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------>
<--- SIP read from UDP:80.75.130.136:5060 --->
SIP/2.0 480 Temporarily Unavailable
Via: SIP/2.0/UDP 10.24.153.12:5060;received=10.24.153.12;branch=z9hG4bK7ade0676;rport=5060
From: "sale-2" <sip:73433619971@10.24.153.12:5060>;tag=as09661d3d
To: <sip:73432189230@80.75.130.136>;tag=2899311355-3792780656-380390839-4943405
Call-ID: 1d029136752274ba29473f4874439192@10.24.153.12:5060
CSeq: 102 INVITE
Contact: <sip:73432189230@10.133.29.18:5061>
Server: TS-v4.5.1-06f
Reason: SIP;cause=480;text="Temporarily Unavailable"
Reason: Q.850;cause=31;text="Normal, unspecified"
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
-- Got SIP response 480 "Temporarily Unavailable" back from 80.75.130.136:5060
set_destination: Parsing <sip:73432189230@80.75.130.136:5060;transport=udp> for address/port to send to
set_destination: set destination to 80.75.130.136:5060
Transmitting (NAT) to 80.75.130.136:5060:
ACK sip:73432189230@80.75.130.136:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.24.153.12:5060;branch=z9hG4bK7ade0676;rport
Max-Forwards: 70
From: "sale-2" <sip:73433619971@10.24.153.12>;tag=as09661d3d
To: <sip:73432189230@80.75.130.136>;tag=2899311355-3792780656-380390839-4943405
Contact: <sip:73433619971@10.24.153.12:5060>
Call-ID: 1d029136752274ba29473f4874439192@10.24.153.12:5060
CSeq: 102 ACK
User-Agent: FPBX-2.10.1(11.1.0)
Content-Length: 0
---
-- SIP/mtt-000002be is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [s@macro-dialout-trunk:23] NoOp("SIP/122-000002bc", "Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 19") in new stack
-- Executing [s@macro-dialout-trunk:24] Goto("SIP/122-000002bc", "s-CONGESTION,1") in new stack
-- Goto (macro-dialout-trunk,s-CONGESTION,1)
-- Executing [s-CONGESTION@macro-dialout-trunk:1] Set("SIP/122-000002bc", "RC=19") in new stack
-- Executing [s-CONGESTION@macro-dialout-trunk:2] Goto("SIP/122-000002bc", "19,1") in new stack
-- Goto (macro-dialout-trunk,19,1)
-- Executing [19@macro-dialout-trunk:1] Goto("SIP/122-000002bc", "continue,1") in new stack
-- Goto (macro-dialout-trunk,continue,1)
-- Executing [continue@macro-dialout-trunk:1] GotoIf("SIP/122-000002bc", "1?noreport") in new stack
-- Goto (macro-dialout-trunk,continue,3)
-- Executing [continue@macro-dialout-trunk:3] NoOp("SIP/122-000002bc", "TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 19 - failing through to other trunks") in new stack
-- Executing [continue@macro-dialout-trunk:4] Set("SIP/122-000002bc", "CALLERID(number)=122") in new stack
-- Executing [73432189230@from-internal:7] Macro("SIP/122-000002bc", "outisbusy,") in new stack
-- Executing [s@macro-outisbusy:1] Progress("SIP/122-000002bc", "") in new stack
-- Executing [s@macro-outisbusy:2] GotoIf("SIP/122-000002bc", "0?emergency,1") in new stack
-- Executing [s@macro-outisbusy:3] GotoIf("SIP/122-000002bc", "0?intracompany,1") in new stack
-- Executing [s@macro-outisbusy:4] Playback("SIP/122-000002bc", "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack
-- <SIP/122-000002bc> Playing 'all-circuits-busy-now.ulaw' (language 'ru')
Really destroying SIP dialog '1d029136752274ba29473f4874439192@10.24.153.12:5060' Method: INVITE
-- <SIP/122-000002bc> Playing 'pls-try-call-later.ulaw' (language 'ru')
-- Executing [s@macro-outisbusy:5] Congestion("SIP/122-000002bc", "20") in new stack
<--- Reliably Transmitting (NAT) to 10.24.154.10:3604 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 10.24.154.10:3604;branch=z9hG4bK-d8754z-e1188c5f6827d164-1---d8754z-;received=10.24.154.10;rport=3604
From: "122"<sip:122@10.24.153.12>;tag=8802805a
To: "73432189230"<sip:73432189230@10.24.153.12>;tag=as08985e73
Call-ID: NjM0M2FhNjQ2NjUxNjlmOWRhZWIxOGI4ZWYzNDg5YzA.
CSeq: 2 INVITE
Server: FPBX-2.10.1(11.1.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-Asterisk-HangupCause: User alerting, no answer
X-Asterisk-HangupCauseCode: 19
Content-Length: 0
<------------>
== Spawn extension (macro-outisbusy, s, 5) exited non-zero on 'SIP/122-000002bc' in macro 'outisbusy'
== Spawn extension (from-internal, 73432189230, 7) exited non-zero on 'SIP/122-000002bc'
-- Executing [h@from-internal:1] Hangup("SIP/122-000002bc", "") in new stack
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/122-000002bc'
== MixMonitor close filestream (mixed)
<--- SIP read from UDP:10.24.154.10:3604 --->
ACK sip:73432189230@10.24.153.12 SIP/2.0
Via: SIP/2.0/UDP 10.24.154.10:3604;branch=z9hG4bK-d8754z-e1188c5f6827d164-1---d8754z-;rport
To: "73432189230"<sip:73432189230@10.24.153.12>;tag=as08985e73
From: "122"<sip:122@10.24.153.12>;tag=8802805a
Call-ID: NjM0M2FhNjQ2NjUxNjlmOWRhZWIxOGI4ZWYzNDg5YzA.
CSeq: 2 ACK
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog 'NjM0M2FhNjQ2NjUxNjlmOWRhZWIxOGI4ZWYzNDg5YzA.' Method: ACK
== End MixMonitor Recording SIP/122-000002bc
Really destroying SIP dialog '38d5adf940df0e87552123a875724268@multifon.ru' Method: INVITE
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