How to Produce a SIP Message Log to Send to my SIP Trunk Provider?

madmac63
Posts: 6
Member Since:
2010-02-19

Team:

I posted this in the Help Forum a week ago, and go no resonses. Hopefully you can help.

I'm running:
TrixBox Rev 2.8.0.3 installed from the ISO with the OS/Asterisk/TrixBox all preinstalled
Kernel Version 2.6.18-164.11.1.el5xen (SMP)
Distro Name CentOS release 5.4 (Final)

On a Dell PowerEdge T510 w/ dual Quad-Core CPUs and 8 GB of RAM, as well as 2 Ethernet Interfaces.

I have a SIP Trunk from PaeTec running over 2 T1's, bonded together, which also carries Internet Access. They placed a managed router in our facility with 2 internal Ethernet Interface, one for the Internet Traffic, and one for the SIP traffic.
I can't make outbound calls on the SIP Trunk. I get circuit busy messages in the asterisk -vr output when I try and place calls, and I get the recording all circuits are busy. The asterisk -vr output is included below.

PaeTec's Techs want me to send them the SIP Log. I did a sip history enable, and that's working. But I can't figure out how to run the sip show history call-id, since I have no idea what the call-id is for my SIP trunk. Any idea how to find that out?
Immediately below is a sample of the SIP Log on their end that they sent me. This is what they want to see from me on my end. How do I produce it? Will sip show history call-id do the trick, once I figure out what the call-id is?

Thanks

A Sample SIP Message Log from PaeTec, my SIP Trunk Provider.

INVITE sip:3129941300@63.253.71.204:5060;transport=UDP SIP/2.0
Via:SIP/2.0/UDP 64.198.149.137;branch=z9hG4bK-BroadWorks.kscamo54h00si03-63.253.71.204V5060-0-1062047485-1918920785-1266991963640-
From:"PAETEC";tag=1918920785-1266991963640-
To:"WRIGHT INSTITUTE ."
Call-ID:BW061243640240210-1797847308@64.198.149.137
CSeq:1062047485 INVITE
Contact:
Allow:ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Accept:multipart/mixed,application/media_control+xml,application/sdp
Supported:timer
Min-SE:60
Max-Forwards:48
Content-Type:application/sdp
Content-Length:340
v=0
o=BroadWorks 210977088 1 IN IP4 64.198.46.204
s=-
c=IN IP4 64.198.46.204
t=0 0
m=audio 24432 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=x-cxc:aT0wOGMzOGM3Zjk3OGU3YTQxLTE7Yj0xMC42MS4yNTMuMjE4O2Q9MTAuNjEuMjUzLjIxODoyMjI0O3Q9dmxhbjQwMDthPTY0LjE5OC40Ni4yMDQ6MjQ0MzI7
a=sendrecv

Asterisk -vr output when placing and outbound SIP Call and getting the Circuit Busy and Congestion Messages:

Connected to Asterisk 1.6.0.22-samy-r60 currently running on trixbox1 (pid = 3086)
Verbosity is at least 3
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP TOS bits 136
== Using SIP VRTP CoS mark 6
-- Executing [917734862577@from-internal:1] Macro("SIP/201-00000002", "user-callerid,SKIPTTL,") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/201-00000002", "AMPUSER=201") in new stack
-- Executing [s@macro-user-callerid:2] GotoIf("SIP/201-00000002", "0?report") in new stack
-- Executing [s@macro-user-callerid:3] ExecIf("SIP/201-00000002", "1?Set(REALCALLERIDNUM=201)") in new stack
-- Executing [s@macro-user-callerid:4] Set("SIP/201-00000002", "AMPUSER=201") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/201-00000002", "AMPUSERCIDNAME=Doug MacFarlane") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/201-00000002", "0?report") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/201-00000002", "AMPUSERCID=201") in new stack
-- Executing [s@macro-user-callerid:8] Set("SIP/201-00000002", "CALLERID(all)="Doug MacFarlane" ") in new stack
-- Executing [s@macro-user-callerid:9] Set("SIP/201-00000002", "REALCALLERIDNUM=201") in new stack
-- Executing [s@macro-user-callerid:10] ExecIf("SIP/201-00000002", "0?Set(CHANNEL(language)=)") in new stack
-- Executing [s@macro-user-callerid:11] GotoIf("SIP/201-00000002", "1?continue") in new stack
-- Goto (macro-user-callerid,s,20)
-- Executing [s@macro-user-callerid:20] NoOp("SIP/201-00000002", "Using CallerID "Doug MacFarlane" ") in new stack
-- Executing [917734862577@from-internal:2] Set("SIP/201-00000002", "_NODEST=") in new stack
-- Executing [917734862577@from-internal:3] Macro("SIP/201-00000002", "record-enable,201,OUT,") in new stack
-- Executing [s@macro-record-enable:1] GotoIf("SIP/201-00000002", "1?check") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-record-enable:4] AGI("SIP/201-00000002", "recordingcheck,20100223-232259,1266988979.2") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck,20100223-232259,1266988979.2: Outbound recording not enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing [s@macro-record-enable:5] MacroExit("SIP/201-00000002", "") in new stack
-- Executing [917734862577@from-internal:4] Macro("SIP/201-00000002", "dialout-trunk,2,17734862577,,") in new stack
-- Executing [s@macro-dialout-trunk:1] Set("SIP/201-00000002", "DIAL_TRUNK=2") in new stack
-- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/201-00000002", "0?sub-pincheck,s,1") in new stack
-- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/201-00000002", "0?disabletrunk,1") in new stack
-- Executing [s@macro-dialout-trunk:4] Set("SIP/201-00000002", "DIAL_NUMBER=17734862577") in new stack
-- Executing [s@macro-dialout-trunk:5] Set("SIP/201-00000002", "DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing [s@macro-dialout-trunk:6] Set("SIP/201-00000002", "OUTBOUND_GROUP=OUT_2") in new stack
-- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/201-00000002", "0?nomax") in new stack
-- Executing [s@macro-dialout-trunk:8] GotoIf("SIP/201-00000002", "0?chanfull") in new stack
-- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/201-00000002", "0?skipoutcid") in new stack
-- Executing [s@macro-dialout-trunk:10] Set("SIP/201-00000002", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing [s@macro-dialout-trunk:11] Macro("SIP/201-00000002", "outbound-callerid,2") in new stack
-- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/201-00000002", "0?Set(CALLERPRES()=)") in new stack
-- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/201-00000002", "0?Set(REALCALLERIDNUM=201)") in new stack
-- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/201-00000002", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,6)
-- Executing [s@macro-outbound-callerid:6] Set("SIP/201-00000002", "USEROUTCID=3129941200") in new stack
-- Executing [s@macro-outbound-callerid:7] Set("SIP/201-00000002", "EMERGENCYCID=") in new stack
-- Executing [s@macro-outbound-callerid:8] Set("SIP/201-00000002", "TRUNKOUTCID=3129941200") in new stack
-- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/201-00000002", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,12)
-- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/201-00000002", "1?Set(CALLERID(all)=3129941200)") in new stack
-- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/201-00000002", "1?Set(CALLERID(all)=3129941200)") in new stack
-- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/201-00000002", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
-- Executing [s@macro-dialout-trunk:12] ExecIf("SIP/201-00000002", "1?AGI(fixlocalprefix)") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
-- AGI Script fixlocalprefix completed, returning 0
-- Executing [s@macro-dialout-trunk:13] Set("SIP/201-00000002", "OUTNUM=17734862577") in new stack
-- Executing [s@macro-dialout-trunk:14] Set("SIP/201-00000002", "custom=SIP/PaeTec SIP") in new stack
-- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/201-00000002", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^))") in new stack
-- Executing [s@macro-dialout-trunk:16] Macro("SIP/201-00000002", "dialout-trunk-predial-hook,") in new stack
-- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/201-00000002", "") in new stack
-- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/201-00000002", "0?bypass,1") in new stack
-- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/201-00000002", "0?customtrunk") in new stack
-- Executing [s@macro-dialout-trunk:19] Dial("SIP/201-00000002", "SIP/PaeTec SIP/17734862577,300,") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP TOS bits 136
== Using SIP VRTP CoS mark 6
-- Called PaeTec SIP/17734862577
-- SIP/PaeTec SIP-00000003 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [s@macro-dialout-trunk:20] Goto("SIP/201-00000002", "s-CONGESTION,1") in new stack
-- Goto (macro-dialout-trunk,s-CONGESTION,1)
-- Executing [s-CONGESTION@macro-dialout-trunk:1] GotoIf("SIP/201-00000002", "1?noreport") in new stack
-- Goto (macro-dialout-trunk,s-CONGESTION,3)
-- Executing [s-CONGESTION@macro-dialout-trunk:3] NoOp("SIP/201-00000002", "TRUNK Dial failed due to CONGESTION - failing through to other trunks") in new stack
-- Executing [917734862577@from-internal:5] Macro("SIP/201-00000002", "outisbusy,") in new stack
-- Executing [s@macro-outisbusy:1] Playback("SIP/201-00000002", "all-circuits-busy-now,noanswer") in new stack
-- Playing 'all-circuits-busy-now.ulaw' (language 'en')
== Spawn extension (macro-outisbusy, s, 1) exited non-zero on 'SIP/201-00000002' in macro 'outisbusy'
== Spawn extension (from-internal, 917734862577, 5) exited non-zero on 'SIP/201-00000002'
-- Executing [h@from-internal:1] Macro("SIP/201-00000002", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] ResetCDR("SIP/201-00000002", "vw") in new stack
-- Executing [s@macro-hangupcall:2] NoCDR("SIP/201-00000002", "") in new stack
-- Executing [s@macro-hangupcall:3] GotoIf("SIP/201-00000002", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [s@macro-hangupcall:6] GotoIf("SIP/201-00000002", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] GotoIf("SIP/201-00000002", "1?theend") in new stack
-- Goto (macro-hangupcall,s,11)
-- Executing [s@macro-hangupcall:11] Hangup("SIP/201-00000002", "") in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/201-00000002' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/201-00000002'



SkykingOH
Posts: 9541
Member Since:
2007-12-17
simple - 1 - Turn off

simple -

1 - Turn off verbosity 'core set verbose 0'
2 - Turn on sip debug 'sip set debug'

THe log is in /var/log/asterisk/full

DOn't forget to turn the sip debugging off when you are done.

--

Scott

aka "Skyking"



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