First things first; I am a hobbyist in this area so, while persistent, I am not a professional. I fully admit that I might say dumb things; please forgive me.
This is my second set up of Asterisk/FreePBX to run my home office. A grand total of one phone with 2 lines (I even readily admit that this is overkill and dumb, forgive me; some people do crossword puzzles, I tinker with tech).
I worked on the first one for days on end and, it suddenly started working. I changed so much, that i have no idea how I got it to work. I followed the advice of dozens of people on this and other boards and something worked. Unfortunately, my router hardware failed and I took the opportunity to rebuild it. It is running pfSense router/firewall software. Everything except Asterisk was working fine. Router replaced; Asterisk failed; implies router is the problem… I get that.
I had every problem that you could have, no dial through, no sound, etc. I played with that for a while and then decided to start from scratch again. Pulled down the Incredible PBX 11 virtual machine for virtual box. Set it up, defined my trunks, extensions, etc and in a few minutes was able to dial out without problems. Asterisk replaced; Almost all problems go away; implies asterisk is the problem… I get that too. Obviously these two work in close concert.
Dialing in is where my issue is. When I dial in from an outside number, the call connects, I have full two way sound, but the call drops after roughly 6 seconds. I went into asterisk and did a SIP trace and it is clear (at least as best I can decipher) that the problem occurs when the dialed number side of the connection does not reply to the SIP 200 OK message. The message is retransmitted 6 times (matches my config setting) and then drops the call.
From reading numerous posts, I am a firm believer that all bad things come from NAT. So I would not be surprised if this is the same. However, reading the log file, it seems odd that only this one SIP message is being ignored. The 100 Trying and 180 Ringing messages seem to be flowing as expected. I can provide the pfSense setup information if that would be helpful, however, I don’t see any blocked traffic to/from my asterisk server in the logs.
In general, I have port forwarding set for UDP 5060 and UDP 10000-20000 to the asterisk server with asterisk settings in lock step. Again, I would think that if those were wrong, I would see it in one-way or no audio. My Asterisk sip settings are for NAT=yes, Dynamic IP (have tried static), the local network is fine and I am using only ulaw codec.
Below is a copy of my trace file. I have masked the phone numbers and my internet facing IP address. I started this just as I dialed and stopped it when the call dropped.
Any assistance is greatly appreciated.
pbx*CLI> sip set debug on
SIP Debugging enabled
<— SIP read from UDP:216.115.69.144:5060 —>
INVITE sip:<inbound#>@:5060 SIP/2.0
Record-Route: sip:216.115.69.144;lr
Record-Route: sip:216.115.69.132;lr
To: <sip:+<inbound#>@flowroute.com>
From: "<outbound#> " <sip:+<outbound#>@flowroute.com>;tag=gK0d0eb901
Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bK6e65.ac5315e7aca572674bbfa33e56b6e8c2.0
Via: SIP/2.0/UDP 216.115.69.131;branch=z9hG4bK6e65.b5a06a00cf08337f09b2be770364648f.0
Via: SIP/2.0/UDP 216.115.69.132;branch=z9hG4bK6e65.699a9a83fb2f8ad5001f2c4fdd8f7ebf.0
Via: SIP/2.0/UDP 4.55.17.35:5060;branch=z9hG4bK0dBb118609f52c22324
Call-ID: [email protected]
CSeq: 25438 INVITE
Max-Forwards: 61
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
Contact: “<outbound#>” <sip:+<outbound#>@4.55.17.35:5060>
Content-Length: 213
Content-Type: application/sdp
P-Asserted-Identity: "<outbound#> " <sip:+<outbound#>@flowroute.com>
v=0
o=- 31654 30110 IN IP4 4.55.17.2
s=-
c=IN IP4 4.55.17.2
t=0 0
m=audio 6472 RTP/AVP 0 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=maxptime:20
<------------->
— (17 headers 11 lines) —
Sending to 216.115.69.144:5060 (NAT)
Using INVITE request as basis request - [email protected]
Found peer ‘flowrouteLV’ for ‘+<outbound#>’ from 216.115.69.144:5060
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 101
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw), peer - audio=(ulaw|g729)/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 4.55.17.2:6472
Looking for <inbound#> in from-trunk (domain )
list_route: hop: sip:216.115.69.144;lr
list_route: hop: sip:216.115.69.132;lr
<— Transmitting (NAT) to 216.115.69.144:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bK6e65.ac5315e7aca572674bbfa33e56b6e8c2.0;received=216.115.69.144;rport=5060
Via: SIP/2.0/UDP 216.115.69.131;branch=z9hG4bK6e65.b5a06a00cf08337f09b2be770364648f.0
Via: SIP/2.0/UDP 216.115.69.132;branch=z9hG4bK6e65.699a9a83fb2f8ad5001f2c4fdd8f7ebf.0
Via: SIP/2.0/UDP 4.55.17.35:5060;branch=z9hG4bK0dBb118609f52c22324
Record-Route: sip:216.115.69.144;lr
Record-Route: sip:216.115.69.132;lr
From: "<outbound#> " <sip:+<outbound#>@flowroute.com>;tag=gK0d0eb901
To: <sip:+<inbound#>@flowroute.com>
Call-ID: [email protected]
CSeq: 25438 INVITE
Server: FPBX-2.11.0beta2(11.1.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:<inbound#>@:5060>
Content-Length: 0
<------------>
– Executing [<inbound#>@from-trunk:1] Set(“SIP/flowrouteLV-0000000a”, “__FROM_DID=<inbound#>”) in new stack
– Executing [<inbound#>@from-trunk:2] Set(“SIP/flowrouteLV-0000000a”, “CIDSFSCHEME=QUxMfEFMTA==”) in new stack
– Executing [<inbound#>@from-trunk:3] AGI(“SIP/flowrouteLV-0000000a”, “/var/www/html/admin/modules/superfecta/agi/superfecta.agi”) in new stack
– Launched AGI Script /var/www/html/admin/modules/superfecta/agi/superfecta.agi
/var/www/html/admin/modules/superfecta/agi/superfecta.agi: CID Superfecta is Answering the Channel
/var/www/html/admin/modules/superfecta/agi/superfecta.agi: CID Superfecta: Scheme is ALL
/var/www/html/admin/modules/superfecta/agi/superfecta.agi: CID Superfecta: The DID passed from Asterisk is: <inbound#>
/var/www/html/admin/modules/superfecta/agi/superfecta.agi: CID Superfecta: The number passed from Asterisk is: <inbound#>
/var/www/html/admin/modules/superfecta/agi/superfecta.agi: CID Superfecta: The CID name passed from Asterisk is: <outbound#>
/var/www/html/admin/modules/superfecta/agi/superfecta.agi: CID Superfecta: Executing Scheme…
– <SIP/flowrouteLV-0000000a>AGI Script /var/www/html/admin/modules/superfecta/agi/superfecta.agi completed, returning 0
– Executing [<inbound#>@from-trunk:4] Set(“SIP/flowrouteLV-0000000a”, “CALLERID(name)=”) in new stack
– Executing [<inbound#>@from-trunk:5] Gosub(“SIP/flowrouteLV-0000000a”, “app-blacklist-check,s,1()”) in new stack
– Executing [s@app-blacklist-check:1] GotoIf(“SIP/flowrouteLV-0000000a”, “0?blacklisted”) in new stack
– Executing [s@app-blacklist-check:2] Set(“SIP/flowrouteLV-0000000a”, “CALLED_BLACKLIST=1”) in new stack
– Executing [s@app-blacklist-check:3] Return(“SIP/flowrouteLV-0000000a”, “”) in new stack
– Executing [<inbound#>@from-trunk:6] Set(“SIP/flowrouteLV-0000000a”, “CDR(did)=<inbound#>”) in new stack
– Executing [<inbound#>@from-trunk:7] ExecIf(“SIP/flowrouteLV-0000000a”, “1 ?Set(CALLERID(name)=+<outbound#>)”) in new stack
– Executing [<inbound#>@from-trunk:8] Set(“SIP/flowrouteLV-0000000a”, “__CALLINGPRES_SV=allowed_not_screened”) in new stack
– Executing [<inbound#>@from-trunk:9] Set(“SIP/flowrouteLV-0000000a”, “CALLERPRES()=allowed_not_screened”) in new stack
– Executing [<inbound#>@from-trunk:10] Goto(“SIP/flowrouteLV-0000000a”, “from-did-direct,105,1”) in new stack
– Goto (from-did-direct,105,1)
– Executing [105@from-did-direct:1] Set(“SIP/flowrouteLV-0000000a”, “__RINGTIMER=15”) in new stack
– Executing [105@from-did-direct:2] Macro(“SIP/flowrouteLV-0000000a”, “exten-vm,novm,105,0,0,0”) in new stack
– Executing [s@macro-exten-vm:1] Macro(“SIP/flowrouteLV-0000000a”, “user-callerid,”) in new stack
– Executing [s@macro-user-callerid:1] Set(“SIP/flowrouteLV-0000000a”, “AMPUSER=+<outbound#>”) in new stack
– Executing [s@macro-user-callerid:2] GotoIf(“SIP/flowrouteLV-0000000a”, “0?report”) in new stack
– Executing [s@macro-user-callerid:3] ExecIf(“SIP/flowrouteLV-0000000a”, “1?Set(REALCALLERIDNUM=+<outbound#>)”) in new stack
– Executing [s@macro-user-callerid:4] Set(“SIP/flowrouteLV-0000000a”, “AMPUSER=”) in new stack
– Executing [s@macro-user-callerid:5] Set(“SIP/flowrouteLV-0000000a”, “AMPUSERCIDNAME=”) in new stack
– Executing [s@macro-user-callerid:6] GotoIf(“SIP/flowrouteLV-0000000a”, “1?report”) in new stack
– Goto (macro-user-callerid,s,13)
– Executing [s@macro-user-callerid:13] GotoIf(“SIP/flowrouteLV-0000000a”, “0?continue”) in new stack
– Executing [s@macro-user-callerid:14] Set(“SIP/flowrouteLV-0000000a”, “__TTL=64”) in new stack
– Executing [s@macro-user-callerid:15] GotoIf(“SIP/flowrouteLV-0000000a”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,26)
– Executing [s@macro-user-callerid:26] Set(“SIP/flowrouteLV-0000000a”, “CALLERID(number)=+<outbound#>”) in new stack
– Executing [s@macro-user-callerid:27] Set(“SIP/flowrouteLV-0000000a”, “CALLERID(name)=+<outbound#>”) in new stack
– Executing [s@macro-user-callerid:28] Set(“SIP/flowrouteLV-0000000a”, “CDR(cnum)=+<outbound#>”) in new stack
– Executing [s@macro-user-callerid:29] Set(“SIP/flowrouteLV-0000000a”, “CDR(cnam)=+<outbound#>”) in new stack
– Executing [s@macro-user-callerid:30] Set(“SIP/flowrouteLV-0000000a”, “CHANNEL(language)=en”) in new stack
– Executing [s@macro-exten-vm:2] Set(“SIP/flowrouteLV-0000000a”, “RingGroupMethod=none”) in new stack
– Executing [s@macro-exten-vm:3] Set(“SIP/flowrouteLV-0000000a”, “__EXTTOCALL=105”) in new stack
– Executing [s@macro-exten-vm:4] Set(“SIP/flowrouteLV-0000000a”, “__PICKUPMARK=105”) in new stack
– Executing [s@macro-exten-vm:5] Set(“SIP/flowrouteLV-0000000a”, “RT=”) in new stack
– Executing [s@macro-exten-vm:6] Gosub(“SIP/flowrouteLV-0000000a”, “sub-record-check,s,1(exten,105,)”) in new stack
– Executing [s@sub-record-check:1] GotoIf(“SIP/flowrouteLV-0000000a”, “1?check”) in new stack
– Goto (sub-record-check,s,6)
– Executing [s@sub-record-check:6] Set(“SIP/flowrouteLV-0000000a”, “__MON_FMT=wav”) in new stack
– Executing [s@sub-record-check:7] GotoIf(“SIP/flowrouteLV-0000000a”, “1?next”) in new stack
– Goto (sub-record-check,s,10)
– Executing [s@sub-record-check:10] ExecIf(“SIP/flowrouteLV-0000000a”, “0?Return()”) in new stack
– Executing [s@sub-record-check:11] GotoIf(“SIP/flowrouteLV-0000000a”, “0?exten,1”) in new stack
– Executing [s@sub-record-check:12] Set(“SIP/flowrouteLV-0000000a”, “__REC_STATUS=INITIALIZED”) in new stack
– Executing [s@sub-record-check:13] ExecIf(“SIP/flowrouteLV-0000000a”, “0?Set(__REC_POLICY_MODE=)”) in new stack
– Executing [s@sub-record-check:14] Set(“SIP/flowrouteLV-0000000a”, “NOW=1361123813”) in new stack
– Executing [s@sub-record-check:15] Set(“SIP/flowrouteLV-0000000a”, “__DAY=17”) in new stack
– Executing [s@sub-record-check:16] Set(“SIP/flowrouteLV-0000000a”, “__MONTH=02”) in new stack
– Executing [s@sub-record-check:17] Set(“SIP/flowrouteLV-0000000a”, “__YEAR=2013”) in new stack
– Executing [s@sub-record-check:18] Set(“SIP/flowrouteLV-0000000a”, “__TIMESTR=20130217-115653”) in new stack
– Executing [s@sub-record-check:19] Set(“SIP/flowrouteLV-0000000a”, “__FROMEXTEN=+<outbound#>”) in new stack
– Executing [s@sub-record-check:20] Set(“SIP/flowrouteLV-0000000a”, “__CALLFILENAME=exten-105-+<outbound#>-20130217-115653-1361123812.10”) in new stack
– Executing [s@sub-record-check:21] Goto(“SIP/flowrouteLV-0000000a”, “exten,1”) in new stack
– Goto (sub-record-check,exten,1)
– Executing [exten@sub-record-check:1] GotoIf(“SIP/flowrouteLV-0000000a”, “0?callee”) in new stack
– Executing [exten@sub-record-check:2] Set(“SIP/flowrouteLV-0000000a”, “__REC_POLICY_MODE=dontcare”) in new stack
– Executing [exten@sub-record-check:3] GotoIf(“SIP/flowrouteLV-0000000a”, “1?caller”) in new stack
– Goto (sub-record-check,exten,10)
– Executing [exten@sub-record-check:10] Set(“SIP/flowrouteLV-0000000a”, “__REC_POLICY_MODE=”) in new stack
– Executing [exten@sub-record-check:11] GosubIf(“SIP/flowrouteLV-0000000a”, “0?record,1(exten,105,+<outbound#>)”) in new stack
– Executing [exten@sub-record-check:12] Return(“SIP/flowrouteLV-0000000a”, “”) in new stack
– Executing [s@macro-exten-vm:7] GotoIf(“SIP/flowrouteLV-0000000a”, “1?macrodial”) in new stack
– Goto (macro-exten-vm,s,13)
– Executing [s@macro-exten-vm:13] GosubIf(“SIP/flowrouteLV-0000000a”, “0?clrheader,1()”) in new stack
– Executing [s@macro-exten-vm:14] Macro(“SIP/flowrouteLV-0000000a”, “dial-one,Ttr,105”) in new stack
– Executing [s@macro-dial-one:1] Set(“SIP/flowrouteLV-0000000a”, “DEXTEN=105”) in new stack
– Executing [s@macro-dial-one:2] Set(“SIP/flowrouteLV-0000000a”, “DIALSTATUS_CW=”) in new stack
– Executing [s@macro-dial-one:3] GosubIf(“SIP/flowrouteLV-0000000a”, “0?screen,1()”) in new stack
– Executing [s@macro-dial-one:4] GosubIf(“SIP/flowrouteLV-0000000a”, “0?cf,1()”) in new stack
– Executing [s@macro-dial-one:5] GotoIf(“SIP/flowrouteLV-0000000a”, “1?skip1”) in new stack
– Goto (macro-dial-one,s,8)
– Executing [s@macro-dial-one:8] GotoIf(“SIP/flowrouteLV-0000000a”, “0?nodial”) in new stack
– Executing [s@macro-dial-one:9] GotoIf(“SIP/flowrouteLV-0000000a”, “0?continue”) in new stack
– Executing [s@macro-dial-one:10] Set(“SIP/flowrouteLV-0000000a”, “EXTHASCW=ENABLED”) in new stack
– Executing [s@macro-dial-one:11] GotoIf(“SIP/flowrouteLV-0000000a”, “0?next1:cwinusebusy”) in new stack
– Goto (macro-dial-one,s,23)
– Executing [s@macro-dial-one:23] GotoIf(“SIP/flowrouteLV-0000000a”, “1?next3:continue”) in new stack
– Goto (macro-dial-one,s,24)
– Executing [s@macro-dial-one:24] ExecIf(“SIP/flowrouteLV-0000000a”, “0?Set(DIALSTATUS_CW=BUSY)”) in new stack
– Executing [s@macro-dial-one:25] GotoIf(“SIP/flowrouteLV-0000000a”, “0?nodial”) in new stack
– Executing [s@macro-dial-one:26] GosubIf(“SIP/flowrouteLV-0000000a”, “1?dstring,1():dlocal,1()”) in new stack
– Executing [dstring@macro-dial-one:1] Set(“SIP/flowrouteLV-0000000a”, “DSTRING=”) in new stack
– Executing [dstring@macro-dial-one:2] Set(“SIP/flowrouteLV-0000000a”, “DEVICES=105”) in new stack
– Executing [dstring@macro-dial-one:3] ExecIf(“SIP/flowrouteLV-0000000a”, “0?Return()”) in new stack
– Executing [dstring@macro-dial-one:4] ExecIf(“SIP/flowrouteLV-0000000a”, “0?Set(DEVICES=05)”) in new stack
– Executing [dstring@macro-dial-one:5] Set(“SIP/flowrouteLV-0000000a”, “LOOPCNT=1”) in new stack
– Executing [dstring@macro-dial-one:6] Set(“SIP/flowrouteLV-0000000a”, “ITER=1”) in new stack
– Executing [dstring@macro-dial-one:7] Set(“SIP/flowrouteLV-0000000a”, “THISDIAL=SIP/105”) in new stack
– Executing [dstring@macro-dial-one:8] GosubIf(“SIP/flowrouteLV-0000000a”, “1?zap2dahdi,1()”) in new stack
– Executing [zap2dahdi@macro-dial-one:1] ExecIf(“SIP/flowrouteLV-0000000a”, “0?Return()”) in new stack
– Executing [zap2dahdi@macro-dial-one:2] Set(“SIP/flowrouteLV-0000000a”, “NEWDIAL=”) in new stack
– Executing [zap2dahdi@macro-dial-one:3] Set(“SIP/flowrouteLV-0000000a”, “LOOPCNT2=1”) in new stack
– Executing [zap2dahdi@macro-dial-one:4] Set(“SIP/flowrouteLV-0000000a”, “ITER2=1”) in new stack
– Executing [zap2dahdi@macro-dial-one:5] Set(“SIP/flowrouteLV-0000000a”, “THISPART2=SIP/105”) in new stack
– Executing [zap2dahdi@macro-dial-one:6] ExecIf(“SIP/flowrouteLV-0000000a”, “0?Set(THISPART2=DAHDI/105)”) in new stack
– Executing [zap2dahdi@macro-dial-one:7] Set(“SIP/flowrouteLV-0000000a”, “NEWDIAL=SIP/105&”) in new stack
– Executing [zap2dahdi@macro-dial-one:8] Set(“SIP/flowrouteLV-0000000a”, “ITER2=2”) in new stack
– Executing [zap2dahdi@macro-dial-one:9] GotoIf(“SIP/flowrouteLV-0000000a”, “0?begin2”) in new stack
– Executing [zap2dahdi@macro-dial-one:10] Set(“SIP/flowrouteLV-0000000a”, “THISDIAL=SIP/105”) in new stack
– Executing [zap2dahdi@macro-dial-one:11] Return(“SIP/flowrouteLV-0000000a”, “”) in new stack
– Executing [dstring@macro-dial-one:9] Set(“SIP/flowrouteLV-0000000a”, “DSTRING=SIP/105&”) in new stack
– Executing [dstring@macro-dial-one:10] Set(“SIP/flowrouteLV-0000000a”, “ITER=2”) in new stack
– Executing [dstring@macro-dial-one:11] GotoIf(“SIP/flowrouteLV-0000000a”, “0?begin”) in new stack
– Executing [dstring@macro-dial-one:12] Set(“SIP/flowrouteLV-0000000a”, “DSTRING=SIP/105”) in new stack
– Executing [dstring@macro-dial-one:13] Return(“SIP/flowrouteLV-0000000a”, “”) in new stack
– Executing [s@macro-dial-one:27] GotoIf(“SIP/flowrouteLV-0000000a”, “0?nodial”) in new stack
– Executing [s@macro-dial-one:28] GotoIf(“SIP/flowrouteLV-0000000a”, “0?skiptrace”) in new stack
– Executing [s@macro-dial-one:29] GosubIf(“SIP/flowrouteLV-0000000a”, “1?ctset,1():ctclear,1()”) in new stack
– Executing [ctset@macro-dial-one:1] Set(“SIP/flowrouteLV-0000000a”, “DB(CALLTRACE/105)=+<outbound#>”) in new stack
– Executing [ctset@macro-dial-one:2] Return(“SIP/flowrouteLV-0000000a”, “”) in new stack
– Executing [s@macro-dial-one:30] Set(“SIP/flowrouteLV-0000000a”, “D_OPTIONS=Ttr”) in new stack
– Executing [s@macro-dial-one:31] ExecIf(“SIP/flowrouteLV-0000000a”, “0?SIPAddHeader(Alert-Info: )”) in new stack
– Executing [s@macro-dial-one:32] ExecIf(“SIP/flowrouteLV-0000000a”, “0?SIPAddHeader()”) in new stack
– Executing [s@macro-dial-one:33] ExecIf(“SIP/flowrouteLV-0000000a”, “0?Set(CHANNEL(musicclass)=)”) in new stack
– Executing [s@macro-dial-one:34] GosubIf(“SIP/flowrouteLV-0000000a”, “0?qwait,1()”) in new stack
– Executing [s@macro-dial-one:35] Set(“SIP/flowrouteLV-0000000a”, “__CWIGNORE=”) in new stack
– Executing [s@macro-dial-one:36] Set(“SIP/flowrouteLV-0000000a”, “__KEEPCID=TRUE”) in new stack
– Executing [s@macro-dial-one:37] GotoIf(“SIP/flowrouteLV-0000000a”, “0?usegoto,1”) in new stack
– Executing [s@macro-dial-one:38] GotoIf(“SIP/flowrouteLV-0000000a”, “1?godial”) in new stack
– Goto (macro-dial-one,s,42)
– Executing [s@macro-dial-one:42] Dial(“SIP/flowrouteLV-0000000a”, “SIP/105,Ttr”) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Audio is at 17838
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.0.1.13:5060:
INVITE sip:[email protected]:5060;user=phone;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.0.1.12:5060;branch=z9hG4bK11916800
Max-Forwards: 70
From: “+<outbound#>” <sip:+<outbound#>@10.0.1.12>;tag=as4fd96afc
To: sip:[email protected]:5060;user=phone;transport=udp
Contact: <sip:+<outbound#>@10.0.1.12:5060>
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0beta2(11.1.2)
Date: Sun, 17 Feb 2013 17:56:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 447
v=0
o=root 1907814230 1907814230 IN IP4 10.0.1.12
s=Asterisk PBX 11.1.2
c=IN IP4 10.0.1.12
t=0 0
m=audio 17838 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=ice-ufrag:2f0e5fc111261d4870a40a467a87ea80
a=ice-pwd:0a0797ac2dffa953135838611f766193
a=candidate:Ha00010c 1 UDP 2130706431 10.0.1.12 17838 typ host
a=candidate:Ha00010c 2 UDP 2130706430 10.0.1.12 17839 typ host
a=sendrecv
-- Called SIP/105
<— Transmitting (NAT) to 216.115.69.144:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bK6e65.ac5315e7aca572674bbfa33e56b6e8c2.0;received=216.115.69.144;rport=5060
Via: SIP/2.0/UDP 216.115.69.131;branch=z9hG4bK6e65.b5a06a00cf08337f09b2be770364648f.0
Via: SIP/2.0/UDP 216.115.69.132;branch=z9hG4bK6e65.699a9a83fb2f8ad5001f2c4fdd8f7ebf.0
Via: SIP/2.0/UDP 4.55.17.35:5060;branch=z9hG4bK0dBb118609f52c22324
Record-Route: sip:216.115.69.144;lr
Record-Route: sip:216.115.69.132;lr
From: "<outbound#> " <sip:+<outbound#>@flowroute.com>;tag=gK0d0eb901
To: <sip:+<inbound#>@flowroute.com>;tag=as68ed7049
Call-ID: [email protected]
CSeq: 25438 INVITE
Server: FPBX-2.11.0beta2(11.1.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:<inbound#>@:5060>
Content-Length: 0
<------------>
Retransmitting #1 (no NAT) to 10.0.1.13:5060:
INVITE sip:[email protected]:5060;user=phone;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.0.1.12:5060;branch=z9hG4bK11916800
Max-Forwards: 70
From: “+<outbound#>” <sip:+<outbound#>@10.0.1.12>;tag=as4fd96afc
To: sip:[email protected]:5060;user=phone;transport=udp
Contact: <sip:+<outbound#>@10.0.1.12:5060>
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0beta2(11.1.2)
Date: Sun, 17 Feb 2013 17:56:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 447
v=0
o=root 1907814230 1907814230 IN IP4 10.0.1.12
s=Asterisk PBX 11.1.2
c=IN IP4 10.0.1.12
t=0 0
m=audio 17838 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=ice-ufrag:2f0e5fc111261d4870a40a467a87ea80
a=ice-pwd:0a0797ac2dffa953135838611f766193
a=candidate:Ha00010c 1 UDP 2130706431 10.0.1.12 17838 typ host
a=candidate:Ha00010c 2 UDP 2130706430 10.0.1.12 17839 typ host
a=sendrecv
<— SIP read from UDP:10.0.1.13:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.1.12:5060;branch=z9hG4bK11916800
From: “+<outbound#>” <sip:+<outbound#>@10.0.1.12>;tag=as4fd96afc
To: sip:[email protected]:5060;user=phone;transport=udp
Call-ID: [email protected]:5060
Date: Sun, 17 Feb 2013 17:56:52 GMT
CSeq: 102 INVITE
Server: Cisco-CP7960G/8.0
Contact: sip:[email protected]:5060;user=phone;transport=udp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Content-Length: 0
<------------->
— (11 headers 0 lines) —
<— SIP read from UDP:10.0.1.13:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.1.12:5060;branch=z9hG4bK11916800
From: “+<outbound#>” <sip:+<outbound#>@10.0.1.12>;tag=as4fd96afc
To: sip:[email protected]:5060;user=phone;transport=udp
Call-ID: [email protected]:5060
Date: Sun, 17 Feb 2013 17:56:52 GMT
CSeq: 102 INVITE
Server: Cisco-CP7960G/8.0
Contact: sip:[email protected]:5060;user=phone;transport=udp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Content-Length: 0
<------------->
— (11 headers 0 lines) —
<— SIP read from UDP:10.0.1.13:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.0.1.12:5060;branch=z9hG4bK11916800
From: “+<outbound#>” <sip:+<outbound#>@10.0.1.12>;tag=as4fd96afc
To: sip:[email protected]:5060;user=phone;transport=udp;tag=001b0cdaf88000a67578368a-5a2d6186
Call-ID: [email protected]:5060
Date: Sun, 17 Feb 2013 17:56:52 GMT
CSeq: 102 INVITE
Server: Cisco-CP7960G/8.0
Contact: sip:[email protected]:5060;user=phone;transport=udp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Content-Length: 0
<------------->
— (11 headers 0 lines) —
list_route: hop: sip:[email protected]:5060;user=phone;transport=udp
– SIP/105-0000000b is ringing
<— Transmitting (NAT) to 216.115.69.144:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bK6e65.ac5315e7aca572674bbfa33e56b6e8c2.0;received=216.115.69.144;rport=5060
Via: SIP/2.0/UDP 216.115.69.131;branch=z9hG4bK6e65.b5a06a00cf08337f09b2be770364648f.0
Via: SIP/2.0/UDP 216.115.69.132;branch=z9hG4bK6e65.699a9a83fb2f8ad5001f2c4fdd8f7ebf.0
Via: SIP/2.0/UDP 4.55.17.35:5060;branch=z9hG4bK0dBb118609f52c22324
Record-Route: sip:216.115.69.144;lr
Record-Route: sip:216.115.69.132;lr
From: "<outbound#> " <sip:+<outbound#>@flowroute.com>;tag=gK0d0eb901
To: <sip:+<inbound#>@flowroute.com>;tag=as68ed7049
Call-ID: [email protected]
CSeq: 25438 INVITE
Server: FPBX-2.11.0beta2(11.1.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:<inbound#>@:5060>
Content-Length: 0
<------------>
<— SIP read from UDP:10.0.1.13:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.1.12:5060;branch=z9hG4bK11916800
From: “+<outbound#>” <sip:+<outbound#>@10.0.1.12>;tag=as4fd96afc
To: sip:[email protected]:5060;user=phone;transport=udp;tag=001b0cdaf88000a67578368a-5a2d6186
Call-ID: [email protected]:5060
Date: Sun, 17 Feb 2013 17:56:53 GMT
CSeq: 102 INVITE
Server: Cisco-CP7960G/8.0
Contact: sip:[email protected]:5060;user=phone;transport=udp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Supported: replaces,join,norefersub
Content-Length: 199
Content-Type: application/sdp
Content-Disposition: session;handling=optional
v=0
o=Cisco-SIPUA 18287 0 IN IP4 10.0.1.13
s=SIP Call
t=0 0
m=audio 19146 RTP/AVP 0 101
c=IN IP4 10.0.1.13
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
— (14 headers 10 lines) —
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), 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 10.0.1.13:19146
list_route: hop: sip:[email protected]:5060;user=phone;transport=udp
set_destination: Parsing sip:[email protected]:5060;user=phone;transport=udp for address/port to send to
set_destination: set destination to 10.0.1.13:5060
Transmitting (no NAT) to 10.0.1.13:5060:
ACK sip:[email protected]:5060;user=phone;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.0.1.12:5060;branch=z9hG4bK106eea2b
Max-Forwards: 70
From: “+<outbound#>” <sip:+<outbound#>@10.0.1.12>;tag=as4fd96afc
To: sip:[email protected]:5060;user=phone;transport=udp;tag=001b0cdaf88000a67578368a-5a2d6186
Contact: <sip:+<outbound#>@10.0.1.12:5060>
Call-ID: [email protected]:5060
CSeq: 102 ACK
User-Agent: FPBX-2.11.0beta2(11.1.2)
Content-Length: 0
-- SIP/105-0000000b answered SIP/flowrouteLV-0000000a
Audio is at 10266
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<— Reliably Transmitting (NAT) to 216.115.69.144:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bK6e65.ac5315e7aca572674bbfa33e56b6e8c2.0;received=216.115.69.144;rport=5060
Via: SIP/2.0/UDP 216.115.69.131;branch=z9hG4bK6e65.b5a06a00cf08337f09b2be770364648f.0
Via: SIP/2.0/UDP 216.115.69.132;branch=z9hG4bK6e65.699a9a83fb2f8ad5001f2c4fdd8f7ebf.0
Via: SIP/2.0/UDP 4.55.17.35:5060;branch=z9hG4bK0dBb118609f52c22324
Record-Route: sip:216.115.69.144;lr
Record-Route: sip:216.115.69.132;lr
From: "<outbound#> " <sip:+<outbound#>@flowroute.com>;tag=gK0d0eb901
To: <sip:+<inbound#>@flowroute.com>;tag=as68ed7049
Call-ID: [email protected]
CSeq: 25438 INVITE
Server: FPBX-2.11.0beta2(11.1.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:<inbound#>@:5060>
Content-Type: application/sdp
Content-Length: 453
v=0
o=root 1023448070 1023448070 IN IP4
s=Asterisk PBX 11.1.2
c=IN IP4
t=0 0
m=audio 10266 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=ice-ufrag:01a27a320cbc5fd03eb48cec65059e07
a=ice-pwd:206f13662da5813d6d63b56a6d7def4b
a=candidate:Ha00010c 1 UDP 2130706431 10.0.1.12 10266 typ host
a=candidate:Ha00010c 2 UDP 2130706430 10.0.1.12 10267 typ host
a=sendrecv
<------------>
Retransmitting #1 (NAT) to 216.115.69.144:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bK6e65.ac5315e7aca572674bbfa33e56b6e8c2.0;received=216.115.69.144;rport=5060
Via: SIP/2.0/UDP 216.115.69.131;branch=z9hG4bK6e65.b5a06a00cf08337f09b2be770364648f.0
Via: SIP/2.0/UDP 216.115.69.132;branch=z9hG4bK6e65.699a9a83fb2f8ad5001f2c4fdd8f7ebf.0
Via: SIP/2.0/UDP 4.55.17.35:5060;branch=z9hG4bK0dBb118609f52c22324
Record-Route: sip:216.115.69.144;lr
Record-Route: sip:216.115.69.132;lr
From: "<outbound#> " <sip:+<outbound#>@flowroute.com>;tag=gK0d0eb901
To: <sip:+<inbound#>@flowroute.com>;tag=as68ed7049
Call-ID: [email protected]
CSeq: 25438 INVITE
Server: FPBX-2.11.0beta2(11.1.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:<inbound#>@:5060>
Content-Type: application/sdp
Content-Length: 453
v=0
o=root 1023448070 1023448070 IN IP4
s=Asterisk PBX 11.1.2
c=IN IP4
t=0 0
m=audio 10266 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=ice-ufrag:01a27a320cbc5fd03eb48cec65059e07
a=ice-pwd:206f13662da5813d6d63b56a6d7def4b
a=candidate:Ha00010c 1 UDP 2130706431 10.0.1.12 10266 typ host
a=candidate:Ha00010c 2 UDP 2130706430 10.0.1.12 10267 typ host
a=sendrecv
Retransmitting #2 (NAT) to 216.115.69.144:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bK6e65.ac5315e7aca572674bbfa33e56b6e8c2.0;received=216.115.69.144;rport=5060
Via: SIP/2.0/UDP 216.115.69.131;branch=z9hG4bK6e65.b5a06a00cf08337f09b2be770364648f.0
Via: SIP/2.0/UDP 216.115.69.132;branch=z9hG4bK6e65.699a9a83fb2f8ad5001f2c4fdd8f7ebf.0
Via: SIP/2.0/UDP 4.55.17.35:5060;branch=z9hG4bK0dBb118609f52c22324
Record-Route: sip:216.115.69.144;lr
Record-Route: sip:216.115.69.132;lr
From: "<outbound#> " <sip:+<outbound#>@flowroute.com>;tag=gK0d0eb901
To: <sip:+<inbound#>@flowroute.com>;tag=as68ed7049
Call-ID: [email protected]
CSeq: 25438 INVITE
Server: FPBX-2.11.0beta2(11.1.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:<inbound#>@:5060>
Content-Type: application/sdp
Content-Length: 453
v=0
o=root 1023448070 1023448070 IN IP4
s=Asterisk PBX 11.1.2
c=IN IP4
t=0 0
m=audio 10266 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=ice-ufrag:01a27a320cbc5fd03eb48cec65059e07
a=ice-pwd:206f13662da5813d6d63b56a6d7def4b
a=candidate:Ha00010c 1 UDP 2130706431 10.0.1.12 10266 typ host
a=candidate:Ha00010c 2 UDP 2130706430 10.0.1.12 10267 typ host
a=sendrecv
Retransmitting #3 (NAT) to 216.115.69.144:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bK6e65.ac5315e7aca572674bbfa33e56b6e8c2.0;received=216.115.69.144;rport=5060
Via: SIP/2.0/UDP 216.115.69.131;branch=z9hG4bK6e65.b5a06a00cf08337f09b2be770364648f.0
Via: SIP/2.0/UDP 216.115.69.132;branch=z9hG4bK6e65.699a9a83fb2f8ad5001f2c4fdd8f7ebf.0
Via: SIP/2.0/UDP 4.55.17.35:5060;branch=z9hG4bK0dBb118609f52c22324
Record-Route: sip:216.115.69.144;lr
Record-Route: sip:216.115.69.132;lr
From: "<outbound#> " <sip:+<outbound#>@flowroute.com>;tag=gK0d0eb901
To: <sip:+<inbound#>@flowroute.com>;tag=as68ed7049
Call-ID: [email protected]
CSeq: 25438 INVITE
Server: FPBX-2.11.0beta2(11.1.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:<inbound#>@:5060>
Content-Type: application/sdp
Content-Length: 453
v=0
o=root 1023448070 1023448070 IN IP4
s=Asterisk PBX 11.1.2
c=IN IP4
t=0 0
m=audio 10266 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=ice-ufrag:01a27a320cbc5fd03eb48cec65059e07
a=ice-pwd:206f13662da5813d6d63b56a6d7def4b
a=candidate:Ha00010c 1 UDP 2130706431 10.0.1.12 10266 typ host
a=candidate:Ha00010c 2 UDP 2130706430 10.0.1.12 10267 typ host
a=sendrecv
Retransmitting #4 (NAT) to 216.115.69.144:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bK6e65.ac5315e7aca572674bbfa33e56b6e8c2.0;received=216.115.69.144;rport=5060
Via: SIP/2.0/UDP 216.115.69.131;branch=z9hG4bK6e65.b5a06a00cf08337f09b2be770364648f.0
Via: SIP/2.0/UDP 216.115.69.132;branch=z9hG4bK6e65.699a9a83fb2f8ad5001f2c4fdd8f7ebf.0
Via: SIP/2.0/UDP 4.55.17.35:5060;branch=z9hG4bK0dBb118609f52c22324
Record-Route: sip:216.115.69.144;lr
Record-Route: sip:216.115.69.132;lr
From: "<outbound#> " <sip:+<outbound#>@flowroute.com>;tag=gK0d0eb901
To: <sip:+<inbound#>@flowroute.com>;tag=as68ed7049
Call-ID: [email protected]
CSeq: 25438 INVITE
Server: FPBX-2.11.0beta2(11.1.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:<inbound#>@:5060>
Content-Type: application/sdp
Content-Length: 453
v=0
o=root 1023448070 1023448070 IN IP4
s=Asterisk PBX 11.1.2
c=IN IP4
t=0 0
m=audio 10266 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=ice-ufrag:01a27a320cbc5fd03eb48cec65059e07
a=ice-pwd:206f13662da5813d6d63b56a6d7def4b
a=candidate:Ha00010c 1 UDP 2130706431 10.0.1.12 10266 typ host
a=candidate:Ha00010c 2 UDP 2130706430 10.0.1.12 10267 typ host
a=sendrecv
Retransmitting #5 (NAT) to 216.115.69.144:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bK6e65.ac5315e7aca572674bbfa33e56b6e8c2.0;received=216.115.69.144;rport=5060
Via: SIP/2.0/UDP 216.115.69.131;branch=z9hG4bK6e65.b5a06a00cf08337f09b2be770364648f.0
Via: SIP/2.0/UDP 216.115.69.132;branch=z9hG4bK6e65.699a9a83fb2f8ad5001f2c4fdd8f7ebf.0
Via: SIP/2.0/UDP 4.55.17.35:5060;branch=z9hG4bK0dBb118609f52c22324
Record-Route: sip:216.115.69.144;lr
Record-Route: sip:216.115.69.132;lr
From: "<outbound#> " <sip:+<outbound#>@flowroute.com>;tag=gK0d0eb901
To: <sip:+<inbound#>@flowroute.com>;tag=as68ed7049
Call-ID: [email protected]
CSeq: 25438 INVITE
Server: FPBX-2.11.0beta2(11.1.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:<inbound#>@:5060>
Content-Type: application/sdp
Content-Length: 453
v=0
o=root 1023448070 1023448070 IN IP4
s=Asterisk PBX 11.1.2
c=IN IP4
t=0 0
m=audio 10266 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=ice-ufrag:01a27a320cbc5fd03eb48cec65059e07
a=ice-pwd:206f13662da5813d6d63b56a6d7def4b
a=candidate:Ha00010c 1 UDP 2130706431 10.0.1.12 10266 typ host
a=candidate:Ha00010c 2 UDP 2130706430 10.0.1.12 10267 typ host
a=sendrecv
Retransmitting #6 (NAT) to 216.115.69.144:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bK6e65.ac5315e7aca572674bbfa33e56b6e8c2.0;received=216.115.69.144;rport=5060
Via: SIP/2.0/UDP 216.115.69.131;branch=z9hG4bK6e65.b5a06a00cf08337f09b2be770364648f.0
Via: SIP/2.0/UDP 216.115.69.132;branch=z9hG4bK6e65.699a9a83fb2f8ad5001f2c4fdd8f7ebf.0
Via: SIP/2.0/UDP 4.55.17.35:5060;branch=z9hG4bK0dBb118609f52c22324
Record-Route: sip:216.115.69.144;lr
Record-Route: sip:216.115.69.132;lr
From: "<outbound#> " <sip:+<outbound#>@flowroute.com>;tag=gK0d0eb901
To: <sip:+<inbound#>@flowroute.com>;tag=as68ed7049
Call-ID: [email protected]
CSeq: 25438 INVITE
Server: FPBX-2.11.0beta2(11.1.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:<inbound#>@:5060>
Content-Type: application/sdp
Content-Length: 453
v=0
o=root 1023448070 1023448070 IN IP4
s=Asterisk PBX 11.1.2
c=IN IP4
t=0 0
m=audio 10266 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=ice-ufrag:01a27a320cbc5fd03eb48cec65059e07
a=ice-pwd:206f13662da5813d6d63b56a6d7def4b
a=candidate:Ha00010c 1 UDP 2130706431 10.0.1.12 10266 typ host
a=candidate:Ha00010c 2 UDP 2130706430 10.0.1.12 10267 typ host
a=sendrecv
[2013-02-17 11:57:01] WARNING[30149]: chan_sip.c:4164 retrans_pkt: Retransmission timeout reached on transmission [email protected] for seqno 25438 (Critical Response) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6400ms with no response
[2013-02-17 11:57:01] WARNING[30149]: chan_sip.c:4193 retrans_pkt: Hanging up call [email protected] - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
– Executing [h@macro-dial-one:1] Macro(“SIP/flowrouteLV-0000000a”, “hangupcall,”) in new stack
– Executing [s@macro-hangupcall:1] GotoIf(“SIP/flowrouteLV-0000000a”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,3)
– Executing [s@macro-hangupcall:3] ExecIf(“SIP/flowrouteLV-0000000a”, “0?Set(CDR(recordingfile)=)”) in new stack
– Executing [s@macro-hangupcall:4] Hangup(“SIP/flowrouteLV-0000000a”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on ‘SIP/flowrouteLV-0000000a’ in macro ‘hangupcall’
== Spawn extension (macro-dial-one, h, 1) exited non-zero on 'SIP/flowrouteLV-0000000a’
Scheduling destruction of SIP dialog ‘[email protected]:5060’ in 6400 ms (Method: INVITE)
set_destination: Parsing sip:[email protected]:5060;user=phone;transport=udp for address/port to send to
set_destination: set destination to 10.0.1.13:5060
Reliably Transmitting (no NAT) to 10.0.1.13:5060:
BYE sip:[email protected]:5060;user=phone;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.0.1.12:5060;branch=z9hG4bK51118442
Max-Forwards: 70
From: “+<outbound#>” <sip:+<outbound#>@10.0.1.12>;tag=as4fd96afc
To: sip:[email protected]:5060;user=phone;transport=udp;tag=001b0cdaf88000a67578368a-5a2d6186
Call-ID: [email protected]:5060
CSeq: 103 BYE
User-Agent: FPBX-2.11.0beta2(11.1.2)
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
== Spawn extension (macro-dial-one, s, 42) exited non-zero on ‘SIP/flowrouteLV-0000000a’ in macro ‘dial-one’
== Spawn extension (macro-exten-vm, s, 14) exited non-zero on ‘SIP/flowrouteLV-0000000a’ in macro ‘exten-vm’
== Spawn extension (from-did-direct, 105, 2) exited non-zero on 'SIP/flowrouteLV-0000000a’
Scheduling destruction of SIP dialog ‘[email protected]’ in 6400 ms (Method: INVITE)
set_destination: Parsing sip:216.115.69.144;lr for address/port to send to
set_destination: set destination to 216.115.69.144:5060
Reliably Transmitting (NAT) to 216.115.69.144:5060:
BYE sip:+<outbound#>@4.55.17.35:5060 SIP/2.0
Via: SIP/2.0/UDP :5060;branch=z9hG4bK1ebf15aa;rport
Route: sip:216.115.69.144;lr,sip:216.115.69.132;lr
Max-Forwards: 70
From: <sip:+<inbound#>@flowroute.com>;tag=as68ed7049
To: "<outbound#> " <sip:+<outbound#>@flowroute.com>;tag=gK0d0eb901
Call-ID: [email protected]
CSeq: 102 BYE
User-Agent: FPBX-2.11.0beta2(11.1.2)
X-Asterisk-HangupCause: No user responding
X-Asterisk-HangupCauseCode: 18
Content-Length: 0
<— SIP read from UDP:10.0.1.13:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.1.12:5060;branch=z9hG4bK51118442
From: “+<outbound#>” <sip:+<outbound#>@10.0.1.12>;tag=as4fd96afc
To: sip:[email protected]:5060;user=phone;transport=udp;tag=001b0cdaf88000a67578368a-5a2d6186
Call-ID: [email protected]:5060
Date: Sun, 17 Feb 2013 17:57:00 GMT
CSeq: 103 BYE
Server: Cisco-CP7960G/8.0
Content-Length: 0
<------------->
— (9 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]:5060’ Method: INVITE
Retransmitting #1 (NAT) to 216.115.69.144:5060:
BYE sip:+<outbound#>@4.55.17.35:5060 SIP/2.0
Via: SIP/2.0/UDP :5060;branch=z9hG4bK1ebf15aa;rport
Route: sip:216.115.69.144;lr,sip:216.115.69.132;lr
Max-Forwards: 70
From: <sip:+<inbound#>@flowroute.com>;tag=as68ed7049
To: "<outbound#> " <sip:+<outbound#>@flowroute.com>;tag=gK0d0eb901
Call-ID: [email protected]
CSeq: 102 BYE
User-Agent: FPBX-2.11.0beta2(11.1.2)
X-Asterisk-HangupCause: No user responding
X-Asterisk-HangupCauseCode: 18
Content-Length: 0
<— SIP read from UDP:216.115.69.144:5060 —>
SIP/2.0 200 OK
From: <sip:+<inbound#>@flowroute.com>;tag=as68ed7049
To: "<outbound#> " <sip:+<outbound#>@flowroute.com>;tag=gK0d0eb901
Via: SIP/2.0/UDP :5060;branch=z9hG4bK1ebf15aa;rport=18714
Call-ID: [email protected]
CSeq: 102 BYE
Record-Route: sip:216.115.69.132:5060;lr
Record-Route: sip:216.115.69.144:5060;lr
Content-Length: 0
<------------->
— (9 headers 0 lines) —
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog ‘[email protected]’ Method: INVITE
<— SIP read from UDP:216.115.69.144:5060 —>
SIP/2.0 200 OK
From: <sip:+<inbound#>@flowroute.com>;tag=as68ed7049
To: "<outbound#> " <sip:+<outbound#>@flowroute.com>;tag=gK0d0eb901
Via: SIP/2.0/UDP :5060;branch=z9hG4bK1ebf15aa;rport=18714
Call-ID: [email protected]
CSeq: 102 BYE
Record-Route: sip:216.115.69.132:5060;lr
Record-Route: sip:216.115.69.144:5060;lr
Content-Length: 0
<------------->
— (9 headers 0 lines) —
pbxCLI> sip set debug off
SIP Debugging Disabled
pbxCLI>