Greetings to all!
I have finally have a 99% working server but incoming calls are being dropped. I have Googled and found no solutions that really address this issue. Firewall has all the required settings based on everything I have read. Can someone be so kind as to shed some light on the subject here. Please see bellow and thank you in advance.
[2011-12-01 14:18:00] VERBOSE[29418] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
[2011-12-01 14:18:00] VERBOSE[29418] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[2011-12-01 14:18:00] VERBOSE[29418] chan_sip.c:
<— Reliably Transmitting (no NAT) to xx.xx.xx.xx:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP xx.xx.xx.xx;branch=z9hG4bKc8ee.6a141044.0;received=xx.xx.xx.xx
Via: SIP/2.0/UDP 81.201.84.195:5060;branch=z9hG4bKd9e7b4a4e8e8edeb6ea2e7d9f8751579
Record-Route: sip:xx.xx.xx.xx;ftag=68693;nat=yes;lr=on
From: “18005551212” sip:[email protected];tag=68693
To: sip:[email protected];tag=as2b11125f
Call-ID: [email protected]
CSeq: 102 INVITE
Server: FPBX-2.9.0(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 283
v=0
o=root 2053286403 2053286403 IN IP4 xx.xxx.xx.xx
s=Asterisk PBX 1.8.7.0
c=IN IP4 xx.xxx.xx.xx
t=0 0
m=audio 16954 RTP/AVP 18 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------>
[2011-12-01 14:18:00] VERBOSE[2792] chan_sip.c: Retransmitting #1 (no NAT) to xx.xx.xx.xx:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP xx.xx.xx.xx;branch=z9hG4bKc8ee.6a141044.0;received=xx.xx.xx.xx
Via: SIP/2.0/UDP 81.201.84.195:5060;branch=z9hG4bKd9e7b4a4e8e8edeb6ea2e7d9f8751579
Record-Route: sip:xx.xx.xx.xx;ftag=68693;nat=yes;lr=on
From: “18005551212” sip:[email protected];tag=68693
To: sip:[email protected];tag=as2b11125f
Call-ID: [email protected]
CSeq: 102 INVITE
Server: FPBX-2.9.0(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 283
v=0
o=root 2053286403 2053286403 IN IP4 xx.xxx.xx.xx
s=Asterisk PBX 1.8.7.0
c=IN IP4 xx.xxx.xx.xx
t=0 0
m=audio 16954 RTP/AVP 18 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
[2011-12-01 14:18:01] VERBOSE[2792] chan_sip.c: Retransmitting #2 (no NAT) to xx.xx.xx.xx:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP xx.xx.xx.xx;branch=z9hG4bKc8ee.6a141044.0;received=xx.xx.xx.xx
Via: SIP/2.0/UDP 81.201.84.195:5060;branch=z9hG4bKd9e7b4a4e8e8edeb6ea2e7d9f8751579
Record-Route: sip:xx.xx.xx.xx;ftag=68693;nat=yes;lr=on
From: “18005551212” sip:[email protected];tag=68693
To: sip:[email protected];tag=as2b11125f
Call-ID: [email protected]
CSeq: 102 INVITE
Server: FPBX-2.9.0(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 283
v=0
o=root 2053286403 2053286403 IN IP4 xx.xxx.xx.xx
s=Asterisk PBX 1.8.7.0
c=IN IP4 xx.xxx.xx.xx
t=0 0
m=audio 16954 RTP/AVP 18 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
[2011-12-01 14:18:03] VERBOSE[2792] chan_sip.c: Retransmitting #3 (no NAT) to xx.xx.xx.xx:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP xx.xx.xx.xx;branch=z9hG4bKc8ee.6a141044.0;received=xx.xx.xx.xx
Via: SIP/2.0/UDP 81.201.84.195:5060;branch=z9hG4bKd9e7b4a4e8e8edeb6ea2e7d9f8751579
Record-Route: sip:xx.xx.xx.xx;ftag=68693;nat=yes;lr=on
From: “18005551212” sip:[email protected];tag=68693
To: sip:[email protected];tag=as2b11125f
Call-ID: [email protected]
CSeq: 102 INVITE
Server: FPBX-2.9.0(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 283
v=0
o=root 2053286403 2053286403 IN IP4 xx.xxx.xx.xx
s=Asterisk PBX 1.8.7.0
c=IN IP4 xx.xxx.xx.xx
t=0 0
m=audio 16954 RTP/AVP 18 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
[2011-12-01 14:18:05] VERBOSE[29420] manager.c: == Manager ‘admin’ logged on from 127.0.0.1
[2011-12-01 14:18:05] VERBOSE[29420] manager.c: == Manager ‘admin’ logged off from 127.0.0.1
[2011-12-01 14:18:05] VERBOSE[2792] chan_sip.c: Reliably Transmitting (no NAT) to xxx.xx.xxx.xxx:5060:
OPTIONS sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP FreePBXIPADD:5060;branch=z9hG4bK48f4a7cb
Max-Forwards: 70
From: “Unknown” sip:Unknown@FreePBXIPADD;tag=as73bfc3bd
To: sip:[email protected]:5060
Contact: sip:Unknown@FreePBXIPADD:5060
Call-ID: 52754b52469b12ab1109edcc015da6d6@FreePBXIPADD:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.9.0(1.8.7.0)
Date: Thu, 01 Dec 2011 19:18:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
[2011-12-01 14:18:05] VERBOSE[2792] chan_sip.c:
<— SIP read from UDP:xxx.xx.xxx.xxx:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP FreePBXIPADD:5060;branch=z9hG4bK48f4a7cb
From: “Unknown” sip:Unknown@FreePBXIPADD;tag=as73bfc3bd
To: “Ramon F McDougall” sip:[email protected]:5060;tag=F72BC885-E820DD40
CSeq: 102 OPTIONS
Call-ID: 52754b52469b12ab1109edcc015da6d6@FreePBXIPADD:5060
Contact: sip:[email protected]:5060
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Supported: 100rel,replaces,100rel,timer,replaces,norefersub
User-Agent: PolycomSoundPointIP-SPIP_331-UA/3.2.4.0267
Accept-Language: en
Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Content-Length: 0
<------------->
[2011-12-01 14:18:05] VERBOSE[2792] chan_sip.c: — (14 headers 0 lines) —
[2011-12-01 14:18:05] VERBOSE[2792] chan_sip.c: Really destroying SIP dialog ‘52754b52469b12ab1109edcc015da6d6@FreePBXIPADD:5060’ Method: OPTIONS
[2011-12-01 14:18:06] VERBOSE[29422] manager.c: == Manager ‘admin’ logged on from 127.0.0.1
[2011-12-01 14:18:07] VERBOSE[29422] manager.c: == Manager ‘admin’ logged off from 127.0.0.1
[2011-12-01 14:18:07] VERBOSE[2792] chan_sip.c: Retransmitting #4 (no NAT) to xx.xx.xx.xx:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP xx.xx.xx.xx;branch=z9hG4bKc8ee.6a141044.0;received=xx.xx.xx.xx
Via: SIP/2.0/UDP 81.201.84.195:5060;branch=z9hG4bKd9e7b4a4e8e8edeb6ea2e7d9f8751579
Record-Route: sip:xx.xx.xx.xx;ftag=68693;nat=yes;lr=on
From: “18005551212” sip:[email protected];tag=68693
To: sip:[email protected];tag=as2b11125f
Call-ID: [email protected]
CSeq: 102 INVITE
Server: FPBX-2.9.0(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 283
v=0
o=root 2053286403 2053286403 IN IP4 xx.xxx.xx.xx
s=Asterisk PBX 1.8.7.0
c=IN IP4 xx.xxx.xx.xx
t=0 0
m=audio 16954 RTP/AVP 18 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
[2011-12-01 14:18:11] VERBOSE[2792] chan_sip.c: Retransmitting #5 (no NAT) to xx.xx.xx.xx:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP xx.xx.xx.xx;branch=z9hG4bKc8ee.6a141044.0;received=xx.xx.xx.xx
Via: SIP/2.0/UDP 81.201.84.195:5060;branch=z9hG4bKd9e7b4a4e8e8edeb6ea2e7d9f8751579
Record-Route: sip:xx.xx.xx.xx;ftag=68693;nat=yes;lr=on
From: “18005551212” sip:[email protected];tag=68693
To: sip:[email protected];tag=as2b11125f
Call-ID: [email protected]
CSeq: 102 INVITE
Server: FPBX-2.9.0(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 283
v=0
o=root 2053286403 2053286403 IN IP4 xx.xxx.xx.xx
s=Asterisk PBX 1.8.7.0
c=IN IP4 xx.xxx.xx.xx
t=0 0
m=audio 16954 RTP/AVP 18 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
[2011-12-01 14:18:15] VERBOSE[2792] chan_sip.c: Retransmitting #6 (no NAT) to xx.xx.xx.xx:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP xx.xx.xx.xx;branch=z9hG4bKc8ee.6a141044.0;received=xx.xx.xx.xx
Via: SIP/2.0/UDP 81.201.84.195:5060;branch=z9hG4bKd9e7b4a4e8e8edeb6ea2e7d9f8751579
Record-Route: sip:xx.xx.xx.xx;ftag=68693;nat=yes;lr=on
From: “18005551212” sip:[email protected];tag=68693
To: sip:[email protected];tag=as2b11125f
Call-ID: [email protected]
CSeq: 102 INVITE
Server: FPBX-2.9.0(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 283
v=0
o=root 2053286403 2053286403 IN IP4 xx.xxx.xx.xx
s=Asterisk PBX 1.8.7.0
c=IN IP4 xx.xxx.xx.xx
t=0 0
m=audio 16954 RTP/AVP 18 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
[2011-12-01 14:18:19] VERBOSE[2792] chan_sip.c: Retransmitting #7 (no NAT) to xx.xx.xx.xx:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP xx.xx.xx.xx;branch=z9hG4bKc8ee.6a141044.0;received=xx.xx.xx.xx
Via: SIP/2.0/UDP 81.201.84.195:5060;branch=z9hG4bKd9e7b4a4e8e8edeb6ea2e7d9f8751579
Record-Route: sip:xx.xx.xx.xx;ftag=68693;nat=yes;lr=on
From: “18005551212” sip:[email protected];tag=68693
To: sip:[email protected];tag=as2b11125f
Call-ID: [email protected]
CSeq: 102 INVITE
Server: FPBX-2.9.0(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 283
v=0
o=root 2053286403 2053286403 IN IP4 xx.xxx.xx.xx
s=Asterisk PBX 1.8.7.0
c=IN IP4 xx.xxx.xx.xx
t=0 0
m=audio 16954 RTP/AVP 18 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
[2011-12-01 14:18:20] VERBOSE[2792] chan_sip.c: Really destroying SIP dialog ‘[email protected]’ Method: OPTIONS
[2011-12-01 14:18:23] VERBOSE[2792] chan_sip.c: Retransmitting #8 (no NAT) to xx.xx.xx.xx:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP xx.xx.xx.xx;branch=z9hG4bKc8ee.6a141044.0;received=xx.xx.xx.xx
Via: SIP/2.0/UDP 81.201.84.195:5060;branch=z9hG4bKd9e7b4a4e8e8edeb6ea2e7d9f8751579
Record-Route: sip:xx.xx.xx.xx;ftag=68693;nat=yes;lr=on
From: “18005551212” sip:[email protected];tag=68693
To: sip:[email protected];tag=as2b11125f
Call-ID: [email protected]
CSeq: 102 INVITE
Server: FPBX-2.9.0(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 283
v=0
o=root 2053286403 2053286403 IN IP4 xx.xxx.xx.xx
s=Asterisk PBX 1.8.7.0
c=IN IP4 xx.xxx.xx.xx
t=0 0
m=audio 16954 RTP/AVP 18 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
[2011-12-01 14:18:23] VERBOSE[2792] chan_sip.c:
<— SIP read from UDP:xx.xx.xx.xx:5060 —>
OPTIONS sip:xx.xxx.xx.xx:1345 SIP/2.0
Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=0
From: sip:[email protected];tag=fc6826d1
To: sip:xx.xxx.xx.xx:1345
Call-ID: [email protected]
CSeq: 1 OPTIONS
Content-Length: 0
<------------->
[2011-12-01 14:18:23] VERBOSE[2792] chan_sip.c: — (7 headers 0 lines) —
[2011-12-01 14:18:23] VERBOSE[2792] chan_sip.c: Looking for s in from-sip-external (domain xx.xxx.xx.xx:1345)
[2011-12-01 14:18:23] VERBOSE[2792] chan_sip.c:
<— Transmitting (NAT) to xx.xx.xx.xx:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=0;received=xx.xx.xx.xx;rport=5060
From: sip:[email protected];tag=xxxxxxxx
To: sip:xx.xxx.xx.xx:1345;tag=as2342c90b
Call-ID: [email protected]
CSeq: 1 OPTIONS
Server: FPBX-2.9.0(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:xx.xxx.xx.xx:5060
Accept: application/sdp
Content-Length: 0
<------------>
[2011-12-01 14:18:23] VERBOSE[2792] chan_sip.c: Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: OPTIONS)
[2011-12-01 14:18:27] VERBOSE[2792] chan_sip.c: Retransmitting #9 (no NAT) to xx.xx.xx.xx:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP xx.xx.xx.xx;branch=z9hG4bKc8ee.6a141044.0;received=xx.xx.xx.xx
Via: SIP/2.0/UDP 81.201.84.195:5060;branch=z9hG4bKd9e7b4a4e8e8edeb6ea2e7d9f8751579
Record-Route: sip:xx.xx.xx.xx;ftag=68693;nat=yes;lr=on
From: “18005551212” sip:[email protected];tag=68693
To: sip:[email protected];tag=as2b11125f
Call-ID: [email protected]
CSeq: 102 INVITE
Server: FPBX-2.9.0(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 283
v=0
o=root 2053286403 2053286403 IN IP4 xx.xxx.xx.xx
s=Asterisk PBX 1.8.7.0
c=IN IP4 xx.xxx.xx.xx
t=0 0
m=audio 16954 RTP/AVP 18 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
[2011-12-01 14:18:31] VERBOSE[29427] manager.c: == Manager ‘admin’ logged on from 127.0.0.1
[2011-12-01 14:18:31] VERBOSE[29427] manager.c: == Manager ‘admin’ logged off from 127.0.0.1
[2011-12-01 14:18:31] VERBOSE[2792] chan_sip.c: Retransmitting #10 (no NAT) to xx.xx.xx.xx:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP xx.xx.xx.xx;branch=z9hG4bKc8ee.6a141044.0;received=xx.xx.xx.xx
Via: SIP/2.0/UDP 81.201.84.195:5060;branch=z9hG4bKd9e7b4a4e8e8edeb6ea2e7d9f8751579
Record-Route: sip:xx.xx.xx.xx;ftag=68693;nat=yes;lr=on
From: “18005551212” sip:[email protected];tag=68693
To: sip:[email protected];tag=as2b11125f
Call-ID: [email protected]
CSeq: 102 INVITE
Server: FPBX-2.9.0(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 283
v=0
o=root 2053286403 2053286403 IN IP4 xx.xxx.xx.xx
s=Asterisk PBX 1.8.7.0
c=IN IP4 xx.xxx.xx.xx
t=0 0
m=audio 16954 RTP/AVP 18 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
[2011-12-01 14:18:32] WARNING[2792] chan_sip.c: Retransmission timeout reached on transmission [email protected] for seqno 102 (Critical Response) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
[2011-12-01 14:18:32] WARNING[2792] chan_sip.c: Hanging up call [email protected] - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
[2011-12-01 14:18:32] VERBOSE[29418] pbx.c: – Executing [h@macro-dial-one:1] Macro(“SIP/VoipVoip-00000075”, “hangupcall,”) in new stack
[2011-12-01 14:18:32] VERBOSE[29418] pbx.c: – Executing [s@macro-hangupcall:1] GotoIf(“SIP/VoipVoip-00000075”, “1?theend”) in new stack
[2011-12-01 14:18:32] VERBOSE[29418] pbx.c: – Goto (macro-hangupcall,s,3)
[2011-12-01 14:18:32] VERBOSE[29418] pbx.c: – Executing [s@macro-hangupcall:3] Hangup(“SIP/VoipVoip-00000075”, “”) in new stack
[2011-12-01 14:18:32] VERBOSE[29418] app_macro.c: == Spawn extension (macro-hangupcall, s, 3) exited non-zero on ‘SIP/VoipVoip-00000075’ in macro ‘hangupcall’
[2011-12-01 14:18:32] VERBOSE[29418] features.c: == Spawn extension (macro-dial-one, h, 1) exited non-zero on ‘SIP/VoipVoip-00000075’
[2011-12-01 14:18:32] VERBOSE[29418] chan_sip.c: Scheduling destruction of SIP dialog ‘01ab2e205f843b0e00b31cd556f1e4a7@FreePBXIPADD:5060’ in 6400 ms (Method: INVITE)
[2011-12-01 14:18:32] VERBOSE[29418] chan_sip.c: set_destination: Parsing sip:[email protected]:5060 for address/port to send to
[2011-12-01 14:18:32] VERBOSE[29418] chan_sip.c: set_destination: set destination to xxx.xx.xxx.xxx:5060
[2011-12-01 14:18:32] VERBOSE[29418] chan_sip.c: Reliably Transmitting (no NAT) to xxx.xx.xxx.xxx:5060:
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP FreePBXIPADD:5060;branch=z9hG4bK1e85e8ef
Max-Forwards: 70
From: “18005551212” sip:18005551212@FreePBXIPADD;tag=as1a25bd1e
To: sip:[email protected]:5060;tag=F502CC4-D8002607
Call-ID: 01ab2e205f843b0e00b31cd556f1e4a7@FreePBXIPADD:5060
CSeq: 103 BYE
User-Agent: FPBX-2.9.0(1.8.7.0)
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
[2011-12-01 14:18:32] VERBOSE[29418] app_macro.c: == Spawn extension (macro-dial-one, s, 42) exited non-zero on ‘SIP/VoipVoip-00000075’ in macro ‘dial-one’
[2011-12-01 14:18:32] VERBOSE[29418] app_macro.c: == Spawn extension (macro-exten-vm, s, 7) exited non-zero on ‘SIP/VoipVoip-00000075’ in macro ‘exten-vm’
[2011-12-01 14:18:32] VERBOSE[29418] pbx.c: == Spawn extension (from-did-direct, 2000, 2) exited non-zero on ‘SIP/VoipVoip-00000075’
[2011-12-01 14:18:32] VERBOSE[29418] chan_sip.c: Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: INVITE)
[2011-12-01 14:18:32] VERBOSE[29418] chan_sip.c: set_destination: Parsing sip:xx.xx.xx.xx;ftag=68693;nat=yes;lr=on for address/port to send to
[2011-12-01 14:18:32] VERBOSE[29418] chan_sip.c: set_destination: set destination to xx.xx.xx.xx:5060
[2011-12-01 14:18:32] VERBOSE[29418] chan_sip.c: Reliably Transmitting (no NAT) to xx.xx.xx.xx:5060:
BYE sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP xx.xxx.xx.xx:5060;branch=z9hG4bK1a603d48
Route: sip:xx.xx.xx.xx;ftag=68693;nat=yes;lr=on
Max-Forwards: 70
From: sip:[email protected];tag=as2b11125f
To: “18005551212” sip:[email protected];tag=68693
Call-ID: [email protected]
CSeq: 102 BYE
User-Agent: FPBX-2.9.0(1.8.7.0)
X-Asterisk-HangupCause: Protocol error, unspecified
X-Asterisk-HangupCauseCode: 111
Content-Length: 0
[2011-12-01 14:18:32] VERBOSE[2792] chan_sip.c:
<— SIP read from UDP:xxx.xx.xxx.xxx:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP FreePBXIPADD:5060;branch=z9hG4bK1e85e8ef
From: “18005551212” sip:18005551212@FreePBXIPADD;tag=as1a25bd1e
To: “Ramon F McDougall” sip:[email protected]:5060;tag=F502CC4-D8002607
CSeq: 103 BYE
Call-ID: 01ab2e205f843b0e00b31cd556f1e4a7@FreePBXIPADD:5060
Contact: sip:[email protected]:5060
User-Agent: PolycomSoundPointIP-SPIP_331-UA/3.2.4.0267
Accept-Language: en
Content-Length: 0
<------------->
[2011-12-01 14:18:32] VERBOSE[2792] chan_sip.c: — (10 headers 0 lines) —
[2011-12-01 14:18:32] VERBOSE[2792] chan_sip.c: Really destroying SIP dialog ‘01ab2e205f843b0e00b31cd556f1e4a7@FreePBXIPADD:5060’ Method: INVITE
[2011-12-01 14:18:32] VERBOSE[2792] chan_sip.c:
<— SIP read from UDP:xx.xx.xx.xx:5060 —>
SIP/2.0 200 OK
Record-Route: sip:xx.xx.xx.xx;ftag=as2b11125f;lr=on
Via: SIP/2.0/UDP xx.xxx.xx.xx:5060;branch=z9hG4bK1a603d48;rport=1345
From: sip:[email protected];tag=as2b11125f
To: “18005551212” sip:[email protected];tag=68693
Call-ID: [email protected]
CSeq: 102 BYE
Content-Length: 0
<------------->
[2011-12-01 14:18:32] VERBOSE[2792] chan_sip.c: — (8 headers 0 lines) —
[2011-12-01 14:18:32] VERBOSE[2792] chan_sip.c: SIP Response message for INCOMING dialog BYE arrived
[2011-12-01 14:18:32] VERBOSE[2792] chan_sip.c: Really destroying SIP dialog ‘[email protected]’ Method: INVITE
[2011-12-01 14:18:32] VERBOSE[29429] manager.c: == Manager ‘admin’ logged on from 127.0.0.1
Thank you all!