malcolmd wrote:What does Asterisk's SIP debug say? Does Lync have some kind of debugging beyond that message?
Here is my Asterisk debug log. For Lync log i will try to find it too.
Lync Server and asteriskNOW server side public IP: 2xx.xxx.1xx.xx1
Client side public ip: 2yy.yyy.1yy.x5
Softphone: 3CX
[Feb 15 10:37:45] VERBOSE[2484] chan_sip.c:
<--- SIP read from UDP:2yy.yyy.1yy.x5:5060 --->
REGISTER sip:2xx.xxx.1xx.xx1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.36.18:5060;branch=z9hG4bK-d8754z-eb06a36727765b64-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:101@2yy.yyy.1yy.x5:5060;rinstance=d7d50749ac51fbee>
To: <sip:101@2xx.xxx.1xx.xx1:5060>
From: <sip:101@2xx.xxx.1xx.xx1:5060>;tag=4b62eb7e
Call-ID: ZWUxZWJjNTBhMmJlOGMyZDdjNjViZTE4Nzg1ZGY1YTk.
CSeq: 134 REGISTER
Expires: 120
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Supported: replaces
User-Agent: 3CXPhone 5.0.14900.0
Authorization: Digest username="101",realm="asterisk",nonce="5ce9efbf",uri="sip:2xx.xxx.1xx.xx1:5060",response="cb4e74749e4c0e35ea99e0807e352da5",algorithm=MD5
Content-Length: 0
<------------->
[Feb 15 10:37:45] VERBOSE[2484] chan_sip.c: --- (14 headers 0 lines) ---
[Feb 15 10:37:45] VERBOSE[2484] chan_sip.c: Sending to 2yy.yyy.1yy.x5 : 5060 (NAT)
[Feb 15 10:37:45] VERBOSE[2484] chan_sip.c:
<--- Transmitting (NAT) to 2yy.yyy.1yy.x5:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.36.18:5060;branch=z9hG4bK-d8754z-eb06a36727765b64-1---d8754z-;received=2yy.yyy.1yy.x5;rport=5060
From: <sip:101@2xx.xxx.1xx.xx1:5060>;tag=4b62eb7e
To: <sip:101@2xx.xxx.1xx.xx1:5060>;tag=as57857641
Call-ID: ZWUxZWJjNTBhMmJlOGMyZDdjNjViZTE4Nzg1ZGY1YTk.
CSeq: 134 REGISTER
Server: FPBX-2.7.0(1.6.2.16.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3b6a26ac"
Content-Length: 0
<------------>
[Feb 15 10:37:45] VERBOSE[2484] chan_sip.c: Scheduling destruction of SIP dialog 'ZWUxZWJjNTBhMmJlOGMyZDdjNjViZTE4Nzg1ZGY1YTk.' in 32000 ms (Method: REGISTER)
[Feb 15 10:37:45] VERBOSE[2484] chan_sip.c:
<--- SIP read from UDP:2yy.yyy.1yy.x5:5060 --->
REGISTER sip:2xx.xxx.1xx.xx1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.36.18:5060;branch=z9hG4bK-d8754z-26379b6e7d297d55-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:101@2yy.yyy.1yy.x5:5060;rinstance=d7d50749ac51fbee>
To: <sip:101@2xx.xxx.1xx.xx1:5060>
From: <sip:101@2xx.xxx.1xx.xx1:5060>;tag=4b62eb7e
Call-ID: ZWUxZWJjNTBhMmJlOGMyZDdjNjViZTE4Nzg1ZGY1YTk.
CSeq: 135 REGISTER
Expires: 120
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Supported: replaces
User-Agent: 3CXPhone 5.0.14900.0
Authorization: Digest username="101",realm="asterisk",nonce="3b6a26ac",uri="sip:2xx.xxx.1xx.xx1:5060",response="b5ef97207dc7f1a3d0ad3f697c76af52",algorithm=MD5
Content-Length: 0
<------------->
[Feb 15 10:37:45] VERBOSE[2484] chan_sip.c: --- (14 headers 0 lines) ---
[Feb 15 10:37:45] VERBOSE[2484] chan_sip.c: Sending to 2yy.yyy.1yy.x5 : 5060 (NAT)
[Feb 15 10:37:45] VERBOSE[2484] chan_sip.c: Reliably Transmitting (NAT) to 2yy.yyy.1yy.x5:5060:
OPTIONS sip:101@2yy.yyy.1yy.x5:5060;rinstance=d7d50749ac51fbee SIP/2.0
Via: SIP/2.0/UDP 2xx.xxx.1xx.xx1:5060;branch=z9hG4bK6ff3a36d;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@2xx.xxx.1xx.xx1>;tag=as089178d7
To: <sip:101@2yy.yyy.1yy.x5:5060;rinstance=d7d50749ac51fbee>
Contact: <sip:Unknown@2xx.xxx.1xx.xx1>
Call-ID:
67d11097592ab77840887ff14bf32919@2xx.xxx.1xx.xx1
CSeq: 102 OPTIONS
User-Agent: FPBX-2.7.0(1.6.2.16.1)
Date: Tue, 15 Feb 2011 08:37:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
---
[Feb 15 10:37:45] VERBOSE[2484] chan_sip.c:
<--- Transmitting (NAT) to 2yy.yyy.1yy.x5:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.36.18:5060;branch=z9hG4bK-d8754z-26379b6e7d297d55-1---d8754z-;received=2yy.yyy.1yy.x5;rport=5060
From: <sip:101@2xx.xxx.1xx.xx1:5060>;tag=4b62eb7e
To: <sip:101@2xx.xxx.1xx.xx1:5060>;tag=as57857641
Call-ID: ZWUxZWJjNTBhMmJlOGMyZDdjNjViZTE4Nzg1ZGY1YTk.
CSeq: 135 REGISTER
Server: FPBX-2.7.0(1.6.2.16.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 60
Contact: <sip:101@2yy.yyy.1yy.x5:5060;rinstance=d7d50749ac51fbee>;expires=60
Date: Tue, 15 Feb 2011 08:37:45 GMT
Content-Length: 0
<------------>
[Feb 15 10:37:45] VERBOSE[2484] chan_sip.c: Scheduling destruction of SIP dialog 'ZWUxZWJjNTBhMmJlOGMyZDdjNjViZTE4Nzg1ZGY1YTk.' in 32000 ms (Method: REGISTER)
[Feb 15 10:37:45] VERBOSE[2484] chan_sip.c:
<--- SIP read from UDP:2yy.yyy.1yy.x5:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 2xx.xxx.1xx.xx1:5060;branch=z9hG4bK6ff3a36d;rport=5060
Contact: <sip:192.168.36.18:5060>
To: <sip:101@2yy.yyy.1yy.x5:5060;rinstance=d7d50749ac51fbee>;tag=68672505
From: "Unknown"<sip:Unknown@2xx.xxx.1xx.xx1>;tag=as089178d7
Call-ID:
67d11097592ab77840887ff14bf32919@2xx.xxx.1xx.xx1
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Supported: replaces
Allow-Events: presence, message-summary, tunnel-info
Content-Length: 0
<------------->
[Feb 15 10:37:45] VERBOSE[2484] chan_sip.c: --- (13 headers 0 lines) ---
[Feb 15 10:37:45] VERBOSE[2484] chan_sip.c: Really destroying SIP dialog
'67d11097592ab77840887ff14bf32919@2xx.xxx.1xx.xx1' Method: OPTIONS
[Feb 15 10:37:52] VERBOSE[9451] chan_sip.c:
<--- SIP read from TCP:192.168.68.91:60257 --->
INVITE sip:101@asterisknow01.mydomain.lt;user=phone SIP/2.0
FROM: "Artūras Rimonis"<sip:2001@fe01.mydomain.lt;user=phone>;epid=03259E13E5;tag=3bca2e7b83
TO: <sip:101@asterisknow01.mydomain.lt;user=phone>
CSEQ: 23319 INVITE
CALL-ID: 843c3b3a-20a8-425a-9804-2668f9d2a1c9
MAX-FORWARDS: 70
VIA: SIP/2.0/TCP 192.168.68.91:60257;branch=z9hG4bKadf9fb25
CONTACT: <sip:fe01.mydomain.lt:5068;transport=Tcp;maddr=192.168.68.91;ms-opaque=a8e975a9c1cf1341>
CONTENT-LENGTH: 343
SUPPORTED: 100rel
USER-AGENT: RTCC/4.0.0.0 MediationServer
CONTENT-TYPE: application/sdp
ALLOW: ACK
Allow: CANCEL,BYE,INVITE,PRACK,UPDATE
v=0
o=- 148 1 IN IP4 192.168.68.91
s=session
c=IN IP4 192.168.68.91
b=CT:1000
t=0 0
m=audio 57422 RTP/AVP 97 101 13 0 8
c=IN IP4 192.168.68.91
a=rtcp:57423
a=label:Audio
a=sendrecv
a=rtpmap:97 RED/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
<------------->
[Feb 15 10:37:52] VERBOSE[9451] chan_sip.c: --- (14 headers 18 lines) ---
[Feb 15 10:37:52] VERBOSE[9451] netsock.c: == Using SIP RTP TOS bits 184
[Feb 15 10:37:52] VERBOSE[9451] netsock.c: == Using SIP RTP CoS mark 5
[Feb 15 10:37:52] VERBOSE[9451] chan_sip.c: Sending to 192.168.68.91 : 60257 (NAT)
[Feb 15 10:37:52] VERBOSE[9451] chan_sip.c: Using INVITE request as basis request - 843c3b3a-20a8-425a-9804-2668f9d2a1c9
[Feb 15 10:37:52] VERBOSE[9451] chan_sip.c: Found peer 'LYNC' for '2001' from 192.168.68.91:60257
[Feb 15 10:37:52] VERBOSE[9451] chan_sip.c: Found RTP audio format 97
[Feb 15 10:37:52] VERBOSE[9451] chan_sip.c: Found RTP audio format 101
[Feb 15 10:37:52] VERBOSE[9451] chan_sip.c: Found RTP audio format 13
[Feb 15 10:37:52] VERBOSE[9451] chan_sip.c: Found RTP audio format 0
[Feb 15 10:37:52] VERBOSE[9451] chan_sip.c: Found RTP audio format 8
[Feb 15 10:37:52] VERBOSE[9451] chan_sip.c: Found audio description format RED for ID 97
[Feb 15 10:37:52] VERBOSE[9451] chan_sip.c: Found audio description format telephone-event for ID 101
[Feb 15 10:37:52] VERBOSE[9451] chan_sip.c: Found audio description format CN for ID 13
[Feb 15 10:37:52] VERBOSE[9451] chan_sip.c: Found audio description format PCMU for ID 0
[Feb 15 10:37:52] VERBOSE[9451] chan_sip.c: Found audio description format PCMA for ID 8
[Feb 15 10:37:52] VERBOSE[9451] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x40c (ulaw|alaw|ilbc)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
[Feb 15 10:37:52] VERBOSE[9451] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x3 (telephone-event|CN), combined - 0x1 (telephone-event)
[Feb 15 10:37:52] VERBOSE[9451] chan_sip.c: Peer audio RTP is at port 192.168.68.91:57422
[Feb 15 10:37:52] VERBOSE[9451] chan_sip.c: Looking for 101 in from-lync (domain asterisknow01.mydomain.lt)
[Feb 15 10:37:52] VERBOSE[9451] chan_sip.c: list_route: hop: <sip:fe01.mydomain.lt:5068;transport=Tcp;maddr=192.168.68.91;ms-opaque=a8e975a9c1cf1341>
[Feb 15 10:37:52] VERBOSE[9451] chan_sip.c:
<--- Transmitting (NAT) to 192.168.68.91:60257 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 192.168.68.91:60257;branch=z9hG4bKadf9fb25;received=192.168.68.91
From: "Artūras Rimonis"<sip:2001@fe01.mydomain.lt;user=phone>;epid=03259E13E5;tag=3bca2e7b83
To: <sip:101@asterisknow01.mydomain.lt;user=phone>
Call-ID: 843c3b3a-20a8-425a-9804-2668f9d2a1c9
CSeq: 23319 INVITE
Server: FPBX-2.7.0(1.6.2.16.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:101@192.168.68.156;transport=TCP>
Content-Length: 0
<------------>
[Feb 15 10:37:52] VERBOSE[9452] pbx.c: -- Executing [101@from-lync:1] Answer("SIP/LYNC-00000012", "") in new stack
[Feb 15 10:37:52] VERBOSE[9452] chan_sip.c: Audio is at 192.168.68.156 port 10084
[Feb 15 10:37:52] VERBOSE[9452] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
[Feb 15 10:37:52] VERBOSE[9452] chan_sip.c: Adding codec 0x8 (alaw) to SDP
[Feb 15 10:37:52] VERBOSE[9452] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Feb 15 10:37:52] VERBOSE[9452] chan_sip.c:
<--- Reliably Transmitting (NAT) to 192.168.68.91:60257 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 192.168.68.91:60257;branch=z9hG4bKadf9fb25;received=192.168.68.91
From: "Artūras Rimonis"<sip:2001@fe01.mydomain.lt;user=phone>;epid=03259E13E5;tag=3bca2e7b83
To: <sip:101@asterisknow01.mydomain.lt;user=phone>;tag=as10e2af07
Call-ID: 843c3b3a-20a8-425a-9804-2668f9d2a1c9
CSeq: 23319 INVITE
Server: FPBX-2.7.0(1.6.2.16.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:101@192.168.68.156;transport=TCP>
Content-Type: application/sdp
Content-Length: 265
v=0
o=root 514945289 514945289 IN IP4 192.168.68.156
s=Asterisk PBX 1.6.2.16.1
c=IN IP4 192.168.68.156
t=0 0
m=audio 10084 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
<------------>
[Feb 15 10:37:52] VERBOSE[9451] chan_sip.c:
<--- SIP read from TCP:192.168.68.91:60257 --->
ACK sip:101@192.168.68.156;transport=TCP SIP/2.0
FROM: <sip:2001@fe01.mydomain.lt;user=phone>;epid=03259E13E5;tag=3bca2e7b83
TO: <sip:101@asterisknow01.mydomain.lt;user=phone>;tag=as10e2af07
CSEQ: 23319 ACK
CALL-ID: 843c3b3a-20a8-425a-9804-2668f9d2a1c9
MAX-FORWARDS: 70
VIA: SIP/2.0/TCP 192.168.68.91:60257;branch=z9hG4bK9a2162f8
CONTENT-LENGTH: 0
USER-AGENT: RTCC/4.0.0.0 MediationServer
<------------->
[Feb 15 10:37:52] VERBOSE[9451] chan_sip.c: --- (9 headers 0 lines) ---
[Feb 15 10:37:53] VERBOSE[9452] pbx.c: -- Executing [101@from-lync:2] Dial("SIP/LYNC-00000012", "SIP/101,20,tr") in new stack
[Feb 15 10:37:53] VERBOSE[9452] netsock.c: == Using SIP RTP TOS bits 184
[Feb 15 10:37:53] VERBOSE[9452] netsock.c: == Using SIP RTP CoS mark 5
[Feb 15 10:37:53] VERBOSE[9452] chan_sip.c: Audio is at 2xx.xxx.1xx.xx1 port 10056
[Feb 15 10:37:53] VERBOSE[9452] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
[Feb 15 10:37:53] VERBOSE[9452] chan_sip.c: Adding codec 0x8 (alaw) to SDP
[Feb 15 10:37:53] VERBOSE[9452] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Feb 15 10:37:53] VERBOSE[9452] chan_sip.c: Reliably Transmitting (NAT) to 2yy.yyy.1yy.x5:5060:
INVITE sip:101@2yy.yyy.1yy.x5:5060;rinstance=d7d50749ac51fbee SIP/2.0
Via: SIP/2.0/UDP 2xx.xxx.1xx.xx1:5060;branch=z9hG4bK0e5c6c2e;rport
Max-Forwards: 70
From: "Artūras Rimonis" <sip:2001@2xx.xxx.1xx.xx1>;tag=as34a8831a
To: <sip:101@2yy.yyy.1yy.x5:5060;rinstance=d7d50749ac51fbee>
Contact: <sip:2001@2xx.xxx.1xx.xx1>
Call-ID:
3e411fd70af178cd24db2bc348ac7db1@2xx.xxx.1xx.xx1
CSeq: 102 INVITE
User-Agent: FPBX-2.7.0(1.6.2.16.1)
Date: Tue, 15 Feb 2011 08:37:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 265
v=0
o=root 31703178 31703178 IN IP4 2xx.xxx.1xx.xx1
s=Asterisk PBX 1.6.2.16.1
c=IN IP4 2xx.xxx.1xx.xx1
t=0 0
m=audio 10056 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
---
[Feb 15 10:37:53] VERBOSE[9452] app_dial.c: -- Called 101
[Feb 15 10:37:53] VERBOSE[2484] chan_sip.c:
<--- SIP read from UDP:2yy.yyy.1yy.x5:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 2xx.xxx.1xx.xx1:5060;branch=z9hG4bK0e5c6c2e;rport=5060
Contact: <sip:101@2yy.yyy.1yy.x5:5060;rinstance=d7d50749ac51fbee>
To: <sip:101@2yy.yyy.1yy.x5:5060;rinstance=d7d50749ac51fbee>;tag=7f79e84a
From: "Artūras Rimonis"<sip:2001@2xx.xxx.1xx.xx1>;tag=as34a8831a
Call-ID:
3e411fd70af178cd24db2bc348ac7db1@2xx.xxx.1xx.xx1
CSeq: 102 INVITE
User-Agent: 3CXPhone 5.0.14900.0
Content-Length: 0
<------------->
[Feb 15 10:37:53] VERBOSE[2484] chan_sip.c: --- (9 headers 0 lines) ---
[Feb 15 10:37:53] VERBOSE[9452] app_dial.c: -- SIP/101-00000013 is ringing
[Feb 15 10:37:57] VERBOSE[2484] chan_sip.c:
<--- SIP read from UDP:2yy.yyy.1yy.x5:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 2xx.xxx.1xx.xx1:5060;branch=z9hG4bK0e5c6c2e;rport=5060
Contact: <sip:101@2yy.yyy.1yy.x5:5060;rinstance=d7d50749ac51fbee>
To: <sip:101@2yy.yyy.1yy.x5:5060;rinstance=d7d50749ac51fbee>;tag=7f79e84a
From: "Artūras Rimonis"<sip:2001@2xx.xxx.1xx.xx1>;tag=as34a8831a
Call-ID:
3e411fd70af178cd24db2bc348ac7db1@2xx.xxx.1xx.xx1
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: 3CXPhone 5.0.14900.0
Content-Length: 237
v=0
o=3cxVCE 151176870 205173900 IN IP4 2yy.yyy.1yy.x5
s=3cxVCE Audio Call
c=IN IP4 2yy.yyy.1yy.x5
t=0 0
m=audio 10030 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
<------------->
[Feb 15 10:37:57] VERBOSE[2484] chan_sip.c: --- (12 headers 10 lines) ---
[Feb 15 10:37:57] VERBOSE[2484] chan_sip.c: Found RTP audio format 0
[Feb 15 10:37:57] VERBOSE[2484] chan_sip.c: Found RTP audio format 8
[Feb 15 10:37:57] VERBOSE[2484] chan_sip.c: Found RTP audio format 101
[Feb 15 10:37:57] VERBOSE[2484] chan_sip.c: Found audio description format PCMU for ID 0
[Feb 15 10:37:57] VERBOSE[2484] chan_sip.c: Found audio description format PCMA for ID 8
[Feb 15 10:37:57] VERBOSE[2484] chan_sip.c: Found audio description format telephone-event for ID 101
[Feb 15 10:37:57] VERBOSE[2484] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
[Feb 15 10:37:57] VERBOSE[2484] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Feb 15 10:37:57] VERBOSE[2484] chan_sip.c: Peer audio RTP is at port 2yy.yyy.1yy.x5:10030
[Feb 15 10:37:57] VERBOSE[2484] chan_sip.c: list_route: hop: <sip:101@2yy.yyy.1yy.x5:5060;rinstance=d7d50749ac51fbee>
[Feb 15 10:37:57] VERBOSE[2484] chan_sip.c: set_destination: Parsing <sip:101@2yy.yyy.1yy.x5:5060;rinstance=d7d50749ac51fbee> for address/port to send to
[Feb 15 10:37:57] VERBOSE[2484] chan_sip.c: set_destination: set destination to 2yy.yyy.1yy.x5, port 5060
[Feb 15 10:37:57] VERBOSE[2484] chan_sip.c: Transmitting (NAT) to 2yy.yyy.1yy.x5:5060:
ACK sip:101@2yy.yyy.1yy.x5:5060;rinstance=d7d50749ac51fbee SIP/2.0
Via: SIP/2.0/UDP 2xx.xxx.1xx.xx1:5060;branch=z9hG4bK2870a1ca;rport
Max-Forwards: 70
From: "Artūras Rimonis" <sip:2001@2xx.xxx.1xx.xx1>;tag=as34a8831a
To: <sip:101@2yy.yyy.1yy.x5:5060;rinstance=d7d50749ac51fbee>;tag=7f79e84a
Contact: <sip:2001@2xx.xxx.1xx.xx1>
Call-ID:
3e411fd70af178cd24db2bc348ac7db1@2xx.xxx.1xx.xx1
CSeq: 102 ACK
User-Agent: FPBX-2.7.0(1.6.2.16.1)
Content-Length: 0
---
[Feb 15 10:37:57] VERBOSE[9452] app_dial.c: -- SIP/101-00000013 answered SIP/LYNC-00000012
[Feb 15 10:38:00] VERBOSE[2484] chan_sip.c:
<--- SIP read from UDP:192.168.68.152:49890 --->
<------------->
[Feb 15 10:38:00] VERBOSE[2484] chan_sip.c: Really destroying SIP dialog 'N2I1MjI1ZmQxZTViY2ViNGFiOTMwNmU1Yzc5MTI2MWI.' Method: REGISTER
[Feb 15 10:38:04] VERBOSE[9451] chan_sip.c:
<--- SIP read from TCP:192.168.68.91:60257 --->
BYE sip:101@192.168.68.156;transport=TCP SIP/2.0
FROM: <sip:2001@fe01.mydomain.lt;user=phone>;epid=03259E13E5;tag=3bca2e7b83
TO: <sip:101@asterisknow01.mydomain.lt;user=phone>;tag=as10e2af07
CSEQ: 23320 BYE
CALL-ID: 843c3b3a-20a8-425a-9804-2668f9d2a1c9
MAX-FORWARDS: 70
VIA: SIP/2.0/TCP 192.168.68.91:60257;branch=z9hG4bK305b846a
CONTACT: <sip:fe01.mydomain.lt:5068;transport=Tcp;maddr=192.168.68.91>
CONTENT-LENGTH: 0
USER-AGENT: RTCC/4.0.0.0 MediationServer
<------------->
[Feb 15 10:38:04] VERBOSE[9451] chan_sip.c: --- (10 headers 0 lines) ---
[Feb 15 10:38:04] VERBOSE[9451] chan_sip.c: Sending to 192.168.68.91 : 60257 (NAT)
[Feb 15 10:38:04] VERBOSE[9451] chan_sip.c:
<--- Transmitting (NAT) to 192.168.68.91:60257 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 192.168.68.91:60257;branch=z9hG4bK305b846a;received=192.168.68.91
From: <sip:2001@fe01.mydomain.lt;user=phone>;epid=03259E13E5;tag=3bca2e7b83
To: <sip:101@asterisknow01.mydomain.lt;user=phone>;tag=as10e2af07
Call-ID: 843c3b3a-20a8-425a-9804-2668f9d2a1c9
CSeq: 23320 BYE
Server: FPBX-2.7.0(1.6.2.16.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
<------------>
[Feb 15 10:38:04] VERBOSE[9452] chan_sip.c: Scheduling destruction of SIP dialog
'3e411fd70af178cd24db2bc348ac7db1@2xx.xxx.1xx.xx1' in 8576 ms (Method: INVITE)
[Feb 15 10:38:04] VERBOSE[9452] chan_sip.c: set_destination: Parsing <sip:101@2yy.yyy.1yy.x5:5060;rinstance=d7d50749ac51fbee> for address/port to send to
[Feb 15 10:38:04] VERBOSE[9452] chan_sip.c: set_destination: set destination to 2yy.yyy.1yy.x5, port 5060
[Feb 15 10:38:04] VERBOSE[9452] chan_sip.c: Reliably Transmitting (NAT) to 2yy.yyy.1yy.x5:5060:
BYE sip:101@2yy.yyy.1yy.x5:5060;rinstance=d7d50749ac51fbee SIP/2.0
Via: SIP/2.0/UDP 2xx.xxx.1xx.xx1:5060;branch=z9hG4bK78a3367e;rport
Max-Forwards: 70
From: "Artūras Rimonis" <sip:2001@2xx.xxx.1xx.xx1>;tag=as34a8831a
To: <sip:101@2yy.yyy.1yy.x5:5060;rinstance=d7d50749ac51fbee>;tag=7f79e84a
Call-ID:
3e411fd70af178cd24db2bc348ac7db1@2xx.xxx.1xx.xx1
CSeq: 103 BYE
User-Agent: FPBX-2.7.0(1.6.2.16.1)
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
[Feb 15 10:38:04] VERBOSE[9452] pbx.c: == Spawn extension (from-lync, 101, 2) exited non-zero on 'SIP/LYNC-00000012'
[Feb 15 10:38:04] VERBOSE[2484] chan_sip.c:
<--- SIP read from UDP:2yy.yyy.1yy.x5:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 2xx.xxx.1xx.xx1:5060;branch=z9hG4bK78a3367e;rport=5060
Contact: <sip:101@2yy.yyy.1yy.x5:5060;rinstance=d7d50749ac51fbee>
To: <sip:101@2yy.yyy.1yy.x5:5060;rinstance=d7d50749ac51fbee>;tag=7f79e84a
From: "Artūras Rimonis"<sip:2001@2xx.xxx.1xx.xx1>;tag=as34a8831a
Call-ID:
3e411fd70af178cd24db2bc348ac7db1@2xx.xxx.1xx.xx1
CSeq: 103 BYE
User-Agent: 3CXPhone 5.0.14900.0
Content-Length: 0
<------------->
[Feb 15 10:38:04] VERBOSE[2484] chan_sip.c: --- (9 headers 0 lines) ---
[Feb 15 10:38:04] VERBOSE[2484] chan_sip.c: Really destroying SIP dialog '843c3b3a-20a8-425a-9804-2668f9d2a1c9' Method: BYE
[Feb 15 10:38:04] VERBOSE[2484] chan_sip.c: Really destroying SIP dialog
'3e411fd70af178cd24db2bc348ac7db1@2xx.xxx.1xx.xx1' Method: INVITE
[Feb 15 10:38:04] VERBOSE[2484] chan_sip.c:
<--- SIP read from UDP:2yy.yyy.1yy.x5:5060 --->
<------------->