
<--- SIP read from UDP:172.30.1.25:4415 --->
REGISTER sip:172.30.0.135:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.1.25:4415;branch=z9hG4bK-d8754z-ef71871059068151-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:20001@172.30.1.25:4415;rinstance=311f4d1534f84e89>
To: <sip:20001@172.30.0.135:5060>
From: <sip:20001@172.30.0.135:5060>;tag=3770b241
Call-ID: YTA0NzU1OTI4NjlmMTBlMDY1MDUzZTIyMjI5NDhlYWE.
CSeq: 1 REGISTER
Expires: 1800
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO
Supported: replaces
User-Agent: 3CXPhone 4.0.10858.0
Content-Length: 0


<------------->
localhost*CLI> --- (13 headers 0 lines) ---
localhost*CLI> Sending to 172.30.1.25 : 4415 (no NAT)
localhost*CLI> 
<--- Transmitting (NAT) to 172.30.1.25:4415 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.1.25:4415;branch=z9hG4bK-d8754z-ef71871059068151-1---d8754z-;received=172.30.1.25;rport=4415
From: <sip:20001@172.30.0.135:5060>;tag=3770b241
To: <sip:20001@172.30.0.135:5060>;tag=as72fdc88e
Call-ID: YTA0NzU1OTI4NjlmMTBlMDY1MDUzZTIyMjI5NDhlYWE.
CSeq: 1 REGISTER
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="17ab0506"
Content-Length: 0


<------------>
localhost*CLI> Scheduling destruction of SIP dialog 'YTA0NzU1OTI4NjlmMTBlMDY1MDUzZTIyMjI5NDhlYWE.' in 32000 ms (Method: REGISTER)
localhost*CLI> 
<--- SIP read from UDP:172.30.1.25:4415 --->
REGISTER sip:172.30.0.135:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.1.25:4415;branch=z9hG4bK-d8754z-454f5610f1147138-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:20001@172.30.1.25:4415;rinstance=311f4d1534f84e89>
To: <sip:20001@172.30.0.135:5060>
From: <sip:20001@172.30.0.135:5060>;tag=3770b241
Call-ID: YTA0NzU1OTI4NjlmMTBlMDY1MDUzZTIyMjI5NDhlYWE.
CSeq: 2 REGISTER
Expires: 1800
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO
Supported: replaces
User-Agent: 3CXPhone 4.0.10858.0
Authorization: Digest username="20001",realm="asterisk",nonce="17ab0506",uri="sip:172.30.0.135:5060",response="8ec06c943d465ae65ee7ca2a9e15d7c4",algorithm=MD5
Content-Length: 0


<------------->
localhost*CLI> --- (14 headers 0 lines) ---
localhost*CLI> Sending to 172.30.1.25 : 4415 (NAT)
localhost*CLI>     -- Registered SIP '20001' at 172.30.1.25 port 4415
localhost*CLI>        > Saved useragent "3CXPhone 4.0.10858.0" for peer 20001
localhost*CLI> 
<--- Transmitting (NAT) to 172.30.1.25:4415 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.1.25:4415;branch=z9hG4bK-d8754z-454f5610f1147138-1---d8754z-;received=172.30.1.25;rport=4415
From: <sip:20001@172.30.0.135:5060>;tag=3770b241
To: <sip:20001@172.30.0.135:5060>;tag=as72fdc88e
Call-ID: YTA0NzU1OTI4NjlmMTBlMDY1MDUzZTIyMjI5NDhlYWE.
CSeq: 2 REGISTER
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 1800
Contact: <sip:20001@172.30.1.25:4415;rinstance=311f4d1534f84e89>;expires=1800
Date: Mon, 31 Jan 2011 08:02:18 GMT
Content-Length: 0


<------------>
localhost*CLI> Scheduling destruction of SIP dialog 'YTA0NzU1OTI4NjlmMTBlMDY1MDUzZTIyMjI5NDhlYWE.' in 32000 ms (Method: REGISTER)
localhost*CLI> 
<--- SIP read from UDP:172.30.1.25:4415 --->
SUBSCRIBE sip:20001@172.30.0.135:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 172.30.1.25:4415;branch=z9hG4bK-d8754z-f5031c0d1e1b7d6a-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:20001@172.30.1.25:4415;rinstance=311f4d1534f84e89>
To: <sip:20001@172.30.0.135:5060>
From: <sip:20001@172.30.0.135:5060>;tag=9a569959
Call-ID: MWY5NGY1Zjc2NjViYWRmY2QyNjQxZmM5N2ViMzViOGE.
CSeq: 1 SUBSCRIBE
Expires: 120
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO
Supported: replaces
User-Agent: 3CXPhone 4.0.10858.0
Event: message-summary
Content-Length: 0


<------------->
localhost*CLI> --- (14 headers 0 lines) ---
localhost*CLI> Creating new subscription
localhost*CLI> Sending to 172.30.1.25 : 4415 (no NAT)
localhost*CLI> list_route: hop: <sip:20001@172.30.1.25:4415;rinstance=311f4d1534f84e89>
localhost*CLI> Found peer '20001' for '20001' from 172.30.1.25:4415
localhost*CLI> 
<--- Transmitting (NAT) to 172.30.1.25:4415 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.1.25:4415;branch=z9hG4bK-d8754z-f5031c0d1e1b7d6a-1---d8754z-;received=172.30.1.25;rport=4415
From: <sip:20001@172.30.0.135:5060>;tag=9a569959
To: <sip:20001@172.30.0.135:5060>;tag=as76930b7d
Call-ID: MWY5NGY1Zjc2NjViYWRmY2QyNjQxZmM5N2ViMzViOGE.
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="04ad7a78"
Content-Length: 0


<------------>
localhost*CLI> Scheduling destruction of SIP dialog 'MWY5NGY1Zjc2NjViYWRmY2QyNjQxZmM5N2ViMzViOGE.' in 32000 ms (Method: SUBSCRIBE)
localhost*CLI> 
<--- SIP read from UDP:172.30.1.25:4415 --->
SUBSCRIBE sip:20001@172.30.0.135:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 172.30.1.25:4415;branch=z9hG4bK-d8754z-b360414ca507fe18-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:20001@172.30.1.25:4415;rinstance=311f4d1534f84e89>
To: <sip:20001@172.30.0.135:5060>
From: <sip:20001@172.30.0.135:5060>;tag=9a569959
Call-ID: MWY5NGY1Zjc2NjViYWRmY2QyNjQxZmM5N2ViMzViOGE.
CSeq: 2 SUBSCRIBE
Expires: 120
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO
Supported: replaces
User-Agent: 3CXPhone 4.0.10858.0
Authorization: Digest username="20001",realm="asterisk",nonce="04ad7a78",uri="sip:20001@172.30.0.135:5060;transport=UDP",response="9d2599e00811bd553be7e57267de96d2",algorithm=MD5
Event: message-summary
Content-Length: 0


<------------->
localhost*CLI> --- (15 headers 0 lines) ---
localhost*CLI> Creating new subscription
localhost*CLI> Sending to 172.30.1.25 : 4415 (NAT)
localhost*CLI> Found peer '20001' for '20001' from 172.30.1.25:4415
localhost*CLI> 
<--- Transmitting (NAT) to 172.30.1.25:4415 --->
SIP/2.0 404 Not found (no mailbox)
Via: SIP/2.0/UDP 172.30.1.25:4415;branch=z9hG4bK-d8754z-b360414ca507fe18-1---d8754z-;received=172.30.1.25;rport=4415
From: <sip:20001@172.30.0.135:5060>;tag=9a569959
To: <sip:20001@172.30.0.135:5060>;tag=as76930b7d
Call-ID: MWY5NGY1Zjc2NjViYWRmY2QyNjQxZmM5N2ViMzViOGE.
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
localhost*CLI> [Jan 31 17:02:18] NOTICE[27653]: chan_sip.c:21476 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 20001
localhost*CLI> Really destroying SIP dialog 'MWY5NGY1Zjc2NjViYWRmY2QyNjQxZmM5N2ViMzViOGE.' Method: SUBSCRIBE
localhost*CLI> 
<--- SIP read from UDP:172.30.0.55:2263 --->
REGISTER sip:172.30.0.135:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.55:2263;branch=z9hG4bK-d8754z-1b45073b9950755e-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:20000@172.30.0.55:2263;rinstance=b4798b847c64aa89>
To: "3CX VoIP Phone"<sip:20000@172.30.0.135:5060>
From: "3CX VoIP Phone"<sip:20000@172.30.0.135:5060>;tag=d96a2c6e
Call-ID: MWY1OTYyMzg5MDA5NWMwNTExYjc4ZGU3YzU4ZjI3M2E.
CSeq: 1 REGISTER
Expires: 1800
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO
Supported: replaces
User-Agent: 3CXPhone 4.0.10858.0
Content-Length: 0


<------------->
localhost*CLI> --- (13 headers 0 lines) ---
localhost*CLI> Sending to 172.30.0.55 : 2263 (no NAT)
localhost*CLI> 
<--- Transmitting (NAT) to 172.30.0.55:2263 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.55:2263;branch=z9hG4bK-d8754z-1b45073b9950755e-1---d8754z-;received=172.30.0.55;rport=2263
From: "3CX VoIP Phone"<sip:20000@172.30.0.135:5060>;tag=d96a2c6e
To: "3CX VoIP Phone"<sip:20000@172.30.0.135:5060>;tag=as0bf4a9a6
Call-ID: MWY1OTYyMzg5MDA5NWMwNTExYjc4ZGU3YzU4ZjI3M2E.
CSeq: 1 REGISTER
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="015fc30e"
Content-Length: 0


<------------>
localhost*CLI> Scheduling destruction of SIP dialog 'MWY1OTYyMzg5MDA5NWMwNTExYjc4ZGU3YzU4ZjI3M2E.' in 32000 ms (Method: REGISTER)
localhost*CLI> 
<--- SIP read from UDP:172.30.0.55:2263 --->
REGISTER sip:172.30.0.135:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.55:2263;branch=z9hG4bK-d8754z-ec58c45024762f41-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:20000@172.30.0.55:2263;rinstance=b4798b847c64aa89>
To: "3CX VoIP Phone"<sip:20000@172.30.0.135:5060>
From: "3CX VoIP Phone"<sip:20000@172.30.0.135:5060>;tag=d96a2c6e
Call-ID: MWY1OTYyMzg5MDA5NWMwNTExYjc4ZGU3YzU4ZjI3M2E.
CSeq: 2 REGISTER
Expires: 1800
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO
Supported: replaces
User-Agent: 3CXPhone 4.0.10858.0
Authorization: Digest username="20000",realm="asterisk",nonce="015fc30e",uri="sip:172.30.0.135:5060",response="1b625659a9605a83f6b037b2d5be39bb",algorithm=MD5
Content-Length: 0


<------------->
localhost*CLI> --- (14 headers 0 lines) ---
localhost*CLI> Sending to 172.30.0.55 : 2263 (NAT)
localhost*CLI>     -- Registered SIP '20000' at 172.30.0.55 port 2263
localhost*CLI>        > Saved useragent "3CXPhone 4.0.10858.0" for peer 20000
localhost*CLI> 
<--- Transmitting (NAT) to 172.30.0.55:2263 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.55:2263;branch=z9hG4bK-d8754z-ec58c45024762f41-1---d8754z-;received=172.30.0.55;rport=2263
From: "3CX VoIP Phone"<sip:20000@172.30.0.135:5060>;tag=d96a2c6e
To: "3CX VoIP Phone"<sip:20000@172.30.0.135:5060>;tag=as0bf4a9a6
Call-ID: MWY1OTYyMzg5MDA5NWMwNTExYjc4ZGU3YzU4ZjI3M2E.
CSeq: 2 REGISTER
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 1800
Contact: <sip:20000@172.30.0.55:2263;rinstance=b4798b847c64aa89>;expires=1800
Date: Mon, 31 Jan 2011 08:02:24 GMT
Content-Length: 0


<------------>
localhost*CLI> Scheduling destruction of SIP dialog 'MWY1OTYyMzg5MDA5NWMwNTExYjc4ZGU3YzU4ZjI3M2E.' in 32000 ms (Method: REGISTER)
localhost*CLI> 
<--- SIP read from UDP:172.30.0.55:2263 --->
SUBSCRIBE sip:20000@172.30.0.135:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 172.30.0.55:2263;branch=z9hG4bK-d8754z-13416316e253c402-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:20000@172.30.0.55:2263;rinstance=b4798b847c64aa89>
To: "3CX VoIP Phone"<sip:20000@172.30.0.135:5060>
From: "3CX VoIP Phone"<sip:20000@172.30.0.135:5060>;tag=8d60397e
Call-ID: ZjRkM2JjNzg2YWM4NTYxMGNmMjZjYmJmMzgyZWUzMmU.
CSeq: 1 SUBSCRIBE
Expires: 120
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO
Supported: replaces
User-Agent: 3CXPhone 4.0.10858.0
Event: message-summary
Content-Length: 0


<------------->
--- (14 headers 0 lines) ---
Creating new subscription
Sending to 172.30.0.55 : 2263 (no NAT)
list_route: hop: <sip:20000@172.30.0.55:2263;rinstance=b4798b847c64aa89>
Found peer '20000' for '20000' from 172.30.0.55:2263

<--- Transmitting (NAT) to 172.30.0.55:2263 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.30.0.55:2263;branch=z9hG4bK-d8754z-13416316e253c402-1---d8754z-;received=172.30.0.55;rport=2263
From: "3CX VoIP Phone"<sip:20000@172.30.0.135:5060>;tag=8d60397e
To: "3CX VoIP Phone"<sip:20000@172.30.0.135:5060>;tag=as2574d55e
Call-ID: ZjRkM2JjNzg2YWM4NTYxMGNmMjZjYmJmMzgyZWUzMmU.
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="29773465"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'ZjRkM2JjNzg2YWM4NTYxMGNmMjZjYmJmMzgyZWUzMmU.' in 32000 ms (Method: SUBSCRIBE)
localhost*CLI> 
<--- SIP read from UDP:172.30.0.55:2263 --->
SUBSCRIBE sip:20000@172.30.0.135:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 172.30.0.55:2263;branch=z9hG4bK-d8754z-e159a94bf478c269-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:20000@172.30.0.55:2263;rinstance=b4798b847c64aa89>
To: "3CX VoIP Phone"<sip:20000@172.30.0.135:5060>
From: "3CX VoIP Phone"<sip:20000@172.30.0.135:5060>;tag=8d60397e
Call-ID: ZjRkM2JjNzg2YWM4NTYxMGNmMjZjYmJmMzgyZWUzMmU.
CSeq: 2 SUBSCRIBE
Expires: 120
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO
Supported: replaces
User-Agent: 3CXPhone 4.0.10858.0
Authorization: Digest username="20000",realm="asterisk",nonce="29773465",uri="sip:20000@172.30.0.135:5060;transport=UDP",response="b8011c6e01aee9e6390f9dc7b0ebb340",algorithm=MD5
Event: message-summary
Content-Length: 0


<------------->
--- (15 headers 0 lines) ---
Creating new subscription
Sending to 172.30.0.55 : 2263 (NAT)
Found peer '20000' for '20000' from 172.30.0.55:2263

<--- Transmitting (NAT) to 172.30.0.55:2263 --->
SIP/2.0 404 Not found (no mailbox)
Via: SIP/2.0/UDP 172.30.0.55:2263;branch=z9hG4bK-d8754z-e159a94bf478c269-1---d8754z-;received=172.30.0.55;rport=2263
From: "3CX VoIP Phone"<sip:20000@172.30.0.135:5060>;tag=8d60397e
To: "3CX VoIP Phone"<sip:20000@172.30.0.135:5060>;tag=as2574d55e
Call-ID: ZjRkM2JjNzg2YWM4NTYxMGNmMjZjYmJmMzgyZWUzMmU.
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
[Jan 31 17:02:24] NOTICE[27653]: chan_sip.c:21476 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 20000
Really destroying SIP dialog 'ZjRkM2JjNzg2YWM4NTYxMGNmMjZjYmJmMzgyZWUzMmU.' Method: SUBSCRIBE
localhost*CLI> 
<--- SIP read from UDP:172.30.1.25:4415 --->



<------------->
localhost*CLI> Really destroying SIP dialog 'YTA0NzU1OTI4NjlmMTBlMDY1MDUzZTIyMjI5NDhlYWE.' Method: REGISTER
localhost*CLI> 
<--- SIP read from UDP:172.30.0.55:2263 --->



<------------->
localhost*CLI> Really destroying SIP dialog 'MWY1OTYyMzg5MDA5NWMwNTExYjc4ZGU3YzU4ZjI3M2E.' Method: REGISTER
localhost*CLI> 
<--- SIP read from UDP:172.30.1.25:4415 --->



<------------->
localhost*CLI> 
<--- SIP read from UDP:172.30.0.55:2263 --->



<------------->
localhost*CLI>   == Using SIP RTP CoS mark 5
localhost*CLI> Audio is at 172.30.0.135 port 12368
localhost*CLI> Adding codec 0x4 (ulaw) to SDP
localhost*CLI> Adding non-codec 0x1 (telephone-event) to SDP
localhost*CLI> Reliably Transmitting (NAT) to 172.30.0.55:2263:
INVITE sip:20000@172.30.0.55:2263;rinstance=b4798b847c64aa89 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.135:5060;branch=z9hG4bK0769cacb;rport
Max-Forwards: 70
From: "20000" <sip:20000@172.30.0.135>;tag=as2e0eb0a5
To: <sip:20000@172.30.0.55:2263;rinstance=b4798b847c64aa89>
Contact: <sip:20000@172.30.0.135>
Call-ID: 49a55c811df68ca96d333a8a2eae9978@172.30.0.135
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.10
Date: Mon, 31 Jan 2011 08:03:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 264

v=0
o=root 1888806412 1888806412 IN IP4 172.30.0.135
s=Asterisk PBX 1.6.2.10
c=IN IP4 172.30.0.135
t=0 0
m=audio 12368 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
localhost*CLI> 
<--- SIP read from UDP:172.30.0.55:2263 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.30.0.135:5060;branch=z9hG4bK0769cacb;rport=5060
Contact: <sip:20000@172.30.0.55:2263;rinstance=b4798b847c64aa89>
To: <sip:20000@172.30.0.55:2263;rinstance=b4798b847c64aa89>;tag=6f492b16
From: "20000"<sip:20000@172.30.0.135>;tag=as2e0eb0a5
Call-ID: 49a55c811df68ca96d333a8a2eae9978@172.30.0.135
CSeq: 102 INVITE
User-Agent: 3CXPhone 4.0.10858.0
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
localhost*CLI> 
<--- SIP read from UDP:172.30.1.25:4415 --->



<------------->
localhost*CLI> 
<--- SIP read from UDP:172.30.0.55:2263 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.135:5060;branch=z9hG4bK0769cacb;rport=5060
Contact: <sip:20000@172.30.0.55:2263;rinstance=b4798b847c64aa89>
To: <sip:20000@172.30.0.55:2263;rinstance=b4798b847c64aa89>;tag=6f492b16
From: "20000"<sip:20000@172.30.0.135>;tag=as2e0eb0a5
Call-ID: 49a55c811df68ca96d333a8a2eae9978@172.30.0.135
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: 3CXPhone 4.0.10858.0
Content-Length: 207

v=0
o=3cxVCE 158361660 246825930 IN IP4 172.30.0.55
s=3cxVCE Audio Call
c=IN IP4 172.30.0.55
t=0 0
m=audio 40002 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

<------------->
--- (12 headers 9 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 - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 172.30.0.55:40002
list_route: hop: <sip:20000@172.30.0.55:2263;rinstance=b4798b847c64aa89>
set_destination: Parsing <sip:20000@172.30.0.55:2263;rinstance=b4798b847c64aa89> for address/port to send to
set_destination: set destination to 172.30.0.55, port 2263
Transmitting (NAT) to 172.30.0.55:2263:
ACK sip:20000@172.30.0.55:2263;rinstance=b4798b847c64aa89 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.135:5060;branch=z9hG4bK18bc0d1b;rport
Max-Forwards: 70
From: "20000" <sip:20000@172.30.0.135>;tag=as2e0eb0a5
To: <sip:20000@172.30.0.55:2263;rinstance=b4798b847c64aa89>;tag=6f492b16
Contact: <sip:20000@172.30.0.135>
Call-ID: 49a55c811df68ca96d333a8a2eae9978@172.30.0.135
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.10
Content-Length: 0


---
localhost*CLI>        > Channel SIP/20000-00000010 was answered.
localhost*CLI>     -- Executing [20001@tenant01:1] Dial("SIP/20000-00000010", "SIP/20001,60") in new stack
localhost*CLI>   == Using SIP RTP CoS mark 5
localhost*CLI> Audio is at 172.30.0.135 port 18082
localhost*CLI> Adding codec 0x4 (ulaw) to SDP
localhost*CLI> Adding non-codec 0x1 (telephone-event) to SDP
localhost*CLI> Reliably Transmitting (NAT) to 172.30.1.25:4415:
INVITE sip:20001@172.30.1.25:4415;rinstance=311f4d1534f84e89 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.135:5060;branch=z9hG4bK2c7d0415;rport
Max-Forwards: 70
From: "20000" <sip:20000@172.30.0.135>;tag=as448aed01
To: <sip:20001@172.30.1.25:4415;rinstance=311f4d1534f84e89>
Contact: <sip:20000@172.30.0.135>
Call-ID: 5a4442797e5af13a2fdaa74a71663b21@172.30.0.135
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.10
Date: Mon, 31 Jan 2011 08:03:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 264

v=0
o=root 1184516194 1184516194 IN IP4 172.30.0.135
s=Asterisk PBX 1.6.2.10
c=IN IP4 172.30.0.135
t=0 0
m=audio 18082 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
localhost*CLI>     -- Called 20001
localhost*CLI> 
<--- SIP read from UDP:172.30.1.25:4415 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.30.0.135:5060;branch=z9hG4bK2c7d0415;rport=5060
Contact: <sip:20001@172.30.1.25:4415;rinstance=311f4d1534f84e89>
To: <sip:20001@172.30.1.25:4415;rinstance=311f4d1534f84e89>;tag=21289b08
From: "20000"<sip:20000@172.30.0.135>;tag=as448aed01
Call-ID: 5a4442797e5af13a2fdaa74a71663b21@172.30.0.135
CSeq: 102 INVITE
User-Agent: 3CXPhone 4.0.10858.0
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
localhost*CLI>     -- SIP/20001-00000011 is ringing
localhost*CLI> 
<--- SIP read from UDP:172.30.0.55:2263 --->
BYE sip:20000@172.30.0.135 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.55:2263;branch=z9hG4bK-d8754z-ff1976683549fb02-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:20000@172.30.0.55:2263;rinstance=b4798b847c64aa89>
To: "20000"<sip:20000@172.30.0.135>;tag=as2e0eb0a5
From: <sip:20000@172.30.0.55:2263;rinstance=b4798b847c64aa89>;tag=6f492b16
Call-ID: 49a55c811df68ca96d333a8a2eae9978@172.30.0.135
CSeq: 2 BYE
User-Agent: 3CXPhone 4.0.10858.0
Reason: SIP;description="User Hung Up"
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Sending to 172.30.0.55 : 2263 (NAT)

<--- Transmitting (NAT) to 172.30.0.55:2263 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.55:2263;branch=z9hG4bK-d8754z-ff1976683549fb02-1---d8754z-;received=172.30.0.55;rport=2263
From: <sip:20000@172.30.0.55:2263;rinstance=b4798b847c64aa89>;tag=6f492b16
To: "20000"<sip:20000@172.30.0.135>;tag=as2e0eb0a5
Call-ID: 49a55c811df68ca96d333a8a2eae9978@172.30.0.135
CSeq: 2 BYE
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
localhost*CLI> Scheduling destruction of SIP dialog '5a4442797e5af13a2fdaa74a71663b21@172.30.0.135' in 32000 ms (Method: INVITE)
Reliably Transmitting (NAT) to 172.30.1.25:4415:
CANCEL sip:20001@172.30.1.25:4415;rinstance=311f4d1534f84e89 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.135:5060;branch=z9hG4bK2c7d0415;rport
Max-Forwards: 70
From: "20000" <sip:20000@172.30.0.135>;tag=as448aed01
To: <sip:20001@172.30.1.25:4415;rinstance=311f4d1534f84e89>
Call-ID: 5a4442797e5af13a2fdaa74a71663b21@172.30.0.135
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 1.6.2.10
Content-Length: 0


---
Scheduling destruction of SIP dialog '5a4442797e5af13a2fdaa74a71663b21@172.30.0.135' in 32000 ms (Method: INVITE)
  == Spawn extension (tenant01, 20001, 1) exited non-zero on 'SIP/20000-00000010'
localhost*CLI> 
<--- SIP read from UDP:172.30.1.25:4415 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.135:5060;branch=z9hG4bK2c7d0415;rport=5060
Contact: <sip:20001@172.30.1.25:4415;rinstance=311f4d1534f84e89>
To: <sip:20001@172.30.1.25:4415;rinstance=311f4d1534f84e89>;tag=21289b08
From: "20000"<sip:20000@172.30.0.135>;tag=as448aed01
Call-ID: 5a4442797e5af13a2fdaa74a71663b21@172.30.0.135
CSeq: 102 CANCEL
User-Agent: 3CXPhone 4.0.10858.0
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '49a55c811df68ca96d333a8a2eae9978@172.30.0.135' Method: BYE

<--- SIP read from UDP:172.30.1.25:4415 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 172.30.0.135:5060;branch=z9hG4bK2c7d0415;rport=5060
To: <sip:20001@172.30.1.25:4415;rinstance=311f4d1534f84e89>;tag=21289b08
From: "20000"<sip:20000@172.30.0.135>;tag=as448aed01
Call-ID: 5a4442797e5af13a2fdaa74a71663b21@172.30.0.135
CSeq: 102 INVITE
User-Agent: 3CXPhone 4.0.10858.0
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Transmitting (NAT) to 172.30.1.25:4415:
ACK sip:20001@172.30.1.25:4415;rinstance=311f4d1534f84e89 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.135:5060;branch=z9hG4bK2c7d0415;rport
Max-Forwards: 70
From: "20000" <sip:20000@172.30.0.135>;tag=as448aed01
To: <sip:20001@172.30.1.25:4415;rinstance=311f4d1534f84e89>;tag=21289b08
Contact: <sip:20000@172.30.0.135>
Call-ID: 5a4442797e5af13a2fdaa74a71663b21@172.30.0.135
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.10
Content-Length: 0


---
Rlocalhost*CLI> eally destroying SIP dialog '5a4442797e5af13a2fdaa74a71663b21@172.30.0.135' Method: INVITE
localhost*CLI> 
<--- SIP read from UDP:172.30.0.55:2263 --->



<------------->
localhost*CLI>   == Using SIP RTP CoS mark 5
localhost*CLI> Audio is at 172.30.0.135 port 12882
localhost*CLI> Adding codec 0x4 (ulaw) to SDP
localhost*CLI> Adding non-codec 0x1 (telephone-event) to SDP
localhost*CLI> Reliably Transmitting (NAT) to 172.30.0.55:2263:
INVITE sip:20000@172.30.0.55:2263;rinstance=b4798b847c64aa89 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.135:5060;branch=z9hG4bK723179ef;rport
Max-Forwards: 70
From: "20000" <sip:20000@172.30.0.135>;tag=as32365614
To: <sip:20000@172.30.0.55:2263;rinstance=b4798b847c64aa89>
Contact: <sip:20000@172.30.0.135>
Call-ID: 210fb10f60a5661c4a1d47605f95f261@172.30.0.135
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.10
Date: Mon, 31 Jan 2011 08:04:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 911803435 911803435 IN IP4 172.30.0.135
s=Asterisk PBX 1.6.2.10
c=IN IP4 172.30.0.135
t=0 0
m=audio 12882 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
localhost*CLI> 
<--- SIP read from UDP:172.30.0.55:2263 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.30.0.135:5060;branch=z9hG4bK723179ef;rport=5060
Contact: <sip:20000@172.30.0.55:2263;rinstance=b4798b847c64aa89>
To: <sip:20000@172.30.0.55:2263;rinstance=b4798b847c64aa89>;tag=a074f151
From: "20000"<sip:20000@172.30.0.135>;tag=as32365614
Call-ID: 210fb10f60a5661c4a1d47605f95f261@172.30.0.135
CSeq: 102 INVITE
User-Agent: 3CXPhone 4.0.10858.0
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
localhost*CLI> 
<--- SIP read from UDP:172.30.0.55:2263 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.135:5060;branch=z9hG4bK723179ef;rport=5060
Contact: <sip:20000@172.30.0.55:2263;rinstance=b4798b847c64aa89>
To: <sip:20000@172.30.0.55:2263;rinstance=b4798b847c64aa89>;tag=a074f151
From: "20000"<sip:20000@172.30.0.135>;tag=as32365614
Call-ID: 210fb10f60a5661c4a1d47605f95f261@172.30.0.135
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: 3CXPhone 4.0.10858.0
Content-Length: 204

v=0
o=3cxVCE 17752110 7752660 IN IP4 172.30.0.55
s=3cxVCE Audio Call
c=IN IP4 172.30.0.55
t=0 0
m=audio 40004 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

<------------->
--- (12 headers 9 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 - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 172.30.0.55:40004
list_route: hop: <sip:20000@172.30.0.55:2263;rinstance=b4798b847c64aa89>
set_destination: Parsing <sip:20000@172.30.0.55:2263;rinstance=b4798b847c64aa89> for address/port to send to
set_destination: set destination to 172.30.0.55, port 2263
Transmitting (NAT) to 172.30.0.55:2263:
ACK sip:20000@172.30.0.55:2263;rinstance=b4798b847c64aa89 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.135:5060;branch=z9hG4bK403edab8;rport
Max-Forwards: 70
From: "20000" <sip:20000@172.30.0.135>;tag=as32365614
To: <sip:20000@172.30.0.55:2263;rinstance=b4798b847c64aa89>;tag=a074f151
Contact: <sip:20000@172.30.0.135>
Call-ID: 210fb10f60a5661c4a1d47605f95f261@172.30.0.135
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.10
Content-Length: 0


---
localhost*CLI>        > Channel SIP/20000-00000012 was answered.
localhost*CLI>     -- Executing [20001@tenant01:1] Dial("SIP/20000-00000012", "SIP/20001,60") in new stack
localhost*CLI>   == Using SIP RTP CoS mark 5
localhost*CLI> Audio is at 172.30.0.135 port 13294
localhost*CLI> Adding codec 0x4 (ulaw) to SDP
localhost*CLI> Adding non-codec 0x1 (telephone-event) to SDP
localhost*CLI> Reliably Transmitting (NAT) to 172.30.1.25:4415:
INVITE sip:20001@172.30.1.25:4415;rinstance=311f4d1534f84e89 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.135:5060;branch=z9hG4bK7aea1c27;rport
Max-Forwards: 70
From: "20000" <sip:20000@172.30.0.135>;tag=as4e760b43
To: <sip:20001@172.30.1.25:4415;rinstance=311f4d1534f84e89>
Contact: <sip:20000@172.30.0.135>
Call-ID: 6dc4686c77d8c0fe03a9afae4dcad221@172.30.0.135
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.10
Date: Mon, 31 Jan 2011 08:04:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 338239026 338239026 IN IP4 172.30.0.135
s=Asterisk PBX 1.6.2.10
c=IN IP4 172.30.0.135
t=0 0
m=audio 13294 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
localhost*CLI>     -- Called 20001
localhost*CLI> 
<--- SIP read from UDP:172.30.1.25:4415 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.30.0.135:5060;branch=z9hG4bK7aea1c27;rport=5060
Contact: <sip:20001@172.30.1.25:4415;rinstance=311f4d1534f84e89>
To: <sip:20001@172.30.1.25:4415;rinstance=311f4d1534f84e89>;tag=437d7d60
From: "20000"<sip:20000@172.30.0.135>;tag=as4e760b43
Call-ID: 6dc4686c77d8c0fe03a9afae4dcad221@172.30.0.135
CSeq: 102 INVITE
User-Agent: 3CXPhone 4.0.10858.0
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
localhost*CLI>     -- SIP/20001-00000013 is ringing
localhost*CLI> 
<--- SIP read from UDP:172.30.1.25:4415 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.135:5060;branch=z9hG4bK7aea1c27;rport=5060
Contact: <sip:20001@172.30.1.25:4415;rinstance=311f4d1534f84e89>
To: <sip:20001@172.30.1.25:4415;rinstance=311f4d1534f84e89>;tag=437d7d60
From: "20000"<sip:20000@172.30.0.135>;tag=as4e760b43
Call-ID: 6dc4686c77d8c0fe03a9afae4dcad221@172.30.0.135
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: 3CXPhone 4.0.10858.0
Content-Length: 205

v=0
o=3cxVCE 60873195 68625855 IN IP4 172.30.1.25
s=3cxVCE Audio Call
c=IN IP4 172.30.1.25
t=0 0
m=audio 40010 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

<------------->
--- (12 headers 9 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 - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 172.30.1.25:40010
list_route: hop: <sip:20001@172.30.1.25:4415;rinstance=311f4d1534f84e89>
set_destination: Parsing <sip:20001@172.30.1.25:4415;rinstance=311f4d1534f84e89> for address/port to send to
set_destination: set destination to 172.30.1.25, port 4415
Transmitting (NAT) to 172.30.1.25:4415:
ACK sip:20001@172.30.1.25:4415;rinstance=311f4d1534f84e89 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.135:5060;branch=z9hG4bK378cdc80;rport
Max-Forwards: 70
From: "20000" <sip:20000@172.30.0.135>;tag=as4e760b43
To: <sip:20001@172.30.1.25:4415;rinstance=311f4d1534f84e89>;tag=437d7d60
Contact: <sip:20000@172.30.0.135>
Call-ID: 6dc4686c77d8c0fe03a9afae4dcad221@172.30.0.135
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.10
Content-Length: 0


---
localhost*CLI>     -- SIP/20001-00000013 answered SIP/20000-00000012
    -- Native bridging SIP/20000-00000012 and SIP/20001-00000013
set_destination: Parsing <sip:20000@172.30.0.55:2263;rinstance=b4798b847c64aa89> for address/port to send to
set_destination: set destination to 172.30.0.55, port 2263
Audio is at 172.30.0.135 port 12882
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 172.30.0.55:2263:
INVITE sip:20000@172.30.0.55:2263;rinstance=b4798b847c64aa89 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.135:5060;branch=z9hG4bK642baeb7;rport
Max-Forwards: 70
From: "20000" <sip:20000@172.30.0.135>;tag=as32365614
To: <sip:20000@172.30.0.55:2263;rinstance=b4798b847c64aa89>;tag=a074f151
Contact: <sip:20000@172.30.0.135>
Call-ID: 210fb10f60a5661c4a1d47605f95f261@172.30.0.135
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 911803435 911803436 IN IP4 172.30.1.25
s=Asterisk PBX 1.6.2.10
c=IN IP4 172.30.1.25
t=0 0
m=audio 40010 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
set_destination: Parsing <sip:20001@172.30.1.25:4415;rinstance=311f4d1534f84e89> for address/port to send to
set_destination: set destination to 172.30.1.25, port 4415
Audio is at 172.30.0.135 port 13294
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 172.30.1.25:4415:
INVITE sip:20001@172.30.1.25:4415;rinstance=311f4d1534f84e89 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.135:5060;branch=z9hG4bK4b300cf9;rport
Max-Forwards: 70
From: "20000" <sip:20000@172.30.0.135>;tag=as4e760b43
To: <sip:20001@172.30.1.25:4415;rinstance=311f4d1534f84e89>;tag=437d7d60
Contact: <sip:20000@172.30.0.135>
Call-ID: 6dc4686c77d8c0fe03a9afae4dcad221@172.30.0.135
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 338239026 338239027 IN IP4 172.30.0.55
s=Asterisk PBX 1.6.2.10
c=IN IP4 172.30.0.55
t=0 0
m=audio 40004 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
localhost*CLI> 
<--- SIP read from UDP:172.30.0.55:2263 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.135:5060;branch=z9hG4bK642baeb7;rport=5060
Contact: <sip:20000@172.30.0.55:2263;rinstance=b4798b847c64aa89>
To: <sip:20000@172.30.0.55:2263;rinstance=b4798b847c64aa89>;tag=a074f151
From: "20000"<sip:20000@172.30.0.135>;tag=as32365614
Call-ID: 210fb10f60a5661c4a1d47605f95f261@172.30.0.135
CSeq: 103 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: 3CXPhone 4.0.10858.0
Content-Length: 204

v=0
o=3cxVCE 17752110 7752661 IN IP4 172.30.0.55
s=3cxVCE Audio Call
c=IN IP4 172.30.0.55
t=0 0
m=audio 40004 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

<------------->
--- (12 headers 9 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 - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 172.30.0.55:40004
set_destination: Parsing <sip:20000@172.30.0.55:2263;rinstance=b4798b847c64aa89> for address/port to send to
set_destination: set destination to 172.30.0.55, port 2263
Transmitting (NAT) to 172.30.0.55:2263:
ACK sip:20000@172.30.0.55:2263;rinstance=b4798b847c64aa89 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.135:5060;branch=z9hG4bK69625b22;rport
Max-Forwards: 70
From: "20000" <sip:20000@172.30.0.135>;tag=as32365614
To: <sip:20000@172.30.0.55:2263;rinstance=b4798b847c64aa89>;tag=a074f151
Contact: <sip:20000@172.30.0.135>
Call-ID: 210fb10f60a5661c4a1d47605f95f261@172.30.0.135
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.6.2.10
Content-Length: 0


---

<--- SIP read from UDP:172.30.1.25:4415 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.135:5060;branch=z9hG4bK4b300cf9;rport=5060
Contact: <sip:20001@172.30.1.25:4415;rinstance=311f4d1534f84e89>
To: <sip:20001@172.30.1.25:4415;rinstance=311f4d1534f84e89>;tag=437d7d60
From: "20000"<sip:20000@172.30.0.135>;tag=as4e760b43
Call-ID: 6dc4686c77d8c0fe03a9afae4dcad221@172.30.0.135
CSeq: 103 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: 3CXPhone 4.0.10858.0
Content-Length: 205

v=0
o=3cxVCE 60873195 68625856 IN IP4 172.30.1.25
s=3cxVCE Audio Call
c=IN IP4 172.30.1.25
t=0 0
m=audio 40010 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

<------------->
--- (12 headers 9 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 - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 172.30.1.25:40010
set_destination: Parsing <sip:20001@172.30.1.25:4415;rinstance=311f4d1534f84e89> for address/port to send to
set_destination: set destination to 172.30.1.25, port 4415
Transmitting (NAT) to 172.30.1.25:4415:
ACK sip:20001@172.30.1.25:4415;rinstance=311f4d1534f84e89 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.135:5060;branch=z9hG4bK023a5ec0;rport
Max-Forwards: 70
From: "20000" <sip:20000@172.30.0.135>;tag=as4e760b43
To: <sip:20001@172.30.1.25:4415;rinstance=311f4d1534f84e89>;tag=437d7d60
Contact: <sip:20000@172.30.0.135>
Call-ID: 6dc4686c77d8c0fe03a9afae4dcad221@172.30.0.135
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.6.2.10
Content-Length: 0


---
localhost*CLI> 
<--- SIP read from UDP:172.30.1.25:4415 --->



<------------->
localhost*CLI> 
<--- SIP read from UDP:172.30.0.55:2263 --->



<------------->
localhost*CLI> core show channels
localhost*CLI> Channel              Location             State   Application(Data)             
localhost*CLI> SIP/20001-00000013   (None)               Up      AppDial((Outgoing Line))      
localhost*CLI> SIP/20000-00000012   20001@tenant01:1     Up      Dial(SIP/20001,60)            
localhost*CLI> 2 active channels
localhost*CLI> 1 active call
localhost*CLI> 16 calls processed
localhost*CLI> 
<--- SIP read from UDP:172.30.1.25:4415 --->



<------------->
localhost*CLI> 
<--- SIP read from UDP:172.30.0.55:2263 --->



<------------->
localhost*CLI> 
<--- SIP read from UDP:172.30.1.25:4415 --->



<------------->
localhost*CLI> 
<--- SIP read from UDP:172.30.0.55:2263 --->



<------------->
localhost*CLI> set_destination: Parsing <sip:20000@172.30.0.55:2263;rinstance=b4798b847c64aa89> for address/port to send to
set_destination: set destination to 172.30.0.55, port 2263
Audio is at 172.30.0.135 port 12882
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 172.30.0.55:2263:
INVITE sip:20000@172.30.0.55:2263;rinstance=b4798b847c64aa89 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.135:5060;branch=z9hG4bK2376e067;rport
Max-Forwards: 70
From: "20000" <sip:20000@172.30.0.135>;tag=as32365614
To: <sip:20000@172.30.0.55:2263;rinstance=b4798b847c64aa89>;tag=a074f151
Contact: <sip:20000@172.30.0.135>
Call-ID: 210fb10f60a5661c4a1d47605f95f261@172.30.0.135
CSeq: 104 INVITE
User-Agent: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 911803435 911803437 IN IP4 172.30.0.135
s=Asterisk PBX 1.6.2.10
c=IN IP4 172.30.0.135
t=0 0
m=audio 12882 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
  == Spawn extension (tenant01, 9060000, 1) exited non-zero on 'SIP/20000-00000012'
    -- Executing [9060000@tenant01:1] Set("SIP/20000-00000012", "Routing=start") in new stack
    -- Executing [9060000@tenant01:2] Queue("SIP/20000-00000012", "9060000") in new stack
    -- Started music on hold, class 'default', on SIP/20000-00000012
localhost*CLI>     -- Executing [9060000@tenant01:1] Set("SIP/20001-00000013", "Routing=start") in new stack
localhost*CLI>     -- Executing [9060000@tenant01:2] Queue("SIP/20001-00000013", "9060000") in new stack
    -- Started music on hold, class 'default', on SIP/20001-00000013
localhost*CLI> 
<--- SIP read from UDP:172.30.0.55:2263 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.135:5060;branch=z9hG4bK2376e067;rport=5060
Contact: <sip:20000@172.30.0.55:2263;rinstance=b4798b847c64aa89>
To: <sip:20000@172.30.0.55:2263;rinstance=b4798b847c64aa89>;tag=a074f151
From: "20000"<sip:20000@172.30.0.135>;tag=as32365614
Call-ID: 210fb10f60a5661c4a1d47605f95f261@172.30.0.135
CSeq: 104 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: 3CXPhone 4.0.10858.0
Content-Length: 204

v=0
o=3cxVCE 17752110 7752662 IN IP4 172.30.0.55
s=3cxVCE Audio Call
c=IN IP4 172.30.0.55
t=0 0
m=audio 40004 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

<------------->
--- (12 headers 9 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 - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 172.30.0.55:40004
set_destination: Parsing <sip:20000@172.30.0.55:2263;rinstance=b4798b847c64aa89> for address/port to send to
set_destination: set destination to 172.30.0.55, port 2263
Transmitting (NAT) to 172.30.0.55:2263:
ACK sip:20000@172.30.0.55:2263;rinstance=b4798b847c64aa89 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.135:5060;branch=z9hG4bK1c3d2afe;rport
Max-Forwards: 70
From: "20000" <sip:20000@172.30.0.135>;tag=as32365614
To: <sip:20000@172.30.0.55:2263;rinstance=b4798b847c64aa89>;tag=a074f151
Contact: <sip:20000@172.30.0.135>
Call-ID: 210fb10f60a5661c4a1d47605f95f261@172.30.0.135
CSeq: 104 ACK
User-Agent: Asterisk PBX 1.6.2.10
Content-Length: 0


---
localhost*CLI> 
<--- SIP read from UDP:172.30.1.25:4415 --->



<------------->
localhost*CLI> 
<--- SIP read from UDP:172.30.0.55:2263 --->



<------------->
localhost*CLI> 
<--- SIP read from UDP:172.30.1.25:4415 --->
BYE sip:20000@172.30.0.135 SIP/2.0
Via: SIP/2.0/UDP 172.30.1.25:4415;branch=z9hG4bK-d8754z-8c13105e870f4411-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:20001@172.30.1.25:4415;rinstance=311f4d1534f84e89>
To: "20000"<sip:20000@172.30.0.135>;tag=as4e760b43
From: <sip:20001@172.30.1.25:4415;rinstance=311f4d1534f84e89>;tag=437d7d60
Call-ID: 6dc4686c77d8c0fe03a9afae4dcad221@172.30.0.135
CSeq: 2 BYE
User-Agent: 3CXPhone 4.0.10858.0
Reason: SIP;description="User Hung Up"
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Sending to 172.30.1.25 : 4415 (NAT)

<--- Transmitting (NAT) to 172.30.1.25:4415 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.1.25:4415;branch=z9hG4bK-d8754z-8c13105e870f4411-1---d8754z-;received=172.30.1.25;rport=4415
From: <sip:20001@172.30.1.25:4415;rinstance=311f4d1534f84e89>;tag=437d7d60
To: "20000"<sip:20000@172.30.0.135>;tag=as4e760b43
Call-ID: 6dc4686c77d8c0fe03a9afae4dcad221@172.30.0.135
CSeq: 2 BYE
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
localhost*CLI>     -- Stopped music on hold on SIP/20001-00000013
  == Spawn extension (tenant01, 9060000, 2) exited non-zero on 'SIP/20001-00000013'
localhost*CLI> 
<--- SIP read from UDP:172.30.0.55:2263 --->
BYE sip:20000@172.30.0.135 SIP/2.0
Via: SIP/2.0/UDP 172.30.0.55:2263;branch=z9hG4bK-d8754z-9f3dc61251520b51-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:20000@172.30.0.55:2263;rinstance=b4798b847c64aa89>
To: "20000"<sip:20000@172.30.0.135>;tag=as32365614
From: <sip:20000@172.30.0.55:2263;rinstance=b4798b847c64aa89>;tag=a074f151
Call-ID: 210fb10f60a5661c4a1d47605f95f261@172.30.0.135
CSeq: 2 BYE
User-Agent: 3CXPhone 4.0.10858.0
Reason: SIP;description="User Hung Up"
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Sending to 172.30.0.55 : 2263 (NAT)

<--- Transmitting (NAT) to 172.30.0.55:2263 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.0.55:2263;branch=z9hG4bK-d8754z-9f3dc61251520b51-1---d8754z-;received=172.30.0.55;rport=2263
From: <sip:20000@172.30.0.55:2263;rinstance=b4798b847c64aa89>;tag=a074f151
To: "20000"<sip:20000@172.30.0.135>;tag=as32365614
Call-ID: 210fb10f60a5661c4a1d47605f95f261@172.30.0.135
CSeq: 2 BYE
Server: Asterisk PBX 1.6.2.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
Really destroying SIP dialog '6dc4686c77d8c0fe03a9afae4dcad221@172.30.0.135' Method: BYE
localhost*CLI>     -- Stopped music on hold on SIP/20000-00000012
  == Spawn extension (tenant01, 9060000, 2) exited non-zero on 'SIP/20000-00000012'
localhost*CLI> Really destroying SIP dialog '210fb10f60a5661c4a1d47605f95f261@172.30.0.135' Method: BYE
localhost*CLI> core show channels
localhost*CLI> Channel              Location             State   Application(Data)             
localhost*CLI> 0 active channels
localhost*CLI> 0 active calls
localhost*CLI> 17 calls processed
localhost*CLI> 
localhost*CLI> 
<--- SIP read from UDP:172.30.1.25:4415 --->



<------------->
localhost*CLI> 
<--- SIP read from UDP:172.30.0.55:2263 --->



<------------->
localhost*CLI> 
<--- SIP read from UDP:172.30.1.25:4415 --->



<------------->
localhost*CLI> 
<--- SIP read from UDP:172.30.0.55:2263 --->



<------------->
localhost*CLI>