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<--- SIP read from 10.0.0.1:5060 --->
INVITE sip:23003@10.0.0.2;transport=UDP;TCID-0 SIP/2.0
From: ORAS<sip:748881658@10.0.0.1:5060>;tag=a000001-13c4-16d2e-59298f8-249e
To: <sip:23003@10.0.0.2>
Call-ID: 9794a14c-a000001-13c4-16d2e-59298f8-1920@10.0.0.1
CSeq: 1 INVITE
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK-16d2e-59298f8-6b7f
Max-Forwards: 70
Contact: <sip:748881658@10.0.0.1:5060;transport=UDP>
Content-Type: application/SDP
Content-Length:288
v=0
o=a0000 1234567890 1234567890 IN IP4 10.0.0.1
s=SIP Call test
c=IN IP4 10.0.0.1
t=0 0
m=audio 10002 RTP/AVP 8 97 0 18
a=fmtp:97 0-15
a=fmtp:18 annexb=yes
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
<------------->
--- (10 headers 13 lines) ---
Sending to 10.0.0.1 : 5060 (no NAT)
Using INVITE request as basis request - 9794a14c-a000001-13c4-16d2e-59298f8-1920@10.0.0.1
Found peer 'harris'
Found RTP audio format 8
Found RTP audio format 97
Found RTP audio format 0
Found RTP audio format 18
Peer audio RTP is at port 10.0.0.1:10002
Found description format PCMA for ID 8
Found description format telephone-event for ID 97
Found description format PCMU for ID 0
Found description format G729 for ID 18
Capabilities: us - 0x3f1fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|jpeg|png|h261|h263|h263p|h264), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.0.0.1:10002
Looking for 23003 in default (domain 10.0.0.2)
list_route: hop: <sip:748881658@10.0.0.1:5060;transport=UDP>
<--- Transmitting (NAT) to 10.0.0.1:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK-16d2e-59298f8-6b7f;received=10.0.0.1
From: ORAS<sip:748881658@10.0.0.1:5060>;tag=a000001-13c4-16d2e-59298f8-249e
To: <sip:23003@10.0.0.2>
Call-ID: 9794a14c-a000001-13c4-16d2e-59298f8-1920@10.0.0.1
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:23003@10.0.0.2>
Content-Length: 0
<------------>
-- Executing [23003@default:1] Dial("SIP/5060-0822ef78", "SIP/23003") in new stack
-- Called 23003
-- SIP/23003-0823b6c0 is ringing
<--- Transmitting (NAT) to 10.0.0.1:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK-16d2e-59298f8-6b7f;received=10.0.0.1
From: ORAS<sip:748881658@10.0.0.1:5060>;tag=a000001-13c4-16d2e-59298f8-249e
To: <sip:23003@10.0.0.2>;tag=as5779f29f
Call-ID: 9794a14c-a000001-13c4-16d2e-59298f8-1920@10.0.0.1
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:23003@10.0.0.2>
Content-Length: 0
<------------>
Audio is at 10.0.0.2 port 11854
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Transmitting (NAT) to 10.0.0.1:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK-16d2e-59298f8-6b7f;received=10.0.0.1
From: ORAS<sip:748881658@10.0.0.1:5060>;tag=a000001-13c4-16d2e-59298f8-249e
To: <sip:23003@10.0.0.2>;tag=as5779f29f
Call-ID: 9794a14c-a000001-13c4-16d2e-59298f8-1920@10.0.0.1
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:23003@10.0.0.2>
Content-Type: application/sdp
Content-Length: 300
v=0
o=root 24756 24756 IN IP4 10.0.0.2
s=session
c=IN IP4 10.0.0.2
t=0 0
m=audio 11854 RTP/AVP 8 0 18 97
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-16
a=silenceSupp:off - - - -
a=ptime:20
=sendrecv
<------------>
-- Call on SIP/23003-0823b6c0 left from hold
-- SIP/23003-0823b6c0 answered SIP/5060-0822ef78
Audio is at 10.0.0.2 port 11854
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (NAT) to 10.0.0.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK-16d2e-59298f8-6b7f;received=10.0.0.1
From: ORAS<sip:748881658@10.0.0.1:5060>;tag=a000001-13c4-16d2e-59298f8-249e
To: <sip:23003@10.0.0.2>;tag=as5779f29f
Call-ID: 9794a14c-a000001-13c4-16d2e-59298f8-1920@10.0.0.1
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:23003@10.0.0.2>
Content-Type: application/sdp
Content-Length: 300
v=0
o=root 24756 24757 IN IP4 10.0.0.2
s=session
c=IN IP4 10.0.0.2
t=0 0
m=audio 11854 RTP/AVP 8 0 18 97
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
-- Packet2Packet bridging SIP/5060-0822ef78 and SIP/23003-0823b6c0
voip*CLI>
<--- SIP read from 10.0.0.1:5060 --->
ACK sip:23003@10.0.0.2 SIP/2.0
From: ORAS<sip:748881658@10.0.0.1:5060>;tag=a000001-13c4-16d2e-59298f8-249e
To: <sip:23003@10.0.0.2>;tag=as5779f29f
Call-ID: 9794a14c-a000001-13c4-16d2e-59298f8-1920@10.0.0.1
CSeq: 1 ACK
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK-16d38-592c144-274c
Max-Forwards: 70
Contact: <sip:748881658@10.0.0.1:5060;transport=UDP>
Content-Length:0
<------------->
--- (9 headers 0 lines) ---
voip*CLI>
<--- SIP read from 10.0.0.1:5060 --->
BYE sip:23003@10.0.0.2;TCID-0 SIP/2.0
From: ORAS<sip:748881658@10.0.0.1:5060>;tag=a000001-13c4-16d2e-59298f8-249e
To: <sip:23003@10.0.0.2>;tag=as5779f29f
Call-ID: 9794a14c-a000001-13c4-16d2e-59298f8-1920@10.0.0.1
CSeq: 2 BYE
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK-16d38-592c14a-6caa
Max-Forwards: 70
Content-Length:0
<------------->
--- (8 headers 0 lines) ---
Sending to 10.0.0.1 : 5060 (NAT)
<--- Transmitting (NAT) to 10.0.0.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK-16d38-592c14a-6caa;received=10.0.0.1
From: ORAS<sip:748881658@10.0.0.1:5060>;tag=a000001-13c4-16d2e-59298f8-249e
To: <sip:23003@10.0.0.2>;tag=as5779f29f
Call-ID: 9794a14c-a000001-13c4-16d2e-59298f8-1920@10.0.0.1
CSeq: 2 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:23003@10.0.0.2>
Content-Length: 0
<------------>
== Spawn extension (default, 23003, 1) exited non-zero on 'SIP/5060-0822ef78'
Really destroying SIP dialog '9794a14c-a000001-13c4-16d2e-59298f8-1920@10.0.0.1' Method: BYE
voip*CLI>