rendered paste body[Feb 1 20:19:25] VERBOSE[13909] logger.c:
<--- SIP read from 12.37.6.160:5060 --->
INVITE sip:17185387024@sip-us.ziggoo.net SIP/2.0
Via: SIP/2.0/UDP 12.37.6.160:5060;branch=z9hG4bK1944e697;rport
From: "Espacio" <sip:2655209225@12.37.6.160>;tag=as79712a0e
To: <sip:17185387024@sip-us.ziggoo.net>
Contact: <sip:2655209225@12.37.6.160>
Call-ID: 6ab3df5d298822913fa87a2f788751d0@12.37.6.160
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 02 Feb 2007 01:19:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 309
v=0
o=root 2617 2617 IN IP4 12.37.6.160
s=session
c=IN IP4 12.37.6.160
t=0 0
m=audio 15580 RTP/AVP 0 18 8 97 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
<------------->
[Feb 1 20:19:25] VERBOSE[13909] logger.c: --- (13 headers 14 lines) ---
[Feb 1 20:19:25] VERBOSE[13909] logger.c: Sending to 12.37.6.160 : 5060 (NAT)
[Feb 1 20:19:25] VERBOSE[13909] logger.c: Using INVITE request as basis request - 6ab3df5d298822913fa87a2f788751d0@12.37.6.160
[Feb 1 20:19:25] VERBOSE[13909] logger.c: Found peer 'slogica'
[Feb 1 20:19:25] VERBOSE[13909] logger.c: Found RTP audio format 0
[Feb 1 20:19:25] VERBOSE[13909] logger.c: Found RTP audio format 18
[Feb 1 20:19:25] VERBOSE[13909] logger.c: Found RTP audio format 8
[Feb 1 20:19:25] VERBOSE[13909] logger.c: Found RTP audio format 97
[Feb 1 20:19:25] VERBOSE[13909] logger.c: Found RTP audio format 101
[Feb 1 20:19:25] DEBUG[13909] chan_sip.c: Peer doesn't provide T.38 UDPTL
[Feb 1 20:19:25] VERBOSE[13909] logger.c: Peer audio RTP is at port 12.37.6.160:15580
[Feb 1 20:19:25] VERBOSE[13909] logger.c: Found description format PCMU for ID 0
[Feb 1 20:19:25] VERBOSE[13909] logger.c: Found description format G729 for ID 18
[Feb 1 20:19:25] VERBOSE[13909] logger.c: Found description format PCMA for ID 8
[Feb 1 20:19:25] VERBOSE[13909] logger.c: Found description format iLBC for ID 97
[Feb 1 20:19:25] VERBOSE[13909] logger.c: Found description format telephone-event for ID 101
[Feb 1 20:19:25] VERBOSE[13909] logger.c: Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x50c (ulaw|alaw|g729|ilbc)/video=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729)
[Feb 1 20:19:25] VERBOSE[13909] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Feb 1 20:19:25] VERBOSE[13909] logger.c: Peer audio RTP is at port 12.37.6.160:15580
[Feb 1 20:19:25] VERBOSE[13909] logger.c: Looking for 17185387024 in slogica (domain sip-us.ziggoo.net)
[Feb 1 20:19:25] VERBOSE[13909] logger.c: list_route: hop: <sip:2655209225@12.37.6.160>
[Feb 1 20:19:25] VERBOSE[13909] logger.c:
<--- Transmitting (no NAT) to 12.37.6.160:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 12.37.6.160:5060;branch=z9hG4bK1944e697;received=12.37.6.160;rport=5060
From: "Espacio" <sip:2655209225@12.37.6.160>;tag=as79712a0e
To: <sip:17185387024@sip-us.ziggoo.net>
Call-ID: 6ab3df5d298822913fa87a2f788751d0@12.37.6.160
CSeq: 102 INVITE
User-Agent: Ziggoo
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:17185387024@208.116.0.218>
Content-Length: 0
<------------>
[Feb 1 20:19:25] VERBOSE[14426] logger.c: -- Executing [17185387024@slogica:1] Set("SIP/12.37.6.160-0849a3b0", "CDR(userfield)="Espacio" <2655209225>,0.013,6,6") in new stack
[Feb 1 20:19:25] VERBOSE[14426] logger.c: -- Executing [17185387024@slogica:2] Macro("SIP/12.37.6.160-0849a3b0", "usa|17185387024") in new stack
[Feb 1 20:19:25] VERBOSE[14426] logger.c: -- Executing [s@macro-usa:1] Set("SIP/12.37.6.160-0849a3b0", "CALLERID(all)=2655209225") in new stack
[Feb 1 20:19:25] VERBOSE[14426] logger.c: -- Executing [s@macro-usa:2] Macro("SIP/12.37.6.160-0849a3b0", "dial|SIP/ibell-usa/17185387024") in new stack
[Feb 1 20:19:25] VERBOSE[14426] logger.c: -- Executing [s@macro-dial:1] Dial("SIP/12.37.6.160-0849a3b0", "SIP/ibell-usa/17185387024||") in new stack
[Feb 1 20:19:25] VERBOSE[14426] logger.c: Audio is at 208.116.0.218 port 15746
[Feb 1 20:19:25] VERBOSE[14426] logger.c: Adding codec 0x100 (g729) to SDP
[Feb 1 20:19:25] VERBOSE[14426] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
[Feb 1 20:19:25] VERBOSE[14426] logger.c: Reliably Transmitting (no NAT) to 69.28.243.3:5060:
INVITE sip:17185387024@69.28.243.3 SIP/2.0
Via: SIP/2.0/UDP 208.116.0.218:5060;branch=z9hG4bK0cdb89fe;rport
From: "2655209225" <sip:2655209225@208.116.0.218>;tag=as1e6f28ed
To: <sip:17185387024@69.28.243.3>
Contact: <sip:2655209225@208.116.0.218>
Call-ID: 32295ead63e2966d57419f5b386f6e7a@208.116.0.218
CSeq: 102 INVITE
User-Agent: Ziggoo
Max-Forwards: 70
Date: Fri, 02 Feb 2007 01:19:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 265
v=0
o=root 13853 13853 IN IP4 208.116.0.218
s=session
c=IN IP4 208.116.0.218
t=0 0
m=audio 15746 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
[Feb 1 20:19:25] VERBOSE[14426] logger.c: -- Called ibell-usa/17185387024
[Feb 1 20:19:25] VERBOSE[13909] logger.c:
<--- SIP read from 69.28.243.3:5060 --->
SIP/2.0 407 Proxy Authentication Required
CSeq: 102 INVITE
Via: SIP/2.0/UDP 208.116.0.218:5060;branch=z9hG4bK0cdb89fe;rport
From: "2655209225" <sip:2655209225@208.116.0.218>;tag=as1e6f28ed
Call-ID: 32295ead63e2966d57419f5b386f6e7a@208.116.0.218
To: <sip:17185387024@69.28.243.3>
Contact: <sip:69.28.243.3:5060;user=phone;transport=udp>
Proxy-Authenticate: DIGEST realm="VoipSwitch", nonce="117037916501200803202192504405
Content-Length: 0
<------------->
[Feb 1 20:19:25] VERBOSE[13909] logger.c: --- (9 headers 0 lines) ---
[Feb 1 20:19:25] VERBOSE[13909] logger.c: Transmitting (no NAT) to 69.28.243.3:5060:
ACK sip:17185387024@69.28.243.3 SIP/2.0
Via: SIP/2.0/UDP 208.116.0.218:5060;branch=z9hG4bK0cdb89fe;rport
From: "2655209225" <sip:2655209225@208.116.0.218>;tag=as1e6f28ed
To: <sip:17185387024@69.28.243.3>
Contact: <sip:2655209225@208.116.0.218>
Call-ID: 32295ead63e2966d57419f5b386f6e7a@208.116.0.218
CSeq: 102 ACK
User-Agent: Ziggoo
Max-Forwards: 70
Content-Length: 0
---
[Feb 1 20:19:25] VERBOSE[13909] logger.c: Audio is at 208.116.0.218 port 15746
[Feb 1 20:19:25] VERBOSE[13909] logger.c: Adding codec 0x100 (g729) to SDP
[Feb 1 20:19:25] VERBOSE[13909] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
[Feb 1 20:19:25] VERBOSE[13909] logger.c: Reliably Transmitting (no NAT) to 69.28.243.3:5060:
INVITE sip:17185387024@69.28.243.3 SIP/2.0
Via: SIP/2.0/UDP 208.116.0.218:5060;branch=z9hG4bK4677ce3c;rport
From: "2655209225" <sip:2655209225@208.116.0.218>;tag=as1e6f28ed
To: <sip:17185387024@69.28.243.3>
Contact: <sip:2655209225@208.116.0.218>
Call-ID: 32295ead63e2966d57419f5b386f6e7a@208.116.0.218
CSeq: 103 INVITE
User-Agent: Ziggoo
Max-Forwards: 70
Proxy-Authorization: Digest username="ziggoo", realm="VoipSwitch", algorithm=MD5, uri="sip:17185387024@69.28.243.3", nonce="117037916501200803202192504405", response="bfb56cc7bef14b42f436c61e437676c8", opaque=""
Date: Fri, 02 Feb 2007 01:19:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 265
v=0
o=root 13853 13854 IN IP4 208.116.0.218
s=session
c=IN IP4 208.116.0.218
t=0 0
m=audio 15746 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
[Feb 1 20:19:25] VERBOSE[13909] logger.c:
<--- SIP read from 69.28.243.3:5060 --->
SIP/2.0 100 Trying
CSeq: 103 INVITE
Via: SIP/2.0/UDP 208.116.0.218:5060;branch=z9hG4bK0cdb89fe;rport
From: "2655209225" <sip:2655209225@208.116.0.218>;tag=as1e6f28ed
Call-ID: 32295ead63e2966d57419f5b386f6e7a@208.116.0.218
To: <sip:17185387024@69.28.243.3>;tag=0102190720256601
Contact: <sip:69.28.243.3:5060;user=phone;transport=udp>
Content-Length: 0
<------------->
[Feb 1 20:19:28] VERBOSE[13909] logger.c: --- (10 headers 0 lines) ---
[Feb 1 20:19:28] VERBOSE[13909] logger.c: Really destroying SIP dialog '4ec7cb1073acb95b7fb2952e54f9003f@208.116.0.218' Method: OPTIONS
[Feb 1 20:19:29] VERBOSE[13909] logger.c:
<--- SIP read from 69.28.243.3:5060 --->
SIP/2.0 180 Ringing
CSeq: 103 INVITE
Via: SIP/2.0/UDP 208.116.0.218:5060;branch=z9hG4bK0cdb89fe;rport
From: "2655209225" <sip:2655209225@208.116.0.218>;tag=as1e6f28ed
Call-ID: 32295ead63e2966d57419f5b386f6e7a@208.116.0.218
To: <sip:17185387024@69.28.243.3>;tag=0102190720256601
Contact: <sip:69.28.243.3:5060;user=phone;transport=udp>
Content-Type: application/sdp
Content-Length: 209
v=0
o=VoipSwitch 7600 7600 IN IP4 69.28.243.3
s=VoipSIP
i=Audio Session
c=IN IP4 69.28.243.3
t=0 0
m=audio 6600 RTP/AVP 18 101
a=rtpmap:18 G729/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
[Feb 1 20:19:29] VERBOSE[13909] logger.c: --- (9 headers 10 lines) ---
[Feb 1 20:19:29] VERBOSE[13909] logger.c: Found RTP audio format 18
[Feb 1 20:19:29] VERBOSE[13909] logger.c: Found RTP audio format 101
[Feb 1 20:19:29] DEBUG[13909] chan_sip.c: Peer doesn't provide T.38 UDPTL
[Feb 1 20:19:29] VERBOSE[13909] logger.c: Peer audio RTP is at port 69.28.243.3:6600
[Feb 1 20:19:29] VERBOSE[13909] logger.c: Found description format G729 for ID 18
[Feb 1 20:19:29] VERBOSE[13909] logger.c: Found description format telephone-event for ID 101
[Feb 1 20:19:29] VERBOSE[13909] logger.c: Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
[Feb 1 20:19:29] VERBOSE[13909] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Feb 1 20:19:29] VERBOSE[13909] logger.c: Peer audio RTP is at port 69.28.243.3:6600
[Feb 1 20:19:29] VERBOSE[14426] logger.c: -- SIP/ibell-usa-084baad8 is ringing
[Feb 1 20:19:29] VERBOSE[14426] logger.c:
<--- Transmitting (no NAT) to 12.37.6.160:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 12.37.6.160:5060;branch=z9hG4bK1944e697;received=12.37.6.160;rport=5060
From: "Espacio" <sip:2655209225@12.37.6.160>;tag=as79712a0e
To: <sip:17185387024@sip-us.ziggoo.net>;tag=as4c0b22b8
Call-ID: 6ab3df5d298822913fa87a2f788751d0@12.37.6.160
CSeq: 102 INVITE
User-Agent: Ziggoo
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:17185387024@208.116.0.218>
Content-Length: 0
<------------>
[Feb 1 20:19:29] VERBOSE[14426] logger.c: -- SIP/ibell-usa-084baad8 is making progress passing it to SIP/12.37.6.160-0849a3b0
[Feb 1 20:19:29] VERBOSE[14426] logger.c: Audio is at 208.116.0.218 port 12838
[Feb 1 20:19:29] VERBOSE[14426] logger.c: Adding codec 0x100 (g729) to SDP
[Feb 1 20:19:29] VERBOSE[14426] logger.c: Adding codec 0x4 (ulaw) to SDP
[Feb 1 20:19:29] VERBOSE[14426] logger.c: Adding codec 0x8 (alaw) to SDP
[Feb 1 20:19:29] VERBOSE[14426] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
[Feb 1 20:19:29] VERBOSE[14426] logger.c:
<--- Transmitting (no NAT) to 12.37.6.160:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 12.37.6.160:5060;branch=z9hG4bK1944e697;received=12.37.6.160;rport=5060
From: "Espacio" <sip:2655209225@12.37.6.160>;tag=as79712a0e
To: <sip:17185387024@sip-us.ziggoo.net>;tag=as4c0b22b8
Call-ID: 6ab3df5d298822913fa87a2f788751d0@12.37.6.160
CSeq: 102 INVITE
User-Agent: Ziggoo
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:17185387024@208.116.0.218>
Content-Type: application/sdp
Content-Length: 313
v=0
o=root 13853 13853 IN IP4 208.116.0.218
s=session
c=IN IP4 208.116.0.218
t=0 0
m=audio 12838 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
[Feb 1 20:19:37] VERBOSE[13909] logger.c:
<--- SIP read from 69.28.243.3:5060 --->
SIP/2.0 200 OK
CSeq: 103 INVITE
Via: SIP/2.0/UDP 208.116.0.218:5060;branch=z9hG4bK0cdb89fe;rport
From: "2655209225" <sip:2655209225@208.116.0.218>;tag=as1e6f28ed
Call-ID: 32295ead63e2966d57419f5b386f6e7a@208.116.0.218
To: <sip:17185387024@69.28.243.3>;tag=0102190720256601
Contact: <sip:69.28.243.3:5060;user=phone;transport=udp>
Content-Type: application/sdp
Content-Length: 209
v=0
o=VoipSwitch 7600 7600 IN IP4 69.28.243.3
s=VoipSIP
i=Audio Session
c=IN IP4 69.28.243.3
t=0 0
m=audio 6600 RTP/AVP 18 101
a=rtpmap:18 G729/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
[Feb 1 20:19:37] VERBOSE[13909] logger.c: --- (9 headers 10 lines) ---
[Feb 1 20:19:37] VERBOSE[13909] logger.c: Found RTP audio format 18
[Feb 1 20:19:37] VERBOSE[13909] logger.c: Found RTP audio format 101
[Feb 1 20:19:37] DEBUG[13909] chan_sip.c: Peer doesn't provide T.38 UDPTL
[Feb 1 20:19:37] VERBOSE[13909] logger.c: Peer audio RTP is at port 69.28.243.3:6600
[Feb 1 20:19:37] VERBOSE[13909] logger.c: Found description format G729 for ID 18
[Feb 1 20:19:37] VERBOSE[13909] logger.c: Found description format telephone-event for ID 101
[Feb 1 20:19:37] VERBOSE[13909] logger.c: Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
[Feb 1 20:19:37] VERBOSE[13909] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Feb 1 20:19:37] VERBOSE[13909] logger.c: Peer audio RTP is at port 69.28.243.3:6600
[Feb 1 20:19:37] VERBOSE[13909] logger.c: list_route: hop: <sip:69.28.243.3:5060;user=phone;transport=udp>
[Feb 1 20:19:37] DEBUG[13909] chan_sip.c: Strict routing enforced for session 32295ead63e2966d57419f5b386f6e7a@208.116.0.218
[Feb 1 20:19:37] VERBOSE[13909] logger.c: set_destination: Parsing <sip:69.28.243.3:5060;user=phone;transport=udp> for address/port to send to
[Feb 1 20:19:37] VERBOSE[13909] logger.c: set_destination: set destination to 69.28.243.3, port 5060
[Feb 1 20:19:37] VERBOSE[13909] logger.c: Transmitting (no NAT) to 69.28.243.3:5060:
ACK sip:69.28.243.3:5060;user=phone;transport=udp SIP/2.0
Via: SIP/2.0/UDP 208.116.0.218:5060;branch=z9hG4bK76a7a2f9;rport
From: "2655209225" <sip:2655209225@208.116.0.218>;tag=as1e6f28ed
To: <sip:17185387024@69.28.243.3>;tag=0102190720256601
Contact: <sip:2655209225@208.116.0.218>
Call-ID: 32295ead63e2966d57419f5b386f6e7a@208.116.0.218
CSeq: 103 ACK
User-Agent: Ziggoo
Max-Forwards: 70
Content-Length: 0
---
[Feb 1 20:19:37] VERBOSE[14426] logger.c: -- SIP/ibell-usa-084baad8 answered SIP/12.37.6.160-0849a3b0
[Feb 1 20:19:37] VERBOSE[14426] logger.c: Audio is at 208.116.0.218 port 12838
[Feb 1 20:19:37] VERBOSE[14426] logger.c: Adding codec 0x100 (g729) to SDP
[Feb 1 20:19:37] VERBOSE[14426] logger.c: Adding codec 0x4 (ulaw) to SDP
[Feb 1 20:19:37] VERBOSE[14426] logger.c: Adding codec 0x8 (alaw) to SDP
[Feb 1 20:19:37] VERBOSE[14426] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
[Feb 1 20:19:37] VERBOSE[14426] logger.c:
<--- Reliably Transmitting (no NAT) to 12.37.6.160:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 12.37.6.160:5060;branch=z9hG4bK1944e697;received=12.37.6.160;rport=5060
From: "Espacio" <sip:2655209225@12.37.6.160>;tag=as79712a0e
To: <sip:17185387024@sip-us.ziggoo.net>;tag=as4c0b22b8
Call-ID: 6ab3df5d298822913fa87a2f788751d0@12.37.6.160
CSeq: 102 INVITE
User-Agent: Ziggoo
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:17185387024@208.116.0.218>
Content-Type: application/sdp
Content-Length: 313
v=0
o=root 13853 13854 IN IP4 208.116.0.218
s=session
c=IN IP4 208.116.0.218
t=0 0
m=audio 12838 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
[Feb 1 20:19:37] VERBOSE[14426] logger.c: -- Native bridging SIP/12.37.6.160-0849a3b0 and SIP/ibell-usa-084baad8
[Feb 1 20:19:37] DEBUG[14426] chan_sip.c: Strict routing enforced for session 32295ead63e2966d57419f5b386f6e7a@208.116.0.218
[Feb 1 20:19:37] VERBOSE[14426] logger.c: set_destination: Parsing <sip:69.28.243.3:5060;user=phone;transport=udp> for address/port to send to
[Feb 1 20:19:37] VERBOSE[14426] logger.c: set_destination: set destination to 69.28.243.3, port 5060
[Feb 1 20:19:37] VERBOSE[14426] logger.c: Audio is at 208.116.0.218 port 15746
[Feb 1 20:19:37] VERBOSE[14426] logger.c: Adding codec 0x100 (g729) to SDP
[Feb 1 20:19:37] VERBOSE[14426] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
[Feb 1 20:19:37] VERBOSE[14426] logger.c: Reliably Transmitting (no NAT) to 69.28.243.3:5060:
INVITE sip:69.28.243.3:5060;user=phone;transport=udp SIP/2.0
Via: SIP/2.0/UDP 208.116.0.218:5060;branch=z9hG4bK754c72cc;rport
From: "2655209225" <sip:2655209225@208.116.0.218>;tag=as1e6f28ed
To: <sip:17185387024@69.28.243.3>;tag=0102190720256601
Contact: <sip:2655209225@208.116.0.218>
Call-ID: 32295ead63e2966d57419f5b386f6e7a@208.116.0.218
CSeq: 104 INVITE
User-Agent: Ziggoo
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 261
v=0
o=root 13853 13855 IN IP4 12.37.6.160
s=session
c=IN IP4 12.37.6.160
t=0 0
m=audio 15580 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
[Feb 1 20:19:37] VERBOSE[13909] logger.c:
<--- SIP read from 12.37.6.160:5060 --->
ACK sip:17185387024@208.116.0.218 SIP/2.0
Via: SIP/2.0/UDP 12.37.6.160:5060;branch=z9hG4bK0f52c978;rport
From: "Espacio" <sip:2655209225@12.37.6.160>;tag=as79712a0e
To: <sip:17185387024@sip-us.ziggoo.net>;tag=as4c0b22b8
Contact: <sip:2655209225@12.37.6.160>
Call-ID: 6ab3df5d298822913fa87a2f788751d0@12.37.6.160
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
<------------->
[Feb 1 20:19:37] VERBOSE[13909] logger.c: --- (10 headers 0 lines) ---
[Feb 1 20:19:37] DEBUG[13909] chan_sip.c: Strict routing enforced for session 6ab3df5d298822913fa87a2f788751d0@12.37.6.160
[Feb 1 20:19:37] VERBOSE[13909] logger.c: set_destination: Parsing <sip:2655209225@12.37.6.160> for address/port to send to
[Feb 1 20:19:37] VERBOSE[13909] logger.c: set_destination: set destination to 12.37.6.160, port 5060
[Feb 1 20:19:37] VERBOSE[13909] logger.c: Audio is at 208.116.0.218 port 12838
[Feb 1 20:19:37] VERBOSE[13909] logger.c: Adding codec 0x100 (g729) to SDP
[Feb 1 20:19:37] VERBOSE[13909] logger.c: Adding codec 0x4 (ulaw) to SDP
[Feb 1 20:19:37] VERBOSE[13909] logger.c: Adding codec 0x8 (alaw) to SDP
[Feb 1 20:19:37] VERBOSE[13909] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
[Feb 1 20:19:37] VERBOSE[13909] logger.c: Reliably Transmitting (no NAT) to 12.37.6.160:5060:
INVITE sip:2655209225@12.37.6.160 SIP/2.0
Via: SIP/2.0/UDP 208.116.0.218:5060;branch=z9hG4bK232cfe62;rport
From: <sip:17185387024@sip-us.ziggoo.net>;tag=as4c0b22b8
To: "Espacio" <sip:2655209225@12.37.6.160>;tag=as79712a0e
Contact: <sip:17185387024@208.116.0.218>
Call-ID: 6ab3df5d298822913fa87a2f788751d0@12.37.6.160
CSeq: 102 INVITE
User-Agent: Ziggoo
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 308
v=0
o=root 13853 13855 IN IP4 69.28.243.3
s=session
c=IN IP4 69.28.243.3
t=0 0
m=audio 6600 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
[Feb 1 20:19:37] VERBOSE[13909] logger.c:
<--- SIP read from 12.37.6.160:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.116.0.218:5060;branch=z9hG4bK232cfe62;rport;received=208.116.0.218
From: <sip:17185387024@sip-us.ziggoo.net>;tag=as4c0b22b8
To: "Espacio" <sip:2655209225@12.37.6.160>;tag=as79712a0e
Call-ID: 6ab3df5d298822913fa87a2f788751d0@12.37.6.160
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:2655209225@12.37.6.160>
Content-Type: application/sdp
Content-Length: 309
v=0
o=root 2617 2618 IN IP4 12.37.6.160
s=session
c=IN IP4 12.37.6.160
t=0 0
m=audio 15580 RTP/AVP 0 18 8 97 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
<------------->
[Feb 1 20:19:37] VERBOSE[13909] logger.c: --- (12 headers 14 lines) ---
[Feb 1 20:19:37] VERBOSE[13909] logger.c: Found RTP audio format 0
[Feb 1 20:19:37] VERBOSE[13909] logger.c: Found RTP audio format 18
[Feb 1 20:19:37] VERBOSE[13909] logger.c: Found RTP audio format 8
[Feb 1 20:19:37] VERBOSE[13909] logger.c: Found RTP audio format 97
[Feb 1 20:19:37] VERBOSE[13909] logger.c: Found RTP audio format 101
[Feb 1 20:19:37] DEBUG[13909] chan_sip.c: Peer doesn't provide T.38 UDPTL
[Feb 1 20:19:37] VERBOSE[13909] logger.c: Peer audio RTP is at port 12.37.6.160:15580
[Feb 1 20:19:37] VERBOSE[13909] logger.c: Found description format PCMU for ID 0
[Feb 1 20:19:37] VERBOSE[13909] logger.c: Found description format G729 for ID 18
[Feb 1 20:19:37] VERBOSE[13909] logger.c: Found description format PCMA for ID 8
[Feb 1 20:19:37] VERBOSE[13909] logger.c: Found description format iLBC for ID 97
[Feb 1 20:19:37] VERBOSE[13909] logger.c: Found description format telephone-event for ID 101
[Feb 1 20:19:37] VERBOSE[13909] logger.c: Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x50c (ulaw|alaw|g729|ilbc)/video=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729)
[Feb 1 20:19:37] VERBOSE[13909] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Feb 1 20:19:37] VERBOSE[13909] logger.c: Peer audio RTP is at port 12.37.6.160:15580
[Feb 1 20:19:37] VERBOSE[13909] logger.c: list_route: hop: <sip:2655209225@12.37.6.160>
[Feb 1 20:19:37] DEBUG[13909] chan_sip.c: Strict routing enforced for session 6ab3df5d298822913fa87a2f788751d0@12.37.6.160
[Feb 1 20:19:37] VERBOSE[13909] logger.c: set_destination: Parsing <sip:2655209225@12.37.6.160> for address/port to send to
[Feb 1 20:19:37] VERBOSE[13909] logger.c: set_destination: set destination to 12.37.6.160, port 5060
[Feb 1 20:19:37] VERBOSE[13909] logger.c: Transmitting (no NAT) to 12.37.6.160:5060:
ACK sip:2655209225@12.37.6.160 SIP/2.0
Via: SIP/2.0/UDP 208.116.0.218:5060;branch=z9hG4bK3a50185d;rport
From: <sip:17185387024@sip-us.ziggoo.net>;tag=as4c0b22b8
To: "Espacio" <sip:2655209225@12.37.6.160>;tag=as79712a0e
Contact: <sip:17185387024@208.116.0.218>
Call-ID: 6ab3df5d298822913fa87a2f788751d0@12.37.6.160
CSeq: 102 ACK
User-Agent: Ziggoo
Max-Forwards: 70
Content-Length: 0
---