All pastes #85097 Raw Edit
Unnamed

public text v1 · immutable
·published 2006-07-11 17:32 UTC
aries*CLI>
<-- SIP read from 66.250.69.16:5060:
INVITE sip:8006170141@host41.com SIP/2.0
Via: SIP/2.0/UDP 66.250.69.16:5060;branch=z9hG4bK1c028e1f
From: "4072930335" <sip:4072930335@66.250.69.16>;tag=as6b37a291
To: <sip:8006170141@host41.com>
Contact: <sip:4072930335@66.250.69.16>
Call-ID: 1067c5fb7a827ecc1a6f50421d1fc233@66.250.69.16
CSeq: 102 INVITE
User-Agent: asterlink/1.2
Date: Tue, 11 Jul 2006 17:29:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Type: application/sdp
Content-Length: 469

v=0
o=root 9978 9978 IN IP4 66.250.69.16
s=session
c=IN IP4 66.250.69.16
t=0 0
m=audio 23940 RTP/AVP 10 4 3 0 8 111 5 7 18 110 97 101
a=rtpmap:10 L16/8000
a=rtpmap:4 G723/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

--- (12 headers 20 lines)---
Using INVITE request as basis request - 1067c5fb7a827ecc1a6f50421d1fc233@66.250.69.16
Sending to 66.250.69.16 : 5060 (non-NAT)
Found peer 'asterlink-switch-06'
Found RTP audio format 10
Found RTP audio format 4
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 111
Found RTP audio format 5
Found RTP audio format 7
Found RTP audio format 18
Found RTP audio format 110
Found RTP audio format 97
Found RTP audio format 101
Peer audio RTP is at port 66.250.69.16:23940
Found description format L16
Found description format G723
Found description format GSM
Found description format PCMU
Found description format PCMA
Found description format G726-32
Found description format DVI4
Found description format LPC
Found description format G729
Found description format speex
Found description format iLBC
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x7ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 8006170141 in incoming (domain host41.com)
list_route: hop: <sip:4072930335@66.250.69.16>
Transmitting (no NAT) to 66.250.69.16:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 66.250.69.16:5060;branch=z9hG4bK1c028e1f;received=66.250.69.16
From: "4072930335" <sip:4072930335@66.250.69.16>;tag=as6b37a291
To: <sip:8006170141@host41.com>
Call-ID: 1067c5fb7a827ecc1a6f50421d1fc233@66.250.69.16
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:8006170141@72.232.182.13>
Content-Length: 0


---
Destroying call '29e216df2bea67bf73615d3210d9773d@72.232.182.13'
Destroying call '63a8663921d8836d4e4fd65a636b6526@72.232.182.13'
We're at 72.232.182.13 port 16618
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 66.250.69.16:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 66.250.69.16:5060;branch=z9hG4bK1c028e1f;received=66.250.69.16
From: "4072930335" <sip:4072930335@66.250.69.16>;tag=as6b37a291
To: <sip:8006170141@host41.com>;tag=as522fefca
Call-ID: 1067c5fb7a827ecc1a6f50421d1fc233@66.250.69.16
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:8006170141@72.232.182.13>
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 3534 3534 IN IP4 72.232.182.13
s=session
c=IN IP4 72.232.182.13
t=0 0
m=audio 16618 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
aries*CLI>
<-- SIP read from 66.250.69.16:5060:
ACK sip:8006170141@72.232.182.13 SIP/2.0
Via: SIP/2.0/UDP 66.250.69.16:5060;branch=z9hG4bK0de25ef0
From: "4072930335" <sip:4072930335@66.250.69.16>;tag=as6b37a291
To: <sip:8006170141@host41.com>;tag=as522fefca
Contact: <sip:4072930335@66.250.69.16>
Call-ID: 1067c5fb7a827ecc1a6f50421d1fc233@66.250.69.16
CSeq: 102 ACK
User-Agent: asterlink/1.2
Content-Length: 0


--- (9 headers 0 lines)---
We're at 72.232.182.13 port 14066
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 10 lines
Reliably Transmitting (no NAT) to 72.40.10.150:5060:
INVITE sip:200@72.40.10.150:5060 SIP/2.0
Via: SIP/2.0/UDP 72.232.182.13:5060;branch=z9hG4bK4fd8c5e7;rport
From: "4072930335" <sip:4072930335@72.232.182.13>;tag=as4aa16d7d
To: <sip:200@72.40.10.150:5060>
Contact: <sip:4072930335@72.232.182.13>
Call-ID: 08c965242994ae18514f92cd2c964b17@72.232.182.13
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 11 Jul 2006 17:30:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 216

v=0
o=root 3534 3534 IN IP4 72.232.182.13
s=session
c=IN IP4 72.232.182.13
t=0 0
m=audio 14066 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
Jul 11 13:30:46 WARNING[1949]: interface.c:215 decodeMP3: Junk at the beginning of frame 49443303
aries*CLI>
<-- SIP read from 72.40.10.150:5060:
SIP/2.0 404 Not Found
To: <sip:200@72.40.10.150:5060>;tag=656eae54128438bbi0
From: "4072930335" <sip:4072930335@72.232.182.13>;tag=as4aa16d7d
Call-ID: 08c965242994ae18514f92cd2c964b17@72.232.182.13
CSeq: 102 INVITE
Via: SIP/2.0/UDP 72.232.182.13:5060;branch=z9hG4bK4fd8c5e7
Server: Linksys/PAP2-3.1.3(LS)
Content-Length: 0


--- (8 headers 0 lines)---
Transmitting (no NAT) to 72.40.10.150:5060:
ACK sip:200@72.40.10.150:5060 SIP/2.0
Via: SIP/2.0/UDP 72.232.182.13:5060;branch=z9hG4bK4fd8c5e7;rport
From: "4072930335" <sip:4072930335@72.232.182.13>;tag=as4aa16d7d
To: <sip:200@72.40.10.150:5060>;tag=656eae54128438bbi0
Contact: <sip:4072930335@72.232.182.13>
Call-ID: 08c965242994ae18514f92cd2c964b17@72.232.182.13
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Destroying call '08c965242994ae18514f92cd2c964b17@72.232.182.13'
aries*CLI>