Public feed #603068

Something

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·published 2007-07-04 14:26 UTC
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sip*CLI> sip set debug peer trunk_3
SIP Debugging Enabled for IP: 81.23.228.150:5060
sip*CLI> 
<--- SIP read from 81.23.228.150:5060 --->
INVITE sip:s@217.195.248.252 SIP/2.0
Record-Route: <sip:81.23.228.150;lr=on;ftag=as4bf5f515;fcd=yes>
Via: SIP/2.0/UDP 81.23.228.150;branch=z9hG4bKfeca.2aa53653.0
Via: SIP/2.0/UDP 193.67.155.227:5060;branch=z9hG4bK614bbc71;rport=5060
From: "0651455735" <sip:0651455735@gw02-mci.budgetphone.nl>;tag=as4bf5f515
To: <sip:31107142242@budgetphone.nl>
Contact: <sip:0651455735@193.67.155.227>
Call-ID: 3d5db5dd3d09e82161895f764c36a9f5@gw02-mci.budgetphone.nl
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 69
Date: Wed, 04 Jul 2007 14:15:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 392

v=0
o=root 16089 16089 IN IP4 193.67.155.227
s=session
c=IN IP4 83.149.75.105
t=0 0
m=audio 62650 RTP/AVP 3 18 5 8 0 97 110 101
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:5 DVI4/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:110 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

<------------->
--- (15 headers 17 lines) ---
Sending to 81.23.228.150 : 5060 (no NAT)
Using INVITE request as basis request - 3d5db5dd3d09e82161895f764c36a9f5@gw02-mci.budgetphone.nl
Found peer 'trunk_3'
Found RTP audio format 3
Found RTP audio format 18
Found RTP audio format 5
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 97
Found RTP audio format 110
Found RTP audio format 101
Peer audio RTP is at port 83.149.75.105:62650
Found description format GSM for ID 3
Found description format G729 for ID 18
Found description format DVI4 for ID 5
Found description format PCMA for ID 8
Found description format PCMU for ID 0
Found description format iLBC for ID 97
Found description format speex for ID 110
Found description format telephone-event for ID 101
Capabilities: us - 0x3f1fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|jpeg|png|h261|h263|h263p|h264), peer - audio=0x72e (gsm|ulaw|alaw|adpcm|g729|speex|ilbc)/video=0x0 (nothing), combined - 0x72e (gsm|ulaw|alaw|adpcm|g729|speex|ilbc)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 83.149.75.105:62650
Peer video RTP is at port 83.149.75.105:5128
Looking for s in DID_trunk_3 (domain 217.195.248.252)
list_route: hop: <sip:81.23.228.150;lr=on;ftag=as4bf5f515;fcd=yes>

<--- Transmitting (no NAT) to 81.23.228.150:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 81.23.228.150;branch=z9hG4bKfeca.2aa53653.0;received=81.23.228.150
Via: SIP/2.0/UDP 193.67.155.227:5060;branch=z9hG4bK614bbc71;rport=5060
From: "0651455735" <sip:0651455735@gw02-mci.budgetphone.nl>;tag=as4bf5f515
To: <sip:31107142242@budgetphone.nl>
Call-ID: 3d5db5dd3d09e82161895f764c36a9f5@gw02-mci.budgetphone.nl
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:s@217.195.248.252>
Content-Length: 0


<------------>
    -- Executing [s@DID_trunk_3:1] Goto("SIP/31105117619-b4608d48", "default|102|1") in new stack
    -- Goto (default,102,1)
    -- Executing [102@default:1] Macro("SIP/31105117619-b4608d48", "stdexten|102|SIP/102") in new stack
    -- Executing [s@macro-stdexten:1] Dial("SIP/31105117619-b4608d48", "SIP/102|20") in new stack
    -- Called 102
    -- SIP/102-09434fc8 is ringing

<--- Transmitting (no NAT) to 81.23.228.150:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 81.23.228.150;branch=z9hG4bKfeca.2aa53653.0;received=81.23.228.150
Via: SIP/2.0/UDP 193.67.155.227:5060;branch=z9hG4bK614bbc71;rport=5060
From: "0651455735" <sip:0651455735@gw02-mci.budgetphone.nl>;tag=as4bf5f515
To: <sip:31107142242@budgetphone.nl>;tag=as62fba3c0
Call-ID: 3d5db5dd3d09e82161895f764c36a9f5@gw02-mci.budgetphone.nl
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:s@217.195.248.252>
Content-Length: 0


<------------>
sip*CLI> 
<--- SIP read from 81.23.228.150:5060 --->
CANCEL sip:s@217.195.248.252 SIP/2.0
Record-Route: <sip:81.23.228.150;lr=on;ftag=as4bf5f515>
Via: SIP/2.0/UDP 81.23.228.150;branch=z9hG4bKfeca.2aa53653.0
Via: SIP/2.0/UDP 193.67.155.227:5060;branch=z9hG4bK614bbc71;rport=5060
From: "0651455735" <sip:0651455735@gw02-mci.budgetphone.nl>;tag=as4bf5f515
To: <sip:31107142242@budgetphone.nl>
Contact: <sip:0651455735@193.67.155.227>
Call-ID: 3d5db5dd3d09e82161895f764c36a9f5@gw02-mci.budgetphone.nl
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 69
Content-Length: 0


<------------->
--- (12 headers 0 lines) ---
Sending to 81.23.228.150 : 5060 (no NAT)

<--- Reliably Transmitting (no NAT) to 81.23.228.150:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 81.23.228.150;branch=z9hG4bKfeca.2aa53653.0;received=81.23.228.150
Via: SIP/2.0/UDP 193.67.155.227:5060;branch=z9hG4bK614bbc71;rport=5060
From: "0651455735" <sip:0651455735@gw02-mci.budgetphone.nl>;tag=as4bf5f515
To: <sip:31107142242@budgetphone.nl>;tag=as62fba3c0
Call-ID: 3d5db5dd3d09e82161895f764c36a9f5@gw02-mci.budgetphone.nl
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 81.23.228.150:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 81.23.228.150;branch=z9hG4bKfeca.2aa53653.0;received=81.23.228.150
Via: SIP/2.0/UDP 193.67.155.227:5060;branch=z9hG4bK614bbc71;rport=5060
Record-Route: <sip:81.23.228.150;lr=on;ftag=as4bf5f515>
From: "0651455735" <sip:0651455735@gw02-mci.budgetphone.nl>;tag=as4bf5f515
To: <sip:31107142242@budgetphone.nl>;tag=as62fba3c0
Call-ID: 3d5db5dd3d09e82161895f764c36a9f5@gw02-mci.budgetphone.nl
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:s@217.195.248.252>
Content-Length: 0


<------------>
  == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'SIP/31105117619-b4608d48' in macro 'stdexten'
  == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'SIP/31105117619-b4608d48'
sip*CLI> 
<--- SIP read from 81.23.228.150:5060 --->
ACK sip:s@217.195.248.252 SIP/2.0
Via: SIP/2.0/UDP 81.23.228.150;branch=z9hG4bKfeca.2aa53653.0
From: "0651455735" <sip:0651455735@gw02-mci.budgetphone.nl>;tag=as4bf5f515
Call-ID: 3d5db5dd3d09e82161895f764c36a9f5@gw02-mci.budgetphone.nl
To: <sip:31107142242@budgetphone.nl>;tag=as62fba3c0
CSeq: 102 ACK
User-Agent: OpenSer (1.1.0-tls (i386/linux))
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '3d5db5dd3d09e82161895f764c36a9f5@gw02-mci.budgetphone.nl' Method: ACK
  RTP-stats
* Our Receiver:
  SSRC:          0
  Received packets: 0
  Lost packets:  0
  Jitter:                0.0000
  Transit:               0.0000
  RR-count:      0
* Our Sender:
  SSRC:          998330856
  Sent packets:  0
  Lost packets:  0
  Jitter:                0
  SR-count:      0
  RTT:           0.000000
  RTP-stats
* Our Receiver:
  SSRC:          0
  Received packets: 0
  Lost packets:  0
  Jitter:                0.0000
  Transit:               0.0000
  RR-count:      0
* Our Sender:
  SSRC:          463146790
  Sent packets:  0
  Lost packets:  0
  Jitter:                0
  SR-count:      0
  RTT:           0.000000
  RTP-stats
* Our Receiver:
  SSRC:          0
  Received packets: 0
  Lost packets:  0
  Jitter:                0.0000
  Transit:               0.0000
  RR-count:      0
* Our Sender:
  SSRC:          1504204863
  Sent packets:  0
  Lost packets:  0
  Jitter:                0
  SR-count:      0
  RTT:           0.000000
  RTP-stats
* Our Receiver:
  SSRC:          0
  Received packets: 0
  Lost packets:  0
  Jitter:                0.0000
  Transit:               0.0000
  RR-count:      0
* Our Sender:
  SSRC:          2088594371
  Sent packets:  0
  Lost packets:  0
  Jitter:                0
  SR-count:      0
  RTT:           0.000000