All pastes #415801 Raw Edit

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#415801 ·published 2007-03-29 21:11 UTC
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*CLI> sip debug
SIP Debugging enabled
*CLI>
<-- SIP read from 88.149.128.60:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 88.149.128.60:5060;branch=z9hG4bK4f5e6239;received=81.75.230.211;rport=5060
From: "055451360" <sip:055451360@squillo.it>;tag=as51ef5b15
To: <sip:s@81.75.230.211>;tag=as2fea9748
Call-ID: 0b0abf1766819e1c6d27c9d91e6359f9@squillo.it
CSeq: 102 INVITE
User-Agent: www.squillo.it
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:055451360@88.149.128.60>
Content-Length: 0


--- (10 headers 0 lines) ---

<-- SIP read from 88.149.128.60:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 88.149.128.60:5060;branch=z9hG4bK4f5e6239;received=81.75.230.211;rport=5060
From: "055451360" <sip:055451360@squillo.it>;tag=as51ef5b15
To: <sip:s@81.75.230.211>;tag=as2fea9748
Call-ID: 0b0abf1766819e1c6d27c9d91e6359f9@squillo.it
CSeq: 102 INVITE
User-Agent: www.squillo.it
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:055451360@88.149.128.60>
Content-Length: 0


--- (10 headers 0 lines) ---
12 headers, 0 lines
Reliably Transmitting (NAT) to 88.149.168.208:5060:
OPTIONS sip:1234@192.168.0.30:5060 SIP/2.0
Via: SIP/2.0/UDP 81.75.230.211:5060;branch=z9hG4bK2a65c93f;rport
From: "asterisk" <sip:asterisk@81.75.230.211>;tag=as7f51e0b2
To: <sip:1234@192.168.0.30:5060>
Contact: <sip:asterisk@81.75.230.211>
Call-ID: 79037f1f2d075175711a0c47476ff400@81.75.230.211
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 29 Mar 2007 21:10:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
12 headers, 0 lines
Reliably Transmitting (NAT) to 88.149.168.208:5060:
OPTIONS sip:7199@192.168.0.30:5060 SIP/2.0
Via: SIP/2.0/UDP 81.75.230.211:5060;branch=z9hG4bK5a895172;rport
From: "asterisk" <sip:asterisk@81.75.230.211>;tag=as79bf5fc0
To: <sip:7199@192.168.0.30:5060>
Contact: <sip:asterisk@81.75.230.211>
Call-ID: 0b3e5ee676d6085e7c43412133d28652@81.75.230.211
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 29 Mar 2007 21:10:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---

<-- SIP read from 88.149.128.60:5060:
INVITE sip:s@81.75.230.211 SIP/2.0
Via: SIP/2.0/UDP 88.149.128.60:5060;branch=z9hG4bK26fd695d;rport
From: "055451360" <sip:055451360@squillo.it>;tag=as684ecc32
To: <sip:s@81.75.230.211>
Contact: <sip:055451360@88.149.128.60>
Call-ID: 0806df6f1d5305df37317c8332aad052@squillo.it
CSeq: 102 INVITE
User-Agent: www.squillo.it
Date: Thu, 29 Mar 2007 21:10:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Type: application/sdp
Content-Length: 341

v=0
o=root 25171 25171 IN IP4 88.149.128.60
s=session
c=IN IP4 88.149.128.60
t=0 0
m=audio 32664 RTP/AVP 8 4 18 97 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

--- (12 headers 15 lines) ---
Using INVITE request as basis request - 0806df6f1d5305df37317c8332aad052@squillo.it
Sending to 88.149.128.60 : 5060 (NAT)
Found peer 'squillo'
Found RTP audio format 8
Found RTP audio format 4
Found RTP audio format 18
Found RTP audio format 97
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 88.149.128.60:32664
Found description format PCMA
Found description format G723
Found description format G729
Found description format iLBC
Found description format GSM
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0x404 (ulaw|ilbc), peer - audio=0x50f (g723|gsm|ulaw|alaw|g729|ilbc)/video=0x0 (nothing), combined - 0x404 (ulaw|ilbc)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for s in default (domain 81.75.230.211)
list_route: hop: <sip:055451360@88.149.128.60>
Transmitting (NAT) to 88.149.128.60:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 88.149.128.60:5060;branch=z9hG4bK26fd695d;received=88.149.128.60;rport=5060
From: "055451360" <sip:055451360@squillo.it>;tag=as684ecc32
To: <sip:s@81.75.230.211>
Call-ID: 0806df6f1d5305df37317c8332aad052@squillo.it
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:s@81.75.230.211>
Content-Length: 0


---
    -- Executing Answer("SIP/criolo-08157860", "") in new stack
We're at 81.75.230.211 port 11916
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 88.149.128.60:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 88.149.128.60:5060;branch=z9hG4bK26fd695d;received=88.149.128.60;rport=5060
From: "055451360" <sip:055451360@squillo.it>;tag=as684ecc32
To: <sip:s@81.75.230.211>;tag=as29a617c0
Call-ID: 0806df6f1d5305df37317c8332aad052@squillo.it
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:s@81.75.230.211>
Content-Type: application/sdp
Content-Length: 244

v=0
o=root 15009 15009 IN IP4 81.75.230.211
s=session
c=IN IP4 81.75.230.211
t=0 0
m=audio 11916 RTP/AVP 0 97 101
a=rtpmap:0 PCMU/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
    -- Executing Dial("SIP/criolo-08157860", "SIP/7199") in new stack
Destroying call '389bf50f75ed13260163c22d1a944b01@192.168.0.2'
Mar 29 23:10:51 NOTICE[15035]: app_dial.c:1056 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing Hangup("SIP/criolo-08157860", "") in new stack
  == Spawn extension (default, s, 3) exited non-zero on 'SIP/criolo-08157860'
Scheduling destruction of call '0806df6f1d5305df37317c8332aad052@squillo.it' in 32000 ms

<-- SIP read from 88.149.128.60:5060:
ACK sip:s@81.75.230.211 SIP/2.0
Via: SIP/2.0/UDP 88.149.128.60:5060;branch=z9hG4bK16442e1d;rport
From: "055451360" <sip:055451360@squillo.it>;tag=as684ecc32
To: <sip:s@81.75.230.211>;tag=as29a617c0
Contact: <sip:055451360@88.149.128.60>
Call-ID: 0806df6f1d5305df37317c8332aad052@squillo.it
CSeq: 102 ACK
User-Agent: www.squillo.it
Content-Length: 0


--- (9 headers 0 lines) ---
set_destination: Parsing <sip:055451360@88.149.128.60> for address/port to send to
set_destination: set destination to 88.149.128.60, port 5060
Reliably Transmitting (NAT) to 88.149.128.60:5060:
CANCEL :s@81.75.230.211 SIP/2.0
Via: SIP/2.0/UDP 81.75.230.211:5060;branch=z9hG4bK19e8983e;rport
From: <sip:s@81.75.230.211>
To: "055451360" <sip:055451360@squillo.it>;tag=as684ecc32
Call-ID: 0806df6f1d5305df37317c8332aad052@squillo.it
CSeq: 101 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Scheduling destruction of call '0806df6f1d5305df37317c8332aad052@squillo.it' in 32000 ms
Retransmitting #1 (NAT) to 88.149.168.208:5060:
OPTIONS sip:1234@192.168.0.30:5060 SIP/2.0
Via: SIP/2.0/UDP 81.75.230.211:5060;branch=z9hG4bK2a65c93f;rport
From: "asterisk" <sip:asterisk@81.75.230.211>;tag=as7f51e0b2
To: <sip:1234@192.168.0.30:5060>
Contact: <sip:asterisk@81.75.230.211>
Call-ID: 79037f1f2d075175711a0c47476ff400@81.75.230.211
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 29 Mar 2007 21:10:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
Retransmitting #1 (NAT) to 88.149.168.208:5060:
OPTIONS sip:7199@192.168.0.30:5060 SIP/2.0
Via: SIP/2.0/UDP 81.75.230.211:5060;branch=z9hG4bK5a895172;rport
From: "asterisk" <sip:asterisk@81.75.230.211>;tag=as79bf5fc0
To: <sip:7199@192.168.0.30:5060>
Contact: <sip:asterisk@81.75.230.211>
Call-ID: 0b3e5ee676d6085e7c43412133d28652@81.75.230.211
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 29 Mar 2007 21:10:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---

<-- SIP read from 88.149.128.60:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 88.149.128.60:5060;branch=z9hG4bK26fd695d;received=81.75.230.211;rport=5060
From: "055451360" <sip:055451360@squillo.it>;tag=as684ecc32
To: <sip:s@81.75.230.211>;tag=as29a617c0
Call-ID: 0806df6f1d5305df37317c8332aad052@squillo.it
CSeq: 102 INVITE
User-Agent: www.squillo.it
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:055451360@81.75.230.211>
Content-Length: 0


--- (10 headers 0 lines) ---

<-- SIP read from 88.149.128.60:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 81.75.230.211:5060;branch=z9hG4bK19e8983e;received=81.75.230.211;rport=5060
From: <sip:s@81.75.230.211>
To: "055451360" <sip:055451360@squillo.it>;tag=as684ecc32
Call-ID: 0806df6f1d5305df37317c8332aad052@squillo.it
CSeq: 101 CANCEL
User-Agent: www.squillo.it
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:055451360@88.149.128.60>
Content-Length: 0


--- (10 headers 0 lines) ---
Retransmitting #2 (NAT) to 88.149.168.208:5060:
OPTIONS sip:1234@192.168.0.30:5060 SIP/2.0
Via: SIP/2.0/UDP 81.75.230.211:5060;branch=z9hG4bK2a65c93f;rport
From: "asterisk" <sip:asterisk@81.75.230.211>;tag=as7f51e0b2
To: <sip:1234@192.168.0.30:5060>
Contact: <sip:asterisk@81.75.230.211>
Call-ID: 79037f1f2d075175711a0c47476ff400@81.75.230.211
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 29 Mar 2007 21:10:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
Retransmitting #2 (NAT) to 88.149.168.208:5060:
OPTIONS sip:7199@192.168.0.30:5060 SIP/2.0
Via: SIP/2.0/UDP 81.75.230.211:5060;branch=z9hG4bK5a895172;rport
From: "asterisk" <sip:asterisk@81.75.230.211>;tag=as79bf5fc0
To: <sip:7199@192.168.0.30:5060>
Contact: <sip:asterisk@81.75.230.211>
Call-ID: 0b3e5ee676d6085e7c43412133d28652@81.75.230.211
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 29 Mar 2007 21:10:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
Retransmitting #3 (NAT) to 88.149.168.208:5060:
OPTIONS sip:1234@192.168.0.30:5060 SIP/2.0
Via: SIP/2.0/UDP 81.75.230.211:5060;branch=z9hG4bK2a65c93f;rport
From: "asterisk" <sip:asterisk@81.75.230.211>;tag=as7f51e0b2
To: <sip:1234@192.168.0.30:5060>
Contact: <sip:asterisk@81.75.230.211>
Call-ID: 79037f1f2d075175711a0c47476ff400@81.75.230.211
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 29 Mar 2007 21:10:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
Retransmitting #3 (NAT) to 88.149.168.208:5060:
OPTIONS sip:7199@192.168.0.30:5060 SIP/2.0
Via: SIP/2.0/UDP 81.75.230.211:5060;branch=z9hG4bK5a895172;rport
From: "asterisk" <sip:asterisk@81.75.230.211>;tag=as79bf5fc0
To: <sip:7199@192.168.0.30:5060>
Contact: <sip:asterisk@81.75.230.211>
Call-ID: 0b3e5ee676d6085e7c43412133d28652@81.75.230.211
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 29 Mar 2007 21:10:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---

<-- SIP read from 88.149.128.60:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 88.149.128.60:5060;branch=z9hG4bK26fd695d;received=81.75.230.211;rport=5060
From: "055451360" <sip:055451360@squillo.it>;tag=as684ecc32
To: <sip:s@81.75.230.211>;tag=as29a617c0
Call-ID: 0806df6f1d5305df37317c8332aad052@squillo.it
CSeq: 102 INVITE
User-Agent: www.squillo.it
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:055451360@88.149.128.60>
Content-Length: 0


--- (10 headers 0 lines) ---
Retransmitting #4 (NAT) to 88.149.168.208:5060:
OPTIONS sip:1234@192.168.0.30:5060 SIP/2.0
Via: SIP/2.0/UDP 81.75.230.211:5060;branch=z9hG4bK2a65c93f;rport
From: "asterisk" <sip:asterisk@81.75.230.211>;tag=as7f51e0b2
To: <sip:1234@192.168.0.30:5060>
Contact: <sip:asterisk@81.75.230.211>
Call-ID: 79037f1f2d075175711a0c47476ff400@81.75.230.211
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 29 Mar 2007 21:10:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
Destroying call '79037f1f2d075175711a0c47476ff400@81.75.230.211'
Retransmitting #4 (NAT) to 88.149.168.208:5060:
OPTIONS sip:7199@192.168.0.30:5060 SIP/2.0
Via: SIP/2.0/UDP 81.75.230.211:5060;branch=z9hG4bK5a895172;rport
From: "asterisk" <sip:asterisk@81.75.230.211>;tag=as79bf5fc0
To: <sip:7199@192.168.0.30:5060>
Contact: <sip:asterisk@81.75.230.211>
Call-ID: 0b3e5ee676d6085e7c43412133d28652@81.75.230.211
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 29 Mar 2007 21:10:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
Destroying call '0b3e5ee676d6085e7c43412133d28652@81.75.230.211'
Beginning asterisk shutdown....
Executing last minute cleanups
  == Destroying musiconhold processes
Asterisk cleanly ending (2).