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Paste #1211: sip debug
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Pasted by chrivers
*CLI>
<-- SIP read from 10.0.0.103:5060:
INVITE sip:1234@10.0.0.3 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.103:5060;branch=z9hG4bK-9fe8d557
From: "Sikkerhed.org" <sip:chrivers@10.0.0.3>;tag=6a8747c231dbdb1fo0
To: <sip:1234@10.0.0.3>
Call-ID: c9f382be-f6b67c53@10.0.0.103
CSeq: 101 INVITE
Max-Forwards: 70
Contact: "Sikkerhed.org" <sip:chrivers@10.0.0.103:5060>
Expires: 240
User-Agent: Linksys/SPA942-5.1.15(a)
Content-Length: 391
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
Content-Type: application/sdp
v=0
o=- 269690 269690 IN IP4 10.0.0.103
s=-
c=IN IP4 10.0.0.103
t=0 0
m=audio 16422 RTP/AVP 0 2 4 8 18 96 97 98 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
--- (14 headers 18 lines) ---
Using INVITE request as basis request - c9f382be-f6b67c53@10.0.0.103
Sending to 10.0.0.103 : 5060 (non-NAT)
Reliably Transmitting (no NAT) to 10.0.0.103:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.0.0.103:5060;branch=z9hG4bK-9fe8d557;received=10.0.0.103
From: "Sikkerhed.org" <sip:chrivers@10.0.0.3>;tag=6a8747c231dbdb1fo0
To: <sip:1234@10.0.0.3>;tag=as18c52318
Call-ID: c9f382be-f6b67c53@10.0.0.103
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="76bff424"
Content-Length: 0
---
Scheduling destruction of call 'c9f382be-f6b67c53@10.0.0.103' in 15000 ms
Found user 'chrivers'
<-- SIP read from 10.0.0.103:5060:
ACK sip:1234@10.0.0.3 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.103:5060;branch=z9hG4bK-9fe8d557
From: "Sikkerhed.org" <sip:chrivers@10.0.0.3>;tag=6a8747c231dbdb1fo0
To: <sip:1234@10.0.0.3>;tag=as18c52318
Call-ID: c9f382be-f6b67c53@10.0.0.103
CSeq: 101 ACK
Max-Forwards: 70
Contact: "Sikkerhed.org" <sip:chrivers@10.0.0.103:5060>
User-Agent: Linksys/SPA942-5.1.15(a)
Content-Length: 0
--- (10 headers 0 lines) ---
<-- SIP read from 10.0.0.103:5060:
INVITE sip:1234@10.0.0.3 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.103:5060;branch=z9hG4bK-e71e76b7
From: "Sikkerhed.org" <sip:chrivers@10.0.0.3>;tag=6a8747c231dbdb1fo0
To: <sip:1234@10.0.0.3>
Call-ID: c9f382be-f6b67c53@10.0.0.103
CSeq: 102 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest username="chrivers",realm="asterisk",nonce="76bff424",uri="sip:1234@10.0.0.3",algorithm=MD5,response="15a054e241de10f8fe0c45758b9b7e75"
Contact: "Sikkerhed.org" <sip:chrivers@10.0.0.103:5060>
Expires: 240
User-Agent: Linksys/SPA942-5.1.15(a)
Content-Length: 391
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
Content-Type: application/sdp
v=0
o=- 269690 269690 IN IP4 10.0.0.103
s=-
c=IN IP4 10.0.0.103
t=0 0
m=audio 16422 RTP/AVP 0 2 4 8 18 96 97 98 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
--- (15 headers 18 lines) ---
Using INVITE request as basis request - c9f382be-f6b67c53@10.0.0.103
Sending to 10.0.0.103 : 5060 (non-NAT)
Found user 'chrivers'
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 4
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 101
Peer audio RTP is at port 10.0.0.103:16422
Found description format PCMU
Found description format G726-32
Found description format G723
Found description format PCMA
Found description format G729a
Found description format G726-40
Found description format G726-24
Found description format G726-16
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x51d (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 1234 in demo2 (domain 10.0.0.3)
list_route: hop: <sip:chrivers@10.0.0.103:5060>
Transmitting (no NAT) to 10.0.0.103:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.0.103:5060;branch=z9hG4bK-e71e76b7;received=10.0.0.103
From: "Sikkerhed.org" <sip:chrivers@10.0.0.3>;tag=6a8747c231dbdb1fo0
To: <sip:1234@10.0.0.3>
Call-ID: c9f382be-f6b67c53@10.0.0.103
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:1234@10.0.0.3>
Content-Length: 0
---
-- Executing Answer("SIP/chrivers-08198fa8", "") in new stack
We're at 10.0.0.3 port 13586
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 10.0.0.103:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.103:5060;branch=z9hG4bK-e71e76b7;received=10.0.0.103
From: "Sikkerhed.org" <sip:chrivers@10.0.0.3>;tag=6a8747c231dbdb1fo0
To: <sip:1234@10.0.0.3>;tag=as4befe3d5
Call-ID: c9f382be-f6b67c53@10.0.0.103
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:1234@10.0.0.3>
Content-Type: application/sdp
Content-Length: 232
v=0
o=root 11515 11515 IN IP4 10.0.0.3
s=session
c=IN IP4 10.0.0.3
t=0 0
m=audio 13586 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
-- Executing BackGround("SIP/chrivers-08198fa8", "demo-moreinfo") in new stack
-- Playing 'demo-moreinfo' (language 'en')
<-- SIP read from 10.0.0.103:5060:
ACK sip:1234@10.0.0.3 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.103:5060;branch=z9hG4bK-5eb6f0b0
From: "Sikkerhed.org" <sip:chrivers@10.0.0.3>;tag=6a8747c231dbdb1fo0
To: <sip:1234@10.0.0.3>;tag=as4befe3d5
Call-ID: c9f382be-f6b67c53@10.0.0.103
CSeq: 102 ACK
Max-Forwards: 70
Proxy-Authorization: Digest username="chrivers",realm="asterisk",nonce="76bff424",uri="sip:1234@10.0.0.3",algorithm=MD5,response="15a054e241de10f8fe0c45758b9b7e75"
Contact: "Sikkerhed.org" <sip:chrivers@10.0.0.103:5060>
User-Agent: Linksys/SPA942-5.1.15(a)
Content-Length: 0
--- (11 headers 0 lines) ---
*CLI>
*CLI> ### NO audio here
No such command '###' (type 'help' for help)
*CLI>
<-- SIP read from 10.0.0.103:5060:
BYE sip:1234@10.0.0.3 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.103:5060;branch=z9hG4bK-b8b79cae
From: "Sikkerhed.org" <sip:chrivers@10.0.0.3>;tag=6a8747c231dbdb1fo0
To: <sip:1234@10.0.0.3>;tag=as4befe3d5
Call-ID: c9f382be-f6b67c53@10.0.0.103
CSeq: 103 BYE
Max-Forwards: 70
Proxy-Authorization: Digest username="chrivers",realm="asterisk",nonce="76bff424",uri="sip:1234@10.0.0.3",algorithm=MD5,response="28efdfb05ef8e7fa057575dfb4de9895"
User-Agent: Linksys/SPA942-5.1.15(a)
Content-Length: 0
--- (10 headers 0 lines) ---
Sending to 10.0.0.103 : 5060 (non-NAT)
Transmitting (no NAT) to 10.0.0.103:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.103:5060;branch=z9hG4bK-b8b79cae;received=10.0.0.103
From: "Sikkerhed.org" <sip:chrivers@10.0.0.3>;tag=6a8747c231dbdb1fo0
To: <sip:1234@10.0.0.3>;tag=as4befe3d5
Call-ID: c9f382be-f6b67c53@10.0.0.103
CSeq: 103 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:1234@10.0.0.3>
Content-Length: 0
---
== Spawn extension (demo2, 1234, 2) exited non-zero on 'SIP/chrivers-08198fa8'
Nov 6 13:40:03 ERROR[11607]: cdr_custom.c:127 custom_log: Unable to re-open master file /var/log/asterisk/cdr-custom/Master.csv : Permission denied
Nov 6 13:40:03 ERROR[11607]: cdr_csv.c:237 csv_log: Unable to re-open master file /var/log/asterisk//cdr-csv//Master.csv : Permission denied
Destroying call 'c9f382be-f6b67c53@10.0.0.103'
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