*CLI> sip debug
SIP Debugging enabled
The 'sip debug' command is deprecated and will be removed in a future release. Please use 'sip set debug' instead.
*CLI>
<--- SIP read from 10.2.0.103:5060 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.2.0.103:5060;branch=z9hG4bK9E4B50C767D1435A
From: <sip:[email protected]>;tag=F33A586F4DD0B6EB
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 9 INVITE
Contact: <sip:[email protected];uniq=CD10C114C9C60E2E62B55B7A37995>
Max-Forwards: 70
Expires: 120
User-Agent: AVM FRITZ!Box Fon WLAN 08.04.15 (Jul 12 2006)
Supported: 100rel,replaces,timer
Allow-Events: telephone-event,refer
Allow: INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE
Content-Type: application/sdp
Accept: application/sdp, multipart/mixed
Accept-Encoding: identity
Content-Length: 373
v=0
o=user 14637199 14637199 IN IP4 10.2.0.103
s=call
c=IN IP4 10.2.0.103
t=1220512044 1220515644
m=audio 7078 RTP/AVP 8 0 2 102 100 99 97 101
a=sendrecv
a=rtpmap:2 G726-32/8000
a=rtpmap:102 G726-32/8000
a=rtpmap:100 G726-40/8000
a=rtpmap:99 G726-24/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
a=rtcp:7079
<------------->
--- (17 headers 16 lines) ---
Sending to 10.2.0.103 : 5060 (no NAT)
Using INVITE request as basis request - [email protected]
<--- Reliably Transmitting (no NAT) to 10.2.0.103:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.2.0.103:5060;branch=z9hG4bK9E4B50C767D1435A;received=10.2.0.103
From: <sip:[email protected]>;tag=F33A586F4DD0B6EB
To: <sip:[email protected]>;tag=as7fa79bbd
Call-ID: [email protected]
CSeq: 9 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3ded203d"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: INVITE)
Found user '440'
<--- SIP read from 10.2.0.103:5060 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.2.0.103:5060;branch=z9hG4bK9E4B50C767D1435A
From: <sip:[email protected]>;tag=F33A586F4DD0B6EB
To: <sip:[email protected]>;tag=as7fa79bbd
Call-ID: [email protected]
CSeq: 9 ACK
User-Agent: AVM FRITZ!Box Fon WLAN 08.04.15 (Jul 12 2006)
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from 10.2.0.103:5060 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.2.0.103:5060;branch=z9hG4bK6AE0F077488D5397
From: <sip:[email protected]>;tag=F33A586F4DD0B6EB
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 10 INVITE
Contact: <sip:[email protected];uniq=CD10C114C9C60E2E62B55B7A37995>
Proxy-Authorization: Digest username="440", realm="asterisk", nonce="3ded203d", uri="sip:[email protected]", response="046aaf4bbf7a5aecb2ba13a2bb830a58", algorithm=MD5
Max-Forwards: 70
Expires: 120
User-Agent: AVM FRITZ!Box Fon WLAN 08.04.15 (Jul 12 2006)
Supported: 100rel,replaces,timer
Allow-Events: telephone-event,refer
Allow: INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE
Content-Type: application/sdp
Accept: application/sdp, multipart/mixed
Accept-Encoding: identity
Content-Length: 373
v=0
o=user 14637199 14637199 IN IP4 10.2.0.103
s=call
c=IN IP4 10.2.0.103
t=1220512044 1220515644
m=audio 7078 RTP/AVP 8 0 2 102 100 99 97 101
a=sendrecv
a=rtpmap:2 G726-32/8000
a=rtpmap:102 G726-32/8000
a=rtpmap:100 G726-40/8000
a=rtpmap:99 G726-24/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
a=rtcp:7079
<------------->
--- (18 headers 16 lines) ---
Sending to 10.2.0.103 : 5060 (no NAT)
Using INVITE request as basis request - [email protected]
Found user '440'
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 102
Found RTP audio format 100
Found RTP audio format 99
Found RTP audio format 97
Found RTP audio format 101
Peer audio RTP is at port 10.2.0.103:7078
Found audio description format G726-32 for ID 2
Found audio description format G726-32 for ID 102
Found unknown media description format G726-40 for ID 100
Found unknown media description format G726-24 for ID 99
Found audio description format iLBC for ID 97
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xc0c (ulaw|alaw|g726|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.2.0.103:7078
Looking for 441 in default (domain 10.2.0.3)
list_route: hop: <sip:[email protected];uniq=CD10C114C9C60E2E62B55B7A37995>
<--- Transmitting (no NAT) to 10.2.0.103:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.2.0.103:5060;branch=z9hG4bK6AE0F077488D5397;received=10.2.0.103
From: <sip:[email protected]>;tag=F33A586F4DD0B6EB
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 10 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:[email protected]>
Content-Length: 0
<------------>
-- Executing [441@default:1] NoOp("SIP/440-08215508", "20080904-090159 =default=441="" <440>*=*[email protected]=") in new stack
-- Executing [441@default:2] Dial("SIP/440-08215508", "SIP/441") in new stack
Audio is at 10.2.0.3 port 13118
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.2.0.103:5060:
INVITE sip:[email protected];uniq=CD10C114C9C60E2E62B55B7A37995 SIP/2.0
Via: SIP/2.0/UDP 10.2.0.3:5060;branch=z9hG4bK39bfa259;rport
From: "440" <sip:[email protected]>;tag=as265a3121
To: <sip:[email protected];uniq=CD10C114C9C60E2E62B55B7A37995>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 04 Sep 2008 07:01:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 277
v=0
o=root 3545 3545 IN IP4 10.2.0.3
s=session
c=IN IP4 10.2.0.3
t=0 0
m=audio 13118 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/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
---
-- Called 441
<--- SIP read from 10.2.0.103:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.2.0.3:5060;branch=z9hG4bK39bfa259;rport=5060;received=10.2.0.3
From: "440" <sip:[email protected]>;tag=as265a3121
To: <sip:[email protected];uniq=CD10C114C9C60E2E62B55B7A37995>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: AVM FRITZ!Box Fon WLAN 08.04.15 (Jul 12 2006)
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from 10.2.0.103:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.2.0.3:5060;branch=z9hG4bK39bfa259;rport=5060;received=10.2.0.3
From: "440" <sip:[email protected]>;tag=as265a3121
To: <sip:[email protected];uniq=CD10C114C9C60E2E62B55B7A37995>;tag=0752098D37866FC7
Call-ID: [email protected]
CSeq: 102 INVITE
Contact: <sip:[email protected];uniq=CD10C114C9C60E2E62B55B7A37995>
User-Agent: AVM FRITZ!Box Fon WLAN 08.04.15 (Jul 12 2006)
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
-- SIP/441-0821d0a8 is ringing
<--- Transmitting (no NAT) to 10.2.0.103:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.2.0.103:5060;branch=z9hG4bK6AE0F077488D5397;received=10.2.0.103
From: <sip:[email protected]>;tag=F33A586F4DD0B6EB
To: <sip:[email protected]>;tag=as1829a929
Call-ID: [email protected]
CSeq: 10 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:[email protected]>
Content-Length: 0
<------------>
<--- SIP read from 10.2.0.103:5060 --->
CANCEL sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.2.0.103:5060;branch=z9hG4bK6AE0F077488D5397
From: <sip:[email protected]>;tag=F33A586F4DD0B6EB
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 10 CANCEL
Proxy-Authorization: Digest username="440", realm="asterisk", nonce="3ded203d", uri="sip:[email protected]", response="c4a0dbfdf9777de0c4d36fe76938739a", algorithm=MD5
Reason: Q.850; cause=16; text="(null)"
Max-Forwards: 70
User-Agent: AVM FRITZ!Box Fon WLAN 08.04.15 (Jul 12 2006)
Supported: 100rel,replaces,timer
Allow-Events: telephone-event,refer
Content-Length: 0
<------------->
--- (13 headers 0 lines) ---
Sending to 10.2.0.103 : 5060 (no NAT)
<--- Reliably Transmitting (no NAT) to 10.2.0.103:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 10.2.0.103:5060;branch=z9hG4bK6AE0F077488D5397;received=10.2.0.103
From: <sip:[email protected]>;tag=F33A586F4DD0B6EB
To: <sip:[email protected]>;tag=as1829a929
Call-ID: [email protected]
CSeq: 10 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------>
<--- Transmitting (no NAT) to 10.2.0.103:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.2.0.103:5060;branch=z9hG4bK6AE0F077488D5397;received=10.2.0.103
From: <sip:[email protected]>;tag=F33A586F4DD0B6EB
To: <sip:[email protected]>;tag=as1829a929
Call-ID: [email protected]
CSeq: 10 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:[email protected]>
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: INVITE)
Reliably Transmitting (no NAT) to 10.2.0.103:5060:
CANCEL sip:[email protected];uniq=CD10C114C9C60E2E62B55B7A37995 SIP/2.0
Via: SIP/2.0/UDP 10.2.0.3:5060;branch=z9hG4bK39bfa259;rport
From: "440" <sip:[email protected]>;tag=as265a3121
To: <sip:[email protected];uniq=CD10C114C9C60E2E62B55B7A37995>
Call-ID: [email protected]
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: INVITE)
== Spawn extension (default, 441, 2) exited non-zero on 'SIP/440-08215508'
<--- SIP read from 10.2.0.103:5060 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.2.0.103:5060;branch=z9hG4bK6AE0F077488D5397
From: <sip:[email protected]>;tag=F33A586F4DD0B6EB
To: <sip:[email protected]>;tag=as1829a929
Call-ID: [email protected]
CSeq: 10 ACK
User-Agent: AVM FRITZ!Box Fon WLAN 08.04.15 (Jul 12 2006)
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '[email protected]' Method: ACK
<--- SIP read from 10.2.0.103:5060 --->
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 10.2.0.3:5060;branch=z9hG4bK39bfa259;rport=5060;received=10.2.0.3
From: "440" <sip:[email protected]>;tag=as265a3121
To: <sip:[email protected];uniq=CD10C114C9C60E2E62B55B7A37995>;tag=0752098D37866FC7
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: AVM FRITZ!Box Fon WLAN 08.04.15 (Jul 12 2006)
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Transmitting (no NAT) to 10.2.0.103:5060:
ACK sip:[email protected];uniq=CD10C114C9C60E2E62B55B7A37995 SIP/2.0
Via: SIP/2.0/UDP 10.2.0.3:5060;branch=z9hG4bK39bfa259;rport
From: "440" <sip:[email protected]>;tag=as265a3121
To: <sip:[email protected];uniq=CD10C114C9C60E2E62B55B7A37995>;tag=0752098D37866FC7
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
<--- SIP read from 10.2.0.103:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.2.0.3:5060;branch=z9hG4bK39bfa259;rport=5060;received=10.2.0.3
From: "440" <sip:[email protected]>;tag=as265a3121
To: <sip:[email protected];uniq=CD10C114C9C60E2E62B55B7A37995>
Call-ID: [email protected]
CSeq: 102 CANCEL
User-Agent: AVM FRITZ!Box Fon WLAN 08.04.15 (Jul 12 2006)
Supported: 100rel,replaces,timer
Allow-Events: telephone-event,refer
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '[email protected]' Method: INVITE
<--- SIP read from 10.2.0.112:5060 --->
OPTIONS sip:10.2.0.3 SIP/2.0
Via: SIP/2.0/UDP 10.2.0.112:5060;branch=z9hG4bK763cc418;rport
From: "asterisk" <sip:[email protected]>;tag=as36c5c4fc
To: <sip:10.2.0.3>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 04 Sep 2008 14:08:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------->
--- (13 headers 0 lines) ---
Looking for s in default (domain 10.2.0.3)
<--- Transmitting (no NAT) to 10.2.0.112:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.2.0.112:5060;branch=z9hG4bK763cc418;received=10.2.0.112;rport=5060
From: "asterisk" <sip:[email protected]>;tag=as36c5c4fc
To: <sip:10.2.0.3>;tag=as70379e33
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:10.2.0.3>
Accept: application/sdp
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: OPTIONS)
sip no debug
SIP Debugging Disabled