All pastes #78794 Raw Edit

Unnamed

public text v1 · immutable
·published 2006-07-04 15:46 UTC
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backupPBX*CLI> 
<-- SIP read from 192.168.0.59:5060: 
INVITE sip:123@192.168.0.25;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.59:5060
From: <sip:12@192.168.0.25;user=phone>;tag=920185343
To: <sip:123@192.168.0.25;user=phone>
Call-ID: 2153418752@192.168.0.59
CSeq: 1 INVITE
Contact: <sip:12@192.168.0.59:5060;user=phone;transport=udp>
User-Agent: Cisco ATA 186  v2.16.2 ata18x (030909a)
Expires: 300
Content-Length: 245
Content-Type: application/sdp

v=0
o=12 7905 7905 IN IP4 192.168.0.59
s=ATA186 Call
c=IN IP4 192.168.0.59
t=0 0
m=audio 16384 RTP/AVP 18 8 0 101
a=rtpmap:18 G729/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

--- (11 headers 11 lines)---
Using INVITE request as basis request - 2153418752@192.168.0.59
Sending to 192.168.0.59 : 5060 (non-NAT)
Reliably Transmitting (no NAT) to 192.168.0.59:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.0.59:5060;received=192.168.0.59
From: <sip:12@192.168.0.25;user=phone>;tag=920185343
To: <sip:123@192.168.0.25;user=phone>;tag=as259d571b
Call-ID: 2153418752@192.168.0.59
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:123@192.168.0.25>
Proxy-Authenticate: Digest realm="asterisk", nonce="5fa099b4"
Content-Length: 0


---
Scheduling destruction of call '2153418752@192.168.0.59' in 15000 ms
Found user '12'
backupPBX*CLI> 
<-- SIP read from 192.168.0.59:5060: 
ACK sip:123@192.168.0.25;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.59:5060;received=192.168.0.59
From: <sip:12@192.168.0.25;user=phone>;tag=920185343
To: <sip:123@192.168.0.25;user=phone>;tag=as259d571b
Call-ID: 2153418752@192.168.0.59
CSeq: 1 ACK
User-Agent: Cisco ATA 186  v2.16.2 ata18x (030909a)
Content-Length: 0


--- (8 headers 0 lines)---
backupPBX*CLI> 
<-- SIP read from 192.168.0.59:5060: 
INVITE sip:123@192.168.0.25;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.59:5060
From: <sip:12@192.168.0.25;user=phone>;tag=920185343
To: <sip:123@192.168.0.25;user=phone>
Call-ID: 2153418752@192.168.0.59
CSeq: 2 INVITE
Contact: <sip:12@192.168.0.59:5060;user=phone;transport=udp>
User-Agent: Cisco ATA 186  v2.16.2 ata18x (030909a)
Proxy-Authorization: Digest username="12",realm="asterisk",nonce="5fa099b4",uri="sip:123@192.168.0.25",response="ea44407c623ffceac38ac33b63c53f3e"
Expires: 300
Content-Length: 245
Content-Type: application/sdp

v=0
o=12 7907 7907 IN IP4 192.168.0.59
s=ATA186 Call
c=IN IP4 192.168.0.59
t=0 0
m=audio 16384 RTP/AVP 18 8 0 101
a=rtpmap:18 G729/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

--- (12 headers 11 lines)---
Using INVITE request as basis request - 2153418752@192.168.0.59
Sending to 192.168.0.59 : 5060 (non-NAT)
Found user '12'
Found RTP audio format 18
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 192.168.0.59:16384
Found description format G729
Found description format PCMA
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0x1f07ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|jpeg|png|h261|h263|h263p), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 123 in mycontext (domain 192.168.0.25)
list_route: hop: <sip:12@192.168.0.59:5060;user=phone;transport=udp>
Transmitting (no NAT) to 192.168.0.59:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.59:5060;received=192.168.0.59
From: <sip:12@192.168.0.25;user=phone>;tag=920185343
To: <sip:123@192.168.0.25;user=phone>
Call-ID: 2153418752@192.168.0.59
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:123@192.168.0.25>
Content-Length: 0


---
We're at 192.168.0.25 port 12296
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.0.59:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.59:5060;received=192.168.0.59
From: <sip:12@192.168.0.25;user=phone>;tag=920185343
To: <sip:123@192.168.0.25;user=phone>;tag=as4843cf8d
Call-ID: 2153418752@192.168.0.59
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:123@192.168.0.25>
Content-Type: application/sdp
Content-Length: 285

v=0
o=root 6796 6796 IN IP4 192.168.0.25
s=session
c=IN IP4 192.168.0.25
t=0 0
m=audio 12296 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
backupPBX*CLI> 
<-- SIP read from 192.168.0.59:5060: 
ACK sip:123@192.168.0.25 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.59:5060
From: <sip:12@192.168.0.25;user=phone>;tag=920185343
To: <sip:123@192.168.0.25;user=phone>;tag=as4843cf8d
Call-ID: 2153418752@192.168.0.59
CSeq: 2 ACK
User-Agent: Cisco ATA 186  v2.16.2 ata18x (030909a)
Proxy-Authorization: Digest username="12",realm="asterisk",nonce="5fa099b4",uri="sip:123@192.168.0.25",response="ea44407c623ffceac38ac33b63c53f3e"
Content-Length: 0


--- (9 headers 0 lines)---
set_destination: Parsing <sip:12@192.168.0.59:5060;user=phone;transport=udp> for address/port to send to
set_destination: set destination to 192.168.0.59, port 5060
Reliably Transmitting (no NAT) to 192.168.0.59:5060:
CANCEL :123@192.168.0.25 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.25:5060;branch=z9hG4bK2b86a88a;rport
From: <sip:123@192.168.0.25;user=phone>
To: <sip:12@192.168.0.25;user=phone>;tag=920185343
Contact: <sip:123@192.168.0.25>
Call-ID: 2153418752@192.168.0.59
CSeq: 101 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Scheduling destruction of call '2153418752@192.168.0.59' in 32000 ms
Retransmitting #1 (no NAT) to 192.168.0.59:5060:
CANCEL :123@192.168.0.25 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.25:5060;branch=z9hG4bK2b86a88a;rport
From: <sip:123@192.168.0.25;user=phone>
To: <sip:12@192.168.0.25;user=phone>;tag=920185343
Contact: <sip:123@192.168.0.25>
Call-ID: 2153418752@192.168.0.59
CSeq: 101 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Retransmitting #2 (no NAT) to 192.168.0.59:5060:
CANCEL :123@192.168.0.25 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.25:5060;branch=z9hG4bK2b86a88a;rport
From: <sip:123@192.168.0.25;user=phone>
To: <sip:12@192.168.0.25;user=phone>;tag=920185343
Contact: <sip:123@192.168.0.25>
Call-ID: 2153418752@192.168.0.59
CSeq: 101 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Retransmitting #3 (no NAT) to 192.168.0.59:5060:
CANCEL :123@192.168.0.25 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.25:5060;branch=z9hG4bK2b86a88a;rport
From: <sip:123@192.168.0.25;user=phone>
To: <sip:12@192.168.0.25;user=phone>;tag=920185343
Contact: <sip:123@192.168.0.25>
Call-ID: 2153418752@192.168.0.59
CSeq: 101 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Retransmitting #4 (no NAT) to 192.168.0.59:5060:
CANCEL :123@192.168.0.25 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.25:5060;branch=z9hG4bK2b86a88a;rport
From: <sip:123@192.168.0.25;user=phone>
To: <sip:12@192.168.0.25;user=phone>;tag=920185343
Contact: <sip:123@192.168.0.25>
Call-ID: 2153418752@192.168.0.59
CSeq: 101 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Retransmitting #5 (no NAT) to 192.168.0.59:5060:
CANCEL :123@192.168.0.25 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.25:5060;branch=z9hG4bK2b86a88a;rport
From: <sip:123@192.168.0.25;user=phone>
To: <sip:12@192.168.0.25;user=phone>;tag=920185343
Contact: <sip:123@192.168.0.25>
Call-ID: 2153418752@192.168.0.59
CSeq: 101 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
backupPBX*CLI> 
<-- SIP read from 192.168.0.59:5060: 
BYE sip:123@192.168.0.25 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.59:5060
From: <sip:12@192.168.0.25;user=phone>;tag=920185343
To: <sip:123@192.168.0.25;user=phone>;tag=as4843cf8d
Call-ID: 2153418752@192.168.0.59
CSeq: 3 BYE
User-Agent: Cisco ATA 186  v2.16.2 ata18x (030909a)
Proxy-Authorization: Digest username="12",realm="asterisk",nonce="5fa099b4",uri="sip:123@192.168.0.25",response="27e636e8556778130daec79ac26fba75"
Content-Length: 0


--- (9 headers 0 lines)---
Sending to 192.168.0.59 : 5060 (non-NAT)
Transmitting (no NAT) to 192.168.0.59:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.59:5060;received=192.168.0.59
From: <sip:12@192.168.0.25;user=phone>;tag=920185343
To: <sip:123@192.168.0.25;user=phone>;tag=as4843cf8d
Call-ID: 2153418752@192.168.0.59
CSeq: 3 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:123@192.168.0.25>
Content-Length: 0


---
Retransmitting #6 (no NAT) to 192.168.0.59:5060:
CANCEL :123@192.168.0.25 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.25:5060;branch=z9hG4bK2b86a88a;rport
From: <sip:123@192.168.0.25;user=phone>
To: <sip:12@192.168.0.25;user=phone>;tag=920185343
Contact: <sip:123@192.168.0.25>
Call-ID: 2153418752@192.168.0.59
CSeq: 101 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Jul  4 12:49:25 WARNING[6786]: chan_sip.c:1217 retrans_pkt: Maximum retries exceeded on transmission 2153418752@192.168.0.59 for seqno 101 (Non-critical Request)
Destroying call '2153418752@192.168.0.59'