All pastes #24037 Raw Edit

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public text v1 · immutable
#24037 ·published 2005-09-27 13:34 UTC
rendered paste body
Sip read:
INVITE sip:200@192.168.16.71;transport=UDP;user=phone SIP/2.0
From: "10" <sip:201@192.168.16.70>;tag=4610a8c0-13c4-69b81-19ce5070-5930
To: "200"<sip:200@192.168.16.71>
Call-ID: 1064f598-4610a8c0-13c4-69b81-19ce506e-22e2@192.168.16.70
CSeq: 1 INVITE
Via: SIP/2.0/UDP 192.168.16.70:5060;rport;branch=z9hG4bK-69b81-19ce5070-49a9
Max-Forwards: 70
Supported: replaces,100rel,timer
Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL, OPTIONS
User-Agent: Swissvoice IP10S
Contact: "10" <sip:201@192.168.16.70:5060;transport=UDP>
Session-Expires: 3600
Content-Type: application/SDP
Content-Length: 212

v=0
o=rtp/1 433025 433025 IN IP4 192.168.16.70
s=-
c=IN IP4 192.168.16.70
t=0 0
m=audio 50000 RTP/AVP 18 0 8
a=ptime:40
a=SilenceSupp:on
a=rtpmap:18 g729/8000
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000

14 headers, 11 lines
Using latest request as basis request
Sending to 192.168.16.70 : 5060 (non-NAT)
Reliably Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.16.70:5060;branch=z9hG4bK-69b81-19ce5070-49a9
From: "10" <sip:201@192.168.16.70>;tag=4610a8c0-13c4-69b81-19ce5070-5930
To: "200"<sip:200@192.168.16.71>;tag=as7edd4a42
Call-ID: 1064f598-4610a8c0-13c4-69b81-19ce506e-22e2@192.168.16.70
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:200@192.168.16.71>
Proxy-Authenticate: Digest realm="asterisk", nonce="7236741f"
Content-Length: 0


 to 192.168.16.70:5060
Scheduling destruction of call '1064f598-4610a8c0-13c4-69b81-19ce506e-22e2@192.168.16.70' in 15000 ms
Found user '201'
pbx*CLI>

Sip read:
ACK sip:200@192.168.16.71;transport=UDP SIP/2.0
From: "10" <sip:201@192.168.16.70>;tag=4610a8c0-13c4-69b81-19ce5070-5930
To: "200"<sip:200@192.168.16.71>;tag=as7edd4a42
Call-ID: 1064f598-4610a8c0-13c4-69b81-19ce506e-22e2@192.168.16.70
CSeq: 1 ACK
Via: SIP/2.0/UDP 192.168.16.70:5060;rport;branch=z9hG4bK-69b81-19ce5070-49a9
Max-Forwards: 70
User-Agent: Swissvoice IP10S
Contact: "10" <sip:201@192.168.16.70:5060;transport=UDP>
Content-Length: 0


10 headers, 0 lines
pbx*CLI>

Sip read:
INVITE sip:200@192.168.16.71;transport=UDP;user=phone SIP/2.0
From: "10" <sip:201@192.168.16.70>;tag=4610a8c0-13c4-69b81-19ce5070-5930
To: "200"<sip:200@192.168.16.71>
Call-ID: 1064f598-4610a8c0-13c4-69b81-19ce506e-22e2@192.168.16.70
CSeq: 2 INVITE
Via: SIP/2.0/UDP 192.168.16.70:5060;rport;branch=z9hG4bK-69b81-19ce50fe-76cf
Max-Forwards: 70
Supported: replaces,100rel,timer
Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL, OPTIONS
User-Agent: Swissvoice IP10S
Contact: "10" <sip:201@192.168.16.70:5060;transport=UDP>
Proxy-Authorization: Digest username="201", realm="asterisk", nonce="7236741f", uri="sip:200@192.168.16.71;transport=UDP", response="26226bb482414f37d233eb75fa2b4416", algorithm=MD5
Session-Expires: 3600
Content-Type: application/SDP
Content-Length: 212

v=0
o=rtp/1 433025 433025 IN IP4 192.168.16.70
s=-
c=IN IP4 192.168.16.70
t=0 0
m=audio 50000 RTP/AVP 18 0 8
a=ptime:40
a=SilenceSupp:on
a=rtpmap:18 g729/8000
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000

15 headers, 11 lines
Using latest request as basis request
Sending to 192.168.16.70 : 5060 (non-NAT)
Found user '201'
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 8
Peer audio RTP is at port 192.168.16.70:50000
Found description format g729
Found description format pcmu
Found description format pcma
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined - 0x0 (nothing)
Looking for 200 in sip-users
list_route: hop: <sip:201@192.168.16.70:5060;transport=UDP>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.16.70:5060;branch=z9hG4bK-69b81-19ce50fe-76cf
From: "10" <sip:201@192.168.16.70>;tag=4610a8c0-13c4-69b81-19ce5070-5930
To: "200"<sip:200@192.168.16.71>
Call-ID: 1064f598-4610a8c0-13c4-69b81-19ce506e-22e2@192.168.16.70
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:200@192.168.16.71>
Content-Length: 0


 to 192.168.16.70:5060
Destroying call '5360421d5494a8ae6bce62e33abe5c60@192.168.16.71'
Sep 27 21:36:23 NOTICE[56557]: app_dial.c:777 dial_exec: Unable to create channel of type 'SIP'
pbx*CLI>

Sip read:


0 headers, 0 lines
Sep 27 21:36:33 WARNING[56557]: pbx.c:1952 ast_pbx_run: Timeout, but no rule 't' in context 'sip-users'
Reliably Transmitting (no NAT):
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.16.70:5060;branch=z9hG4bK-69b81-19ce50fe-76cf
From: "10" <sip:201@192.168.16.70>;tag=4610a8c0-13c4-69b81-19ce5070-5930
To: "200"<sip:200@192.168.16.71>;tag=as5501d9cd
Call-ID: 1064f598-4610a8c0-13c4-69b81-19ce506e-22e2@192.168.16.70
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:200@192.168.16.71>
Content-Length: 0


 to 192.168.16.70:5060
pbx*CLI>

Sip read:
ACK sip:200@192.168.16.71;transport=UDP SIP/2.0
From: "10" <sip:201@192.168.16.70>;tag=4610a8c0-13c4-69b81-19ce5070-5930
To: "200"<sip:200@192.168.16.71>;tag=as5501d9cd
Call-ID: 1064f598-4610a8c0-13c4-69b81-19ce506e-22e2@192.168.16.70
CSeq: 2 ACK
Via: SIP/2.0/UDP 192.168.16.70:5060;rport;branch=z9hG4bK-69b81-19ce50fe-76cf
Max-Forwards: 70
User-Agent: Swissvoice IP10S
Contact: "10" <sip:201@192.168.16.70:5060;transport=UDP>
Content-Length: 0


10 headers, 0 lines
Destroying call '1064f598-4610a8c0-13c4-69b81-19ce506e-22e2@192.168.16.70'
pbx*CLI>

Sip read:


0 headers, 0 lines