All pastes #606364 Raw Edit

Miscellany

public text v1 · immutable
·published 2007-07-06 20:15 UTC
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aphrodite*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.
aphrodite*CLI>
<--- SIP read from 192.168.0.101:5060 --->
INVITE sip:3344439@192.168.0.8:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.101;branch=z9hG4bKab540309B5DA95CA
From: "110" <sip:110@192.168.0.8>;tag=F944654F-A24DF74E
To: <sip:3344439@192.168.0.8;user=phone>
CSeq: 1 INVITE
Call-ID: 8cc09acb-d4533b8d-2697220c@192.168.0.101
Contact: <sip:110@192.168.0.101>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.2.0054
Supported: 100rel,replace
Allow-Events: talk,hold,conference
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 251

v=0
o=- 1183752770 1183752770 IN IP4 192.168.0.101
s=Polycom IP Phone
c=IN IP4 192.168.0.101
t=0 0
a=sendrecv
m=audio 2234 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000

<------------->
--- (14 headers 11 lines) ---
Sending to 192.168.0.101 : 5060 (no NAT)
Using INVITE request as basis request - 8cc09acb-d4533b8d-2697220c@192.168.0.101

<--- Reliably Transmitting (NAT) to 192.168.0.101:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.0.101;branch=z9hG4bKab540309B5DA95CA;received=192.168.0.101
From: "110" <sip:110@192.168.0.8>;tag=F944654F-A24DF74E
To: <sip:3344439@192.168.0.8;user=phone>;tag=as253fd41a
Call-ID: 8cc09acb-d4533b8d-2697220c@192.168.0.101
Seq: 1 INVITE>
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1d83f855"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '8cc09acb-d4533b8d-2697220c@192.168.0.101' in 32000 ms (Method: INVITE)
Found user '110'
aphrodite*CLI>
<--- SIP read from 192.168.0.101:5060 --->
ACK sip:3344439@192.168.0.8:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.101;branch=z9hG4bKab540309B5DA95CA
From: "110" <sip:110@192.168.0.8>;tag=F944654F-A24DF74E
To: <sip:3344439@192.168.0.8;user=phone>;tag=as253fd41a
CSeq: 1 ACK
Call-ID: 8cc09acb-d4533b8d-2697220c@192.168.0.101
Contact: <sip:110@192.168.0.101>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.2.0054
Max-Forwards: 70
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
aphrodite*CLI>
<--- SIP read from 192.168.0.101:5060 --->
INVITE sip:3344439@192.168.0.8:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.101;branch=z9hG4bK7f579590390C9811
From: "110" <sip:110@192.168.0.8>;tag=F944654F-A24DF74E
To: <sip:3344439@192.168.0.8;user=phone>
CSeq: 2 INVITE
Call-ID: 8cc09acb-d4533b8d-2697220c@192.168.0.101
Contact: <sip:110@192.168.0.101>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.2.0054
Supported: 100rel,replace
Allow-Events: talk,hold,conference
Proxy-Authorization: Digest username="110", realm="asterisk", nonce="1d83f855", uri="sip:3344439@192.168.0.8:5060;user=phone", response="8aecf0c1671a5f150d1d54867121a04b", algorithm=MD5
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 251

v=0
o=- 1183752770 1183752770 IN IP4 192.168.0.101
s=Polycom IP Phone
c=IN IP4 192.168.0.101
t=0 0
a=sendrecv
m=audio 2234 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000

<------------->
--- (15 headers 11 lines) ---
Sending to 192.168.0.101 : 5060 (NAT)
Using INVITE request as basis request - 8cc09acb-d4533b8d-2697220c@192.168.0.101
Found user '110'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 192.168.0.101:2234
Found description format PCMU for ID 0
Found description format PCMA for ID 8
Found description format G729 for ID 18
Found description format telephone-event for ID 101
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10c (ulaw|alaw|g729)/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 192.168.0.101:2234
Looking for 3344439 in downstairs (domain 192.168.0.8)

<--- Reliably Transmitting (NAT) to 192.168.0.101:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.0.101;branch=z9hG4bK7f579590390C9811;received=192.168.0.101
From: "110" <sip:110@192.168.0.8>;tag=F944654F-A24DF74E
To: <sip:3344439@192.168.0.8;user=phone>;tag=as253fd41a
Call-ID: 8cc09acb-d4533b8d-2697220c@192.168.0.101
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '8cc09acb-d4533b8d-2697220c@192.168.0.101' in 32000 ms (Method: INVITE)
aphrodite*CLI>
<--- SIP read from 192.168.0.101:5060 --->
ACK sip:3344439@192.168.0.8:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.101;branch=z9hG4bK7f579590390C9811
From: "110" <sip:110@192.168.0.8>;tag=F944654F-A24DF74E
To: <sip:3344439@192.168.0.8;user=phone>;tag=as253fd41a
CSeq: 2 ACK
Call-ID: 8cc09acb-d4533b8d-2697220c@192.168.0.101
Contact: <sip:110@192.168.0.101>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.2.0054
Max-Forwards: 70
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
aphrodite*CLI>