All pastes #84423 Raw Edit

Mine

public text v1 · immutable
·published 2006-07-10 22:24 UTC
rendered paste body
Sending to 66.250.69.11 : 5060 (non-NAT)
Found peer 'asterlink-switch-01'
Found RTP audio format 10
Found RTP audio format 4
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 111
Found RTP audio format 5
Found RTP audio format 7
Found RTP audio format 18
Found RTP audio format 110
Found RTP audio format 97
Found RTP audio format 101
Peer audio RTP is at port 66.250.69.11:23426
Found description format L16
Found description format G723
Found description format GSM
Found description format PCMU
Found description format PCMA
Found description format G726-32
Found description format DVI4
Found description format LPC
Found description format G729
Found description format speex
Found description format iLBC
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x7ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 8006170141 in incoming (domain host41.com)
list_route: hop: <sip:4072930335@66.250.69.11>
Transmitting (no NAT) to 66.250.69.11:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 66.250.69.11:5060;branch=z9hG4bK16f07be1;received=66.250.69.11
From: "4072930335" <sip:4072930335@66.250.69.11>;tag=as4536f66d
To: <sip:8006170141@host41.com>
Call-ID: 798134131272577c3f910c4425ee6cb3@66.250.69.11
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:8006170141@72.232.182.13>
Content-Length: 0


---
Jul 10 18:21:07 NOTICE[29960]: chan_sip.c:5342 sip_reregister:    -- Re-registration for  120983@switch-01.asterlink.com
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 66.250.69.11:5060:
REGISTER sip:switch-01.asterlink.com SIP/2.0
Via: SIP/2.0/UDP 72.232.182.13:5060;branch=z9hG4bK1ec289a1;rport
From: <sip:120983@switch-01.asterlink.com>;tag=as61f0ccdb
To: <sip:120983@switch-01.asterlink.com>
Call-ID: 64d8dd26442a40524a79fe062763ab50@72.232.182.13
CSeq: 1438 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="120983", realm="asterisk", algorithm=MD5, uri="sip:switch-01.asterlink.com", nonce="4c9531cd", response="5f70a8a142e8c4d33a17fd1d58448b0c", opaque=""
Expires: 120
Contact: <sip:120983@72.232.182.13>
Event: registration
Content-Length: 0


---
aries*CLI>
<-- SIP read from 66.250.69.11:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 72.232.182.13:5060;branch=z9hG4bK1ec289a1
From: <sip:120983@switch-01.asterlink.com>;tag=as61f0ccdb
To: <sip:120983@switch-01.asterlink.com>
Call-ID: 64d8dd26442a40524a79fe062763ab50@72.232.182.13
CSeq: 1438 REGISTER
User-Agent: asterlink/1.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:120983@66.250.69.11>
Content-Length: 0


--- (10 headers 0 lines)---
aries*CLI>
<-- SIP read from 66.250.69.11:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 72.232.182.13:5060;branch=z9hG4bK1ec289a1
From: <sip:120983@switch-01.asterlink.com>;tag=as61f0ccdb
To: <sip:120983@switch-01.asterlink.com>;tag=as02ea1de8
Call-ID: 64d8dd26442a40524a79fe062763ab50@72.232.182.13
CSeq: 1438 REGISTER
User-Agent: asterlink/1.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:120983@66.250.69.11>
WWW-Authenticate: Digest realm="asterisk", nonce="036614d7"
Content-Length: 0


--- (11 headers 0 lines)---
Responding to challenge, registration to domain/host name switch-01.asterlink.com
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 66.250.69.11:5060:
REGISTER sip:switch-01.asterlink.com SIP/2.0
Via: SIP/2.0/UDP 72.232.182.13:5060;branch=z9hG4bK7c6ea1e6;rport
From: <sip:120983@switch-01.asterlink.com>;tag=as44a8d573
To: <sip:120983@switch-01.asterlink.com>
Call-ID: 64d8dd26442a40524a79fe062763ab50@72.232.182.13
CSeq: 1439 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="120983", realm="asterisk", algorithm=MD5, uri="sip:switch-01.asterlink.com", nonce="036614d7", response="8fdbbdbd65d16cf1804a429c8a670cb5", opaque=""
Expires: 120
Contact: <sip:120983@72.232.182.13>
Event: registration
Content-Length: 0


---
aries*CLI>
<-- SIP read from 66.250.69.11:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 72.232.182.13:5060;branch=z9hG4bK7c6ea1e6
From: <sip:120983@switch-01.asterlink.com>;tag=as44a8d573
To: <sip:120983@switch-01.asterlink.com>
Call-ID: 64d8dd26442a40524a79fe062763ab50@72.232.182.13
CSeq: 1439 REGISTER
User-Agent: asterlink/1.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:120983@66.250.69.11>
Content-Length: 0


--- (10 headers 0 lines)---
aries*CLI>
<-- SIP read from 66.250.69.11:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 72.232.182.13:5060;branch=z9hG4bK7c6ea1e6
From: <sip:120983@switch-01.asterlink.com>;tag=as44a8d573
To: <sip:120983@switch-01.asterlink.com>;tag=as02ea1de8
Call-ID: 64d8dd26442a40524a79fe062763ab50@72.232.182.13
CSeq: 1439 REGISTER
User-Agent: asterlink/1.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Expires: 120
Contact: <sip:120983@72.232.182.13>;expires=120
Date: Mon, 10 Jul 2006 22:19:46 GMT
Content-Length: 0
aries*CLI>

--- (12 headers 0 lines)---
Scheduling destruction of call '64d8dd26442a40524a79fe062763ab50@72.232.182.13' in 32000 ms
Jul 10 18:21:07 NOTICE[29960]: chan_sip.c:9830 handle_response_register: Outbound Registration: Expiry for switch-01.asterlink.com is 120 sec (Scheduling reregistration in 105 s)
We're at 72.232.182.13 port 10610
Adding codec 0x2 (gsm) to SDP
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 66.250.69.11:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 66.250.69.11:5060;branch=z9hG4bK16f07be1;received=66.250.69.11
From: "4072930335" <sip:4072930335@66.250.69.11>;tag=as4536f66d
To: <sip:8006170141@host41.com>;tag=as13bd9c93
Call-ID: 798134131272577c3f910c4425ee6cb3@66.250.69.11
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:8006170141@72.232.182.13>
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 29951 29951 IN IP4 72.232.182.13
s=session
c=IN IP4 72.232.182.13
t=0 0
m=audio 10610 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
aries*CLI>
<-- SIP read from 66.250.69.11:5060:
ACK sip:8006170141@72.232.182.13 SIP/2.0
Via: SIP/2.0/UDP 66.250.69.11:5060;branch=z9hG4bK1ef7d78c
From: "4072930335" <sip:4072930335@66.250.69.11>;tag=as4536f66d
To: <sip:8006170141@host41.com>;tag=as13bd9c93
Contact: <sip:4072930335@66.250.69.11>
Call-ID: 798134131272577c3f910c4425ee6cb3@66.250.69.11
CSeq: 102 ACK
User-Agent: asterlink/1.2
Content-Length: 0


--- (9 headers 0 lines)---
We're at 72.232.182.13 port 18688
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 10 lines
Reliably Transmitting (no NAT) to 72.40.10.150:5060:
INVITE sip:200@72.40.10.150:5060 SIP/2.0
Via: SIP/2.0/UDP 72.232.182.13:5060;branch=z9hG4bK637df9c1;rport
From: "4072930335" <sip:4072930335@72.232.182.13>;tag=as1df7cef9
To: <sip:200@72.40.10.150:5060>
Contact: <sip:4072930335@72.232.182.13>
Call-ID: 290b3a4b583911136c9a4a224278040b@72.232.182.13
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 10 Jul 2006 22:21:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 218

v=0
o=root 29951 29951 IN IP4 72.232.182.13
s=session
c=IN IP4 72.232.182.13
t=0 0
m=audio 18688 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
Jul 10 18:21:19 WARNING[21690]: res_musiconhold.c:896 local_ast_moh_start: No class: default
aries*CLI>
<-- SIP read from 72.40.10.150:5060:
SIP/2.0 404 Not Found
To: <sip:200@72.40.10.150:5060>;tag=656eae54128438bbi0
From: "4072930335" <sip:4072930335@72.232.182.13>;tag=as1df7cef9
Call-ID: 290b3a4b583911136c9a4a224278040b@72.232.182.13
CSeq: 102 INVITE
Via: SIP/2.0/UDP 72.232.182.13:5060;branch=z9hG4bK637df9c1
Server: Linksys/PAP2-3.1.3(LS)
Content-Length: 0


--- (8 headers 0 lines)---
Transmitting (no NAT) to 72.40.10.150:5060:
ACK sip:200@72.40.10.150:5060 SIP/2.0
Via: SIP/2.0/UDP 72.232.182.13:5060;branch=z9hG4bK637df9c1;rport
From: "4072930335" <sip:4072930335@72.232.182.13>;tag=as1df7cef9
To: <sip:200@72.40.10.150:5060>;tag=656eae54128438bbi0
Contact: <sip:4072930335@72.232.182.13>
Call-ID: 290b3a4b583911136c9a4a224278040b@72.232.182.13
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Destroying call '290b3a4b583911136c9a4a224278040b@72.232.182.13'
aries*CLI>