All pastes #101636 Raw Edit

Mine

public text v1 · immutable
#101636 ·published 2006-07-27 01:08 UTC
rendered paste body
SIP Debugging enabled
[Jul 26 20:02:04]     -- Accepting AUTHENTICATED call from 70.103.254.3:
       > requested format = ulaw,
       > requested prefs = (ulaw),
       > actual format = ulaw,
       > host prefs = (ulaw),
       > priority = mine
[Jul 26 20:02:04]     -- Executing Set("IAX2/voip1-1", "ORIGINAL_EXT=16024897001") in new stack
[Jul 26 20:02:04]     -- Executing GotoIf("IAX2/voip1-1", "1?clients:dial_out") in new stack
[Jul 26 20:02:04]     -- Goto (OUTBOUND,16024897001,3)
[Jul 26 20:02:04]     -- Executing Goto("IAX2/voip1-1", "LKR-CLIENTS|6024897001|1") in new stack
[Jul 26 20:02:04]     -- Goto (LKR-CLIENTS,6024897001,1)
[Jul 26 20:02:04]     -- Sent into invalid extension '6024897001' in context 'LKR-CLIENTS' on IAX2/voip1-1
[Jul 26 20:02:04]     -- Executing Goto("IAX2/voip1-1", "OUTBOUND|16024897001|dial_out") in new stack
[Jul 26 20:02:04]     -- Goto (OUTBOUND,16024897001,4)
[Jul 26 20:02:04]     -- Executing Goto("IAX2/voip1-1", "DIAL_OUT|16024897001|1") in new stack
[Jul 26 20:02:04]     -- Goto (DIAL_OUT,16024897001,1)
[Jul 26 20:02:04]     -- Executing Dial("IAX2/voip1-1", "Sip/69114416024897001@telcommone") in new stack
[Jul 26 20:02:04] We're at 70.103.254.2 port 30548
[Jul 26 20:02:04] Adding codec 0x4 (ulaw) to SDP
[Jul 26 20:02:04] Adding non-codec 0x1 (telephone-event) to SDP
[Jul 26 20:02:04] 13 headers, 10 lines
[Jul 26 20:02:04] Reliably Transmitting (no NAT) to 72.232.115.250:5060:
INVITE sip:69114416024897001@sip1.telcommone.net SIP/2.0
Via: SIP/2.0/UDP 70.103.254.2:5060;branch=z9hG4bK50422d4f;rport
From: "Tim Ferguson" <sip:9167821300@70.103.254.2>;tag=as36b2ff08
To: <sip:69114416024897001@sip1.telcommone.net>
Contact: <sip:9167821300@70.103.254.2>
Call-ID: 3f15c46b7bba31307121f1c078f0b0f0@70.103.254.2
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 27 Jul 2006 03:02:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 214

v=0
o=root 3868 3868 IN IP4 70.103.254.2
s=session
c=IN IP4 70.103.254.2
t=0 0
m=audio 30548 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
[Jul 26 20:02:04]     -- Called 69114416024897001@telcommone
[Jul 26 20:02:04] 
<-- SIP read from 72.232.115.250:5060: 
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 70.103.254.2:5060;branch=z9hG4bK50422d4f;received=70.103.254.2;rport=5060
From: "Tim Ferguson" <sip:9167821300@70.103.254.2>;tag=as36b2ff08
To: <sip:69114416024897001@sip1.telcommone.net>
Call-ID: 3f15c46b7bba31307121f1c078f0b0f0@70.103.254.2
CSeq: 102 INVITE
User-Agent: TelCommOne-GW
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:69114416024897001@72.232.115.250>
Content-Length: 0


[Jul 26 20:02:04] --- (10 headers 0 lines)[Jul 26 20:02:04] ---
[Jul 26 20:02:07] 
<-- SIP read from 72.232.115.250:5060: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP 70.103.254.2:5060;branch=z9hG4bK50422d4f;received=70.103.254.2;rport=5060
From: "Tim Ferguson" <sip:9167821300@70.103.254.2>;tag=as36b2ff08
To: <sip:69114416024897001@sip1.telcommone.net>;tag=as3a4b1f47
Call-ID: 3f15c46b7bba31307121f1c078f0b0f0@70.103.254.2
CSeq: 102 INVITE
User-Agent: TelCommOne-GW
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:69114416024897001@72.232.115.250>
Content-Type: application/sdp
Content-Length: 218

v=0
o=root 2486 2486 IN IP4 72.232.115.250
s=session
c=IN IP4 72.232.115.250
t=0 0
m=audio 19230 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

[Jul 26 20:02:07] --- (11 headers 10 lines)[Jul 26 20:02:07] ---
[Jul 26 20:02:07] Found RTP audio format 0
[Jul 26 20:02:07] Found RTP audio format 101
[Jul 26 20:02:07] Peer audio RTP is at port 72.232.115.250:19230
[Jul 26 20:02:07] Found description format PCMU
[Jul 26 20:02:07] Found description format telephone-event
[Jul 26 20:02:07] Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
[Jul 26 20:02:07] Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Jul 26 20:02:07] list_route: hop: <sip:69114416024897001@72.232.115.250>
[Jul 26 20:02:07] set_destination: Parsing <sip:69114416024897001@72.232.115.250> for address/port to send to
[Jul 26 20:02:07] set_destination: set destination to 72.232.115.250, port 5060
[Jul 26 20:02:07] Transmitting (no NAT) to 72.232.115.250:5060:
ACK sip:69114416024897001@72.232.115.250 SIP/2.0
Via: SIP/2.0/UDP 70.103.254.2:5060;branch=z9hG4bK44ed5f7a;rport
From: "Tim Ferguson" <sip:9167821300@70.103.254.2>;tag=as36b2ff08
To: <sip:69114416024897001@sip1.telcommone.net>;tag=as3a4b1f47
Contact: <sip:9167821300@70.103.254.2>
Call-ID: 3f15c46b7bba31307121f1c078f0b0f0@70.103.254.2
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
[Jul 26 20:02:07]     -- SIP/telcommone-09e43458 answered IAX2/voip1-1
[Jul 26 20:02:10] set_destination: Parsing <sip:69114416024897001@72.232.115.250> for address/port to send to
[Jul 26 20:02:10] set_destination: set destination to 72.232.115.250, port 5060
[Jul 26 20:02:10] Reliably Transmitting (no NAT) to 72.232.115.250:5060:
BYE sip:69114416024897001@72.232.115.250 SIP/2.0
Via: SIP/2.0/UDP 70.103.254.2:5060;branch=z9hG4bK3b843a66;rport
From: "Tim Ferguson" <sip:9167821300@70.103.254.2>;tag=as36b2ff08
To: <sip:69114416024897001@sip1.telcommone.net>;tag=as3a4b1f47
Contact: <sip:9167821300@70.103.254.2>
Call-ID: 3f15c46b7bba31307121f1c078f0b0f0@70.103.254.2
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
[Jul 26 20:02:10]   == Spawn extension (DIAL_OUT, 16024897001, 1) exited non-zero on 'IAX2/voip1-1'
[Jul 26 20:02:10]     -- Hungup 'IAX2/voip1-1'
[Jul 26 20:02:10] 
<-- SIP read from 72.232.115.250:5060: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP 70.103.254.2:5060;branch=z9hG4bK3b843a66;received=70.103.254.2;rport=5060
From: "Tim Ferguson" <sip:9167821300@70.103.254.2>;tag=as36b2ff08
To: <sip:69114416024897001@sip1.telcommone.net>;tag=as3a4b1f47
Call-ID: 3f15c46b7bba31307121f1c078f0b0f0@70.103.254.2
CSeq: 103 BYE
User-Agent: TelCommOne-GW
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:69114416024897001@72.232.115.250>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing


[Jul 26 20:02:10] --- (11 headers 0 lines)[Jul 26 20:02:10] ---
[Jul 26 20:02:10] Destroying call '3f15c46b7bba31307121f1c078f0b0f0@70.103.254.2'
teamics0101*CLI> sip no debug