rendered paste body
Verbosity is at least 10
-- Remote UNIX connection
Reliably Transmitting (no NAT) to 64.61.93.87:5060:
OPTIONS sip:connect01.voicepulse.com SIP/2.0
Via: SIP/2.0/UDP 10.18.18.94:5060;branch=z9hG4bK24de7733;rport
From: "asterisk" <sip:asterisk@10.18.18.94>;tag=as54f4eebb
To: <sip:connect01.voicepulse.com>
Contact: <sip:asterisk@10.18.18.94>
Call-ID: 59af2972309e712b3f304d4a457f2f9b@10.18.18.94
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 26 Jun 2007 22:00:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
---
asteriskpbx*CLI>
<--- SIP read from 64.61.93.87:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.18.18.94:5060;branch=z9hG4bK24de7733;received=216.207.245.1;rport=5060
From: "asterisk" <sip:asterisk@10.18.18.94>;tag=as54f4eebb
To: <sip:connect01.voicepulse.com>;tag=as4dd37b10
Call-ID: 59af2972309e712b3f304d4a457f2f9b@10.18.18.94
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:64.61.93.87>
Accept: application/sdp
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
[Jun 26 22:00:27] NOTICE[314]: chan_sip.c:12280 handle_response_peerpoke: Peer 'trunk_1' is now Reachable. (43ms / 2000ms)
Really destroying SIP dialog '59af2972309e712b3f304d4a457f2f9b@10.18.18.94' Method: OPTIONS
asteriskpbx*CLI>
<--- SIP read from 192.168.69.52:5060 --->
INVITE sip:914232405159@192.168.69.1:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.69.52;branch=z9hG4bK3d1d9db1740104D6
From: "New User" <sip:6001@192.168.69.1>;tag=3AC39282-E22FAE3
To: <sip:914232405159@192.168.69.1;user=phone>
CSeq: 1 INVITE
Call-ID: 6c383de7-b0638734-14af0dcd@192.168.69.52
Contact: <sip:6001@192.168.69.52>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundStationIP-SSIP_4000-UA/2.1.1.0052
Supported: 100rel,replaces
Allow-Events: talk,hold,conference
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 225
v=0
o=- 1182895236 1182895236 IN IP4 192.168.69.52
s=Polycom IP Phone
c=IN IP4 192.168.69.52
t=0 0
m=audio 2234 RTP/AVP 0 8 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
<------------->
--- (14 headers 10 lines) ---
Sending to 192.168.69.52 : 5060 (no NAT)
Using INVITE request as basis request - 6c383de7-b0638734-14af0dcd@192.168.69.52
Found peer '6001'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 192.168.69.52:2234
Found description format PCMU for ID 0
Found description format PCMA for ID 8
Found description format telephone-event for ID 101
Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.69.52:2234
Looking for 914232405159 in numberplan-custom-1 (domain 192.168.69.1)
list_route: hop: <sip:6001@192.168.69.52>
<--- Transmitting (no NAT) to 192.168.69.52:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.69.52;branch=z9hG4bK3d1d9db1740104D6;received=192.168.69.52
From: "New User" <sip:6001@192.168.69.1>;tag=3AC39282-E22FAE3
To: <sip:914232405159@192.168.69.1;user=phone>
Call-ID: 6c383de7-b0638734-14af0dcd@192.168.69.52
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:914232405159@192.168.69.1>
Access-URL:
Content-Length: 0
<------------>
-- Executing [914232405159@numberplan-custom-1:1] Macro("SIP/6001-008ea26c", "trunkdial|SIP/trunk_1/14232405159") in new stack
-- Executing [s@macro-trunkdial:1] Dial("SIP/6001-008ea26c", "SIP/trunk_1/14232405159") in new stack
Audio is at 10.18.18.94 port 17640
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x10 (g726aal2) to SDP
Adding codec 0x40 (slin) to SDP
Adding codec 0x800 (g726) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 64.61.93.87:5060:
INVITE sip:14232405159@connect01.voicepulse.com SIP/2.0
Via: SIP/2.0/UDP 10.18.18.94:5060;branch=z9hG4bK4b7c7dd2;rport
From: "New User" <sip:6001@10.18.18.94>;tag=as46e9f30d
To: <sip:14232405159@connect01.voicepulse.com>
Contact: <sip:6001@10.18.18.94>
Call-ID: 6586323f20b32df879fa4923728fa9ca@10.18.18.94
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 26 Jun 2007 22:00:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 373
v=0
o=root 893 893 IN IP4 10.18.18.94
s=session
c=IN IP4 10.18.18.94
t=0 0
m=audio 17640 RTP/AVP 0 3 8 112 10 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:10 L16/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called trunk_1/14232405159
asteriskpbx*CLI>
<--- SIP read from 64.61.93.87:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.18.18.94:5060;branch=z9hG4bK4b7c7dd2;received=216.207.245.1;rport=5060
From: "New User" <sip:6001@10.18.18.94>;tag=as46e9f30d
To: <sip:14232405159@connect01.voicepulse.com>;tag=as1f0ace78
Call-ID: 6586323f20b32df879fa4923728fa9ca@10.18.18.94
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:14232405159@64.61.93.87>
Proxy-Authenticate: Digest realm="asterisk", nonce="3e54ed8a"
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Transmitting (no NAT) to 64.61.93.87:5060:
ACK sip:14232405159@connect01.voicepulse.com SIP/2.0
Via: SIP/2.0/UDP 10.18.18.94:5060;branch=z9hG4bK4b7c7dd2;rport
From: "New User" <sip:6001@10.18.18.94>;tag=as46e9f30d
To: <sip:14232405159@connect01.voicepulse.com>;tag=as1f0ace78
Contact: <sip:6001@10.18.18.94>
Call-ID: 6586323f20b32df879fa4923728fa9ca@10.18.18.94
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Access-URL:
Content-Length: 0
---
Audio is at 10.18.18.94 port 17640
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x10 (g726aal2) to SDP
Adding codec 0x40 (slin) to SDP
Adding codec 0x800 (g726) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 64.61.93.87:5060:
INVITE sip:14232405159@connect01.voicepulse.com SIP/2.0
Via: SIP/2.0/UDP 10.18.18.94:5060;branch=z9hG4bK1d093484;rport
From: "New User" <sip:6001@10.18.18.94>;tag=as46e9f30d
To: <sip:14232405159@connect01.voicepulse.com>
Contact: <sip:6001@10.18.18.94>
Call-ID: 6586323f20b32df879fa4923728fa9ca@10.18.18.94
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Access-URL:
Proxy-Authorization: Digest username="ueD6XhR29T", realm="asterisk", algorithm=MD5, uri="sip:14232405159@connect01.voicepulse.com", nonce="3e54ed8a", response="1d453577ab427d302df38a39327c8ed5", opaque=""
Date: Tue, 26 Jun 2007 22:00:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 373
v=0
o=root 893 894 IN IP4 10.18.18.94
s=session
c=IN IP4 10.18.18.94
t=0 0
m=audio 17640 RTP/AVP 0 3 8 112 10 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:10 L16/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
asteriskpbx*CLI>
<--- SIP read from 64.61.93.87:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.18.18.94:5060;branch=z9hG4bK1d093484;received=216.207.245.1;rport=5060
From: "New User" <sip:6001@10.18.18.94>;tag=as46e9f30d
To: <sip:14232405159@connect01.voicepulse.com>;tag=as1f0ace78
Call-ID: 6586323f20b32df879fa4923728fa9ca@10.18.18.94
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:14232405159@64.61.93.87>
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Transmitting (no NAT) to 64.61.93.87:5060:
ACK sip:14232405159@connect01.voicepulse.com SIP/2.0
Via: SIP/2.0/UDP 10.18.18.94:5060;branch=z9hG4bK1d093484;rport
From: "New User" <sip:6001@10.18.18.94>;tag=as46e9f30d
To: <sip:14232405159@connect01.voicepulse.com>;tag=as1f0ace78
Contact: <sip:6001@10.18.18.94>
Call-ID: 6586323f20b32df879fa4923728fa9ca@10.18.18.94
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Access-URL:
Content-Length: 0
---
-- SIP/trunk_1-008eb7f0 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [s@macro-trunkdial:2] Goto("SIP/6001-008ea26c", "s-CONGESTION|1") in new stack
-- Goto (macro-trunkdial,s-CONGESTION,1)
-- Executing [s-CONGESTION@macro-trunkdial:1] NoOp("SIP/6001-008ea26c", "") in new stack
== Auto fallthrough, channel 'SIP/6001-008ea26c' status is 'CONGESTION'
<--- Transmitting (no NAT) to 192.168.69.52:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.69.52;branch=z9hG4bK3d1d9db1740104D6;received=192.168.69.52
From: "New User" <sip:6001@192.168.69.1>;tag=3AC39282-E22FAE3
To: <sip:914232405159@192.168.69.1;user=phone>;tag=as6ae5e9e7
Call-ID: 6c383de7-b0638734-14af0dcd@192.168.69.52
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:914232405159@192.168.69.1>
Access-URL:
Content-Length: 0
X-Asterisk-HangupCause: Unallocated (unassigned) number
X-Asterisk-HangupCauseCode: 1
<------------>
Really destroying SIP dialog '6586323f20b32df879fa4923728fa9ca@10.18.18.94' Method: INVITE
Really destroying SIP dialog '6c383de7-b0638734-14af0dcd@192.168.69.52' Method: INVITE
asteriskpbx*CLI>
<--- SIP read from 192.168.69.52:5060 --->
ACK sip:914232405159@192.168.69.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.69.52;branch=z9hG4bK3d1d9db1740104D6
From: "New User" <sip:6001@192.168.69.1>;tag=3AC39282-E22FAE3
To: <sip:914232405159@192.168.69.1;user=phone>;tag=as6ae5e9e7
CSeq: 1 ACK
Call-ID: 6c383de7-b0638734-14af0dcd@192.168.69.52
Contact: <sip:6001@192.168.69.52>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundStationIP-SSIP_4000-UA/2.1.1.0052
Max-Forwards: 70
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---