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-- Starting simple switch on 'Zap/1-1'
-- Executing [914232405159@numberplan-custom-1:1] Macro("Zap/1-1", "trunkdial|SIP/trunk_1/14232405159") in new stack
-- Executing [s@macro-trunkdial:1] Dial("Zap/1-1", "SIP/trunk_1/14232405159") in new stack
Audio is at 10.18.18.94 port 12754
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=z9hG4bK5fbe8397;rport
From: "New User" <sip:666@10.18.18.94>;tag=as4678a846
To: <sip:14232405159@connect01.voicepulse.com>
Contact: <sip:666@10.18.18.94>
Call-ID: 29e9f45c03d6317204453df72034549a@10.18.18.94
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 26 Jun 2007 22:06:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 373
v=0
o=root 923 923 IN IP4 10.18.18.94
s=session
c=IN IP4 10.18.18.94
t=0 0
m=audio 12754 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=z9hG4bK5fbe8397;received=216.207.245.1;rport=5060
From: "New User" <sip:666@10.18.18.94>;tag=as4678a846
To: <sip:14232405159@connect01.voicepulse.com>;tag=as3ff7a6e0
Call-ID: 29e9f45c03d6317204453df72034549a@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="6afabf39"
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=z9hG4bK5fbe8397;rport
From: "New User" <sip:666@10.18.18.94>;tag=as4678a846
To: <sip:14232405159@connect01.voicepulse.com>;tag=as3ff7a6e0
Contact: <sip:666@10.18.18.94>
Call-ID: 29e9f45c03d6317204453df72034549a@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 12754
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=z9hG4bK1a97a88e;rport
From: "New User" <sip:666@10.18.18.94>;tag=as4678a846
To: <sip:14232405159@connect01.voicepulse.com>
Contact: <sip:666@10.18.18.94>
Call-ID: 29e9f45c03d6317204453df72034549a@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="6afabf39", response="d4b71bd9f76105feeaabc666b87c6042", opaque=""
Date: Tue, 26 Jun 2007 22:06:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 373
v=0
o=root 923 924 IN IP4 10.18.18.94
s=session
c=IN IP4 10.18.18.94
t=0 0
m=audio 12754 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=z9hG4bK1a97a88e;received=216.207.245.1;rport=5060
From: "New User" <sip:666@10.18.18.94>;tag=as4678a846
To: <sip:14232405159@connect01.voicepulse.com>;tag=as3ff7a6e0
Call-ID: 29e9f45c03d6317204453df72034549a@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=z9hG4bK1a97a88e;rport
From: "New User" <sip:666@10.18.18.94>;tag=as4678a846
To: <sip:14232405159@connect01.voicepulse.com>;tag=as3ff7a6e0
Contact: <sip:666@10.18.18.94>
Call-ID: 29e9f45c03d6317204453df72034549a@10.18.18.94
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Access-URL:
Content-Length: 0
---
-- SIP/trunk_1-008f06a0 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [s@macro-trunkdial:2] Goto("Zap/1-1", "s-CONGESTION|1") in new stack
-- Goto (macro-trunkdial,s-CONGESTION,1)
-- Executing [s-CONGESTION@macro-trunkdial:1] NoOp("Zap/1-1", "") in new stack
== Auto fallthrough, channel 'Zap/1-1' status is 'CONGESTION'
Really destroying SIP dialog '29e9f45c03d6317204453df72034549a@10.18.18.94' Method: INVITE
-- Hungup 'Zap/1-1'