All pastes #350644 Raw Edit

Mine

public text v1 · immutable
·published 2007-02-11 16:31 UTC
rendered paste body
ASTERISK SERVER  = AAA.AAA.AAA.AAA
OUT GOING ROUTE  = BBB.BBB.BBB.BBB
CISCO ATA IP     = CCC.CCC.CCC.CCC
GATEWAY IP       = DDD.DDD.DDD.DDD


Date: Sun, 11 Feb 2007 16:21:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
Destroying call '01aa801751b812ad2e5bc98331e6fcc6@AAA.AAA.AAA.AAA'
    -- SIP Seeding peer from astdb: '101038' at 101038@DDD.DDD.DDD.DDD:51062 for 3600
12 headers, 0 lines
Reliably Transmitting (NAT) to DDD.DDD.DDD.DDD:51062:
OPTIONS sip:101038@CCC.CCC.CCC.CCC:8891;user=phone;transport=udp SIP/2.0
Via: SIP/2.0/UDP AAA.AAA.AAA.AAA:5060;branch=z9hG4bK6468304d;rport
From: "asterisk" <sip:asterisk@AAA.AAA.AAA.AAA>;tag=as1a6cb282
To: <sip:101038@CCC.CCC.CCC.CCC:8891;user=phone;transport=udp>
Contact: <sip:asterisk@AAA.AAA.AAA.AAA>
Call-ID: 250c73cb7eaa91ac017a000d65609fd2@AAA.AAA.AAA.AAA
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 11 Feb 2007 16:21:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
Destroying call '250c73cb7eaa91ac017a000d65609fd2@AAA.AAA.AAA.AAA'
We're at AAA.AAA.AAA.AAA port 15652
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding codec 0x1 (g723) to SDP
Adding codec 0x100 (g729) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x10 (g726) to SDP
Adding codec 0x20 (adpcm) to SDP
Adding codec 0x40 (slin) to SDP
Adding codec 0x80 (lpc10) to SDP
Adding codec 0x200 (speex) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 21 lines
Reliably Transmitting (NAT) to DDD.DDD.DDD.DDD:51062:
INVITE sip:101038@CCC.CCC.CCC.CCC:8891;user=phone;transport=udp SIP/2.0
Via: SIP/2.0/UDP AAA.AAA.AAA.AAA:5060;branch=z9hG4bK17fab459;rport
From: "8009598923" <sip:8009598923@AAA.AAA.AAA.AAA>;tag=as469ca43d
To: <sip:101038@CCC.CCC.CCC.CCC:8891;user=phone;transport=udp>
Contact: <sip:8009598923@AAA.AAA.AAA.AAA>
Call-ID: 41f0016844c24bdf40c5c2c26e852f64@AAA.AAA.AAA.AAA
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 11 Feb 2007 16:21:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 490

v=0
o=root 13205 13205 IN IP4 AAA.AAA.AAA.AAA
s=session
c=IN IP4 AAA.AAA.AAA.AAA
t=0 0
m=audio 15652 RTP/AVP 0 97 4 18 3 8 111 5 10 7 110 101
a=rtpmap:0 PCMU/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:110 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
    -- Called 101038
Retransmitting #1 (NAT) to DDD.DDD.DDD.DDD:51062:
INVITE sip:101038@CCC.CCC.CCC.CCC:8891;user=phone;transport=udp SIP/2.0
Via: SIP/2.0/UDP AAA.AAA.AAA.AAA:5060;branch=z9hG4bK17fab459;rport
From: "8009598923" <sip:8009598923@AAA.AAA.AAA.AAA>;tag=as469ca43d
To: <sip:101038@CCC.CCC.CCC.CCC:8891;user=phone;transport=udp>
Contact: <sip:8009598923@AAA.AAA.AAA.AAA>
Call-ID: 41f0016844c24bdf40c5c2c26e852f64@AAA.AAA.AAA.AAA
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 11 Feb 2007 16:21:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 490

v=0
o=root 13205 13205 IN IP4 AAA.AAA.AAA.AAA
s=session
c=IN IP4 AAA.AAA.AAA.AAA
t=0 0
m=audio 15652 RTP/AVP 0 97 4 18 3 8 111 5 10 7 110 101
a=rtpmap:0 PCMU/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:110 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
Feb 11 10:21:39 NOTICE[13437]: chan_sip.c:11200 handle_request_register: Registration from '143000 <sip:143000@ip-pabx.com:8891>' failed for '70.154.21.166' - Username/auth name mismatch
Retransmitting #2 (NAT) to DDD.DDD.DDD.DDD:51062:
INVITE sip:101038@CCC.CCC.CCC.CCC:8891;user=phone;transport=udp SIP/2.0
Via: SIP/2.0/UDP AAA.AAA.AAA.AAA:5060;branch=z9hG4bK17fab459;rport
From: "8009598923" <sip:8009598923@AAA.AAA.AAA.AAA>;tag=as469ca43d
To: <sip:101038@CCC.CCC.CCC.CCC:8891;user=phone;transport=udp>
Contact: <sip:8009598923@AAA.AAA.AAA.AAA>
Call-ID: 41f0016844c24bdf40c5c2c26e852f64@AAA.AAA.AAA.AAA
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 11 Feb 2007 16:21:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 490

v=0
o=root 13205 13205 IN IP4 AAA.AAA.AAA.AAA
s=session
c=IN IP4 AAA.AAA.AAA.AAA
t=0 0
m=audio 15652 RTP/AVP 0 97 4 18 3 8 111 5 10 7 110 101
a=rtpmap:0 PCMU/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:110 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
Retransmitting #3 (NAT) to DDD.DDD.DDD.DDD:51062:
INVITE sip:101038@CCC.CCC.CCC.CCC:8891;user=phone;transport=udp SIP/2.0
Via: SIP/2.0/UDP AAA.AAA.AAA.AAA:5060;branch=z9hG4bK17fab459;rport
From: "8009598923" <sip:8009598923@AAA.AAA.AAA.AAA>;tag=as469ca43d
To: <sip:101038@CCC.CCC.CCC.CCC:8891;user=phone;transport=udp>
Contact: <sip:8009598923@AAA.AAA.AAA.AAA>
Call-ID: 41f0016844c24bdf40c5c2c26e852f64@AAA.AAA.AAA.AAA
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 11 Feb 2007 16:21:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 490

v=0
o=root 13205 13205 IN IP4 AAA.AAA.AAA.AAA
s=session
c=IN IP4 AAA.AAA.AAA.AAA
t=0 0
m=audio 15652 RTP/AVP 0 97 4 18 3 8 111 5 10 7 110 101
a=rtpmap:0 PCMU/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:110 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
Feb 11 10:21:41 NOTICE[13437]: chan_sip.c:2010 auto_congest: Auto-congesting SIP/101038-0889b200
    -- SIP/101038-0889b200 is circuit-busy
Scheduling destruction of call '41f0016844c24bdf40c5c2c26e852f64@AAA.AAA.AAA.AAA' in 32000 ms
    -- SIP Seeding peer from astdb: '101038' at 101038@DDD.DDD.DDD.DDD:51062 for 3600
12 headers, 0 lines
Reliably Transmitting (NAT) to DDD.DDD.DDD.DDD:51062:
OPTIONS sip:101038@CCC.CCC.CCC.CCC:8891;user=phone;transport=udp SIP/2.0
Via: SIP/2.0/UDP AAA.AAA.AAA.AAA:5060;branch=z9hG4bK648764bc;rport
From: "asterisk" <sip:asterisk@AAA.AAA.AAA.AAA>;tag=as5dd3b2a9
To: <sip:101038@CCC.CCC.CCC.CCC:8891;user=phone;transport=udp>
Contact: <sip:asterisk@AAA.AAA.AAA.AAA>
Call-ID: 7b327fc71584729f14151cbf4cc1f6ff@AAA.AAA.AAA.AAA
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 11 Feb 2007 16:21:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
Destroying call '7b327fc71584729f14151cbf4cc1f6ff@AAA.AAA.AAA.AAA'
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing VoiceMail("SIP/101033-08895cc0", "b101038") in new stack
    -- SIP Seeding peer from astdb: '101033' at 101033@DDD.DDD.DDD.DDD:51062 for 3600
12 headers, 0 lines
Reliably Transmitting (NAT) to DDD.DDD.DDD.DDD:51062:
OPTIONS sip:101033@CCC.CCC.CCC.CCC:8891;user=phone;transport=udp SIP/2.0
Via: SIP/2.0/UDP AAA.AAA.AAA.AAA:5060;branch=z9hG4bK02c3e4fb;rport
From: "asterisk" <sip:asterisk@AAA.AAA.AAA.AAA>;tag=as7f5abd72
To: <sip:101033@CCC.CCC.CCC.CCC:8891;user=phone;transport=udp>
Contact: <sip:asterisk@AAA.AAA.AAA.AAA>
Call-ID: 554f67c4164758093870e0e0141d87e7@AAA.AAA.AAA.AAA
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 11 Feb 2007 16:21:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
Destroying call '554f67c4164758093870e0e0141d87e7@AAA.AAA.AAA.AAA'
Feb 11 10:21:41 WARNING[13914]: app_voicemail.c:2461 leave_voicemail: No entry in voicemail config file for '101038'
    -- Executing Hangup("SIP/101033-08895cc0", "") in new stack
  == Spawn extension (PBX, 101038, 103) exited non-zero on 'SIP/101033-08895cc0'
    -- SIP Seeding peer from astdb: '101033' at 101033@DDD.DDD.DDD.DDD:51062 for 3600
12 headers, 0 lines
Reliably Transmitting (NAT) to DDD.DDD.DDD.DDD:51062:
OPTIONS sip:101033@CCC.CCC.CCC.CCC:8891;user=phone;transport=udp SIP/2.0
Via: SIP/2.0/UDP AAA.AAA.AAA.AAA:5060;branch=z9hG4bK60435e42;rport
From: "asterisk" <sip:asterisk@AAA.AAA.AAA.AAA>;tag=as3cc9888c
To: <sip:101033@CCC.CCC.CCC.CCC:8891;user=phone;transport=udp>
Contact: <sip:asterisk@AAA.AAA.AAA.AAA>
Call-ID: 116b197060933f9f512bfc57233a639c@AAA.AAA.AAA.AAA
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 11 Feb 2007 16:21:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
Destroying call '116b197060933f9f512bfc57233a639c@AAA.AAA.AAA.AAA'




[PBX]
exten => *100,1,DeadAGI,SPSayBalance.pl|${EXTEN}|${ACCOUNTCODE}
exten => 100,1,DeadAGI,SPSayBalance.pl|${EXTEN}|${ACCOUNTCODE}

exten => _1XXXXX,1,Dial(SIP/${EXTEN},1000,tr)
exten => _1XXXXX,2,Voicemail(b${EXTEN})
exten => _1XXXXX,3,Hangup
exten => _1XXXXX,102,Voicemail(b${EXTEN})
exten => _1XXXXX,103,Hangup

exten => _2XXXXX,1,Dial(SIP/${EXTEN},1000,tr)
exten => _2XXXXX,2,Voicemail(b${EXTEN})
exten => _2XXXXX,3,Hangup
exten => _2XXXXX,102,Voicemail(b${EXTEN})
exten => _2XXXXX,103,Hangup

exten => _300XXXXXXX,1,Dial(SIP/${EXTEN}@phoneopia.com)
exten => _400XXXXXXX,1,Dial(SIP/${EXTEN}@phoneopia.com)
exten => 111,1,Dial(SIP/101010@ip-pabx.com:8891)
exten => *123,1,Answer
exten => *123,2,Wait(2)
exten => *123,3,VoicemailMain(${CALLERIDNUM})
exten => *100,1,DeadAGI(SayBalance.pl,${CALLERIDNUM},${EXTEN})
exten => _78XXXX,1,Dial(SIP/${EXTEN}@superphone.us,100,tr)
exten => _78XXXX,2,Hangup
exten => _68XXXX,1,Dial(SIP/${EXTEN}@superphone.us,100,tr)
exten => _68XXXX,2,Hangup
;exten => _X.,1,DeadAGI(pbxOut.pl,${CALLERIDNUM},${EXTEN})
;exten => _X.,2,Hangup

exten => _X.,1,DeadAGI(pbxOut.pl,${ACCOUNTCODE},${EXTEN})
exten => _X.,2,Hangup