All pastes #514293 Raw Edit

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#514293 ·published 2007-05-27 17:25 UTC
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 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 67.15.10.28:5060;branch=z9hG4bK0e4d4315;rport
From: "asterisk" <sip:asterisk@67.15.10.28>;tag=as480750a6
To: <sip:101033@CCC.CCC.CCC.CCC:8891;user=phone;transport=udp>
Contact: <sip:asterisk@67.15.10.28>
Call-ID: 538c245814b4ed5c1c0fc44d1fd84fcc@AAA.AAA.AAA.AAA
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 11 Feb 2007 15:53:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
Destroying call '538c245814b4ed5c1c0fc44d1fd84fcc@AAA.AAA.AAA.AAA'
    -- Registered SIP '101023' at 70.154.21.166 port 63189 expires 3600
    -- Saved useragent "IP SIP Phone/2.0.6" for peer 101023
Feb 11 09:54:09 WARNING[13437]: chan_sip.c:1226 retrans_pkt: Maximum retries exceeded on transmission 2244356000@CCC.CCC.CCC.CCC for seqno 2 (Critical Response)
Feb 11 09:54:09 WARNING[13437]: chan_sip.c:1243 retrans_pkt: Hanging up call 2244356000@CCC.CCC.CCC.CCC - no reply to our critical packet.
    -- AGI Script ippbx/pbxOut.pl completed, returning 0
    -- 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=z9hG4bK45612b24;rport
From: "asterisk" <sip:asterisk@AAA.AAA.AAA.AAA>;tag=as1e7b5810
To: <sip:101033@CCC.CCC.CCC.CCC:8891;user=phone;transport=udp>
Contact: <sip:asterisk@AAA.AAA.AAA.AAA>
Call-ID: 3c7a86493666291265544d3066174e4e@AAA.AAA.AAA.AAA
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 11 Feb 2007 15:54:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
Destroying call '3c7a86493666291265544d3066174e4e@AAA.AAA.AAA.AAA'
server1*CLI>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

    -- Executing DeadAGI("SIP/101033-088748e8", "pbxOut.pl|101033|18005558355") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/pbxOut.pl
    -- 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=z9hG4bK15b46b86;rport
From: "asterisk" <sip:asterisk@AAA.AAA.AAA.AAA>;tag=as41ebeadc
To: <sip:101033@CCC.CCC.CCC.CCC:8891;user=phone;transport=udp>
Contact: <sip:asterisk@AAA.AAA.AAA.AAA>
Call-ID: 1f2d628621ab72a8761242e25e232140@AAA.AAA.AAA.AAA
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 11 Feb 2007 15:53:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
Destroying call '1f2d628621ab72a8761242e25e232140@AAA.AAA.AAA.AAA'
  pbxOut.pl|101033|18005558355: [Demo]: 18005558355 : 101
  pbxOut.pl|101033|18005558355: TalkTime is: 2000.
    --  pbxOut.pl|101033|18005558355: [Arfeen]: CallRate: 0.019
    -- AGI Script Executing Application: (SetCIDNum) Options: (8009598923)
    -- AGI Script Executing Application: (DIAL) Options: (SIP/18005558355@BBB.BBB.BBB.BBB|30|HL(63120000:7000:5000))
    -- Limit Data for this call:
    -- - timelimit     = 63120000
    -- - play_warning  = 7000
    -- - play_to_caller= yes
    -- - play_to_callee= no
    -- - warning_freq  = 5000
    -- - start_sound   = UNDEF
    -- - warning_sound = timeleft
    -- - end_sound     = UNDEF
    -- Called 18005558355@BBB.BBB.BBB.BBB
    -- SIP/BBB.BBB.BBB.BBB-08879e28 is making progress passing it to SIP/101033-088748e8
    -- SIP/BBB.BBB.BBB.BBB-08879e28 answered SIP/101033-088748e8
    -- SIP Seeding peer from astdb: '101033' a
b3