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mchristo

Asterisk output when calling another phone

Aug 29th, 2015
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  1. == Using SIP RTP CoS mark 5
  2. -- Executing [69511XXXXX@phones:1] Set("SIP/IMSIXXXXXXXXXXXXXXX-00000005", "Name=IMSIYYYYYYYYYYYYYYY") in new stack
  3. -- Executing [69511XXXXX@phones:2] GotoIf("SIP/IMSIXXXXXXXXXXXXXXX-00000005", "0 ?outbound-trunk,69511XXXX,1") in new stack
  4. -- Executing [69511XXXXX@phones:3] Set("SIP/IMSIXXXXXXXXXXXXXXX-00000005", "IPAddr=127.0.0.1") in new stack
  5. -- Executing [69511XXXXX@phones:4] GotoIf("SIP/IMSIXXXXXXXXXXXXXXX-00000005", "0 ?outbound-trunk,69511XXXXX,1") in new stack
  6. -- Executing [69511XXXXX@phones:5] Dial("SIP/IMSIXXXXXXXXXXXXXXX-00000005", "SIP/IMSIYYYYYYYYYYYYYYY@127.0.0.1:5062") in new stack
  7. == Using SIP RTP CoS mark 5
  8. -- Called SIP/IMSIYYYYYYYYYYYYYYY@127.0.0.1:5062
  9. -- SIP/127.0.0.1:5062-00000006 is ringing
  10. -- SIP/127.0.0.1:5062-00000006 answered SIP/IMSIXXXXXXXXXXXXXXX-00000005
  11. > 0x8bb3328 -- Probation passed - setting RTP source address to 127.0.0.1:16516
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