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Cisco Bug: CSCub06422 - Call fails as SIP implementation sets local rtp port to 0

Last Modified

Apr 25, 2020

Products (64)

  • Cisco IOS
  • Cisco 861W Integrated Services Router
  • Cisco 892W Integrated Services Router
  • Cisco 1905 Serial Integrated Services Router
  • Cisco 888W Integrated Services Router
  • Cisco AS5400XM Universal Gateway
  • Cisco 2951 Integrated Services Router
  • Cisco VG224 Analog Voice Gateway
  • Cisco VG204 Analog Voice Gateway
  • Cisco 1803 Integrated Services Router
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Known Affected Releases


Description (partial)

Call flow: PSTN---pri---Voice Gateway---sip---SIP server

After running fine for 6-7 days,  calls through voice gateway fail (100% of the calls fail).
On a call that comes in through the PRI, INVITE is sent with "m=audio 0..".
Then, on getting "200 OK" from the other end, gateway disconnects the call.

Router up and running for 6-7 days
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