A Cisco router running Session Initiation Protocol (SIP) is correctly playing an Real Time Protocol (RTP) stream to the PSTN. This stream has a destination address of the LAN interface on the router. If two RTP streams are presented on the same User Datagram Protocol (UDP) port on the router, the router mixes the two audio streams and plays the two streams to the PSTN party. The person on the PSTN leg hears the two audio streams mixed together perfectly. The person on the IP leg is unaware that the PSTN party is hearing a mix of two audio streams.
For a call coming from SIP trunk and terminating on the Cisco CallManager Express IP phone, if several mid-call re-invites come in with different remote addresses and ports, Cisco CallManager Express uses the same local port to listen to these media streams. If the remote side does not shut down the old media stream before switch to new media stream, Cisco CallManager Express automatically mixes them and sends them to the IP phone.
This issue is a slight variation of Cisco bug ID CSCei58858.
This bug is fixed in IOS versions 12.4(4.7)PI3a, 124(4.7)T and beyond.
To resolve this issue, upgrade the Cisco IOS Software Release to 12.4(4.7)T. In this version, these Cisco IOS commands can be issued:
im configuring a H323 E1 voice gateway but whenever i input this command "network-clock select 1 controller E1 0/1/0" i get this message "G.781 based clock selection process is enabled Please unconfigure G.781 based configuration before con...