I have the following VoIP architecture: I have two sites with a PABX in each of them. The PABX is connected to a E1 interface on a Cisco gateway, using QSIG protocol. The Cisco gateway sends voice calls (RTP) through an IP network to the other Cisco gateway, and this passes the packets to the remote PABX.
Additionally, there is videoconference equipment in each site. The videoconference connects to a BRI interface on the PABX, using H.320. The videoconference is configured to establish a rate of 64 kbps for the data packets.
I would like to know how I have to configure the Cisco gateway (E1 interface, dial-peers for the videoconference numbers, etc.) to maintain the voice call routing functionality on the Cisco gateways and to pass the videoconference data transparently through the Cisco gateways.
No, that's not what I am looking for. I know the basic of how dial-peers work and how to configure a serial interface of an E1 interface. What I want to know if I need some 'special' commands to permit that videoconferencing packets pass the VoIP infrastructure transparently.
I have one configuration that I'm going to attach to the post. I need to confirm that it is OK for my environment. Some things that I want to know are:
- Can the command "isdn incoming-voice voice" affect the videoconferencing stream?
- Do I need to specify the codec clear-channel in the dial-peer? Will this codec be used by default? Is it not neccesary?
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The short answer is that you don't.... That isn't entirely true while at
the same time it kind of is, but for the most part you don't configure
the softkeys. You enable or disable them via TCL. Here is the long
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Phone Models 802.1X support? 802.1x flavor Addtl Comment EAP-MD5 EAP-TLS
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