Cisco 2811 as a VoIP originating SIP Gateway

[hadbihas]

Hello VoIPers,

I would like to use the new Cisco 2811 as a VoIP originating SIP Gateway connected to PSTN using a T1 and the Internet using the FE interface over DSL. Can you please verify if the VWIC-2MFT-T1-DI module with a PVDM2-48 is sufficient enough to start with? I only selected the 2T1 module for scalability later along with another PVDM2-48. Also, can you please provide a sample config on how to configure a 2811 to be a stand alone VoIP SIP gateway (it will be pointed to a 3rd party for IVR who will provide termination as well).

Thanks in advance,

[hattingh]

A VWIC and a PVDM2 are all you need to connect the PSTN yes, and an IOS image of at least IP Voice. On the 2800s wou will need 12.3.8T4 or later.

You do not require the VWIC-2MFT-T1-DI module (although it will certainly also work), the more basic VWIC-1/2MFT-T1 module is good enough for PSTN connectivity. The -DI card is a superset of the more basic T1 card.

The PVDM2-48 will give you 48 channels of G.711, or 24 of G.729A or 18 of G.729. If you need more calls than that, pls add more DSPs. You can use the DSP calculator at http://www.cisco.com/cgi-bin/Support/DSP/cisco_dsp_calc.pl to determine the correct amount of DSPs for your application.

You can find IOS SIP configuration info at:

http://www.cisco.com/univercd/cc/td/doc/product/software/ios123/123cgcr/vvfax_c/callc_c/sip_c/sipc1_c/index.htm

At the very basics, a SIP config looks just like an H.323 one. You have a VoIP dial-peer pointing to the IP address of where you want the calls to go based on the dial-plan. The protocol is just SIP and not H.323 (use the "session protocol sipv2" CLI on the dial-peer).

The last note is jsut to be careful of voice QoS on the Internet. There are no QoS guarantees on Internet links, so the voice quality may vary and the GW has no control over this.

Hope that helps,

CH