SIP and H323 Dial-Peers on same gateway

[swensonj]

Has anyone ever run both SIP and H323 dial-peers on the same gateway in unison.  I'm having to transfer our phones about 400 of them over to the VoIPnow server and since were running H323 on the Call Manager setup we having to run SIP on the VoIPnow server I was wondering if I could use the existing gateways and just run both, and or if this was a possible way to migrate these phones over without to much difficulty.

[Venkatesh Perumal]

SIP and H.323 dial peers can be configured on the same gateway, but call routing between the two types of dial peers is disabled by default. To enable this routing, enter voice service configuration mode and issue the command allow-connections from-type to to-type. Options for both the from-type and the to-type are h323 and sip. Below example shows a router that is configured to allow multiple types of VoIP connections.

Configuring H.323 to SIP Connections

H323-SIP-GW(config)#voice service voip
H323-SIP-GW (conf-voi-serv)# allow-connections h323 to h323
H323-SIP-GW (conf-voi-serv)# allow-connections h323 to sip
H323-SIP-GW (conf-voi-serv)# allow-connections sip to h323
H323-SIP-GW (conf-voi-serv)# allow-connections sip to sip

-Venkatesh

[Nadeem Ahmed]

allowing connection whether its H.323 or SIP it just enable interoperability to communicate with each other through this GW. TDM to IP may not require this command unless we need to allow IP-IP communication(H.323 or SIP). By default h.323-H.323 or SIP-SIP connections are disabled an POTS-to-any and Any-to Pots connection are enabled.

having said above by Venkatest you can enable both protocol but keeping in mind the routing that's it.

Br,

Nadeem AHmed

[swensonj]

All I'm trying to do is allow calls to come and go to the PSTN.  I hadn't thought about as we transfer phones to the VoIPnow system going from SIP to H323 and back again.  The way I have the Call Manager set up right now extension dialing is only done between intercompany calling using translation patterns, everything else goes out to PSTN then back in.  Its how our PRI provider requires.

[Dennis Mink]

If you have the option, ditch H323 between your gateway and turn it into SIP, this generally provides more interoperability.  which doesnt mean H323<>SIP wont work.

I had some issues with SIP early offer and H323 fast start some years ago, and decided to try to avoid the two protocols from working together as much as I can:

http://ciscoshizzle.blogspot.com.au/2012/12/sip-early-offer-cucm-trunk.html

[swensonj]

Getting rid of H323 and the CallManagers is the goal. I do like the CallManager system but it is not affordable for the future for small providers. I just want to be able to continue to use my existing gateways of which there are three and use both protocols while I migrate over to the Inhouse VoIPnow server.  Anyone have a sample or suggested SIP Config for the gateway.

[Dennis Mink]

In that case, you will need H323 dialpeers to your CUCM and SIP dialpeers to your Voip now server. So if you move for example your phones in the 33XX range (for example) from CUCM to the VOIPnow server, you add a route pattern on CUCM to point to the relevant GW and from that gateway you have a 33.. SIP dialpeer to point to the Voipnow server.