CCM5.1 to SIP voip gateway

[tripoli-e]

Running CCM5.1, all SCCP Cisco IP phones.  Have a new device, an  Octtel 4 port VoIP gateway.  The device support SIP trunking and  extensions, and has two FXO and 2 FXS ports, but being unfamiliar with  SIP configuration, I cannot get the device to register to the Call  Manager.  Here's how I'd like the configuration to work:

The SCCP phones on CCM have extension numbers 100-499

The phones on the VoIP gateway have extension numbers 501 and 502

The VoIP gateway and CCM will connect over the public Internet, and both will be behind their respective firewalls

An SCCP phone can call either SIP extension directly, i.e. dialing 501 rings the analog phone on FXS port 1

The  analog phones on the VoIP gateway can likewise call the SCCP phones  directly by extension, e.g. calling 110 rings the SCCP phone with  extension 110

I would also expect voicemail to work properly, if enabled on the SCCP extensions

What I've done so far:

Created a simple SIP extension in CCM, assigned an extension number

Created an End User in CCM, assigned a password, associated the SIP extension

Associated the End User to the SIP extension as well

Created an SIP Trunk, assigned public IP address of the VoIP gateway

What I haven't done:

Created the route for the SIP extensions

The problem:

The  VoIP gateway never registers.  I wouldn't expect the missing route to  prevent registration, but I could be wrong.  If screenshots would be  helpful, they can be provided.  Points will be awarded to those who help towards a solution.

[Christopher Graham]

I'm not quite sure I understand. From what you've said, this is what you have:

IP phone--sccp--CUCM---SIP (over public internet)--Voip gateway--analog phone

Is this correct?  If you are just configuring a sip trunk between cucm and the voip gateway, there really isn't a 'registration' that takes place.  You are just configuring the call routing for CUCM to send outbound SIP invites to the voip gateway, and to accept inbound SIP invites from the voip gateway.

Besides, if you can already call between CUCM and the voip gateway, I'm not sure what the issue is.

Let me know if I'm getting this wrong.

[tripoli-e]

Thanks Chris, I should have included a diagram - but yours is correct.  However, the VoIP gateway will not be registered with a service provider, it will only be used to connect to the CUCM.  I believe this is why the vendor requires the device to register to CUCM, using what it calls 'SIP proxy server'.  Since this is a brand-new install, we have not yet been able to pass any calls.

[Jonathan Schulenberg]

If you want this third-party analog gateway to register to CUCM as the SIP proxy you will need to follow instructions such as these: http://www.netcraftsmen.net/component/content/article/70-unified-communications/774-sip-endpoints-in-cisco-communications-manager-call-manager-express-x-lite.html

A route pattern will not be involved in this scenario as you are not creating a SIP trunk. CUCM does not accept registrations on a trunk.

Also, you need to get this working on the local LAN first. It is far more complex to do across the internet because of the application-level awareness (SIP SDP rewriting for example) the firewalls/NAT devices must have. CUCM is not intended to be internet-facing at all by the way. I suggest that you build an IPsec tunnel or other VPN session to connect these two devices without NAT or firewalls.

[tripoli-e]

I currently have the VoIP gateway sitting on the public side of the firewall.  I'll move it inside and see if that helps.  Good to know I won't have to do trunking or route patterns, but the vendor seems to indicate otherwise with it's product page.  I'll follow-up with some results and screenshots.

[tripoli-e]

So, it's working, but I'm only in control of the CallManager side.  However, the vendor switched the VoIP gateway unit without notification, and still maintains that it supports trunking.  You guys have been a great help so far, and if you have any further guidance on setting up the trunking, I'd greatly appreciate it.  Here's the response from the vendor on the other side regarding their Auvtech AVX1200 (IPPBX1200).

The Shanghai office have 48 extensions, and need make call to USA. The 
Shanghai office is a gateway the USA is a Cisco's CallManager. There is two way to 
handlethis: 1, The gateway use 48 SIP extensions register to the Cisco CallManager. But 
when add, modify, delete the extensionsin Shanghai we need change the gateway's and the 
cisco CallManager's configuration, that's very trouble. 2, The gateway and Cisco 
CallManager use sip to Link Establishment. And the CallManager just need configure this: 
when make a call to Shanghai just uset the Link Establishment trunk. That's means when 
you call 500-547 just send the call to gateway use the Link Establishment trunk. That's 
very easy. And i know Cisco's Call Manager can do this, but i don't know how to configure , 
so maybe you need get help from CISCO. 

UPDATE:  It would seem that connecting a 3rd party SIP 'gateway' to a CUCM is rare, based on the lack on information - or at least the overabundance of VoIP Service Provider trunking information.  I've mananged to piece together that I need at least three things in CUCM for a private SIP trunk: 1) The SIP Trunk, 2) The SIP Route Pattern, 3) a standard Route Pattern.  I'm sure I'm missing some details, because while I can see the 5XX route associated to the SIP trunk, in the Route Plan Report, dialing any 5XX number from an SCCP phone results in 'Your call cannot be completed as dialed'.

UPDATE2: I no longer receive the '...call cannot be completed...' message, by setting the standard Route Pattern to use our internal phone Route Partition.  Now, any call to a 5XX extension results in a 5 second pause after dialing, then a busy signal.