[afnog] VoIP

[Bill Woodcock]

    > Please look at the Visio or JPEG and advise accordingly!
    > How can I achieve ability to call PSTN, GSM and also international from the
    > IP network, i.e. A to be able to call B, C or D! 

To summarize in ASCII (and please tell me if I've misunderstood):

You have a Cisco Skinny network (7900 phones running against a legacy Call 
Manager back-end instead of a SIP back-end), with a Skinny-to-FXO gateway 
interconnecting the Skinny network to an Ericsson PBX.  You have tried 
both analog and ISDN FXO ports.

The Ericsson PBX has both analog and digital proprietary Ericsson 
station sets on it, as well as some sort of dial-out to your GSM network, 
and some sort of gateway to an international voice minutes wholesaler.

The entire Ericsson PBX-to-GSM and PBX-to-international part works fine,
and the Ericsson extensions can call each other, and the Skinny devices 
can call each other.  

What's not working is the FXO gateway between the Skinny network and 
everything else.

We presume that if the Skinny phones were able to reach anything on the 
Ericsson PBX, they'd be able to reach everything on the PBX, and the 
problem would be solved.

So this may be stating the obvious, but that would lead me to look really 
closely at the interface between the Ericsson PBX and the Skinny Call 
Manager.  Am I interpreting your diagram correctly, that you've tried both 
analog and ISDN BRI FXO interfaces on the 1760?  (Both Cisco part VIC-2FXO 
or VIC2-2FXO and Cisco part VIC-2BRI-NT/TE or VIC2-2BRI-NT/TE.)  

Have you tried an E&M (Ear & Mouth or Earth & Metal, the ill-defined 
inter-PBX trunking "standard") interface between the 1760 and the 
Ericsson?  The Ericsson line card will be a more expensive one than an 
extension line card, or will have fewer ports, but it will be designed 
expressly for this purpose.  Which isn't to say that it can be made to 
work easily, E&M interfaces are notoriously difficult to configure 
(different ways of passing caller-ID, different ways of sharing ground 
between the devices, different pinouts, et cetera), but at least it's an 
interface that Ericsson _knows_ is supposed to talk to another 
non-Ericsson PBX device, and will give you support for that purpose, 
however mediocre it may be.

Ideally, the handoff between the Ericsson and the Skinny gateway would be 
digital (i.e. ISDN), so that there won't be any loss in voice quality, and 
so that signalling like dialed-digits and caller-ID will be passed 
cleanly, out-of-band of the voice portion of the call.  Is the interface 
that you're plugging into on the Ericsson side actually specifically ISDN, 
and if so, is it the correct "flavor" of ISDN, intended to talk to a 
station device, and is the 1760 set to match the Ericsson's "switch type?" 
(See 
http://www.cisco.com/en/US/products/sw/iosswrel/ps5013/products_feature_guide09186a0080080de8.html#62113
for a list of the switch types and their corresponding configuration 
keywords.)

Have you checked to make sure that the version of IOS you're running on 
the 1760 fully supports the VIC?  Take a look at the table at 
http://www.cisco.com/en/US/tech/tk652/tk653/technologies_tech_note09186a0080111b16.shtml#ps 
to be sure.

Have you turned on debugging on the Cisco when you've tried to pass calls 
through it, and checked for any error messages?

A useful debugging tool in situations like this is to also have an FXS 
card on the gateway, so you can plug in a simple desk-set for debugging 
purposes.  That would allow you to verify call-completion in both 
directions from the gateway, so you could be absolutely sure which side 
the problem is on, and at least simplify it considerably.  It would be 
unfortunate to put a lot of work into debugging the Ericsson connection, 
only to find out that you had a simple configuration error between the 
gateway and the Skinny Call Manager, for instance.

All of this is in answer to your short-term question, which is, I presume, 
what you're looking for.  In the long term, you probably want to download 
a SIP standards-based software load into the 7900 phones, get rid of the 
Call Manager and the gateway, and have everything talk to everything via 
SIP, as God and the IETF intended.  That would get rid of the long chain 
of devices which you're currently stymied by trying to pass calls through, 
and make everything much simpler and more reliable.  Which is what you 
want, as a service provider.

Good luck.  I hope something above proves helpful.

                                -Bill