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The CLAN NR is defined on the IP interface screen. As kyle555 has suggested, it looks like a DSP for another region is selected because all your regions are interconnected. Why exactly this happening isn't clear to me, but will be something like not having DSPs in a particular region, a codec...

After VMM went EoL Avaya shipped a license for Prognosis with CM (but I think they stopped this a while ago). You can get a 30 day trial of Prognosis at voicequality.com. Alternatives are Nectar or Empirix (I don't know if trial versions are available for those).

There are a lot of things to consider. To start with: What type of gateways do you have? What are the near end and far end regions for your signalling group(s) to SM? Are you using IP or TDM phones? If an IP phone, is it definitely registered in NR 10 (check 'list reg', is the phone IP in the...

Have a look in 'system-parameters ip-options' to see if you have the 'IP DTMF TRANSMISSION MODE' settings (page 2 on newer releases).

From a general perspective you'll need two lists - one for voice traffic (phones, medpros, H248 gateways) and one for signalling traffic (servers, IPSIs, CLANs, phones, H248 gateways). This will allow you to use different priorities for the different traffic types. The problem for you is that...

I don't disagree with any of the suggestions here, but can suggest a few alternatives and extra considerations: - You said you'll be using Exchange for voicemail, how are you going to integrate? If you want to use SIP, then you'll need Session Manager. The alternative would be to use a third...

As some have mentioned there is an official version of what you're proposing, I think with IPO 9 the B5800 terminology is gone and it's called 'IP Office Enterprise Branch'. A few points: 1. You have to use SIP endpoints for the centralised model (which I think is what you want). In normal...

A colleague of mine told me about this. Apparently this happens when returning a missed call from the log after 10 mins. I haven't seen it myself, and don't know when it will be fixed, but just thought I'd let you know you're not alone.

9608 Desk phones aliased as 9630 as recommended by Avaya" I think this has changed, the admin manual dated Aug 2010 says 9630, but a newer one (Feb 2012) says to alias as 9650. Does changing that make a difference?

When I last looked at this issue (six months ago) this was broken for 96x1 phones (H323 and SIP). It isn't a settings issue (there are no settings for this feature), it just doesn't work. What firmware version are you running? I would have thought Avaya would have fixed this by now...

No, call type analysis is something different: change calltype analysis Page 1 of x CALL TYPE DIGIT ANALYSIS TABLE Location: all Dialed String Delete Insert Type Delete Insert Type Match: _____________________ 1: ___ __________ 2: ___ ___________...

>per-location dialplan Only allows UDP" Yes, so... Enter 5666,4,UDP in the location dial plan Don't put an entry in the UDP table 5666 should then be sent to the call type analysis table, from there it can be manipulated into whatever number goes to Asterisk

What CM version are you on? If 5 or higher you can have per-location dialplan, you could use that to reclassify 5666. How do you dial in to the Asterisk VM now?

Similar, but not the same. IGAR sends signalling traffic over the IP network, but voice over the PSTN (when bandwidth between regions can't support VoIP). DPT just sends the call over the PSTN, without the IP signalling connection (when an LSP site is isolated from the rest of the system)...

The phones send keepalives to maintain their registration. The settings can be adjusted on the first page of the IP network region form. There's more information on the actual mechanisms used in the Network Region Configuration Guide and IP Telephony Implementation Guide.

Do you have a non-direct connection configured between branch A and branch B with 250 as the intervening region?

Hey, don't get me wrong, I'm not here to defend SSA. It's just that I've been working on similar issues lately (not involving SSA, but using the RTCP data) and found the problems to be what I mentioned earlier. I'm perfectly happy to accept that SSA is crap [smile]

There are a few things to consider here before dismissing the stats out of hand. I believe these figures are coming from RTCP data, so it's possible there could be a phone firmware or IPO software issue that is causing incorrect stats to be generated. Another possibility is that the figures...

Well if Avaya were going to replace TAPI with anything then it should be DMCC, that has (or at least will have) more functionality than JTAPI/TSAPI. It has Java and .Net interfaces as well.

nmessick - Do you mean for 1st or 3rd party control? 1st party-wise I think Avaya got rid of the TAPI driver for one-X Communicator and have something proprietary instead, but that's CM only. I always thought TAPI made sense for IPO because it's very Windows focused, why would you want...