penelope

I have no experience on the S8300 (yet) and I'm trying to get a new site communicating with an existing G3si v10. The new one is an S8300, actually says G3 V13. I've created the trunk group on the G3si side exactly how I've done it for another site to communicate with a Prologix (that one works VoIP).

I've done the signalling group, node-names, everything that I am aware of. A trace says:
10:04:57 denial event 1012: Destination Unavailable D1=0x7a D2=0xcc
10:04:57 dial 8025
10:04:57 reorder trunk-group 37 cid 0xcc

The status on the S8300 says: 0037/001 T00031 out-of-service-NE
The status on the G3si says: 37/01 T00276 OOS/FE-idle

I've compared it to another site with an S8300 that is communicating with the G3si and I don't know what I'm missing. I've searched all over Avaya and can't find a manual on the S8300. Does anyone have an idea of what's missing or where I can get a manual?

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Penelope

stubborne

I haven't done any 8300 configs either but I do have a couple of questions. I'm assuming this is a standalone server (G350 possibly) and not an LSP? Can you ping the far-end node-name from each device within Communication Manager?

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Stu

PBXtech.info

 

penelope

Yes, it's a G350. From the G3si I did a ping node-name and I get:

PING RESULTS
MILAN 01B0117 ETH-PT PASS 196

I did this from both the CLAN and the MedPRO and it was ok on both. On the Milan system I tried the same and can also ping the node-name of the G3si. The problem is on the Milan side according to both systems... out of service on the trunks. I don't know how to correct that on the Milan side or what I am missing in config.

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Penelope

penelope

A little progress... I had set the network region at 2 but found that milan side was at 1. I changed it to 1 and the trunks on both systems came up as in-service/idle. Tried calling again and on the Milan side I get:

00:00:44 Calling party trunk-group 37 member 1 cid 0x146
00:00:44 Calling Number & Name NO-CPNumber Wxxx, Penelope
00:00:44 active trunk-group 37 member 1 cid 0x146
00:00:44 dial 8025
00:00:44 ring station 8025 cid 0x146
00:00:44 denial event 2309: Drop call codec mismatch D1=0x7f40001f D2=0x146
00:00:44 idle trunk-group 37 member 1 cid 0x146

I don't know a lot about the codec so looks like I will have to do some more reading.

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Penelope

martinyoung

You won't find a manual for the S8300 or any other S8xxx server. What you want is the library CD (which is downloadable) for Communication Manager. That library contains all the switches. The best CD for you is CM2.2 because it includes all switches, including Definity. That link follows.

http://support.avaya.com/japple/css/...&PAGE=Document

Download the CD and install it on your PC.

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Marty
Retired Avaya DSIC tech

martinyoung

The usual problem with codecs is that they need to match at each end. If one codec calls for G711 and the other codec calls for G729, you will never connect. I usually set mine for G711 as first choice and G729 as second choice.

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Marty
Retired Avaya DSIC tech

PBXtech.info

 

penelope

A little more looking around and I think I might have it?. On both trunk groups it says codeset 6, codeset 6 has just one entry of G.711MU.

I then looked at the Network Region 1 on both systems and there I found the difference. For ip-network-region 1, Milan has codeset 1, codec G.711A. On the other system it also has codeset 6 on the trunks with G.711MU. But, on the Network Region 1 it has G.711MU.

I haven't changed anything yet since I do not know if there is anything else that would be affected. If I change the codec on code-set 1 from G.711A to G.711MU, what else might that effect? There are no other VoIP trunks, I'm trying to get it working for the first time on that system. Would it affect anything like communicating with the IALX in Milan?

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Penelope

Arslanalizaidee

HI,

The codec set for both sides should be same....so kindly check for this properly in network regions......G.711 is generally used when communication is done on LAN where as G.729 is used on WAN links.Also check for the IP Hairpining and shuffling parameters in the administered network regions.

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Thx n regards,
Arslan Ali
Telecom engineer

stubborne

"If I change the codec on code-set 1 from G.711A to G.711MU, what else might that effect? There are no other VoIP trunks, I'm trying to get it working for the first time on that system"


Excellent, you are on the right track for sure. Set the codec on the 8300 to match your G3Si...

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Stu

oozenoz

The A and MU at the end of the G.xxx refer to the companding being used on the voice signal - A for A-law, MU for mu-law. (If you've ever accidentally hooked up a digital span between two devices where one side was A-law and the other mu-law and actually placed a call, you would understand why a mistmatch is torn down when attempting to make the call.) It matters not which companding you use - it just has to match.

Make it easy on yourself and define a new network region which uses either A-law or mu-law companding, and use this new region for your trunks.

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- calvin

penelope

I did create a new network region and matched the codec this morning. It works! I had a meeting with an Avaya engineer later and asked him take a look at all that I had set-up and he said it was all good.

After that, I did Hong Kong. Took about 15 minutes to get everything configured on both systems and the first test call went through. Only issue I have is that the extension numbers in HK start with a 0 (don't ask me why my co-worker who went there did that, the extensions match the DIDs). So, I set it up so that my side actually dials 1025 instead of 0025, for example. In the route pattern I have it doing a delete 4 (3-digit AAR code + the 1) and insert the 0 to send 0025. I don't really like that because I just know that users are going to be confused if they dial 1025 but if they address and send a voicemail it would be 0025.

Any other way to get around dialing an extension that starts with 0?

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Penelope

oozenoz

Ouch - overlaps in dialing plans are a real bugger, especially when your DIDs overlap on networked systems (I usuall suggest 5-digit dialing plans globally for those sort of occasions as it gives you more flexibility)

In your type of case, though, I prefer simply expanding the number of dialed digits. Users are pretty willing to understand and accept dialing a trunk access code - one or two digits, then the extension. Or you could force them to dial the entire international number, but you set up your ARS to remove any digits prior to the last 4, and route the call over your VoIP trunks. (ARS would also allow you to trunk the call via PSTN in the event your VoIP trunks ever went OOS).

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- calvin

penelope

Yeah... Too bad when for 5 years since new systems were installed in EVERY office we have been careful to never overlap DID ranges/extensions now, one of the second international site to be brought up with 4-digit dialing throws a wrench into the whole thing.

I'm thinking of rather than make ALL off the offices 5-digit dialing to accomodate that one, I'll begin making the international offices 5-digit. Since Milan and HK are the first to be added users can start out getting used to dialing something in front of the 4-digit extension number for international. Like a 1. Irks me that HK was done like that and between you, me and the forum, the person who did it says his reason was that he did not know all of the other offices DID/ext ranges. He could have asked. He could have looked at the other systems. Any number of ways. Soooooo... he chooses 0? The one certain to conflict with EVERYONE.

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Penelope

oozenoz

You're definitely on the right set of tracks, just be cautious when making a final decision about your dialing plan and scope... there are some pitfals and things you need to sort out before you begin programming - what you are describing is very easy when there is only one trunk path, you will potentially have several for any single site.

Assume that you will have one VoIP trunk path from site A to B, another from site A to C, and one between B and C, and that you continue to have no overlap of extension ranges (adding that would only complicate this discussion unnecessarily).

Let's say you go forward and make all international calling via the format yxxxx where y defines your call as an international, and xxxx is the foreign site extension. Your user at site A wants to call someone at B, how will the system determine which set of IP trunks to go out so that it does not route the call to site C? (ARS can probably do this for you if you use a 2-digit code, where yx are unique to a given site... again this only works if you have no overlaps on extension ranges).

The alternative (which is something I would strongly urge you consider) would be to create a 2-digit trunk access code for each foreign site. So from Milan, let's say you dial 11 to get to Hong Kong, 12 to get to another site, 13 to another, etc. If you think you are ever going to need more than 10 sets of VoIP trunks at any one site, make the access code 3-digit.

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- calvin