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Networking 3 BCM 50's 1

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rval80

Technical User
Sep 23, 2009
386
US
I am networking 3 BCM 50's and trying to use 4 digit dialing between the 3. Site 1 is using 2200 series, site 2 is using 2300, site 3 is using 2400. I have everything programmed to what I believe is correct. Destination digits of 22, 23, 24; routing set up, ip addresses configured. However, I am getting fast busy when I dial out using 4 digit dialing. I've opened the monitor, and all digits are being sent. Any ideas?
 
I have SIP. It turns out I had the gateway wrong.
 
New issue - may be network related. I have one of sites working from site a to site b. When trying to dial site c from a, it does not work. I can ping from BCM to BCM. Could a firewall be blocking something?
 
Update: I have the BCM monitor opened at site A. When dialing from site B, I can see the call going through, however it disconnects after 2 or 3 rings. No voicemail or anything. Any insight would be great.
 
Yes, the private received digits are set.
 
Make sure SIP URI is the same in all 3 BCM's.

It's only dialtone-VZ
 
Few tips:

1) take care that an incoming call is not being routed out on the BCM you are calling.

2) If you have digital sets on the BCMs, try dialling from digital set on one BCM to a digital set on the intended BCM....thay way you are testing the signalling and media path BCM to BCM, and not to/from the IP sets.

3) Look at the feps log (this will give details from the perspective of the voip trunk, where it is SIP or H323)...
on both the BCM you are calling from and to,

a. Take a snap shot of the feps log before the call.
where xx.xx.xx.xx.xx is the IP address of your BCM/SRG.
Enter the nnadmin account and corresponding password as required.

b. Attempt the failed call.

c. Repeat step 1.

Does the feps log on the calling and called BCMs give an indication? Browsing the log file will not be available for BCM 4.0 or possibly earlier versions of 50, so you might need to make some test calls and collect the full set of logs, then extract them and look for feps.log.

regards
Paul
 
Now, I can actually dial from point A to point B, however, now when the person at site B answers, it automatically disconnects!? This is getting frustrating.
 
If you have SIP trunks between your sites, make sure that your codecs are all set the same and that the payload sizes match.

If you're using IP sets to call to/from, make sure the payload sizes match that of you're trunks. I had that problem once and it took me 2 days to figure out what was causing that exact problem. Call would go through but disconnect as soon as the other end answered.
 
Telcodog, you are the best! It was the codecs. Thank you for your help.
 
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