IP Office 406 with 3 VPN Sites - Voip Issues 1

[KevinBugg]

We have 3 sites which are linked via vpn tunnel. IP Office was configured by Avaya Specialists but we have major issues. We can not call between sites successfully via voip. The calls are answered but no two way conversation, or the line is dropped. What can we do to try and test for ourselves. Pointers to test basic settings? Our 2 UK offices are joined by a small community network but even those calls are bad. Our other office has its own dedicated voicemail server hence the reason it isn't part of this network. We have leased lines so the connection between the sites should be ok. We have been having these issues for months and the specialists pointed me here!!

 

[KevinBugg]

We probably have a maximum of 30 extensions at each office (3 offices) and calls between offices are scarce but essential when needed. Is there an easy way to check to see how many vcm's we have at each office and if we get close to using them all? I wasn't trying to be cheeky about copying and pasting, its just that i have looked at the help and got no where (cos im not clued up on this). I appreciate all your help!!

 

[kurthansen]

Take a look at System Status.

 

[IPOfficeIreland]

you can monitor vcm usage easily in both ssa and monitor.

ssa will give you live stats as well as congestion occurences.

monitor will give you logged files so you can go through the outputs. enable development tracing in System, then enable the new tab which appears, vcomp or something like that.

You can now also watch live stats of the vcm usage, codecs used, rtp streams, ip addresses from Monitor with DevTrace enabled. very usefull in fault finding.

 

[Thermus]

Hi Kevin

Had a similar problem between 2 IPOs here. Turned out the firewall had a setting called "stateful inspection of H.323" and it was on. We had it turned off and have not had a problem since.

My issues were that the first call between sites would connect with no problem, but subsequent calls for the next couple of minutes would not establish a speech connection between sites but would still ring at the called party end. Then after say 3 or 4 minutes a call would connect and then following calls would fail again.

 

[kurthansen]

Sounds like different wording for the Fixup h323 that Cisco uses. Also on the new ASA series they have policy maps that inspect the h323 protocol.

 

[KevinBugg]

Does direct media path only affect ip phones and not digital phones / analogue phones? We have a combination of all 3 over the 3 sites. Just trying to get my head around what happens if both telephones are digital or analogue, does this direct media setting come into play?