Problem with Network regions, LAN, WAN? No more ideas

[nurofen]

Hello there,
I have the following scenario.

A S8300 working as media server in a G700 in location A.
A S8300 working as LSP in G700 in location B.
Trunks are in location A.
Two network regions 1 and 2. Codec set 1 (G.711) for intra NR and codec set 2 (G.729) inter NR calls.
Incoming calls for A or B works fine.
Outgoing calls from A or B, works fine.
Calls from A to A or B to B works fine.
Calls from A to B works fine
Calls from B to A one way voice path.
LAN & WAN engineers claim the network is fine.
UPD range open, TCP and UPD port 1719 and 1720 also open, no firewall between the two locations...
When a call is traced and the VoIP resource is from location A, everything is fine, when the VoIP resource is allocated on site B, then the one way communication appears.
If I busy out all DSP resources on site B, no one way path problem appears.
As soon as the DSPs are allocated from B, problems appears.
I review most of the threads on the forum, but couldn't find any similar, we are not using medpros or clan, all processing is on the S8300B. We have around 10 phones on location A and 30 phones in location B, so doesn't seems to be a problem with resources, no fail ip-networks; no significant alarms or errors, nothing that I can see and start to pull the thread.
If I trace both phones, with no DSP resources busy, then I can see that ports allocated in both phones are ok.
To avoid the problem for the time being, as previously stated, all DSP resources on location B are busy at the moment, the IP-IP connection is set to no, to avoid problems.
As on the subject, I have no more ideas to follow. Any suggestiong will be greatly appreciated.
Thanks

 

[big70]

Normaly an call from B to A would not use a dsp if ip-ip is set to yes

I would check al ip-settings and default gateway in CMM and G700.

Also put your pc in the G700 and check if you can ping all ip adresses.

Greets Peter

 

[nurofen]

Thanks Peter for your answer.
I can ping every location, every phone, LSP, Gateway....you name it.
IP settings were chechk at least five times and seems to be correct (seems because now I am not sure of anything)
I will return the IP-IP to yes; but the original setting was IP-IP yes in both NR and station. By the way, I am runing ACM 3.1.2 with latest service pack from Avaya.
Thanks

 

[wildcard100001]

Just a thought... update the integrated VOIP module firmware version, both sides.

Thanks,

Wildcard

 

[mrjedi]

what about show ip route voip v0 on B and make sure there is a route from B to A. Best to put: set ip route 0.0.0.0 0.0.0.0 xxx.xxx.xxx.xxx ( your gateway )

 

[big70]

It must be a routing problem, like you said al trunks are in location A.
And you can be reached an call out with phone,s in A and B.....

So routing to the G700 at location A is fine...

But if you say ip-direct on yes the it tries to go direct to talk to a phone from A to a phone from B and this not working correct..

If you put you laptop in the switch where the phone,s are located in A and B and put this port in same V-lan as the phone,s(important).

Look at what IP-Adress is given to your laptop by DHCP and Gateway addres and see if you can ping the phone,s at side A and B do this at bouth side,s......


Greets Peter

 

[nurofen]

Hello again,

Thanks to all of you that replay to my question.
I try to answer all of you.
wildcard100001; the VoIP cards and internals are to the latest firmware version on both sites.
mrjedi; the routes are as you mention, in both machines pointing to the default gateway.
big70; both routing for A and B are exactly the same, I checked several times.Yes, is correct all trunks are in A and there are no problems with incoming calls or outgoing calls. If I say IP to IP depends on who iniciate the call, if is location B, then a DSP resource from B is in charge of communications, so the call will fail; that is the reason why all DSP are busy out in location B and IP to IP set to no, to at least cause the minimun inconvenience to the customer.
I already place a computer in both ends, on the VLANs for voice and take some traces with the sniffer, but no relevant information was found.
More information about routes; these are the routes on B:
show ip route
0.0.0.0 Default gateway IP@
show ip route mgp
0.0.0.0 0.0.0.0 Default Gateway IP@
network 255.255.255.0 media gateway IP@
show ip route static
0.0.0.0 0.0.0.0 default gateway IP@
show ip route voip
0.0.0.0 0.0.0.0 default gateway IP@
network 255.255.255.0 DSP IP@
The IP routing was manually setup, as my first impression was that something was going on with the routing.
I was even thinking that the config we setup wasn't allow or used, but since the desing was "Avaya made"...
More information, I test all DSPs from location B one by one, busy out all except one and then making calls, also DSP resources were exchanged with location A and as result we now know that DSP are OK
LSP is correctly registered, MG also registered, translation save correctly...
LSP is in NR2 as the MG and phones for location B.
I don't really know were to look more.
Now a question (another)thinking that the system works fine. When you pick the phone in location B, a VoIP resource from B is assigned to provide the phone with dial-tone; so, two ports on the range of UDP ports for NR2 are assigned, one to the phone and another for the DSP.
The phone rings in A, then a port is assigned to the telephone and DSP Port +1 is assigned on the DSP resource. That's is how I was teached the system works. It is correct?
OK, enough. I thank you again and waiting for more answers.

Thanks

 

[mrjedi]

quick question. How are you testing the DSP resource and how are you swapping the DSP resource for Loc a and b ?

 

[nurofen]

Mrjedi,
the way we test the DSP resources is getting them in busy state and leave one in release, then make calls and look how many resources are in use. Also we physically swap the DSPs from one media gateway to another (just the cards, not the V0). Also we replace the MGs. I know that maybe doesn't sound quite a scientific test, but that is the best we can do.

Thanks

 

[mrjedi]

can you run : show voip v0 and see the state of the card

 

[nurofen]

MrJedi,
I run that command serveral times (at least twice everytime we busy and release the cards) and shows no faults/released or no faults/busy.
As I mention before, we also swap the cards from location A to B and the result is the same. We have four VoIP card, not only the V0 on-board DSP and usually used these cards for testing but also the built-in DSPs; I find dificult to have all of them with problems.

Thanks

 

[mrjedi]

what about removing all the DSP and just using the V0 as a test. Sorry, I am sure you must have done this but just trying to see any anlge you have missed out