Voip trunks connecting 2 BCM50es

[ojaber]

Good morning,I have described this on another thread but am still no closer to the solution.A client recently changed to a cloud server.This involved them getting all new IP addresses.Previously the client had 2 offices connected for call tranfers over 4 VOIP trunks.While inputting the new IP addresses the harddrive in the local office crashed for no reason(the other office changes were a cinch).After the crash,performed a level 1 reset,recovered the keycodes and rebuilt the system,inputted the new IP address.Now the remote office can intercom the local office,target any particular station,can be heard when answered,yet cannot hear the local party.The local office cannot intercom the far office.I can see the session begin while using BCM monitor logged in to the remote office but after 3 seconds nothing.Is there anybody who might be able to assist.You can call for more details 321-453-0730. Thank you Ben

 

[cook1082]

Patches?

 

[ojaber]

Thanks cook,yes all patches were added after keycode recovery.There most be something very simple overlooked or the crash fouled something. thank you Ben

 

[ironhorse17]

Have you tested to see if it is isolated to one or all trunks?

 

[ojaber]

Ironhorse,briefly no.The VOIP trunks have been pooled.So when an attempt is made to intercom the remote office,info pops up briefly,and if the more key is pressed it shows line 4,again there are 4 VOIP trunks.No attempts have been made to try to intercom from 2 stations at the same moment. Thanks Ben

 

[ironhorse17]

Take them out of the pool one at the time.

 

[ojaber]

Was out on calls yesterday,I will try both suggestions Ironhorse.I can remotely remove the VOIP lines.I have to be on site to try two calls at the same time and will not see the client again until Thurs. Thank you for any and all suggestions. Ben

 

[ironhorse17]

Test with all 4 lines in a pool and remove one at a time, this way you could test them individually.you do not need to make two calls at once.

 

[telcodog]

This is just a shot in the dark but you might want to check the packetization time for the codecs. The BCM default for G.711 is 30 msecs but some network providers require it to be 20 msecs. I've had it happen that even though it was set correctly, after patching it, the system was reset to the default and this customer had the same symptoms you're describing. Only takes a second to check.

 

[ojaber]

Good morning telcodog,not sure where to look for codec timing in system.This is a BCM50e version 1.00.2.04.1.Any help at this point is most appreciated.Considering how smooth everything went in the remote office and now this far office can target any ext. in the local office and be heard indicates to me that after the crash something very simple is being overlooked. Thanks Ben

 

[telcodog]

Look under Resources-Telephony Resources- IP Trunks. Look at the bottom screen (this is for both H.323 and SIP). Look at the codec settings. They are labelled as payload size. I think your systems are release 1 and the defaults back then were 30msecs but the release 6 now has the default (for G.711 anyway) set at 20.

 

[ojaber]

telcodog,under media parameters in the local office there was a mismatch for G.729 payload size and G.711 payload size so I made the 2 systems match(using the remote office as the model since it seems to work).The remote office being a different version the screens are different(tabs at the top).In the remote office enable voice activity detection is not checked in the local office it is does this matter? Thanks Ben

 

[telcodog]

No, it shouldnt matter but make them both the same. Can't hurt.

Yes, the screens will be different depending on the release of software. I think it was release 5 when made the IP trunks a separate entry in the navigation screen. Doesn't matter though, the end result is the same.

Did changing those payload sizes make a difference? Some providers care about, others don't so it's just a shot in the dark.

If it does make a difference, make sure you set the parameters for any ip sets you have as well because if they're different, the system will use the setting for the phone. Got stung on that one myself a while back.

 

[ojaber]

telcodog,no one in the local office to try changes right now(I am off site). Does it make a difference if the two screens on the left are different? Under codec preferences the remote office(the apparently working office)at the top of available list has G.723 with G.711-aLaw under.Under selected list is G711-uLaw with G.729 under.The local office has G729 over G711uLaw in the available list with G723 over G711-uLaw under in the selected list box.Does this matter? Thank you for your answers. Ben

 

[telcodog]

The available list on the left is for exactly what it says. It's a list of codecs that are available to be used by the system. The list on the right is a list of the ones you have selected for the systems to negotiate between them. As long as bandwidth is not an issue, I always put all from the list on the left into the list on the right on both systems and let them negotiate whatever codec they want to use.

It is a problem if your right hand list in both systems don't have at least one codec in common. To move them from one list to the other, just click on an entry and the Add Del arrows light up. You'll see what I mean when you try it.

 

[ojaber]

Thanks telcodog,have to run service calls now. Will have a chance to try these changes on Fri. morning.If anything else comes to mind I appreciate all suggestions.Why can't they all be as easy as a CICS? Computers will be the death of us all. Thank you Ben

 

[ojaber]

telcodog,different day,new results.After changing packet times the remote office can now intercom the local office and can hear us.The remote office can now also transfer calls off the public network to the local office.The local office cannot intercom the remote office.The intercom calls are going out on line 4(VOIP bloc A)but shutdown after about 3 seconds.Obviously no calls off the local CO lines can be transferred.Since we have 2 way communication one way we must be a lot closer.From here it should be something simple overlooked? Any and all suggestions are appreciated. Thank you Ben

 

[telcodog]

Do the calls actually disconnect on their own or is the far end just hanging up because thet think no one is there.

Given that you can actually call the far end, it would indicate that all the routing info is correct so I would start looking at default gateways. It sounds like the voice packets aren't getting through and that is usually caused by a wrong gateway address somewhere, or the audio ports are blocked at a router somewhere.

 

[ojaber]

telcodog,the call disconnects on it's on.I'll check the gateway address.I can see the call come in to the remote BCM by signing in to the system and using BCM monitor.wouldn't that indicate the gateway is correct? Thank you Ben