ISDN Connection Problem

[Airwave1]

Hi,

I am having a particular problem with one of our Sony Contact 1600's that is served by ISDN6 off our mitel phone system.

The problem is as follows:

When you make a ISDN6 bonded call to the unit from another unit that is on the same phone system (ie an internal call) there is no problem at all and the call bonds with all 6 ISDN channels. However when you call the same unit from the same other unit but dial in the full dial code (ie an external call) the call does not bond. I am dialing from a Tandberg 880 and when you watch the call status the call connects on the first ISDN channel then the status says H.221 but then fails to connect.

Anyone have any ideas? :(
It seems like a problem with the phone system to me but tis proving it! Our Mitel phone engineer has made a trace on the call and only see's the first ISDN channel being rung and conected and not the others so say's its not his equipment!

Thanks in advance everyone! :)

Gareth

[mazzarak]

how do the extensions for internal and external match up? because if the numbers are slightly different, and that difference isn't in the numbers list, then it might cause a problem.

[robertk]

I would also investigate if the PBX route the call as a AUDIO(analog) call or as DATA call.

It's typical if you only are connected on one channel and the units never negotiate any DATA protocols, such as G711, H261 (and ofcourse the BONDING protocol). Then you can suspect that you only have a ANALOG connection between the sites. IE it rings and the system answers, but the data can't be transmitted between the systems.

//Robert

[Airwave1]

Thanks guys,

The numbers are definetly not the issue as they have been the same for a few years and this problem has only apeared in the past 3 weeks.

I think its definetly to do with the protocol's not getting through or not being recognised/supported.

Have any of you had experience making ISDN Traces using Tandberg units (880 in this case). I have read over how to make the ISDN trace and understand that but don't know how to translate the info?
Once again any ideas?

Thanks

Gareth

[pbraatelien]

As an added note (and this coming from too many years of troubleshooting). What has changed? If it was working previously and just (3 weeks ago) started messing up, some kind of programming, software release, patch, etc may be the culprit. I would try that out as a starting point.
Paul

[Airwave1]

Unfortunately there have been many changes in the past 3 weeks and this is just one of a few problems we have had to tackle.
The problem is proving what is causing the issue.

I will let you know how I get on.

Thanks again.

Gareth

[dndtech]

Airwave,
Since you are successful between devices served by the same
PBX, your local PBX - VTC connections and protocol settings are good. Don't touch them.

You mention that there were several changes made lately. pbraatelien is bang on with his recommendation of looking for what's changed. You can narrow this further by looking for what has changed on the carrier side of your PBX.

Has the trunk route been changed by your service provider?
Have the PBX trunking parameters been changed?
Did a PBX SW upgrade affect your carrier settings?
Can you make a successful call to a VTC device outside your PBX.
Can an outside VTC device successfully call your internal VTC?

Try these and get back with the results.

I have several loopbcak numbers that I use. These allow me to ISDN dial out across the public network and then the call returns to my device. It's like having a VTC with yourself. It proves my device as well as my carrier connections. I'm in the Colony but you can try this across the pond - 613-941-1112. Others in NA feel free to use this number for yourselves. I think it works up to 512Kbs.

[mazzarak]

Hi Gareth

are you on Cable and Wireless? I'd make them pull their weight if you are. Some of their other clients have had that "For your convenience and security, we have cut off your external ISDN capability" and all that shenanigans.

I suppose the alternative is to get someone completely off your network to try it, to see if anyone can reach it externally.

[Airwave1]

Yes I am on Cable & Wireless and it cause me no end of grief!

Last month I had an issue where by we were getting no far end camera control when we connected 2 units accross the C&W VPN. We could however get far end camera control if we routed the calls off the VPN and accross some external British Telecom lines we have in place for resilience.
C&W made various traces and tests and even installed some "high end" monitoring equipment on site with us to check things further. This showed up nothing and when I mention the protocol's of video conferencing they were reveived with blank expressions!!
They have since reset various peices of equipment and now the problem seems to have disapeared, but I expect it to come back at anytime as we never got to the bottom of what caused it.

Anyway back to my current problem!

When I make an incoming call to this unit from any external device, whether over out VPN or from those BT lines I mentioned we still can't bond the call and it fails.
So this proves its not the C&W VPN in this case and that is more than likely to do with the PBX.

I will enquire about the carrier settings tomorrow as thats me away for the day now!

Thanks again everyone :)

[dndtech]

Airwave,
You said you can make a call internally. This proves the PBX ISDN configuration is good. By getting the same results with your unsuccessful external calls using different carriers, your problem could be in the PBX but on the carrier side. Focus on any changes that were made to your carrier side (both software and hardware). I suggest software as your physical connection to separate carriers are more than likely on different cards within your PBX. Confirm this as well.

Good luck and keep us posted on your findings......

[mazzarak]

this might appear a stupid question, but have you tried 128 and 256k calls in the same way? For bonded calls if one channel falls over, often all of them will, so reducing the bandwidth might lead you to narrow it down to one faulty line rather than 3.

[Entropy3XD]

Originally posted by mazzarak@May 13 2004, 06:16 AM
this might appear a stupid question, but have you tried 128 and 256k calls in the same way? For bonded calls if one channel falls over, often all of them will, so reducing the bandwidth might lead you to narrow it down to one faulty line rather than 3.
I agree with Mazzarak. I had a similar issue at a client's site the other day. Calls would not bond on outbound. I tested each channel individually by having someone from the outside dial each number at 64k. Everything checked out there.

I then took each line individually and configured them on port one, then tested them at 128k one at a time. Lines 2 & 3 bonded, but line 1 failed. I moved line 1 to port 3 so they are now able to bond up to 5 channels until telco arrives. It is an old VTel ESA, which always wants to force calls out of port 1 first.

[dndtech]

Yikes!! Should have realized this from the top.

Since ISDN is derived from traditional voice technology, each ISDN line has a telephone number assigned. You can simply use a telephone to call each individual ISDN line to test that is servicable. This will at least prove your PBX configuration is correct. Once proven, you can focus on the bonding issue, which is the performed by the codec. Use the procedure for reduced bandwidth calls as outlined by mazzarack and Entropy. (64K, 128K, 256K etc.)

I'm not sure if this is pertinent in Europe, but you may also try a restricted call (forcing your lines to use 56K vice 64K). We have several locations that are not yet served by ISDN and we must use 56K restricted calls to connect in either direction. There is nil D Channel used in a 56K restricted call and all signalling is performed within the 56K channel. A successful 56K call will point to a D Channel problem.

Keep us informed on your findings........

[dndtech]

Airwave? What's happening over there?

[Airwave1]

Hi,

Sorry guys been swamped with other work at moment and have not tested this yet!
Im also away for most of next week as well.

Its a typical example of my work load here as because this service is used very rarely it has to be prioritsed and therefore takes a back seat at the minute.
Im not forgetting about it though!

Thanks

Gareth

[Airwave1]

Hi Everyone,

I have had a bit of time today and have made soem of the tests you have suggested, so here goes:

First test I made was calling the units 3 lines seperately using a standard audio telephone, I made these calls as an external call as this is where the problem lies.
All 3 calls connected correctly.

Second test was to dial the unit from another video conference unit (Tandberg 880) externally trying the different bandwidths (64K,128K,256K and 384K). All call failed to establish, even the 64K.
This suggests its not a bonding issue.

Thirdly I called from the 880 this time limiting the call to 56K, I tried this at all bandwidths (64K,128K,256K and 384K) but once again had no joy as the calls failed to connect.

I am now away to connect another peice of Sony equipment in place of the Tandberg 880 to see if this makes any difference.

I will let you all know how it goes.

Thanks again,

Gareth

[robertk]

Again!

You should REALLY talk to your network provider and ask them if there are any issues using the DATA service on your ISDN network.

The problem you have is typical when for example a service provider choose to limit a customer to ONLY use the ANALOG service on a network.

You can dial in using a normal phone, but as soon you try to send data it fails. It's a text book example :)

//Robert

[Airwave1]

Just to add this problem is only when I call into the system and I can call out from it (externally).
If the lines were set to analogue only would this not effect it in both directions?

Thanks

[mazzarak]

Gareth

when you say you're calling at 128k and 256k and 384k, have you been trying 2x64, 4x64 and 6x64?

If you like I can try calling you from our bridge - see what happens...

[Airwave1]

I have tried 64K, 2x64K, 128K,256K & 384K, as well as limiting the ISDN use to 56K.
With the Tandberg 880 you can not, as far as I know do an unbonded call above 2 x 64K.
I can not get through on even a 64K so it suggests to me its not a bonding issue, more of a data issue.

If you want to try ring me that would be helpful though.

You can ring me 01224 559269 if you want.

Thanks

[George]

Have you talked to your local switch techs? Someone that has visibility on your ISDN circuits outside of your room?

[robertk]

Originally posted by Airwave1@Jun 1 2004, 11:24 AM
Just to add this problem is only when I call into the system and I can call out from it (externally).
If the lines were set to analogue only would this not effect it in both directions?

Thanks

Not neccecary... it depends on how they are configured.

//Robert

[vtjoe]

Originally posted by Airwave1@May 10 2004, 09:05 AM
Have any of you had experience making ISDN Traces using Tandberg units (880 in this case). I have read over how to make the ISDN trace and understand that but don't know how to translate the info?
Once again any ideas?

I have some experience in trying to read the traces. However, there is no documentation in english (maybe in norwegian) by tandberg on how to read the traces. If you understand BONDING and ISDN it is my understanding you may be able to decipher the traces. See if your PBX guy can help.

Otherwise, your best bet is to work with Tandberg tech support. When I had issues in the past, they sent the traces to the engineers in Norway.

[Airwave1]

I have made an ISDN trace fairly recently and sent it to Tandberg for them to analyse, in the end they needed far more other info than I sent and by then it was too late as the problem had fixed itself!

I have my Tandberg rep visiting me next week so Im hoping to have a chat about Trace testing and H.221 Dumps and how they can help me out. I also need to talk to him about the TMS software as although we have it installed we are not using it in anger yet!

Anyone else use TMS??

[mazzarak]

We're currently using a demo version. Looks quite funky, but you can't do much if you can only put 3 systems on the demo...

I set up another discussion about GMS v TMS somewhere in the forum... cant remember where now... :unsure:

[vtjoe]

Originally posted by Airwave1@Jun 2 2004, 03:13 AM
Anyone else use TMS??
We use TMS all the time and are very happy with it. What kind of problems are you having?

[Airwave1]

With TMS its not so much problems more limitations.

At the moment all our bookings are made the old fashioned way using a paper diary as it has been used like this for a number of years, this works well and we have little problems with bookings.
I am looking at moving it all over to TMS but the migration seems to be all or nothing as you have to book all the conferences corectly as each of the 3 dial in lines go into "Idle" mode when a booked conference ends and don't become active until the next booked one commences. This causes a problem as at the moment all conferences are dial in, not out, so if the bridge is in "idle" mode then no one can dial in to it. This would cause a big problem.
The next issue is we have conferences booked right up to the end of the year, and some beyond!, and all have been issued with the correct dial in number. I try to give certain groups the same number as this means they get used to it and it becomes routine. With TMS it automatcially issues the first free number (the first one) and you can not specify what number to give to a particular conference. If you could specify what number to give a conference at the time off booking this would be far better for me and cause a lot les hassle in the long run.

Another thing I'd like to be able to change is when making a booking with any system you should be able to specify the call route, at the moment you can only do this if the site is IP enabled (ie you TMS can control it). The way around this is to input this site as a dial in Video but it would be easier to be able to use a system already in TMS and just allow it to dial in on its own, this way your system monitoring is kept up to date.

You guys spotted any changes you'd like to see?
I'll be mentioning these to the Tandberg Rep next week so hopefully they can make these changes in time for the next software release, thy have done this before and are always listenning to customer comments/suggestions.

Thanks

Gareth

[vtjoe]

At the moment all our bookings are made the old fashioned way using a paper diary as it has been used like this for a number of years, this works well and we have little problems with bookings.


If you are passing on suggestions, how about some extra customized fields in TMS. For example, someone wants to register for a conference but it needs to be approved first - their name, phone number, e-mail, outside 3rd party information. I talked to someone who hacked into the database to make something like this work. Not a bad idea.

I am looking at moving it all over to TMS but the migration seems to be all or nothing as you have to book all the conferences corectly as each of the 3 dial in lines go into "Idle" mode when a booked conference ends and don't become active until the next booked one commences. This causes a problem as at the moment all conferences are dial in, not out, so if the bridge is in "idle" mode then no one can dial in to it. This would cause a big problem.


I'm confused why this is an issue - just schedule the conference to start earlier. Or are you saying you want ad-hoc ability when other conferences are not scheduled? If so I agree.



With TMS it automatcially issues the first free number (the first one) and you can not specify what number to give to a particular conference. If you could specify what number to give a conference at the time off booking this would be far better for me and cause a lot les hassle in the long run.


I agree with you 100% on this one.

You guys spotted any changes you'd like to see?


Yes, I agree with all of your changes. The integration between TMS and the MCU is not as seemless as I would like to see. Resources are allocated by TMS, but TMS does not control 100% all the features of the MCU. For example, during important conferences when we need to have someone manually switching between Continuous Presence and voice switching, I or one of my techs go direclty to the MCU web page. However, TMS should be able to handle this functionality.

Also, if a conference is schedule in the MCU, then the MCU should know this and not allow you to make TMS resource related changes - except through TMS.


I'll be mentioning these to the Tandberg Rep next week so hopefully they can make these changes in time for the next software release, thy have done this before and are always listenning to customer comments/suggestions.



Pass these MCU changes along...
Ability to have an admin site connect and their video and audio is not shown (not even a black square for Continous Presence), until their video/audio is turned on/off in the MCU. This way I could monitor a conference completely without interupting it, until needed.

DuoVideo On/Off should be schedule in TMS through the MCU. This can mess you up in a mixed environment. If a legacy system connects to a conference, and a Tandberg codec send DuoVideo, the legacy system only recieves the main image (people) and not the second image (content). Better yet, ask them to have the MCU send the content to the legacy systems.

[robertk]

Originally posted by vtjoe@Jun 3 2004, 03:57 PM
Better yet, ask them to have the MCU send the content to the legacy systems.


This really isn't a TMS feature, but a MCU feature... and I agree with you to 110%. This need to be fixed by Tandberg.

If you are on a Legacy system connect to a Tandberg MCU and someone is showing content/dup video then all you see is the MAIN video source.

//Robert

[Airwave1]

[QUOTE
I am looking at moving it all over to TMS but the migration seems to be all or nothing as you have to book all the conferences corectly as each of the 3 dial in lines go into "Idle" mode when a booked conference ends and don't become active until the next booked one commences. This causes a problem as at the moment all conferences are dial in, not out, so if the bridge is in "idle" mode then no one can dial in to it. This would cause a big problem.




I'm confused why this is an issue - just schedule the conference to start earlier. Or are you saying you want ad-hoc ability when other conferences are not scheduled? If so I agree.
]

This is really something that would help me out as I almost always know if there are conferences on at anytime so if I know there is not one on on 1 number and I am off site I can use it to test call with. Yes I could make a sperate booking for this but I can not always do this as I am often off site.
This not a huge problem but it is something I would like to change, I don't think they will though.

As for all your other suugestions I agree with them all, although we do still use Duo Video occasionally with a mixture of systems. An example is if someone is showing a Powerpoint slide as the MAIN image and the main camera as the duo video then any sysytem that does not support Duo Video will just see the Powerpoint, that works ok as long as the person presenting is aware that some sites will only the Powerpoint.

I'd definetly like to see more fields for text as at the moment we have to put all our info in the one text space for Booking Name.

I will print this all off and hand it over to the rp next week, I'll let you all know how I get on!

Thanks

Gareth

[Airwave1]

How do I use the quote feature??

Cheers

[George]

Originally posted by Airwave1@Jun 4 2004, 02:46 AM
How do I use the quote feature??

Cheers
In the same row as the user's name that posted the message, look to the right and you'll see a Quote button.

It'll place the quoted information in a new box under your reply box on the next page.

You can also simply copy what you want to quote and paste it inside quote tags like

[*QUOTE]words to quote here[*/QUOTE]

Without the *

[Airwave1]

Just to let you all know I met up with th Tandberg Rep yesterday and made a whole host of suggestions to him. He has taken a note of these and will pass them on to the techy guy's and see what happens.

I have arranged a video conference meeting with him for a few weeks time when hopefully he will be able to tell me what the techy guys had to say about my suggestions.

In terms of new developments he spoke about a new Duo Video style that allowed a standard screen monitor (in single screen mode)to display 2 video sources at equal size, at the moment you can do this but its using the picture in picture style. He also said that they are working on widescreen (16x9 aspect ratio) support which would tie in with this new duo video style.

Thanks

Gareth

[sdevoy]

Gareth,
Send me the traces: bonding;mitel;tandberg;sony
Bonding will show the local directory numbers sent or received.

I know that SONY are one of the few codecs that still use H.221 for 4 to 6B, if a bonded call fails. Tandberg, do not support anything but 2B.

I used to work for Madge video with lost of Mitel exp. where are you in the UK?

Cheers

Steve@Devoy.net :)

[Airwave1]

Sdevoy,

Thanks very much for the offer, I will get another set of traces made for you today and get them emailed to you.

Thanks

Gareth

[Airwave1]

I work for the NHS in Aberdeen.

[ashok1357]

hi,
check up with your exchange.see if they have configured all the lines for voice ,data and video.I think the problem lies with your local exchange on routing

[sdevoy]

Ashok,
Sounds like it.
Gareth beware of what Tandberg offer, remember they are not Mitel and it doesn'* sound like a VTC problem.

Steve

[Airwave1]

Due to holidays etc. it had been a number of weeks before I could sit down with my Telecoms engineer and resolve this issue but it is now repaired.

As many of you had thought it was down to the routing within the PABX, or rather the lack of it!

Once again thanks to everyone for the help.

Gareth