[RESOLVED]TE410P Channel Negotiation?[RESOLVED]

Hi All,

I was wondering if anyone knew if the TE410P can support Channel Negotiation, and if so how do you enable it on this card?

I have looked through the config files but was unable to find any reference to Channel negotiation.

The problem i am having, is that we an outbound call is placed, the card chooses a Channel that is busy when there are almost 80+ other channels that are not, now instead of searching for a channel that is vacant it just returns Circuit Busy/Congested signal back. This form of Collisions seems to be unusual for a card of this caliber, i would have though that the Digium cards can support Channel Negotiations.

I cant even think of a work around at the moment to prevent this problem from happening, either my mind has gone blank, or there just isnt any and i may have to look to the other cards that work with Asterisk as my source of E-1 cards, i would be greatly disappointed though that the Digium cards are flawed like this.

Cheers,

David.

in other words- when you place a call and have two used channels and a handful of free ones, you want asterisk to choose a free channel and use that one to place the call instead of trying on the used on and failing, yes?

This is done via zaptel groups. In zapata.conf define a group where you are defining your PRI channels.

Then when you dial, dial something like Zap/g1/numbertodial

the result is it will use a free channel from zaptel group 1 and make the call on that.

hope that helps!

[quote=“IronHelix”]

This is done via zaptel groups. In zapata.conf define a group where you are defining your PRI channels.

Then when you dial, dial something like Zap/g1/numbertodial

the result is it will use a free channel from zaptel group 1 and make the call on that.

hope that helps![/quote]

Hello Iron,

Yeah roughly what you said :wink:

However we have been doing groups from day one, but the collisions still happens, it is quite a disturbing thing really as this should be a very basic function of any TDM card.

Today we are going to try something different, if this doesn’t work, i guess the Digium card will be side lined, i have a card arriving to me from China next week, which has an on board DSP card, and its own Configuration system which allows us to switch on Channel negotiation.

Don’t get me wrong Digium cards are good, but they have their limitation, and for what you pay for them i can see them being a viable commercial product, they seem to lack the features that their competing cards have, which has greatly disappointed me, i would have rather used something that was built by Digium to work in Asterisks.

I guess the good news is that the card wont be a total waste, we should be able to use it as n incoming card only, and then have all our outgoing go over another card, not sure if this can be supported between two different brands or not, but if it cant, then i will install into different boxes and do IAX or SIP trunks and use the call forward command, there is always a way i guess.

Cheers,

David.

sounds like there might be a config mismatch somewhere, this really shouldn’t happen.

Try using the opposite order tho… swap g for G or vice versa. I forget which is which but toggling the case changes if it tries to use lower # first or higher # first. This way maybe you will have better luck as a call won’t try to come in on the same channel at the same time?

You could also try a Sangoma card, they are also good…

Hi Iron,

We cant find a config Mismatch, that was one of the things that came to my mind at first, that has been eliminated from the fault finding equation already.

We have tried the following groups: g1, G1, r1, R1 neither of them solved the problem for us.

However i have had long discussions with our carrier about this problem, due to the card not supporting Channel negotiation, they suggested we have all inbound calls start Span-1 Channel-1 and ascend, and for outbound calls we configure it at our end to start at Span-4 Channel-124 and have it descend.

We both felt this would reduce any possibilities of Call Collisions, so far they have reduced significantly but, another problem has surfaced (IT just never ends).

The new problem is, Asterisk is now starting to choose channels that are already in use, instead of going to the next free channel it just drops the call, arrrrgggg so damn frustrating this is.

I am thinking of using the Sagnoma cards as well, since they have confirmed for me that they do in fact support Channel Negotiation (YAY). However there is this card coming out of China that support FULL DSP and support all the features we need, which is less then half the cost of a Digium Card and still cheaper then the Sagnoma card, i will get most likely one of each.

I appreciate your advice and your replies, they help me more then you probably know :wink:

Cheers,

David.

Digium cards DO support this. Actually this is all done within libpri/asterisk. There has to be some kind of conflict with the software and the telco, not the hardware. If you provide a “pri debug span X” (X being the number of the span the call is trying to go out on) I will take a look at it, otherwise you can contact Digium support and they will find out what is going on.You can enable the debug on all spans since u won’t know exactly which span it will use.

Hey Angler,

Thanks for your reply, i will try to get you the PRI Debug, it is a bit difficult as the problem only happens when there are a few calls going through the system, and when that happens trying to copy the info is something of a challenge.

We are beginning testing with the carrier now as to find out if this problem we are getting is a tare down issue at their end, for starters we have begun to do a call retry process where if the call gets rejected our system will try a total of 3 times to get through to the number. On our end we have discovered that the calls end up going through usually on the second or third retry, this indicates to me that the carrier is either not taring down the calls properly or when they interconnect to another carrier that carrier is not taring down the call, i should add as well that the outbound calls are 90% to mobiles, so this issue is new to me.

Cheers,

David.

Have you gotten any more info?

Hey Angler,

Thank you for the follow up, i actually forgot to put Resolved on this one, the problem turned out to be due to us using the .Callfiles setup, the .Callfiles setup was causing Asterisk to choke as soon as we went to 40 call requests per minute and would struggle to get past 2 full E-1’s in fact the only time we could get there was when we reached to 100+ call requests per minute.

We resolved the problem by using AMI via Telnet, it is by far more superior and efficient over call files when dealing will high call volumes.

Oh and i should also mention we swapped out cards from the Digium to a Sangoma 104D, the Sangoma card is by far more robust then the Digium card we had, but thats not to say the Digium card is not a good one, it was just not suited for the amount of outbound calls we had to process, but it is finding a new home now just managing the inbound traffic which it is humming along nicely playing that part.

But in saying that, the Sangoma card is not without its fault, due to its architecture it is also a far more complex card and working with it takes time and patience, but when each problem is resolved the card shows more of its true colours.

Once again thank you for following up, that is much appreciated.

Cheers,

David.