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Can only receive 7 calls at a time on SIP

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Bivman

Technical User
Jan 13, 2008
60
US
I have an IP Office 500. I think the software is 4.0 (15)
I have 124 VCM channels. When I get a large call volume, usually Thursday nights, I can only register 7 calls and then I start getting congestion reports saying not enough VCM channels available. Any ideas?

Thanks,
-Jon
 
Do you have enought bandwitdh ?
Are your licenses valid for the additional vcm channels ?


RTFM.gif



ACS - Implement IP Office
ACA - Implement IP Telephony -- ACA - Design IP Telephony
ACA - Voice Services Management
______________
Women and cats can do as they please and men and dogs should relax and get used to the idea!
 
Using Bandwidth.com who wasn't originally friendly w/Avaya. They installed an IAD unit to translate from SIP to PRI. However, their IAD seems to cause trouble with call quality.
We now want to move to SIP direct. We have a DS3 which should provide ample bandwidth, plus the IAD allows plenty of incoming calls (92 on 4 PRIs). We have to sets of 60 VCM channels with valid licenses. We have 3 sets of 20 SIP channels, also with valid licenses. Not sure what else could be going on. As far as I know, I only need one extension with the 800 number in the SIP field to allow the number to register.
Thanks,
-Jon
 
So are you saying you are using direct sip now or pri?

are the calls clearing down properly? To view active rtp sessions and Voice Compression T1 sessions, open monitor, enable development tracing, then on the menu bar, select Status and choose the above 2 options.

If also direct, is the 8th call hitting the pbx, as in, are you receiving invite messages. capture a trace of when this happens (SIP only).

 
Have you ever used a Quintum box before?

30+ SIP channels --> PRI

costs less than 30 SIP licences and reasonably easy to configure.
 
4.0(15) is not the version of the IPO but of the VMPro, 4.0 is old and has a lot of bugs. Better start upgrading to at least 4.2.x

Avaya_Red.gif

___________________________________________
It works! Now if only I could remember what I did...
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OK... The version info looks like 6.0(14) for the manager software, 4.0(14) for the IPO and 3.0 for VM Pro. I don't know anything about the Quintum box. I kinda want to get away from an interface box anyway, because we seem to have trouble with the interface we have. It's managed by the service provider, and I'd really like to have the IPO sending the outgoing number IDs so we can have the different
sections of our business being able to send out their respective DIDs. I don't know if an interface box can provide that service or not. The service provider did a trace and said the IPO was sending some kind of "call is queued" message that they did not support. I have two different IP addresses at this time, one to the IAD interface, and one direct to the IPO. I have moved our FAX DIDs to the IPO and that has solved the problems we were having with the FAXes going through the IAD. Other numbers on the switch are working ok, but they only get minimal traffic. The main 800 number works fine as far as call clearing through the IAD interface, but falls apart when the calls are directed straight to the IPO.
Did I cover everything?
Thanks,
-Jon
 
6.0 is for the manager and is always 2 versions higher and the same is for modules
This has a history

VMPRo should be 4.0 too and not 3.0 !!!
This can cause problems but i am not sure if that can be your problem, i don't think so

If i am right you have your trunk split up in two now ?


RTFM.gif



ACS - Implement IP Office
ACA - Implement IP Telephony -- ACA - Design IP Telephony
ACA - Voice Services Management
______________
Women and cats can do as they please and men and dogs should relax and get used to the idea!
 
What I did was have the ITSP change the routing of the calls from the IP for their IAD interface, to the IP for the IPO direct, so the calls would be presented to our switch directly instead of to the IAD. I have seven other DIDs coming to our IPO without any trouble, but those numbers only get occasional use as they are FAX lines. The problem occurs only when more than 7 calls are presented at a time. In the Office System Status screen, I see in the VCM section, that there are 124 available channels. Once there are 7 in use, I start seeing the congestion count going up and only seven calls at a time are presented on the switch. I had the ITSP switch the IP where the calls are presented back to the IAD and the calls started coming in normally. (Routed through the IAD to the PRI inputs on the switch.) I have 60 valid sip licences and 124 valid VCMs. It almost seems that the IPO is only validating 7 VCMs before it whacks out.
I did see a lot of these:
1216945306mS CMMap: Mapper::AllocateCodec unable to allocate resource
1216945382mS CMMap: Mapper::AllocateCodec unable to allocate resource
Those are taken from the system monitor.
Don't know if that might shed some light...
If the VM is not the same version as the IPO, and the VM box is the first place the calls go before they are routed into the call center, could that be the problem?


Thanks,
-Jon
 
Also, does Avaya provide the VMPro upgrade like they do the IPO? I upgraded the IPO once last year on the advice of our equipment provider. (Who by the way said that they didn't recommend upgrading until they had fully tested an upgrade.)
They didn't, however, give me the VMPro upgrade...

Thanks,
-Jon
 
There have been a lot of bugs in software re SIP which have been rectified in 4.1 and 4.2. I suggest you contact an alternative provider who knows what they are doing.
 
are you sure your SIP provider does send more then 7 calls simultaneously?
mine sends 4 and any more they want more money.
Try upgrading as suggested and test again and like TT said get a provider that has a clue what they are doing if you need their help to troubleshoot.
Also you might want to look for a BP in your area to look over the system and upgrade you to the latest version and maybe recomment some changes in the setup of the programming as you are clearly like lots of guys knowledgeable enough to be dangerous but there are still some things you should consult with people on site rather then here.
Please don't take this as an insult to you but paying a BP 2-3 hours might help you save 40 hours of headache at the end.

Joe W.

FHandw., ACS

If you can't be good, be good at it!
 
We have a contract for 92 SIPs so that isn't the problem, and when the signals are sent to the IAD box, we get the calls no-trouble, so the problem is definately in the IPO.
As far as our "service" guys go, I've experimented with the system and given them information that I've discovered to make things work. The major problem we have is that we are so far out in the sticks, that we have to import sticks to SAY we are in the sticks! No insult taken, but I'd like to figure this thing out myself if I can. The more I know about the system quirks, the faster I can react to other trouble. Our guys didn't even know HOW to set up the SIP in the first place. (They'd never put one in place I guess..)
I guess no one has seen this trouble before...
(I should deliniate between the SIP provider and the "service" guys... The SIP guys seem to be pretty responsive and willing to test and all... The "service" guys who sold us the system aren't as good, and are 200 miles away...)
Any Ideas on the CMMap error on my previous post? Is that "unable to allocate resource" error possibly where things are falling apart?
Thanks all for trying...
-Jon
 
Hi Jon
I am glad you don't take my comment personal, and being out in the sticks sounds like a dream to me at the moment.
I would try definitely the upgrade before anything else, because Avaya will not start troubleshooting 4.0 in any which way shape or form, your business partner should have at least tried to upgrade you if nothing else as they know too that that is step nimber 1 if the release is not supported any more.
The "unable to allocate resource" sounds to me like there is either a bandwidth, VCM or a license problem but it seems that this is all covered with adequate resources so if the upgrade doesn't bring the desired success then the next thing I would try is take a configuration and save it, then default the box and just put the SIP trunk into the box and test the calling behaviour without all the programming in it. If it is still there then you know it is nothing with the programming.

Joe W.

FHandw., ACS

If you can't be good, be good at it!
 
When you say you have one extn with the 800 number in, is that against a user???

Have you tried putting the 800 number as a URI in the SIP line tab?? Line>SIP Line>SIP URI>Add Channel. Put the local URI as the 800 number and create a new i/c and o/g group ids. Then add an incomming call route entry for the 800 number ot route to where you want.

I am wondering if this 7 call limit is related to the number being registered against a user. The IPO might think that 7 calls to someones DDI is quite enough for one person!!!!


I agree with Westi, "CMMap: Mapper::AllocateCodec unable to allocate resource" sounds like you do not have enough VCM channels available.

Just guessing

Jamie Green

ACA:Implement - IP Office
ACS:Implement - IP Office

Football is not a matter of life and death-It is far more important!!!!
 
It is possible that there is a sip uri, in your line, that has 7 channels enabled

This does exactly what now is happening

Check that first and if that isn ot the case then upgrade the system

Upgrading the ip office is 2-3 minutes works
Upgrading vmpro is a bit more but still not much work if done by the rules


RTFM.gif



ACS - Implement IP Office
ACA - Implement IP Telephony -- ACA - Design IP Telephony
ACA - Voice Services Management
______________
Women and cats can do as they please and men and dogs should relax and get used to the idea!
 
tlpeter
I want one of your dutch clocks please :), 2-3 minutes for upgrading sounds like a dream (OK it is only a quick one after you installed the Admin software)
Bivman
check in the URI please and confirm that you don't have only 7 channels for the URI you are using as TLPETER said.
Please say it ain't so or we have to shower TLPETER with stars.

Joe W.

FHandw., ACS

If you can't be good, be good at it!
 
Jamie and tlpeter, I looked at the URI under the SIP channel. I have only one channel setup with 100 lines authorized. (Maybe I should set it to the number of SIP licenses, duh...) I looked at adding a line for the 800, but I don't see how that would be done because there doesn't seem to be a selection for it. Only "Use User Data" or "Use Authentication Name" Maybe a newer version of the software addresses that. Also, the 800 number call-route goes to an ACD hunt, but only one user in the switch has the 800 number in it's SIP name field. (I found that the switch wouldn't accept calls to a number that wasn't somewhere in a user's SIP name, even though it was in the call route table.) I'll add the number to another user and see if the calls increase to 14 or so. If that is the case, I'll just create several phantom extensions with the 800 and main numbers in their SIP name fields. (As well as upgrading the switch and VMPro softwares.)
I'll let you know what happens. And if there is something else you think of, let me know?
Thanks,
-Jon
 
Westi, thanks too.. I was editing my previous post and your comment came in before I was done... Didn't see it till I had hit the send.
-Jon
 
No worries, but you can set the number of channels to anything you like even without licenses (I use 4 channels but only have 2 licenses here) it still works but only until the number of licenced channels is reached. So with your 60 SIP licenses you chould be definitely good for the 7 calls.
I guess upgrade is the next step
this is where you can get the 4.2 firmware.

Joe W.

FHandw., ACS

If you can't be good, be good at it!
 
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