Question

Asterisk Call Limit Problem...

Asked by: latdon

I'm working on an Asterisk system and I'm finding that the maximum number of incoming calls on any channel is 6.  I need this channel to communicate with a conferencing system with over 20 simultaineous calls.  I've set the call-limit and incominglimit parameters on the channel but they have no affect.  When I check "sip show inuse" i can see the status of 6/0 and "Limit" shows 200.  Any ideas?

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Asked On
2007-10-23 at 14:48:51ID22913190
Tags

asterisk

,

limit

,

call

Topics

Voice Over IP

,

Open Source Programming

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Answers

 

by: tvman_odPosted on 2007-10-23 at 20:49:08ID: 20136231

Could you tell which version of Asterisk do you run. Some versions had issues with it.

 

by: grbladesPosted on 2007-10-24 at 00:39:37ID: 20136928

In addition please clarify if you are using standard Asterisk or Trixbox.

 

by: feptiasPosted on 2007-10-24 at 02:28:35ID: 20137334

Perhaps the limit is being imposed by your SIP service provider, especially if you are using an account type that is intended for use with a SIP phone rather than a PBX.

 

by: latdonPosted on 2007-10-24 at 05:01:40ID: 20137969

The system is an Asterisk only implementation running version is 1.4.6.  I had the SIP provider run their analizer on the failed calls and they're sending me the invites but my system is either not seeing them or ignoring them.  The provider is very good to work with and they send me the call ladders from their Hammer.

 

by: feptiasPosted on 2007-10-24 at 05:26:39ID: 20138122

I suggest you use the CLI command: sip debug
This allows you to see the SIP packets (and responses) reaching your Asterisk server.
You switch it off again using: sip no debug

 

by: tvman_odPosted on 2007-10-24 at 06:01:30ID: 20138401

try to capture traffic in the SIP signalling port, 5060 default. Make calls until it failes and review the exchange. You should be able to see INVITEs and answers. At this point you would be able to see if INVITEs mare all the way to your box and answers sent back to your VoIP provider. If this initial fase looks ok, start Asterisk debug and as an option increase logging level.
Another question, are this calls will be transcoded or it's passthrough to SIP phones? It might be a limit of available channels for G.729 codec if you purchased it.

 

by: tvman_odPosted on 2007-10-24 at 06:02:44ID: 20138412

BTW, which Asterisk application do you use for conferencing?

 

by: latdonPosted on 2007-10-24 at 06:56:17ID: 20138935

Thanks - that's my plan.  I'm going to get a packet capture on the server network to see what's happenning to the packets. I have "SIP Debug" running on the console and I don't even see the failed calls coming in.  No mesaging at all.  The conferencing server is Cisco Meeting Place Express.

 

by: latdonPosted on 2007-10-31 at 06:29:49ID: 20185232

Problem Solved - the packet capture revealed that the firewall was blocking the traffic after all.  Be aware that the Juniper SSG5 firewalls have an ALG default SIP call limit of 16 but if you call their tech support they'll show you how to boost that to 48 - the command isn't anywhere in the documentation BTW.  An additional license key can get you to over 100 but at an extra cost.   Thanks for all the help.

 

by: Computer101Posted on 2008-09-10 at 18:52:45ID: 22445176

PAQed with points refunded (250)

Computer101
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20120131-EE-VQP-002

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