[Asterisk-Users] problems with 1.2 Beta1

[Dan Fernandez]

Greetings!

I am running a small callcenter with 10 analog lines, aprox. 15 agents and using Asterisk 1.2beta1. We have 10 sipura 3000s connected to the PSTN and a few linksys PAP2s.

The ports connected to phones are configured as SIP/200s and SIP/300s and the ones connected to the PSTN as SIP/900s.

When an agents makes a call, asterisks bridges a SIP/200 with a SIP/900. However, every now and then I see calls bridges between two SIP/900s which of course should not occur. The agents claim then that sometimes when they are on a call other agents can sneak in the call.

Previously, when I was using version 1.0.9 and had a similar problem which I fixed it with SetGroup and CheckGroup. When I upgraded to 1.2Beta1 I replaced those two funtions with the corresponding functions in the new version, but it appears these two functions don't work as they used to, and that's why the lines are getting mixed. My extensions.conf looks like:

[macro-stdial]
exten => s,1,NoOp(${GROUP_COUNT(L_${ARG1})})
exten => s,2,Set(GROUP()=L_${ARG1})
exten => s,3,NoOp(${GROUP_COUNT(L_${ARG1})})
exten => s,4,GotoIF($[${GROUP_COUNT(L_${ARG1})}>1]?${EXTEN}|106:${EXTEN}|5)
exten => s,5,Dial(SIP/${ARG1}/${ARG2},45,grTH)
exten => s,6,AGI(calif.agi)
exten => s,7,hangup
exten => s,106,NoOP

The agents also claim that the calls sometimes hangup abruptly while they are on the phone. I don't have more info than that, other than this occurs on just any ATA device. Any ideas on how can i debug these problems?

Thanks much
Dan









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