Swich channels


#1

Hi
I have Asterisk 1.0.9 working fine, when a call is made to PSTN is using a ITSP ,I have a 256 Internet connection and a E1 card adapter, so i can make only 3 concurrents calls to PSTN, i want to make calls by E1 when is bussy the ITSP Connection,
Its possible to swich sip and zap channels???
Any ideas???

Thank you


#2

That’s an easy one.

You would create a macro that will dial your ITSP the way that you normally would, and when your ITSP lines are all busy, would jump to the E1 trunks (at priority x+101) and try your E1 trunks.

[macro-dialitspoverflow]
exten => s,1,Dial(${ITSP}/${MACRO_EXTEN})
exten => s,2,Congestion
exten => s,102,Dial(${E1_TRUNK}/${MACRO_EXTEN})
exten => s,203,Busy

Then you just refer to the macro:

[default]
exten => _NXXNXXXXXX,1,Macro(dialitspoverflow)


#3

thank you for your answer, the trouble in concurrent calls is that quality goes down when more calls, i ve notice that the optimal quality is with 3 concurrent calls , so if i implement your macro it will let me make 5 calls before send me to congestion event, do you know a way to fix that???


#4

You need to set a “calls in progress” value anytime you place a call.

Just use a SetVar() command to do that.

On hangup, just decrement the variable.

(Or, load Asterisk onto a better computer. Either should solve your porblem.)