Asterisk expert needed.2 - repost

Cancelled Posted Dec 24, 2013 Paid on delivery
Cancelled Paid on delivery

First you have to set the parameter 'qualify=yes' on all the sip extensions at sip.conf. This parameter will become your extension reachable to your server.

Check your dialplan rule at extensions.conf. Must Have something like this:

exten => _2X,1,Dial(SIP/${EXTEN},300,Ttr)

exten => _2X,n,Hangup()

At the end, but very important, check the codec you are using. You are using the 3G network, your connection probably will have an high latency.

Set on your softphone codecs that will give you a better compression like GSM or G729.

Just a last note: If you have firewall, check if the sip ports are opened in two ways, and not only out-side to in-side.

Asterisk PBX Linux VoIP

Project ID: #5258410

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Remote project Active Dec 24, 2013