> Thanks for the tip, that made things work, it is really > difficult for me to understand the different config files and > especially the extensions.conf, it is very confusing. I am > trying to learn though. > > Now that I have got outgoing calls to work from the sip > phone. How can I route incoming calls on the pstn line > (x100p) to the sip phone? > > Thanks!
First, I would dial the telephone number of the line plugged into the X101P and make sure that the demo answers to verify that things are working correctly...assuming that works, you just need to modify your extensions.conf a little bit... Your [default] context includes [demo] which has an answer line in it, followed by the rest of the items necessary to playback the demo. So if you want an incoming call to ring directly to your x-lite, I would remove the include for [demo] from your [default] context (but leave the include for [local] so that you can make outbound calls!...then inside your [default] context (just below the include for [local] for example) add lines that will answer the phone and ring your x-lite: (note that below, the SIP/1000 is just an example...the '1000' should be whatever name you gave your x-lite in sip.conf) exten => s,1,Wait exten => s,2,Answer exten => s,3,Dial(SIP/1000,20,r) Save the changes and reload asterisk, try calling the line connected to the X101P and if your x-lite has registered with asterisk correctly, it should ring there...look on the wiki (www.voip-info.org) for the specific syntax of the Dial command and it's options, also the above is a very basic config, with no timeouts specified, etc...it should work, but should/could be made more robust after you get it working initially. Marty _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
