[Asterisk-Users] PSTN Gateway X101P

[Marty Mastera]

> 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