[Asterisk-Users] PSTN Gateway X101P

[Jason Armentrout]

Thanks Marty,
That works now, the caller id on Xlite only shows the name for some reason, not
the number, but anyway it now rings in.

When I call the pstn number, the zaptel picks up the line on the first ring and
then forwards it to the sip phone and rings it. Is there anyway to prevent the
zaptel from picking up the line until the sip phone actully answers the call.
This way I could answer the phone either locally on a regular analog handset or
through the sip phone.

The way it is now, it only rings my phones in the house 1 time.

Jason


Quoting Marty Mastera <mmastera at m3resources.com>:

> > 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
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