[Asterisk-Users] Q: How to dial out / transfer calls with manager

[Don Fanning]

Greetings,

Here's my issue.  My local free VSP isn't transfering proper DTMF
(inband or converting to RFC2833) so I'm stuck with making a php
interface so my roommates whom are not using softphone/ata devices to
call out via * (and thusly get the better deals in Long Distance).

I've tried using the Manager interface to creating the connection
however when I create a Channel: it needs to be something virtually
attached to the system.  I'm trying to see if there is a way around it.

IE: Currently I drop

 fputs($socket, "Secret: ibanez\r\n\r\n");
 fputs($socket, "Action: Originate\r\n");
 fputs($socket, "Channel: $mytelephone\r\n");
 fputs($socket, "Exten: 1$callnumber\r\n");
 fputs($socket, "Priority: 1\r\n\r\n");

>From a php script with $mytelephone being the home phone via sip like
SIP/1235551212 at sipprovider and $callnumber is the destination number
which would default to my $TRUNK.  However since the channel isn't
registered on the system it will fail.

Is there a way of cheating this via callpark or meetme?  How about a
dummy iaxclient to originate then dumps to a meetme with the $callnumber
doing the same?  I find this very limiting as I can't route calls the
way I want to.  (the DTMF issue is worse... Don't get me started. ;)

Ideas?  Thanks.