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

[Moises Silva]

Your problem is related to not reading the documentation in voip-info.org :)

You can originate a call to anyplace doing:

- First a LoginAction.
- Then an Originate action with the proper arguments.

In the example you put, you are doing neither of them. You can test manually
how the protocol works doing a login from a telnet client:

telnet localhost 5038 (in case your in the asterisk box)

Action: Login\r\n
Username: someuser\r\n
Secret: somesecretpassword\r\n\r\n

Action: Originate
Channel: SIP/13 <-- this should be the first phone you want to ring
(your own phone usually)
Context: somecontextwithoutbountpatterns
Exten: XXXXXXXXXXXX <--- extension that will make your call
Priority: 1 (usually one is fine)


Best Regards


On 1/2/06, Don Fanning <don at 00100100.net> wrote:
> 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.
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>


--
"Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org"