SIP URI routing within dialplan?

[snoms]

Hi,
one nice thing about voip is making free calls to other user having their own SIP URI.
Image the following setup:
- several hard and softphones are connected to a 3cx system
- there´s one Mediatrix voip gateway for incoming and outgoing landline calls.
- For outgoing calls to landline numbers, users have to to add a "0" before every actual number.
- ...

There are many SIP based external service, e.g. conferencing.
What I´d like to have is: I want my users being able calling SIP-URIs from their e.g. hardphones, so instead of typing a number they dial sip:[email protected]

So the phone basically takes this and gives the SIP URI to the registrar, which is in this case, the 3cx voip server. Next thing is that the server has to deal with this within the dialplan.

Is this possible with 3cx? How can this be enabled within the dialplan?

 

[leejor]

It would depend on how many different domains that you are calling. If it is a handful, then you could set up a trunk-group for each, that would send a dialled number out to that URL. If you are sending to an alpha-numeric address you may need to have the outbound rules do the translation for you. i know that this works for alpha-numeric Skype calls but haven't tried it for any others.

Dial 123, strip 3, prepend with "JBrown", send out on trunk ABC, that sends out [email protected].

It would be like a speedcall but would be somewhat limited, and really only feasible there were a limited number of destinations that you were calling repeatedly.
There may be other ways to do it as well...just one, off the top of my head.

 

[snoms]

Hi,

thank you for your answer. But indeed I want to have unlimited domains. I do not want ot predefine all SIP domains.
What about this, for each SIP URI call, the user types a prefix, e.g.

*789SIP:[email protected]

Within dialplan *789 is stripped , take the rest and work with it...

 

[leejor]

The problem is that the outbound rules are numeric based and pick up the @domain.com part from whichever trunk group the rule points to. It does not seem to be intended to do random direct SIP calls. Someone else may have come up with some sort of work-around though.