Sip uri

FreePBX Installation / Upgrade
1

Hey guys,

I have asked a couple of times for help in the Trixbox forums but have always guy not very intelligent answers.

I’m no Linux guru nor am I an expert of dial plans (that’s why I use Free PBX) but I’m trying to get SIP URI working with no luck.

Can anyone help me please? I have Trixbox 2.6.3 (Asterisk 1.4)

With thanks

2

Heres what I did, might suit you might not.

I set up a DYNDNS account [email protected], the 684646464 can be anything you want.

In FreePBX I turned on accept anonymous inbound sip, its under general settings.

I set up a incoming route

Description: Sip
DID Number: 684646464

and pointed it to a ring group, extension or whatever you want.

I also set up another route

Description: Catch All
DID Number: _.

and pointed it to terminate call hang up. This will stop random sip calls getting through, you will need to have a route defined for all your trunks so they don’t fall into this catch all.

That’s it, someone calls me on [email protected] and they get through.

Your original thread has some other options

http://trixbox.org/forums/trixbox-forums/open-discussion/sip-url

if none of these are what you want, you will have to do a better job of explaining your needs.

3

You can also put a name in the alias field. The you can call [email protected]

4

At this point I’m still working on Outbound SIP URI, there seems to be conflicting information and are unsure if I have to modify the dial plan or what is involved.

Thanks

Alan Scott

5

hi all, hi alan,

please do post back if you have solved this mystery – i have plenty of [email protected] “phone” numbers, but cannot dial …

thanks

m

6

It would be excellent if someone could point us in the right direction for implementing outbound SIP/IAX URI dialing through freepbx. Being able to take advantage of the peer-to-peer part of SIP/IAX would be excellent.

Also, it might be a good idea to set up the catch-all incoming route to direct users to a PIN or an IVR, rather than congestion or hangup. I guess it all depends on what you use the system for.

Tom

7

A quick method is to create a new extension, for example 2222, save it and then edit it again, in the dial field put sip/[email protected] or whatever sip uri you want to dial. Save and reload.

Dial extension 2222 and it will dial the sip uri.

8

Well, with the inbound route and the custom extension, that ought to work. How about being able to dial anyone with a URI, not just someone you call often enough to program a custom extension for them?

Also, does anyone have a good link for a discussion of the risks associated with allowing anonymous SIP connections inbound? I presume potential toll-fraud and other issues are concerns. What else?

Tom

9

Hi

I searched a lot for finding a way to call SIP URI from Freepbx, and I just found what Mikael mentioned, by adding the address in dial field and then calling that extension.
But as I’m trying to call a SIP URI from follow me option and subsequently using Fixed CID , but this solution as it’s like an internal call, the freepbx doesn’t change the CID!!
Do you have any solution to be able to make a SIP URI call with fixed CID or do it like an external call and then using followme fixed cid?
thanks

10

I found this
< http://projects.colsolgrp.net/projects/urihand/wiki/Installing_SIP_URI_Handling_Module_for_FreePBX#Original-Manual-Method >

and seems to work !!!

Is a part o a project to integrate a URI module in FreePBX, but seems abandoned (last edit dec 2010)

In any case i’m not able to receive URI call .

Bye,

Charles

11

You can create a trunk that goes to a SIP URI as well. This would solve your Caller ID problems.

Custom Trunk, use this as the custom dial string

SIP/name@ipaddres

i.e.

SIP/[email protected]

You can even get really fancy and use:

SIP/[email protected]

That entry will pass the phone number that is dialed thorugh the trunk in the $OUTNUM field.

12

To receive URI call populate SIP alias field and enable anonymous SIP.