Sip uri

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

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

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

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

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.


Alan Scott

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 …



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.


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.

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?



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?

I found this
< >

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 .



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/[email protected]


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.

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