255 — 2009-10-04T03:17:33-04:00 — #1
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)
tdf — 2009-10-04T22:02:50-04:00 — #2
Heres what I did, might suit you might not.
I set up a DYNDNS account email@example.com, 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
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 firstname.lastname@example.org 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.
skykingoh — 2009-10-04T23:43:05-04:00 — #3
You can also put a name in the alias field. The you can call email@example.com
255 — 2009-10-09T19:59:22-04:00 — #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.
mayak — 2010-01-04T12:56:07-05:00 — #5
hi all, hi alan,
please do post back if you have solved this mystery -- i have plenty of firstname.lastname@example.org "phone" numbers, but cannot dial ...
rymes — 2010-01-20T09:41:34-05:00 — #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.
mickecarlsson — 2010-01-20T11:10:04-05:00 — #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 email@example.com or whatever sip uri you want to dial. Save and reload.
Dial extension 2222 and it will dial the sip uri.
rymes — 2010-01-20T14:10:19-05:00 — #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?
sepehr741 — 2011-08-05T07:10:04-04:00 — #9
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?
cguasco — 2011-11-24T10:40:12-05:00 — #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 .
adhominem — 2011-11-24T13:01:13-05:00 — #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
You can even get really fancy and use:
That entry will pass the phone number that is dialed thorugh the trunk in the $OUTNUM field.
skykingoh — 2011-11-24T23:28:07-05:00 — #12
To receive URI call populate SIP alias field and enable anonymous SIP.