Thaddeus, FYI, doing an API from the dialplan is a piece of cake: http://wiki.freeswitch.org/wiki/Mod_commands#From_the_Dialplan In your case put this line in your dialplan before you do the sofia_contact check: <action application="set" data="api_result=${sofia profile internal check_sync}"/> That should solve the problem of people wandering off and their SIP clients not unregging. Give it a try and let us know how it works. -MC On Tue, May 8, 2012 at 8:11 AM, thaddeus@thogan.com <thaddeus@thogan.com>wrote:
Thanks for the information, I have explored the sofia_contact option and it is promising, but I am unsure of how to make it work in my dialplan.
I think I need to elaborate on the indended behavior some more. A user will have three possible routes by which they can be reached. Two SIP accounts and a POTS number. Each user has three routes but only two devices, a desk phone and a cell phone. The cell phone is a target of the POTS number and is also one of the SIP clients. I am afraid I did not mention the desk phone initially, and I apologize for my question being incomplete.
Given this setup I want to select one of these two dialplans:
1. If SIP on cell phone is registered, ring <DESK SIP> && <CELL SIP>.
2. If SIP on cell is not registered, ring <DESK SIP> && <CELL POTS>
sofia_contact looks promising, however it returns a string indicating that a user is registered long after they have become unavailable. Example: User walks out of building and cell phone loses the Wifi connection. The SIP client did not un-register, it just went away. I have found that "sofia check_sync" will check all registrations and correctly identify clients that have gone off the network as no longer being registered.
However, I cannot figure out how to run sofia_check sync before testing the output of sofia_contact in my dialplan, any thoughts on how to do this? Or is there a better way to get my above listed call routing to occur based on the availability of a SIP client?
Also, I did investigate using continue_on_fail to check for hangup causes that did involve the SIP client being unreachable, but I could not make the above logic occur using only that setting. That is, I always want to ring <DESK SIP>, but only one of <CELL POTS> or <CELL SIP>, and never both simultaneously or in sequence.
Might I need to go into a Lua script for this kind of logic?
Thanks again for your responses!
-- Thaddeus
On Friday, May 4, 2012 12:34 PM CDT, Michael Collins <msc@freeswitch.org> wrote:
Hi Thaddeus,
You definitely have several options here. While you could use the sofia_contact API, I'm wondering if perhaps you could let the dialplan and the bridge app do the work for you.
You could bridge directly to the Android user and then if it fails it could then continue in the dialplan and ring the mobile number. This is covered briefly on the wiki**, however **it is covered very nicely in the new FreeSWITCH Cookbook** in chapter 1.
Try out the basic failover (pipe-separated list of endpoints) and if that doesn't work let us know. Maybe you could hop on IRC and talk live with other community members. Of course, it can't hurt to raise your karma by getting the FreeSWITCH books. :) (See freeswitch.org, upper left corner.)
Thanks, Michael
On Fri, May 4, 2012 at 10:10 AM, thaddeus@thogan.com **wrote:
I was wondering if there is a way to tell if a SIP client is registered with a dialplan condition? Or maybe a better way to accomplish the following?:
The users' cell phones (Android) are running a SIP client and connect to freeswitch via Wifi when in the office. I want calls destined for a given user to ring only their extension if they are connected via SIP, and ring only their mobile number when they are not connected.
Currently I just take every incoming call and bridge it back out to their mobile number, but 95% of the time these users are on the Wifi network, and I could just pass the call to their phones via SIP.
I have tested hunt groups but the callers hear ringing for far too long. Ring groups don't work because it is confusing when the user simultaneously receiving the same call via SIP and mobile on the same phone.
Thanks in advance for any insights and help!
-- Thaddeus _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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