Dialplan condition to test if SIP client is registered
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<http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridge#Implementing_Failover>, however <shameless_plug>it is covered very nicely in the new FreeSWITCH Cookbook</shameless_plug> 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 <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
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Thaddeus & Michael: As Michael stated there are many ways to fail over if a phone/user isn't registered. However sometimes you may want to actually check the registration yourself. Or even do something based on a contact being registered, that has nothing to do with the actual one you are calling. I normally use the API: sofia_contact([profile/]<user>@domain). It returns the channel string you would pass to bridge. If the user isn't registered it returns "error/<the error>" so it would use the logical "error" endpoint to handle the right treatment. If the regex "^error" match is true, then the user isn't registered. The extra thing you get from that API call is that you can use a "*" for the profile, and find out which SIP profile the user is registered in. For example in the case of an Android client, that may be using WIFI or 3G, and you have an external and internal profile they can register, you can tell if the user is "on site" or "off site" by the profile it returns, and can be more restrictive of the calls you send to someone when they're out of the office. --Dave _____ From: Michael Collins [mailto:msc@freeswitch.org] To: FreeSWITCH Users Help [mailto:freeswitch-users@lists.freeswitch.org] Sent: Fri, 04 May 2012 10:34:31 -0700 Subject: Re: [Freeswitch-users] Dialplan condition to test if SIP client is registered 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 <shameless_plug>it is covered very nicely in the new FreeSWITCH Cookbook</shameless_plug> 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 <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 FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Dave, Thanks for the extra info. I had forgotten that we added the sofia_contact(*/user@domain) syntax to let you figure out which profile they're on. -MC On Fri, May 4, 2012 at 11:24 AM, Dave R. Kompel <drk@drkngs.net> wrote:
** Thaddeus & Michael:
As Michael stated there are many ways to fail over if a phone/user isn't registered. However sometimes you may want to actually check the registration yourself. Or even do something based on a contact being registered, that has nothing to do with the actual one you are calling.
I normally use the API: sofia_contact([profile/]<user>@domain). It returns the channel string you would pass to bridge. If the user isn't registered it returns "error/<the error>" so it would use the logical "error" endpoint to handle the right treatment. If the regex "^error" match is true, then the user isn't registered.
The extra thing you get from that API call is that you can use a "*" for the profile, and find out which SIP profile the user is registered in. For example in the case of an Android client, that may be using WIFI or 3G, and you have an external and internal profile they can register, you can tell if the user is "on site" or "off site" by the profile it returns, and can be more restrictive of the calls you send to someone when they're out of the office.
--Dave
Thanks for the information. I got something working today using sofia_contact, but am having issues with SIP registrations surviving for a good deal of time after the user has disconnected from the network. Basically, the SIP client is no longer available when the user leaves the building and is disconnected from Wifi. When this happens, sofia_contact continues to return a value indicating the client is registered for quite some time. I found the command "sofia profile <profile_name> check_sync", which I used successfully to invalidate registrations for users that have disconnected from the network. But I could not figure out how to make this execute before my "sofia_contact" condition in the dialplan. I am assuming this is because I need it to run before the ROUTING stage, and I can only get it to run in the EXECUTE stage. So I have a couple questions. Is check_sync quick enough to run on every inbound call prior to making routing decisions? Also, according to the comments here http://wiki.freeswitch.org/wiki/FS_weekly_2011_06_29 I should be able to give check_sync a user@domain argument, but this does not seem to work. Are there other ways to determine if a bridge failed because a user's SIP client was not reachable, versus a timeout or call rejection? Thanks for helping with this! -- Thaddeus On 5/4/2012 1:24 PM, Dave R. Kompel wrote:
Thaddeus & Michael: As Michael stated there are many ways to fail over if a phone/user isn't registered. However sometimes you may want to actually check the registration yourself. Or even do something based on a contact being registered, that has nothing to do with the actual one you are calling. I normally use the API: sofia_contact([profile/]<user>@domain). It returns the channel string you would pass to bridge. If the user isn't registered it returns "error/<the error>" so it would use the logical "error" endpoint to handle the right treatment. If the regex "^error" match is true, then the user isn't registered. The extra thing you get from that API call is that you can use a "*" for the profile, and find out which SIP profile the user is registered in. For example in the case of an Android client, that may be using WIFI or 3G, and you have an external and internal profile they can register, you can tell if the user is "on site" or "off site" by the profile it returns, and can be more restrictive of the calls you send to someone when they're out of the office. --Dave
------------------------------------------------------------------------ *From:* Michael Collins [mailto:msc@freeswitch.org] *To:* FreeSWITCH Users Help [mailto:freeswitch-users@lists.freeswitch.org] *Sent:* Fri, 04 May 2012 10:34:31 -0700 *Subject:* Re: [Freeswitch-users] Dialplan condition to test if SIP client is registered
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 <http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridge#Implementing_Failover>, however <shameless_plug>it is covered very nicely in the new FreeSWITCH Cookbook</shameless_plug> 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 <http://freeswitch.org/>, upper left corner.)
Thanks, Michael
On Fri, May 4, 2012 at 10:10 AM, thaddeus@thogan.com <mailto:thaddeus@thogan.com> <thaddeus@thogan.com <mailto: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 <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com <http://www.cudatel.com/>
Official FreeSWITCH Sites http://www.freeswitch.org <http://www.freeswitch.org/> http://wiki.freeswitch.org <http://wiki.freeswitch.org/> http://www.cluecon.com <http://www.cluecon.com/>
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org <http://www.freeswitch.org/>
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
What about continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION 2012/5/5 Thaddeus Hogan <thaddeus@thogan.com>
Thanks for the information. I got something working today using sofia_contact, but am having issues with SIP registrations surviving for a good deal of time after the user has disconnected from the network.
Basically, the SIP client is no longer available when the user leaves the building and is disconnected from Wifi. When this happens, sofia_contact continues to return a value indicating the client is registered for quite some time.
I found the command "sofia profile <profile_name> check_sync", which I used successfully to invalidate registrations for users that have disconnected from the network. But I could not figure out how to make this execute before my "sofia_contact" condition in the dialplan. I am assuming this is because I need it to run before the ROUTING stage, and I can only get it to run in the EXECUTE stage.
So I have a couple questions. Is check_sync quick enough to run on every inbound call prior to making routing decisions? Also, according to the comments here http://wiki.freeswitch.org/wiki/FS_weekly_2011_06_29 I should be able to give check_sync a user@domain argument, but this does not seem to work.
Are there other ways to determine if a bridge failed because a user's SIP client was not reachable, versus a timeout or call rejection?
Thanks for helping with this!
-- Thaddeus
On 5/4/2012 1:24 PM, Dave R. Kompel wrote:
Thaddeus & Michael:
As Michael stated there are many ways to fail over if a phone/user isn't registered. However sometimes you may want to actually check the registration yourself. Or even do something based on a contact being registered, that has nothing to do with the actual one you are calling.
I normally use the API: sofia_contact([profile/]<user>@domain). It returns the channel string you would pass to bridge. If the user isn't registered it returns "error/<the error>" so it would use the logical "error" endpoint to handle the right treatment. If the regex "^error" match is true, then the user isn't registered.
The extra thing you get from that API call is that you can use a "*" for the profile, and find out which SIP profile the user is registered in. For example in the case of an Android client, that may be using WIFI or 3G, and you have an external and internal profile they can register, you can tell if the user is "on site" or "off site" by the profile it returns, and can be more restrictive of the calls you send to someone when they're out of the office.
--Dave
------------------------------ *From:* Michael Collins [mailto:msc@freeswitch.org <msc@freeswitch.org>] *To:* FreeSWITCH Users Help [mailto:freeswitch-users@lists.freeswitch.org<freeswitch-users@lists.freeswitch.org> ] *Sent:* Fri, 04 May 2012 10:34:31 -0700 *Subject:* Re: [Freeswitch-users] Dialplan condition to test if SIP client is registered
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<http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridge#Implementing_Failover>, however <shameless_plug>it is covered very nicely in the new FreeSWITCH Cookbook</shameless_plug> 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 <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
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services:consulting@freeswitch.orghttp://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com
Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.frees...
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
****
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
participants (5)
-
Anton Kvashenkin
-
Dave R. Kompel
-
Michael Collins
-
Thaddeus Hogan
-
thaddeus＠thogan.com