Hello, I have been making extensive use of the callee_id_name and callee_id_number variables so that our IP phones update the display to show who they are actually talking to on the other end. Generally, it works very well, but it only seems to work if there is a bridge. Is there a way to set the callee_id in a single legged call? ie. I want the name of the company to come up when the caller is in the main IVR, or the name of the conference to come up if they are in the conference room. Is there a way to do this? Thanks! -Tim
Hi Tim, would love this as well. I guess the answer application should see if effective_callee_id_* is set and trigger a display update on the calling phone. It would also be nice to be able to trigger display updates to the a leg during (ivr) execution. Asterisk's CONNECTEDLINE() allows all of this. It's using Reinvites/Updates w/ PAI-Header and the syntax is cumbersome as everything asterisk is, but at least it works with most phones. (Snom 3xx, Snom M3, Siemens OpenStage, Polycom confirmed as i'm working with these every day) FreeSWITCH display updates seem to rely on workarounds (grep polycom or snom in mod_sofia). I would like to be able to disable the INFO/sipfrag thing for Snom, use PAI instead of RDNIS in the A-Leg and send Reinvites for Display Update etc., but it all seems to be hard coded right now. Regards, Markus On Wed, Mar 28, 2012 at 16:02, Tim St. Pierre <fs-list@communicatefreely.net> wrote:
Hello,
I have been making extensive use of the callee_id_name and callee_id_number variables so that our IP phones update the display to show who they are actually talking to on the other end.
Generally, it works very well, but it only seems to work if there is a bridge. Is there a way to set the callee_id in a single legged call? ie. I want the name of the company to come up when the caller is in the main IVR, or the name of the conference to come up if they are in the conference room.
Is there a way to do this?
Thanks!
-Tim
_________________________________________________________________________ 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
Its not a workaround, its because each phone does it differently and other phones consider each other's way a fatal err and hangup calls. set effective_callee_id_name and effective_callee_id_number variables before you execute ring_ready, pre_answer or answer or before you run an app that may answer the call and you will get what you are looking for. If the call is already up you can use the send_display app <action application="send_display" data="This will be the name|1234"/> And from ESL or cli uuid_display <uuid> This will be the name|1234 On Thu, Mar 29, 2012 at 2:32 AM, Markus Lindenberg <markus.lindenberg@gmail.com> wrote:
Hi Tim,
would love this as well.
I guess the answer application should see if effective_callee_id_* is set and trigger a display update on the calling phone. It would also be nice to be able to trigger display updates to the a leg during (ivr) execution.
Asterisk's CONNECTEDLINE() allows all of this. It's using Reinvites/Updates w/ PAI-Header and the syntax is cumbersome as everything asterisk is, but at least it works with most phones. (Snom 3xx, Snom M3, Siemens OpenStage, Polycom confirmed as i'm working with these every day)
FreeSWITCH display updates seem to rely on workarounds (grep polycom or snom in mod_sofia). I would like to be able to disable the INFO/sipfrag thing for Snom, use PAI instead of RDNIS in the A-Leg and send Reinvites for Display Update etc., but it all seems to be hard coded right now.
Regards, Markus
On Wed, Mar 28, 2012 at 16:02, Tim St. Pierre <fs-list@communicatefreely.net> wrote:
Hello,
I have been making extensive use of the callee_id_name and callee_id_number variables so that our IP phones update the display to show who they are actually talking to on the other end.
Generally, it works very well, but it only seems to work if there is a bridge. Is there a way to set the callee_id in a single legged call? ie. I want the name of the company to come up when the caller is in the main IVR, or the name of the conference to come up if they are in the conference room.
Is there a way to do this?
Thanks!
-Tim
_________________________________________________________________________ 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
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
Great, thanks, confirmed that works for me. After i downgraded Snom firmware to 8.4.22 (later Firmware is broken in different ways on 370 and 360), send_display works (also for one-legged calls) and connectedline gets updated after bridge/transfer. The only thing that doesn't work for Snom phones is connected lind info in ringing phase. Freeswitch is sending RPID to the caller, Snom expects PAI. Polycom works great. Aastra doesn't display anything at all before 200OK, i'll have to investigate what they expect. Setting sip_cid_type just affects the called party. If i tell the Snom phones that they're connected to a Cisco Callmanager, they will read RPID. But i don't know about possible side effects. On Thu, Mar 29, 2012 at 18:49, Anthony Minessale <anthony.minessale@gmail.com> wrote:
Its not a workaround, its because each phone does it differently and other phones consider each other's way a fatal err and hangup calls.
set effective_callee_id_name and effective_callee_id_number variables before you execute ring_ready, pre_answer or answer or before you run an app that may answer the call and you will get what you are looking for.
If the call is already up you can use the send_display app
<action application="send_display" data="This will be the name|1234"/>
And from ESL or cli
uuid_display <uuid> This will be the name|1234
On Thu, Mar 29, 2012 at 2:32 AM, Markus Lindenberg <markus.lindenberg@gmail.com> wrote:
Hi Tim,
would love this as well.
I guess the answer application should see if effective_callee_id_* is set and trigger a display update on the calling phone. It would also be nice to be able to trigger display updates to the a leg during (ivr) execution.
Asterisk's CONNECTEDLINE() allows all of this. It's using Reinvites/Updates w/ PAI-Header and the syntax is cumbersome as everything asterisk is, but at least it works with most phones. (Snom 3xx, Snom M3, Siemens OpenStage, Polycom confirmed as i'm working with these every day)
FreeSWITCH display updates seem to rely on workarounds (grep polycom or snom in mod_sofia). I would like to be able to disable the INFO/sipfrag thing for Snom, use PAI instead of RDNIS in the A-Leg and send Reinvites for Display Update etc., but it all seems to be hard coded right now.
Regards, Markus
On Wed, Mar 28, 2012 at 16:02, Tim St. Pierre <fs-list@communicatefreely.net> wrote:
Hello,
I have been making extensive use of the callee_id_name and callee_id_number variables so that our IP phones update the display to show who they are actually talking to on the other end.
Generally, it works very well, but it only seems to work if there is a bridge. Is there a way to set the callee_id in a single legged call? ie. I want the name of the company to come up when the caller is in the main IVR, or the name of the conference to come up if they are in the conference room.
Is there a way to do this?
Thanks!
-Tim
_________________________________________________________________________ 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
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
_________________________________________________________________________ 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 (3)
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Anthony Minessale
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Markus Lindenberg
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Tim St. Pierre