the command only supports sending name, it will use the destination_number from caller profile. If you use esl or another way to generate the session message you can send both name and number. check out the code in mod_sofia.c at 1636 if you want to follow the logic how it decides what to send. Also, we could take a patch to send_display to let you specify both if you can figure out a backwards compatible way to specify it. See uuid_send_message for an example of how to set multiple args in that message.
On May 22, 2015, at 4:10 AM, Joshua Gigg <giggsey@gmail.com> wrote:
In mod_commands.c, it appears to split the argv by spaces, and pass that to the switch. mod_sofia will take the second argument as the phone number.
When I try it from the API (uuid_display callid Name 1234), it sends the sip as:
From: "Name 1234" ThisCallToAddress
Instead of splitting it up.
My snom does then display Name 1234 on the screen, but as a text name, instead of updating the number.
This can work, but I'd rather be able to update the number too.
On 21 May 2015 at 15:26, Michael Jerris <mike@jerris.com <mailto:mike@jerris.com>> wrote: from mod_dptools:
SWITCH_ADD_APP(app_interface, "send_display", "Send session a new display", "Send session a new display.", display_function, "<text>", SAF_SUPPORT_NOMEDIA);
there is a similar api command in mod_commands. I think if you have ignore_dispaly_updates it may not work, try it out to be sure.
On May 21, 2015, at 4:05 AM, Joshua Gigg <giggsey@gmail.com <mailto:giggsey@gmail.com>> wrote:
I do have ignore_dispaly_updates=true, but that's on outbound phones, my displays were updating with the number being bridged out to.
Also, I want to trigger it manually rather than just automatically when a transfer takes place.
On 20 May 2015 at 17:26, Michael Jerris <mike@jerris.com <mailto:mike@jerris.com>> wrote: this should happen by default unless you have ignore_dispaly_updates=true.
On May 20, 2015, at 12:18 PM, Joshua Gigg <giggsey@gmail.com <mailto:giggsey@gmail.com>> wrote:
Digging up an old thread, but I've been looking into this a bit more.
What I want to do is described in RFC 4916. What I want to try to use an application / api command to generate a SIP INFO/UPDATE packet that would update the Caller ID.
On 29 January 2015 at 20:06, Joshua Gigg <giggsey@gmail.com <mailto:giggsey@gmail.com>> wrote: When a transfer completes, FreeSWITCH will send an UPDATE message to the SIP server updating the caller id.
Is there a way of making FreeSWITCH send this message via a dialplan/command?
On 29 January 2015 at 17:58, Michael Collins <msc@freeswitch.org <mailto:msc@freeswitch.org>> wrote: Could you expound upon this question a bit? What does "update the CLI" mean? Thanks, MC
On Thu, Jan 29, 2015 at 1:20 AM, Joshua Gigg <giggsey@gmail.com <mailto:giggsey@gmail.com>> wrote: Hi,
Is it possible to update the CLI at will once a Freeswitch originated call has been answered?
I know it can update during a transfer, but I want to be able to control it directly myself.
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