No application 'Dial' for extension

Hello, everyone!
I hope to everything is very well.
I had a problem. I am going to implement IVR from FreePBX to other phone.
For example, I’m trying to implement FreePBX to automatically call a skype phone and then answer with an IVR.
So, I set the Trunk , Outbound Routes below as:


First, I tried to test with MicroSIP. I mean - MicroSIP->skype phone.
But I got the following error.
image

Please help me.
Thanks,

On Thursday 24 October 2024 at 15:40:44, Panda via Asterisk Community wrote:

I am going to implement IVR from FreePBX

You’re probably far better off asking this sort of question at
https://community.freepbx.org since they know how the FreePBX dialplan works,
and how you can be creative with it whereas here we deal only with Asterisk
itself.

Antony.


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Please provide logs as plain text.

You haven’t installed Asterisk correctly for FreePBX. If there is no app_dial, your modules.conf is non-standard and broken for most normal uses of Asterisk.

Hello, @david551 Thanks for your message.
This is logs.

[2024-10-25 15:00:08] NOTICE[583471] res_pjsip/pjsip_distributor.c: Request 'REGISTER' from '"7700" <sip:7700@4.242.27.76>' failed for '62.210.113.73:5083' (callid: 1750957905) - No matching endpoint found after 76 tries in 1.943 ms
[2024-10-25 15:00:08] NOTICE[583471] res_pjsip/pjsip_distributor.c: Request 'REGISTER' from '"7700" <sip:7700@4.242.27.76>' failed for '62.210.113.73:5083' (callid: 1750957905) - Failed to authenticate
[2024-10-25 15:00:08] NOTICE[504160] res_pjsip/pjsip_distributor.c: Request 'REGISTER' from '"7700" <sip:7700@4.242.27.76>' failed for '62.210.113.73:5083' (callid: 1632912763) - No matching endpoint found after 77 tries in 1.978 ms
[2024-10-25 15:00:08] NOTICE[504160] res_pjsip/pjsip_distributor.c: Request 'REGISTER' from '"7700" <sip:7700@4.242.27.76>' failed for '62.210.113.73:5083' (callid: 1632912763) - Failed to authenticate
[2024-10-25 15:00:22] ERROR[583471] res_pjsip.c: Unable to apply outbound proxy on request OPTIONS to endpoint sipvoice_trunk as outbound proxy URI 'phxpripbx.sipvoice.com' is not valid
[2024-10-25 15:00:22] ERROR[583471] res_pjsip/pjsip_options.c: Unable to create request to qualify contact sip:208.77.63.131:5060 on AOR sipvoice_trunk
[2024-10-25 15:01:22] ERROR[504160] res_pjsip.c: Unable to apply outbound proxy on request OPTIONS to endpoint sipvoice_trunk as outbound proxy URI 'phxpripbx.sipvoice.com' is not valid
[2024-10-25 15:01:22] ERROR[504160] res_pjsip/pjsip_options.c: Unable to create request to qualify contact sip:208.77.63.131:5060 on AOR sipvoice_trunk
[2024-10-25 15:02:22] ERROR[504160] res_pjsip.c: Unable to apply outbound proxy on request OPTIONS to endpoint sipvoice_trunk as outbound proxy URI 'phxpripbx.sipvoice.com' is not valid
[2024-10-25 15:02:22] ERROR[504160] res_pjsip/pjsip_options.c: Unable to create request to qualify contact sip:208.77.63.131:5060 on AOR sipvoice_trunk

Specifically, it must be as it would be used in a Route header, so should normally include the lr parameter and it may also need a hide parameter, to stop the Route headers actually being generated. The ; needs to be escaped.

Note this doesn’t address the previously reported failure to load the Dial application, or the new report of the failure to identify the registrant.

It also doesn’t invalidate the point that people on this forum cannot be expected to give advice on using the FreePBX GUI, or interpret screenshots, from it.

Thanks for your help. @david551
This is my final plan.
I want to set up FreePBX to make a call to someone else’s phone.
Then I want to play an existing voice file with IVR.
Is this possible?
If so, how can I do it?
I think I will do this using AMI.
Could you explain it in detail, please?

Hello, @david551 !
I connected MicroSIP to FreePBX.
But Below errors occured.

Please use the FreePBX forum.

There is no app_macro in the latest version of Asterisk, and recent earlier versions do not build/load it, without explicit configuration. The latest version of Asterisk requires the latest version of FreePBX.

When providing log, provide them as plain text.

Thank you very much and sorry. David.