Is it possible?

Hello, everyone. I am a beginner of FreePBX.
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?

Yes, almost everything is possible.
Read the documentations here.

https://sangomakb.atlassian.net/wiki/spaces/PG/pages/13172873/PBX+GUI+Home

https://sangomakb.atlassian.net/wiki/spaces/PG/pages/23953486/IVR+Module

Thanks for your support. I have already set Trunk, Outbound Routes, IVR.
I’m trying to call my skype phone via MicroSIP and it’s decliend.
why and how to fix it?

Now I am using Asterisk 20.10, FreePBX 16.0.

Need further details.
MicroSIP is used as a simple softphone for calls?
The calls don’t work only for Skype calls or any calls?

Anyway, you need to check Asterisk logs and try to check if everything is registered and up.
If you have an issue with the registration, then chcek the Firewall and allow the Responsive Firewlal is enable OR if your IPs addresses are allowed in the trusted zone.
Also, maybe the IP address has been banned.

If the outbound route is correct, and if your SIP provider allows the good CID number, there is no reason to break calls.

This is log of asterisk when I calling.

[2024-10-28 09:41:53] VERBOSE[2174046] res_pjsip/pjsip_configuration.c: Endpoint 0001 is now Unreachable
[2024-10-28 09:41:54] VERBOSE[2168304] res_pjsip_registrar.c: Added contact 'sip:[email protected]:17118;ob;x-ast-orig-host=192.168.104.171:52536' to AOR '0001' with expiration of 300 seconds
[2024-10-28 09:41:54] VERBOSE[2168304] res_pjsip/pjsip_configuration.c: Endpoint 0001 is now Reachable
[2024-10-28 09:41:54] VERBOSE[2168304] res_pjsip/pjsip_options.c: Contact 0001/sip:[email protected]:17118;ob;x-ast-orig-host=192.168.104.171:52536 is now Reachable.  RTT: 259.396 msec
[2024-10-28 09:42:20] ERROR[2168304] 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
[2024-10-28 09:42:20] ERROR[2168304] res_pjsip/pjsip_options.c: Unable to create request to qualify contact sip:208.77.63.131:5060 on AOR sipvoice_trunk
[2024-10-28 09:42:57] VERBOSE[2168304] res_pjsip/pjsip_configuration.c: Endpoint 0001 is now Unreachable
[2024-10-28 09:42:57] VERBOSE[2168304] res_pjsip/pjsip_options.c: Contact 0001/sip:[email protected]:17118;ob;x-ast-orig-host=192.168.104.171:52536 is now Unreachable.  RTT: 0.000 msec
[2024-10-28 09:43:20] ERROR[2168304] 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
[2024-10-28 09:43:20] ERROR[2168304] res_pjsip/pjsip_options.c: Unable to create request to qualify contact sip:208.77.63.131:5060 on AOR sipvoice_trunk
[2024-10-28 09:44:12] VERBOSE[2168304] netsock2.c: Using SIP RTP Audio TOS bits 184
[2024-10-28 09:44:12] VERBOSE[2168304] netsock2.c: Using SIP RTP Audio TOS bits 184 in TCLASS field.
[2024-10-28 09:44:12] VERBOSE[2168304] netsock2.c: Using SIP RTP Audio CoS mark 5
[2024-10-28 09:44:12] WARNING[2174361][C-00000006] pbx.c: No application 'Macro' for extension (from-internal, +15032183424, 1)
[2024-10-28 09:44:12] VERBOSE[2174361][C-00000006] pbx.c: Spawn extension (from-internal, +15032183424, 1) exited non-zero on 'PJSIP/0001-00000005'
[2024-10-28 09:44:12] WARNING[2174361][C-00000006] pbx.c: No application 'Macro' for extension (from-internal, h, 1)
[2024-10-28 09:44:12] VERBOSE[2174361][C-00000006] pbx.c: Spawn extension (from-internal, h, 1) exited non-zero on 'PJSIP/0001-00000005'

and then firewall status is inactive.

Looking at logs.

Unable to apply outbound proxy on request OPTIONS to endpoint sipvoice_trunk as outbound proxy URI 'phxpripbx.sipvoice.com' is not valid

Your trunk is not correctly configured.

You didn’t compile Asterisk with app-macro or you are using the wrong version of FreePBX

But I can calling from my skype phone to freepbx.

Incoming calls don’t use the proxy setting.

Thanks. everyone.
And then how to fix it?

[2024-10-28 09:44:12] WARNING[2174361][C-00000006] pbx.c: No application 'Macro' for extension (from-internal, +15032183424, 1)

How did you install asterisk?

I installed asterisk using compile the source.

Then in the make menuselect bit check app-macro

Is this right?

But I got the same error.

Which error? You have two unrelated errors (at least) in the logs.

I mean

[2024-10-28 09:44:12] WARNING[2174361][C-00000006] pbx.c: No application 'Macro' for extension (from-internal, +15032183424, 1)

You’re following a prescription for advanced users. May I suggest you start over with a new FreePBX 17 install using the install script published by the project for Debian.

1 Like

Hello, Thanks.
I am trying to install FreePBX 17.
I already installed PHP 8.2. and then I got error when “sudo ./install”.

PHP Fatal error:  Uncaught Error: Call to undefined function FreePBX\Install\simplexml_load_file() in /usr/local/src/freepbx/install:19
Stack trace:
#0 {main}
  thrown in /usr/local/src/freepbx/install on line 19

How to fix that?

Start over, carefully read the documentation twice, it is unlikely that it only doesn’t work just for you.