i was working on IVR menus on FreePBX-17 using my ubuntu OS pc. and i created the IVR and when dialing to 100 extension i can hear the welcome voice and all records of options. but when i try to send a DMTF from zoiper of my mobile nothing is happing.
Have you checked Zoiper side other DTMF modes? If not pls check it.
Asterisk side by default DTMF is " RFC2833
i changed the dtmf to both freepbx and my mobile sip phone app to “SIP INFO”. but still i can here only the IVR menu. just for details this is how i did the IVR.
– i just uploaded multiple sounds to the system recording
– then i went to ivr menu and selected the recording then i added one entry , like when user clicks 1, agent with 1001 will rang.
am testing the freepbx using my phone in the laptop and connected to the FreePBX using 127.0.0.1 addr.
and also i have a phone with sip phones installed and connected it with my laptop throught the laptops hotspot and it has an ip of 10.42.0.1 , and i use this ip as a server to my mobile sip phone.
Have you checked Asterisk logs side when you are testing DTMF tone ?
yes yes, i show me some thing like this
-- <PJSIP/1003-00000024> Playing 'custom/welcome.slin' (language 'en')
-- <PJSIP/1003-00000024> Playing 'custom/01_Language.slin' (language 'en')
-- <PJSIP/1003-00000024> Playing 'custom/02_General_Service.slin' (language 'en')
-- <PJSIP/1003-00000024> Playing 'custom/03_Meet_Staff.slin' (language 'en')
Show us your IVR setup pls.
You have to see at your Asterisk logs side similer this info when you pressed any IVR Digits.
-- Executing [s@ivr-1:12] Read("PJSIP/1001-00000178", "IVREXT,,,,0,10") in new stack
> 0x7ff9680a6630 -- Strict RTP learning complete - Locking on source address 192.168.19.222:55090
-- User entered '3'
-- Executing [s@ivr-1:13] GotoIf("PJSIP/1001-00000176", "0?#,1") in new stack
-- Executing [s@ivr-1:14] GotoIf("PJSIP/1001-00000176", "0?t,1") in new stack
-- Executing [s@ivr-1:15] ExecIf("PJSIP/1001-00000176", "0?Set(LOCALEXT=1)") in new stack
-- Executing [s@ivr-1:16] GotoIf("PJSIP/1001-00000176", "0?i,1") in new stack
-- Executing [s@ivr-1:17] GotoIf("PJSIP/1001-00000176", "0?from-did-direct-ivr,3,1") in new stack
-- Executing [s@ivr-1:18] Goto("PJSIP/1001-00000176", "3,1") in new stack
-- Goto (ivr-1,3,1)
---
> 0x7ff950010a20 -- Strict RTP learning complete - Locking on source address 192.168.19.222:56899
-- User entered '2'
-- Executing [s@ivr-1:12] GotoIf("PJSIP/1001-00000175", "0?#,1") in new stack
-- Executing [s@ivr-1:13] GotoIf("PJSIP/1001-00000175", "0?t,1") in new stack
-- Executing [s@ivr-1:14] ExecIf("PJSIP/1001-00000175", "0?Set(LOCALEXT=1)") in new stack
-- Executing [s@ivr-1:15] GotoIf("PJSIP/1001-00000175", "0?i,1") in new stack
-- Executing [s@ivr-1:16] GotoIf("PJSIP/1001-00000175", "0?from-did-direct-ivr,2,1") in new stack
-- Executing [s@ivr-1:17] Goto("PJSIP/1001-00000175", "2,1") in new stack
-- Goto (ivr-1,2,1)
You had install FreePBX-17 to Ubuntu? If is that case you have to find other your issues by self.
I do have Debian-12 install FreePBX system.
Okay, How you are calling from Internal Extension to IVR?
oh, it worked now, the problem were free PBX didn’t know datas coming from my hotpots IP 10.42.0.1. so registered this IP in the asterisk sip setting ( NAT Settings). then it worked. thank for being with me and for supporting me
Well Done. Have a good day,