Hello,
I am a newbie and am attempting to set up a AWS hosted FreePBX instance - so far I have done some of the configuration which I will summarise below however I am not quite achieving the desired functionality.
Objective: User can dial in and is directed to an IVR, and then to a DISA, and are instructed to “dial 0” which will then trigger a series of prompts requesting them to input the number they are calling as well as the caller ID they wish to use for their outbound call - the outbound call is then made using the provided details.
Current state: When the call is made to the PBX, I am hearing the IVR announcement, am then connected to the DISA and hear the dial tone - after dialing “0” I hear the audio prompt to enter in Caller ID - which I do and the callerID is read back to me.
I hear the audio prompt to enter number to call - which I do and it is then read back to me
When it comes to connecting the call using Dial - I then hear an engaged tone.
custom_extensions.conf
[from-internal-custom]
include => cli-change-custom
[cli-change-custom]
exten => 0,1,Answer
exten => 0,2,Wait(2)
exten => 0,3,Playback(custom/3disaAnnouncement)
exten => 0,4(collect),Read(digito,,11)
exten => 0,5,Playback(custom/4disaConfirmingCID)
exten => 0,6,SayDigits(${digito})
exten => 0,7,Set(CALLER ID(number)=${digito})
exten => 0,8,Answer
exten => 0,9,Wait(2)
exten => 0,10,Playback(custom/5disaEnterOutbound)
exten => 0,11(collect),Read(digito1,,11)
exten => 0,12,Playback(custom/6disaConfirmingOutbound)
exten => 0,13,SayDigits(${digito1})
exten => 0,14,Playback(custom/7disaConnectingCall)
exten => 0,15,Dial(PJSIP/${digito1}@[*SIP provider address*],40)
I am unsure about that last line. Here is what my pjsip.endpoint.conf contains (assuming this is relavant):
[0]
#include pjsip.endpoint_custom.conf
[new_pjsiptrunk]
type=endpoint
transport=0.0.0.0-udp;
context=from-pstn
disallow=all
allow=ulaw,alaw,gsm,g726,g722,h264,mpeg4
aors=new_pjsiptrunk
send_connercted_line=no
rtp_keepalive=0
language=en_AU
outbound_auth=new_pjsiptrunk
trust_id_inbound=no
rtp_symmetric=yes
dtmf_mode=auto
And finally, the log - after I make a test call inbound, here is the output of var/log/asterisk/full:
Updated Pastebin Link
Edit:
I have just logged into Asterisk CLI and then attempted to call in again - When the call is made to the PBX, I am hearing the IVR announcement, am then connected to the DISA and hear the dial tone - after dialing “0” I hear the audio prompt to enter in Caller ID - which I do and the callerID is read back to me.
I hear the audio prompt to enter number to call - which I do and it is then read back to me
When it comes to connecting the call using Dial - I then hear an engaged tone.
This was the output in the Asterisk Console:
[2024-08-15 14:12:45] WARNING[164394][C-00000001]: app.c:3054 parse_options: Unrecognized option: 'e'
[2024-08-15 14:12:47] WARNING[164399][C-00000001]: channel.c:6247 request_channel: No channel type registered for 'CUSTOM'
[2024-08-15 14:12:47] NOTICE[164399][C-00000001]: app_dial.c:2703 dial_exec_full: Unable to create channel of type 'CUSTOM' (cause 66 - Channel not implemented)