Hi and thank you for responding!
Based upon the above I’ve done some research and am posting below what I tried in hopes someone can point out where I’m going wrong.
I verified PJSIP has:
dtmf_mode=rfc4733
;dtmf_mode=rfc2833
Just to test, I tried each individually and then just 2833 without any DTMF detected.
Next I (hopefully correctly interpreted and) followed the above instructions:
*CLI> logger show channels
Logger queue limit: 1000
Channel Type Formatter Status Configuration
------- ---- --------- ------ -------------
/var/log/asterisk/messages File default Enabled - NOTICE WARNING ERROR
Console default Enabled - NOTICE WARNING ERROR VERBOSE
# update logger.conf
console => notice,warning,error <--from
console => notice,warning,error,dtmf,debug <--to
*CLI> logger reload
*CLI> logger show channels
Logger queue limit: 1000
Channel Type Formatter Status Configuration
------- ---- --------- ------ -------------
/var/log/asterisk/messages File default Enabled - NOTICE WARNING ERROR
Console default Enabled - DEBUG NOTICE WARNING ERROR VERBOSE DTMF
A test call displayed the following (which is what I am hoping shows I correctly set-up DTMF debugging):
[Aug 23 08:34:55] DTMF[234643][C-0000004c]: channel.c:3980 __ast_read: DTMF begin '5' received on PJSIP/flowroute
[Aug 23 08:34:55] DTMF[234643][C-0000004c]: channel.c:3984 __ast_read: DTMF begin ignored '5' on PJSIP/flowroute
[Aug 23 08:34:56] DTMF[234643][C-0000004c]: channel.c:3866 __ast_read: DTMF end '5' received on PJSIP/flowroute, duration 130 ms
[Aug 23 08:34:56] DTMF[234643][C-0000004c]: channel.c:3955 __ast_read: DTMF end passthrough '5' on PJSIP/flowroute
Linphone–no dtmf displayed
Here is the console out for the session:
Please note I changed numbers for this post (I don’t use ext 1000 or 7045551234).
-- Executing [xxxx@my-phone:1] VoiceMailMain("PJSIP/1000-0000006e", "") in new stack
> 0x7f4bb80529f0 -- Strict RTP learning after remote address set to: 192.168.1.x:xxxx
-- <PJSIP/1000-0000006e> Playing 'vm-login.ulaw' (language 'en')
> 0x7f4bb80529f0 -- Strict RTP switching to RTP target address 192.168.1.x:xxxx as source
> 0x7f4bb80529f0 -- Strict RTP learning complete - Locking on source address 192.168.1.x:xxxx
-- <PJSIP/1000-0000006e> Playing 'vm-password.ulaw' (language 'en')
-- Incorrect password '' for user '7045551234' (context = default)
-- <PJSIP/1000-0000006e> Playing 'vm-incorrect-mailbox.ulaw' (language 'en')
[Aug 23 08:40:55] WARNING[234656][C-0000004d]: app_voicemail.c:11235 vm_authenticate: Couldn't read username
I think the “Incorrect password for user 7045551234” also indicates no DTMF detected because I believe DTMF would cause the number to change from the default to any detected DTMF–but that’s just my guess.
I also did some client-research. Here’s what I found going through my Linphone settings:
Settings --> Audio
Software echo cancellation - on
Adaptive rate control - on
bitrate limit 36 kbits/s
and a list of codecs:
opus 48kHz - on
speex 16kHz - off
speex 8kHz - off
PCMU 8kHz - on
PCMA 8kHz - on
GSM 8kHz - on
G722 8kHz - on
iLBC 8kHz - off
G729 8kHz - on
iSAC 16kHz - off
speex 32kHz - off
L16 44.1kHz - off
L16 44.1kHz - off
Settings --> Call
encryption - all off, set to none
Send out-band DTMFs (SIP INFO)
Send in-band DTMFs (RFC 2833)
With regards to the above 2 settings I have tried:
Send out-band DTMFs (SIP INFO) - off
Send in-band DTMFs (RFC 2833) - off
Send out-band DTMFs (SIP INFO) - off
Send in-band DTMFs (RFC 2833) - on
Send out-band DTMFs (SIP INFO) - on
Send in-band DTMFs (RFC 2833) - off
Send out-band DTMFs (SIP INFO) - on
Send in-band DTMFs (RFC 2833) - on
No DTMF messages and the following line each attempt:
WARNING[234688][C-00000051]: app_voicemail.c:11235 vm_authenticate: Couldn't read username
Thank you very much for your help!