Thanks for the answer.
How is the channel being created/initiated ?
As I told you, I’m really not an expert in Asterisk (this is a part of the pipeline I’m working on). So I might
I atttached the conf files I have access to, hoping they are relevant to answer the question:
extensions.conf (dialplan)
[global]
[default]
;Empty for security
[call-file-test]
exten => _X.,1,Verbose(1,${CONTEXT} - ${STRFTIME(${EPOCH},%C%y-%m-%d %H:%M:%S)} - ${CALLERID(ALL)} - ${EXTEN} - ${remoteid})
same => n,Wait(1)
;same=> n, Set(VOLUME(tx)=10)
same=> n, Monitor(wav,/tmp/${CDR(uniqueid)})
same => n,Playback(/opt/s3/audio/${audiofile})
same => n,Wait(4)
same => n,Hangup()
[callbackout]
exten => _123,1,Answer()
same => n,Set(__AUDIOFILENAME=${audiofile})
same => n,Set(__DBID=${dbid})
same => n,Set(__GRAMMAR=${grammar})
same => n,Set(__MRCP_OPTIONS=${mrcp_options})
;same => n,Set(GROUP()=ast1)
;same => n,Set(__COUNT_CHANNEL=${GROUP_COUNT(ast1)})
same => n,Dial(PJSIP/200@local,1,b(predialhandler^addheader^1))
[from-local]
exten => _200,1,Verbose(1,${CONTEXT} - ${STRFTIME(${EPOCH},%C%y-%m-%d %H:%M:%S)} - ${CALLERID(ALL)} - ${EXTEN} – ${remoteid})
;; Asterisk Manual
; asterisk-unimrcp/app-unimrcp/app_mrcprecog.c at master · unispeech/asterisk-unimrcp · GitHub
same => n,Set(__audiofilename=${PJSIP_HEADER(read,X-AudioName)})
same => n,Set(__dbid=${PJSIP_HEADER(read,X-DbId)})
same => n,Set(__grammar=${PJSIP_HEADER(read,X-Grammar)})
same => n,Set(__mrcp_options=${PJSIP_HEADER(read,X-mrcp-options)})
same => n,Set(GROUP()=ast1)
same => n,Set(__count_channel=${GROUP_COUNT(ast1)})
same => n,MRCPRecog(${grammar}, ${mrcp_options})
same => n,Hangup
exten => h,1,AGI(/opt/scripts/send_data.py)
[predialhandler]
exten => addheader,1,Set(PJSIP_HEADER(add,X-AudioName)=${AUDIOFILENAME})
same => n,Set(PJSIP_HEADER(add,X-DbId)=${DBID})
same => n,Set(PJSIP_HEADER(add,X-Grammar)=${GRAMMAR})
same => n,Set(PJSIP_HEADER(add,X-mrcp-options)=${MRCP_OPTIONS})
same => n,return()
mrcp.conf
[general]
; Default ASR and TTS profiles.
default-asr-profile = speech-nuance5-mrcp2
default-tts-profile = speech-nuance5-mrcp2
; UniMRCP logging level to appear in Asterisk logs. Options are:
; EMERGENCY|ALERT|CRITICAL|ERROR|WARNING|NOTICE|INFO|DEBUG →
log-level = DEBUG
max-connection-count = 100
max-shared-count = 100
offer-new-connection = 1
; rx-buffer-size = 1024
; tx-buffer-size = 1024
; request-timeout = 5000
; speech-channel-timeout = 30000
;
; Profile for Nuance Speech Server MRCPv1
;
[speech-nuance5-mrcp1]
; MRCP version.
version = 1
; === RTSP settings ===
; Must be set to the IP address of the MRCP server.
server-ip = XXX.XXX.XXX.XXX
; RTSP port on the MRCP server.
server-port = 4900
; force-destination = 1
resource-location = media
speechsynth = speechsynthesizer
speechrecog = speechrecognizer
; === RTP factory ===
; rtp-ip = 0.0.0.0
; Must be set to the IP address of the MRCP client.
rtp-ip = XXX.XXX.XXX.XXX
; rtp-ext-ip = auto
; RTP port range on the MRCP client.
rtp-port-min = 4000
rtp-port-max = 5000
; === Jitter buffer settings ===
playout-delay = 50
; min-playout-delay = 20
max-playout-delay = 200
; === RTP settings ===
ptime = 20
codecs = L16/96/8000 telephone-event/101/8000
; === RTCP settings ===
rtcp = 1
rtcp-bye = 2
rtcp-tx-interval = 5000
rtcp-rx-resolution = 1000
;
; Profile for Nuance Speech Server MRCPv2
;
[uni2]
; MRCP version.
version = 2
; === SIP settings ===
; Must be set to the IP address of the MRCP server.
; PREPROD
server-ip = XXX.XXX.XXX.XXX
; PROD
;server-ip = XXX.XXX.XXX.XXX
; SIP port on the MRCP server.
server-port = 8060
; server-username = test
force-destination = 0
; === SIP agent ===
client-ip = 0.0.0.0
; Must be set to the IP address of the MRCP client.
; client-ext-ip = auto
; SIP port on the MRCP client.
client-port = 5093
; SIP transport either UDP or TCP.
sip-transport = tcp
; ua-name = Asterisk
sdp-origin = stresstester
; sip-t1 = 500
; sip-t2 = 4000
; sip-t4 = 4000
; sip-t1x64 = 32000
; sip-timer-c = 185000
; === RTP factory ===
rtp-ip = 0.0.0.0
; Must be set to the IP address of the MRCP client.
; rtp-ext-ip = auto
; RTP port range on the MRCP client.
rtp-port-min = 4000
rtp-port-max = 5000
; === Jitter buffer settings ===
playout-delay = 50
; min-playout-delay = 20
max-playout-delay = 200
; === RTP settings ===
ptime = 20
codecs = L16/96/8000 telephone-event/101/8000
; === RTCP settings ===
rtcp = 1
rtcp-bye = 2
rtcp-tx-interval = 5000
rtcp-rx-resolution = 1000
pjsip.conf
; PJSIP Configuration
;[global]
[general]
bindport = 5060
[transport-udp]
type = transport
protocol = udp
bind = 0.0.0.0:5060
[local]
type = aor
contact = sip:127.0.0.1
qualify_frequency = 60
default_expiration = 1800
[local]
type = identify
endpoint = local
match = 127.0.0.1
[local]
type = endpoint
context = from-local
dtmf_mode = rfc4733
disallow = all
allow = alaw
direct_media = no
language = fr
aors = local