Asterisk-Unimrcp-Amazon Lex Bot Integration.

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fayaz patel

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Apr 29, 2020, 8:47:15 AM4/29/20

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Hi,

I am integrating Asterisk with Amazon Lex Bot using unimrcp. I have installed the required plugins.

My asterisk version is 16.6.1 and unimrcp version is 1.7.

I tried with an sample example as mentioned in https://www.unimrcp.org/manuals/html/AsteriskAWSUsageManual.html document.

Example:

MRCPSynth

Use the application MRCPSynth to speech synthesis.

[app-unimrcp-2]

exten => s,1,Answer

exten => s,2,MRCPSynth(Welcome to Asterisk,l=en-US&p=uni2)

In the Asterisk dialplan, associate the provided sample to an extension, for example, 802.

exten => 802,1,Goto(app-unimrcp-2,s,1)

Place a test call and listen to the synthesized message.

I am unable to listen the response output.

================================================\

Extensions.conf file:

[from-internal

exten => 802,1,Goto(app-unimrcp-2,s,1)

[app-unimrcp-2]

exten => s,1,Answer

exten => s,2,MRCPSynth(Welcome to Asterisk,l=en-US&p=uni2)

===========================================================

Asterisk CLI output:

CLI> -- Time to scan old dialplan and merge leftovers back into the new: 0.000050 sec

-- Time to restore hints and swap in new dialplan: 0.000006 sec

-- Time to delete the old dialplan: 0.000011 sec

-- Total time merge_contexts_delete: 0.000067 sec

-- pbx_config successfully loaded 4 contexts (enable debug for details).

== Setting global variable 'SIPDOMAIN' to '192.168.0.109'

-- Executing [802@from-internal:1] Goto("PJSIP/6001-00000001", "app-unimrcp-2,s,1") in new stack

-- Goto (app-unimrcp-2,s,1)

-- Executing [s@app-unimrcp-2:1] Answer("PJSIP/6001-00000001", "") in new stack

> 0x29c8630 -- Strict RTP learning after remote address set to: 192.168.0.104:8000

> 0x29c8630 -- Strict RTP switching to RTP target address 192.168.0.104:8000 as source

-- Executing [s@app-unimrcp-2:2] MRCPSynth("PJSIP/6001-00000001", "Welcome to Asterisk,l=en-US&p=uni2") in new stack

[Apr 29 04:39:08] NOTICE[7425][C-00000002]: app_mrcpsynth.c:514 app_synth_exec: MRCPSynth() prompt: Welcome to Asterisk

[Apr 29 04:39:08] NOTICE[7425][C-00000002]: app_mrcpsynth.c:529 app_synth_exec: MRCPSynth() options: l=en-US&p=uni2

[Apr 29 04:39:08] NOTICE[7425][C-00000002]: src/mrcp_application.c:117 Create MRCP Handle 0x7f84c8016b08 [uni2]

[Apr 29 04:39:08] NOTICE[7425][C-00000002]: src/mrcp_client_session.c:131 Create Channel TTS-0 <new>

[Apr 29 04:39:08] NOTICE[7387]: src/mrcp_client_session.c:385 Receive App Request TTS-0 <new> [2]

[Apr 29 04:39:08] NOTICE[7387]: src/mrcp_client.c:697 Add MRCP Handle TTS-0 <new>

[Apr 29 04:39:08] NOTICE[7387]: src/mrcp_client_session.c:717 Add Control Channel TTS-0 <new@speechsynth>

[Apr 29 04:39:08] NOTICE[7387]: src/mrcp_client_session.c:409 Send Offer TTS-0 <new> [c:1 a:1 v:0] to 192.168.0.106:8060

[Apr 29 04:39:08] NOTICE[7387]: src/mrcp_sofiasip_client_agent.c:357 Local SDP TTS-0 <new>

v=0

o=Asterisk 0 0 IN IP4 192.168.0.109

s=-

c=IN IP4 192.168.0.109

t=0 0

m=application 9 TCP/MRCPv2 1

a=setup:active

a=connection:new

a=resource:speechsynth

a=cmid:1

m=audio 28000 RTP/AVP 0 8 96 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:96 L16/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=recvonly

a=ptime:20

a=mid:1

[Apr 29 04:39:08] NOTICE[7392]: src/mrcp_sofiasip_client_agent.c:617 Receive SIP Event [nua_i_state] Status 0 INVITE sent [uni2]

[Apr 29 04:39:08] NOTICE[7392]: src/mrcp_sofiasip_client_agent.c:555 SIP Call State TTS-0 [calling]

[Apr 29 04:39:08] NOTICE[7392]: src/mrcp_sofiasip_client_agent.c:617 Receive SIP Event [nua_r_invite] Status 406 Not Acceptable [uni2]

[Apr 29 04:39:08] NOTICE[7392]: src/mrcp_sofiasip_client_agent.c:617 Receive SIP Event [nua_i_state] Status 406 Not Acceptable [uni2]

[Apr 29 04:39:08] NOTICE[7392]: src/mrcp_sofiasip_client_agent.c:555 SIP Call State TTS-0 [terminated]

[Apr 29 04:39:08] NOTICE[7387]: src/mrcp_client_session.c:149 Receive Answer TTS-0 <new> [c:0 a:0 v:0] Status 406

[Apr 29 04:39:08] NOTICE[7387]: src/mrcp_client_session.c:453 Raise App Response TTS-0 <new> [2] FAILURE [2]

[Apr 29 04:39:08] ERROR[7387]: app_mrcpsynth.c:208 speech_on_channel_add: (TTS-0) Channel error status=2, response code=406!

[Apr 29 04:39:08] NOTICE[7387]: src/mrcp_client_session.c:385 Receive App Request TTS-0 <new> [1]

[Apr 29 04:39:08] NOTICE[7387]: src/mrcp_client_session.c:828 Terminate Session TTS-0 <new>

[Apr 29 04:39:08] NOTICE[7387]: src/mrcp_client_session.c:207 Session Terminated TTS-0 <new>

[Apr 29 04:39:08] NOTICE[7387]: src/mrcp_client.c:707 Remove MRCP Handle TTS-0 <new>

[Apr 29 04:39:08] NOTICE[7387]: src/mrcp_client_session.c:453 Raise App Response TTS-0 <new> [1] SUCCESS [0]

[Apr 29 04:39:08] NOTICE[7387]: src/mrcp_application.c:211 Destroy MRCP Handle TTS-0

[Apr 29 04:39:08] NOTICE[7425][C-00000002]: app_mrcpsynth.c:479 mrcpsynth_exit: MRCPSynth() exiting status: ERROR on PJSIP/6001-00000001

-- Auto fallthrough, channel 'PJSIP/6001-00000001' status is 'UNKNOWN'

================================================================================

As i am new to this concept, can anyone help me to figure out what i am missing.?

Regards,

Fayaz Patel

fayaz patel

unread,

Apr 29, 2020, 8:47:16 AM4/29/20

to UniMRCP

Hi,

I want to integrate Asterisk with Amazon Lex Voice Bot using Unimrcp. I have installed asterisk, unimrcp and lex plugins.

I am using asterisk 16.6.1 and unimrcp version 1.7.

I tried to run basic example which is in https://www.unimrcp.org/manuals/html/AsteriskAWSUsageManual.html document.

Example is below:

Use the application MRCPSynth to speech synthesis.

[app-unimrcp-2]

exten => s,1,Answer

exten => s,2,MRCPSynth(Welcome to Asterisk,l=en-US&p=uni2)

In the Asterisk dialplan, associate the provided sample to an extension, for example, 802.

exten => 802,1,Goto(app-unimrcp-2,s,1)

As per document i am unable to hear the response output.

===================================================

My extensions.conf file is below:

[from-internal]

exten => 802,1,Goto(app-unimrcp-2,s,1)

[app-unimrcp-2]

exten => s,1,Answer

exten => s,2,MRCPSynth(Welcome to Asterisk,l=en-US&p=uni2)

===============================================================

Asterisk CLI output:

*CLI> -- Time to scan old dialplan and merge leftovers back into the new: 0.000050 sec

=============================================================================================================

As i am new to this concept, can anyone help me to figure it out what i am missing..?

Regards,

Fayaz Patel

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