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What steps will reproduce the problem?
1. Run MRCPRecog using i=any
2. Press a dtmf key
What version of the product are you using? On what operating system?
UniMRCP Modules for Asterisk 1.1.0
Asterisk 10.7.0
Below is output using builtin:dtmf/number , but the problem is with any grammar
and also with and without telephone-event set in codecs. Also with any value of
b
The error is produced as soon as a dtmf key is pressed, giving ERROR[5338]
app_mrcprecog.c: (ASR-45) Recognition terminated prematurely
[Apr 14 11:05:12] NOTICE[5338] app_mrcprecog.c: MRCPRecog() grammar:
builtin:dtmf/number
[Apr 14 11:05:12] NOTICE[5338] app_mrcprecog.c: MRCPRecog() options:
i=any,t=5000&f=beep&b=1
[Apr 14 11:05:12] NOTICE[5338] app_unimrcp.c: Create MRCP Handle 0xe146a88
[telisma-mrcp2]
[Apr 14 11:05:12] NOTICE[3414] app_unimrcp.c: Add Control Channel ASR-45
<new@speechrecog>
[Apr 14 11:05:12] NOTICE[3418] app_unimrcp.c: SIP Call State ASR-45 [calling]
[Apr 14 11:05:12] NOTICE[3418] app_unimrcp.c: SIP Call State ASR-45 [ready]
[Apr 14 11:05:12] NOTICE[3415] app_unimrcp.c: Established TCP/MRCPv2 Connection
127.0.0.1:40821 <-> 127.0.0.1:1544
[Apr 14 11:05:12] NOTICE[3414] app_mrcprecog.c: (ASR-45) Channel ready
codec=PCMU, sample rate=8000
[Apr 14 11:05:12] WARNING[5338] format_pcm.c: offset too large 3406, truncating
to 3405
[Apr 14 11:05:12] NOTICE[5338] app_mrcprecog.c: Stream file on
SIP/jase-00000012 length:3405
[Apr 14 11:05:12] NOTICE[5338] app_mrcprecog.c: (ASR-45) Recognizing, enable
DTMFs: 1, start input timers: 0
[Apr 14 11:05:13] ERROR[5338] app_mrcprecog.c: (ASR-45) Recognition terminated
prematurely
[Apr 14 11:05:13] WARNING[5338] app_mrcprecog.c: (ASR-45) Unable to retrieve
result
[Apr 14 11:05:13] NOTICE[3418] app_unimrcp.c: SIP Call State ASR-45 [terminated]
[Apr 14 11:05:13] NOTICE[3414] app_unimrcp.c: Destroy TCP/MRCPv2 Connection
127.0.0.1:40821 <-> 127.0.0.1:1544
[Apr 14 11:05:13] NOTICE[3414] app_unimrcp.c: Destroy MRCP Handle ASR-45
[Apr 14 11:05:13] NOTICE[5338] app_mrcprecog.c: MRCPRecog() exiting status:
ERROR on SIP/jase-00000012
Original issue reported on code.google.com by [email protected] on 14 Apr 2014 at 12:45
The text was updated successfully, but these errors were encountered:
Support for the option i=any seems to be broken. However, everything should
work normally in case DTMFs are transparently sent to the server to be matched
against active grammars. This is the default and recommended way. So, if your
server supports either in-band or out-of-band DTMFs, then just omit the option
i.
Original issue reported on code.google.com by
[email protected]
on 14 Apr 2014 at 12:45The text was updated successfully, but these errors were encountered: