Re: [asterisk-users] Asterisk not working with Festival
Hi,
I'm experiencing the same problem, with identical symptoms.
I also noticed that after making a call attempt, I see this stuck TCP
connection pair until I stop and restart the asterisk server process.
# netstat -an | grep 1314
tcp0 0 0.0.0.0:13140.0.0.0:*
LISTEN
tcp 46 0 127.0.0.1:52206 127.0.0.1:1314
CLOSE_WAIT
tcp0 0 127.0.0.1:1314 127.0.0.1:52206
FIN_WAIT2
Mark
On Thu, Aug 12, 2010 at 02:41:50PM +0530, Davinder Kumar Meen wrote:
I tried it but I still cannot hear any sound created from Festival()
function. I can hear only a voice saying one which was working earlier
as well. Here is log of asterisk console:
-- Attempting call on SIP/011xx...@gafachi1a for
s...@connect-to-me:1 (Retry 1)
-- Executing [...@connect-to-me:1] Answer(SIP/gafachi1a-,
) in new stack
-- Executing [...@connect-to-me:2] Wait(SIP/gafachi1a-,
7) in new stack
-- Executing [...@connect-to-me:3]
SayDigits(SIP/gafachi1a-, '1') in new stack
-- SIP/gafachi1a- Playing 'digits/1.slin' (language 'en')
-- Executing [...@connect-to-me:4] Festival(SIP/gafachi1a-,
hello john) in new stack
== Parsing '/usr/local/etc/asterisk/festival.conf': == Found
On 11/08/10 11:22 PM, Danny Nicholas da...@debsinc.com wrote:
From: asterisk-users-boun...@lists.digium.com
[[1]mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Davinder Kumar Meen
Subject: Re: [asterisk-users] Asterisk not working with Festival
Can anyone help please on this?
snip
[connect-to-me]
exten = s,1,Answer
Exten = s,n,SayDigits(`1')
exten = s,n,Festival(hello john)
exten = s,n,Hangup
snip
When you call in from your mobile, you are using a DAHDI channel
which introduces a 3-7 second delay into the process, unless you
have one of the blessed phone companies that offers call
supervision. If you put a wait(7) in front of SayDigits, you should
hear the call normally.
This is what I would suggest
[connect-to-me]
exten = s,1,Answer
Exten = s,n,Gotoif($[${EXTEN}:0:3) = SIP]?4:3
Exten = s,n,wait(7)
Exten = s,n,SayDigits(`1')
exten = s,n,Festival(hello john)
exten = s,n,Hangup
Thanks,
Davinder Kumar Meen
Partner Project Manager
Impinge Solutions, F-250, Phase 8B, Mohali (India)
www.impingesolutions.com
We also provide server hosting services. Please checkout our website
www.goforspace.com
References
1. mailto:asterisk-users-boun...@lists.digium.com]
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Mark G. Thomas (m...@misty.com)
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Tel: +1-215-512-0112 US: 877-512-0112
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Re: [asterisk-users] Asterisk not working with Festival
I tried it but I still cannot hear any sound created from Festival()
function. I can hear only a voice saying one which was working earlier as
well. Here is log of asterisk console:
-- Attempting call on SIP/011xx...@gafachi1a for
s...@connect-to-me:1 (Retry 1)
-- Executing [...@connect-to-me:1] Answer(SIP/gafachi1a-, ) in
new stack
-- Executing [...@connect-to-me:2] Wait(SIP/gafachi1a-, 7) in
new stack
-- Executing [...@connect-to-me:3] SayDigits(SIP/gafachi1a-,
'1') in new stack
-- SIP/gafachi1a- Playing 'digits/1.slin' (language 'en')
-- Executing [...@connect-to-me:4] Festival(SIP/gafachi1a-,
hello john) in new stack
== Parsing '/usr/local/etc/asterisk/festival.conf': == Found
On 11/08/10 11:22 PM, Danny Nicholas da...@debsinc.com wrote:
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Davinder Kumar
Meen
Subject: Re: [asterisk-users] Asterisk not working with Festival
Can anyone help please on this?
snip
[connect-to-me]
exten = s,1,Answer
Exten = s,n,SayDigits(1¹)
exten = s,n,Festival(hello john)
exten = s,n,Hangup
snip
When you call in from your mobile, you are using a DAHDI channel which
introduces a 3-7 second delay into the process, unless you have one of the
³blessed² phone companies that offers call supervision. If you put a wait(7)
in front of SayDigits, you should hear the call ³normally².
This is what I would suggest
[connect-to-me]
exten = s,1,Answer
Exten = s,n,Gotoif($[³${EXTEN}:0:3)² = ³SIP²]?4:3
Exten = s,n,wait(7)
Exten = s,n,SayDigits(1¹)
exten = s,n,Festival(hello john)
exten = s,n,Hangup
Thanks,
Davinder Kumar Meen
Partner Project Manager
Impinge Solutions, F-250, Phase 8B, Mohali (India)
www.impingesolutions.com
We also provide server hosting services. Please checkout our website
www.goforspace.com
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
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Re: [asterisk-users] Asterisk not working with Festival
) on
'407cd1eb69781303206dd7d43ca34...@202.164.37.235' Request 104: Found
[Aug 12 14:50:01] DEBUG[7342]: chan_sip.c:4098 __sip_ack: Stopping
retransmission on '407cd1eb69781303206dd7d43ca34...@202.164.37.235' of
Request 104: Match Found
[Aug 12 14:50:01] DEBUG[7342]: chan_sip.c:3273 initialize_initreq:
Initializing already initialized SIP dialog
407cd1eb69781303206dd7d43ca34...@202.164.37.235 (presumably reinvite)
[Aug 12 14:50:01] DEBUG[7342]: chan_sip.c:4139 __sip_semi_ack: (Provisional)
Stopping retransmission (but retaining packet) on
'35cba4547c9dca423fd4c80401085...@202.164.37.235' Request 104: Found
[Aug 12 14:50:01] DEBUG[7342]: chan_sip.c:4098 __sip_ack: Stopping
retransmission on '35cba4547c9dca423fd4c80401085...@202.164.37.235' of
Request 104: Match Found
[Aug 12 14:50:01] DEBUG[7342]: chan_sip.c:3273 initialize_initreq:
Initializing already initialized SIP dialog
35cba4547c9dca423fd4c80401085...@202.164.37.235 (presumably reinvite)
[Aug 12 14:50:02] DEBUG[7342]: chan_sip.c:4139 __sip_semi_ack: (Provisional)
Stopping retransmission (but retaining packet) on
'407cd1eb69781303206dd7d43ca34...@202.164.37.235' Request 105: Found
[Aug 12 14:50:02] DEBUG[7342]: chan_sip.c:4098 __sip_ack: Stopping
retransmission on '407cd1eb69781303206dd7d43ca34...@202.164.37.235' of
Request 105: Match Found
[Aug 12 14:50:02] DEBUG[7342]: chan_sip.c:18218 handle_response_register:
Registration successful
[Aug 12 14:50:02] DEBUG[7342]: chan_sip.c:18220 handle_response_register:
Cancelling timeout 19
[Aug 12 14:50:02] DEBUG[7342]: chan_sip.c:4139 __sip_semi_ack: (Provisional)
Stopping retransmission (but retaining packet) on
'35cba4547c9dca423fd4c80401085...@202.164.37.235' Request 105: Found
[Aug 12 14:50:02] DEBUG[7342]: chan_sip.c:4098 __sip_ack: Stopping
retransmission on '35cba4547c9dca423fd4c80401085...@202.164.37.235' of
Request 105: Match Found
[Aug 12 14:50:02] DEBUG[7342]: chan_sip.c:18218 handle_response_register:
Registration successful
And there is no log entry related to call terminated by remote.
On 11/08/10 11:22 PM, Danny Nicholas da...@debsinc.com wrote:
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Davinder Kumar
Meen
Subject: Re: [asterisk-users] Asterisk not working with Festival
Can anyone help please on this?
snip
[connect-to-me]
exten = s,1,Answer
Exten = s,n,SayDigits(1¹)
exten = s,n,Festival(hello john)
exten = s,n,Hangup
snip
When you call in from your mobile, you are using a DAHDI channel which
introduces a 3-7 second delay into the process, unless you have one of the
³blessed² phone companies that offers call supervision. If you put a wait(7)
in front of SayDigits, you should hear the call ³normally².
This is what I would suggest
[connect-to-me]
exten = s,1,Answer
Exten = s,n,Gotoif($[³${EXTEN}:0:3)² = ³SIP²]?4:3
Exten = s,n,wait(7)
Exten = s,n,SayDigits(1¹)
exten = s,n,Festival(hello john)
exten = s,n,Hangup
Thanks,
Davinder Kumar Meen
Partner Project Manager
Impinge Solutions, F-250, Phase 8B, Mohali (India)
www.impingesolutions.com
We also provide server hosting services. Please checkout our website
www.goforspace.com
--
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Re: [asterisk-users] Asterisk not working with Festival
Can anyone help please on this? I tried same configuration on CentOS as well and got same result i.e. No sound and hangup. On 04/08/10 5:58 PM, Davinder Kumar Meen davin...@impingeonline.com wrote: Hello, I am having a Mac 10.6.4 (Snow Leopard). I have compiled and built Asterisk 1.6.2.9 and Festival 2.0.95:beta on my machine. Asterisk is working fine with SIP channels without Festival. I have written following context in extension.conf: [connect-to-me] exten = s,1,Answer Exten = s,n,SayDigits(1¹) exten = s,n,Festival(hello john) exten = s,n,Hangup I use call files to make calls to my mobile and once call is answered then asterisk attaches it to ³connect-to-me² context. But after that, I can hear only a voice saying ³one² but nothing after that. Please find below details on configuration files: festival.conf: ; Festival Configuration [general] host=localhost port=1314 usecache=yes cachedir=/var/lib/asterisk/festivalcache/ festivalcommand=(tts_textasterisk %s 'file)(quit)\n And, festival.scm : (define (tts_textasterisk string mode) (tts_textasterisk STRING MODE) Apply tts to STRING. This function is specifically designed for use in server mode so a single function call may synthesize the string. This function name may be added to the server safe functions. (let ((wholeutt (utt.synth (eval (list 'Utterance 'Text string) (utt.wave.resample wholeutt 8000) (utt.wave.rescale wholeutt 5) (utt.send.wave.client wholeutt))) I have placed the above text before the last line which is (provide 'festival). Below is the debug log shown on asterisk console : [Aug 4 17:50:11] Channel SIP/gafachi1a- was answered. [Aug 4 17:50:11] DEBUG[17094]: pbx.c:3692 pbx_extension_helper: Launching 'Answer' [Aug 4 17:50:11] -- Executing [...@connect-to-me:1] Answer(SIP/gafachi1a-, ) in new stack [Aug 4 17:50:11] DEBUG[17094]: pbx.c:3692 pbx_extension_helper: Launching 'SayDigits' [Aug 4 17:50:11] -- Executing [...@connect-to-me:2] SayDigits(SIP/gafachi1a-, '1') in new stack [Aug 4 17:50:11] DEBUG[17094]: channel.c:3881 set_format: Set channel SIP/gafachi1a- to write format slin [Aug 4 17:50:11] DEBUG[17094]: rtp.c:3878 ast_rtp_write: Ooh, format changed from unknown to ulaw [Aug 4 17:50:11] DEBUG[17094]: rtp.c:3904 ast_rtp_write: Created smoother: format: 4 ms: 20 len: 160 [Aug 4 17:50:11] DEBUG[17094]: channel.c:2488 ast_settimeout: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Aug 4 17:50:11] -- SIP/gafachi1a- Playing 'digits/1.slin' (language 'en') [Aug 4 17:50:12] DEBUG[17094]: channel.c:2488 ast_settimeout: Scheduling timer at (571 requested / 100 actual) timer ticks per second [Aug 4 17:50:12] DEBUG[17094]: channel.c:2488 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 4 17:50:12] DEBUG[17094]: channel.c:2488 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 4 17:50:12] DEBUG[17094]: channel.c:2488 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 4 17:50:12] DEBUG[17094]: channel.c:3881 set_format: Set channel SIP/gafachi1a- to write format ulaw [Aug 4 17:50:12] DEBUG[17094]: pbx.c:3692 pbx_extension_helper: Launching 'Festival' [Aug 4 17:50:12] -- Executing [...@connect-to-me:3] Festival(SIP/gafachi1a-, hello john) in new stack [Aug 4 17:50:12] == Parsing '/usr/local/etc/asterisk/festival.conf': [Aug 4 17:50:12] DEBUG[17094]: config.c:1330 config_text_file_load: Parsing /usr/local/etc/asterisk/festival.conf [Aug 4 17:50:12] == Found [Aug 4 17:50:12] DEBUG[17094]: app_festival.c:376 festival_exec: Text passed to festival server : hello john [Aug 4 17:50:12] DEBUG[17094]: app_festival.c:446 festival_exec: Cache file exists, strln=10, strlen=10 [Aug 4 17:50:12] DEBUG[17094]: app_festival.c:448 festival_exec: Size OK [Aug 4 17:50:12] DEBUG[17094]: app_festival.c:467 festival_exec: Reading from cache... [Aug 4 17:50:12] DEBUG[17094]: app_festival.c:491 festival_exec: Passing data to channel... [Aug 4 17:50:12] DEBUG[17094]: app_festival.c:513 festival_exec: Festival WV command [Aug 4 17:50:12] DEBUG[17094]: channel.c:3881 set_format: Set channel SIP/gafachi1a- to write format slin [Aug 4 17:50:34] DEBUG[17094]: chan_sip.c:3562 __sip_xmit: Trying to put 'SIP/2.0 200' onto UDP socket destined for 67.216.35.162:5060 And, festival server console looks like following: $ ./bin/festival --server serverWed Aug 4 17:49:04 2010 : Festival server started on port 1314 client(1) Wed Aug 4 17:50:12 2010 : accepted from localhost client(1) Wed Aug 4 17:50:12 2010 : disconnected I have to end the call after sometime. Festival works fine if I got into its console and type SayText(³hello john²) Please let me know how I can fix this. Thanks,
Re: [asterisk-users] Asterisk not working with Festival
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Davinder Kumar
Meen
Subject: Re: [asterisk-users] Asterisk not working with Festival
Can anyone help please on this?
snip
[connect-to-me]
exten = s,1,Answer
Exten = s,n,SayDigits('1')
exten = s,n,Festival(hello john)
exten = s,n,Hangup
snip
When you call in from your mobile, you are using a DAHDI channel which
introduces a 3-7 second delay into the process, unless you have one of the
blessed phone companies that offers call supervision. If you put a
wait(7) in front of SayDigits, you should hear the call normally.
This is what I would suggest
[connect-to-me]
exten = s,1,Answer
Exten = s,n,Gotoif($[${EXTEN}:0:3) = SIP]?4:3
Exten = s,n,wait(7)
Exten = s,n,SayDigits('1')
exten = s,n,Festival(hello john)
exten = s,n,Hangup
--
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[asterisk-users] Asterisk not working with Festival
Hello, I am having a Mac 10.6.4 (Snow Leopard). I have compiled and built Asterisk 1.6.2.9 and Festival 2.0.95:beta on my machine. Asterisk is working fine with SIP channels without Festival. I have written following context in extension.conf: [connect-to-me] exten = s,1,Answer Exten = s,n,SayDigits(1¹) exten = s,n,Festival(hello john) exten = s,n,Hangup I use call files to make calls to my mobile and once call is answered then asterisk attaches it to ³connect-to-me² context. But after that, I can hear only a voice saying ³one² but nothing after that. Please find below details on configuration files: festival.conf: ; Festival Configuration [general] host=localhost port=1314 usecache=yes cachedir=/var/lib/asterisk/festivalcache/ festivalcommand=(tts_textasterisk %s 'file)(quit)\n And, festival.scm : (define (tts_textasterisk string mode) (tts_textasterisk STRING MODE) Apply tts to STRING. This function is specifically designed for use in server mode so a single function call may synthesize the string. This function name may be added to the server safe functions. (let ((wholeutt (utt.synth (eval (list 'Utterance 'Text string) (utt.wave.resample wholeutt 8000) (utt.wave.rescale wholeutt 5) (utt.send.wave.client wholeutt))) I have placed the above text before the last line which is (provide 'festival). Below is the debug log shown on asterisk console : [Aug 4 17:50:11] Channel SIP/gafachi1a- was answered. [Aug 4 17:50:11] DEBUG[17094]: pbx.c:3692 pbx_extension_helper: Launching 'Answer' [Aug 4 17:50:11] -- Executing [...@connect-to-me:1] Answer(SIP/gafachi1a-, ) in new stack [Aug 4 17:50:11] DEBUG[17094]: pbx.c:3692 pbx_extension_helper: Launching 'SayDigits' [Aug 4 17:50:11] -- Executing [...@connect-to-me:2] SayDigits(SIP/gafachi1a-, '1') in new stack [Aug 4 17:50:11] DEBUG[17094]: channel.c:3881 set_format: Set channel SIP/gafachi1a- to write format slin [Aug 4 17:50:11] DEBUG[17094]: rtp.c:3878 ast_rtp_write: Ooh, format changed from unknown to ulaw [Aug 4 17:50:11] DEBUG[17094]: rtp.c:3904 ast_rtp_write: Created smoother: format: 4 ms: 20 len: 160 [Aug 4 17:50:11] DEBUG[17094]: channel.c:2488 ast_settimeout: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Aug 4 17:50:11] -- SIP/gafachi1a- Playing 'digits/1.slin' (language 'en') [Aug 4 17:50:12] DEBUG[17094]: channel.c:2488 ast_settimeout: Scheduling timer at (571 requested / 100 actual) timer ticks per second [Aug 4 17:50:12] DEBUG[17094]: channel.c:2488 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 4 17:50:12] DEBUG[17094]: channel.c:2488 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 4 17:50:12] DEBUG[17094]: channel.c:2488 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 4 17:50:12] DEBUG[17094]: channel.c:3881 set_format: Set channel SIP/gafachi1a- to write format ulaw [Aug 4 17:50:12] DEBUG[17094]: pbx.c:3692 pbx_extension_helper: Launching 'Festival' [Aug 4 17:50:12] -- Executing [...@connect-to-me:3] Festival(SIP/gafachi1a-, hello john) in new stack [Aug 4 17:50:12] == Parsing '/usr/local/etc/asterisk/festival.conf': [Aug 4 17:50:12] DEBUG[17094]: config.c:1330 config_text_file_load: Parsing /usr/local/etc/asterisk/festival.conf [Aug 4 17:50:12] == Found [Aug 4 17:50:12] DEBUG[17094]: app_festival.c:376 festival_exec: Text passed to festival server : hello john [Aug 4 17:50:12] DEBUG[17094]: app_festival.c:446 festival_exec: Cache file exists, strln=10, strlen=10 [Aug 4 17:50:12] DEBUG[17094]: app_festival.c:448 festival_exec: Size OK [Aug 4 17:50:12] DEBUG[17094]: app_festival.c:467 festival_exec: Reading from cache... [Aug 4 17:50:12] DEBUG[17094]: app_festival.c:491 festival_exec: Passing data to channel... [Aug 4 17:50:12] DEBUG[17094]: app_festival.c:513 festival_exec: Festival WV command [Aug 4 17:50:12] DEBUG[17094]: channel.c:3881 set_format: Set channel SIP/gafachi1a- to write format slin [Aug 4 17:50:34] DEBUG[17094]: chan_sip.c:3562 __sip_xmit: Trying to put 'SIP/2.0 200' onto UDP socket destined for 67.216.35.162:5060 And, festival server console looks like following: $ ./bin/festival --server serverWed Aug 4 17:49:04 2010 : Festival server started on port 1314 client(1) Wed Aug 4 17:50:12 2010 : accepted from localhost client(1) Wed Aug 4 17:50:12 2010 : disconnected I have to end the call after sometime. Festival works fine if I got into its console and type SayText(³hello john²) Please let me know how I can fix this. Thanks, Davinder -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update
