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<TITLE>Re: [asterisk-users] Asterisk not working with Festival</TITLE>
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<FONT SIZE="4"><FONT FACE="Calibri, Verdana, Helvetica, Arial"><SPAN STYLE='font-size:11pt'>I just noticed that if I change the dialplan (connect-to-me) to following (shifting places of SayDigits and Festival() ):<BR>
<BR>
[connect-to-me]<BR>
exten => s,1,Answer<BR>
exten => s,n,Festival(hello john)<BR>
exten => s,n,SayDigits('1')<BR>
exten => s,n,Hangup<BR>
<BR>
I do not hear any voice, not even anyone saying “one”. It appears that application hangs while processing festival command. Here is asterisk log:<BR>
<BR>
<BR>
-- Attempting call on SIP/011xxxxxxxxxxxx@gafachi1a for s@connect-to-me:1 (Retry 1)<BR>
[Aug 12 14:48:16] DEBUG[7342]: chan_sip.c:23198 sip_request_call: Asked to create a SIP channel with formats: 0x40 (slin)<BR>
[Aug 12 14:48:16] DEBUG[7342]: chan_sip.c:7336 sip_alloc: Allocating new SIP dialog for 6743a3cc12499c6d7fb77bcd58ab7a02@202.164.37.235 - INVITE (With RTP)<BR>
[Aug 12 14:48:16] DEBUG[7342]: chan_sip.c:5014 do_setnat: Setting NAT on RTP to Off<BR>
[Aug 12 14:48:16] DEBUG[7342]: chan_sip.c:5440 sip_call: Outgoing Call for 011xxxxxxxxxxxx<BR>
[Aug 12 14:48:16] DEBUG[7342]: chan_sip.c:10180 add_sdp: ** Our capability: 0x4 (ulaw) Video flag: False Text flag: False<BR>
[Aug 12 14:48:16] DEBUG[7342]: chan_sip.c:10181 add_sdp: ** Our prefcodec: 0x40 (slin) <BR>
[Aug 12 14:48:16] DEBUG[7342]: chan_sip.c:3275 initialize_initreq: Initializing initreq for method INVITE - callid 581443ae14ba9e13288dd66720a6a9f4@202.164.37.235<BR>
[Aug 12 14:48:17] DEBUG[7342]: chan_sip.c:4061 __sip_ack: Acked pending invite 102<BR>
[Aug 12 14:48:17] DEBUG[7342]: chan_sip.c:4098 __sip_ack: Stopping retransmission on '581443ae14ba9e13288dd66720a6a9f4@202.164.37.235' of Request 102: Match Found<BR>
[Aug 12 14:48:17] DEBUG[7342]: chan_sip.c:10180 add_sdp: ** Our capability: 0x4 (ulaw) Video flag: False Text flag: False<BR>
[Aug 12 14:48:17] DEBUG[7342]: chan_sip.c:10181 add_sdp: ** Our prefcodec: 0x40 (slin) <BR>
[Aug 12 14:48:18] DEBUG[7342]: chan_sip.c:4098 __sip_ack: Stopping retransmission on '581443ae14ba9e13288dd66720a6a9f4@202.164.37.235' of Request 102: Match Not Found<BR>
[Aug 12 14:48:18] DEBUG[7342]: chan_sip.c:4139 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '581443ae14ba9e13288dd66720a6a9f4@202.164.37.235' Request 103: Found<BR>
[Aug 12 14:48:20] DEBUG[7342]: chan_sip.c:4139 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '581443ae14ba9e13288dd66720a6a9f4@202.164.37.235' Request 103: Found<BR>
[Aug 12 14:48:33] DEBUG[7342]: chan_sip.c:4061 __sip_ack: Acked pending invite 103<BR>
[Aug 12 14:48:33] DEBUG[7342]: chan_sip.c:4098 __sip_ack: Stopping retransmission on '581443ae14ba9e13288dd66720a6a9f4@202.164.37.235' of Request 103: Match Found<BR>
> Channel SIP/gafachi1a-00000000 was answered.<BR>
[Aug 12 14:48:33] DEBUG[7342]: pbx.c:3692 pbx_extension_helper: Launching 'Answer'<BR>
-- Executing [s@connect-to-me:1] Answer("SIP/gafachi1a-00000000", "") in new stack<BR>
[Aug 12 14:48:33] DEBUG[7342]: pbx.c:3692 pbx_extension_helper: Launching 'Festival'<BR>
-- Executing [s@connect-to-me:2] Festival("SIP/gafachi1a-00000000", "hello john") in new stack<BR>
== Parsing '/usr/local/etc/asterisk/festival.conf': [Aug 12 14:48:33] DEBUG[7342]: config.c:1330 config_text_file_load: Parsing /usr/local/etc/asterisk/festival.conf<BR>
== Found<BR>
[Aug 12 14:48:33] DEBUG[7342]: app_festival.c:376 festival_exec: Text passed to festival server : hello john<BR>
[Aug 12 14:48:33] DEBUG[7342]: app_festival.c:446 festival_exec: Cache file exists, strln=10, strlen=10<BR>
[Aug 12 14:48:33] DEBUG[7342]: app_festival.c:448 festival_exec: Size OK<BR>
[Aug 12 14:48:33] DEBUG[7342]: app_festival.c:467 festival_exec: Reading from cache...<BR>
[Aug 12 14:48:33] DEBUG[7342]: app_festival.c:491 festival_exec: Passing data to channel...<BR>
[Aug 12 14:48:33] DEBUG[7342]: app_festival.c:513 festival_exec: Festival WV command<BR>
<FONT COLOR="#FF0000"><B>[Aug 12 14:48:33] DEBUG[7342]: channel.c:3881 set_format: Set channel SIP/gafachi1a-00000000 to write format slin<BR>
</B></FONT><B><FONT COLOR="#0000FF">==========================>[DAVINDER] - there is difference of 2 minutes in the log line above and below. I have to end the call as well after waiting so long. <BR>
</FONT><FONT COLOR="#FF0000">[Aug 12 14:50:01] DEBUG[7342]: chan_sip.c:7336 sip_alloc: Allocating new SIP dialog for 407cd1eb69781303206dd7d43ca348bc@202.164.37.235 - REGISTER (No RTP)<BR>
</FONT></B>[Aug 12 14:50:01] DEBUG[7342]: chan_sip.c:11744 transmit_register: Scheduled a registration timeout for 67.216.45.45 id #19 <BR>
[Aug 12 14:50:01] DEBUG[7342]: chan_sip.c:3275 initialize_initreq: Initializing initreq for method REGISTER - callid 407cd1eb69781303206dd7d43ca348bc@202.164.37.235<BR>
[Aug 12 14:50:01] DEBUG[7342]: chan_sip.c:7336 sip_alloc: Allocating new SIP dialog for 35cba4547c9dca423fd4c804010856cc@202.164.37.235 - REGISTER (No RTP)<BR>
[Aug 12 14:50:01] DEBUG[7342]: chan_sip.c:11744 transmit_register: Scheduled a registration timeout for 67.216.35.162 id #21 <BR>
[Aug 12 14:50:01] DEBUG[7342]: chan_sip.c:3275 initialize_initreq: Initializing initreq for method REGISTER - callid 35cba4547c9dca423fd4c804010856cc@202.164.37.235<BR>
[Aug 12 14:50:01] DEBUG[7342]: chan_sip.c:4139 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '407cd1eb69781303206dd7d43ca348bc@202.164.37.235' Request 104: Found<BR>
[Aug 12 14:50:01] DEBUG[7342]: chan_sip.c:4098 __sip_ack: Stopping retransmission on '407cd1eb69781303206dd7d43ca348bc@202.164.37.235' of Request 104: Match Found<BR>
[Aug 12 14:50:01] DEBUG[7342]: chan_sip.c:3273 initialize_initreq: Initializing already initialized SIP dialog 407cd1eb69781303206dd7d43ca348bc@202.164.37.235 (presumably reinvite)<BR>
[Aug 12 14:50:01] DEBUG[7342]: chan_sip.c:4139 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '35cba4547c9dca423fd4c804010856cc@202.164.37.235' Request 104: Found<BR>
[Aug 12 14:50:01] DEBUG[7342]: chan_sip.c:4098 __sip_ack: Stopping retransmission on '35cba4547c9dca423fd4c804010856cc@202.164.37.235' of Request 104: Match Found<BR>
[Aug 12 14:50:01] DEBUG[7342]: chan_sip.c:3273 initialize_initreq: Initializing already initialized SIP dialog 35cba4547c9dca423fd4c804010856cc@202.164.37.235 (presumably reinvite)<BR>
[Aug 12 14:50:02] DEBUG[7342]: chan_sip.c:4139 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '407cd1eb69781303206dd7d43ca348bc@202.164.37.235' Request 105: Found<BR>
[Aug 12 14:50:02] DEBUG[7342]: chan_sip.c:4098 __sip_ack: Stopping retransmission on '407cd1eb69781303206dd7d43ca348bc@202.164.37.235' of Request 105: Match Found<BR>
[Aug 12 14:50:02] DEBUG[7342]: chan_sip.c:18218 handle_response_register: Registration successful<BR>
[Aug 12 14:50:02] DEBUG[7342]: chan_sip.c:18220 handle_response_register: Cancelling timeout 19<BR>
[Aug 12 14:50:02] DEBUG[7342]: chan_sip.c:4139 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '35cba4547c9dca423fd4c804010856cc@202.164.37.235' Request 105: Found<BR>
[Aug 12 14:50:02] DEBUG[7342]: chan_sip.c:4098 __sip_ack: Stopping retransmission on '35cba4547c9dca423fd4c804010856cc@202.164.37.235' of Request 105: Match Found<BR>
[Aug 12 14:50:02] DEBUG[7342]: chan_sip.c:18218 handle_response_register: Registration successful<BR>
<BR>
<BR>
And there is no log entry related to call terminated by remote.<BR>
<BR>
<BR>
<BR>
On 11/08/10 11:22 PM, "Danny Nicholas" <danny@debsinc.com> wrote:<BR>
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<FONT SIZE="4"><FONT FACE="Tahoma, Verdana, Helvetica, Arial"><SPAN STYLE='font-size:10pt'><B>From:</B> asterisk-users-bounces@lists.digium.com [<a href="mailto:asterisk-users-bounces@lists.digium.com]">mailto:asterisk-users-bounces@lists.digium.com]</a> <B>On Behalf Of </B>Davinder Kumar Meen<BR>
<B>Subject:</B> Re: [asterisk-users] Asterisk not working with Festival<BR>
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</SPAN></FONT></FONT><FONT SIZE="4"><FONT FACE="Calibri, Verdana, Helvetica, Arial"><SPAN STYLE='font-size:11pt'>Can anyone help please on this?<BR>
<BR>
<FONT COLOR="#000080"><snip><BR>
></FONT>[connect-to-me]<BR>
<FONT COLOR="#000080">></FONT>exten => s,1,Answer<BR>
<FONT COLOR="#000080">></FONT>Exten => s,n,SayDigits(‘1’)<BR>
<FONT COLOR="#000080">></FONT>exten => s,n,Festival(hello john)<BR>
<FONT COLOR="#000080">></FONT>exten => s,n,Hangup<BR>
<FONT COLOR="#000080"><snip><BR>
</FONT></SPAN></FONT><FONT COLOR="#000080"><FONT FACE="Arial"><SPAN STYLE='font-size:10pt'>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”. <BR>
This is what I would suggest<BR>
</SPAN></FONT></FONT><FONT FACE="Calibri, Verdana, Helvetica, Arial"><SPAN STYLE='font-size:11pt'>[connect-to-me]<BR>
exten => s,1,Answer<BR>
Exten => s,n,Gotoif($[“${EXTEN}:0:3)” = “SIP”]?4:3<BR>
Exten => s,n,wait(7)<BR>
Exten => s,n,SayDigits(‘1’)<BR>
exten => s,n,Festival(hello john)<BR>
exten => s,n,Hangup<BR>
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Thanks,<BR>
Davinder Kumar Meen<BR>
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Impinge Solutions, F-250, Phase 8B, Mohali (India)<BR>
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