RE: [Asterisk-Users] Why I can't hear anything from my sjphone as anh323 endpoint?
That's your problem, u need mpg123 and not 321. There are instructions on the wiki. Donny -Original Message- From: Willis Wang [mailto:[EMAIL PROTECTED] Sent: October 27, 2004 10:41 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Why I can't hear anything from my sjphone as anh323 endpoint? When I call asterisk(registered as an endpoint on gnugk) from sjphone(also registered on gnugk), I can see following on the asterisk console: *CLI == Starting H323/ip$192.168.1.125:3260/4096 at default,20030060,1 failed so falling back to exten 's' -- Executing Wait(H323/ip$192.168.1.125:3260/4096, 1) in new stack -- Executing Answer(H323/ip$192.168.1.125:3260/4096, ) in new stack -- Executing DigitTimeout(H323/ip$192.168.1.125:3260/4096, 5) in new stack -- Set Digit Timeout to 5 -- Executing ResponseTimeout(H323/ip$192.168.1.125:3260/4096, 10) in new stack -- Set Response Timeout to 10 -- Executing BackGround(H323/ip$192.168.1.125:3260/4096, demo-congrats) in new stack Is there anything wrong with my mpg123? I can't hear anything from sjphone, and after I dropped the call, I can't use 'stop now' to quit asterisk, and there will always be a process called mpg123 running: mpg123 -q -s --mono -r 8000 -b 2048 -f 4096 fpm-calm-river.mp3 fpm-sunshine.mp3 My linux version is debian woody 3.0 2.4.27-1-686, and the mpg321 package version is 0.2.10.3 thanks a lot! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Bandwidth Load Balancing / Dundi
App_conference worked well for me, but after upgading to 1.0.2 this evening, it would no longer compile. I will take a closer look at it soon. But it was much better then app_meetme the most important thing being AGC. (automatic gain control) -Original Message- From: Kenneth Shaw [mailto:[EMAIL PROTECTED] Sent: October 26, 2004 5:03 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Bandwidth Load Balancing / Dundi Just wondering what other kinds of solutions people have considered/implemented for load balancing bandwidth and IAX connections over the net. Ideas? Results? Suggestions? Experience? -- Kenneth Shaw [EMAIL PROTECTED] ExpiTrans, Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Bandwidth Load Balancing / Dundi
Correction, with the real 1.0.2 (not the .9 or whatever got accidentally released) It works fine. Check it out, look it up on the wiki. Donny -Original Message- From: Donny Kavanagh Sent: October 27, 2004 1:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Bandwidth Load Balancing / Dundi App_conference worked well for me, but after upgading to 1.0.2 this evening, it would no longer compile. I will take a closer look at it soon. But it was much better then app_meetme the most important thing being AGC. (automatic gain control) -Original Message- From: Kenneth Shaw [mailto:[EMAIL PROTECTED] Sent: October 26, 2004 5:03 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Bandwidth Load Balancing / Dundi Just wondering what other kinds of solutions people have considered/implemented for load balancing bandwidth and IAX connections over the net. Ideas? Results? Suggestions? Experience? -- Kenneth Shaw [EMAIL PROTECTED] ExpiTrans, Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] chan_sip changes affecting ACK? - Bug?
You may want to post this to asterisk-dev and possibly open a bug, if all that is said is correct. Donny -Original Message- From: Chad Brown [mailto:[EMAIL PROTECTED] Sent: Monday, October 25, 2004 1:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] chan_sip changes affecting ACK? - Bug? The INGATE engineer is pointing the finger firmly at asterisk. Any comments from the Asterisk folks? See below: Chad, The problem is that the Asterisk server is not following the RFC. Not only is it not following it in the bad call, but it is not following it in the good call either. It just so happens that they are doing it wrong differently in each case. In the case of the good call, things seem to work anyway despite the incorrect format of the ACK. As you will see in the excerpts from the RFC, assuming that the Asterisk is acting as a loose router, then the the remote target of the route set of this dialog is set by the Contact: header of the 200 OK. That URI should be used by the UA as the Request URI of the ACK, but it isn't. (The Asterisk is populating it Request URI with sip:[EMAIL PROTECTED] rather than sip:[EMAIL PROTECTED] .10.0.5). Therefore, the SIParator is sending the ACK where it is being told. It is just being told the wrong place. If it had received the correct URI, it would have decrypted it and sent it to the correct place. In the case of the Good Call, the Asterisk IS populating the Request URI correctly, however, it should then include a Route header with the route set values in order. Instead, it is adding the Route header and populating it with the contents of the Contact field(sip:e_RY4_466QliT14zp26IqP6KYbo9s6ZERZM0fQuq8nzGMs71r0jwT2UOVGyjPo [EMAIL PROTECTED]) It should be populating it with sip:[EMAIL PROTECTED];lr. From RFC3261. 13.2.2.4 2xx Responses [...] The header fields of the ACK are constructed in the same way as for any request sent within a dialog (see Section 12) with the exception of the CSeq and the header fields related to authentication. 12.2.1.1 Generating the Request [...] The UAC uses the remote target and route set to build the Request-URI and Route header field of the request. If the route set is empty, the UAC MUST place the remote target URI into the Request-URI. The UAC MUST NOT add a Route header field to the request. If the route set is not empty, and the first URI in the route set contains the lr parameter (see Section 19.1.1), the UAC MUST place the remote target URI into the Request-URI and MUST include a Route header field containing the route set values in order, including all parameters. If the route set is not empty, and its first URI does not contain the lr parameter, the UAC MUST place the first URI from the route set into the Request-URI, stripping any parameters that are not allowed in a Request-URI. The UAC MUST add a Route header field containing the remainder of the route set values in order, including all parameters. The UAC MUST then place the remote target URI into the Route header field as the last value.. 200OK Sent to the Asterisk by the SIParator. (Bad Call) SIP/2.0 200 OK To: sip:[EMAIL PROTECTED];tag=3307485377-144837 From: Chad Brown sip:[EMAIL PROTECTED];tag=as1ce965cb Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Contact: sip:[EMAIL PROTECTED] 0.10.0.5 Content-Type: application/sdp Via: SIP/2.0/UDP 10.10.0.6:5060;branch=z9hG4bK03f598de Record-Route: sip:[EMAIL PROTECTED];lr Content-Length: 187 v=0 o=NexTone-MSW 1234 467330188 IN IP4 10.10.0.5 s=sip call c=IN IP4 10.10.0.5 t=0 0 m=audio 58024 RTP/AVP 0 a=silenceSupp:off a=ecan:b on g168 a=ptime:20 a=rtpmap:0 PCMU/8000 ACK sent back by the Asterisk.(Bad Call) ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.10.0.6:5060;branch=z9hG4bK78bbf6a8 Route: sip:[EMAIL PROTECTED] 0.10.0.5 From: Chad Brown sip:[EMAIL PROTECTED];tag=as1ce965cb To: sip:[EMAIL PROTECTED];tag=3307485377-144837 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 In summary, the problem is being caused by the incorrect format of the ACKs coming from the Asterisk. This should be corrected there. Please let me know if you have any questions. Thanks Shane Cleckler Mgr Systems Engineering Ingate Systems -Original Message- From: Chad Brown [mailto:[EMAIL PROTECTED] Sent: Friday, October 22, 2004 10:00 PM To: [EMAIL PROTECTED] Cc: Shane Cleckler Subject: chan_sip changes affecting ACK? Are there any changes to chan_sip since 09/16/04 in the stable branch that could affect the way Asterisk issues an ACK? The reason I ask...I have a product by INGATE called the Siparator which assists in NAT traversal. It worked great until I upgraded to Asterisk v1.0. After comparing the
RE: [Asterisk-Users] answer on # key?
Title: [Asterisk-Users] answer on # key? Did you see the code I posted a day or two ago, it should do exactly what you want by running a macro before the calls are joined. However I tested it on Zap so I have no idea if it works between other mediums, please check it out and let me know. If you missed the posting, email me personally and I will hook you up with the code. Donny From: Marty Mastera [mailto:[EMAIL PROTECTED] On Behalf Of Marty Mastera Sent: Thursday, October 21, 2004 5:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] answer on # key? Matthew: That feature is referred to as Answer Supervision, and the dial flag is c...beware however, it only works when the outbound call is made via a Zap channel. I have the exact same need as you, however the call is made using either sip or iax, and currently answer supervision isn't an option. I posted to a bug report asking about making this feature channel type independant, but I don't think it's going anywhere yet. Marty From: [EMAIL PROTECTED] on behalf of Matthew Simpson Sent: Thu 10/21/2004 11:47 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] answer on # key? I thought I read somewhere on the Wiki that one could give Dial() an argument that would first dial the extension, but not bridge the connection until the called party hit the # key. It must have been during one of those late night coding sessions because now I can't find anything to do with that other than options to allow hangup of the call by hitting *. Does such an option exist? If not, is anyone using a Macro to do that? I have a system that attempts to do a Dial out to a cell phone number with a 15 second timer as a find me type of application. If the cell phone is off or out of range, the 15 seconds of ring time isn't reached and the caller gets connected to the cell phone's voicemail instead of the Asterisk voicemail like I want. Having the # to connect option would fix this problem. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] (no subject)
I've been working on the asterisk manager for a few days, today I
started on a little prototype click to talk on the web via php thing. I
have it working properly except for one thing. I do a SetVar:
CTTN=somenumber over the socket and when the call gets to the socket
its empty. Is this intended? Is it supposed to work?
Heres my code.
I put the number I wanted to dial into the Account variable and it works
fine, however, when I set it into CTTN using set var, no such luck. Am
I doing something wrong? Is this feature not intended? Or is it a bug,
anyone have any ideas? The socket also returns no errors from the call.
Thanks in advance,
Donny
-- code --
PRE
?php
$phonenumber1=6164;
$phonenumber2=88626327;
echo (SetVar: CTTN=$phonenumber2\r\n);
echo (Account: $phonenumber2\r\n);
$socket = fsockopen(192.168.0.52,5038, $errno, $errstr, $timeout);
fputs($socket, Action: Login\r\n);
fputs($socket, UserName: test\r\n);
fputs($socket, Secret: test\r\n);
fputs($socket, Events: off\r\n\r\n);
# fputs($socket, \r\n\r\n);
fputs($socket, Action: Originate\r\n);
fputs($socket, Channel: Zap/g1/$phonenumber1\r\n);
fputs($socket, MaxRetries: 3\r\n);
fputs($socket, RetryTime: 30\r\n);
fputs($socket, WaitTime: 30\r\n);
fputs($socket, Context: clicktotalk-f\r\n);
fputs($socket, Extension: s\r\n);
fputs($socket, Priority: 1\r\n);
fputs($socket, SetVar: CTTN=$phonenumber2\r\n);
fputs($socket, Account: $phonenumber2\r\n);
fputs($socket, Callerid: CRA CTT (613) 111-\r\n\r\n);
fputs($socket, Action: Logoff\r\n\r\n);
while (!feof($socket)) {
$wrets .= fread($socket, 8192);
}
fclose($socket);
echo $wrets;
?
/pre
-- and the context --
[clicktotalk-f]
exten= s,1,Answer
exten= s,2,Wait(2)
exten= s,3,Noop(${CTTN})
exten= s,4,Dial(Zap/g1/${ACCOUNTCODE})
exten= s,5,Hangup
-- the socket reports --
SetVar: CTTN=88626327
Account: 88626327
Asterisk Call Manager/1.0
Response: Success
Message: Authentication accepted
Response: Success
Message: Originate successfully queued
Response: Goodbye
Message: Thanks for all the fish.
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RE: [Asterisk-Users] answer on # key?
I did this for a click to talk prototype application, hope this helps.
There is no error checking in this prototype aka it will time out
indefinitally on the menu until they hangup etc. It was just a proof of
concept. Temp.agi called text2wave from festival and generated a wav
file. Badly named, but this was hacked together in an afternoon.
In macro-clicktotalk Read seems to require you play a file before input
(or at least that's how I was able to get it working) hence the playing
of beep, ideally all these messages would be recorded and not using TTS
there for the message you have a CTT call would just be played in the
read.
Some of the options in the dial may not be required, I was playing
around with this code somewhat trying to get it to transfer so it
woudn't use up to zap channels while the call was active. The ideal
solution is just to transfer, however then you have no ability to
control both sides of the call.
I wrote my own agi because calling Festival() was finicky and only
seemed to work for me via the pri/zap and not sip?? This agi will work
on unpatched festival as well. If you have any further questions, or
want the code for temp.agi (should be named tts.agi) feel free to
contact me off list if you choose to.
Donny
-- code --
[clicktotalk]
exten= s,1,Answer
exten= s,2,Wait(2)
exten= s,3,AGI,temp.agi|This is a call from the C T T service of where
I work
exten= s,4,AGI,temp.agi|To accept the call press 1 to refuse the call
press 2.
exten= 1,1,AGI,temp.agi|Please wait while we make contact with an agent
exten= 1,2,AGI,temp.agi|You may expirence a short wait and a peroid of
silence. Please remain on the line.
exten= 1,3,Goto(clicktotalk-agent,s,1)
exten= 2,1,AGI,temp.agi|Since you have indicated you do not wish to
accept this call
exten= 2,2,AGI,temp.agi|your request will be canceled.
exten= 2,3,Hangup
exten= t,1,Goto(clicktotalk,s,1)
[clicktotalk-agent]
exten= s,1,Dial(Zap/g1/82979022,30,tTmgM(clicktotalk))
exten= s,2,Hangup
[macro-clicktotalk]
exten= s,1,AGI,temp.agi|You have a CTT Call. Please press any key.
exten= s,2,Read(XInput,beep,1)
exten= s,3,Gotoif(${XInput}?10:20)
exten= s,10,Goto(Z,1)
exten= s,20,Goto(s,1)
exten= t,1,Goto(s,1)
exten= Z,1,AGI,temp.agi|The refrence number is 1 1 1 1.
exten= Z,2,AGI,temp.agi|We are now connecting you to the client.
exten= Z,3,Wait(5)
-Original Message-
From: Matthew Simpson [mailto:[EMAIL PROTECTED]
Sent: Thursday, October 21, 2004 1:47 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] answer on # key?
I thought I read somewhere on the Wiki that one could give Dial() an
argument that would first dial the extension, but not bridge the
connection
until the called party hit the # key. It must have been during one of
those late night coding sessions because now I can't find anything to do
with that other than options to allow hangup of the call by hitting *.
Does such an option exist?
If not, is anyone using a Macro to do that?
I have a system that attempts to do a Dial out to a cell phone number
with a
15 second timer as a find me type of application. If the cell phone is
off
or out of range, the 15 seconds of ring time isn't reached and the
caller
gets connected to the cell phone's voicemail instead of the Asterisk
voicemail like I want. Having the # to connect option would fix this
problem.
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RE: [Asterisk-Users] RE: answer on # key?
See my post using a macro, it will loop a message waiting for the other end to press a dtmf to accept the call, you can optionally also play more information (in my case it was a ticket number so the agent could pull up additional information) The only limitation for me was there is no ability to play a message when both parties are connected since dial is a blocking call. It would be ideal if there were a flag to make Dial() non blocking, and then when that flag is used, we should be able to call another blocking function eg WaitForCall such that when we are finished running the context the parties wont be disconnected until they hang up. Say for example the non blocking dial flag is B (maybe that's used for something else, I'm just being hypothetical) We would do [context] exten= 1,1,Dial(Zap/g1/,30,B) exten= 1,2,Wait(2) exten= 1,3,BackGround(youarenowconnected) exten= 1,4,AGI,some-agi-to-log-the-sucessful-connection.agi exten= 1,5,WaitForCall exten= 1,6,Hangup This would be ideal. What does everyone think of that. Donny -Original Message- From: Christopher Jacob [mailto:[EMAIL PROTECTED] Sent: Thursday, October 21, 2004 2:25 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] RE: answer on # key? I was asking about this about a week ago. What I found out is that the # option is in the ZAP channel not the dial() command. Ie. Dial(Zap/g1c/5551212) It does work as advertised, but in my mind has some limitations. It sits silently waiting for user input. There is a bug filled, but I don't know the status. In my mind you should be able to play a file. (Press # to accept this call) Others have suggested that it be moved to the dial() command so that it could be used across all channels. I don't know if this is possible. Hope this helps... ~chris Message: 9 Date: Thu, 21 Oct 2004 12:47:15 -0500 From: Matthew Simpson [EMAIL PROTECTED] Subject: [Asterisk-Users] answer on # key? To: [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; format=flowed; charset=iso-8859-1; reply-type=original I thought I read somewhere on the Wiki that one could give Dial() an argument that would first dial the extension, but not bridge the connection until the called party hit the # key. It must have been during one of those late night coding sessions because now I can't find anything to do with that other than options to allow hangup of the call by hitting *. Does such an option exist? If not, is anyone using a Macro to do that? I have a system that attempts to do a Dial out to a cell phone number with a 15 second timer as a find me type of application. If the cell phone is off or out of range, the 15 seconds of ring time isn't reached and the caller gets connected to the cell phone's voicemail instead of the Asterisk voicemail like I want. Having the # to connect option would fix this problem. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] app_conference
I found this patch a few days ago (on a mailing list), and patched it against the latest cvs which I downloaded for app conference. With these changes I believe everything compiled fine no other tweaks required other then the include dir for asterisk in the make file. On a side note, id like to see a few more features in this module beep on entry part is a big one, however my c/c++ isnt savy enough. I made some attempts but without any luck. If anyone feels bored, try it out and let me know. Donny -Original Message- From: Darren Wiebe [mailto:[EMAIL PROTECTED] Sent: October 20, 2004 7:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] app_conference I don't really like swapping binaries but... I have an app_conference.so binary file I could send to you if you like. It is working on the latest stable cvs as of a few days ago. If you would like it, please let me know and I will get it available. Darren Wiebe [EMAIL PROTECTED] Steve Kann wrote: Shawn Dillon wrote: Thanks to all who have helped me build and test out Asterisk installation thus far. I needed to move my * installation to a new box , due to the fact my test machine would not support PCI 2.2 ( which I am told is required to use my TDM11B). I have * up and running and I am attempting to compile the app_conference source. The MeetMe app has too much echo. I am running Debian 2.4.26 and get tons of compile errors. If I compile right from the CVS of app_conference I get: chatterbox:/usr/src/app_conference# make gcc -pipe -std=c99 -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/root/local/asterisk/asterisk/include -D_REENTRANT -D_GNU_SOURCE -O3 -ffast-math -funroll-all-loops -fprefetch-loop-arrays -fsingle-precision-constant -mcpu=7450 -faltivec -mabi=altivec -mdynamic-no-pic -DCRYPTO -DAPP_CONFERENCE_DEBUG -Ilibspeex -DSILDET=2 -c -o app_conference.o app_conference.c cc1: error: invalid option `abi=altivec' cc1: error: invalid option `dynamic-no-pic' cc1: error: unrecognized option `-faltivec' cc1: error: bad value (7450) for -mcpu= switch cc1: warning: -fprefetch-loop-arrays not supported for this target (try -march switches) make: *** [app_conference.o] Error 1 I then fix the mcpu ( I am on a Pentium4 Box). I comment out the line. I run make clean and make and then get the following. gcc -pipe -std=c99 -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/root/local/asterisk/asterisk/include -D_REENTRANT -D_GNU_SOURCE -O3 -ffast-math -funroll-all-loops -fprefetch-loop-arrays -fsingle-precision-constant -DCRYPTO -DAPP_CONFERENCE_DEBUG -Ilibspeex -DSILDET=2 -c -o app_conference.o app_conference.c cc1: warning: -fprefetch-loop-arrays not supported for this target (try -march switches) gcc -pipe -std=c99 -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/root/local/asterisk/asterisk/include -D_REENTRANT -D_GNU_SOURCE -O3 -ffast-math -funroll-all-loops -fprefetch-loop-arrays -fsingle-precision-constant -DCRYPTO -DAPP_CONFERENCE_DEBUG -Ilibspeex -DSILDET=2 -c -o conference.o conference.c cc1: warning: -fprefetch-loop-arrays not supported for this target (try -march switches) conference.c:29: error: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) conference.c:32: error: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) make: *** [conference.o] Error 1 I change my two lines in conference.c as per http://lists.digium.com/pipermail/asterisk-users/2004-September/06376 5.html I run make clean, make and get the following error chatterbox:/usr/src/app_conference# make gcc -pipe -std=c99 -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/root/local/asterisk/asterisk/include -D_REENTRANT -D_GNU_SOURCE -O3 -ffast-math -funroll-all-loops -fprefetch-loop-arrays -fsingle-precision-constant -DCRYPTO -DAPP_CONFERENCE_DEBUG -Ilibspeex -DSILDET=2 -c -o app_conference.o app_conference.c cc1: warning: -fprefetch-loop-arrays not supported for this target (try -march switches) gcc -pipe -std=c99 -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/root/local/asterisk/asterisk/include -D_REENTRANT -D_GNU_SOURCE -O3 -ffast-math -funroll-all-loops -fprefetch-loop-arrays -fsingle-precision-constant -DCRYPTO -DAPP_CONFERENCE_DEBUG -Ilibspeex -DSILDET=2 -c -o conference.o conference.c cc1: warning: -fprefetch-loop-arrays not supported for this target (try -march switches) gcc -pipe -std=c99 -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/root/local/asterisk/asterisk/include -D_REENTRANT -D_GNU_SOURCE -O3 -ffast-math -funroll-all-loops -fprefetch-loop-arrays -fsingle-precision-constant -DCRYPTO -DAPP_CONFERENCE_DEBUG -Ilibspeex -DSILDET=2 -c -o member.o member.c cc1: warning:
RE: [Asterisk-Users] DUNDi in stable? (New subject)
Current CVS has some realtime changes for voicemail sip (pull from database, no reload required!) can we expect to see more fo the same? Could the dial plan eventually be databased? Or Is this even possible due to the complexity of it. (its obviously possible, but would it just confuse things more) -Original Message- From: Matt Riddell [mailto:[EMAIL PROTECTED] Sent: October 19, 2004 10:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] DUNDi in stable? (New subject) Brian West wrote: Yes but I'm just saying that if you want it get a checkout it from now as in THIS POINT IN TIME otherwise you're gonna have fun in the next few weeks once the major changes start going in. Oooh /me is getting excited about the new changes...any hints of things to come? -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dialogic D/300JCT-E1 support
Do the dialogic drivers from digium require those lame redhat 7.2/7.3 only drivers that intel released? Donny -Original Message- From: Brian West [mailto:[EMAIL PROTECTED] Sent: October 14, 2004 8:19 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Dialogic D/300JCT-E1 support You would do well to ebay the card if you don't otherwise need it and then buy a Digium card. And you have to sign an NDA to get the drivers for a Dialogic card from Digium. bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TTS Voice Rec (sphinx)
Im not sure if this is a bug, or if Im doing
something wrong but I followed the instructions on the wiki for festival, I installed
1.4.3 applied two patches, one for it not compiling on fedora core2 (newer gcc)
as well as one patch to make it work with cmd Festival. This seemed to
work well however, it only works when called via the zaptel card. When I call
via sip, I would see the Festival cmd execute on the cli, but I heard nothing,
and the sip call hung up.
I ended up writing an agi myself which I believe will run on
an unpatched copy of festival. It also works fine on both versions I have
on my system 1.4.3 as well as 1.95beta.
As a side note, perhaps it would be a cool idea to be able
to map agis to be functions within extensions.
--snip
#!/usr/bin/perl
#make a tts dir inside your sounds dir.
use Asterisk::AGI;
use File::Basename;
use Digest::MD5 qw(md5_hex);
$AGI = new Asterisk::AGI;
my %input = $AGI-ReadParse();
my ($text)[EMAIL PROTECTED];
my $hash = md5_hex($text);
my $sounddir = /var/lib/asterisk/sounds/tts;
my $wavefile =
$sounddir/.tts-$hash.wav;
my $t2wp=
/root/www.cstr.ed.ac.uk/download/festival/1.95/festival/bin/;
unless (-f $wavefile) {
open(fileOUT,
$sounddir./say-text-$hash.txt);
print fileOUT
$text;
close(fileOUT);
my
$execf=$t2wp.text2wave $sounddir/say-text-$hash.txt -F 8000 -o
$wavefile;
system($execf);
unlink($sounddir./say-text-$hash.txt);
}
$AGI-stream_file('tts/'.basename($wavefile,.wav));
--snip
As for sphinx Ive done some research via google,
searched the mailing list and Ive found very little information on this
subject.
Looking at eagi-sphinx-test.c I think I see where the voice detection
occurs, however Im not sure if it just listens for sound, or if it will
return a word it thinks its heard, or if we provide a word list of what we are
looking for etc. I am just doing some preliminary research before I go
ahead and install it.
Has anyone else installed sphinx? Do you have any
working examples to post etc, if you know anything share it because theres
a definite lack of information at the moment.
Thanks,
Donny
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RE: [Asterisk-Users] TDM01B Goes missing after reboot
When you do that shut down asterisk first J I made the mistake of unloading the specific module for my card (which did unload) but when I tried to unload zaptel it said it was in use, realizing asterisk was still loaded I connected and did a stop now. Bye bye linux, Nice kernel panic stack trace for me, which was on a machine I wasnt sitting next to! Lesson learned J Donny From: Ian D. Wlloughby [mailto:[EMAIL PROTECTED] Sent: Wednesday, October 13, 2004 5:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] TDM01B Goes missing after reboot Hmm, Didn't think about unloading the driver, sounds like a plan. I will give it a go when I get home. Thanks Ian From: [EMAIL PROTECTED] Sent: Wed 13/10/2004 02:17 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] TDM01B Goes missing after reboot On Oct 12, 2004, at 7:38 PM, Ian D. Wlloughby wrote: Hi All, I have just installed a TDM01B to fix my UK callerid and echo problems. In this respect everything is wonderful, however when I reboot wcfxs fails to load due to No Device found. If I power off and on everything is fine. I noticed that wctdm does not appear in /proc/interrupts after the reboot but does after power off/on. This seems similar to other peoples problems, do I have a duff card (Revision H) or is this a bug in wcfxs ? Regards IanIan,I responded to a similar posting today. With any luck, this workaroundwill also work for you.http://lists.digium.com/pipermail/asterisk-users/2004-October/ 067004.htmlNiles ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TTS via text2wave
I was refering to cacheing the generated wave files so they didn't have to be created everytime. It would be ideal if they did, save plenty of cpu time. Is this possible? Donny -Original Message- From: Paul Dugas [mailto:[EMAIL PROTECTED] Sent: October 11, 2004 10:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] TTS via text2wave Donny Kavanagh said: Could these files be cached as well? Not sure what files you're refering to but the AGI Perl script isn't being cached as I've been able to change it and call the extension to see the changes without a reload. No res_perl going on here unless it magically part of the stock build now; don't think so. I don't think the sound files are being cached as their names are pretty unique as generated by the Perl File::Temp module. Is there a way to enable additional debugging of the activity in * due to the STREAM FILE command from my AGI? Doing a set verbose and set debug with really big numbers doen's give me anything useful. Thanks again, Paul -- Paul A. Dugas Dugas Enterprises, LLC email: [EMAIL PROTECTED]1711 Indian Ridge Drive phone: 404.932.1355 fax: 770.516-4841 Woodstock, GA 30189 USA [ onsite at the Georgia DOT's West Annex, 404.463.2860 x158 ] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TTS via text2wave
Could these files be cached as well?
Donny
-Original Message-
From: Paul Dugas [mailto:[EMAIL PROTECTED]
Sent: October 10, 2004 12:56 PM
To: Asterisk Mailing List
Subject: [Asterisk-Users] TTS via text2wave
Tinkering with getting a text-to-speech component worked into my dial
plan without having to run the Festival server. Entries in
extensions.conf and the tts.agi script are below. I know I must be
close as it worked once.
Was shocked as it was after about 50 different tries. Tried it again
and it didn't work. Been tinkering since so the script is not exactly
the same but it's close.
Thinking the file may have been deleted before it was played, I've tried
clearing the UNLINK option to tempfile() but still no luck.
When the file was being created in '/tmp/ but * was lookig in
.../sounds, I would get an eror indicating it couldn't find the file so
I know the STREAM FILE command was being sent. When I fixed the
paths, the error went away.
Log message from the console were:
-- Executing Wait(SIP/106-5a8d, 1) in new stack
-- Executing AGI(SIP/106-5a8d, tts.agi|Shall we play a game?) in
new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/tts.agi
tts.agi|Shall we play a game?: FORKING text2wave
tts.agi|Shall we play a game?: CREATED
/var/lib/asterisk/sounds/HykFjnCdwu-tts.wav
tts.agi|Shall we play a game?: DONE
-- AGI Script tts.agi completed, returning 0
-- Executing Hangup(SIP/106-5a8d, ) in new stack
extensions.conf entries like so:
exten = 7003,1,Wait(1)
exten = 7003,2,AGI(tts.agi|Shall we play a game?)
exten = 7003,3,Hangup
tts.agi script like so:
#!/usr/bin/perl -w
#
# tts.agi - AGI script for test-to-speech using Festival's text2wave
#
# modules
use strict;
use Asterisk::AGI;
use File::Basename;
use File::Temp;
# defined
use constant SOUNDS= '/var/lib/asterisk/sounds';
use constant TEXT2WAVE = '/root/work/festival/bin/text2wave';
use constant TEXT = join( ' ', @ARGV );
# init AGI
my $AGI = new Asterisk::AGI;
my %input = $AGI-ReadParse();
# temp file
my (undef, $tmpfile) = File::Temp::tempfile( DIR= SOUNDS,
SUFFIX = '-tts.wav',
UNLINK = 1 );
$AGI-verbose( FORKING text2wave\n, 4 );
# fork
my $pid = open my $pipe, |-;
die fork() failed: $! unless defined $pid;
if ( !$pid )
{
# child
open STDOUT, $tmpfile or die can't redir to $tmpfile: $!;
exec TEXT2WAVE, '-F', '8000', '-';
die exec() failed: $!;
}
else
{
# parent
print $pipe TEXT;
close $pipe;
waitpid $pid, 0;
$AGI-verbose( CREATED $tmpfile\n, 4 );
$AGI-stream_file( basename( $tmpfile, '.wav' ) );
}
# outahere
$AGI-verbose( DONE\n, 4 );
0;
Any suggestions appreciated,
Paul
--
Paul A. Dugas Dugas Enterprises, LLC
email: [EMAIL PROTECTED]1711 Indian Ridge Drive
phone: 404.932.1355 fax: 770.516-4841 Woodstock, GA 30189 USA
[ onsite at the Georgia DOT's West Annex, 404.463.2860 x158 ]
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