[asterisk-users] SVN
Hello, everyone. Anybody know when that svn will be available again? Regards *Alex Montoanelli* Administração e Gerência de Redes Unetvale Conectividade http://www.unetvale.net +55 48 3263 8700 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SVN
I was trying a 'svn ls http://svn.digium.com/svn/', and was receiving a 403 - Forbiden. But a rising level could access the content. Thank you and hugs Regards *Alex Montoanelli* On Wed, Nov 26, 2008 at 12:17 PM, Atis Lezdins [EMAIL PROTECTED] wrote: On Wed, Nov 26, 2008 at 1:32 PM, Michiel van Baak [EMAIL PROTECTED] wrote: On 09:06, Wed 26 Nov 08, Alex Montoanelli wrote: Hello, everyone. Anybody know when that svn will be available again? Regards Hey, I can checkout stuff fine from svn.digium.com. Maybe you can provide some more info about how it's not working for you. Probably it's that http://svn.digium.com/ gives 403 error. As i recall, it showed up when some search engine tried to indexing whole SVN ignoring robots.txt, so Digium disabled root page. Now you can access it by adding /view/ to URL. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IVR Menu
If the variables *digittimeout, *and *responsetimeout* not response, try use this exten after the last background * exten = s,n,background(silence/7) exten = s,n,goto(s,1) *play a silence for 7 seconds, if the time end, goto the begin of IVR. Alex, Doug Lytle wrote: Dov Bigio wrote: Hi, I made a simple menu using the Background application and some wav files. I converted the wav files using for a in *.wav; do sox $a -r 8000 -c1 `echo $a|sed -e s/wav//`gsm; done (from http://www.voip-info.org/wiki/index.php?page=Convert%20WAV%20audio%20files%20for%20use%20in%20Asterisk) The first two files 01/bemvindo and 01/menu_top are good. But the third file (01/menu_top), fails in the end of the sentence, and this message Auto fallthrough, channel 'SIP/dov.bigio-ae4a' status is 'UNKNOWN' appears in the console. In extensions.conf: If priorityjumping is set to 'yes', then applications that support jumping' to a different priority based on the result of their operations will do so (this is backwards compatible behavior with pre-1.2 releases of Asterisk). Individual applications can also be requested to do this by passing a 'j' option in their arguments. priorityjumping=no Doug ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users begin:vcard fn:Alex Montoanelli n:Montoanelli;Alex org:Unetvale adr;dom:;;;Tijucas;SC;8820-000 email;internet:[EMAIL PROTECTED] tel;quoted-printable;work:48=C2=B732630013 tel;quoted-printable;cell:47=C2=B791498260 x-mozilla-html:TRUE version:2.1 end:vcard ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] wcfxo md3200 problem...
hello all, i have a * 1.2.1, in a lab, only for test, with 4fxo clone - md3200 - intel537, connect to pstn. All work well, but, 1 once day 2 of this cards, stop make call, and receiv call thought. i kill the asterisk, remove modules, wcfxo and zaptel, mount the modules again, and start the *, for resolv the problem. No message in logs, the asterisk or system, dmesg, i use slackware 10.1 with kernel 2.4.29 any idea? thanks all,and sorry for bad english Alex, begin:vcard fn:Alex Montoanelli n:Montoanelli;Alex org;quoted-printable:Unetvale Internet =C2=B7 Agente Autorizado Brasil Telecom;Programador e Administrador de Redes email;internet:[EMAIL PROTECTED] tel;quoted-printable;work:48=C2=B73263 0013 tel;quoted-printable;cell:47=C2=B791498260 x-mozilla-html:TRUE version:2.1 end:vcard ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] modify a cdr values..
Is it possible to modify CDR variables before insert into MySQL or CVS? how? thanks all; Alex begin:vcard fn:Alex Montoanelli n:Montoanelli;Alex org;quoted-printable:Unetvale Internet =C2=B7 Agente Autorizado Brasil Telecom;Programador e Administrador de Redes email;internet:[EMAIL PROTECTED] tel;quoted-printable;work:48=C2=B73263 0013 tel;quoted-printable;cell:47=C2=B791498260 x-mozilla-html:TRUE version:2.1 end:vcard ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PHP Manager
try this ?php $socket = fsockopen(localhost,5038, $errno, $errstr, $timeout); fputs($socket, Action: Login\r\n); fputs($socket, UserName: 1212\r\n); fputs($socket, Secret: 1212\r\n\r\n); fputs($socket, Action: Command\r\n); fputs($socket, Command: reload\r\n\r\n); * fputs($socket, Action: Command\r\n);* fputs($socket, Command: show channels\r\n\r\n); $wrets=fgets($socket,128); ? Code Lover wrote: Hi all, I have a small problem to execute Asterisk Commands in Asterisk Manager using PHP. I am able to run all Asterisk Manager command but the problem is comming with asterisk command. here is the code i am trying to run. ?php $socket = fsockopen(localhost,5038, $errno, $errstr, $timeout); fputs($socket, Action: Login\r\n); fputs($socket, UserName: 1212\r\n); fputs($socket, Secret: 1212\r\n\r\n); fputs($socket, Action: Command\r\n); fputs($socket, Command: reload\r\n\r\n); #Working well fputs($socket, Command: show channels\r\n\r\n); #Not working Working well fputs($socket, Command: 'show channels'\r\n\r\n); #Not working Working well $wrets=fgets($socket,128); ? If you see in my code when i am calling only reload command working but when i am trying to call piar command it is just prompting : == Manager '1212' logged off from localhost without showing channels Please advice me to solve this problem. -- Thank You, Code Lover ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FOP led Colors
Hello guy´s
I´m trying to create a extension do modify the led colors of
a button on a FOP,
Via manager command, liked PHP, but I do not have a good
result, I have set de astdb family in op_astdb.conf, but never,
My Php script, and my extension.conf is bellow
Thanks for all
By
//php script
fputs($socket,
Action: Originate\r\n);
fputs($socket, Channel: Local/[EMAIL PROTECTED]/n\r\n);
fputs($socket,
Context: features \r\n);
fputs($socket, Exten:
*79\r\n);
fputs($socket, Priority:
1\r\n);
fputs($socket, Callerid:
$_SESSION[type]/$_SESSION[extension]\r\n); type like SIP or IAX
fputs($socket,
Timeout: 3\r\n\r\n);
//extensions.conf
[features]
exten = *78,1,UserEvent(ASTDB|Channel:
${CALLERID}^Family: dnd^State: On)
exten = *78,2,SetVar(temp=${CALLERID})
exten = *78,3,Cut(temp=temp,,1)
exten = *78,4,DBPut(dnd/${temp}=On)
exten = *78,5,Hangup
exten = 123,1,answer
exten = 123,n,hangup
Alex Montoanelli
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