Re: [asterisk-users] [AsteriskBrasil] [Elastix-pt] Melhor Chipeira para Integrar com Elastix
Acrescentando o report do Dell, os equipamentos da Khomp são homologados pela Anatel - funcionamento normalmente nas implementações de Asterisk puro, FreePBX ou Elastix. Caso desejem mais informações sobre equipamentos da Khomp, consultem a CAM Tecnologia. A CAM Tecnologia atua com revenda ou venda direta da khomp para o cliente final. Contato: Rubens Duarte de Andrade Tel: (21) 3189-1050 cont...@camtecnologia.com.br www.camtecnologia.com.br Att, Thiago Maluf. -- Thiago Maluf Resende Tel: +55 21 9700-9113 e-mail: malu...@gmail.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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AEL Eswitches
Hi List,
Would someone can to explain me the main difference in SWITCHES or
ESWITCHES in AEL.
context default {
switches {
DUNDi/e164;
IAX2/box5;
};
eswitches {
IAX2/context@${CURSERVER};
};
};
All the best,
Thiago
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Tel: +55 21 9700-9113
e-mail: malu...@gmail.com
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Re: [asterisk-users] IVR and MySQL
Hi Fabio, of course that you can. One way to do it is working with app MYSQL(), where you will put your sql as argumment. read more in http://www.voip-info.org/wiki/view/Asterisk+cmd+MYSQL good luck, Thiago Maluf Resende. 2007/8/14, Fabio Ardeola [EMAIL PROTECTED]: Hi Does somebody know if I can save the answers made by the caller person on the IVR menu in a MySQL Database? If yes, can I save the CallerID as well? Thanks, Fabio Luggage? GPS? Comic books? Check out fitting gifts for grads at Yahoo! Search http://search.yahoo.com/search?fr=oni_on_mailp=graduation+giftscs=bz ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- THIAGO MALUF RESENDE Consultor Voip e Programador WEB (Voip Developer and Web Developer) Tel: +55 21 86042100 e-mail: [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How strip +1 from caller id on inbound call
If you want remove in CALLERID.
you can remove it this way:
exten= _X./_+1X.,1, Set()
ok?
good luck!
Thiago Maluf.
2007/8/14, Anselm Martin Hoffmeister [EMAIL PROTECTED]:
Am Sonntag, den 12.08.2007, 21:16 -0400 schrieb C F:
you can do like this:
exten = _X.,1,GoSubIf($[${LEN(${CALLERID(num)})}10]?strip1);if it's
longer than grab the last 10 digits of the CIDNUM
exten =
_X.,50(strip1),Set(CALLERID(num)=${CALLERID(num):$[${LEN(${CALLERID(num)})}-10]});this
grabs the last 10 digits of CALLERID(num) and sets it to CALLERID(num)
exten = _X.,n,Return()
Argh! You do not ever get international calls, do you? (Well, Canada
does not count here for obvious reasons)
The clean solution to the question
I get some calls with a leading +1. If that is the case, how do I
strip that off?
is of course
If the CALLERID(num) starts +1, re-set it to the same value, offset 2:
...
exten = _X.,n,GoSubIf($[${CALLERID(num):0:2} = +1]?strip1)
...
exten = _X.,n(strip1),Set(CALLERID(num)=${CALLERID(num):2})
exten = _X.,n,Return()
Which leaves international calls for themselves. Of course you still
could replace the leading + for all other numbers by 011, if you
like.
Your code would probably handle
+12125551212
correctly, would work OK with
+495924236
(which might or might not be one of the old, short numbers still present
in some places in Germany), leaving it intact, but not with
+4916177554224
which would be remapped to a Boston MA number (actually a Cingular cell
phone number) instead of mapping it to a german mobile phone.
Variable handling (offset et al) is documented on
http://www.voip-info.org/wiki/view/Asterisk+variables
BR
Anselm
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Consultor Voip e Programador WEB (Voip Developer and Web Developer)
Tel: +55 21 86042100
e-mail: [EMAIL PROTECTED]
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Re: [asterisk-users] Query
Hi Sanchal, 115 in your case is just DIALLED NUMBER and it will be searched by you E1 trunk. If you want change your CALLERID, you would insert one default or would insert one to each user. the command is the same sendt by Todd: callerid=Your Name 5554441212 but you can work with function callerid and set up it in the same extensions. more informations about it, you have in http://www.voip-info.org/wiki/index.php?page=Asterisk+func+callerid all the best and good luck, Thiago Maluf. 2007/8/8, [EMAIL PROTECTED] [EMAIL PROTECTED]: Hi, I am running asterisk PBX ( digium TE120P card configured) on one system. It is connected to E1 card running application on the other system. After establishing sync between two card, I am able to place call from sip phone to E1 card running application. I want to pass the callerid, when calling from sip phone to E1 card running application. Which all configuration files is to be changed in the asterisk. I am doing the following changes in extensions.conf exten=115,1,Dial(ZAP/g1/115,20) So, extension 115 is received at other end as callerid. Is it correct. Can any body help in how to configure for callerid with digium card. thanks and regards sanchal ___ --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 -- THIAGO MALUF RESENDE Consultor Voip e Programador WEB (Voip Developer and Web Developer) Tel: +55 21 86042100 e-mail: [EMAIL PROTECTED] ___ --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] Call Center SoftPhone with Auto Answer
Ola Joao, tem um modo do Asterisk fazer isso sim. Entre em contato no meu GTALK por esse e-mail e eu te dou mais informações. Abs! Hi List, The asterisk have one way to do it. just put one script to discovery if this user is online or offline. case is offline play one music. if not, call the user. understand? thiago! 2007/8/6, Joao Pereira [EMAIL PROTECTED]: Hello I need a Softphone with auto answer where users can't turn it off. Does someone knows a softphone where users can't turn the auto answer off? Or is there any way Asterisk could force the clients to answer the phone? Thanks Regards Joao Pereira ___ --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 -- THIAGO MALUF RESENDE Consultor Voip e Programador WEB (Voip Developer and Web Developer) Tel: +55 21 86042100 e-mail: [EMAIL PROTECTED] ___ --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
[asterisk-users] RE: In Asterisk 1.4.x, Why Digium has two H323 channels?
Hi Users, Administrators and Pavel Jezek, You prefer chan_h323 from asterisk tree and it's of course that use channels by tree is very good. But in 1.2.x, the chan_h323 is very simple and the chan_oh323 is so bad. And I work with chan_ooh323, that it's too from Digium and work good! And I am Studing one possible change to Asterisk 1.4.x , but in 1.4.x the oh323 channel don't have more, and studing the chan_h323, it's the old chan_oh323 I not wanna work with add_ons but in Asterisk 1.4.x, will I have work? Somebody confirm it, have the same opinion. Or this new chan_h323 work fine without the problems that had the H323 or OH323 channels. Thanks in Advanced. Thiago Maluf Resende. -- Date: Mon, 12 Mar 2007 15:57:42 +0100 From: Pavel Jezek [EMAIL PROTECTED] Subject: Re: [asterisk-users] In Asterisk 1.4.x, Why Digium has two H323Channels To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed as I know, ooh323 is external project from objective systems, anyway, for 1.4 I prefer chan_h323 from asterisk tree. Thiago Maluf wrote: Now, the H323 Channels is updated and your bugs fixed. But Digium still develop your OOH323 Channel. My question is why? What is the better in Asterisk 1.4.x.? I know that in Asterisk 1.2.x OOH323 is very better than H323 or OH323. Thanks in advanced. Thiago. -- -- -- THIAGO MALUF RESENDE Consultor Voip e Programador WEB (Voip Developer and Web Developer) Tel: +55 21 86042100 e-mail: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users THIAGO MALUF RESENDE Consultor Voip e Programador WEB (Voip Developer and Web Developer) Tel: +55 21 86042100 e-mail: [EMAIL PROTECTED] ___ --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] In Asterisk 1.4.x, Why Digium has two H323 Channels
Now, the H323 Channels is updated and your bugs fixed. But Digium still develop your OOH323 Channel. My question is why? What is the better in Asterisk 1.4.x.? I know that in Asterisk 1.2.x OOH323 is very better than H323 or OH323. Thanks in advanced. Thiago. -- THIAGO MALUF RESENDE Consultor Voip e Programador WEB (Voip Developer and Web Developer) Tel: +55 21 86042100 e-mail: [EMAIL PROTECTED] ___ --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] AGI PHP
I am using the source following :
write(GET VARIABLE SIPCALLID); $a = read(); $ip = substr($a,36);
function read() { global $in, $debug, $stdlog; $input = str_replace(\n, , fgets($in, 4096)); if ($debug) fputs($stdlog, read: $input\n);
errlog(read: $input\n); return $input; }
function errlog($line) { //gloWbal $err; //fwrite($stdlog, VERBOSE \$line\\n); echo VERBOSE \$line\\n; }
function write($line) { global $debug, $stdlog; if ($debug) fputs($stdlog, write: $line\n); echo $line.\n; }
But it return 510 Invalid or unknown command
somebody can help me! Please!
Thanks,
Thiago Resende.
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