[Asterisk-Users] Realtime goto problem
Hi, Sample database ++---+---+--+-+- -+ | id | context | exten | priority | app | appdata | ++---+---+--+-+- -+ | 1 | incoming| 6069 |1 | Goto| incoming-next|6069|1 | | 2 | incoming| 6069 |2 | Hangup | | | 3 | incoming-next | 6069 |1 | DigitTimeout| 10 | | 4 | incoming-next | 6069 |2 | ResponseTimeout | 30 | | 5 | incoming-next | 6069 |3 | Background | welcome If i dont declare the incoming-next context in extensions.conf I get: Channel 'Zap/21-1' sent into invalid extension '1' in context 'incoming-next ', but no invalid handler. But if I put on extensions.conf: [incoming-next] Switch = Realtime/@ ,it works fine. Do we need to declare all contexts in extensions.conf so we can use it on Realtime?? Another question: Its possible to include contexts in Realtime like we made on extensions.conf? Thanks in advance, Pedro Nunes ___ --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] Leading 0 on caller ID with internal S0 (HFC)
Hi Sven, Same problem... Not solved... With CAPI and mISDN. I think it as to do with nationalprefix=0 internationalprefix=00 on capi.conf/misdn.conf. I already try to nationalprefix= but always get that damn 0. If I change nationalprefix=5 I get a leading 5 and so on... But without any leading digit I couldn't do it yet. Pedro Nunes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sven Fischer Sent: quinta-feira, 9 de Fevereiro de 2006 9:23 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Leading 0 on caller ID with internal S0 (HFC) Hi all, I have a problem: On my internal S0 where phones are connected via HFC I get all the number with a leading 0 (either from internal SIP phones or external dialins via CAPI). I don't know where to look for this 0. Any ideas? Greetings, Sven -- Sven Fischer (Dipl.-Phys.) - FACIT Consulting GmbH Hausinger Str. 6 - 40764 Langenfeld Tel: 02173/16700-55 Fax: 02173/16700-60 ___ --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] mISDN errors on asterisk CLI
Hi there guys, Does anyone know what this is?? Every time a mISDN channel connects to anything, I get this message on the CLI of asterisk. Unhandled Message: prim 281 len -22 from addr 51400101, dinfo 0 on port: 1 Thanks Pedro Nunes ___ --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] Starting RTP with Dial and MusicOnHold
Hello, Do you try Answer() and then Dial(SIP/xyz,,m)??? Exten = ???,1,Answer() Exten = ???,2,Dial(SIP/xyz,,m) You need to answer the call before you can hear music on hold. Pedro Nunes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aaron Clauson Sent: quinta-feira, 15 de Dezembro de 2005 4:45 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Starting RTP with Dial and MusicOnHold Hi, I'm trying to get Asterisk working with a supplier's Cerpack switch and everything is working except audio ringback for calls coming from Cerpack to Asterisk. The Cerpack switch only does out of band progress indication (seems a bit strange for SIP to SIP calls?!) so I've spent the last two days trying to find a way to force Asterisk to send an RTP stream to Cerpack for ring back. Theoretically the Dial command with the m option looks to be exactly what I need: Dial(SIP/xyz,,m) This should play musiconhold back to the caller and in my case I just took a recording of the PSTN tones I wanted to play and created a musiconhold class for them. The command will work correctly when dialled from a SIP phone connected to Asterisk but not for calls coming from Cerpack. As far as I can tell this is because Asterisk won't initiate the RTP stream and waits for a packet from the client before starting to play the musiconhold, perhaps assuming the connection is not available until it gets a packet. In this case Cerpack isn't sending a packet so no audio is heard until the call is answered. Has anybody seen anything like this before? Thanks, Aaron ___ --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] mISDN Caller ID problem
Hello everyone, I am trying mISDN driver with asterisk 1.2.1 but when i call from SIP to mISDN and from mISDN to SIP, the caller ID appears always with a leading 0 (0X). I think the problem is with nationalprefix. How can I remove that zero Here is my config. [general] debug=0 trace_calls=false trace_dir=/var/log/ bridging=yes stop_tone_after_first_digit=yes append_digits2exten=yes l1_info_ok=yes clear_l3=no method=standard dynamic_crypt=no crypt_prefix=** crypt_keys=test,muh [default] context=default language=en nationalprefix=0 internationalprefix=00 rxgain=0 txgain=0 te_choose_channel=no dialplan=0 use_callingpres=yes ;always_immediate=no ;immediate=no ;hold_allowed=yes ;callgroup=1 ;pickupgroup=1 ;presentation=not_screened ;echocancel=no echocancelwhenbridged=no echotraining=yes [group1] ports=1 context=bri_card_1 msns=* Thanks in advance Pedro Nunes ___ --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] Need advice on BRI
Hello all, I need to install a production server with BRI support. I know that exists bristuff, misdn, chan_capi ... I have hcfpci based cards. For a very stable environment, what driver should I use?? Thanks in advance Pedro Nunes ___ --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] Fritz!Card PCI ver2.0
What chipset that card use?? Pedro Nunes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Arulraj Sent: terça-feira, 1 de Novembro de 2005 23:30 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Fritz!Card PCI ver2.0 Anyone knows how I can use this ISDN card for asterisk as a BRI trunk interface? Thanks, Stephen ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Bristuff question
Hi there, I have 2 ISDN modems (HFC chipset). I use bristuff from junghanns. Its possible to load one cards as NT (T-Bus) and the other as S-Bus. When I do make load the 2 cards loads as S-Bus and when I do make loadNT the 2 cards loads as T-Bus. Can someone help me?? Thanks in advance Pedro Nunes ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Bristuff question
Giordano, Thanks, stupid question. Ive look to that page 100 of times but I do not remember that part of the page about loading more than one card :S. Thanks again From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giordano Grandis Sent: quinta-feira, 27 de Outubro de 2005 15:30 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: R: [Asterisk-Users] Bristuff question http://www.voip-info.org/wiki-Asterisk+zaphfc look this Giordano Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] Per conto di Pedro Nunes Inviato: giovedì 27 ottobre 2005 16.23 A: asterisk-users@lists.digium.com Oggetto: [Asterisk-Users] Bristuff question Hi there, I have 2 ISDN modems (HFC chipset). I use bristuff from junghanns. Its possible to load one cards as NT (T-Bus) and the other as S-Bus. When I do make load the 2 cards loads as S-Bus and when I do make loadNT the 2 cards loads as T-Bus. Can someone help me?? Thanks in advance Pedro Nunes ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] ACD/queues question
Thanks, That will fix my problem... And agent skills, is that possible too?? Thanks again Pedro Nunes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lorenzo Emilitri Sent: quinta-feira, 13 de Outubro de 2005 8:17 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] ACD/queues question Hello Pedro, you should do this using agent priority groups; this way first all low priority agents are filled, then another group is used up. Thanks l. On Wed, 12 Oct 2005 19:30:43 +0200, Pedro Nunes [EMAIL PROTECTED] wrote: Hi there, Does anyone know how to setup an overflow queue? When a call rings on the queue A, if all agents were busy, the call goes to the queue B. If all agents in queue B were busy, then the call stays on both queues until somebody answers it. I think this is a basic ACD feature available on most PBX that support ACD functionality. Does anybody knows how to do it with asterisk?? Thanks in advance Pedro Nunes -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] ACD/queues question
Thanks, That will fix my problem... And agent skills, is that possible too?? Thanks again Pedro Nunes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Rymes Sent: quarta-feira, 12 de Outubro de 2005 23:39 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] ACD/queues question On Oct 12, 2005, at 1:30 PM, Pedro Nunes wrote: Hi there, Does anyone know how to setup an overflow queue? When a call rings on the queue A, if all agents were busy, the call goes to the queue B. If all agents in queue B were busy, then the call stays on both queues until somebody answers it. I think this is a basic ACD feature available on most PBX that support ACD functionality. Does anybody knows how to do it with asterisk?? Thanks in advance Pedro Nunes What we have done is to set up a single queue that all calls come into. For the agents that we want to be our Front Line (i.e.: Customer Service Reps), we give them a penalty of 0. Our Overflow group (i.e.: Customer service reps who are also dealing with walk-in customers and therefore should not be bothered unless we're really busy) gets a penalty of 1, and our Last Resort (i.e.: Everyone else) people get a penalty of 2. That way, all of the calls are answered by our front line people, unless they are all busy/unavailable. Then, and only then, the calls start going to our overflow people, and if they are also all unavailable, the calls go to our last resort people. Seeing as how we have more than 23 people between the three groups, there should technically be no waiting on hold in the queue, even with the PRI saturated. I don't know if this is what you are looking for, but it works extremely well for us. To whomever coded this feature, THANK YOU! To set this up, just edit the queues.conf file and add the penalty to each agent's member = line like this: ; Front-line - Penalty of 0 member = 100,0 ; Overflow - Penalty of 1 member = 101,1 ;Last Resort - Penalty of 2 member = 102,2 Hope that proves useful to someone Tom ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] AGI and set_callerid for number and name
Curse,
Look at this php script ...
Contactlookup.agi
#!/usr/local/bin/php -q
?php
ob_implicit_flush(true);
set_time_limit(6);
$in = fopen(php://stdin,r);
$stdlog = fopen(/var/log/asterisk/my_agi.log, w);
// toggle debugging output (more verbose)
$debug = true;
// Do function definitions before we start the main loop
function read() {
global $in, $debug, $stdlog;
$input = str_replace(\n, , fgets($in, 4096));
if ($debug) fputs($stdlog, read: $input\n);
return $input;
}
function errlog($line) {
global $err;
echo VERBOSE \$line\\n;
}
function write($line) {
global $debug, $stdlog;
if ($debug) fputs($stdlog, write: $line\n);
echo $line.\n;
}
// parse agi headers into array
while ($env=read()) {
$s = split(: ,$env);
$agi[str_replace(agi_,,$s[0])] = trim($s[1]);
if (($env == ) || ($env == \n)) {
break;
}
}
// main program
echo VERBOSE \Here we go!\ 2\n;
read();
$session = mssql_connect('mssql server' , 'username' , 'password' );
$result = mssql_query(select * from ContactDB WHERE
extension=.$agi['callerid'],$session );
$row = mssql_fetch_array($result);
mssql_close($session);
if ($row['Name'] == ){
write('SET VARIABLE NAME Not Found');
read();
} else {
write('SET VARIABLE NAME '.$row['Name'].'');
read();
}
fclose($in);
fclose($stdlog);
And in extensions.conf
[extensions]
exten = 4501,1,agi,contactlookup.agi
exten = 4501,2,SetCIDName(${Name})
exten = 4501,3,Dial(SIP/421,15)
It looks to an mssql DB, try to find the callerID number in table
extensions, and then sets a variable named Name to the value of
table Name. Cool hah...
Pedro Nunes
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Serge
Lhermitte
Sent: quarta-feira, 12 de Outubro de 2005 17:57
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] AGI and set_callerid for number and name
Hi,
I've been trying to use the set_callerid function in the AGI. It sets
the CallerIDname properly but I can't figure out how to set the
CallerIDnumber.
Is it at at possible ?
Cheers.
SL
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[Asterisk-Users] ACD/queues question
Hi there, Does anyone know how to setup an overflow queue? When a call rings on the queue A, if all agents were busy, the call goes to the queue B. If all agents in queue B were busy, then the call stays on both queues until somebody answers it. I think this is a basic ACD feature available on most PBX that support ACD functionality. Does anybody knows how to do it with asterisk?? Thanks in advance Pedro Nunes ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
