similar to: SPRINTF option : format %1$s not supported

Gentlemen, I have googled, searched the mailing list archives, and even spoke on the IRC channel, but have not found an answer to the following problem. I am attempting to retrieve multiple columns in an ODBC query using ARRAY per the solutions offered by many individuals. My dialplan code is as follows: exten => _.,n,Set(ARRAY(var1,var2,var3)=${ODBC_LOOKUP(${KEYVAL})}) exten =>

I'm trying to declare a few date-related global variables to ease my dialplan. When I declare the following in the [globals] context of extensions.conf, I get unexpected results: YEAR = ${STRFTIME(${EPOCH},,%Y)} MONTH = ${STRFTIME(${EPOCH},,%m)} DAY = ${STRFTIME(${EPOCH},,%d)} TIMESTAMP = ${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)} If I evaluate these variables in the dialplan later, using exten

Hello, I have been trying to completely disable music on hold on my asterisk system. When a call is put on hold I do not want any music on hold, but I would like the remote user to get informed of this event (depending on the technology e.g. with a SIP reinvite and an SDP indicating the call is on hold). I have searched and tried out various approaches, but when putting the call on hold

Hi, I've got a brand new Asterisk 11 setup for which I would like to keep the number of loaded modules to a minimum. My goal is to this setup in a pure SIP environment, for switching incoming calls to outgoing tSIP trunks. When I leave autoload=yes in /etc/asterisk/modules.conf, I can handle an incoming SIP call with a Playback app. When I leave autoload=no in /etc/asterisk/modules.conf, it

Hello, I'm trying to match a number in international format, like +49XXXX... The regexp string "^\+49" doesn't work. Both in $["+49..." : "^\+49"] and ${REGEX("^\+49" ${NUMBER})}. The error is: WARNING[12486]: func_strings.c:138 regex: Malformed input REGEX(): Invalid preceding regular expression. The regexp expression "^49\+" works.

Does the application PGSQL has been removed from Asterisk? Couldn't find it on Asterisk source and addons. Atenciosamente, Vin?cius Fontes Gerente de Seguran?a da Informa??o Canall Tecnologia em Comunica??es Passo Fundo - RS - Brasil +55 54 2104-7000 Information Security Manager Canall Tecnologia em Comunica??es Passo Fundo - RS - Brazil +55 54 2104-7000

Hi folks, Is there anywhere I can possibly get a model of the exam itself, maybe possible scenarios for the prac, etc? To people who have done the exam....any helpful hints ? Thanks,

Hi all, Here is the script i want to make - Caller call to a number to record a message - Asterisk answer and start recording message as following + User press * to start recording + Record is finished if: + User press # + OR message duration reach 60 second + Hangup How do you counting down 60s, and how to detect # (i make a test using Read() but it cant read #) Thanks in advance

I am using asterisk 1.6.1.6 and have been setting up a system to use a Postgresql database as the realtime DB via the ODBC route. I have got extensions and voicemail working but am having trouble with SIP The problem seems to be with using a schema. If I put the table "sip" in the schema "foo" then I add this entry to extconfig.conf sippeers => odbc,psqldb,foo.sip Restart

Hi All, Is there a way to change the mappings of disconnect reasons to certain SIP messages? E.G. I need to change the mapping for SIP 402 ?Payment Required? from 16 (normal termination) like it is in 1.4.24 to 21 (call rejected) as defined in RFC 3398. For me this is a big issue because my dial plan will look for alternative termination in the event of network error (e.g. reason 3 or 21 which is

Hoping someone can help me understand what is happening here; we start asterisk as a service at boot (actually, with heartbeat) on CentOS using the asterisk init script installed with "make config" upon reboot of the server (when the asterisk service is first started by heartbeat) we get color in the console when we connect to it using asterisk -r after the execution of

Hi, When I was testing an IVR, I realized I miss a function I would call GotoIfTimeWithOffset. Today, this IVR is using function AEL GotoIfTime in several places. The problem is if it's 11pm at the moment I'm testing this IVR, I can't nicely test the 9am or 2pm branch. GotoIfTimeWithOffset would get 2 incoming arguments : - the first is a time range (just like GotoIfTime), - the

I need to create 30 mins of GSM file for Asterisk . Silent / Blank file. Whats the best way to create it ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100305/b92821c0/attachment.htm

Hello Everyone, We are running asterisk 1.2.x version in production environment since last 5 year and we have no issue at all, But now time to upgrade. and i heard about 1.8 which has introduce many features. I am wondering should I use asterisk 1.8 in production ? or should I go with 1.4 or 1.6 stable version? I would like if you suggest me which version would be good for production since

Hello list, I have a file to be played in wav-format. I thought Asterisk would automatically take the wav-file and translate it to the codec used, but I see this : [Aug 28 11:16:29] WARNING[2705]: file.c:664 ast_openstream_full: File /var/lib/asterisk/sounds/vprompts/*zip-code.wav* does not exist in any format [Aug 28 11:16:29] WARNING[2705]: file.c:991 ast_streamfile: Unable to open

Hi all, I am having trouble getting cdr_adaptive_odbc to work. I have correctly configured the odbc drivers and dsn (I have tested this by connecting directly using isql). I have also configured /etc/asterisk/cdr_adaptive_odbc.conf like so: [test-asterisk] connection=test-asterisk-odbc table=cdr I have tested the ODBC connection test-asterisk-odbc and it works correctly However when I try to

Not sure if this is a bug yet, so I wanted to ask around to see if anyone else was having this issue. I have pollmailboxes=yes set in voicemail.conf but externnotify is not called. I know it isn't the externnotify script because if the changes are done in asterisk then it is called properly, if the changes are done via our webserver then it is not. Also, we use odbc voicemail storage. Thanks

"make menuconfig" does not show cdr_odbc as a selectable compile option. I have compiled and installed both unixODBC and freetds from source and have verified both successfully connect to my sql server. Both were installed to standard locations (/usr/lib). I had no problem compiling cdr_odbc on my test server(CentOS 4.6), however following the same steps on my production server (CentOS

It appears that there's no way to get the return value from a GOSUB into an AGI script. Is that correct?

Hi folks, How would I go about running a stored procedure call from asterisk via func_odbc. I'm after an example entry in func_odbc if possible for ast 1.4 Also, if someone could post an insert statement that actually works, would be nice. Thanks, :)