similar to: app_directory broken in 1.6

I've checked the wiki, searched the mailing list, Mantis (bug tracker), and looked through the source code, but found not mention of this issue: If using ODBC Storage for voicemail messages, name-playback by app_directory breaks. app_directory has no ODBC Storage code and therefore only looks for the greet.wav file (recording of user's name) in the filesystem, whereas app_voicemail saves

Hi Everyone, Sorry in advance if this not the correct place to ask this question, feel free to point me somewhere more appropriate to ask. We have an Asterisk 1.2.7.1 server (about a year old version of Asterisk @ Home with FreePBX) running the phone system for our small office (roughly 15 extensions). I'm trying to hide a couple of extensions from the app_directory generated

Hello, Using 1.6.2-rc5, my settings include: [local-phone](!) context=mylocal type=friend nat=no canreinvite=no host=dynamic qualify=yes dtmf=info language=fr call-limit=5 subscribecontext=subs disallow=all allow=alaw t38pt_udptl=no setvar=accountcode=foo [168](local-phone) defaultuser=168 secret=pass168 callerid=John Doe<168>

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 list, I hope someone could help me. I've started using Asterisk 1.6.0.14 to get queue logs in real time with odbc (our databases are all PostgreSQL) but it's not working. However, cdr odbc is working well. When asterisk starts next message appears: WARNING[4217] config.c: Realtime mapping for 'queue_log' found to engine 'odbc', but the engine is not available My

Digium has discontinued their ATA iaxy adapter; don't blame them, too expensive so they can not compete. The adapter is upgraded automaticaly when it is connected to new asterisk version; since this adapter is discontinued will it still work with asterisk 1.6 and beyond or will it be\ just a "door stopper"? -- Joseph

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

How to enable cdr_mysql.conf in Asterisk 1.6? I have installed asterisk-addons which compiled mysql support, "module show" is showing "cdr_addon_mysql.so" but cdr_mysql.conf was not created in /asterisk directory Is there any configuration file to enable mysql support? Comping cdr_mysql.conf from previous installation does not do anything, calls aren't recorded. --

Hi, I'm having a bit of a problem with storing voicemail messages in an odbc database. I *think* I've got everything configured correctly but messages are stored on the asterisk server instread of in the database. System info 64 bit redhat RHEL 5.1 Asterisk 1.4.26 unixODBC installed used makemenuselect to instal res_odbc and cdr_odbc Back end database DB2 Database name voiceml

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

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

Hi All, I'm running 1.2.9.1 stable. I'm wondering has this patch been applied to stable release or is it still only in CVS. Will this file patch apply correctly to 1.2.9.1 stable? Which file do I patch? I'm guessing app_directory_realtime_1.6.1.patch <http://bugs.digium.com/file_download.php?file_id=4915&type=bug> and config.h.patch

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

Hi. I'm trying to use the Directory app. I was wondering how can I enter the first 3 letters from the phone keypad... I need a keypad with also the ABCD tones? or a standard one is ok? Doing some tests I never been able to select an entry till now... (using the demo voicemail.conf in asterisk) Thanks for any help. -- Brancaleoni Matteo <mbrancaleoni@espia.it> Espia - Emmgi Srl

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

Dr Murrell and others, It seems grid.ls() fails after any use of grid.remove(). It gives an infinite recursion error even in the simplest cases, and no matter what arguments are passed to grid.ls. > library(grid) > grid.newpage() > grid.lines(name="foo") > grid.ls() foo > grid.remove("foo") > grid.ls() Error: evaluation nested too deeply: infinite recursion

Hi Folks, Are all the astricon presentations up? I'm especially after the one that tilghman did. I caught the tail end of the prez when I decided to skip the session I was attending and go for that one. :)