similar to: Hint for realtime peers

Hello, I noticed that changes to realtime sip peers are not applied until a 'reload'. A 'sip reload' does not make any changes to realtime sip peers. When changing for instance the mailbox-parameter in the realtime sip_buddies table, the change is not applied with a 'sip reload'. For every change there is a complete 'reload' necessary. Why does a 'sip

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

I'm getting the following message in my full log at startup and my realtime sip peers aren't being found. My realtime extensions have no errors. The table sippeers exists in the database. Is this a known problem? res_config_mysql.c: Table sippeers not found in database. This table should exist if you're using realtime.

Starting with Asterisk 1.2 I have always used realtime static to load my extensions.conf into Asterisk. It worked perfectly up to version 1.6.0.X but starting from 1.6.1.X and upwards it simply does nothing. I can see that the extensions.conf file is mapped to the database: == Parsing '/etc/asterisk/asterisk.conf': == Found == Parsing '/etc/asterisk/extconfig.conf': ==

Hi All, I'm testing some realtime extension apps with Asterisk 1.4.28 and addons 1.4.10 using res_mysql. Localhost database is 5.0.32 with Debian Etch. The apps are working fine all syntax is proper, using Set with (REALTIME) function, Set with (CUT) function, calling a Macro with s extensions, and using a few pattern matching extensions as well. I can certainly detail all database rows if

Hello. >> ubuntu*CLI> sip show peers >> Name/username Host Dyn Nat ACL Port Status >> voipprovider xxx.xxx.xxx.xxx A 5060 Unmonitored I've ben trying to connect an asterisk server to a voip provider, and I'm currently wondering what the 'A' in the ACL field of the 'sip show peers' command might

Hi List, I have a problem with Asterisk 1.6.1.6 realtime (MySQL databases hosted on a separate machine). When Asterisk is in verbose mode, it prints messages saying "failed to extend from 512 to 664" (quite a few lines in a block) and then the last message is mostly "failed to extend from 512 to 663". The number of lines varies unpredictably. The full message (in the logs)

I am using Asterisk 1.4.36 with Realtime Voicemail from a MySQL database, and whilst I have it all working, I am unable to find a way to customize the content of the email that gets sent to a user when they receive a voicemail. In the past I just edited it in the voicemail.conf file and made the customizations in there, but now that I am using Realtime voicemail from MySQL, my voicemail.conf file

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,

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 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

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

Hi Friends, i have installed and configure asterisk-1.8.0. When i have tried asterisk start get below errors and not able to start asterisk. *FD 32767 exceeds the maximum size of ast_fdset!* Thanks in advance. -- Best Regards, Rajnikant Vanza -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi guys, Having issue with getting CDR to write to MS-SQL via ODBC. > cdr_odbc: Connected to freetds-connector > cdr_odbc: Error in PREPARE -1 > cdr_odbc: Query FAILED Call not logged! == Spawn extension (cisco, ##########, 2) exited non-zero on 'IAX2/astYYYY-507 Isql test: [xxx at YYYY asterisk]# isql freetds-connector XXXXXXX YYYYYYYYY

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

On Wed, Jan 20, 2010 at 2:42 PM, JR Richardson <jmr.richardson at gmail.com> wrote: > Hi All, > > I'm using Asterisk 1.4 branch and checking the status of some SIP > Peers with the functions ${SIPPEER(101:status)} and the result is "OK > (48 ms)". ?Seems to work fine. > > Now I would like to use the function CUT to set a variable with the > 'OK'

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

Hello, In asterisk-addons-1.6.1.2's changelog, you can see that 6 changes were committed between versions 1.6.1.1 and 1.6.1.2. But if I'm not mistaken, you cannot read anything there about Asterisk to Asterisk-addons compatibility. What is the rule for Asterisk to Asterisk-addons compatibility ? Is this rule implicit ("any Asterisk-addons 1.6.1.X is compatible with any Asterisk

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