similar to: realtime versus static

Ok I'm a little confused about realtime static. The wiki has the database schema but no explanation of what fields are for what. I would appreciate if someone can confirm or deny how I think the schema works. cat_metric = sort order for category var_metric = sort order for vars filename = equivalent .conf file in /etc/asterisk category = anything that would be enclosed in braces ([])

I am trying to find where the authoritative sql definitions for Asterisk Realtime Architecture ARA are located. I have found many locations but each and everyone seems to be different. http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html/asterisk-book.html http://www.open-voip.org/index.php?title=Asterisk_Full_RealTime_example Files included with the distribution:

I've added an introduction article about the ARA on my web site http://www.voip-forum.com/ The same text is now also added to CVS head as README.realtime. On the same site, you will also find the news item about how we used Asterisk for a call from an airline jet above Greenland to Stockholm, Sweden. The world is getting smaller and more connected every day! /Olle

Hi All, I'm upgrading some PBX's from 1.2 to 1.4 and having a bit of trouble with converting the Realtime application to the REALTIME function. I have the method down and understand simplistically what is going on, at least enough to get my old 1.2 apps to run in 1.4 functions. I do not understand why change from the app to the func? What the benefits? To me, the app seemed so

Hello list, what is the use of realtime SIP peers when you always need to reload the sip configuration as if you were just putting your SIP peers in sip.conf ?? My SIP peers are now defined in a mysql-DB and when I add a mailbox in the field 'mailbox', the change is not active untill a do a "sip reload" or a "module reload chan_sip.so". Doing a "sip

On the wiki it say's that if you use the Goto commands you need to replace ',' with '|' in the app data field. But in the examples it uses '|' in place of ',' in the Dial command also in a couple of places. Is it safe to replace ',' with '|' everywhere in the app data field when using realtime? Or should I still to substituting ','

Does realtime work reliably on Asterisk 1.2.24? Are there any definitive guides, I can only find bits and pieces here and there. Any accurate howtos would be of great help. I am missing func_realtime.so. Where does this file come from? Asterisk or asterisk-addons? I saw in one of the howtos that it is needed. Is it needed for 1.2.X or 1.4.X. Also, what about the switch lines in the

Trying to use realtime sip for the first time, and it's not working as expected. I have one user entry in the sip database. Everthing else is still in sip.conf. When I get an incoming call, this is the database query: SELECT * FROM ast_home_sip_realtime WHERE name = '+18003859169' The 800# is the caller id of the caller, which doesnt' make any sense to me. Is there any

Hello everybody, I'd like to hear some success stories about the use of Asterisk Realtime in medium-large contexts, like > 50 extensions. Don't you think that in those contexts the system could be overloaded from the excessive number of queries to the DB? So.. is anybody using ARA in those kind of deployments? Thanks in advance, -- Andrea Spadaccini Multimedia Technologies

Anyone had Realtime working with IAX2 and RSA authentication to connect two PBXs, please? It seems that inkeys/outkey fields are not read at all and the following warning is logged when dialing: Jun 2 02:41:36 WARNING[6299] chan_iax2.c: I don't know how to authenticate ******** to XXX.XXX.XXX.XXX Using iax.conf it perfectly works. Maybe a bug in Realtime? TIA, Alex

Hi guys We're using asterisk 1.8.23.1 on CentOS 5 and are trying to create a trunk to MS Lync server. If I create the peer in sip.conf the trunk connects with no problem. However, we prefer to use ARA. Whenever we define the peer in our peers table, the trunk does not work, even if we use sip show peer <peer-name> load. Has anyone got any experience of connecting to Lync using ARA?

Hi, list. We have a problem with loss peers after 'sip reload', our configuration: Asterisk 11.6-cert1, SIP realtime peers, sip.conf: - rtcachefriends=yes - rtsavesysname=yes - rtupdate=yes - rtautoclear=yes When we do 'sip reload' , peers are removing from available. Before `sip reload` : srv-pbx2*CLI> sip show peers Name/username Host

Way back in the mists of time, I built my asterisk installation with SNMP support. It's a bit tedious to get the sub-agent for snmpd set up but once you have it you can poll the OID for the asterisk sub-agent and it will tell you how many calls are up at that moment in time. That said, I actually prefer ARA/ARI to flat file configuration of endpoints and dialplans.  Changes are more or less

Hello Everyone, The documentation suggests using unixodbc for asterisk realtime. Is there any way we can just use native database clients such as libmysqlclient from MySQL? The native clients tend to be more up-to-date. Thanks in Advance, Nick.

Sunny wrote: > Hello, > > > In Asterisk 11 it is possible to set extensions on DB table (sipppers) > and also in sip.conf. > > But in Asterisk 13 apparently this is not possible: as I tried to set in > ps_endpoints and also in pjsip.conf but only the realtime endpoints are > loaded. > > Is there a way to use realtime + static modes at the same time for the >

I was looking for MySQL table structures for ARA (Asterisk 1.8.X). I found one for SIP friends on, https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime,+MySQL+table+structure But it seems that it is not as per the Asterisk 1.8 SIP options. i.e. it contains 'call-limit' which is deprecated in 1.8 and not the 'callcounter' as one of the fields. Pardon my ignorance, but are

I just upgraded from 1.6.0.14 to 1.6.1.6 and now my realtime static configuration for extensions.conf will not load. All other realtime configs work (SIP, IAX2, Voicemail). I cannot find any reference or documentation about the structure of the realtime static database for 1.6.1.x but I have used the same table structure since 1.4.x. CREATE TABLE `ast_config` ( `id` int(11) NOT NULL

Hi All I am trying to get voicemail switched over to ARA on version 13 and notice that the password is not stored in the db when it is changed. A little hair pulling and playing around and I think the problem is in the function ast_update2_realtime in main/config.c. Issued source is ==> int ast_update2_realtime(const char *family, ...) { RAII_VAR(struct ast_variable *,

Hi list, still wondering for a couple of days how to handle context includes in realtime architecture i've tried to patch my pbx_realtime.c with a patch on the digium issue tracker (http://bugs.digium.com/view.php?id=6014) but it does not seem to work or may be i'm using the wrong way does anyone has a solution???? thanks for reply BR -- Cheikhou DIAW -------------- next part

On 2 February 2015 at 15:12, Joshua Colp <jcolp at digium.com> wrote: > Sunny wrote: > >> Hello, >> >> >> In Asterisk 11 it is possible to set extensions on DB table (sipppers) >> and also in sip.conf. >> >> But in Asterisk 13 apparently this is not possible: as I tried to set in >> ps_endpoints and also in pjsip.conf but only the