similar to: Confused about SIP Realtime Updates

Does anyone know if realtime extensions allows extensions in the format callerid/extension yet? ie the extensions.conf file allows you to do: 5551212/1000 => exten ... and it matches against extension 1000 when the caller id is 5551212. Last time I checked, realtime didn't support this yet. That's a show stopper for us. -------------- next part -------------- An HTML

It seems that when you use Realtime static and possibly realtime realtime for sip users, that Asterisk fails to create the regexten context for DUNDi. Someone else had the same problem back in July. Doesn't look like they ever had a resolution. <http://lists.digium.com/pipermail/asterisk-users/2006-July/160105.html>

We are hosting multiple companies with Asterisk. For a high degree of control, each company has many contexts that are included from a main context. I had wanted to use realtime, but realised very soon that it didn't scale. For each context that you put a realtime switch statement in, Asterisk has to go and query the database. If you include 10 contexts, and each one of those has a realtime

Is anyone using realtime sip for friends (ie phones) with multiple Asterisk boxes all pointing to the one central MySQL database? Does it work? Are phones that are registered to the database from Asterisk box able to reach phones registered to the database from another Asterisk box? Doug.

Boy, am I stuck... I'm officially ready to toss Asterisk out the window. I have to admit it isn't necessarily all the fault of Asterisk either. It just seems that every option I turn to suddenly ends in failure. I don't know if it's me that's bitten of more than I can chew with this project, or maybe Asterisk just isn't mature enough yet. Nothing complicated really....

Hi All, I'm running 1.2.9.1. I can prune sip realtime peers and users and iax realtime peers but no command to prune iax realtime users. Was this implemented in a later version? Thanks. JR -- JR Richardson Engineering for the Masses

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi guys, I've been working on a little dialplan fragment for roaming extensions, however the customer wants us to set the MWI indicator for the roaming extension that has just logged in. We're using MySQL realtime, so I've figured out that RealTimeUpdate will happily update the realtime database with the correct mailbox. My problem

I'd love to understand what the function of the peer list returned by 'sip show peers' is for, especially when Realtime is used. If I start Asterisk with realtime enabled, a 'sip show peers' yields none. As each peer (phone) registers, or a call is made to the peer, Asterisk adds them to the list returned by 'sip show peers'. Correct? Apparently Asterisk doesn't

Trying to get SIP realtime working here... I'm connected to the database... *CLI> realtime mysql status Connected to vox180internal@db1.ipt.XXX.com, port 3306 with username voxadmin for 6 seconds. I can get information for the extension in question... *CLI> realtime load sipusers name 2944093 Column Name Column Value

Has anyone considered the idea of splitting the sip registration information in a realtime database from the actual configuration of the peers? I mean, instead of having a table full of the configuration information (i.e. name, regexten, secret, etc) and registration information (i.e. ipaddr, fullcontact, etc), you have separate tables with their own information. This way, you can have separate

Hello, I am just asking this because I am note sure if the problem is on my side or not, I saw some comments on SIP realtime today so I was wondering, has anybody has SIP realtime working with a softfone ? If yes, please confirm, that would give me a light. My previous message to the list is below. Thanks. Frederic ----- Original Message ----- From: Frederic Jean To:

On Fri, 2006-12-08 at 04:26 -0700, Douglas Garstang wrote: > Hi Steve. > > Thanks, but unfortunately, I can't be involved in that. We are > running Asterisk in a production environment and we're using > 1.2, not 1.4. I don't have the resources to work with 1.4. > Last time I filed a bug against 1.2 I got told off. >

Hi, Is there something wrong with REALTIME (ARA) when used with rtcachefriends parameter? In my sip.conf (Asterisk 1.2.0): rtcachefriends=yes rtupdate=yes rtautoclear=yes Desired configuration is realtime configuration (via odbc) for SIP phones + MWI. Realtime means the following: when I make changes to db they should apply with no extra commands executed in CLI. In order to use MWI with

Oh heck. It really looks like realtime has been seriously screwed up. When a call comes in to Asterisk, I can see asterisk executing these queries. SELECT * FROM ast_sip_peers WHERE host = '2XX.YYY.142.205' SELECT * FROM ast_sip_peers WHERE name = '2944093' SELECT * FROM ast_sip_peers WHERE name = '2944093' So, the first thing it does is check and see if there are any

"Douglas Garstang" <dgarstang@oneeighty.com> wrote: >Boy, am I stuck... > >I'm officially ready to toss Asterisk out the window. I have to admit it isn't necessarily all the fault of Asterisk either. It just seems that every option I turn to suddenly ends in failure. I don't know if it's me that's bitten of more than I can chew with this project, or

I am using realtime on two boxes, one running 1.4.10.1 and one running 1.4.11. Everything works fine except for when I make a database change, such as a phones password. I change the DB, I prune the peer, I see it is gone and then I see it show up again in "sip show peer xxxx", but everything is not being updated. The phone will not register even though the DB and the phone have

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

I've been using realtime for sip users information. I noticed that when you are doing this, if you do a 'reload' or restart asterisk, the information in a 'sip show peers' goes away. When I do this, MWI stops working. I always though MWI used the astdb file ('database show') to determine where to send MWI but it must be using 'sip show peers' because when this

Hi, I'm running asterisk realtime, i had prob when a user does not unregister properly. I tested with SPA942 and a PAP2, when phone is registered, i call using the SPA using x-lite no problem, but when i unplugged the power, it does not unregister properly, so asterisk think SPA942 is still registered, when i call using x-lite, asterisk tries to call it.so it gets stuck at [Aug 11

And after reload ALL your phones are unreachable for 2 minutes! Imagine you have several thousands devices unreachable for 2 minutes. How much calls will fail during that time? Regards, Mindaugas Kezys Kolmisoft UAB=20 VoIP Billing Solutions e-mail: info at kolmisoft.com URL: http://www.kolmisoft.com -----Original Message----- From: asterisk-users-bounces at lists.digium.com =