similar to: RealTime Configuration for extensions.conf

Hi All, Had anyone got an idea of there exists an LDAP backend for SIP and IAX? I've read that there is a patch for LDAP realtime, but I hadn't seen any type of relevant configuration information. Any information on the above would be highly appreciated. Regards, Nir S -------------- next part -------------- An HTML attachment was scrubbed... URL:

hi, i'm a bit puzzled because i do not get include and hint to work with the new realtime enginge (cvs-head from 2004-12-09). other things (sipfriends and "normal" extensions) work perfect with the realtime engine. the entries in the static extensions.conf file i used before where: exten => 183,hint,SIP/snom220 exten => 183,1,Macro(stdexten,443,SIP/snom220,183) exten =>

Howdy, My first forray into using res_mysql.conf for realtime access of sip users and extensions. I have the following relevant section of extensions.conf: --- [trunklocal] exten => _NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}}) [local] include => trunklocal include => trunktollfree [longdistance] include => local include => trunkld [international] include

As seen in the following URL: http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions and as I also tested some time ago with an old release of Asterisk, RealTime Extensions didn't support the Ex-Girlfriend syntax. Is it already working in recent 1.4 or 1.2.15 releases? Is there any other way that I can use to do the same thing but only using contexts, for example? If yes, please

Hi, Is it possible to filter the calling user with the usage of mysql realtime the same as it is done in extensions.conf file: exten => some_exten/calling-user.... is there some flag which activates this extra check?? Cheers Tomasz

Hi All, I'm testing different ways to implement a LCR/OCN tabe to shift calls to multiple carriers for better rates. I'm using realtime mysql database access, asterisk 1.2.9.1 with mysql 3.23. Scenario 1: I send outgoing calls with with a Goto statement into the context with the realtime switch to dip into the mysql lcr database, currently I have ~13K records/routes. The database

Hi to all the ML. I'm new here. I start to use asterisk with realtime configuration, with pgsql backend connected via odbc. The connection between asterisk and pgsql works fine. I create a table sip_conf with 2 user (for testing purpose), 1401 and 1501. Those are the records: asterisk=> SELECT name,host,type,context,secret,defaultuser from sip_conf; name | host | type | context |

Is it possible to use the "include" command to include other contexts if you are using realtime for extensions? I've searched voip-info and some people were asking about it, but I didn't find a real answer anywhere.

One of the frequently asked-for features in pbx_realtime is the ability not to have to have an extensions.conf, because you want realtime to auto-register its contexts. There is now such a patch out there, for testing. The faster that people test it and give feedback, the sooner it can make its way into Asterisk. http://bugs.digium.com/view.php?id=14158 -- Tilghman

Can realtime extensions be used within AEL contexts? I've tried without success. Also did some Googling and looking around on the Wiki, but couldn't find much. Thanks, Mike Clark

Hi, I need to make use of BLF feature on Aastra 6757i phones but its an Asterisk 1.4 using realtime architecture. Extensions are defined in realtime database and dial plan is in AEL. I am able to correctly setup hints in the dialplan, but they don't work. Did some research and found out that hints don't work work with realtime extensions. Is there any work around? On voip-info I read

Is the template capability in sip.conf compatible with realtime sip.conf entries such as users in a database? I notice that contrib/realtime/mysql/sippeers.sql and the wiki page don't mention a template column: https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure while some third-party examples do suggest that a column named "template" is permitted:

Hi - I'm looking into realtime and I'm having a bit of a problem with the SIP part. My review of the posts seems to indicate that I should use realtime static for the [general] part of my sip.conf including the registration commands: register=><did>:<secret>@<domain>/<did context> and use realtime realtime (funny name!) for peers and friends: [myprovider]

Hi, Case1: --------- --> extensions.conf exten => 1023,1,Voicemail(101) exten => 1023/101,1,MeetMe(200) Case2: --------- -> extensions table (using realtime extensions) +----+---------+----------+--------+----------+---------+ | id | context | exten |priority| app | appdata | +----+---------+----------+--------+----------+---------+ | 29 | default | 1023 | 1 |

I have the following setup: - UAs registered with SER/OpenSER - SIP peers (non cached), extensions, voicemail setup (not message storage) defined in Asterisk 1.2 using Realtime When a message is left in the user's mailbox, no Notify message is sent to SER. 1. If the SIP peer is defined in sip.conf with a host=ser.domain.com then the notfy is sent to SER. 2. If realtimecache=yes is set in

I currently have my phones setup in the sip.conf file. I use templates to describe the specific settings to each phone type. For instance in sip.conf, I have: [generic_phone](!) ... ... [polycom501](!,generic_phone) ... ... [grandstream](!,generic_phone) ... ... ;;;;;begin subscribers [200](polycom501) ... ... [201](grandstream) ... ... I am using asterisk 1.4.21.2 I would like to move my

hi in ael the [general] equivalent is globals { } ? what is the equivalent in realtime??? can i use realtime and static ael or conf??? using realtime when i do sip show peers asterisk didnt show the realtime peers what can i do? Thanks David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your (")_(")signature to help him gain world domination. -------------- next part

Hi - I'm looking into realtime and I'm having a bit of a problem with the SIP part. My review of the posts seems to indicate that I should use realtime static for the [general] part of my sip.conf including the registration commands: register=><did>:<secret>@<domain>/<did context> and use realtime realtime (funny name!) for peers and friends: [myprovider]

Hello list, scrambling different pieces of info together I've come with the following : I want to have my "register =>" statements in a MySQL-database, so I've made the following table. table ast_config : id 1 cat_metric 0 var_metric 0 commented 0 filename sip.conf category general var_name register var_val username:password at sip.provider.net In ext_config

I am currently in the process of testing out realtime support for sip.conf. I have followed all of the directions that are listed in the Wiki, but for some reason this does not work. When utilizing a flat file, I am able to register endpoints without any problems, and calls can proceed. One interesting side effect that I have noticed is that when I am using realtime for sip, I am unable to see