similar to: SIP registration fails with realtime

Hi, I have installed asterisk real time and sip buddies information is being stored in DB. Now I have a question, Asterisk Realtime Server -A Third party SIP server-B Question: Is there any configuration in * RT that it can register with defined sip user on Server-B I was only able to find sip users information in DB not about user registration on other server. -ag -------------- next part

I have read and followed as much as I can find but I am missing something. What I want to do is get as much as I can running from mysql and keep the *.conf files for static things. So I have setup a SIP users/peers table in a mysql database and I have populated it with a few peers. I have configured asterisk addons and from the asterisk CLI I am able to search the sip users / peers tables using

Hi! I use the following configuration to register my asterisk server to my SIP provider: register => 12345:passwd@sip.provider.com/12345 sip.conf: [sipout-test] type=peer username=12345 fromuser=12345 fromdomain=provider.com secret=passwd insecure=very host=sip.provider.com qualify=yes context=test-incoming extensions.conf: exten => 12345,1,Dial(SIP/10) exten =>

Hello list, I'm using asterisk 1.6.2.11 with realtime SIP (mysql DB). I notice that when the SIP peer registers, the fields 'fullcontact', 'ipaddr', 'port', 'regserver', 'regseconds', 'lastms' are filled with values. But after a while, these fields become empty. Asterisk CLI shows : asterisk*CLI> sip show peers Name/username

Apparently, the realtime system in asterisk is faulty. Implementing realtime, begins a host of seeding messages along with registration messages visible at the CLI prompt. This is not the case with .conf file configuration Unfortunately, it is not clear where the bug originates but is shows it's head while calling the register_verify function, (which there are 2 one in chan_sip.c and one in

Hello there I'd like to use sip users and peers realtime. I think I done all I need to get asterisk works fine in realtime: res_odbc.conf configuration. extconfig.conf sippeers => odbc,asterisk,sipclient sipusers => odbc,asterisk,sipclient sip.conf [general] rtcachefriends=yes The sipclient table as suggest in this article: SIP Realtime, MySQL table structure (

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

Hi all, We are looking for information about Dynamic Realtime Asterisk, We have install some Snom phone 360 (SIP) for our customer , but we have a problem when we want to register 2 accounts on the same phone and on the same Asterisk PBX. The problem when we register two phone line in realtime it doesn't work, we can't make a call the registration failed when we place a call. Can

Dear All, I am currently working on asterisk cvs-head version in order to use realtime with mysql, 2 asterisk servers with duplicate mysql databases, one asterisk server is serving the sip phones and the data is logged to the database and replicated to the other asterisk database, when the first server fails though it has the sip phones data in it's database the sip phones need to re-register

When I do a "sip show peer <peer> load" command in the Asterisk CLI I get the information about the peer I requested, however, there is a line that says "Realtime peer: No". All the other information is correct. According to "help sip show peer" the "Option "load" forces lookup of peer in realtime storage.". Also, this particular peer is

-----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

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]

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:

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 - 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 everybody, I am having a problem with realtime SIP peers. On Asterisk 1.8, I had SIP peers for external SIP providers configured in database and additional register lines in sip.conf so they would register. Now I upgraded to Asterisk 11.3.0, partly because of the promised callbackextension feature for realtime peers (https://reviewboard.asterisk.org/r/1717/). Removed the 'register'

I'm confused about a few things relating to realtime, SIP and config in general. As I understand it, with the exception of extensions.conf, I can either have a config file completely in text or completely in a database. Is that correct? I can't find documentation for exactly what "switch =>" does but is that only in the dialplan and a way to have it partly from a file and

I have a problem using asterisk 1.6 with realtime sip. When I add sip channel (my sip provider) to asterisk using realtime sip (http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip), incoming calls don't work for me. In asterisk CLI I get message: NOTICE[19805]: chan_sip.c:21250 handle_request_invite: Sending fake auth rejection for device "test" <sip:test at

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.

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