similar to: Realtime patch

I am using Realtime on Asterisk 11.5 with a SQLite3 backend. While everything seems to be working fine I keep getting this error on my log files: [2014-02-17 19:55:18] WARNING[20569] res_config_sqlite3.c: Could not execute 'UPDATE "sip_buddies" SET "ipaddr" = '192.168.2.23', "port" = '5060', "regseconds" = '1392692118',

Hello all, I've been testing a Kamailio Asterisk Realtime integration, and found a strange situation. My problem is that when using the integration, everything seems ok but Asterisk does not see the clients as registered. Kamailio and the clients report registered clients. Also calls fail. In Asterisk cli sip show peers shows nothing but for example realtime load sipusers name 660 shows the

Hello I change this from chan_sip.conf (see ipsvr): static void realtime_update_peer(const char *peername, struct sockaddr_in *sin, const char *username, const char *fullcontact, int expirey) { char port[10]; char ipaddr[20]; char regseconds[20]; time_t nowtime; -> char ipsvr[20]; time(&nowtime); nowtime += expirey;

Hi All, I am trying to configure asterisk realtime. But i am unable to get the extensions listed successfully when i type "sip show peers" in the asterisk CLI . i am unable to see any failure logs when i do a reload i can able to connect to the data source through "odbc show" in the CLI, Any hep in this regard is highly appreciated. Following is the configuration

When I try to use 1.6.1.1 with ODBC and MySQL, I get these: [Jun 19 17:19:22] WARNING[5882] res_config_odbc.c: Realtime table supporten_sip at asterisk: column type (-9) unrecognized for column 'name' [Jun 19 17:19:22] WARNING[5882] res_config_odbc.c: Realtime table supporten_sip at asterisk: column type (-9) unrecognized for column 'ipaddr' [Jun 19 17:19:22] WARNING[5882]

Hello, I have a problem with a call between 2 webrtc clients. Asterisk removes the ice-related lines from the sdp when it sends the INVITE out, and the called webrtc client rejects the INVITE due to the missing ice lines. Both webrtc clients are defined exactly the same way, same values in all fields except the number of the peer. There's probably something I've changed that causes this

Hi, I am using Realtime for sip configuration. When there is an INVITE which arrives at asterisk asterisk makes the following selects: Nov 20 10:37:10] DEBUG[31852]: res_config_mysql.c:651 mysql_reconnect: MySQL RealTime: Everything is fine. [Nov 20 10:37:10] DEBUG[31852]: res_config_mysql.c:138 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_conf WHERE name = 'tzl' [Nov

Hi guys, I'm trying to use Asterisk with LDAP integration. I created some schemas and it seems to work fine for sip.conf replacement. When I try to register a softphone to test the service, it seems ok from the softphone point of view (user registred) but when I do a "sip show peers", no one is registered (nor sip show subrscriptions, users...) I put my Asterisk on full debug and I

Hi, I'm wondering - why SIP device state doesn't get updated to anything else, except Not In Use. For queue call (with Local channel) i get: app_queue.c: Device 'SIP/21168' changed to state '1' (Not in use) app_queue.c: Device 'SIP/21168' changed to state '1' (Not in use) app_queue.c: The device state of this queue member, Agent/21168, is still 'Not in

Guys, I'm having a hellava time getting realtime to work, focused on sipusers right now, followed the wiki and other examples but still no luck. Using mysql on a seperate server, asterisk actually sees the database and can poll the table "realtime load sipusers" at the cli but asterisk realtime engine is no pulling the user info. I'm using 1.2.4 stable and have the database

Hi all, From the web, I can find a table scheme of sipusers for ARA using. However, I can't find any meaning of each field, especially for the field regserver which is new in the table. Can any tell me more detail about the usage of each field? CREATE TABLE `sip_buddies` ( `id` int(11) NOT NULL auto_increment, `name` varchar(80) NOT NULL default '', `host` varchar(31) NOT NULL

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

Hi everyone, in ldap realtime sip peers i need "fullcontact" set to "sip:exten@userip" for asterisk to correctly match the peers (at least for the natted peers to reach them)... anyway, how do I populate fullcontact "on the fly" with information from exten and userip? of course, i could just do it staticaly on ldap but since the info is already there why not

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:

Hi everybody, First of all i am sorry my English :) i want to configure my asterisk server as a sip server that stores sip users in the mysql database connecting directly over odbc driver. My odbc configuration works as below [root at ao042 asterisk]# isql -v asterisk +---------------------------------------+ | Connected! | | | |

I'm trying to get sipusers working with a realtime odbc database on Asterisk 1.6. We have sippeers working from the database, but need sipusers to be in a separate table for other implementation reasons. sip show user test load returns results from the database. CLI> sip show user test load * Name : test Secret : <Set> MD5Secret : <Not set>

I have configured asterisk realtime to work with two servers and a seperate MySQL DB. Each sip client registers which server it is connected to in the MySQL DB. This works great as long as the clients are 1. On the same network 2. Behind a NAT and connected to the same asterisk server as the caller. However I need this configuration to work for "NAT-ed" clients on different asterisk

Hi guys, im using realtime and I want to show registered users or online users on a webpage and offline users. Im taking regseconds field to make this happend If regseconds value is 0 then user appers offline, it regseconds is something else then its online, but sometimes this works and sometimes it does not. Im using the following options rtcachefriends=yes rtnoupdate=yes

Hello, I installed 1.2.5 and realtime SIP. The connection to the DB is OK because I can get the values from the CLI. Here are my 3 different cases: 1- If I put an unexisting user, I get 404 and I am not able to dial. 2- If I check "Disable registration" within Firefly it does not register but I am able to dial a destination (...) 3- If I leave registration ON, I get the 404 message

Hi All, Has anyone else noticed that when a sip phone sitting behind a nat registers to asterisk using realtime database, the private IP of the phone is put into the fullcontact field instead of the public contact IP. The database has the correct public IP in the ipaddr field and correct port number in the port field, which is actually what asterisk uses to to contact the device. This