similar to: 1.6.2.11 realtime sip registrations disappear from DB

Hello, I'm trying to get calls working between websocket clients and sip clients. For clients I have sip.js based clients on chrome, Zoipers and a Grandstream phone. Challenge here is I'd like to have Kamailio and rtpengine to handle the bridging between different rtp profiles but Asterisk changes them in the sdp bodies along the way. I'm using Asterisk 11.11.0. Is there a way to

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 noticed a strange thing while testing my Asterisk-Kamailio Realtime setup. In an INVITE the From and To headers contain the same number when calling through a Realtime integration setup. This happens when the INVITE leaves Asterisk. Can you guys tell me what might be causing this? I have 660 at testers.com as a websocket client and 700 at testers.com (caller) using a Zoiper client (db

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

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',

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

Greetings. Attempting to connect Asterisk to LDAP database using res_config_ldap module. While trying to register sip client (Ekiga softphone), according to slapd.log, asterisk connects to LDAP server, asks for some attributes to modify (they do exist, and asterisk user has all permissions to do that, etc). And then asterisk application just crashes. Without ldap (using just static users'

Greetings, I've noticed a problem that might originate from my Asterisk configuration, could use a hand in sorting it out. Problem is a 488 response from Asterisk whenever it gets RTP/SAVPF profile in the SDP. My current setup has Asterisk Kamailio realtime integration, and Kamailio uses dispatcher to route calls for Asterisk to handle. Now I have only one Asterisk, on the same machine as

Hello list, using Asterisk 1.4.30. I want to set the SIP-header Remote-Party-ID to display the name of the calling party on my phone in stead of the number. This is the dialplan : exten => 10,1,NoOp() exten => 10,n,SIPAddHeader(Remote-Party-ID: "eric" <sip:10 at 192.168.1.150>;party=called ) exten => 10,n,Dial(SIP/test2) This is what the CLI shows : /[Jul 12

We have several Asterisk servers (1.6.2.15) all configured for Realtime, all backed by the same database. The Asterisk servers are all listed under DNS SRV records, and SIP ATAs find us this way. Normally, no matter which Asterisk server an ATA connects to, we get our database fields filled out correctly, such as "regseconds", "lastms", "ipadr", etc. However, with

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

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

Why pay 1.1 cent's a minute for interconnecting to another Asterisk server for a high volume call center. Do people really understand what they are trying to sale and take an honest look into what they advertise. As a high volume user like a call center I would not connect my Asterisk Box to there Asterisk Box to a third Sip provider who then hands of to the Level 3 and so forth. With LD

Hi all, I have a problem with using remote MySQL database server with two Asterisk (1.4.17) servers. PhoneA registers with Asterisk#1 using realtime into MySQL on remote server and everything is working fine and when I call Phone A from Phone B (also registered with Asterisk#1) call is established. Problem is when I call PhoneA (which is registered with Asterisk#1) from PhoneC (which is

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

Hello list, asterisk 1.4.30 2 situations in which call-limit should work, but it does not : [Jul 8 09:15:49] WARNING[11132]: app_queue.c:3272 try_calling: The device state of this queue member, test12, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings. In sip.conf I have : limitonpeer = yes In my realtime sip_buddies

Hello, I'd appreciate your comments on the following problem I'm having, please forgive me if this is something obvious, I've been scratching my head on this for a while: I have Asterisk+Kamailio setup where I'm currently testing inbound calls from outside. I have both webrtc and sip clients, where webrtc peers are defined according to sip.js instructions (

All, I have multiple Asterisk servers in various locations running various 1.4 and 1.6 versions (lab and production) and am having trouble with a new Aastra 6739i (3.0.1.2015) registering. Below is my request to support and they have looked it over and don't see anything wrong: Support, Can not get a 6739i to register with 3 different Asterisk servers with varying configurations/versions

Problems with acl in dovecot-1.2.15 was not solved! See my configs below. And I see the new bug - still something wrong with configs: cat dovecot-acl: user=t1 lrwsti #user=oper-olegs lrwsti #user=oper-antona lrwsti user=operdss lr #user=operdss lrwsti User t1 can't create subfolers in inbox. He can only create subfolders in the first levels: Inbox test3 test4 Outbox Sent Trash Test1 - test

Hello, Here is how I do this. The prerequisits are: - MySQL to hold the extensions realtime database. MySQL is synchronized among all servers using the Master/slave replication model. - The phones are spread by some external algorithm over the Asterisk servers (statefull load balancer, statically defined in the config file of the phone, etc.). The idea is to locate on which server the