Hi All I have configured asterisk with the addons and setup my config files so that i can pull sip extensions (phones) from a mysql database. I have followed all the docs and have editted my extconfig.conf res_mysql.conf and sip.conf to contain all that is advised.>From the CLI i can see realtime has a connection and is able to load the user but when I plug in the voip phone it fails to register. My database content matches that of other phones in the sip.conf. If i remove the database user and add it direct into sip.conf the phone connects fine.Any help would be appreciated as its driving me mad now Very Happy CLI> realtime mysql status Connected to asterisk@192.168.1.26, port 3306 with username scott for 22 minutes, 47 seconds. CLI> realtime load sipusers name 114 Column Name Column Value -------------------- -------------------- id 1 name 114 callerid 114 canreinvite yes context default defaultip 192.168.10.136 dtmfmode info fromuser 114 fullcontact 114 host 192.168.10.136 nat no secret 114 type friend username 114 disallow all allow g729 allow ilbc allow gsm allow ulaw allow alaw regseconds 0 cancallforward yes Nov 21 12:52:55 NOTICE[15585]: chan_sip.c:10793 handle_request_register: Registration from '<sip:114@192.168.1.26>' failed for '192.168.10.136' - Username/auth name mismatch
Hi Thank you for your reply. I have tried various definitions in the sipusers table but none seem to be working :-( I have attached mey structure and content export below for your attention. Many thanks Scott Pinhorne -- -- Table structure for table `sip_users` -- CREATE TABLE `sip_users` ( `id` int(11) NOT NULL auto_increment, `name` varchar(80) NOT NULL default '', `accountcode` varchar(20) default NULL, `amaflags` varchar(7) default NULL, `callgroup` varchar(10) default NULL, `callerid` varchar(80) default NULL, `canreinvite` char(3) default 'yes', `context` varchar(80) default NULL, `defaultip` varchar(15) default NULL, `dtmfmode` varchar(7) default NULL, `fromuser` varchar(80) default NULL, `fromdomain` varchar(80) default NULL, `fullcontact` varchar(80) default NULL, `host` varchar(31) NOT NULL default '', `insecure` varchar(4) default NULL, `language` char(2) default NULL, `mailbox` varchar(50) default NULL, `md5secret` varchar(80) default NULL, `nat` varchar(5) NOT NULL default 'no', `deny` varchar(95) default NULL, `permit` varchar(95) default NULL, `mask` varchar(95) default NULL, `pickupgroup` varchar(10) default NULL, `port` varchar(5) NOT NULL default '', `qualify` char(3) default NULL, `restrictcid` char(1) default NULL, `rtptimeout` char(3) default NULL, `rtpholdtimeout` char(3) default NULL, `secret` varchar(80) default NULL, `type` varchar(6) NOT NULL default 'friend', `username` varchar(80) NOT NULL default '', `disallow` varchar(100) default 'all', `allow` varchar(100) default 'g729;ilbc;gsm;ulaw;alaw', `musiconhold` varchar(100) default NULL, `regseconds` int(11) NOT NULL default '0', `ipaddr` varchar(15) NOT NULL default '', `regexten` varchar(80) NOT NULL default '', `cancallforward` char(3) default 'yes', PRIMARY KEY (`id`), UNIQUE KEY `name` (`name`), KEY `name_2` (`name`) ) ENGINE=MyISAM DEFAULT CHARSET=latin1 ROW_FORMAT=DYNAMIC AUTO_INCREMENT=2 ; -- -- Dumping data for table `sip_users` -- INSERT INTO `sip_users` VALUES (1, '114', NULL, NULL, NULL, '114', 'yes', 'default', '192.168.10.136', 'info', NULL, NULL, '114', '192.168.10.136', NULL, NULL, NULL, NULL, 'no', NULL, NULL, NULL, NULL, '', NULL, NULL, NULL, NULL, '114', 'friend', '114', 'all', 'g729;ilbc;gsm;ulaw;alaw', NULL, 0, '', '', 'yes'); -----Original message----- From: Are london3@gmail.com Date: Mon, 21 Nov 2005 09:07:43 -0600 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Realtime Problems> You have error: Username/auth name > mismatch<http://fast.turbosite.net/phpmyadmin/tbl_properties_structure.php?> lang=en-utf-8&server=1&collation_connection=utf8_general_ci&db=mans> ionpbx&table=asv_sip> > > So there is clearly and issue with the content in your table. > > In our setup the column name and username have the same value = 114 > the fromuser and authuser column = NULL > > If this is not helping send your table definition and the content of your > record 114 and we will sort it out. > > -- > Are Casilla > http://astartelecom.com - Independent VOIP Telecoms Broker. Asterisk > Consultants > http://astbill.com - Open Source Billing, Routing and Management software > for Asterisk and VOIP > AstBill DEMO: http://demo.astbill.com >
Did you see the mysql.log file? I was having a similar problem, and i saw a problem with an update in a mysql table when a user was trying to register a phone. Sixto ----- Original Message ----- From: "scott" <scott@mysecretworld.co.uk> To: <asterisk-users@lists.digium.com> Sent: Monday, November 21, 2005 4:24 AM Subject: [Asterisk-Users] Realtime Problems Hi All I have configured asterisk with the addons and setup my config files so that i can pull sip extensions (phones) from a mysql database. I have followed all the docs and have editted my extconfig.conf res_mysql.conf and sip.conf to contain all that is advised.>From the CLI i can see realtime has a connection and is able to load theuser but when I plug in the voip phone it fails to register. My database content matches that of other phones in the sip.conf. If i remove the database user and add it direct into sip.conf the phone connects fine. Any help would be appreciated as its driving me mad now Very Happy CLI> realtime mysql status Connected to asterisk@192.168.1.26, port 3306 with username scott for 22 minutes, 47 seconds. CLI> realtime load sipusers name 114 Column Name Column Value -------------------- -------------------- id 1 name 114 callerid 114 canreinvite yes context default defaultip 192.168.10.136 dtmfmode info fromuser 114 fullcontact 114 host 192.168.10.136 nat no secret 114 type friend username 114 disallow all allow g729 allow ilbc allow gsm allow ulaw allow alaw regseconds 0 cancallforward yes Nov 21 12:52:55 NOTICE[15585]: chan_sip.c:10793 handle_request_register: Registration from '<sip:114@192.168.1.26>' failed for '192.168.10.136' - Username/auth name mismatch _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
You have error: Username/auth name mismatch<http://fast.turbosite.net/phpmyadmin/tbl_properties_structure.php?lang=en-utf-8&server=1&collation_connection=utf8_general_ci&db=mansionpbx&table=asv_sip> So there is clearly and issue with the content in your table. In our setup the column name and username have the same value = 114 the fromuser and authuser column = NULL If this is not helping send your table definition and the content of your record 114 and we will sort it out. -- Are Casilla http://astartelecom.com - Independent VOIP Telecoms Broker. Asterisk Consultants http://astbill.com - Open Source Billing, Routing and Management software for Asterisk and VOIP AstBill DEMO: http://demo.astbill.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051121/9da74db5/attachment.htm
scott wrote:> Hi > > Thank you for your reply. > I have tried various definitions in the sipusers table but none seem to be working :-( > > I have attached mey structure and content export below for your attention. >You should have a look at this page : http://www.asteriskguru.com/tutorials/realtime_pgsql.html. -- Benoit Merouze Network Software Developer at IPercom benoit.merouze@ipercom.com
I have a fairly simple menu structure, three options branch to submenus. There is a long (several seconds) delay between pressing a key and getting the next menu. This happens on 2 out of 3 of my menus for no apparent reason. I am kind of at a loss as to what to look at. Any suggestions would be appreciated. I am using Asterisk 1.2, CentOS 4.2. 2.6ghz machine with 1gb of RAM. -Kerry
Kerry Garrison wrote:> I have a fairly simple menu structure, three options branch to submenus. > There is a long (several seconds) delay between pressing a key and getting > the next menu. This happens on 2 out of 3 of my menus for no apparent > reason. I am kind of at a loss as to what to look at. Any suggestions would > be appreciated. >The problem is that asterisk does not know if it needs to wait for additional digits so is waiting for a timeout. When someone dials 3, are they done or could they also dial 30 or 301? The way to get rid of this wait is to make sure any other numbers in this context begin with a different digit. Only have 1 <option 1> 2 <option 2> 3 <option 3> and not additional lines like 301 <extension 301> Extensions in the same context would begin with a 4, 5, 6, 7, 8, 9. Don Pobanz