similar to: how to load phone registration information

maybe a dumb question but what do we have here??? sip-friends.sql # # Table structure for table `sipfriends` # CREATE TABLE `sipfriends` ( `name` varchar(40) NOT NULL default '', `secret` varchar(40) NOT NULL default '', `context` varchar(40) NOT NULL default '', `username` varchar(40) default '', `ipaddr` varchar(20) NOT NULL default '',

i am getting 403 Forbidden message from asterisk when it try to register my user agent. i am basically useing mysql through ODBC. i hvae checked ODBC connecteion with 'ODBC Show' command. ------------------------------------------------------ *CLI> odbc show Name: mysql1 DSN: asteriskdsn Connected: yes *CLI> ------------------------------------------------------ and user is added to

Hi there asterisk goes to 90% cpu usage when trying to authenticate a sip friend using realtime mysql, no other message does appear at cli and asterisk hungs; here some info: *CLI> realtime load sipfriends name 104 Jan 13 11:52:21 DEBUG[8928]: res_config_mysql.c:109 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sipfriends WHERE name = '104' Jan 13 11:52:21 DEBUG[8928]:

Hi Guys, We have just set up Asterisk (CVS Head) for a realtime enviorment using MySQL & Asterisk Addons. I have populated the "sip_buddies" table with the same information that is came from our sip.conf, however registration seems to fail for the softphone we have set up. Does anyone have any idea as to what I should be looking for here? I'm not getting any error messages

Hello I need to add sip extensions from my UI so without going through sip.conf so i created table CREATE TABLE `sipfriends` ( `name` varchar(40) NOT NULL default '', `username` varchar(40) default '', `secret` varchar(40) NOT NULL default '', `context` varchar(40) NOT NULL default '', `ipaddr` varchar(20) NOT NULL default '', `port`

Thanks a lot for the help. But if Asterisk has private IP address and the only way to access it from remote sites is to have vpn connection to the site that asterisk existed (the site has vpn), then how that will happen from the Mobile to be able to run the softphone from the mobile? Any help? Regards Bilal ----------------- I installed out of curiosity today, and guess what? You can do SIP over

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

Having problems getting realtime working, I'm trying to use odbc for all of this. I've got Fedora 3 and have been fighting with odbc for a day now. I think I got it working correctly, however I can't seem to get the realtime portion working. In asterisk 'odbc show' shows it connected, I see it on my (odbc) mysql server connected and all, it connects and just idles. So, without

Hi all, Is there a way to keep track in Asterisk of which phones are online in realtime using some MySQL DB table for exemple, much like "sip show peers" does in the CLI? Regards, Ricardo.

You are reading the instructions for the STABLE 1.0 version of asterisk and are using the CVS version. Goto the wiki and read the instructions for RealTime. -Matthew ----- Original Message ----- From: "Muhammad Rizwan Khan" <rizwan@advcomm.net> To: <Asterisk-Dev@lists.digium.com> Sent: Wednesday, January 05, 2005 12:42 PM Subject: [Asterisk-Dev] Asterisk with MySQL >

Hello all, I recently noticed the realTime effort and must say it is a nice idea! I would appreciate any help to get it running .. I downloaded the code & patches and succefully patched my asterisk (CVS-HEAD-11/29/04-12). - created a DB called asterisk, and a table sip using the schema supplied at http://bugs.digium.com/bug_view_page.php?bug_id=0002613. - entered an entry: insert into

I'm having trouble with the Realtime setup. I've followed the instructions on voip-info using odbc but I get this message during asterisk boot: Parsing '/etc/asterisk/sip.conf': Not found (No such file or directory) Dec 14 16:11:37 NOTICE[8868]: chan_sip.c:8462 reload_config: Unable to load config sip.conf, SIP disabled == Registered channel type 'SIP' (Session

Greetings, Is there a way to tie a specific sip username to a IP address when authenticating against mysql sipfriends table? (USE_MYSQL_FRIENDS=1 USE_SIP_MYSQL_FRIENDS=1 in channels/Makefile) The reason is that I'm using Wellgate FXSes that have second/third/fourth FXS ports bugged when I use a password, but work ok when there is no password. Linking the username to a specific ip could

Hi, I'm deploying a SER + Asterisk architecture, where SER is used as SIP registrar, and Asterisk is used for voicemail and PSTN gateway. This system is already able to make Call Transfers (Blind and Attended) internally between phones registered in SER, although, I can't make Call Transfers in some scenarios involving PSTN numbers (which need to pass through Asterisk). The problem

Hi, How can I configure my asterisk server so that I can receive incomming calls comming from the same IP from where my server also receives phone registrations? The problem is that since the moment any extension registers at that IP (actually I have a registration proxy running at that IP), asterisk no more accepts calls coming from a SIP trunk I also have at that IP, replying back with

I am using mysql sipfriends and can't seem to get the MWI to work. From what I've read it seems this is not supported with that dynamic system, and probably never will be. I was thinking of just setting a cron job or something to check every minute for voicemail and set our sip NOTIFY messages as needed. Also, the PAP2-NA has the ability to reboot via a sip notify and I would like to be

Hi, I have some mySQL-sipfriends and connectivity to PSTN. When a call from PSTN comes, it shows a callerid, and that callerid is displayed at the called sip phone. When the call comes from another sip user (defined as mySQL-sipfriend), no callerid is displayed at the called sip phone. I turned on sip debug and discovered, that in the last case in the SIP-header to the called phone: From:

Dear all, I'm trying to enable Asterisk to work with FAX using T38. I've tried Asterisk 1.2.4 with the available patch found at URL http://bugs.digium.com/view.php?id=5090 and also with the new 1.4 Beta3 that is announced to support it too. With both Asterisk versions, I've sent with success FAXes between two FAX machines each one attached to an ATA interface, both registered in

Hi, I manage an SBC which stands between my company server farm and some SIP telco trunks. The system works fine, for inbound and outbound calls. Now I've configured the SBC to also act as a registration proxy, forwarding SIP registrations coming from the Internet to my asterisk servers. It all seems fine, but it doesn't work well, because by the time at least one phone registers through