similar to: sip_buddies mysql table

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 '',

It seems that when setting qualify = 200 or qualify = yes in the database for a sip friend/peer, RealTime does not update the registration status like it should. I also have several peers which have been offline and Asterisk still reports them as registered, even though the registration seconds are only 200. Asterisk Ver: CVS HEAD 12/1/2004 Layout of sip_buddies: mysql> describe

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

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 all! So far I've always used plaintext passwords for SIP, but now I've decided to use MD5 encryption. For each client I edited its section as follows, then: auth=md5 md5secret=hashed_passwd ;secret=plaintext_passwd where hashed_passwd is the output of echo -n "user:realm:plaintext_passwd" | md5sum When the first SIP clients registers with Asterisk after a "sip

Hello, Right now there is not a table build script at: http://www.voip-info.org/wiki-Asterisk+RealTime+IAX Therefore I have taken the SIP build script and added a few fields that I use from my iax.conf (could be more out there, please see the complete build script below): `dbsecret` varchar(100) default '', `notransfer` varchar(100) default '', `inkeys` varchar(100)

ok so my table looks like this... REATE TABLE `sip` ( `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`

from the asterisk messages log: Registration from '<sip:40852@192.168.70.2>' failed for '192.168.70.25' the only place I can see extension 40852 linked to the ip is in the phone's configuration. sip.conf [40852] ;for a grandstream bt100 musicclass=homeline pedantic=yes accountcode = 40852 amaflags = billing ;callgroup = callerid = 40852 <40852> canreinvite = no

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

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, I have some peers configured in SIP.CONF file with parameters incominglimit and outgoinglimit set up to 10. By doing that, I expect that this peer will not be allowed to handle more than 10 incoming calls and 10 outgoing calls at the same time. However, since I upgraded to Asterisk 1.2.17, I started to face a problem. Sometimes, calls to those peers are not connected. When I check the

I am running asterisk 1.0.1 with OH323 compiled in. We have a 323 trunk to CallManager with a mgcp controlled pri router. When using sip phones (directly registered with asterisk) to call out the 323 trunnk to PSTN, calls timeout after 3 rings. If I answer b4 3 rings - no problem, otherwise I get "no one is available to answer at this time" on the consoel and it redirects to an

For those that don't read every line of source code here's something I found out today... -------- Deprecated incominglimit and outgoinglimit Incominglimit = number of calls the local extension can originate to Asterisk. Outgoinglimit = number of calls Asterisk will terminate to local extension. End of Life for these commands announced**, please use setgroup and checkgroup, that will

I have 2 issues which I need to resolve on our production Asterisk server: We are currently using Polycom IP600 VOIP phones for our office which are capable of handling 2 calls per SIP registration. What we're finding is when staff are on the phone, Asterisk will pass them a second call which will show up on their display, and an audible beep is heard over the phone (regular call waiting). I

I've been googling and talking with Libretel about my setup and the fact that incoming calls to my asterisk box through the Libretel number reach my box (I hear the greeting being played) but then don't accept DTMF. Here is a rough diagram of my setup: Asterisk | server | NAT <------------ Libretel | router | Note that there are NO SIP

Starting with Asterisk 1.2 I have always used realtime static to load my extensions.conf into Asterisk. It worked perfectly up to version 1.6.0.X but starting from 1.6.1.X and upwards it simply does nothing. I can see that the extensions.conf file is mapped to the database: == Parsing '/etc/asterisk/asterisk.conf': == Found == Parsing '/etc/asterisk/extconfig.conf': ==

Hi, all. I have problems with iax2, when try to communicate with one third server, asterisk reports the following errors in server's, could help me? Server A it speaks It with C in iax and Server B it speaks with D in iax, but Server A it does not obtain to speak with B in iax, reports the following error in server B "chan_iax2.c:5749 socket_read: Host 200.xxx.xxx.xxx failed you

Hi all I've been doing some testing on realtime using mysql, an have a little question that could not find the answer to or maybe its not posible at this time. Is there a way use "register=>......" on a DB using realtime. For the moment I use it in sip.conf. It will help me a lot if this could be store on a DB somehow. commets or sugestions .... ? thanks Billy

hi there, i have posted earlier on the list but got no satisfying answer. the problem is not big. I have asterisk server directly connected with internet (79.80.x.x) and clients are behind router. clients/users are registered with asterisk and are using sipura and xlite softphone. Now problem is that when a user calls other by dialing his IP:Port (sip uri), call is connected fine and he can

Hi! I am proving Asterisk 1.2.24 in realtime with MySQL 5.0.27 using Idefisk softphones. I followed the steps of "how to" of voip-org but always have this error: Sep 25 20:29:07 WARNING[12000]: res_config_mysql.c:360 update_mysql: MySQL RealTime: Failed to query database. Check debug for more info. Sep 25 20:29:07 WARNING[12000]: res_config_mysql.c:360 update_mysql: MySQL RealTime: