similar to: Problems to use asterisk with mysql /odbc

Hi! I have benn playing a little with quesues tonight and I found out if there are at least one member-extension free the announcement with p'the place in the queue wont be played to the person who called in. Is this possible to change so the announcement will be played even if there are free member-extensions? I think that would be nice (well it's not how ACD-groups usually works but

Hello folks, I'm running 1.8.11 on a Centos 6 system with an adjacent Hylafax server using softmodems: Noticed this in the Asterisk log when trying to send a fax from Hylafax to Asterisk: [Apr 3 01:53:09] WARNING[29184]: chan_sip.c:8926 process_sdp: Multiple audio streams are not supported I've googled a few asterisk tickets that may suggest that yes, multiple audio streams are not

have you set up the db schema? and have you entered any sip data into the db? Sean -----Original Message----- From: Chandra [mailto:chandra@digital.com.np] Sent: Tue 1/6/2004 10:57 PM To: asterisk-users@lists.digium.com Cc: Subject: [Asterisk-Users] no results. i have been working with the retrieve_sip_conf_from_mysql.pl file and i have set everything as required. but when i

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]:

Hello all! I am trying to load sip.conf from mysql database. I have followed the instructions at <http://www.voip-info.org/wiki-Asterisk+sip+mysql+peers>. Seems that the authentication (user & psw) works fine but I would like to get more information from mysql and I don't know how to retrieve it. Could anybody help me? Any idea about how to do it? Regards, Victor.

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`

hi using sipfriends from mysql from asterisk 1.0 branch, how can I make asterisk show the true channel's current codec with SIP SHOW CHANNELS? This does not seem to work, and bkw_ said sipfriends from mysql didn't have that info at all. For what it may seem, asterisk uses G.726 as told, giving me a -- Format for call is g726 at the start of the call, but in SIP SHOW CHANNELS all these

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`

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

Anyone know the command that tells * to load a sipfriend from the realtime db rather than saying no such host? I've tried various combinations of the rt commands: rtcachefriends=yes; ;rtcache=yes ;rtAutoClear=yes ;rtautoreg=yes ;rtIgnoreRegExpire=yes ;rtupdate=yes rtfromcontact=yes Basically I have a group of 4 * servers all routing calls, but only two are hearing the phones

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

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

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

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:

After updating to the latest CVS Feb 17 15:20:03 WARNING[15317]: config.c:819 read_config_maps: The 'sipfriends' table is obsolete, update your config to use sipusers and sippeers, though they can point to the same table. == Binding sipusers to mysql/asterisk/sip == Binding sippeers to mysql/asterisk/sip Feb 17 15:20:03 WARNING[15317]: config.c:823 read_config_maps: The

Hello I was wandering If I let 2 asterisk boxes (let's name them ast01 and ast02) connect to one SQL realtime iaxfriends/sipfriends database What happens if I register my client to ast01, The ast01 box will update the client's record in the iaxfriends database (ipaddr/port/regseconds) Let's say there is an incoming call then for this client but this call arrives on ast02 (the box

Hi all, I am new to asterisk and I can not find any detailed info on using SIP MySQL support (sipfriends) with clients behind NAT. I've heard that I have to patch chan_sip.c and Makefile to get it working. I tried on voip-info.org but found no answer for my questions. I found some answer on Digium mail list archive: http://lists.digium.com/pipermail/asterisk-cvs/2004-January/000854.html

I was looking for MySQL table structures for ARA (Asterisk 1.8.X). I found one for SIP friends on, https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime,+MySQL+table+structure But it seems that it is not as per the Asterisk 1.8 SIP options. i.e. it contains 'call-limit' which is deprecated in 1.8 and not the 'callcounter' as one of the fields. Pardon my ignorance, but are

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

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 '',