similar to: sipfriends in MySQL question/request

Greetings, I'm running latest * from CVS on FreeBSD 4.10 box. We've just bought several WellGate 3502A FXSes to play with till welltech guys fix the 3504a's registration bug. So far everything is working as expected, except the fact only ulaw and alaw codecs work with *. If I add allow=gsm or allow=g723.1 in FXS's ports entries in the sip.conf, no voice is heard from both

hi, i.m. newbie in asterisk. asterisk 1.0.3 is my current version. i like to store usernames and passwords in a sql database. i like to log failed authentification-passwords, to create a blacklist for securityreasons. i thingk a sql-database is a good way to log these actions. i don.t find debugging-options to output invalid login-passwords. Ok, i have made the following: debian is my OS.

Hello there, I have configured my asterisk to run on Mysql backend. But the Asterisk was unable to pick the peer details from the database. This is how I configured the Asterisk to run with mysql on the backend. Edit /usr/src/asterisk/channels/Makefile, change it to enable the MYSQL_FRIENDS USE_MYSQL_FRIENDS=1 USE_SIP_MYSQL_FRIENDS=1 cd /usr/src/asterisk make

Hi, No one seems to have any issue with the following posting. Can any one suggest how to install/configure channel drivers to work. Basically I am trying to send the SIP calls to GNUGK but Asterisk reports the error "No channel driver found". >>> I was trying to compile the oh323 channel driver but unable to compile the openh323_1_13_5 (which is the only required version as

I'm authenticating against sipfriends in MySQL, and have just noticed that none of the below commands return any output: sip show users sip show inuse sip show active sip show subscriptions Is this a bug or something wrong on my side? I'm using the stable 1.0 cvs Vahan

I'm loosing hair at cosmic speed now for the past 10 days. Welltech's Wellgate 38xx/35xx FXO/FXS SIP hardware versions seem to have very buggy firmware possibly due to hastely done porting from H.323 firmware. Is there anyone on this mailing list who was able to: 1. setup a 35xxA FXS with all ports authenticating properly with *? or 2. setup a 38xx FXO to work as dial-in from pstn to

Where I can find the firmware for the Wellgate 3804 ? The files are: - 2m4sipfxo.103 - 4fxosip.103 I don't have a password to pick up it at the welltech site. Kind regards, Miguel

I've just compilied th latest CVS of * with USE_MYSQL_FRIENDS enabled ("1"). During startup * tells me that it connects to the db, so this should be fine. Nevertheless I don't see any users from the db when I run "sip show users" or "iax2 show users" although I configured some. It is also not possible to call them. Any hints?

11-10-2004 there was a subject: Re: Where did USE_SIP_MYSQL_FRIENDS go?: on asterisk.user list. >All db specific code has been removed from the code in favor of the >currently-in-development "RealTime" method of configuration from >database. >You are most likely not using the 1.0 stable branch. >You need to use the new RealTime configuration method. And currently,

Any feedback on this brand and in particular on doing business with WelltechUSA? I am looking to the Wellgate 3701A which is a 1FXS-1FXO arrangement. I am hoping to replace the near worthless Grandstream HT-488. This company is telling me that I need to wire $ directly into there bank account. Most unusual. Thanks for any feedback on this, Marty

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

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

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:

Anyone else has problems with CVS HEAD's from today with voicemail hanging up silently without any debug/error messages when checked? It also keeps insisting that the user's voice mailbox is full and can't store more messages even if I clear/rebuild the /var/spool/asterisk/voicemail stuff. I've tried falling back to voicemail.conf entries from realtime voicemail with the same

Hello to all, Im trying to use DeadAGI to implement billing with Asterisk2Billing. Before the billing, I had something like: exten => _2XXXXXXXX,1,Dial(SIP/${EXTEN}@voiprovider) Now, with Asterisk2Billing would be something like this? exten => _2XXXXXXXX,1,Answer exten => _2XXXXXXXX,2,Wait,2 exten => _2XXXXXXXX,3,DeadAGI,a2billing.php exten => _2XXXXXXXX,4,Wait,2 exten =>

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]:

I downloaded asterisk and was trying to compile fresh, It end up in error, Any help appreciated. cvs checkout asterisk cd asterisk make clean make END UP with following error, (Previously I was able to compile without any errors. After a make clean stopped compiling.) gcc -shared -Xlinker -x -o chan_iax2.so chan_iax2.o iax2-parser.o -lmysqlclient -lz /usr/bin/ld: cannot find -lmysqlclient

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