similar to: res_odbc.conf and odbc show

Hi, I have some issue with voice mail with ODBC on asterisk 11.7 box. I may not understand database functionality on asterisk fully. The most suspected area is func_odbc. I already googled but not luck. Your guide is warmly welcomed *Error messages when I make call and leave message.* -- <SIP/1ffa9-00000007> Playing 'auth-thankyou.g722' (language 'en') [2014-05-28

Hi I have On func_odbc [EXEC] readhandle=ressqlserver writehandle=ressqlserver readsql=${ARG1} writesql=${ARG1} I'm trying an update on dialplan: exten=> 141,3,Set(dummy=${ODBC_EXEC(UPDATE Tabla set campo = ${EXTEN})}) On Cli: WARNING[3579]: func_odbc.c:353 acf_odbc_read: Error -1 in FETCH [UPDATE Tabla set campo = 4356] Any idea why is this?? The query

i have installed the asterisk 1.6 before that installed the necessary packages in Debian, * i followed the steps as follows, root at astserver: ~# apt-get install unixodbc unixodbc-dev odbc-postgresql postgresql-8.1 postgresql-contrib postgresql-dev * then i installed the asterisk 1.6 version with the odbc modules as in the selected list. * then i created the database as asterisk and the user

I have an issue where * tries to route a call over IAX to another server even if the server is down. I have included the relevant entries from my iax.conf, extensions.conf, and some debug output. If someone could tell me what I have configured incorrectly, I would appreciate it. Thanks, Stephen -----------iax.conf on voip2---------- [voip1] type=friend username=voip1 host=x.x.x.x (ip

Hi, I just upgraded from 1.2 to 1.4. In 1.2, when I did a "show uptime" I used to see a second line telling me the time since the last reload. Has this been removed in 1.4? The following is the output of my two test boxes: Connected to Asterisk 1.4.18.1 currently running on voip2 (pid = 10605) Verbosity is at least 3 voip2*CLI> show uptime System uptime: 15 hours, 55 seconds

Hello, Now I'm testing func_odbc and hash. My configurations are: func_odbc.conf [GETNUMBER] dsn=sqlserver ;mode=multirow ;rowlimit=10 readsql=SELECT number,real_number1,real_number2,status FROM ivr.dbo.numbers WHERE number=${SQL_ESC(${ARG1})} extensions.conf exten => s,1,Ringing exten => s,n,Wait(4) exten => s,n,Answer exten => s,n,Set(NUMERIS=37037210602) exten =>

I'm looking to use * for a HQ/branch office topology with fairly few calls over the WAN. The questions I have all pertain to the following architectural pic: http://www.45891.com/misc/arch.jpg I'm looking at a distributed architecture so users are somewhat functional when the link to HQ is down, with a centralized voicemail server to allow for transfer of voicemail messages from user to

This *should* be something that's easy to fix, but apparently I'm not doing something right. Our SIP long distance provider is telling us to only use formats G.723 and G.729, so I've set up their trunk configuration in sip.conf as such: [t564] type=friend host=XXX.XX.56.4 context=default disallow=all allow=g723 allow=g729 However, the Dial application gives the following error:

I have the following pair of macros defined to handle outbound calls from *. Rather than specifying full dialstrings in the main body of extensions.conf, outbound dial commands are made using a macro call as follows: Macro (outbound,number_to_dial,callerid_to_present,gateway1,gateway2,gateway3,gate way4) The final gateway defined is nearly always a fallback to PSTN if none of the IAX or SIP

I have setup iax2 between two servers without success. when I launch asterisk with the asterisk -vvvvvgcd command I see serveral wanings listed below. Is this why I cannot make connections?? My question is, how do I setup zaptel timming without any cards if possible? Does anyone have the steps? Thanks for any information. James [chan_iax2.so] => (Inter Asterisk eXchange (Ver 2)) Apr

Hello list, I notice that the dialplan method SIPAddHeader is not working : in dialplan : /exten => s,n,SIPAddHeader(Privacy: id)/ in SIP invite no trace of this header : /INVITE sip:0473 at sip.domain.be SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:5063;branch=z9hG4bK-5b2b1b97 From: "VC" <sip:voip2 at sip.domain.be>;tag=729476652f511c67o2 To: <sip:0473 at sip.domain.be>

Hi I have an account with mynetphone (australia), which gives me two voip (sip) accounts, which i used to have connected to a spa9000. this is behind a firewall, so on the spa9000 I would listen on another port apart from 5060. so on the firewall 5060 would go to voip1 and 5061 to voip2. I moved to asterisk (+tdm410) and the machine was also the firewall and I had no problem - well atleast it

I have successfully configured two servers and I am now trying to connect via iax. When I attempt to call from one ext, 2006(server viop1) to extension 3006 (server voip2) I receive a timeout or "call failed 403 forbidden. The information I am receiving from the console is below. Apr 26 10:53:32 WARNING[311313]: channel.c:1745 ast_request: No channel type registered for 'IAX'

I constantly get the following error messages in /var/log/asterisk/messages: Dec 8 10:52:57 WARNING[1009521664]: File chan_iax.c, Line 3324 (iax_ack_registry): Received unsolicited registry ack from '192.168.0.1' Dec 8 10:52:57 WARNING[1009521664]: File chan_iax.c, Line 4181 (socket_read): Registration failure Where 192.168.0.1 is another asterisk server. Below are the local and

Hi all asterisk 1.8.23 I have odbc all setup to mysql but cant figure out why the dialplan wont write to the odbc function fubc_odbc.conf [DEVICES] dsn=device-conn ;dsn in res_odbc not odbc.ini readsql=SELECT call.callNum, call.city, devices.callId, devices.id FROM call INNER JOIN devices ON call.id = devices.callId WHERE deviceNumber = '${ SQL_ESC(${ARG1})}'

Hello, I have a bunch of Polycom 601's and Asterisk 1.4.13. The problem is that the MWI indicators will never go off (The blinking red light and envelope in the LCD). I have tried to upgrade to 1.4.14 and all different SIP versions on the Polycoms. I am now at 1.6.7 Here is the SIP Message that turns on the lights: Scheduling destruction of SIP dialog '

All, Thank you in advance for any help. I have a customer in need of a conferencing system. A requirement is for users to each have their own PIN for the same bridge. So, I put the list of users, PINs bridges into a MYSQL DB and used an ODBC connector to parse the table. Asterisk is connected and reads the rows as expected. The problem is that if a user enters a PIN that is NOT in the table,

Good morning, I'm having suddenly cut-offs and I don`t know why. It's been hapenning since I enabled cdr_odbc/func_odbc in my system. We use func_odbc to register some queue member's events (login, pause, etc.) at an external DB ('remoto' connector) and to uptade local tables at a local DB ('local' connector). Currently we are usind cdr_odbc to Postgresql and cdr_addon

Okay, so here's the configuration I have for MySQL Realtime (Asterisk version 1.6.2.17): In /etc/asterisk/extconfig.conf: sipusers => mysql,mya2billing,cc_sip_buddies In /etc/asterisk/res_mysql.conf: [mya2billing] dbhost = localhost dbname = mya2billing dbuser = a2billinguser dbpass = REDACTED dbport = 3306 And here's the error messages I get: voip2*CLI> realtime mysql status

Hey, all... If this is too off-topic, I'd be grateful for directions to a more appropriate mailing list. I'm trying to set up Asterisk and some Sipura boxes. I've got an SPA-3000 which is registering twice with Asterisk - once for its FXS/Line1/VoIP1 and once for its FXO/PSTN/VoIP2. My eventual goal is to have inbound calls on its FXO ring four times on its FXS and then fail over to