similar to: Realtime Extensions and "Include"

I am using realtime on two boxes, one running 1.4.10.1 and one running 1.4.11. Everything works fine except for when I make a database change, such as a phones password. I change the DB, I prune the peer, I see it is gone and then I see it show up again in "sip show peer xxxx", but everything is not being updated. The phone will not register even though the DB and the phone have

Does anybody have realtime queue members working? Not the queues themselves, just the members. I have realtime working for voicemail and sippeers, but I can't get queue members to work. Here is what I have: res_mysql.conf: [general] dbhost = 127.0.0.1 dbname = ASTERISK dbuser = myuser dbpass = mypass dbport = 3306 dbsock = /tmp/mysql.sock queues.conf: [general]

I am in the process of setting up several * servers using realtime and connecting to mysql. I am trying to figure out if I should just use one database and one set of tables for all of the user data. Or if I should have separate databases for each * box. The boxes are independent of each other in that customerA only connects to box A. They will never fail over to box B or anything like

Is anybody using 1.2 (or 1.2.1) in a production network using Realtime (voicemail, sip or extensions) with 100+ SIP phones? If so, what are your experiences? We've been running 1.0.3 for about a year and it's been rock-solid. We'd like to upgrade to Realtime and 1.2, but I'm afraid of killing our stability. Obviously, we'd do it in stages (upgrade to 1.2, then realtime

Hi, I think ive found a bug but need someone to double check. Whenever I issue a "reload" in Asterisk, any realtime extensions stop receiving calls. I have to reboot the sip phones in order to get them to re-register. Can anyone see if they have a similar problem? Asterisk 1.4.32 Mysql realtime. Thanks Dan -------------- next part -------------- An HTML attachment was scrubbed...

Is there any way to see the called number when a call gets redirected to the 'a' extension from voicemail? Say x123 calls x456 and it rolls to voicemail. x123 hits * and gets dumped into the 'a' extension in the original context. I need some logic in 'a' to do a database lookup based on the original called number (x456). Any ideas? When I do a test, it appears

I know that you can set it up to where a user hits 0 from their mailbox and goes to an operator, but can you set up other options as well? Could I have 0 for an operator and 1 to go to another extension? I know you can do this by building an AA, but I don't want to have to do that for every user as there are about 40 people that want this. They won't all go to the same number.

hi, i'm a bit puzzled because i do not get include and hint to work with the new realtime enginge (cvs-head from 2004-12-09). other things (sipfriends and "normal" extensions) work perfect with the realtime engine. the entries in the static extensions.conf file i used before where: exten => 183,hint,SIP/snom220 exten => 183,1,Macro(stdexten,443,SIP/snom220,183) exten =>

Does anybody know the sql type for the "call-limit" field under sip peers? Everything on voip-info is missing that entry.

Is the storage of actual voicemail messages in a database still limited to ODBC? If so, why? And is the use of mySQL and ODBC at the same time still a bad idea? If so, why? I want to store all of my voicemail stuff in a database so that I can give users web access to it, but I don't want to run web services on my * server itself. If it is all in a DB, I can have a web box and a

I have a customer with a Fortinet Firewall that is having stability issues with Asterisk and SIP endpoints (PAP2T) outside his network. The first issue I see is that Asterisk sees all phones as the IP address of the Fortinet. Since the parameter "localnet" defines the local network and that address falls in that range, how will Asterisk treat the endpoints? I have

We've had a few people complain that the "beep" before leaving a voicemail is not loud enough and too short. Does anybody have a recorded beep that they can share, that is a little louder and a little longer? We've had this box in production for 2+ years, so I hate to mess with the gain on the PRI or anything like that because everything else works fine. I know nothing

Howdy, My first forray into using res_mysql.conf for realtime access of sip users and extensions. I have the following relevant section of extensions.conf: --- [trunklocal] exten => _NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}}) [local] include => trunklocal include => trunktollfree [longdistance] include => local include => trunkld [international] include

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:

Hi, I'm testing realtime right now, it does not seem to me that realtime contexts can be included in normal context, like this [sip] include=>sip-dial exten=>i,1,Hangup [sip-dial] switch=>Realtime/sip-dial Am I getting it wrong ? Tnx ! -- Best regards, Alessio mailto:afoc@interconnessioni.it

I am running 1.4.10.1. I have a macro that is called from default for a certain extension (both below). I added NoCDR to s to try and stop extra CDR records, but I am still getting them. Any idea how to stop them? extensions.conf: [macro-STDEXT] exten =s,1,NoCDR() exten =s,2,Dial(${ARG1},30,Tt) exten =s,3,Goto(s-${DIALSTATUS},1) exten =s-NOANSWER,1,Voicemail(${ARG2}|u) exten

I am trying to configure RealTime Voicemail with MySQL. I downloaded compiled and installed the CVS HEAD for asterisk, and for asterisk-addons. MySQL seems to be loading correctly (the cdr table is recording incoming calls). But the RealTime Voicemail doesn't seem to be checking the database table for the voicemail users. When trying to login to voicemailMain if I use a user in the

Our asterisk server has been up and running for over a year and it works great. I have emails going to my account as an attachment and I can listen to them on the desktop and it works fine. I just got a T-Mobile MDA that runs Windows Pocket (or whatever they call it) and it can check email. If I have it download the email, it gets the attachment, but it can't seem to play it (it CAN

I have 20-30 GXP2000's connected to * over a T1 line. Neither end is NAT'd and there is plenty of bandwidth available over the line. The GXP's are 1.1.5.15, which is the latest. I have a problem where the phones keep dropping off of * and I get a "failed to register" message in the log of *. Sometimes they eventually connect and sometimes, I have to reboot them to

As seen in the following URL: http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions and as I also tested some time ago with an old release of Asterisk, RealTime Extensions didn't support the Ex-Girlfriend syntax. Is it already working in recent 1.4 or 1.2.15 releases? Is there any other way that I can use to do the same thing but only using contexts, for example? If yes, please