similar to: Multiple servers using realtime

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 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

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 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:

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

Is it possible to use the "include" command to include other contexts if you are using realtime for extensions? I've searched voip-info and some people were asking about it, but I didn't find a real answer anywhere.

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

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

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

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...

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

Does anybody happen to know the input power specs for the Polycom IP 500 and IP 600? We've mixed up our power supplies and we've got a whole box of them and can't figure out which go to the Polycoms. I would rather not kill the phones by trying random ones....

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.

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

We have users with Cisco 7900 phones running sip. When user A calls user B, we want user B's name to appear on user A's phone. It shows the extension they call, but not the internal name of the called user. Is this possible? We have some people that used to be on an MGCP based system and they would get the callee's name popup on their phone when they called someone. I

I am trying to use Asterisk Manager via php to record auto attendant greetings and I just can't figure out how to do it. I've got the php page working and I can click to call between two phones. However if I click to call just a single phone and then try to enable "monitor", when I pick up the ringing phone, it just hangs up and doesn't record anything. I'm sure I

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

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

How does the g729 encoder/decoder count in regards to the total number of licenses and how does it count an encoder/decoder? I looked on the wiki and don't really see anything that explains it. In other words, how do the calls below count (assume no reinvite)? g729 phone calls into voicemail g729 phone calls g711 phone g729 phone calls other g729 phone

I am trying to setup a queue in a very specific way and I can't quite figure it out. I'm sure someone else has done this. I want calls to come into a queue and do a ringall on a number of phones (let's say 3). So ring them for 20 seconds or so. If there is no answer, I want it to ring a second set of phones for 20 seconds. If no answer, then go back to the first set of phones.