similar to: Voicemail WAV to PDA Problems

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 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

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

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

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 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....

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

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 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 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

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

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.

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.

I've got a very strange problem and I can't figure it out. I have a Cisco PRI gateway connected to * via SIP. When I debug on the Cisco, I see callerID name, but it is not getting to * via SIP. I am running * 1.4.2 and the latest Cisco IOS for my router. Here is what is happening: A call comes into the gateway. It sends a SIP INVITE to * with "pending" as the callerID

I have a little call recording script that I am running and it works fine, but I have one problem. I get CDR when a user calls into the extension, but I do not get CDR for the call that it makes outbound on # 17. Any idea why? Here is the extensions info: [default] exten => 2211,1,Answer exten => 2211,2,Wait(1) exten => 2211,3,Playback(/etc/asterisk/recording/getshop) exten =>

Is there a way to use privacy manager without requiring the user to enter their name? Essentially I am just looking for a way to force the called user to enter "1" to accept the call. I don't need a name recording. I want a call to come in, a message to be played, music on hold, call out to the called party, then enter "1" to accept, "2" to reject. Peder

I am using the Find-me/Follow-me example below with screening: http://www.voip-info.org/wiki/view/Asterisk+tips+findme Here is my actual config: [macro-screen] exten => s,1,Wait(1) exten => s,n,Background(press-1-to-be-connected-to-the-caller) exten => s,n,Set(TIMEOUT(response=5)) exten => 1,1,NoOp(Caller accepted) exten => i,1,Set(MACRO_RESULT=CONTINUE) exten =>