similar to: Playing Sound during dial

Hi, I'm using gotoiftime on asterisk, but it seems  there is a difference between the asterisk time and the system time. could it be because i adjusted the system timezone on my linux? do asterisk not detect the change of timezone on the system? How can I fix this prob? Regards, nhadie -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Is it possible to assign a plus sign on the callerid(num) ? currently this is what i do CALLERID(num)=+6523450017 but telco is denying calls, coz they said they are seeing "bs523450017" instead of +6523450017. i tried putting it inside double quotes CALLERID(num)="+6523450017" telco says the same thing. is this possible? thank you Regards, nhadie

if i have this setup: [sip users] -- [asterisk] --- [as5300] --- [pstn] asterisk will talk to as5300 using sip. i will use as5300 as a trunk on the asterisk so sip users can call out to pstn. what i would like to is do prepaid on those trunks, not on the sip users. sip users can call any other sip users . i want to do it that way coz i'm trying to build a multi-tenant pbx, and i will use

Hello ! I want to call a line and play a sound from the callee before putting it in connection with the caller. Is this possible? Example: Dial(SIP/111111, m) // Ring or Music... if(call==ANSWERED) Play(announce) // Play 'announce' to the called // To connect caller and called ? Best regards, Mickael. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I am running an AGI script in a command dial, or call a SIP trunk. I want to execute after 10 minutes a voice message (stream file) on the channel to warn the person that the call is about to end. How to do that? Thank you, Mickael. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Is there a way to check on AMI if a user is currently engage on the phone? i would like to display on my portal whether a user is calling or not. thank you regards Ron

Hi All. got this problem that IP phones could not re-register to my server. even if device is power cycled it still would not register. the solution i found was to change the SIP port settings on the phone and it will register. but after registration expires and its time to re-register the same thing will happen, so i have to update the port settings again just to make it work which is

Hi, what software do i need to convert an mp3 to a g729 format? I have a portal where a user can upload their own MP3, but when a user is using a g729 codec, the music on hold has a crackly sound. using g711 it's very clear. so what i'd like to do is when they upload an MP3 i will make a copy on g729 format, so that asterisk can choose which file to play depending on what codec is

hi, for my multi-tenant pbx, i would like to approach prepaid like this: when a customer dials number, i have an AGI that will determine what country was dialed and retrieve the rate from the rate table, once the rate is retrieved, i will get the remaining balance of that customer nd compute how much time remaining based on the rte and the remaining balance. then i set that as an absolute

Hey folks! Does any one have an idea as to when centOS 5.3 will be out and if so then I would like to hear from you coz I am waiting for it day and night coz I heard that the kernel of centos 5.3 will have inbuilt support for intel pro/wireless 3945 wifi card ( I have been trying to install the drivers desperately with no fruit ). And by the way does any body know how I can safely upgrade from

Hi, Is what i'm thinking possible, i will setup an asterisk server, i will let 2 or more offices use it, e.g. Office 1 will have extension 100 - 110 Office 2 will have extension 100 - 105 Office 3 will also have 100 - 105 my questions are: 1. will it have conflict on the asterisk having the same extension number for each office? 2. can i define trunk/s that will be dedicated only for

hi all, i recently subscribe for an isdn and terminate it on a 3825 router. i used it as a sip trunk for my asterisk. i'm a newbie when it comes to ISDN. and i've been experiencing some issues: 1. Call Hangup: When hangup is initiated from the outside the extension (softphone/ip phone) does not hangup, is this normal? shouldn't asterisk hangup the extension as well when it

hi everyone. can anyone provide me concrete examples on how to use the GotoIf application? can't figure out how to use it in my dialplan coz im having errors....thanks! : ) __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com -------------- next part -------------- An HTML attachment was

Hi, i'm currently using Originate command on AMI, i can call a certain channel like a SIP user SIP/1000 then once 1000 is answered it dials out to amobile or landline. Would just like to know if i can use AMI to dialout to a mobile or landline first (instead of SIP user) and once answered, dial another mobile or landline again. If not is it possible to call a macro from the AMI? i think

Hi, Is it possible to use realtime for Music On Hold? Is it also possible to store the music/audio files on the database, same way a voicemail can be stored on the database? Thank You Regards, Nhadie

hey guys, wanna know where to find my graphics card driver for centos5.5 coz the 5.4 support it and xserver start automaticly but the 5.5 doesnot start and tried to reconfigure the xorg.conf but it doesnot work too. and i noticed that the hardware prob process find a different driver for my card . mine is arrandle.intel! my system is a dell studio laptop 1569! thnx in advance -------------- next

Hi, I'm quite familiar w/ mp3 cbr/abr/vbr encoding, as well as mpeg4 (cbr/vbr,etc). And I can always calc the bit rate for a given file size with: file size * 8000 / length in seconds = kbits/sec Works great w/ mpeg4 + mp3. BUT FOR THE LIFE OF ME: I cannot get oggenc (1.0x version) to give me the file size I want. I calc. it with the above formula, and nothing comes out right. Then I do

Hi, Is it possible to detect where the call came from and route it out to different sip trunks. e.g. i have user 100300 when that user calls outbound i will make him use of [sip-trunk-100] another user, 101300 when that users calls outbound i will make him use of [sip-trunk-101] actually the 100 and 101 at the beginning of the username is the accountcode i used for cdr. hope my question

Hi, Anyone knows a DID provider that can do both outbound and inbound? Regards Nhadie

Hi, I'm running asterisk realtime, i had prob when a user does not unregister properly. I tested with SPA942 and a PAP2, when phone is registered, i call using the SPA using x-lite no problem, but when i unplugged the power, it does not unregister properly, so asterisk think SPA942 is still registered, when i call using x-lite, asterisk tries to call it.so it gets stuck at [Aug 11