similar to: Single * multiple offices

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 All, I have setup 2 trunks using 2 different voip providers using sip. the first one i have no problem calling inbound then redirected to an IVR, i can hear the IVR. the second one has issues, inbound works going to IVR as i can see it on the CLI, but i don't hear anything. i tried redirecting it to an extension not an IVR just to see if inbound really works, and it rings the

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

Are there any standards for setting up pbx dialplans for businesses/offices? What I mean is that, which numbers are reserved for a specific use ex. 0 for operator ? Putting Zero for operator in the dialplan seems to be the common practice of businesses. If there is such a standard, * and # are used for what ? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All, How can i play a sound during dial while waiting for it to connect? Coz currently i'm using SIP providers from other countries, when i send them the call there is a bit of delay to connect. I would like my users to hear a music first then when the call connects the sound gets canceled out. coz some users think the phone does not work coz they just hear a long silence but they

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, 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

Good <fill in local time of day> I'm looking for a piece of hardware that we can place in two offices that have decent bandwidth, but are in two different US states. There are phone systems on both sides, that have extra CO analog line ports that I'd like to connect through. One side has an IVR, the other side does not use one during office hours. The best configuration would

Asterisk is an excellent PBX system, and makes a very good endpoint in the SIP chain for all sorts of things -- IVR systems, voicemail applications, automated messages, etc. It has an extremely well-written CDR engine, so many people mesh it with billing applications to produce accurate accounting information. It also is fully aware of the media stream, which means it's capable of cutting

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

Good Day list, I have a friend who is interested in implementing an asterisk implementation at his offices. The configuration would consist of the following Site A ---- Asterisk Box With 12 incoming lines and 15 phones Extensions 101-115 Site B ---- Asterisk Box With 4 incoming lines and 7 phones Extensions 201-207 Site C ---- Asterisk Box With 4 incoming lines and 6 phones

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 All, is it possible to choose outbound route by checking the extension of the caller? e.g extension that starts with 3 goes to outbound route 1 extension that starts with 4 goes to outbound route 2. Basically, i'm hosting two(2) office, extension 3XXX is office 1 and extensions 4XX is office 2, they both have the same dialling pattern so i need to choose route based on source.

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

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

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

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

I would like connect two offices where one office have 4 PSTN Analog lines and another office without any PSTN. Both the offices will have two separate Asterisk server with TDM400P cards (4 ports FXS & FXO). My questions is that how to configure Asterisk to forward the PSTN calls directly to another Asterisk which has the TDM400P card without pressing the extension number. Diagram like