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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

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, We run 1.1.3 and I use Outlook to connect to my IMAP box and a couple of times per day (randomly) when I click certain folders IMAP will start retrieving ALL headers for that folder again ('retrieving new headers')! I don't understand this. What triggers IMAP to do so, why not just retrieve only the NEW headers? Is it Dovecot or Outlook (probably) and what can be done

Hi All, kind of need help on the conference module, i'm using freepbx and enabled conferencing, i created a conference number, 6000. when i dial to it, my phone says it is connected but i'm hearing nothing, maybe logs below can help you. also, when i hang up the phone, the conference did not disconnect me. how can i end a conference? thank you -- Executing

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, hope any one can help create a trunk, i'm talking to a voip gateway provider right now, they gave me the IP address of their server a prefix to authenticate calls. How can i create such a trunk? example prefix is 1234# and IP address is 1.1.1.1, in ser i was able to do it by just simply rewriting the host part. Hope you guys can help me. TIA Regards, Ron

Hi, I have this weird problem i cant explain. i have two asterisk, i'm using realtime table for my sip/user accounts. my database is on a mysql cluster. my prob is if i register on phone on asterisk 1 it is ok, but on second asterisk it can't, Registration from '"122144" <sip:122144 at 10.10.10.131:5060>' failed for '12.34.56.78' - Wrong password

Hi All, i have 3 asterisk server in a cluster using a cluster of mysql server via realtime, users can register via DNS SRV. I send/receive calls to an AS5400 via a SIP trunk defined on the realtime sip table, the trunk has call-limit=5. Problem i encountered is each of the 3 asterisk servers will 5 channels each to them instead of 5 for all 3 servers. Is there any solution to this?

Hi All, Any solution on how I can implement prepaid billing on asterisk? But not the calling card type, just a simple Custome rwill buy credit, consume then buy again. Also, is there a solution for that when you combine asterisk with ser? Regards, Ronald

Hi, I'm trying to install asterisk(v1.2.22) with FreepBX(v2.2.3) with a 4-Port FXO Sangoma card A200. I'm using Fedora 7 (x86_64) kernel version 2.6.22.1-27.fc7, but i'm having these errors: $ ztcfg -vvvv Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04: FXS

Hi, Is there a specific latency that asterisk accepts? I encountered a problem wherein when the latency was unusually high,my xlite's (i have 2 xlite) cannot register. but when the link suddenly went stable, the x-lite just registered. what i forgot to look at is if the registration packet is reaching my asterisks. ------ when xlite cannot register --------------- Pinging

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