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911 Help! I accidentially deleted all directories under /var/spool/asterisk I did use the backup facility not too long ago but cannot find the process for restore. However, I don't believe a full restore is needed -- I just need to know the names of the directories under /var/spool/asterisk and re-create them (I hope!). Can some kind soul give me some direction or tell me the directory

I have a setup which allows users to access my asterisk box via FWD. That is, a user in say, France can call into a local access number for FWD, then hit number 7 which dumps them into a DISA request for a password, which then dumps them into my internal extension so they can dial out through a VOIP provider. Everything works fine until they enter their tones for the number they are calling

Strange things. When I run the RxFAX command through an internally dialed extension, I can *hear* fax tones, meaning, I presume, that the RxFAX application is running. In fact, doing a show application confirms that. So, I'm presuming RxFAX application is talking as it should. However, inbound fax calls (tones) are not being detected. I know that my extensions file is corect. I am

I have a little confusion about the general settings (other than the register values) in the SIP General area. I understand that for examle in a SIP context like [FWD] or [BROADVOICE] the entries in those areas are ths settings that take effect in any communication woth FWD and/or BROADVOICE. However, I'm confused as to the purpose of the "general" settings -- to what or which

...& VPN trouble (gw@adcomcorp.com) > 4. Re: Native MoH patch for 1.0.8? (Juan Jose > Comellas) > 5. Re: Polycom & VPN trouble (Tim Pushor) > 6. Re: Re: teliax [Was: LiveVoip is Bankrupt] > (r00t) > 7. Newbie Confusion on Call Forward and > DBput/DBdel (Jeffrey Starin) > 8. Eicon equipment, BRI Server or PRI? > (gw@adcomcorp.com) > 9. Re: Level 3 SIP <--> asterisk (Max Clark) > 10. Re: Asterisk server with remote monitoring > capabilities > (Max Clark) > 11. How to get the outbound data of agent in queue > (Gary Li)...

Hi! I have MeetMe working fine except I thought that if I used the "i" option, there would be some kind of announcement made upon a new meetme person joining a conference. Is that correct? Is it the "i" option? Because I'm not hearing an annoucement when a new person joins the conference. Also, if I combine the "i" with the "x" option, there is

I have MeetMe working fine except I have a little confusion over the "i" option. Is that supposed to allow for announcing to the conference the recorded name of the individual who is entering the conference? If so, it's not working in my verions of Asterisk HEAD. I am using ztdummy because I have no digium stuff onboard. Everything else works fine. Is there some trick to

I have the standard script for activating call forward and when I do a database show, I indeed see: /CFIM/2000 :12125553434 so I presume that means call forwarding is in effect. However, when anyone dials extension 2000, it rings and no forwarding takes place. Is there something basic I'm missing here? Does one have to define, first, what CFIM is? The Newbie Thanks! B.

I've recently upgraded *@home to CVS HEAD and in addition to losing the ability to use the MySQL database, I've noticed that my MOH has degraded significantly. I've tried all sorts of remedies -- removing the x100p card and loading asterisk without the zaptel drivers and such -- still have terrible MOH. I've used .raw. sln. mu. .gsm .mp3 files for the music on hold and it