similar to: where is addmailbox now?

Hello, I checked out the asterisk version from the CVS. But I dont seem to have the addmailbox script. How can I setup a mailbox without this utility. Regards, Rajesh Golani

Hi, I am a new user to both Linux and Asterisk and would be grateful for any help and advice anyone has to offer. I have installed Linux and asterisk as per Andy Powell's excellent getting started guide. The problem I have is that the addmailbox utility does not work and I cannot find the file anywhere on the machine. I downloaded the files via CVS so assume I have the current

Hi * users, I want to know if STUN suport is available with Asterisk. Kindly let me know. I have posted this also in DEV list but none replied to me. thanks, Somesh -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050808/e26855c9/attachment.htm

Tie line is type em_w (old school stuff) to TE110P. Phone rings on the asterisk side, but the calling party does not hear the ring through sound. If I pick it up within the first two rings it goes through and I can talk otherwise our old switch drops the call. Anyhow...here is my config if anyone can shed some light on it. It used to work with HEAD a few weeks ago. -Matt

The cheap ones on EBay won't work with the SC420 server. I have one and can't make any of the clones work. I do have one TDM40B card for analog stations that works well. The problem with the SC420 is that it won't let you set the interrupts yourself and you end up with interrupts being shared. =============================================================== Message: 26 Date:

Hello, I am trying to let someone know that is being called from a specified location. For that, the command: exten => _107.,1,Dial(SIP/SIPGW/${EXTEN:3},30,A(Anounce.gsm)) should let the called person hear Anounce.gsm as soon as he/she answers. (Only calls with prefix 107 are given this notice). The call proceeds fine, but no one hears AnounceSPF.gsm. I tried putting this file in every

I just bough a Sipura SPA-2100 to use with Asterisk. When I use the analog handset plugged into the SPA-2100, the person on the other end can hardly hear me. I check the SPA-2100 setup and their is no mic/spk gain control. Is this a problem with the SPA-2100 or with Asterisk? Any way for asterisk to compensate for the poor audio level (if the problem is the SPA-2100)? Thanks, Mike

I've just gotten off a skype conference call and it pisses me off that the quality of skype is higher than my asterisk calls. Is there such a thing as a super high bandwidth codec? In a situation that you have the bandwidth to share is there something that I can use for important calls when the situation warrants it? TIA, Dean -------------- next part -------------- An

is it possible to register asterisk in a sip proxy as if it were a terminal (like a cisco ATA)? how? Thanx Jenna ;) ___________________________________________________________ 1GB gratis, Antivirus y Antispam Correo Yahoo!, el mejor correo web del mundo http://correo.yahoo.com.ar

Hi I am trying to make asterisk load config from database, so far I get the sip, extension working, but voicemail seems still looking for config file, not from the database. the extconfig.conf looks like ... sipusers => mysql,asterisk,sip_buddies sippeers => mysql,asterisk,sip_buddies extensions => mysql,asterisk,extensions_table voicemail => mysql,asterisk,voicemail_users .. the

Hi, Everybody, I can modprobe TDM400P card without feeding power supply , so what is the purpose of providing power supply in that card. Can any body tell me Regards Lokesh Portugal mail -lokeshkumar80@yahoo.co.in ____________________________________________________ Send a rakhi to your brother, buy gifts and win attractive prizes. Log on to http://in.promos.yahoo.com/rakhi/index.html

Hi peoples Can anyone tell me if the Zultys Zip 4x5 supports iax protocols or if they have configured one before for iax. If you have a sample config file that would be great. Any assistance would be nice Scott -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050811/6c243053/attachment.htm

Hello, If a parked call times out, it will ring the original extension again. If this fails for any reason (e.g. the extension is busy), then the caller receives silence until the timeout time, and then gets the busy signal and is disconnected. I consider disconnecting callers like this very unprofessional, and could happen for instance if you receive a new call when you just parked the

I am using asterisk (2 week old CVS) am for the first time have been starting to experiment with busy and congestion. At this point I am only using sip endpoints PAP2-NA devices. All testing of this is being done on a local network. my test extension looks like this: exten => 7777,1,Answer exten => 7777,2,busy(35) exten => 7777,3,Hangup Or like this: exten => 7777,1,Answer

We have recently started routing about 3 PRI's worth of traffic thru our asterisk box. The text on the console now flys by so damn fast, I can't really see what the heck is going on. Even with verbosity 0 and debug 0 it is still so fast. Is there some way I can attach to the console in a way that will allow me to grep or otherwise filter the text so I can focus on something in

hello according to this user guide found at http://www.automated.it/guidetoasterisk.htm#_Toc49248768 it says the following Voicemail - Please leave a message after the tone... Ok, so you've got the basics going, and it's great - if you happen to sit by you phone all the time. What happens if you are out/away from your desk/sleeping you'll miss those vital calls. We need to set up

We have been running into problems here, we have 2 PRI's when they fillup, All channels in use, and we dial more calls asterisk becomes unstable and crashes alot. We are currently on: Asterisk CVS-v1-0-10/26/04-09:35:31 built by root@localhost on a i686 running Linux I know I need to upgrade. Is this a know issue?? Kyle

I'm trying to get this zap dial to work. I want to send DNIS and ANI to other system (ZAP/g2) at answer, while the caller hears ring (RBT). I hear the RBT, but callerid and exten is not sent to other T1 - The ZAP/g2 T1 is standard D4, SF, E&M Wink Start. - At ZAP/g2 wink, asterisk should send DTMF "*ANI*DNIS*" exten => _XXXX,1,NoOp,${CALLERID} exten =>

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How come an outgoing call using my TDM400P immediately say the call is answered? I'd like to be able to detect when the call is actually picked up, is this possible? If this is normal with analog cards, does the same thing happen with T1 cards? -L ____________________________________________________ Yahoo! Sports Rekindle the Rivalries. Sign up for Fantasy Football