RE: [Asterisk-Users] Web based Asterisk management tool

Why not share with the community? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Chandler Sent: Wednesday, February 09, 2005 9:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Web based Asterisk managemen

[Asterisk-Users] back to operator

Is there a way to have a caller go back to the operator once they are in the voicemail directory or are they stuck? IE I call in and don't' know the extension but go to the company directory and can't find who I want, how do I get back to the operator?

[Asterisk-Users] IVR---if you know your parties extension you may dial it now

I have searched and I have my IVR working when it has to fork off to another application but how do I get it to allow callers to dial the extension directly instead of going though the directory? [mainmenu] exten => s,1,Answer exten => s,2,SetMusicOnHold(default) exten => s,3,DigitTimeout,5

RE: [Asterisk-Users] Bellster - IAX-based interchange -- lets you callanywhere for free

I get a 500 Internal Server Error when I try to register. :( -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ed Guy Sent: Friday, January 21, 2005 2:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Bellster - IAX

RE: [Asterisk-Users] Xfering a call

Well that didn't workI now get this error Jan 12 16:56:21 NOTICE[4989]: app_dial.c:746 dial_exec: Unable to create channel of type 'SIP' == Everyone is busy/congested at this time -- Executing VoiceMail("IAX2/[EMAIL PROTECTED]:4569/5", "b") in new stackJan 12 16:56:21 WARNING[4989]: app

[Asterisk-Users] Xfering a call

I'm having an issue when I transfer a call to another SIP extension it sees that the sip phone is not there and goes to voicemail but in my case it transfers to the main voicemail instead of the users voicemail. Here is what my SIP extensions look like in the extension.conf file exten => 3957,1,D

[Asterisk-Users] OT question

Can someone help me answer this question? Where would you most likely find a file with the line "+::"? What does it do? I have been racking my brain a buddy of mine is testing me and I don't want him to win. Thanks ___ Asterisk-Users mailing list

[Asterisk-Users] FWD to IVR

I'm trying to setup my FWD# so that when it reaches my * my IVR plays. Currently this works by me having it go to a dummy SIP/# as shown here in my extensions.conf exten => ${FWDNUMBER},1,SetMusicOnHold(default) exten => ${FWDNUMBER},2,Dial(SIP/},10,tr) exten => ${FWDNUMBER},3,Wait(2) exten =>

[Asterisk-Users] Can't dial out on X100P

I've been beating my head against the wall all night on this. I have gone through about 200 google searches trying to figure this out and am at a loss. I did do the instructions outlined in the readme.linux26 for FC2. Below is the error I get when I try to dail out. Dec 7 22:28:49 WARNING[22

[Asterisk-Users] cisco 7940 help

Does anyone have any simple documentation on converting a 7940 to SIP and making it function with *? I have been beating my head on a wall on this and have gone no where. Thanks Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium

RE: [Asterisk-Users] asterisk gui?

I agree it was nice once it was configured but it did blow up my install of *. Since it was on a test server I went and nuked it all, but will just keep my eye on it as it evolves over time. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Oliver Stone Sen