similar to: addmailbox2 (Attached)

Hi Lee, You need to specify the VM context that you are using.. so using your examples.. extensions.conf entry.. exten => 1000,1,Dial(SIP/1000,20) exten => 1000,2,Voicemail2(u1000) exten => 1000,102,Voicemail2(b1000) exten => 1000,103,Hangup should be.. exten => 1000,1,Dial(SIP/1000,20) exten => 1000,2,Voicemail2(u1000@sip) exten => 1000,102,Voicemail2(b1000@sip) exten

I have googled this one to death, and can't find anything. I added a number of new users to my asterisk (current CVS) system. I am using the "Voicemail2" family. I added entries in extensions.conf and voicemail.conf for my new users, and I have tested leaving and retrieving new voicemails for them. All of this works fine. But if one of the new users tries to "Administer

Hallo.. if you have one more for me...thanky you.... Stephan -----Urspr?ngliche Nachricht----- Von: WipeOut . [mailto:wipeout@linuxmail.org] Gesendet: Donnerstag, 10. April 2003 19:28 An: asterisk-users@lists.digium.com Betreff: [Asterisk-Users] Asterisk-Redhat 9 install guide. Hi, Not sure if anyone will be intersted but I have put together an install guide for Asterisk on RedHat 9.. Its

Ok.. I have done some more digging and the problem seems to be caused by chan_capi not detecting that the call has been answered.. I downgraded chan_capi from 0.2.3b to 0.2.2 and the system is working fine.. Kapjod, do you know of this problem? Later.. > I seem to be having problems with my 2 asterisk systems.. > > This works.. Call originating from the PSTN and then routed to the

Look like SNOM have rolled back the firmware version of the 200's from 1.16w to 1.16q.. Anyone know why? -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

I see in the changelog that ENUM support has been added.. anyone know what this is? -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

Hi.. How do you accept user input while waiting or playing moh? My Dialplan is as follows.. ring,ring,.. Hello thanks for calling blah blah... Please enter the extention number blah blah... WaitMusicOnHold(10) If no input pass call to operator.. The problem is that the user has to input the extension while they are being told what to do.. any input during Wait or WaitMusicOnHold is ignored...

Hi, I installed s fresh system yesterday and it seems that zttool did not install!! ztcfg is there.. Anyone else had this problem or is it just me? -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

Can the system command be used to retrieve a variable from a mysql database using the mysql command line client?? or would it be simpler to write some sort of AGI type application?? -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

Callgroups/pickupgroups are allocated per channel, not in the dialplan. sip.conf and zapata.conf are the two files you're interested in. -wade ---- Original Message ---- From: wipeout@linuxmail.org To: asterisk-users@lists.digium.com, Subject: RE: [Asterisk-Users] Using callgroups (was: Taking a call for someone elses extensionfrom my extension) Date: Sun, 20 Apr 2003 16:39:15 +0000

Subject: Re: [Asterisk-Users] IAX Bandwidth Question Hi, I have changed the codec to lpc10 (I see what they mean by Mr. Roboto!!) and the ping times generally dont go over 300ms. It seems very odd that GSM saturates the link. I would love to give g.723.1 a try but have no idea where to get the proprietary codec... can anyone help? Thanks Jay > ----- Original Message ----- >

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

Hi, I have just been playing with the latest X-Lite.. It works fine with Asterisk.. As for codecs I tested G.711a/u, GSM and iLBC... iLBC is the only one that didn't work.. not sure why.. But the bigger problem is that when I call another extension that is using a Snom200 the call connects but there is no audio in either direction.. I have tried G.711a/u and GSM and while X-Lite shows that

Hi, Is Asterisk (or can it be set up as) a SIP proxy? Thanks -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

Hi, Is there and trick to getting a conference room up and running.. I have 'conf => 7500' in the meetme.conf file and 'exten => 7500,1,MeetMe(7500)' in the extensions.conf file (in the same context as my phone extensions).. When I dial extension 7500 I get the voice saying "That is not a valid conference number, Please try again.." <beep> <beep>

Hey, Anyone have any idea what the cost to visit the VON exhibition in London next week? Not the Conference thats just way to far out of my budget.. Thanks.. -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

Hi, Does any one have any pointers on improving moh quality?? Symptoms are crackling and hissing as the sound comes and goes.. I installed mpg123 this morning.. I have tried various MP3's sampled at 160k, 128k, 32k and 8k and they all sounded terrible... The PC is a P4 so its got plenty of processing power.. I have tried a few different types of classical music (Piano, Violin and full

I have been using chan_capi and a Fritz AVM (passive) card for a little while now.. So far so good.. There are still a couple of things I am not totally sure about.. First.. What is the difference between "softdtmf=0" and "softdtmf=1"?? Which is the preffered/recommended option?? Second.. What is Early B3 all about?? Should I be using it in my dial string?? Which is the

Hi, I have VoiceMailMain on extention 1001 so it would be nice to get that sent to the phone instead of asterisk@[IP Address] when there is a message waiting.. Is it possible to change the extention that is sent to the phone when MWI is lit up?? Thanks.. -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by

Hi, What is the correct chan_capi dial syntax?? This is what I think it is.. exten => _90.,1,Dial(CAPI/[msn here]:${EXTEN:1}) This seems to work for local numbers.. but I have an access number for cheap long distance calls.. wich gets dialed and then the number I want to call is sent as DTMF after a few waits (w).. On my X100P I used the following.. exten => _9001.,1,Dial(Zap/1/[access