similar to: Strange voicemail problem when call forwarding off local PBX

I would like to create a simple menu that would allow a caller to decide whether they want to leave a message or be forwarded to another number (i.e cell phone). Thanks in advance for any insight. Here's my current extension.conf [general] static=yes writeprotect=yes [globals] [default] exten => 101,1,Dial(SIP/101,20) exten => 101,n,Voicemail(101 at default) ;This automatically

On my 7905s I can configure a "voicemail" number, which in turn activates a "Messages" softkey. When pressed, it goes to that exten, which in turn is configured to go to VoiceMailMain. Great, works like a charm. However, the phones also have a "go to voicemail timeout" after which, the phone diverts the call to voicemail via a "temporarily moved". This

lqbal, I do plan on having alot of users. Two markets I'm trying to get some volume users from are: residential consumers and business users. Residential consumers should get basic line services such as their own DID, voicemail, caller-id, call-waiting, three-way calling, and basically, all the standard features you get from companies like Vonage, etc. This particular market base

Waldo, How do you let your customers manage 'their' PBX. I too have a setup like you. However, I installed a * server for each customer, via vserver. I'd like to now what kind of software/webbased package you use for this. I also have SER installed as a front-end server for the * servers. But, as I'm still not very into SER, don't know exactly how this fits in. Should I use

> -----Original Message----- > From: JR Richardson [mailto:jmr.richardson@gmail.com] > Sent: Saturday, June 17, 2006 2:30 PM > To: asterisk-users@lists.digium.com; Douglas Garstang > Subject: Voicemail with NFS (working, I think) > > I'm using a stand-alone VM server and exporting the VM files ro for > MWI function only. All my registration servers mount the remote

Can anyone point me to info on how to change the functionality of the SIP (7.4) 7960's. We previously had an SCCP firmware on the phone and the users want the phone to work like it used too. Here are some examples: The users do not want to push the new call softkey or the speaker button in order to dial a call. They want to be able to just begin dialing the number. The users do not want

thanks to those who replied. I have managed to get the functionality I was looking for working with a series of Macros. However, it doesn't work as simply as I would like. There are two issues I've run into: (1)Goto provides no way to pass variables between one context and another. (2)I can't find any way to Goto a specific point within a Macro when calling it. Mostly this is a

Howdy, My first forray into using res_mysql.conf for realtime access of sip users and extensions. I have the following relevant section of extensions.conf: --- [trunklocal] exten => _NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}}) [local] include => trunklocal include => trunktollfree [longdistance] include => local include => trunkld [international] include

Hi All, Strange issue, recently I started getting a lot of .lock files in the voicemail /INBOX folder preventing proper access to voicemail. I can delete the .lock files and everything is normal. After searching around, I found some SIP lock file stuff but nothing specific to voicemail. Can someone point me in the right direction to resolve this? I'm runnning 1.2.9 on Debian Sarge.

Is there any way to program a soft key in ADSI to put a caller on hold. Then able to retreive that caller. Example - Softkey Hold Softkey Retreive Call Softkey End Call -gcc

I've got a problem that keeps popping up with my reception phone. It is a IP 650 and the receptionist - on three occassions - has accidentally hit the "Forward" softkey just before she enters the "Page All" keystrokes and then all future calls get routed as an overhead page. I will admit, the first time it happened, I was totally stumped. Why the heck did I have

Hi All, A couple of customers called complaining that folks were dialing into their PBX trying to use the Directory to locate users, from a Blackberry, and getting frustrated due to the incompatibility of dialing alpha characters on the the qwerty keyboard and not getting through. The issue of course is the Directory application only recognizes numeric digit tones, not alpha characters (not sure

> Just wanted to let you know you were just left a 2236:45 long > message (number 1) in mailbox 1000 from an unknown caller, on Monday, > February 21, 2005 at 05:38:14 AM so you might want to check it when you > get a chance. Thanks! > > --Asterisk Hmmm... Call came in either via a SPA-3000 or an X100P Zaptel FXO interface. Got 2200 minutes of beep-beep-beep-...; the caller

Update When no greeting is recorded the default you have reached ext # greeting is cropped. When there is a greeting it is just ignored and not played at all. Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 ---------------------------------------- From: "Bryant Zimmerman" <BryantZ at zktech.com> Sent: Saturday, November 23, 2013 8:32 AM To: asterisk-users at

We upgraded to Asterisk 1.4.13 earlier this week, and now an odd problem has cropped up with our Polycom phones. Basically, performing a transfer via the transfer softkey fails on transfers to extensions beginning with a *. When you try, you just get a fast-busy, and next to nothing on the console (usually a message about the peer being unable to authenticate) This is only affecting

We are installing Aastra phones (480's and 57i's) into a fairly simple asterisk setup. Although call park & pickup work fine using xfer to 700 (to park), dial 701 (to pickup), we are unable to make the park/pickup softkey feature work on the aastra's. Although we've programmed the softkeys per the manuals, they seem to have no effect (just dead). For example, our 57i is

Hey all I have been beating on this all weekend long. Any feed back would be appreciated. We stood up a 11.6 system. We tested everything we could think of. We moved over to it and all seemed to be working good than a customer told us that they were not hearing our vociemail greetings. When we call into the system and it drops to voicemail we just get a beep no greeting played. We checked

I found on the web that there is way to customize the softkeys for the 7941/7961 phones. In the SEP<mac>.xml there is a section called "softKeyFile" where you can specify an xml file for the softkeys. I couldn't find any examples of this softkey file or the format to this file. Does anyone have a call manager enviroment where they can look at this file and send me an

Some of you may remember back in May the thread on using Asterisk as a voicemail server for a Cisco Callmanager system. My own Callmanager system is integrated into an Asterisk server for voicemail (and other things). Back in May I was using H323 for integration, but since I've upgraded to CCM 4.1 I have switched over to SIP. The integration with H323 required using Call forwarding to send

Hello; I am using vicidial which is using asterisk 1.8, mean while when the extension has voicemail, I always see the red light on the Polycom and hear the beep sound (toot toot) in period time. Also, I can see at the LCD an option to select it for accessing the voicemail ?but I am facing the following problems: 1) The red light and the beep: How I can let the Phone only have the red light