similar to: Voicemail() application returning -1 on a hangup

I would like to have my asterisk ring my cell phone and let me know when a new voicemail arrives. In fact if it would automatically put into the voicemail menu that would be cool too. In the future I will probably want it to IM me. Are there good examples somewhere of doing stuff automatically on the arrival of a new voicemail ? I noticed a place for the pager email address in voicemail.conf,

Hi, I set up a ring group. I would like for people who select a certain voice menu option to ring a list of extensions (I have just one extension in there at the moment) and if it doesn't answer to go to an extension's voice mail. I am using a version of asterisk from CVS, last updated a couple of weeks ago. This line in extensions_addtional.conf sends the call to ringgroup 3 if

There was a post in the 'wiki' for an application to provide an outcall when there is a voicemail is left on asterisk. I am having a problem that this application will only work if the caller presses the pound sign at the end of recording. As most people just hang up, this application isn't working. Can any offer suggestions to accomplish this out call?

On Tue, Nov 25, 2014 at 10:21 AM, James Lamanna <jlamanna at gmail.com> wrote: > > On Tue, Nov 25, 2014 at 8:14 AM, Matthew Jordan <mjordan at digium.com> > wrote: > >> On Mon, Nov 24, 2014 at 2:12 PM, James Lamanna <jlamanna at gmail.com> >> wrote: >> > Also, how big does the cache in frame.c grow to? >> > I've recompiled with

Todd How do you have the context in the phones sip configs set? Bryant From: "Todd Reese" treese65 at gmail.com Hi all, I've been have problems with getting this system on line and would like to acquire some help with the extensions.conf. My current problem is that the phones won't dialout.on the VOIP lines listed as dialout1, dialout2, dialout3. This version of asterisk

On Tue, Nov 25, 2014 at 8:14 AM, Matthew Jordan <mjordan at digium.com> wrote: > On Mon, Nov 24, 2014 at 2:12 PM, James Lamanna <jlamanna at gmail.com> wrote: > > Also, how big does the cache in frame.c grow to? > > I've recompiled with MALLOC_DEBUG on that server: > > > > asterisk -rx "memory show summary" > > > > .... > >

Hello , I want to enable recording for a few extensions. In sip.conf it is defined as record_out=Always record_in=Always under the section of extension.but it doesn't work. Extensions are defined in the extension_additional.conf file like exten => 10,1,Macro(exten-vm,10,10) exten => 10,hint,SIP/10 exten => ${VM_PREFIX}10,1,Macro(vm,10,DIRECTDIAL) I can't be sure

Hello Xen users, I want to purchase some gigabit ethernet cards. Our local Fry''s is having a $5.99 special on Airlink gigabit PCI cards with the Realtek chipset. Any experiences with this chipset vis-a-vis xen that would preclude its use? Thanks for your advice, Mike Wright :m) _______________________________________________ Xen-users mailing list Xen-users@lists.xensource.com

This might be a newbie question... I'm using a SIP trunk and trying to get DID line information on an incoming call. All I hear is a nice lady saying 'Zero' - then the call continues... Any suggestions? thanks Todd exten => s,n,Set(DIDID=(<${FROM_DID}>)) exten => s,n,SayNumber(DIDID) or exten => s,n,Set(FROM_DID=${EXTEN}) exten =>

Hello all, I think I have most of my AAH 1.3 setup running (Asterisk 1.0.9), but somehow something is not quite right with my vMail setup. I would have sworn this was all working, but maybe I was just dreaming. Anyway here is what is happening, say I am on extension 200 and I want to call to extension 201. If extension 201 is no connected, then it rolls right into vMail with the message the

Hi, using asterisk@home, with quadBri from junghanns.net I am facing a strange problem: I have set incoming routes for some extension / DID: [ext-did] include => ext-did-custom exten => 23,1,SetVar(FROM_DID=23) exten => 23,2,Goto(ext-local,23,1) exten => 57,1,SetVar(FROM_DID=57) exten => 57,2,Goto(ext-local,57,1) exten => 66,1,SetVar(FROM_DID=66) exten =>

I'm in the process of writing a simple autodialer to dial a list of numbers and play a message. One of the options I want to give them is a way to "dial X to have a customer service representative call you" Looking for a simple way to pass the number that I dialed to a script in extensions.conf... something like this: [serviceinterruption] exten =>

Ok, not exactly an Asterisk problem, but... I picked up some SNOM 200 phones because SNOM's have been recommended for use with Asterisk and they have line buttons that can subscribe to presence. However, they don't appear to power up when connected to my Negear FS108P, which is an 802.3af Power-over-ethernet capable hub. I am pretty sure these are the SNOM 200b, in that the ethernet

I'm trying to create a new contex for incomming calls from certain trunks. My problem is this calls are not checked through ext-did (for incoming routing). The calls from standard trunks are filtered correctly but these ones are not. Is there some way to debug what file/line is being executed by asterisk? My custom context is this: [from-pstn-nofax] include => from-pstn-custominclude

Hi Guys, I'm having sound problems when diverting a call using asterisk@home 1.5. I am using the following configuration in extensions_custom.conf, extensions_additional.conf and extensions.conf [custom-Sales] exten => s,1,SetVar(DivertNumber=02XXXXXXXX) exten => s,2,Dial(SIP/116, 15) exten => s,3,Goto(outrt-010-outside3,9${DivertNumber},1) (i have replaced the diverted phone

Hi Bastian, I'd like to propose backporting the following changesets from upstream xen-unstable into the xen-3 package. With these it is possible to boot a default installation of Squeeze (using d-i) in a domU using pygrub. 20480:c2c2e67b8198 pygrub: if default entry is "saved" then use first entry. 20481:8f4e0adc2b3b pygrub: expands tabs before displaying menus. 20485:086a6a0c3f37

We're having an issue where we call ReceiveFax in a context that includes a hangup extension and half the time dialplan execution doesn't continue after the fax is received successfully. Am I missing something simple here? Below is a sample call where this happened: The last log line for this channel/call is: [Feb 21 09:10:53] VERBOSE[13730] res_fax_digium.c: -- Channel

Using RHL 6.0 (2.2.5 kernel) with Win95 systems, smbfs doesn't recognize (immediately) changes made on the remote the remote system by a user on the remote computer. After a share has been mounted, if some computer other than the computer using smbfs to mount the share adds/removes a file to a directory in the share, an 'ls' of the mounted directory doesn't show the new/removed

Hello Guys, I'm somewhat of a newbie and am desperately seeking for some help... I've managed to get asterisk up and running on my server, and signed up with a broadvoice account... I'm having no problem dialing and communicating between extensions, but whenever anyone tries to call my broadvoice account, they are greeted by no ring or anything, but rather simply a direct to

Hi All, I have registered "abhijit" for SIP in asterisk Server. I am able to register my softphone (SJPhone) to the server using the name "abhijit". But whenever I try to make any calls I am gettinh the following error message:- *CLI> -- Registered SIP 'abhijit' at 172.20.28.85 port 5060 expires 120 May 26 07:34:52 NOTICE[2761]: pbx.c:1738 pbx_extension_helper: