similar to: How to find out if a new voicemail exists

Hi, 1. How do you then, synced then unread message presence with custom device status ? From an external program ? When a user leaves VoiceMailMan application ? Using externnotify ? 2. What is MWI:101 at default expression for (see [2] ? Cheers [2] https://wiki.freepbx.org/display/FPG/Subscribe+a+BLF+button+to+Monitor+a+Voicemail+Box 2017-11-21 17:58 GMT+01:00 John Kiniston <johnkiniston at

Someone posted a message suggesting someone try sendtext() and so I thought I'd see if it was useful. Much searching through help at the CLI has failed to find any help for sendtext, but I did find that: "core show function vmcount" fails but: "core show function VMCOUNT" works. Is that a bug and if so, has it been reported? Ira

Greetings, Quick question: I am about to acquire some disks for use with ZFS (currently using zfs-fuse v0.7.0). I''m aware of some 4k alignment issues with Western Digital advanced format disks. As far as I can tell, the Hitachi Deskstar 7K3000 (HDS723030ALA640) uses 512B sectors and so I presume does not suffer from such issues (because it doesn''t lie about the physical layout

Hi, Here's the deal. When someone leaves me a voicemail message I want Asterisk to call me on my cellphone by dialing my cellphone number and tell me I have a message. Is this possible? Can anyone cite examples? Most commercial voicemail systems produced in the last 10 years can do this. Any help would be much appreciated. Regards, Mike -------------- next part -------------- An HTML

Hi We are currently using Asterisk 1.2.4 with IAX and app_meetme for conferencing, but are looking to move to SIP because of issues with an IAX control we're using. The reason we moved from SIP to IAX in the first place was because of the poor NAT traversal with SIP. At that stage we were using Asterisk 1.0.*. How does Asterisk 1.2.4 handle NAT traversal and firewalls compared to the older

Hi all, is it possible to bridge a call between a Junghanns quadBRI card and a TDM400 in the same server? It should be I think, -- I am trying this and when an incoming call comes in, it hangs both up at the moment the bridge is attempted (and a subsequent 'qozap: dropped audio' error is show in the /var/log/messages) Any thoughts appreciated -- I've seen posts, but no clear

Hi Is it possible to dial from within voicemail to reach another extension. I would like my customers to have a choice of dialing 1 to get my cell phone while in voicemail or to just leave a message at the tone. Thanks Kelly

I am having major problems with the first run of Snom 360s that rolled out last month. I am working with the US vendor and they in turn are working with Snom but I wanted to see of anyone else was using these or having issues with them. Issues: Speakerphone/Hands Free volume spikes up and down during a call. You have to manually set the volume during every call. This makes it totally unusable.

I will be on vacation from <22/06/06> to <30/06/06>. I will not be reachable on my mobile. I will have limited access to mails, and please expect a delayed response. In my absence, please contact the following: Ray Richard or Safeer Mohammed Thanks H.Gireesh

On Saturday, April 15, 2006 3:17 PM Remco Barende wrote: > I heard that Junghanns is working on such an interconnection. It is > already possible to connect their PRI cards, and they are working on > BRI<->PRI. Correct. The next driver generation is supposed to support this fully. > I ise their bristuff for an HFC-S BRI card and am not happy at all > with the way they

Hi, my oi151 based home NAS is approaching a frightening "drive space" level. Right now the data volume is a 4*1TB Raid-Z1, 3 1/2" local disks individually connected to an 8 port LSI 6Gbit controller. So I can either exchange the disks one by one with autoexpand, use 2-4 TB disks and be happy. This was my original approach. However I am totally unclear about the 512b vs 4Kb issue.

Hi friends ! I want to add stun functionality in asterisk. can anybody give me some hint that how can i start that. thanks in advance Deepak Dhiman

Is there a way to find out the no of parked calls in a parking lot by name (in a multiple parking lot environment) from within the dialplan (not CLI) other than writing a custom function (like VMCOUNT)? Thanks, Matt -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, is the following possible? I would like to transfer a call to my "personal" MeetMe conference room and get transferred there automatically as well. Currently I can transfer the call to the conference, have to hangup and then call the conference number myself. I would love to have this in one quick function. Moreover is there a way to disable the "You are currently the only

Hi Please help! I have one X101P and TDM400P in my asterisk Box When i make a call from * to PSTN, everything goes Ok, When the PSTN hangups or * hangups, the busy tone is detected and * disconnects the channel without problems. The problem occurs when the call comes from PSTN. When * hangups, the other end (at pstn) does not hangup, it only presents silence. Please tell me how to solve this

Hi, I am investigating a difference in code generation between release and assert builds of llvm. The culprit is IndVarSimplify that comes up with different behavior on the same input: in the assertion build, it does do an extra 'INDVARS: Rewriting loop exit condition' After digging around, it seems that following change is the culprit: ----- Author: Philip Reames <listmail at

Hello, In sip.conf, each peer/friend/user entry gathers several parameters such as type, canreinvite or mailbox. How can you specifically access to mailbox value from dialplan ? I know how to access custom parameters (ie setvar=FOO=value) but I don't know to access standard parameters. I'm specifically concerned to access to mailbox's value (from a given entry) but would be

I have "externnotify = /var/lib/asterisk/agi-bin/vm_notify.pl" so that when someone is left a voicemail it will call the person's mobile phone and prompt them with the new message. The perl script simply originates a call to a persons mobile phone and connects it to their voicemail using VoiceMailMain. Problem is when user hangs up from checking their messages, it runs the

I was flipping through the code recently and noticed that we still have the double whammy of allocating dir entry positions with parent_dir->counter++ and that weird setting of f_pos to 2^31-1. So after enough creates (and deletes :)) in a directory we end up with an entry item whose key is past that value. f_pos gets rewound instead of being set to that magical EOF. readdir() gets stuck

Hello, I am using the voicemail module of asterisk. When I did some test calls from my mobile phone, sometimes the beginning of the prompt was missing, e.g. instead of something like "number 12345 not available" I was only hearing "345 not available". Verbose level 5 on the asterisk console didn't give me any hint on this, it only shows that playback of the prompt started