similar to: Is it possible to Voicemail menus (not just audio files) ?

Hi, Has anyone met any success, installing localized (ie non-english) menus within SIP firmware enabled Cisco 7941 ? Those phones seem to be trying to download localized menus from Cisco Call Manager but as they are managed by an Asterisk server, I'm looking for a workaround. Any advice ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Is there a way to edit the options available in the voicemail menu trees? My users are complaining that it's too complicated (I know, it's not really complicated), and I wanted to remove some of the options if this is possible. So far I havent' found any info on the wiki or searches, not that it isn't out there.. I just cant' seem to find it.. Any pointers? Thanks Dan

[ I'm not in the list because I didn't find a digested version; please move the lists to sourceforge.net, and we would have the digested version. I read the replies from the archive. ] Hello. Would it be possible to allow Vorbis use the same codebook for multiple files? I could keep a 650 MB codebook on CD-R and use that for all my audiofiles. If that is possible, how much the

Hi Is the keep alive activated on the phone? On Thu, Aug 11, 2016, 5:36 PM Dovid Bender <dovid at telecurve.com> wrote: > 1) Does it happen every time at the 5 minute work? > 2) Have you done a dump on the client side to see if the NAT device is > dropping the packets? > 3) Is the phone behind a load balance internet connection and is the RTP > port changing? > > >

Hello, In voicemail.conf, it's possible to edit the voicemail message, but when I define a pager email address, I get the message from "Asterisk PBX", and the content is fixed by the system. Is there a way to manipulate this message, as well? Thanks, David -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi R-ers, I'm wondering if anyone has investigated a method for embedding audio files in R graphs (pdf format), and allowing their playback to be triggered interactively (by clicking on a graph element for instance). I know how to do this in latex pdfs with the multimedia package, but it seems that R would provide a more appropriate platform for many reasons. Thanks for any help you

I need to authenticate users to make long distance calls. Basically,when the user dials a long distance dialplan pattern, I want to prompt for his pin and look it up against a table of pins:usernames in a file. If it exists, I'll use the username in the cdr accountcode and permit the call. Authenticate() looked very promising nut I couldn't get the ma options to work. Any help is

Hi all, Just installed Asterisk 13 on CentOS 7 and have run into a problem. The Scenario is this: Asterisk is on the internet the Phone, a D40, is behind NAT So someone calls the number and Asterisk routes the call to the D40 Everything works fine and the call is established, but then after 5 min. the caller stops getting audio from the D40 but there is still audio to the D40. using both

When my follow me or transferred calls come out to me they appear as if they are coming from one of my lines rather than showing the caller id of the initial caller. I believe there is a way to make it forward the initial caller id information isn't there? Is it just that my voip provider is not allowing me to do this and if so does anyone have any suggestions on some voip providers that

I quote from "Principles of Digital Audio" by Ken C. Pohlmann: "Because perceptual coders tailor the coded signal to the ear's acuity, they similarly tailor the required response of the playback system itself. Live music does not pass through amplifiers and loudspeakers, it goes directly to the ear. But recorded music must pass through the playback signal chain. Much of the

Hi, I can't find a way to tailor DHCP/TFTP/HTTP environment so that brand new Aastra SIP phones can be auto-provisioned when config files are stored in a specific TFTP subdirectory instead of TFTP root directory. For instance, TFTP root directory is /srv/tftp. When config files are stored in /srv/tftp, a new Aastra can find its config files. When config files are stored in /srv/tftp/aastra,

Hi All, I'm going to be upgrading our 1.2 Asterisk system. At the moment we use the Enicomms SLN files. Are there major differences in the 1.4 default voicefile packs, or will I be able to re-use Enicomms?? In the Make menuselect, I noticed theres no .SLN voicefile selection for the basic audiofiles - has SLN been depreciated? Thanks Adrian

I'm running debian testing, so samba 3.05 atm. As of a few upgrades ago, which i think coincided with a library restructure or somewhat samba is no longer honouring read/write/admin lists on share definitions. Using the below share as an example, previously all users could access the share with read access, and those in the ntadmins group had write access. Since the upgrade users can only

I don't mean to be a troll in any way shape or form. I was on IRC last night and I observed the following convo. below. What do you guys make of it ? [02:14] <bkw__> Let me tell you how chidlish digium and Mark Spencer is. I walk into a restaurant with them all here at Astricon wearing my sangoma shirt and he asked me to leave. [02:15] <Dovid> u serious ? [02:15] *** mog

In PHP something like: $pid = pcntl_fork(); if ($pid != 0) { // we are the parent // do parent stuff exit; } // we are the child, detatch from terminal $sid = posix_setsid(); if ($sid < 0) { die; } // do child stuff On 04/19/2019 02:00 PM, Mark Wiater wrote: > On 4/19/2019 1:49 PM, Dovid Bender wrote: >> Mark, >> >> I am using PHP agi and when forking

Hi list, Can anyone who has successfully ran asterisk on a home router please give me the modell number as well as how they did it ? Thanks. Dovid -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061207/92c7f946/attachment.htm

Hi, Does anyone know of any projects that would allow you to use Redis in place of AstDB? By in place of I don't mean for what Asterisk needs but to store values. For instance for CNAM currently we need to use an AGI to connect to redis to pull CNAM. So in place of: Set(CALLERID(name)=${DB(CNAM/${CALLERID(num)})} it would be done with redis for example:

Hi List, I was thinking the other day that even with T.38 there are still some issues with faxing. I was thinking of a protocol that instead of just sending down the fax tones an ATA or "VOIP fax machine" would get the entire fax convert it into some sort of image and pass it down the line to the receiving end. I got the idea from RFC2833. Yes I know that fax machines send bit by bit and

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.

Anyone know of any equipment that I can use to connect a laptop running asterisk to a POTS line (RJ11) ? Regards, Dovid __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com