Re: [asterisk-users] Asterisk Voicemail changes

2017-08-31 Thread Tim Turpin
Is there a way that I can modify the source code for the voicemail application? I need to change some of the options in the user’s interface to make it work like an existing system that I’m replacing. Thanks. Tim From: asterisk-users-boun...@lists.digium.com

Re: [asterisk-users] Asterisk Voicemail changes

2017-08-31 Thread Jonathan H
What about just using the built-in options? In http://doxygen.asterisk.org/trunk/voicemail.conf.html; you can configure the following: ; listen-control-forward-key=# ; Customize the key that fast-forwards message playback ; listen-control-reverse-key=* ; Customize the key that rewinds message

Re: [asterisk-users] Asterisk Voicemail changes

2017-08-31 Thread Tim Turpin
I’m looking to change the TUI, the Telephone User Interface. In other words, instead of pressing ‘1’ to play a message, I want to press ‘7’, etc., etc. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonathan H Sent: Thursday,

Re: [asterisk-users] Asterisk Voicemail changes

2017-08-31 Thread Jonathan H
Well, yes, anyone can recompile anything! But what exactly is it that the current voicemail can't do or be modified to do through normal dialplan and config? On 31 August 2017 at 23:07, Tim Turpin wrote: > Thanks for the info, but not really what I’m looking for. If

Re: [asterisk-users] Asterisk Voicemail changes

2017-08-31 Thread Tim Turpin
Thanks for the info, but not really what I’m looking for. If possible, I’d like to modify the source and re-compile the existing voicemail to make it match what I have today. Thanks. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On

Re: [asterisk-users] Asterisk Voicemail changes

2017-08-31 Thread Jonathan H
What about MiniVM? http://doxygen.asterisk.org/trunk/App_minivm.html Example: http://doxygen.asterisk.org/trunk/Config_minivm_examples.html That said, I don't know if it's actually actively developed or stable (docs last updated 2015 - Asterisk team?) Also make sure your Asterisk is up to date

Re: [asterisk-users] ERROR during high volume MoH dialplan

2017-08-31 Thread Pete Mundy
>> On Thu, 31 Aug 2017, Joseph Smith wrote: >> >> So I am looking for a better way to allow several thousand callers to listen >> to this IVR menu at the same time. > On 1/09/2017, at 7:10 AM, Steve Edwards wrote: > > I'm thinking multiple hosts. > > I'm not a fan

Re: [asterisk-users] ERROR during high volume MoH dialplan

2017-08-31 Thread Steve Edwards
On Thu, 31 Aug 2017, Joseph Smith wrote: So I am looking for a better way to allow several thousand callers to listen to this IVR menu at the same time. I'm thinking multiple hosts. I'm not a fan of 4,000 eggs in one basket. -- Thanks in advance,

[asterisk-users] AST-2017-007: Remote Crash Vulerability in res_pjsip

2017-08-31 Thread Asterisk Security Team
Asterisk Project Security Advisory - AST-2017-007 ProductAsterisk SummaryRemote Crash Vulerability in res_pjsip Nature of Advisory Denial of Service

[asterisk-users] AST-2017-005: Media takeover in RTP stack

2017-08-31 Thread Asterisk Security Team
Asterisk Project Security Advisory - AST-2017-005 ProductAsterisk SummaryMedia takeover in RTP stack Nature of Advisory Unauthorized data disclosure

[asterisk-users] AST-2017-006: Shell access command injection in app_minivm

2017-08-31 Thread Asterisk Security Team
Asterisk Project Security Advisory - AST-2017-006 ProductAsterisk SummaryShell access command injection in app_minivm Nature of Advisory Unauthorized command execution

[asterisk-users] Asterisk 11.25.2, 13.17.1, 14.6.1, 11.6-cert17, 13.13-cert5 Now Available (Security Release)

2017-08-31 Thread Asterisk Development Team
The Asterisk Development Team has announced security releases for Asterisk 11, 13, and 14, and for Certified Asterisk 11.6 and 13.13. The available security release versions are 11.25.2, 13.17.1, 14.6.1, 11.6-cert17, and 13.13-cert5. These releases are available for immediate download at

Re: [asterisk-users] ERROR during high volume MoH dialplan

2017-08-31 Thread Daniel Tryba
On Thu, Aug 31, 2017 at 05:54:43PM +, Joseph Smith wrote: > > So I am looking for a better way to allow several thousand callers to listen > to this IVR menu at the same time. > An alternative that comes to mind is to have 1 conference with 1 channel playing MoH in it and then add callers

Re: [asterisk-users] ERROR during high volume MoH dialplan

2017-08-31 Thread Joseph Smith
It is meant to simulate simultaneous calls on an IVR. I have also tested with a separate set of audio files closer to what the actual IVR menu. This produced the same result. I apologize for not clearly stating the use case up front. I will try to give a bit more detail on that now. I

Re: [asterisk-users] ERROR during high volume MoH dialplan

2017-08-31 Thread Antony Stone
On Thursday 31 August 2017 at 18:15:54, Joseph Smith wrote: > I was hoping Asterisk would handle more than 4k simultaneous calls. I know from experience that Asterisk can handle more than 4k simultaneous calls, however it's an extreme case to have all of them playing music on hold. I think

Re: [asterisk-users] ERROR during high volume MoH dialplan

2017-08-31 Thread Richard Mudgett
On Thu, Aug 31, 2017 at 11:15 AM, Joseph Smith wrote: > Is there any more information I can provide to give insight to these > errors? > > Any further advice on avoiding these during high call volume? > > > I was hoping Asterisk would handle more than 4k simultaneous

Re: [asterisk-users] ERROR during high volume MoH dialplan

2017-08-31 Thread Joseph Smith
Is there any more information I can provide to give insight to these errors? Any further advice on avoiding these during high call volume? I was hoping Asterisk would handle more than 4k simultaneous calls. Thanks Joseph From:

[asterisk-users] Asterisk Voicemail changes

2017-08-31 Thread Tim Turpin
Is there a way that I can modify the source code for the voicemail application? I need to change some of the options in the user's interface to make it work like an existing system that I'm replacing. Thanks. Tim -- _ --

[asterisk-users] SayUnixTime plays nothing if say.conf mode=new and a format is specified

2017-08-31 Thread marek cervenka
hi, is there somebody who is using say.conf mode=new in Asterisk 13? i'm searching for tips what to try in https://issues.asterisk.org/jira/browse/ASTERISK-15421 Marek -- _ -- Bandwidth and Colocation Provided by