similar to: Set(CHANNEL(SAY_DTMF_INTERRUPT)=1)

Hi, I am trying to use SAY_DTMF_INTERRUPT with Asterisk 20.0.1. I see that I asked about it here https://www.spinics.net/lists/asterisk/msg174142.html and Sean was nice enough to create a patch. I am trying it on 20.0.1 by doing in the dial plan: CHANNEL(SAY_DTMF_INTERRUPT)=on and I get back: Unknown or unavailable item requested: 'SAY_DTMF_INTERRUPT' When I do: "core show function

I changed in asterisk.conf mindtmfduration = 50 The inter-digit duration is for this function SendDTMF when we dial out dtmf The question is, how do I change it without changing the source code? On Sat, Jun 7, 2014 at 1:00 PM, <asterisk-users-request at lists.digium.com> wrote: > Send asterisk-users mailing list submissions to > asterisk-users at lists.digium.com > > To

> On Jan 17, 2017, at 4:36 PM, Hal Finkel via llvm-dev <llvm-dev at lists.llvm.org> wrote: > > > On 01/17/2017 05:36 PM, Wael Yehia via llvm-dev wrote: >> Hi. Regarding the token approach, I've read some documentation (review D11861, EH in llvm, and Reid and David's presentation) but couldn't answer the following question. >> Does the intrinsic or the

Hi, We are running asterisk 1.2.16 and need to connect two channels which are already established. We are currently using app_meetme to achieve that, but we are sometimes unhappy, as app_meetme provides functionality that produces load that we do not need in our two party conferences. I figured out that there is an alternative called app_changrab.

The other day I was working on an example which used tempfile() to create file for use by the graphics device. And while I love tempfile()---as it is portable and clever and the files get cleaned by R and all that---I noticed one missing feature I would like to see: beside a starting name pattern, and an optional directory, an 'file extension' argument would be nice to have. As e.g. in

/**************************************************************************** * * * Always Be Conferencing (ABC) * * * * Creator: chris @ Penguin PBX Solutions * *

> From the provided documentation I understood that in memory data > structures of a PNaCl program are incompatible to the host program because > ABIs are different (e.g. PNaCl pointers are always 32-bit even when running > on x86_64 platform). > So PNaCl program can't access any data structures of the host program > directly. The only communication way is by using syscalls,

Am 19.06.2013 18:01, schrieb JF Bastien: > > From the provided documentation I understood that in memory data > structures of a PNaCl program are incompatible to the host program > because ABIs are different (e.g. PNaCl pointers are always 32-bit > even when running on x86_64 platform). > So PNaCl program can't access any data structures of the host >

Hello Michael, you are referring to the following behavior - did I get it correctly?: outbound broken: asterisk offers g722 / g711 to provider (callee), callee answers g711. Asterisk now transcodes between caller and callee (g722 <-> g711). inbound works: call from provider: g711 -> asterisk drops g722 and passes g711 to internal callee -> no transcoding. As far as I know,

> You used -noinhibit-exec to ignore the diagnostic, which is usually a bad thing. I certainly agree with that. The point I was trying to make in my original email is that, specifically for kernel objects, this diagnostic is incorrect. R_X86_64_PC32 can be used safely against the symbol foo in that specific context, and should be possible without ignoring diagnostics. I wondered if there

Hello, I have the following situation: I have a PRI with 200 DID numbers and I have set up 200 sip extensions that matches the last 4 digit of the corresponding DID numbers so that when any of the 200 DID number is called, asterisk can pass the call to the respective sip extension. Incomming has been fine. But when making out going calls I want the called party to always see the same number

On Tue, Mar 10, 2020 at 2:30 PM David Blaikie <dblaikie at gmail.com> wrote: > > > On Tue, Mar 10, 2020 at 8:31 AM Alina Sbirlea <alina.sbirlea at gmail.com> > wrote: > >> Hi Dave, >> >> It may be possible to do this with the current API, but what I was >> looking for is a common API for existing block types. For example there is >> no

Hi guys, Greetings for the new year The problem faced by me is simple, I cannot get channel bonding to work perfectly with Xen. I am trying to boost the bandwidth of guest domains by using link aggregation (a.k.a channel bonding). I have tried round robin (mode 0) and high availability (mode 1). I know that mode 1 will not give me bandwidth boost. But strangely once i try mode 1 for the guest, I

Hi guys, Greetings for the new year The problem faced by me is simple, I cannot get channel bonding to work perfectly with Xen. I am trying to boost the bandwidth of guest domains by using link aggregation (a.k.a channel bonding). I have tried round robin (mode 0) and high availability (mode 1). I know that mode 1 will not give me bandwidth boost. But strangely once i try mode 1 for the guest, I

On Tue, Jan 27, 2015 at 05:02:31PM +0100, Hannes Frederic Sowa wrote: > On Di, 2015-01-27 at 09:26 -0500, Vlad Yasevich wrote: > > On 01/27/2015 08:47 AM, Hannes Frederic Sowa wrote: > > > On Di, 2015-01-27 at 10:42 +0200, Michael S. Tsirkin wrote: > > >> On Tue, Jan 27, 2015 at 02:47:54AM +0000, Ben Hutchings wrote: > > >>> On Mon, 2015-01-26 at 09:37

On Tue, Jan 27, 2015 at 05:02:31PM +0100, Hannes Frederic Sowa wrote: > On Di, 2015-01-27 at 09:26 -0500, Vlad Yasevich wrote: > > On 01/27/2015 08:47 AM, Hannes Frederic Sowa wrote: > > > On Di, 2015-01-27 at 10:42 +0200, Michael S. Tsirkin wrote: > > >> On Tue, Jan 27, 2015 at 02:47:54AM +0000, Ben Hutchings wrote: > > >>> On Mon, 2015-01-26 at 09:37

Hi What is the easiest method to set up CDRs for inbound calls? Can this be achieved without use of AGI and programming? Thanks for your help. James -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060215/cbff0714/attachment.htm

Hi, I am running several QEMU VMs on a host machine. There's different persons being responsible for different VMs. I would like to find a way to allow everybody to manage just the VMs they are assigned to (i.e. insert some ISO file, stop the machine and so on) without allowing them to disturb the other VMs. Also, I would like to allow these persons to set up new VMs (there's plenty

I am not proxying the media, but never the less I am forced to store the source media IP in my CDR, for regulatory reasons. Asterisk gets that information when the reinvite comes, but how do I store it? If I don't figure this out my next email will be from Federal Prison. Kindly help me stay away from those guys. Eventually we all need to save that information or we shall not be able to stay

Sorry for the delay; I have been waiting for SourceForge to come back on-line. lvqcl wrote: > 1) It seems that some programs (eac3to) write the value of > WAVEFORMATEXTENSIBLE_CHANNEL_MASK tag with uppercase 'x' (e.g. 0X3) > (see <http://forum.doom9.org/showthread.php?p=1728852#post1728852>) > > Also, from MediaInfo changelog