similar to: How do you hangup a call without terminating your session?

In the following example exten => _1NXXNXXXXXX,1,Set(GROUP(outbound)=myprovider) exten => _1NXXNXXXXXX,n,Set(COUNT=${GROUP_COUNT(myprovider at outbound)}) exten => _1NXXNXXXXXX,n,NoOp(There are ${COUNT} calls for myprovider) exten => _1NXXNXXXXXX,n,GotoIf($[ ${COUNT} > 2 ]?denied : continue) exten => _1NXXNXXXXXX,n(denied),NoOp(There are too many calls up) exten =>

Hello guys. Have 2 external numbers that required registration on provider server, trunk1: 73432260005 at 80.75.132.66 trunk2: 73432260050 at 80.75.132.66 Thing is I can?t figure out how to route them to different IVRs by default Asterisk can?t match endpoint Request from '<sip:+ 73432260005 at 80.75.132.66>' failed for '80.75.132.66:5060' (callid:

On 6/17/23 08:47, Steve Matzura wrote: > > Both Background() and WaitExten()  allow the caller to enter DTMF > digits. Asterisk then attempts to find an extension in the current > context that matches the digits that the caller entered. If Asterisk > finds a match, it will send the call to that extension. > > > My question then is, is "*" a valid exension, as

Hello, I have a recording started in the dialplan with the MixMonitor application. I want to be able to stop it during a call and maybe restart it. I tried using the value defined in [featuremap] but it starts another MixMonitor application even if there already one instead of stopping it. Any idea on how I can stop the MixMonitor application while it is running? [featuremap] automixmon =>

Doug, This is where the weeds start growing. On 6/17/2023 4:55 AM, Doug Lytle wrote: > > For both capabilities, you can use Background() instead of Playback() > for audio prompts.  Background() allows for interrupting the prompts > and continue on with your dialplan. > > Understood. From the book: The most common use of the Background() application is to create basic

Hello Doug, Friday, December 13, 2019, 11:03:37 AM, you wrote: >> This is exactly what I do - “press 1 for a human” >> Works great > I do this as well, but I also do a database lookup to see if the number > is on our speeddial list and if so, pass the call directly on without > the IVR prompts. I do something similar for calls without caller ID, but I was still getting

Hi people, I need to detect when the user presses twice *... In the dialplan I added the following, but it doesn't work. Could you help me with that? exten => **,1,..... Anahi Ludue?a _________________________________________________________________ ?Un navegador seguro buscando est?s? ?Protegete ya en www.ayudartepodria.com! www.ayudartepodria.com -------------- next

I'm wondering whether there's a problem with the blindxfer and atxfer commands. I was using Asterisk STABLE and pressing the # key to transfer calls worked fine, except of course when you called up FedEx and they asked "Enter the number of packages, followed by the Pound key". I found on the wiki (http://www.voip-info.org/tiki-index.php?page=Asterisk+config+features.conf) that

Have you enabled DTMF logging and seen the DTMF codes being recognised by Asterisk? I had a bunch of soft phones that I had to change to using ?sip info? for the DTMF signalling as the RFC signalling was not always being recognised. This would cause transfers to appear as if the user had not dialled any digits. On 20/12/2014 20:52, "sean darcy" <seandarcy2 at gmail.com> wrote:

On 9/12/18 1:22 PM, Joshua Colp wrote: > On Wed, Sep 12, 2018, at 2:19 PM, sean darcy wrote: >> I understand that HangUp() hangs up the calling channel. I want to >> hangup the called channel. >> >> SIP/mycall-xxxxx calls and bridges with DAHDI/1-1. >> >> I send SIP/.... to listen to a long, very long, file. > > Define "send". How are you

Hi, I am trying to use the attended transfer. So I put this in my feature.conf: [general] [featuremap] atxfer => *0 blindxfer => #0 I completly restart asterik, and not just make a RELOAD. But during a call, when I press # it runs a blind transfer and if I press * I am disconnected. I am using the CVS version of * get as explain here

Is features.conf included in the cvs as of 8-1-04? I have updated, but am not seeing it?

I understand that HangUp() hangs up the calling channel. I want to hangup the called channel. SIP/mycall-xxxxx calls and bridges with DAHDI/1-1. I send SIP/.... to listen to a long, very long, file. GoSub(play-long-file,s,1) [play-long-file] exten=s,1, ;;; Here I want to hangup DAHDI/1-1, the called channel same=n,Playback(very-long-file) same=n,Hangup() How do I hangup the called channel,

On 12/19/2014 09:42 AM, Rusty Newton wrote: > On Wed, Dec 17, 2014 at 1:09 PM, sean darcy <seandarcy2 at gmail.com> wrote: >> I've got a confbridge set up which works if dialed locally: >> >> -- Executing [266 at internal:1] Answer("DAHDI/1-1", "") in new stack >> -- Executing [266 at internal:2] SendDTMF("DAHDI/1-1",

thanks for help, xend start is OK but xm list failed as follows How to fix it ?? I am look forward to your directions [root@tgh Twisted-1.3.0]# xend start [root@tgh Twisted-1.3.0]# xm list (111, ''Connection refused'') Error: Error connecting to xend, is xend running? _______________________________________________ Xen-users mailing list Xen-users@lists.xensource.com

Hello how can I set the fromuser field of the SIP INVITE when using the Dial()-command in the dialplan ? None of the below Dial() command give the correct result : exten => _XX.,n,Dial(SIP/${EXTEN}:passwdk5j6::user762 at myprovider.biz) exten => _XX.,n,Dial(SIP/${EXTEN}:passwdk5j6::user762 at myprovider.biz/${EXTEN}) exten => _XX.,n,Dial(SIP/user762:passwdk5j6::user762 at

Hello a call goes out and is answered : [Sep 17 11:29:52] VERBOSE[23420][C-00000051] app_dial.c: SIP/myprovider-0000010b is making progress passing it to SIP/mysippeer-00000108 [Sep 17 11:30:05] VERBOSE[23420][C-00000051] app_dial.c: SIP/myprovider-0000010b answered SIP/mysippeer-00000108 [Sep 17 11:30:05] VERBOSE[23522][C-00000051] bridge_channel.c: Channel SIP/myprovider-0000010b joined

Hi, I would like to use the new atxfer (native assisted transfer, see mantis item #3241) , but I've partially been able to make it work. I can receive a call and then having the caller hear MOH while talking with another extension (the one I want to transfer to), but then I can't make the caller and the trasferred talk hanging up or pressing any key combination I'm aware of. My

hello, Is it possible to make attendant transfer (not blind) with ZAP channels ? bartek

? Hi, I am working on a small inbound call center solution that uses an ACD system. I might add an IVR system later on. I only have 2 extensions set up (extensions 1000 and 1001), I want the system to put new calls in a queue if both extensions are busy. I am currently subscribed with a SIP trunk provider and can successfully recieve calls. I want?to design a system where customers?can call my