similar to: callprogress option

David, I guess I'm a little confused here. Are you asking me to provide a "pencil copy" of an LOA for your review? I don't understand why you need an LOA from us. We need an LOA from you to order circuits that will be billed to us that will be attached to your CFA. It was also my understanding that you had an LOA ready to be given to us, which had already been reviewed by

My outgoing FXO calls are answered but have no audio in either direction if I have callprogress=no in chan_dahdi.conf. If I change to callprogress=yes then the audio returns. My chan_dahdi.conf file is listed below. Can anyone point-out why callprogress=no isn't working? #cat /tmp/a [trunkgroups] [channels] language=en context=incoming toneduration=40 ;usedistinctiveringdetection=yes

http://lists.digium.com/pipermail/asterisk-users/2004-October/065540.html I saw this post about the M(x) option for the Dial command, but I could not find a reply questions posed here. I am wanting to pass the Zap channel that the original call came from to my macro embedded in the Dial command. I've tried to add arguments to the macro by using the syntax M(x,arg1), and I always get the

Here is some more stuff to add to the confusion about the "callprogress" option. I currently have my * system operating with a T100P talking to an ADTRAN TSU600 channel bank with 8 FXO ports connecting to the outside world and Grandstream SIP phones as handset extensions. At first I naively set "callprogress=yes" in zapata.conf. The results were typical of what many

I'm running Asterisk 1.6.2.1 with DAHDI 2.2.1 using a TDM410P. I have callprogress=yes in chan_dahdi.conf because, from everything I've read, it is needed when using call files over PSTN, which I DO use occasionally. I know that callprogress=yes is "experimental" and causes some issues. We've never experienced any problems when making local calls over PSTN with callprogress

Hi, Would anyone happen to have experience with the callprogress option? What I'm trying to do is use a couple normal POTS lines in a queue setup where it will call the queue members to pass the call to them. Of course the problem I'm running into is that under normal conditions the lines register as answered immediately and the caller gets transferred to the ringing line which

Location = US asterisk/zaptel from CVS. Updated last week some time. Currently rebuilding with todays checkout. I have 2 fxo channels hooked up to outside standard Bell South phone lines. If I configure as so [channels] context=pstn group = 1 signalling = fxs_ks callprogress = yes channel => 4,3 Then any call routed from asterisk to the outside line will ring, and can be picked up, but *

I have a couple of X100P's in my system and while on calls they just randomly hang up for no reason. I have tried messing with the busydetect and callprogress setting them to yes and no same and still random hangups. Is there another setting I should be looking at? My zap config looks like. context = inbound-work include => extensions signalling

Using Digium's "Asterisk Developer's Kit (TDM)", I've been trying to make an outside call by copying sample.call to /var/spool/asterisk/outgoing. I want the VoiceMailMain to run when the call is answered. The call is made correctly but, as you probably know, the application starts as soon as the call is made. I see there are two solutions: Using callprogress=yes in

I'm attempting to use callprogress in my system, and I'm having trouble. Callprogress always can tell if the line is busy or ringing, but when the line is answered, the call does not get bridged. Messages showing that "line is ringing" stop in the console and if the called party hangs up, asterisk reports the line is busy. Are there any settings that I could use to help with

Dear Richard Grainger, I am grateful for your reply and guidance. After reading the following article at nginx.com, I realize that I should use a reverse proxy instead of a load balancer. Article: WHAT IS A REVERSE PROXY VS. LOAD BALANCER? Link: https://www.nginx.com/resources/glossary/reverse-proxy-vs-load-balancer/ Presently, I am using pfSense community edition firewall. Can I install

Dear Promise Kumalo, Thank you for the recommendations. So for Exchange equivalents in the Linux world, it is Zimbra and Kolab. ________________________________ From: CentOS <centos-bounces at centos.org> on behalf of Promise Kumalo <promise.kumalo at outlook.com> Sent: Wednesday, July 18, 2018 11:38 PM To: centos at centos.org Subject: Re: [CentOS] Which is better? Microsoft

I'm trying to set up asterisk so any phone connected to channel 1-16 of my Adit600 channel bank can pick up a call coming in on channel 24. I do not wish to ring any of the 16 channels on an incoming call -- this is strictly so I can pick up the line if I see it ringing and wish to answer at work. I have channel 24 in call group 3, and channels 1-16 in pickup groups 1 and 3. However

Hi, What hypervisor/virtual machine monitor and host operating system are you using? On Tue, 24 Sep 2019 at 19:31, J Martin Rushton via CentOS <centos at centos.org> wrote: > > On 24/09/2019 12:16, Turritopsis Dohrnii Teo En Ming wrote: > > Good evening from Singapore, > > > > Anybody downloaded, installed, and tried CentOS 8.0 1905 yet? > > > >

We have a tenant who has been having issues with a congested connection and in trouble shooting it we've noticed that there seems to be a lot of SIP traffic even when none of the phones are doing anything. We've determined that this traffic is mostly INFO packets generated by setting qualify=2000. I understand that 2000 ms is the default value for the qualification parameter but what

I've been trying to get a Win 2000 RAS server working with my asterisk PBX for quite some time, to no avail. I've googled, I've tried loads of configurations, I've rewired phone lines, and still I am not winning the battle. Here's my config. PRI->T400P->Asterisk->T400P->Adtran 750(L36 Firmware)->RAS Server. I have 4 Zap channels signalled FXO_KS to the 750

Hi to all, I'm using a TDM22B. When i establish an external call to the PSTN through an FXO port, I'm not able to know the status of the call (no answer, busy, ...). If I enable call progress (callprogress=yes) in Zapata.conf, I am able to detect the no answer state but if the callee on the PSTN answers the call asterisk doesn't detect that and it jumps to the NOANSWER state and

Guys, this is weird.. Today I started having some problems with calls been dropped. Im suing X100p cards (clones) and I have this setting on my zatala fle: [channels] language=sp signalling=fxs_ks usecallerid=yes cidsignalling=bell cidstart=ring hidecallerid=no callwaiting=yes usecallingpres=yes ;sendcalleridafter=1 callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes

Hi, I'm having random hangup problems with zap channels. If I turn busydetect off in Zapata.conf, * fails completely to detect a user hangup in the middle of a script. On the other hand, if I turn it on, everything works much better, but long calls tend to be hung up without a motive. Any other parameter that I can try? Any #define that I can tweak and recompile? Will callprogress

Noted with thanks. ________________________________ From: CentOS <centos-bounces at centos.org> on behalf of Phil Wyett <philwyett at kathenas.org> Sent: Friday, May 10, 2019 1:56 PM To: centos at centos.org Subject: Re: [CentOS] Now that Red Hat Enterprise Linux 8.0 has been released... On Fri, 2019-05-10 at 05:46 +0000, Turritopsis Dohrnii Teo En Ming wrote: > Good afternoon