similar to: SIP routing over IAX2

Sipdiscount has replaced their asterisk servers for another thing. Then, no more iax. Ok, but I can't make calls using sip also... I'm getting a "forbidden" error when using sip1.sipdiscount.com. Anybody got it working? -- Alejandro Vargas

I'm configuring a sip trunk. My problem is if I configure the sip device to dial to a sip phone, it works ok but when I dials to "s" or "7777", asterisk picks up the call immediatly and places it's own ring tone instead of waiting until one of the extension configured for answer the call picks up. Is there a way to avoid it? Is it a problem of the sip trunk? Should I

Thanks Bart. Unfortunately "The VIB module does not comply with the security recommendations imposed by VMWare. You lose VMWare support by installing this package." This pretty much rules out this option. :( We need vCenter support, as well. nomad On Thu, May 14, 2020 at 10:06 AM Bart J. Smit <bart at smits.co.uk> wrote: > This shuts down the hosts directly (by-passing

That's certainly on the list of options. I'm doing data gathering right now to see what options there are. I'd prefer not to rashly decide to run off in a direction only to discover this is already a Solved Problem. nomad On Thu, May 14, 2020 at 1:13 PM Bart J. Smit <bart at smits.co.uk> wrote: > What about the REST API to do a graceful shutdown of your VM’s and the >

I want to do inbound routing of calls comming from sip trunks. Is there a way to force the DID that comes from a trunk that does not have DID support? (something like using the outgoing caller-id for the trunk?) My problem is this: I've got several sip trunks (SPA3000). I want to have an IVR in all but one of them, the one that is connected to a cellular adapter. In this line I want to let it

Hi all. I am attempting to setup Asterisk to allow me to press *1 while in a call to use automon to record the call but have had absolutely no success. Is there a trick to this? In extensions.conf [globals] DYNAMIC_FEATURES=>automon [default] exten => 123,2,Dial(SIP/3000,,wW) ; wW allow one-touch recording During the call, I press *1 but it records nothing. David Morrow

I'm using asteriskathome 2.5. I'm using 2 spa3000 for dialing-out. I configured a trunk for each one with maximun channels=1 and an outbound route that includes both trunks. When a second outgoing call is placed, Asterisk tries to place it in the same that is already in use resulting in a busy tone. ?What can be the problem? -- Alejandro Vargas

How is working the automatic fax detection? I'm making tests in asteriskathome and the ivr plays, the fax sends little bips but asterisk don't detects it as a fax. (for testing I routed one caller id to the ivr). -- Alejandro Vargas

Hi all, I just set up a small asterisk box at home and it works as expected, I have some relatives in USA, I usually use skype to speak to them, is there anyway to connect skype and asterisk. I'd like to use skpe as an extension or a channel with asterisk. Thanks in advance for any suggestion. adriano. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All, We are experiencing a a problem when running calls over IAX with g.729. The call flow is as follows: Sip handset -(SIP)> Asterisk1 -(IAX)> Asterisk2 -(SIP)> Carrier The first Asterisk system is running 1.2 and the second is running 1.0. When using g726 from the handset all the way thru to Asterisk2(then 729 for the carrier leg) calls go thru fine, but when using g729, there

Is anyone using NUT to signal/control vCenter and vServer (6.x) for shutdown when batteries are getting low? Any recommendations? We've found https://4sysops.com/archives/startup-and-shutdown-a-vmware-cluster-with-powercli-and-powershell/ but I'd prefer to do this from the master (which is running CentOS 8) and not pollute it by installing PS on it. thanks, nomad -------------- next part

I've googled around and although I get a lot of hits about postfix smarthost authentication with ssl, I can not find out how to actually accomplish the task. I've read through smatterings of postings from Neophasis and the like searching for just the syntax and what file (I assume it's main.cf) I should be using; however, any smtpd_ lines I have tried result in postfix hanging and

Hi I need to set some of our boxes to send mail through our smarthost. In the sendmail.mc file i have set the smarthost and rebuilt the sendmail.cf file. When i send a mail locally on the box though i get this error Diagnostic-Code: SMTP; 553 5.1.8 <root at localhost.localdomain>... Domain of sender address root at localhost.localdomain does not exist What else do i need to edit as

Hello All, Anyone here has experience of accepting a ilbc call and sending it on g711 or g729 I am having problem in VOICE , call goes though but there is no voice. Senario: Call is coming in from Machine A to Machine B, sending to Machine C Machine B is an asterisk box, transcoding it from IBLC to G711 and g729. Problem: Voice is not appearing on the sip user sitting on machine A Already

Hi, Just downloaded the latest cvs from zaptel on my sparking new Athlon64 Centos4.2 system, but hitting a stumbling block... (sorry for the long post) #make cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -m64 -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -c -o gendigits.o gendigits.c cc -o gendigits gendigits.o -lm ./gendigits cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -m64

Hi all ! Let's say you have about 6 small different companies sharing the same E1 / Asterisk server, and every company needs its own fax number. Since they don't really need fax machines, what would be the most cost-effective way to handle this (keeping fax-privacy at its best) ? Is there a way to configure Hylafax or sth & one modem behind an ATA-186 to email faxes to different

How to read a xml file? I have this XML source: ------------------------------------------------------- <?xml version="1.0" encoding="UTF-8" standalone="yes"?> <fichas> <ficha> <nombre>Gabriel</nombre> <apellido>Molina</apellido> <direccion>Alfredo Vargas #36</direccion> </ficha>

Hi, I want to insert onto database the Current DateTime Value into database field "Store_date" with datatype (DATETIME). How to do tihs? I am usiing hidden variable into my form as follows:- ======================= <%= hidden_field("store", "store_date", :value =>:datetime ) %> ======================= But when I m submmittng form I couldn''t find

I recently had to uninstall PokerStars and WINE will not allow me to preform the reinstallation. I have also tried installing other programs with no success. I contacted PokerStars support but they said it'll be at least 2 days before their Linux guy was in. What they wanted was my WINE log files by going in through the terminal. Thats all they could tell me until 2 days from now. I have no

Hi, I have installed asterisk-1.8.25.0 on an Ubuntu server which has both an ipv6 ip and ipv4 ip(real ip) assigned. And I have a client ubuntu with only ipv4 ip(local ip) installed asterisk-1.8.25.0 . I want to configure the client asterisk with the server asterisk as IAX2 peer and want to connect to the IPV6 ip. I bind the server with ipv6 and also sending the registration request from the