similar to: call divert to TRUNK , if one number is unregistered?

Hi, I have installed latest stable version of Asterisk. I registered 1 Quintum Gateway and 5 eyeBeam SoftPhone. But I can make just one call at the same time, if i try to make calls from 2 softphones to anotherone, second caller listens " the person have extension xxxx is on the phone ...." . So we couldn't make two or more calls at the same time for a SoftPhone. What should we

Is it possible that a RTP session between two end users (so i want to use asterisk as a signaling proxy and bypass RTP sessions)? I used "canreinvite=yes" but it didn't work. Description from asterisk conf. File; (canreinvite=yes ; allow RTP voice traffic to bypass Asterisk) Thanks Erdem HAKI - erdemh@tesas.com -------------- next part

Hi list, I set up two asterisk servers , 1001 is the first asterisk server's sip user, and 2001 is the second asterisk server's sip user. Each of them work well, but I don't konw how to connect them. I want to let the user in 1th Asterisk can call the user in 2nd Asterisk. First asterisk server ip : 192.168.3.101 Second asterisk server ip : 192.168.3.102 can someone

Hi, I tried to install R-2.4.1 on cygwin system. "./configure" succeeded, but make failed. Below, I provide the output from the process: error message, and info from configure output, in that order. I appreciate that someone can guide me (technically in-sophisticated) through this process. Again, thanks for your help. Michael Niu (1). Output from make make[3]:

Hello, I am new to Asterisk and new to this list. I got Asterisk setup and running using Asterisk@home, and purchased a PolyCom SoundPoint IP500 phone to test out. I cannot get the phone to talk to the Asterisk box. On bootup of the phone, it tells me that it cannot contact boot server. Why is that? It gets an IP fine, and I have also tried manually setting the IP of the phone and the Asterisk

Hi all of you. I am trying to configure voice mail in asterisk and i am facing problems. I have found following warning message in /var/log/asterisk/messages -------------- No application 'Voicemail' for extension (macro-mainmenu, s, 5) I have configured voice mail accordingly in extention.conf [headoffice] -- ------------ ------------- exten => _63,1,Macro(mainmenu)

Hello Haki I fixed this problem following the instructions in /usr/src/zaptel-1.0.9/README.udev. Regards Cec?lia -----Mensagem original----- De: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]Em nome de Erdem HAKI Enviada em: sexta-feira, 15 de julho de 2005 05:11 Para: 'Asterisk Users Mailing List - Non-Commercial Discussion' Assunto:

Dear Xen community, I have Xen 3.0 installed on RedHat Linux Enterprise RHEL4U2. "xend install" runs fine with no error messages. However, when I start "xm cr guest-vmx.conf" I do not get any new window open for the new guest OS. "xm list" shows that the vmx has started and seems to be working fine (just for testing, when I type "xterm" an X window

Was my question not understood? Hello, I would like to connect 2 asterisk boxes together, so this is my scenario: Asterisk Main: it is connected to many sip providers and its main purpose as a call termination forwarder. Asterisk B: it?s connected to E1, and its purpose to terminate calls. It will receive SIP messages from Asterisk_Main, but there will be no voice traffic

Hello, I have a problem with connecting a Digium X100P card to a Brazilian analog line. Can somebody help me out with this problem? My /etc/zaptel.conf is loadzone=br defaultzone=br fxsks=1 My /etc/asterisk/indications.conf [general] country=br [br] description = Brazil ringcadance = 1000,4000 dial = 425 busy = 425/250,0/250 ring = 425/1000,0/4000 congestion =

Folks, I'm fairly new to this whole Linux UPS thingie, but I'd quite like to have a look at getting my UPS to work under Linux and would be grateful for any help in getting a driver. I have a reasonable working knowledge of Linux and software development, and thus am happy to modify config files, alter kernel settings, etc, although I'm no C guru. I have a Kebo UPS-650D, which

Thanks for the feedback! The attached patch addresses the style issues that have been found. From: Jim Grosbach [mailto:grosbach at apple.com] Sent: Wednesday, April 25, 2012 2:22 PM To: Justin Holewinski Cc: llvm-commits at cs.uiuc.edu; llvmdev at cs.uiuc.edu; Vinod Grover Subject: Re: [llvm-commits] [PATCH][RFC] NVPTX Backend Hi Justin, Cool stuff, to be sure. Excited to see this. As a

I'm filtering that is being deliverd to postfix mail server with amavisd-new . I want spam with spam f level 1 - 8 to ad a tag any everything above to be delete is this posebol? If yes how? Met vriendelijk groet, Bas van Dikkenberg GISkit bv BFVD1-RIPE Tel: +3130-6340430 Fax: +3130-6342433 Prive Tel: +3130-6372769 Mob:

Hello, I'm looking for a SIP to XMPP Jingle voice gateway. I see that Asterisk has Jabber and Jingle support, but it looks like Asterisk acts as a Jabber client. Are there any Jabber server solutions, where Jabber users can call SIP users by using the SIP URI and vice versa? -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/

Is there an up and running provider of SIP termination in Brazil? I know that there are some people building on a SIP termination solution. But who as it up and running ? Best regards, Han -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040921/f1043e19/attachment.htm

Hello, I'm trying Meet Me Feature. I read wiki , searched google and i configured my extension.conf and meetme.conf. But I receive "this is not a valid conference number, please try again" message, so what could be the problem? Thanks for your interest. Erdem HAKI -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello - My single line extension users (connected via channel banks) need to be able to hang up faster. If they just flash the hook it doesn't disconnect right away. Any ideas on how to resolve this? Thanks, Dave -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I am using asterisk 1.4.17 which is connected to a SIP trunk supporting rfc2833 dtmf events. Asterisk stays in the media path. In sip.conf I have set dtmfmode=rfc2833 for the outbound sip proxy (SIP Trunk account) and for SIP clients I have set dtmfmode=info. So when I make a call to a cell number using the sip trunk and then press digits I can see the 2833 dtmf events coming to asterisk

Hi I am using QEMU 0.12.1 as the hypervisor in my RHEL installation of 6.5. I wanted to know if there are any side-effects with respect to VM image corruption when i use SIGTERM signal to kill a qemu-kvm process which effectively stops my VM running on the host. Appreciate if you can provide me some valuable information in this regard. Thanks Jatin

So I made the mistake of buying a Carrier Access channel bank without noticing the page on the wiki about the fact that they don't support disconnect supervision (bastards!). However, apart from that, I do have it working fine for incoming calls. Is there some trick to get asterisk to detect the hangup tones from SBC? I've tried busydetect and callprogress as suggested, but neither