similar to: FW: Sending voicemail with qmail and call waiting

you can do that. But are u installing qmail and * on same box. i wont recommend that. i use qmail and *. qmail is strictly for internet email. * is on separate server not exposed to Internet. * box also has sendmail. i hv configured sendmail to use smart host (qmail server). This way its safe and secure. HTH, -B ----- Original Message ----- From: "Ing Isianto Istiadi"

Dear all, I have a question regarding the configuration of *. I have 3 port FXS, and 2 port FXO. I have 4 users that use analog phone connected to FXS (I have 3 phones). I need to limit the user's capability (user A can call International, user B can call long distance, etc). I want to implement the password say to call , he/she needs to puch 9(for outgoing call)2-4 digits password,then

Dear all, I have some questions, I'm sure it's pretty stupid for most of you, but I need you guys to help me. Here are my questions: 1. Music On Hold, it doesn't play any sound on the parked call or hold call. But if I do ps-ax, it shows mpg123 .....( I forgot the exact line). I'm using slackware 9.1 2. I have fxs 3 port, and in my zapata.conf I have included callpickup=1-4,

Dear All, I''m using the kernel 2.6.6, iproute2-2.4.7.20020116, iptables v1.2.9, and gentoo. I have a leased-line 64 kbps. I can see the counter works in iptables, but in the htb, it doesn''t go to the right class (it always go to the default class). Any help will be appreciated here''s my htb conf #!/bin/bash tc qdisc del dev eth1 root tc qdisc add dev eth1 root

Dear all, I'm just finished installing the TDM (2 port) and X100P. I'm using X100P to pstn, and the TDM to the phone. I've loaded the module, and I can also list the card in the /proc/zaptel/ I'm a little confused now. in zapatel.conf, how do I know which channel is which. (TDM or X100P)? Thanks and pardon my English Isianto Istiadi

Hi, I have the following basic configuration: - one X100P card - two IP Phones (connected to an ATA186) When an incomming call on X100P, both phones connected to an ATA186 rings (configured like that in extensions.conf). When a call is established between the two ATA phones and a call occurs from the PSTN line, the callerid is correctly displayed on both phones (as internal callwaiting

Hello, I am looking for information on setting up digium FXO card for use as a PSTN Gateway (H323-PSTN) to work with GNUGk. I am basically looking for the setup and it would be great if anyone can share his experiences with the same. Also, if there are any limitations in going for such a setup and problems that may arise/things that I should keep in consideration. Thanks & Regards, Deepak

Hello, hoping someone out there has some ideas - I have a VOIP line that has call waiting. It is terminated at a Sipura 3000 and the POTS side of that device connects to an FXO port in my * box. I also have a POTS/PSTN line that terminates in another FXO port on my * box. There are two FXS ports which feed cordless phones. I'm using the Zaptel TDM400 card. This gives 2 extensions + 2 lines

I have Asterisk setup in a SOHO environment. I have 2 X100P cards at Zap/1 and Zap/2. I have 1 TDM400P card with Zap/3 - Zap/5. I have subscribed to callwaiting, callerid and calleridcallwaiting from Qwest on the 2 PSTN lines - Zap/1 and Zap/2. My problem is when I'm in an active call to the outside thru Zap/1 or Zap/2, I can't pickup the incoming callwaiting call. I can see the

Hi, I have "callwaiting=yes" in my zapata.conf, and "Call Waiting Serv: Yes" in the PAP2s. However, there's sitll no callwaiting on the PAP2s. Everything else work fine. Any ideas? Am I missing something somewhere? Thank you. AK -------------- next part -------------- An HTML attachment was scrubbed... URL:

I would like to connect a Legacy PBX (Avaya IP Office 406) to an Asterisk Server. The Avaya has 3 CO Ports available. I thought buying a TDM30B card with 3 FXS ports to connect the * to the Avaya CO Ports. Is this the right approach? Does any one have experience with such a configuration? Thanks in advance for all recommandations and suggestions.... Daniel Kocher

Hey guys, can you shead some light on this? I will copy my mgcp.conf and post below, but here is the problem. I can't get call waiting to work with my MGCP device. I already have one call going, and I can hear the second call come in, I flash over to it, but all I get is a dial tone, * puts the 1st call on mute/hold, but I never get the second, and it terminates. I flash back over and pick

I've tried to ask this before but didn't want to waste a lot of bandwidth. However, I have to give the params in order to make the exact question understood. We have 2 ZAP FXO connected to two phone lines(2xX100P) 3 phones connected to a TDM400P with FXS modules Usually a caller will be on ZAP/2 and the first outbound line in the g1 group is on ZAP/1. There are two (main) call waiting

Greetings list, We are having a puzzle with * (asteriskathome 0.5) and SIP phones (SPA2000 ATA's). If callwaiting is enabled, everything (including call waiting) is normal. If callwaiting is turned off, the phone will not accept incoming calls and the call goes straight to whatever is programmed for the busy voicemail response. It doesn't matter whether reinvite is on or off, or

> I have "callwaiting=yes" in my zapata.conf, and "Call Waiting Serv: Yes" > in the PAP2s. > However, there's sitll no callwaiting on the PAP2s. Everything else work > fine. Any ideas? Am I missing something somewhere? Hi Andy, You also need to set "CW Setting: Yes" on the User 1 and User 2 screens. Or, dial from each line, whatever you have set for

I just got a TDM30B, and the documentation doesn't indicate what driver I should be loading for it. Does anyone know? Any tricks I should know about before rolling this thing out? Thanks. B.

This newbie has been trying out Asterisk. It has been both a) surprisingly painful and b) impressive in terms of helpful support from other users. Having got two phones to communicate and then got voicemail MWI going (neither painlessly) I decided the next step was to implement call transfer as per nearly all commercial PBX systems i.e. hold call consult another extension either exit and let

Hi Jean, On Mon Oct 23, 2017 at 10:32:41AM +0100, Jean-Philippe Brucker wrote: > This is version 0.5 of the virtio-iommu specification, the paravirtualized > IOMMU. This version addresses feedback from v0.4 and adds an event virtqueue. > Please find the specification, LaTeX sources and pdf, at: > git://linux-arm.org/virtio-iommu.git viommu/v0.5 >

Hi Linu, On 24/10/17 07:27, Linu Cherian wrote: > Hi Jean, > > On Mon Oct 23, 2017 at 10:32:41AM +0100, Jean-Philippe Brucker wrote: >> This is version 0.5 of the virtio-iommu specification, the paravirtualized >> IOMMU. This version addresses feedback from v0.4 and adds an event virtqueue. >> Please find the specification, LaTeX sources and pdf, at: >>

On 25/10/17 08:07, Linu Cherian wrote: > Hi Jean, > > On Tue Oct 24, 2017 at 10:28:59PM +0530, Linu Cherian wrote: >> Hi Jean, >> Thanks for your reply. >> >> On Tue Oct 24, 2017 at 09:37:12AM +0100, Jean-Philippe Brucker wrote: >>> Hi Linu, >>> >>> On 24/10/17 07:27, Linu Cherian wrote: >>>> Hi Jean, >>>>