similar to: CT1 and callerid

On Tue, 2003-12-23 at 19:22, Brian West wrote: > I'm just double checking.. I was told it wasn't possible but i'm going to > ask just in case. > > Can you set outbound callerid on a channelized T1? > >I think there is a way to do something like DID with the 4 digits of >DTMF passed before the call. It is unlikely though that you will find >someone interested

Hi All, Can we stop this thread pl. This lady has no intentions to learn asterisk. She is just a troll and wasting our time. With her corporate attitude, what she expects is support that available with paid commercial products. Her company has enough money to buy commercial products, let she go there. Hey lady, whoever u are, dont waste our time. this is not for u. Lets move on to something

If you are using Microsoft Outlook and you are reading this message you need to make 500% sure you are not propagating virii. I posted our support (at) nufone d0t net addy on this mailing list last night and have never posted it in an unprotected fashion like that anywhere else. So far today we have received over a hundred email virii to that address. I suggest you upgrade to a more secure

I know that Asterisk supports DID, but does anyone have documentation on how to write the configuration for it? I'll be trying to setup a hybrid system where some incoming numbers will be DID enabled and others won't, so I'll need to be able to sort between the two, i.e. directly connect the DID dialed numbers and route the others to an autoattendant for extension dialing.

Hello, We're using asterisk with a z-plex 10 and a Zap tormentia card at a non-profit home for the aged. Sometimes when we dial out, our numbers are misintepreted with one number being detected twice. For instance 337-4411 becomes 337-7441 7's 4's and 0's seem to be particularly prone to this issue. Any ideas? I've tried turning down rxgain and txgain. A little debugging

We have 1000's of Remote Call Forward #'s across the USA / Canada, which forward into 1000's of 800 #'s in our call center. Is it possible to automate a solution where Asterisk could dial a given number, record the first 3 seconds of the call, save it to disk, and then go on to the next number, and just do this all day long ? We need to regularly check that the numbers work, for

Hello, It is my understanding that on the softphone side, asterisk is only responsible for establishing the session between two phones. If this is the case, does it matter what type of audio codecs the two phones are using? And if it does matter, are there any codecs that cause problems with asterisk bridging two SIP connections? Thanks for your helpful input, Daniel

To all Asterisk guru's... Here is my question. 1. Asterisk PBX - 5 Trunks / Incoming lines 2. 1 Building - 3 Companies (sharing phone system) Ok that's the basic layout. Here's the low down - Each company will have one dedicated channel for there company. The other 2 channels they want to be set aside as rollover's (rotary). That is not an issue. Where my concern is, how

swar sir, can u please unsubscribe me for your list b.regards jihad chalhoub --- asterisk-users-request@lists.digium.com wrote: > Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, > visit > > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a message

Hi, i am recording a message, but it is always cut off at 40 secs. There are no time out configured. Gabriel -- The educated person is not the person who can answer the questions but the person who can question the answer.

I have set up a couple of test Asterisk servers and have never had a problem with sound. I've just done a fresh install on a dual 1GHZ PIII Asus box running Fedora Core3 with the Digium PCI Dev kit and following all the various Core 3 How-To's. I can make calls ok but when any sound is sent from the Asterisk box such as voice prompts and music on hold the sound is completely chopped up in

Hey all, I am dialing a DID through VoicePulse Connect. The number is answered by a main menu type of IVR. The configuration is as specified in both the wiki and VoicePulses documentation. The call comes through without a problem, but when the caller enter any keys they are either not recieved by * or they are ignored. With SIP I would typically put a dtmfmode= line under the peer and

A friend and I have recently added the ability to announce the callers position in the call queue every x seconds.. or even just inject an anouncement every x seconds. All setup in queues.conf and can be setup per queue. My next project is to add the ability to announce the callers estimated wait time. I want some feedback to see whats the best method to calculate that? What do you want just

<P>hi everyone,</P> <P>I know this issue has been raised many times before, i think still the problem remains. When a call is made through a Zap channel, whether it is actually made or not (irrespective of whether, engaged, busy, or actually answered), asterisk logs it in CDRs as a call made. This makes it impossible to do an accurate billing. Has anybody found a way to overcome

from voip-info wiki Asterisk automatic daily restart To automatically restart Asterisk you can add something like this to cron # Restart Asterisk PBX once a day to prevent any problems from piling up 10 7 * * * root /usr/sbin/asterisk -rx "restart now" >/dev/null 2>&1 or 10 7 * * * root /usr/sbin/asterisk -r -x "restart gracefully" >/dev/null 2>&1

Forwarded back to the list so others might get the benefit of the answers, and I get fact checked by others. -------- Forwarded Message -------- > From: Lee <leeb00@gmail.com> > Reply-To: Lee <leeb00@gmail.com> > To: Steven Critchfield <critch@basesys.com> > Subject: Re: [Asterisk-Users] udev or not? > Date: Fri, 10 Dec 2004 13:00:29 -0800 > On Fri, 10 Dec 2004

Supposed scenario: one PC(2GHz CPU), one card 4E1, and one Internet link. There is somebody he know (has experienced) how many concurrent call (Classical Phone->Voip) can handle Asterisk ? Thanks ! -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

Can T400P be configured for nfas (one d-channel providing signaling for more than one span)? Thank you. Alex Zarubin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030522/f2b637a9/attachment.htm

hi guys, do you have any suggestions on how to connect 2 TE410P via E1? (for simulation and testing purposes) asterisk1 --> TE410P ----> ? ---------> ? ---->TE410P -->asterisk2 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030907/698cd499/attachment.htm

Why does this person have my e-mail address ? ----- Original Message ----- From: <markogift@astriholics.org> To: <hackerwacker@cybermesa.com> Sent: Tuesday, November 23, 2004 1:13 PM Subject: Gift for Mark Spencer > Hello everyone! > > We have been thinking about something that we could do for Mark > Spencer as a holiday gift. We have decided to try to orgranize a