similar to: Exposing Zap Channels on Server A to be UsedByServer B

Robert, Essentually I want to be able to have Server B dial the extensions connected to server A as well as route calls to the outbound route on Server A. Server B will have little to no knowledge of what is on Server A. I just want it to dump the calls off. For some reason I keep thinking this was a PRI type of thing. Like there was a module that loaded up as a fake PRI that your

Hello All, I remember there is a way to use two Asterisk servers and allow one to see a virtual trunk that makes it so server B can use the ZAP channels on server A. Does anyone know where I can find this? I am racking my brain trying to remember the terminology. It was like creating a 24 channel virtual T1 connection from server B to Server A that allowed server B to not have any ZAP

Robert, > Is there a product to simulate a PRI trunk? (Robert > Goodyear) TDMoE emulates a T1. ;) Once the TDMoE link is up, Asterisk just sees 24-lines that appear to be a T1 instead of having to deal with all of the complexities of VoIP. This is useful, since probably 75% of the utility of VoIP is really just the fact that it can run over a network. It's also handy because it

Actually doing both sounds good to me. Can you explain further about ringing them all at once? Here is how I tried to make mine work and failed... {global} PHONES0=SIP/2000 PHONES1=SIP/2001 [local] exten => 6001,1,Dial(${PHONES0&PHONES1),20,trf) When I dial 6001 I see my debugger tell me that I am using the wrong syntax. Do you know the correct syntax for ringing them all at once? I

Hi, I recently purchased from a friend 2 Cisco 7905G for testing them with Asterisk. I was able to upgrade one of them with the SIP image, the other hang up during the upgrade process and now it won't boot again. When powered up, the red and green lights keep on and the screen is blank. Does any one know a procedure to fix this ? I do not have a contract with Cisco, I have even call a

That could be it. What I want to do is set a group of callers and have the event cause the phone to ring them in order. I will tie it to my IVR portion and thus I can make sure peole in sales get calls based on our hierarchy in the office. So if I am reading your example right the syntax is.... Exten => 501,1,Dial(SIP/PHONE1&SIP/PHONE2&SIP/PHONE3), rtf) Is that a valid way to cause

Hi, does anyone have the setup for go2call ? I have digium boards and quicknet linejacks and phonejacks. The cards work fine in asterisk without the g729 or g723.1 for the phonejack. I will like to do SIP origination using the codec in the phonejack and linejack g729 or g723 and send the calls to go2call. Anyone has the setup for this ? Or similar setup to a SIP provider using g729 or g723

>From atacomm.... ________________________________ From: Jessee J Holmes [mailto:jholmes@atacomm.com] Sent: Thursday, May 12, 2005 2:24 PM To: Wiley Siler Subject: Re: Got a date yet? No specific release date as of yet; but, we're hoping to have a physical date soon. So far planned release is either in June or July. Right now they developers are cleaning up the echo cancellation

My boss is complaining that the delay between speaking and hearing in a meetme conference is noticeable and doesn't want to roll out our system until I can eliminate the delay. Personally, I don't think the delay is significant, but I don't sign his check. The system consist of 3 1u's, each with a single quad t1 card. Each card has 2 t1's running NFAS. The "t1

So in order to answer the background and backbone questions here is the system as it is. I hope this isn't too much for the list but I'll post it in response to a few inquiries. The current system is quite interesting. We have an office in a town that is about 50 miles from the ski area. The ski area is powered 100% of of generators and the telephone access and internet access goes from

OK. So check out the Wiki here.... http://www.voip-info.org/tiki-index.php?page=Asterisk The archive of this list can be search via google by entering... site:lists.digium.com <some parameter> www.digium.com has a link to all the materials for getting started in the Documentation section of the website. Those are really quite good so I would start there. Most were written prior to

I was trying to establish a TDMoE span of 4 channels between two Asterisk servers. Machine A has a T100P to our PBX. Machine B has no Zaptel hardware. With 4 channels (e&m signalling) the red alarm never clears, and eventually machine A panics. With 24 channels, the TDMoE span seems to work perfectly. Is this hardcoded somewhere? I don't really need a full 1.544 Mbps between my two

LOL - You mean he actually 'met' Newt Gingrich? How dare you not extend him credit!!! I mean seriously... For such a distinguished individual... Hey, not only have I met the heads of several multi-billion dollar corps, I have gotten absolutely blasted drunk with them. So I should get credit, a 40% discount, and your daughters phone number, right??? LOL Seriously, though. I think it

Wiley is definitely right. It would be dangerous not to have a firewall for security reasons. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Wiley Siler Sent: Wednesday, August 10, 2005 2:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Firewall will definately

We have been pretty much able to solve our echo problems, except for the primary mode in which we desire to operate our system. See system diagram at bottom. Prior to making adjustments to cancel echos (all echocancel=no): Call Type Result (Before) --------- --------------- CP <- LEC PRI * TDMoE * FXO -> AP

Hello. I am trying to write an R function to plot the survival function (and associated hazard and density) for a Siler competing hazards model. This model is similar to the Gompertz-Makeham, with the addition of a juvenile component that includes two parameters---one that describes the initial infant mortality rate, and a negative exponential that describes typical mortality decline over the

Before I put myself to the task (next month, maybe) of surveying the CPU costs of transcoding, perhaps someone else has already done this work and would be willing to share it or refer me to a link of previously published data. My reviews of the mailing list with various keywords were unsuccessful in finding adequate references, though I admit I only spent 20 minutes looking. What I seek is

Hello AstLinux seems quite suited for my use. Can you configure more incoming port via a web interface? I'd like to install it to a "normal" hdd. Can that cause any problems? BR Amund Nygaard -----Opprinnelig melding----- Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne av Kristian Kielhofner Sendt: 4. juli 2005 03:23 Til:

Wiley, There are a couple of issues that we saw while not using this option. 1) sip authentication failures as Asterisk is not able to reach Polycom phones. A typical problem description is here: http://lists.digium.com/pipermail/asterisk-users/2004-December/079251.ht ml 2) DTMF issues for Transfers, Hold or simply to dial extensions. This problem is more pronounced when you are using

I have a T400P in my main building M and am going to put asterisk boxes in two buildings A & B. I already have our wan on fiber running between the buildings (including some extra strands) and the distance is beyond straight Ti distance so I would need csu... to use copper. It occurred to me that I could use TDMoE - and it may be less costly. My assumption is that the best to avoid the