similar to: Re: Asterisk-Users Digest, Vol 19, Issue 58

Reason I ask is I may have a non-voice T-1 replacement project going on and I'm investigating my various options. Costs may be about the same for turn-key and DIY. ---- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ----- Original Message ----- From: <asterisk-users-request@lists.digium.com> To: <asterisk-users@lists.digium.com> Sent: Thursday, February

In all the information on Asterisk it takes about TDMoE to link asterisk servers together. Is this IAX??? How would I use TDMoE. Maybe my first question should be, What is it??? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030516/cd74bddb/attachment.htm

XO fixed my caller ID name. I am using FreePBX and I can include a "wait" to my custom extensions. Is there a way to add a wait to the whole PRI? I assume that if I set immediate to yes, I can then have a "s" extension do the wait, but how would it get from the "s" to the DID extension? (also, I would rather not answer every call) Is there a "magic" spot

-----Mensaje original----- De: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]En nombre de asterisk-users-request@lists.digium.com Enviado el: jueves, 02 de febrero de 2006 10:15 Para: asterisk-users@lists.digium.com Asunto: Asterisk-Users Digest, Vol 19, Issue 15 Send Asterisk-Users mailing list submissions to asterisk-users@lists.digium.com To

Please remove this email from your mailing list. UNSUBSCRIBE Thank you. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of asterisk-users-request@lists.digium.com Sent: Thursday, March 29, 2007 9:14 AM To: asterisk-users@lists.digium.com Subject: asterisk-users Digest, Vol 32, Issue 118 Send asterisk-users

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

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

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

Hello, I've been using TDMoE accross 2 * boxes for months. It was working well, but * would crash when anyone would try to connect via h.323. So, I got the latest CVS release and compiled without h.323 support. After I made and installed the latest CVS the TDMoE channels stopped working. They have been working great (especially after we got echo cancel working them, thanks Mark) and the

Hello, I want to link several * boxes together. Some of them are dedicated as "user" servers (SIP and IAX clients connect to them) and some are used as PRI servers (where the PRIs are hooked onto). I think TDMoE is the only channel type where you can group different Interfaces into a single group. E.g. for using Dial(ZAP/g1/12345), I think you cannot group different IAX accounts

I'm just experimenting with dynamic spans using TDMoE, which looks ideal for clustering Asterisk boxes with low latency (i.e. 1ms for TDMoE compared with 20ms for IAX). When using normal E1s and T1s, I've always used ISDN signalling: either EuroISDN or NI2. So naturally I tried that over the TDMoE. I've found that I get a lot of HDLC errors and the D-channel is up and down like a

Hello, We've just started using TDMoE(local T1s connecting between Asterisk servers in the same building over the LAN) to connect a few of our high-availability servers instead of using crossover T1 cables. The 3 servers we have connected to each other over TDMoE are running just fine and we have no audio quality issues or bandwidth issues, but I'm considering using TDMoE to connect 8

I've been experimenting with the zaptel TDMoE stuff and I've got it all working. Calls go from one asterisk box to the other, with no issues, except they don't bring the callerID along with them. I tried the e&m signalling from the wiki and I thought maybe that had something to do with it, so I just changed it to half fxsks and fxoks and that didn't help me any, I still

I am planning to configure * box A with PSTN interface to route faxes to * box B (running spandsp) over TDMoE. I am using 2xGb bonded NIC's for connection between servers. Was wondering - does anybody have experience with TDMoE over bonded interface - ie. does it work ok?. - does anybody have feedback using this scenario for fax? another question, perhaps someone knows what's the

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

Juan, I dont see people here with "attitude" or any need to shut down this list as you suggested. Contrary to your experience, I find people to be quite helpful here. Steven's comment "Start here and then ask a real question" is definitely not arrogant IMHO. He is just trying to knock some etiquette on Luan. Besides, he did give him all the answer Luan was looking for.

Has anyone every attempted to set up a mini PC to achieve much the same functionality as the fonebridge box? The sort of thing I'm imagining is a micro itx board & case in a completely solid state configuration (flash disk, maybe a psu fan but only if really required), with a TE210P (or equiv) card(s). The sole purpose of this would be to be a bridge between E1 lines and TDMoE. The

I just came across http://www.voip-info.org/tiki-index.php? page=Asterisk%20TDMoE and seemed very interesting. It prompted me to question whether it would be more efficient to do TDMoE or IAX2. The application is very simple. I have two asterisk boxes. One is strictly a gateway to the PSTN using multiple T1s. The second is where all my extensions and dialplan are defined (the workhorse for

If I have a system with 1 machine to handle incoming H.323 calls and then multiple machines to distribute them to T1 ports over TDMoE, where does the codec translation take place? Does it take place in the master system or does it take place in each of the slave TDMoE systems? Also, any idea how many concurrent G.729 calls a system like this would handle?

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