similar to: how can I reduce delays in meetme with zap channels

I have 12 T1's going into 3 servers, 4 in each into "Digium, Inc. Wildcard TE410P Quad-Span togglable E1/T1/J1 card 3.3v (rev 02)" cards. Each "group" of T1's have the primary D on 24 and the secondary D on 96. The first server (ts20) and the last server (ts22) can playback "demo-congrats" fine. The "middle" server (ts21) cannot -- just dead air.

This is weird. If I have 2 members call into meetme using zap PRI channels on the box, they can here each other's keypresses. If I have 2 members call into a separate box using the same PRI's and then forward (dial(iax/...)) them to the previous box into the same meetme, they only hear a minor "squelch" for each other's keypresses. How can I completely mute a

TDMoE was it. Thank you!!!! Wiley ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Robert Goodyear Sent: Friday, June 24, 2005 2:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Exposing Zap Channels on Server A to be UsedByServer B

I have Sipura 841's talking to a CVS-HEAD of August 4. If I disconnect the power to the Sipura, Asterisk does not hang up the channel. My sip.conf for this phone looks like: ; [super1] ; Sipura 841 disallow = all allow = ulaw callerid = "super1"

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

Hi, Can someone point me to a zapata.conf example that will create a single DIAL OUT group including all 4 spans on a TE4XXP? One friend says to change the group number all to "1" on all 4 spans. Another suggestions says it is possible to have these unique groups (1-4) and to combine all 4 into a single group "5". I like the second suggestion best. Can you guide me to the

I'm wondering if tweaking [rx|tx]gain would improve my fax reception success rate. Running ztmonitor when receiving a fax shows 4 "octos" and an * on the RX side and nothing on the TX side. At the end of the page, there's a burst where RX goes to about 1/2 and TX goes to about 2/3 of the range displayed. Any opinions? Thanks in advance,

Is there any way to transfer a call from host to host and keep the call's variables intact? -- specifically, UNIQUE_ID and user created variables like CARD_NUMBER, EXPIRATION_DATE, and CVV2? Thanks in advance, ------------------------------------------------------------------------ Steve Edwards sedwards@sedwards.com Voice: +1-760-468-3867 PST Newline

I'm trying to pass a variable between servers using "setvar" in iax.conf. I have a box (ts2) with a t100p in it. It answers the call and dials another box (ast0) via IAX. I want to pass a variable along with the call from ts2 to ast0. I'm running CVS-HEAD-03/07/05 on ts2 and ast0. ts2's iax.conf: [general] disallow = all allow

I'm trying to spy on an agent (Agent/54321). I can "dial(Agent/54321)" successfully. If I "chanspy(Agent/54321)" or "chanspy(Agent)" all I get is a series of beeps. Any clue where I should start looking? Thanks in advance, ------------------------------------------------------------------------ Steve Edwards sedwards@sedwards.com Voice:

I'm not receiving ANI2 (info digits) on my SBC PRI's. SBC said they're sending them. I called Digium support and was told it is not supported. Is anybody receiving ANI2 on a PRI? Thanks in advance, ------------------------------------------------------------------------ Steve Edwards sedwards@sedwards.com Voice: +1-760-468-3867 PST Newline pagesteve@sedwards.com

A lot of people are going for the "VOIP only" approach, but SBC says you have to have an active analog voice circuit before they will sell you DSL. Does anybody know which DSL providers will sell you DSL without making you pay for a voice circuit? Thanks in advance, ------------------------------------------------------------------------ Steve Edwards sedwards@sedwards.com

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

Our setup: We have a DS3 from Global Crossing terminating into a Adtran MX2800 M13 mux. From there, groups of 4 T1's run into T410P digium cards to 7 individual servers. Each trunk is configured as ISDN PRI, B8ZS/ESF, D-channel being chan 96 with B-channels of 1-95 (we're using NFAS). The D channel is up and there are no alarms. We see the connection on the console from the

This has been an interesting discussion for me (except for the sniping). The last post led me, out of curiosity, to this wiki entry: http://www.voip-info.org/wiki-Asterisk+TDMoE I was unaware of this feature, and it looks pretty good. I've been pondering replacing some T1's by leveraging IP capacity but of course have run up against the QoS issue. My idea was different... I

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

Does anyone have a working NFAS config for Zapata and zaptel for 2 NFAS trunks? First two DS1s on tg 1 and other two on tg2? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050825/95961694/attachment.htm

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