similar to: Questions about AGI and Wildcard boards

Hello, I've been playing around with * for quite a while now, and have run into a problem that I just cannot seem to figure out. When using * and any IAX client (I have tested with GnoPhone and both clients from iaxclient.sourceforge.net) I have incredibly bad jitter on the connection. What I'm running is a P3-1Ghz machine with 512mb ram for a server. The other end has been

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

We have 4 analog line and 2 analog trunks. On the trunks we have all the DIDs coming into the current phone system. Trying to get everything moved over to Asterisk but having issues picking up the calls on the analog trunk. We can receive calls on the plain analog lines and we can call out on all analog lines and analog trunks. When a call comes in on the trunk line the ZAP channels don't

I am trying to turn up DID trunks with our local phone company but do not know the correct format of extensions.conf to do this.

Hi all, Muppet from the UK asking for help We are just about to have a T1 line installed in our office in Dallas and "Advantex" the supplier has sent a questionnaire asking a number of questions. I have put the question area at the bottom of the email, we will be using Digium's hardware. could anybody help :-) In the UK when I asked for a E1, number of trunks required and the

I recently purchased a Sipura 2000 and connected a phone to it which is connected to my asterisk box via sip. Calls to the Sipura 2000 work fine from another sip device connected through *, from either an fxo or fxs (via adtran channel bank connected to a T400P card) port. However, when a call comes in from the phone company over a T1 with em_w trunks, the phone on the Sipura will ring but I

Does anyone know the default E&M Wink timings for Nortel DID ports? The default settings on Asterisk are: ; prewink: Pre-wink time (default 50ms) ; preflash: Pre-flash time (default 50ms) ; wink: Wink time (default 150ms) ; flash: Flash time (default 750ms) ; start: Start time (default 1500ms) ; rxwink: Receiver wink time (default 300ms) ;

Hi all, I'm sure something similar has been discussed, but one can only wade through the archives for so long. I'm setting up a T1 and my telco has a bunch of questions it wants me to answer. I know much of the TE110p is configurable to do any of this, but I wanted to know if there is an optimal or preferred setup. Any help would be appreciated. Here is the quiz my telco is giving me:

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Not new: I can remember seeing DEC engineers cleaning up the contacts on memory boards for a VAX 11/782 with a pencil eraser c.1985. It's still a pretty standard first fix to reseat a card or connector. On 18/01/16 15:47, Matt Garman wrote: > That's strange, I expected the SMART test to show some issues. > Personally, I'm still

Hello, I'm trying to make my asterisk box to act as a telco, in order to reproduce a US environment in europe. Our telco provider is giving us those settings: ESF B8ZF Inbound = E&M Immediate Outbound sig =Wink Start Yield to Glare = Yes Those trunks are using CAS for signaling. I have tried many configs/combinations in zaptel.conf and zapata-channels.conf In zaptel.conf, when having

Ok, I'm about to take the plunge, and am trying to decide between Channelized T1 E&M and PRI. I'm getting an "Integrated T1" which will have data and voice capability, all plugged directly into my digium single T1 card. In either case the data piece looks pretty straighforward, just setup the channel properly, hand it off to the linux hdlc layer, and route away.... the

Hello All, I've mostly solved my DID problem from a few days ago. Apparenly the lines weren't configured properly. Now heres the next question. 12 E&M wink lines from telco. I have them all plugging into an Adtran 750 with FXS cards. The Adtran ports are configured DPO. How do I signal this from Zaptel. I have them setup E&M in zaptel.conf and EM_W in zapata.conf. They

Greetings, I have several Tyan S2518UGN (Thunder LE-T) server boards that I've been running for a few yrs. I've always used the SCSI on them, so never really used the ATA ports. I once experimented dual-booting NT2003 & FBSD on one of the ATA ports about a year ago. But ran into the problem I'm running into now. Back then I just blamed it on the drive (even tho it was brand new,

Help. Why is it that I can call out from my GSBudgetone SIP phone but the audio is "one-way'? Why is it that when I call my asterisk phone number, I get a fast busy?

When I send a call from my TE410P using E&M, the legacy PBX answers the call but doesn't route it. Any idea what this could be? I assume the digits aren't being delivered properly to the legacy pbx. Any suggestions on what config settings to muck with? Asterisk SVN-branch-1.2-r40901 built by root @ pbx04 on a i686 running Linux on 2007-01-14 14:05:02 UTC zaptel.conf

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I suspect that the gold layer on edge connectors 30-odd years ago was a lot thicker than on modern cards. We are talking contacts on 0.1" spacing not some modern 1/10 of a knat's whisker. (Off topic) I also remember seeing engineers determine which memory chip was at fault and replacing the chip using a soldering iron. Try that on a DIMM!

Hello All, I'm trying to connect Asterisk to an Executone phone system with an analog DID card and I'm hoping someone can help me figure out what I'm doing wrong. The Executone DID card provides battery to the telco, when the telco wishes to dial a DID it goes off-hook, waits for a wink from the Executone and then dials the last three digits on the number with pulse (as opposed

Several of us locally are puzzling over the following problem: We have a central repository of R packages on our linux system. Occasionally, we'd like to install a different version of the same package (such as making updates to the survival package before it is released to the larger group, or a downloaded beta version). When using the library function, it appears that lib.loc is not

One thing I have noticed when encoding VCDs is that often videos with complex backgrounds don't compress nearly as well as those with simple ones. Part of the problem is that TV cameras seam to have the focus so that the entire image is clear rather than focusing on the foreground (and blurring the backgrounds). The result is that a good number of bits are wasted trying to perfectly

Does anyone have any experience getting a Mitel SX-200 EL/ML PBX system to work with *? I'm not getting inbound or outbound calls to work. (Inbound caller gets line busy tone.) SETTINGS FROM MITEL: I built a Crossover cable and connected it like this: PSTN--T1--ASTRISK--T1--OLD MITEL -Crossover Cable Pin-out: 1 - 4 2 - 5