similar to: asterisk acts as media gateway for existing pabx ?

I've got an old HFC-S card to play with, and I would like to use it in NT mode. I've a problem only: wiring. I can't fully understand the instructions I was able to find online. Someone can point me to a site which explains the whole procedure clearly (like with some schematics, even in ASCII)? TIA, Simone.

-----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Zoltan Szecsei Sent: zaterdag 16 juli 2005 12:54 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Multiple ISDN BRI Units with Asterisk usingBristuff zaphfc in NT mode? >> Maybe this is rather a hardware question,

Good morning all, I have a question: if I want to connect an * to a traditionnal PABX via TDM04B to get PSTN line (incoming and outcoming call), are FXO enough (no analog phones connected to *)? I just want to add PSTN call (in and out) possibility for SIP or IAX UA. From what I saw here http://www.voip-info.org/wiki-FXO I'm right. Thanks. -- Daniel

Hello, Is there some version of FLAC that allows very very fast encoding (i.e. able to process at least 150 MB / second of .wav input data on a standard computer : laptop computer, Core i5/i7, Windows 7 64 bit, 8 GB RAM) ? (It's ok to have a compression ratio which is a little bit lower than traditionnal FLAC) I'm looking for something which is between FLAC (very good ratio, slower than

Hello, I am a newbie in Asterisk IP PBX but I am very impressed by its functionalities. I have read that It can work over IP network and across the PSTN. I am not very sure how it works over the PSTN.. In case if people have not yet the Internet or SDL access, I would like just to know if it is possible to interconnect two IP LANs using the traditionnal analog Network and Asterisk PBX? In such I

Hi list, we want to connect asterisk to an traditionnal PBX (EADS 6550/Matra). People from telco told that they can't connect two PBX's using E1/T1 or only with QSig signaling. I wanted to use EuroISDN. In this case, it was me told that VN6-VN7 would be used. The PBX has a spare ADQ card installed on which we would connect. Has someone a such working setup? Is it working well? More

Dear Listers, Kurt (subject "CRAN packages maintained by you" sent on 02/07/2015) was mentionning to package maintainers that "with current versions of r-devel, one can get a convenient summary of all undefined globals" and describing how to get a summary conveniently. Now I am using R 3.2.3 but cannot get this by using rcmd check --as-cran mypackage, the

Hello, Is there a way to define the width of the axis tick marks on traditionnal plots? I did not find anything specific on this topic in the help and par(lwd=...) does not affect the tick marks. I guess that using axes=FALSE and recreating the axis with the axis() command could do the trick but I wonder if there is no easier way. Thanks in advance Sebastien

Hi, i. After estimating some coefficients using robust regression with rlm() or lqs(), I wonder if there exist some measures of the goodness-of-fit as those for standard linear model(R2)... or evenly if it's a statistics non-sense to look for since I do not find any mention of that in differents chapters on robust and resistant regression or in severals R documentation (Fox, Ripley and

Hello, Does anybody know if Asterisk can support QSIG protocols to be interconnected with a Traditionnal PABX? (Using a HFC chipset based ISDN card to emulate NT Interface) Thank you in advance ;-) Ignace

Hi All, Can Asterisk dial extension which resides in the PABX? (eg. 2000) Sip Phone <-----> Asterisk <------> ATA (FXS) <------> (CO side) PABX <-----> Extension (eg. 1000) (2100 & 2101) can my sip phone call to pabx extension 1000? What will be my dial plan? I know I can connect to 1000 by

Hello all! I'm VERY new in using VoIP. I'm looking for any tip or trick to connect a physically PABX behind an Asterisk-System(or similar) via an SIP to Analog- or ISDN-Converter. The point is, I _need_ to deliver calls to extensions in the connected PABX directly (in ISDN-speech "DDI" (DirectDialIn)) without intervention of an operator. Is this technically possible, and if

Hi, We are trying to port over a PABX to our network. Both PRI's seem to be live however, whenever someone dials out from the PABX Asterisk happens to report : -- Extension '' in context 'samsungincoming' from '736327438' does not exist. Rejecting call on channel 0/31, span 2 If crc4 is turned off, it reports a yellow alarm. Any suggestions? Regards, Sahil

? Dear all, We are connecting the PABX with Application server.What we are trying is that when a nbr 1800 (this is not registered in PABX) is dialled the pabx should route the call to Application server .The PABX should also have intelligence to route the call by itself for its registered clients.For this scenario to work please guide us what are the files we need to change and other necessary

Hi all, I have an Asterisk server connected in a PABX (TELEDATA) by channel Unicall.. I`m having problem when somebody call from PABX to Asterisk.. Eg: When somebody dial 1234, I received 1111112222333333444444 in the Asterisk CLI... If somebody can help me... or already saw this... Everton Goularth Uberlandia - MG - Brazil _______________________________________________________

Hi guys! I'm having one or two issues with queues hosted by an Asterisk machine where the clients are on a legacy PABX - at least for the interim. I fully expect most of these issues to be non-resolvable, but thought I'd at least ask to find out if there is some way of working around the issues. The legacy PABX is an NEC 7400 ICS connected to Asterisk via an E1 ISDN link. Calls are

Hi it's possible to use a Digium TE110P Single T1 / E1 PCI Interface for supply a E1 link to a old PABX ? Thanks

Hi Folks, I am running a SPA-3000 behind a legacy PABX on an analog line. I have been able to set up a dial plan that sends outgoing calls out to the appropriate VSP depending on prefix, and that part and the incoming call handling works fine. I am now trying to implement call pickup (dial 6*) or manual call forwarding (flash, dial extension). On the first of these I have worked out how to get

Hi all I've got a setup where I use a Sangoma PRI card driven via Sangoma WanPipe to connect to a legacy Samsung PABX (I'm not sure which model) form Asterisk 1.8.11.0. The reason is the customer has a large installed base of Samsung phones physically connected to it and on each users desk. They did not want to spring for a complete replace of all their Samsung phones with generic, and

Hi for all, i 've installed asterisk with isdn trunk with alcatel pabx. When alcatel users dial for external numbers, a channel on asterisk is allocated for dial. When we call to an number that is an IVR the digits isn't recognized by IVR. In sip.conf i putted dtmfmode as rfc... and info, inband is only for 64k codecs, and still don't work. How can i resolve this issue ?? Thanks.