similar to: Asterisk and Lucent APX8100 Universal Gateway

Is anyone using a media gateway with a DS3 for TDM/PSTN access? I am trying to determine the best way to scale Asterisk beyond 4 PRI's. What about the TNT Max? Can it be configured with a DS3 PSTN interface, the appropriate software and hardware, and Ethernet ports to support SIP-PSTN/DS3 media gateway services for multiple * boxes? What about TNT - SER - * ? Is there anyone here with real

Is anyone out there using Lucent brand equipment to handle an incomming DS3, converting all 672 calls to SIP (as G729) and sending those to Asterisk/SER over ethernet? If you are and are willing to speak to my boss about your experiences (over the phone) with it, please contact me off list. We have a possible contract with a local CLEC to handle their long distance, and they want to send to

Hello folks, I've been trying to get T.38 over SIP working with calls terminated by a MAX/Lucent/Ascent TNT. As far as I can tell, SIP and T.38 are actually working perfectly; however, I can't get the TNT to properly terminate a FAX call. Does anyone have a working configuration for SIP and T.38 for calls from a TNT or APX? Here's a brief description/diagram of my test setup:

If you are looking for real high density VOIP termination I would look at > something like a Lucent APX 8000, configure correctly it can pass 2500+ > g.729 calls to the PSTN course we paid lots of $ for ours. > > Chris > Chris, My experience has been that the APX and TNT products require a single SIP proxy, how are you load balancing 2500 calls? If all of the traffic is

Hello all, We are looking for some hardware requirements/recommendations to be able to handle a full DS3's worth of TDM -> VoIP traffic. The DS3 would bring 24 calls per T1 x 28 T1s = 672 simultaneous calls. We would then need to convert those calls into G729 SIP VoIP calls to send to our asterisk box over ethernet. Since everything is going in/out of asterisk is 729, and no features

Does anyone have a MAX/APX with working ingress PRI calling name? I recently acquired a MAX TNT on the cheap and it's integrating fine except for one thing. In the 11.0.0 release notes, it is stated that ISDN calling name will, if present and permitted by presentation flags, be added to the From: and Remote-Party-ID: headers of the INVITE. I'm not able to make this happen. Pcap

Hi everybody, I have a client with redhat 3.8 32bits, on this platform he use an old samba version Could you tell me if the new samba's version (Samba 4) can be installed on this platform. Fr?d?ric DOUCHE Groupe APX Vos Architectes ''Services & Technologies'' Mobile:+33.(0)6.98.05.29.85 29 Rue Denis Papin - Parc Club des Pr?s - 59650 Villeneuve D'Ascq

hey asterisk folk, i'm new to asterisk, but not to freebsd as i've been using it and developing on it for quite a few years. we're planning on deploying asterisk as part of a voip provider's service platform and naturally i'd rather be using freebsd instead of a linux based distro. as such, i am very, very excited that there is work going on towards porting the zaptel drivers

Hi all, I've got an xts time series of stock symbols and closing prices. > head(x) symbol close 2010-01-04 "AFB" "13.46" 2010-01-04 "AKP" "12.80" 2010-01-04 "APX" " 8.78" 2010-01-04 "AYN" "13.15" 2010-01-04 "BAF" "13.50" 2010-01-04 "BBF" "12.86" >

has anyone managed to get these three beasties to work together ? we're using ooh323 from asterisk-addons-1.2.2, asterisk 1.2.6 and microsoft netmeeting default from windows xp. the symptoms are that calls from a SIP client to NetMeeting rings on NetMeeting, but upon answering the call in NetMeeting, no audio is passed between the two. eventually, the call times out and hangs up. on a

hey all, am wondering if anyone has successfuly done a SIP attended transfer using the REFER method (after an INVITE obviously) and the Replaces: header. we're writing our own SIP UAC and the asterisk code seems to support it, but we're not really sure if this is so. we plan on the following call flows: 1. incoming call from exten 1111 is sent to our UAC with Dial() 2. our UAC makes

i have quake v1.06 installed and running fine under windows. however, running it in wine gives a bunch of errors. see below: prophet% wine --winver win98 -- QLAUNCH.EXE QUAKEUDP.DLL QUAKE.EXE -mpath fixme:win32:DEVICE_Open Unknown VxD MGenVxD.VXD. Try --winver nt40 or win31 ! prophet% wine --winver nt40 -- QLAUNCH.EXE QUAKEUDP.DLL QUAKE.EXE -mpath fixme:win32:DEVICE_Open Unknown VxD MGenVxD.VXD.

We have a network of apx 200 NT's,W95 & W98 using an NT machine for WINS resolution. Our dept has apx 15 W95/W98 machines hooked to a hub off a switch from the main network. I would like to setup SAMBA to act as a WIN server for just these machines. Is it possible to do without changing subnets? Now to complicate the question. Network login is done via an NT machine as well. If we

I currently use asterisk version 1.0.10 with AMP 1.0.010, our setup is APX 8000 -> Interaction SIP Proxy 3.0.013 -> asterisk server. When I use Asterisk version 10.0.10 everything works perfectly, however when I use 1.2.4 I lose the ability to receive calls from the PSTN. All I get is the following error in my SIP Proxies error logs: SIPSession::proxyResponseImmediately(): Failed to

Hey there everyone, I appreciated everyone''s help so much in the last few posts I decided to post some more! :) We implemented our bandwidth management machine on the network last week and everything looked great. After looking at it today, anything past the machine is losing major packets and the latency is high. I''m using RedHat 9.0 on a Pentium 4 2.8ghz with 512mb of ram.

Dear all I'd wonder why executable's debug information size is bigger than gcc dwarf debug info. When I compiled ARM hello world code with -g option, the arm based assembly and executable files are bigger than gcc results significiantly.(apx. 3x) (source file size was only 3KB) I think that both clang and gcc use same dwarf format ver2 for debugging. But I don't know why the clang

Hello, I have a Lucent MAX TNT, (DS-3, 672 modem ports, 28 PRIs). I'd like to be able to direct an inbound fax call into my TNT, have it answer the fax and send the image file over to Asterisk, or some other system to deliver to an e-mail address(s). I'm not sure if I need Asterisk to any of the call control or not. I'd also like to setup a print queue and have outbound

I'm looking for someone familiar with setting up some of the more advanced features of the Lucent TNT, preferably someone with knowledge of Trunk Groups and choosing outgoing PRI channels based on call type and perhaps NPA-NXX We currently have 8 PRI's. 7 of them are for our dialup pool, the 8th is for our voip. We currently run the dialup PRI's to a seperate TNT We want to

I'm running SIP between my Lucent TNT acting as a gateway, and an asterisk server. We have a PRI coming into the Lucent. Basically the problem I'm having is mostly on inbound calls but some outbound calls as well. I hear echo and sometimes some weird artifacting on calls coming in from the lucent. Everything routed over IAX to VoIP Jet or Nufone sounds fine. It seems like every 3

Hi, we are currently considering different options for rolling out a large scale IP PBX to handle around 3,000 + concurrent calls. Can this be done with Asterisk? Has it been done before? I really would like an input on this. Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: