similar to: OT: "Integrated Access T1" voice problems -is this possible?

Hello, I am currently pricing out various T1 and PRI options for a client of mine. We need voice and data - we want T's. Whether it be two seperate T's, two superate fractional T's, or one combined fractional T, we need it done. We are getting pricing and one provider is telling us that they have quality issues with the "Integrated Access" product. From what they

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

Pardon my ignorance, but I was hoping someone could clear up something for me. - For a few POTS lines, digium has a single port card for that, or a T1 card to a channel bank. - For 10 or more lines, digium has a T1 or E1 card for that too based on PRI channels - For 100's to 1000's of lines, I suspect a soft-switch is in order??? A traditional phone company will sell: - POTS lines for

> The documentation mentions that the Digium channels can be split into some > voice channels and the remainder of the channels used for routing IP > traffic. > > Does any one have this in use in conjunction with Asterisk? Does it work > well? Would you recommend it for a production server? > > Obviously, if this works, this makes for a cost effective platform where

The "telco guys" probably did something non-industry standard and reversed send and receive in the jack that they plugged the CAT5 into. Sure it works, sure it is easier, sure it is not the correct way of doing things. Thanks, Steve ________________________________ From: asterisk-users-bounces@lists.digium.com on behalf of Lacy Moore - Aspendora Sent: Sat 4/22/2006 2:55 PM To: Paul

I agree. I haven't had a problem using CAT-5, even for long runs, however it's not a real T-Carrier cable and I didn't know how old the PBX is. Paul >I have not in my experience seen any problems with using a Good Quality >Cat5 vs. Cat 3 (telco standard) cable for X-connects. YMMV, but you >should be fine. As far as the shielding goes, I use UTP cables and >Connectors

Can anyone offer up a suggestion on a reliable and cost effective customer premise hardware setup to be able to take an inbound IP T1 and deliver a PRI interface to a remote office? Trying to reduce the amount of hardware required to implement this, right now we use a Cisco router to take the IP T1 in on a serial port and then we go Ethernet to a slimmed asterisk box with a single port T1

Looking for someone with a successful experience similar to this; I have a need to cross connect a 3COM NBX PBX PRI interface to asterisk, but over a long distance. We do not need any IP connectivity and the solution requires G.711u audio so there is no benefit to using IP. Has anyone here successfully cross connected any PBX PRI interface expecting NI2 PRI signaling B8ZS/ESF with an

Hello All, I was wondering if I could do the following on asterisk... Get a T1 between 2 locations, and split it into a data channel of like 1024, and use the rest for voice channels. Has anyone done this and had it working well? Or would I need to get a csu that allows a split into two interfaces? Regards, Greg

Hi I looking a good IAX service for a *emerging * voip provider. Better with a test account to try. Thanks in advance. roberto -- Ing. Roberto Pereyra ContenidosOnline Servidores BSD, Solaris y Linux Soporte t?cnico ISPs Jabber ID: rpereyra@lugmen.org.ar For reliable and professional DNS, use DNS Made Easy! http://www.dnsmadeeasy.com/u/14989 -------------- next part

Hi guys! Unfortunately there is no offtopic list but the subject is somehow related to centos as the OS is/will be centos :) So, under this thin cover i ask : Is it possible that for a SAS controler like LSI 3008 that in specs says that : "This high-performance I/O controller supports T-10 data protection model and optical support, PCIe hot plugging, and up to 1,000 connected devices"

I sent a query about this before, but have some further information and am hoping somebody has a suggestion as to what to try next to debug this. I'm using an Asterisk box primarily for MeetMe conferencing. There are two sources: TDM via two Q.SIG T1's and SIP phones. Conferencing works fine between TDM channels. But when a SIP phone calls the conference, there's no voice path *to*

Hi, For the project I've used the Eicon DIVA card. It has 8 BRI ports, and for about 25% of the time there are 7 or 8 in use. So we want to replace it with an E1 card. Only issue is, replace it with what? The idea we have been playing with was to get a Digium E1 card (we already have bought lot of Quad E1 cards :-) and then just put it back to back against Asterisk server. And instead of

Has anyone on the list used a Soekris engineering PC as a TDM - Ethernet bridge? For example something like a net4801 with a TE110p in it and then using TDMoE to get it into a bigger server where the call processing proper will occur. Anyone know if it might handle a quadspan card ok? (no transcoding, just pure PRI to TDMoE bridging). Craig

I''ve decided to try to make a transparent bridge, that should drop packet''s based on some weird rules. I have googled around a bit and couldn''t find any information about what I would like to do. And therefore I now send a mail to this mailinglist in hope that someone can either help me, or point me in the right direction. What I want to do is the following: Make a

I'm looking to expand my bandwidth for my Asterisk PBX. Why should I choose a T1 over DSL for my asterisk server? I found someone offering T1's for $290 a month + Loops or 3 Meg for $561 a month + Loops. Is this a good deal? Thanks Bart -------------- next part -------------- An HTML attachment was scrubbed... URL:

Is anyone using an inline IDS like hogwash or snort-inline to drop packets in a system running shoreline? I _think_ I see how to configure it, but I''d be really interested in finding a howto or something... Thanks! Mike- -- Mornings: Evolution in action. Only the grumpy will survive. -- Please note - Due to the intense volume of spam, we have installed site-wide spam filters at

Hi all, My data genes [,1] [,2] [,3] [,4] [1,] 25 72 23 55 [2,] 34 53 41 33 [3,] 26 43 26 44 [4,] 36 64 64 22 [5,] 47 72 67 34 stu<-t.test(genes[,1:2],genes[,3:4]) > stu$p.value [1] 0.4198002 i get 1 pvalue for the entire col1:col2 Vs col3:col4. I am trying to get 5 p values for the 5 rows i have. I am trying to avoid a for loop coz my

Anyone successfully implemented a solution for allowing ZapBarge call monitoring only for a specific group of agents calls? The issue I see is that the feature only works on zap channels, and all of the agents (in many cases) are IP phones. Allowing ZapBarge and ZapScan on the TDM PSTN (t100p) interface has privacy issues for senior managers, but would allow all outbound zap calls to be

Hello, I have an Asterisk running with a Sangoma A200 card with Hardware Echo cancelling connected to the UK PSTN. If a PSTN call comes in, voice both ways is OK, however if an outgoing call over the PSTN is made I can hear the other party OK but they can not, they can barely understand what I am saying, my voice is unclear fading and skipping. Internal SIP and IAX2 calls are OK,