similar to: Asterisk phone system plan - for review!

Hi all, To solve my need for dial in modems, I've hit upon an idea: buy a used T1 "analog" modem bank like a Lucent Portmaster that takes in a T1 and provides several 56K modems. This is overkill for a lightly used dial in service, but the prices of these boxes is so cheap (~$300) with ISPs going away from dial in service that it makes sense. This is how I imagine it would work:

Hi all, I like to have a dial in modem on my toll free number so that when I or my employees travel, they can always get in for net access to read email if no better method is available. Right now, my Panasonic KX-TD1232D PBX receives the call on a POTS line and routes it to an analog modem. The speeds achieved are around 24 kbps due to the digitization of the KX-TD1232D introducing

First of all - Many, many thanks to Mark for his troubleshooting and fix of bug 1320 (FXO_KS signalled Zap Channels on Adtran 750 Channel Bank Stuck in Rsrvd State). I have heard complaints that once every couple weeks, when a user picks up their analog phone (t1 span off of a TE410P into an Adtran 750 with 6 FXS cards), they don't get dialtone, but instead, hear another conversation.

Maybe of you guys know the answer to this: We have T1's that come from both MCI and Global Crossing as channelized (24 Ports per T) with inband (DTMF) delivery of ANI and DNIS (format = *DNIS*ANI*). My old equipment was set for D4, AMI, SF and Wink Start and so is Asterisk. I've moved these T's to Asterisk TE410P and inbound calls are arriving to external voice mail system

Maybe of you guys know the answer to this: We have T1's that come from both MCI and Global Crossing as channelized (24 Ports per T) with inband (DTMF) delivery of ANI and DNIS (format = *DNIS*ANI*). My old equipment was set for D4, AMI, SF and Wink Start and so is Asterisk. I've moved these T's to Asterisk TE410P and inbound calls are arriving to external voice mail system correctly

All, Thanks for taking time to read. I am wanting someone to tell me that I have a clue here and that the following proposed setup would (possibly) work. Currently existing are 4 pots lines with qty. 10 AT&T 954/854 phones. Would like to convert to asterisk and at the same time do a mild feature upgrade. Proposing to get: PIII 800MHz or better >=256MB RAM

Anybody have suggestions on having a 56K dialpool and VOIP connections with an Asterisk box over the same set of PRIs? We've done the PM3 with PRIs for just dialup, but are looking for a way to integrate our Asterisk box and move our voice calls onto the same PRIs. Ian -- Ian White Victoria Free-Net Association email: iwhite@victoria.tc.ca http://victoria.tc.ca/

Nuthin beats an Atlas: http://www.adtran.com/adtranpx/Doc/0/TUA2HMOPDK3KN6S9LM1FH91169/61200305L2-8 A.pdf Telephony Swiss army knife. You can make it do anything. Be prepared to crap your pants when you see the price, though. -----Original Message----- From: Time Bandit [mailto:timebandit001@gmail.com] Sent: Friday, May 26, 2006 9:23 AM To: Asterisk Users Mailing List - Non-Commercial

Hi all, I'm looking for recommendations on ethernet switches for a new install. Ideally would want switches with at least 24 ports, ideally with a GE uplink, and that support PoE (power over ethernet) on every port. I've seen lots of switches, and lots of "power hubs", but the combination, which makes a lot of sense, seems rare. What is out there? Do the switches need to be

Hi all, This is a NEWBIE question, so all you experienced types that are tired of stupid questions can move on... I've pretty much given up trying to do my entire phone system over IP (including local service), so I have to select and provision my local CO lines. I need about 10-12 lines which can be POTS lines, of course. But, I thought, why not get something digital and expandable

Here's my planed setup : PRI from telco <--> (port 1 of A104d) * (port 2 of A104d) <--> PM3 The PM3, for those who don't know is lucent's portmaster RAS dial-up router. I had setup asterisk, zaptel, libpri, wanpipe (as I have sangoma's cards). In wancfg, I have port 1 as TDM_VOICE, with hardware echo on, span 1. Port 2 is TDM_VOICE, without hw echo, Clock as

In the past we have used Adtran Atlas 550's to break out FXS ports for devices like modems. The great thing about the 550 is that internally it is all TDM so there is absolutely zero latency. We are able to use ATA's for faxes and analog phones but devices that use modems, they fail 99.99% of the time when using an ATA. We tried to migrate to TA908 devices; they have FXS ports built

SYNOPSIS Erratic red alarm T1 PRI on asterisk, but zttool running concurrently during alarm shows no errors, irq misses, or alarms, on any span. Using asterisk and quad Digium T405P, configured as follows: Span 1 connects to ISDN PRI (fractional 8 B channels, D channel 24). Span 2 connects to T1 Mux and analog stations. Span 3 connects to ISDN PRI Nortel BCM hybrid key system digital trunk.

to a fax (modulation and demodulation) over VoIP. Try again, the fact that you succeeded twice was pure luck, and as far as I understand that didn't even work out. Switch back to TDM. Your dial up modems want that magic thing called timing and no jitter that only TDM will give you. On Mon, Aug 29, 2011 at 2:56 PM, Aaron Krohn <akrohn at ewebforce.net> wrote: > We have an Ascend Max

I have a dilemma. I am trying to setup an asterisk setup for about 10 people, but with the ability to expand to 100s. We are looking into hardphones to use for our systems. Here are the requirments I am looking for: SIP Not POE Full duplex Speakerphone $100-$300 We would like to have the cordless and corded option if possible (at least for 3 of the 10 people). So here are the

I've been looking at the two phones Aastra 9112i and 9133i for our offices now. I was just curious if anyone could tell me about the ring tones. Specifically: 1) How many ring tones do they come with? 2) Can you upload your own ring tones? 3) If so, what format and how? (wav, mp3, gsm, etc. and TFTP, web management, flash, etc.) Thanks for any help. -- Tim DeBaillie 812-476-2721

Hello, I ftp'ed the binary RPM for samba from us6.samba.org, and when I tried to verify it, rpm said that MD5 was NOT OK. Has anyone else reported this behavior? Thanks, Hiten Sonpal sonpal@ciholas.com Engineer, CIHOLAS Enterprises Voice: 812 476 2721 x 102 2626 Kotter Ave, Unit #D Fax: 812 476 2881 Evansville, IN 47715 http://www.ciholas.com

Hello all, We are having a really wierd problem with asterisk, basically after about 80 calls, asterisk stops responding. The process itself is still running, you can log into asterisk and events are still logged, however all phones lose dial tone and no incoming or outgoing calls work. Calls come in over a ISDN PRI line and are routed over IAX to another asterisk server that has an Adtran

We are getting reports from the receptionist that callers are having dtmf problems (131 gets read as 113, 117 read as 111) - first digit recognized twice, third digit not read at all). I have tried 6 different analog phones, two cell phones (on two different networks), and a digital phone behind another pbx. I can't reproduce the problem at all. Switch layout: {Channelized T1} ->

I'm specing out a new box to act as a tandem switch. It will have a TE410P with 4 x PRI and support IAX connections to four other boxes using predominantly ilbc and/or gsm. It also has 3 IAX trunks to Teliax for call routing also using gsm. No extensions actually terminate on the tandem, they're all switched to other boxes (highly distributed). On the PRI card, one goes to Embarq, the