similar to: setup questions- many users, little use

When people call into my * box over the T1 interface, they get no ring tone. It rings the SIP phone and when the SIP user picks up, both parties can hear each other ok, its just the PSTN user calling in hears no ring. What could be causing this? I tried setting immediate to yes in zapata.conf, but that causes my DNIS and CallerID to stop being available. T100P with E & M Wink start

We currently have a 15-phone system using Asterisk, a combination of analog phones/Grandstream HandyTone-286 and Grandstream BT101s, and 4 X100Ps connected to analog lines. The system works well except for the occasional echo problem. I have all the echo parameters configured, removed all the extra incoming analog lines except to the PBX, etc. following all the advice on the wiki and on the

Hello all ! Can anyone recommend me ATA device that REALLY has T.38 built in. So far I have heard of Telco Systems Access201, which seems to be impossible to bye in Europe (all resselers are droped Telco systems ATAs for some reason (tried in Germany and in UK so far)), and I have heard that SIPURA SPA-2100 should have T.38 built in into newer firmware, but I wasn't able to confirm that

We have issues with all of the SPA-2000 ATAs we have where incoming calls from only one of our Asterisk servers do not complete. Details: 1- On the CLI we see that when the call is pushed to the ATA it shows Busy/Congested 2- We can make calls to this same server just fine 3- We can receive calls from other Asterisk servers running older CVS versions of Asterisk with the same exact ATA

Hi all. First thing: I want to thank you all for your help over the past month as I've been learning asterisk. This is one of the more helpful lists. Even when I ask questions that have answers in the wiki ( which I missed because I've been over studying ). Second thing is this: My office is scouting out VoIP solutions, and I have suggested an asterisk solution. We will be

First of all, I'm not knocking Sipura/Linksys. I have heard very good things about their products. I'm just wondering if they are the only quality shop on the market. I know about the zoom 5801 where you can't dial out the FXO from SIP, only from the FXS port. And I have heard similar about the HT-488 also. I want to know if anyone else makes ATAs where all of the features work

I have about a dozen SIP phones up and working, now I want to connect the asterisk box to our Fujitsu 9600 PBX. I currently have two dial-up servers conencted to the Fujitsu PBX that I built with mgetty/pppd and have the lines provisioned the same way as those dial-up server, ESF, B8ZS, and E&M wink start, so I have confidence in the guys who set up the PBX. I've built a loop back plug

I bought an Aastra PT480 from digium, but I wanted to see if I could get some more help with this before Monday. Any help would be appreciated. I have the phone connected to the TDM400P card, and I also have the T100P and the X100P in the same box. My problem is, it appears as if the phone and asterisk can't understand each other. The port the phone is connected to always remains

Anyone have call waiting working on the ATA-186 connected to Asterisk? Other VoIP phones seem to work, but I can not get the ATAs to allow call waiting. Christopher M Iarocci Network Admin JD Posillico 631-249-1872 X244

I read the thread about what IP phone is best for business deployment with great interest. Our need is slightly different however. We are deploying VoiP as a value-add with our high speed internet service and are having trouble finding the right SIP analog terminal adapter. In order to support people's existing phones and wiring we need to use an ATA. 1) The first priority is we want

Good Day all, I have a Fractional PRI connected to my Asterisk Box via a T100P card. When I initiate a call out to phone number 123-8888 the call sounds great no echo what so ever. If the person at 123-8888 hangs up and calls me right back (same handset on both sides) same trunk line The call always has echo on it. The Asterisk sip extension hears them selves echoing. The remote party

anyone have a recommendation for a place I can buy cheap motherboards that supports those 64-bit 3.3 volt PCI slots for the T100P ? I can't find them at Fry's or anywhere locally. All I can find online is dual processor server boards that are overkill for this application. I would like to use a P3/ P4/ AMD single processor. No Xeons or dual processor junk. Anyone know why digium

I have come to realize that I don't have to have a g729a license in order to make use of an ATA-186 or 7460 connecting to another 7460. I just need to allow the codec in sip.conf. Now what ramification does that have when I dial out over one of my analog line (connected to * by a channelbank and a T100P) using my 7460 or ATA-186. The only benefit I am looking for is reduced bandwidth

Hi I am interested in setting up an Asterisk PBX in my office with digium hardware, and I just have a few questions in regards to what I would need. It is my understanding that an FXO card is used to interface with an incoming/outgoing phone line, and an FXS card is used for interfacing with a phone within the system. Currently we have 4 incoming/outgoing phone lines and would like to have

This is probably going to sound really silly and I must be confused about it. Maybe someone can set me straight. I've been tinkering for a while with * and a number of different FXO/FXS cards, SIP phones, and ATAs trying to get a feel for what works and what doesn't. In the SIP phone group, I have a Cisco 7940G, a Polycom 500, and now an SPA-841. Each of these allow me to configure at

Hi all, I've seen a couple of posts recently from people who are doing something with Asterisk and a T100P and a Nortel PBX. However it's not clear exactly what they are doing. Does anyone know if it's possible to use a T100P or T400P as a DSX-1 interface to connect to a T1/PRI CO module in a Nortel PBX? We have a Nortel Norstar Modular ICS PBX and I'd like to plug its CO

Hi I am setting up a phone system for a small office. The office will have 5-8 phones and a fax line. There are 4 hunt lines coming into the office. We have made no hardware purchase yet. Being an asterisk newbie, before I suscribed to this list I just assumed that I would buy voip phones and connect all the phones to a private ethernet network. However, I see many people inquiring about FXS

I am looking for an ATA that has had very reliable results when passing FAX over IP. I was thinking of testing the Cisco (not Linksys) ATA 186 I1, ATA 186 I2, ATA 188 I1. This is what I'm looking for: FAX -> PTSN -> through Asterisk -> ATA -> Fax Machine. I have QoS from PSTN entry to ATA on the network so I can assure precedence. What has everyone out there been using

Hi All, I'd really like to see a show of hands with regard to people's experience with FXO interfaces. I own a few X100p cards and have had nothing but problems with them. I also took part in Sipura's beta program, for the SPA-3000. While it can be an improvement over the X100p, it presently has echo problems that make it unusable. Sipura has not acknowledged the problem ( at least

I run a SIP-only shop with a 23 channel PRI and single T100P. Here are my configs: /etc/zaptel.conf: span=1,0,0,esf,b8zs bchan=1-23 dchan=24 loadzone = us defaultzone=us /etc/asterisk/zapata.conf [channels] context=default switchtype=dms100 signalling=pri_cpe pridialplan=unknown rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=no hidecallerid=no callwaiting=no