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

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

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 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

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

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

So I am designing a railsapp that indexes school organizations such as clubs and frats/sororities. I have decided to represent this as a graph, and that each entity is going to be just represented as a node, and the relationship between each node is going to be an edge. so basically, a frat such as "Alpha Beta Gamma" is represented as a node, and its chapters also as seperate nodes,

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

First question: Do you need to use a special phone to use SIP? My setup is X100P and TDM10B (I have 2 computers with the same setup), will I be able to use SIP? I already have the two computers calling each other using IAX, but I would like to see if I can get them to work using SIP. My current setup: On computer 192.168.10.50 **sip.conf*** [general] port=5060 bindaddr=0.0.0.0

Hi, There is any experience using Asterisk with VMWare? I think about installing a virtual linux box over VMWare and then Asterisk over it. Thanks, Dan

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

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

OK, this is probably a dumb question for a lot of you, but I have no experience with digital lines outside of a tiny bit of ISDN, so I'll just bite the bullet and ask some newbie questions. I am attempting to plan an asterisk installation with about 20 SIP phones and the following incoming lines: 1) At least 6 (as many as 10) lines for voice to the SIP phones 2) 2 incoming/outgoing fax 3)

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

Hi, I was going to order the T100P but it is replaced by TE110P. On further reading the TE110P does not need an external router (The one that separates the data from pstn lines ?). Has anyone got it configured? And on the wiki it says that the drivers for some distros don't exist yet. Is redhat supported? And if I need to connect a fax machine will the FXO cards on ebay work (I know to

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031012/59c113df/attachment.htm -------------- next part -------------- Hello, We are planning to buy following Hardware for Asterisk TestBed. Please let me know if this seems fine to you. 1. IP Phones ( 5 in number) CISCO 7940/7960, SNOM 200, Pingtel xpressa 2. Wildcard T100P interface card,

Hey folks, Using FreeBSD 5.2.1 and I've got the current zaptel driver installed from ports (0.8_1) and current ports asterisk (1.0.1). I've set options HZ=1000 in my kernel config, recompiled and rebooted and as far as I can tell, I've done everything right but when I try to use the conference, the audio is very delayed, choppy and segmented -- totally unusable. At the

hi, i have some newbie questions about channel banks. i have an adtran act-1241 sitting around. it accepts D4 modules, and it contains a number of e&m cards. first of all, how does this thing work? a t1 contains 24 channels, and i noticed that the channel bank has space for 24 cards. what do these cards do? what are their outputs? the ones that are in there have some outputs on the front