similar to: 3 way calling

I'm running RedHat 9 with a TDM400 (2FXO, 2FXS). Before I implemented Asterisk, some users were using Bell services to set-up 5 way calling: The user would set up a three way call on one line, switch to the second line, set up another 3 way call and then link the two lines together with the Flash key, thus establishing a 5 way call (the user, 2 others on line 1, another 2 on line 2). How

I'm running Asterisk (stable branch downloaded 2-Sep-05) on RedHat 9 and I have a TDM22B installed (TDM400 w/ 2 FXO and 2 FXS ports). Everything seems to work except threeway calling. I can establish a threeway call, but it uses up BOTH FXO lines. Note that I DO have threeway calling active with my Bell service. Here's a typical scenario: 1) Call 765-1574, 2) When they answer, press

Hello, hoping someone out there has some ideas - I have a VOIP line that has call waiting. It is terminated at a Sipura 3000 and the POTS side of that device connects to an FXO port in my * box. I also have a POTS/PSTN line that terminates in another FXO port on my * box. There are two FXS ports which feed cordless phones. I'm using the Zaptel TDM400 card. This gives 2 extensions + 2 lines

Currently running two POTS lines into an asterisk system. Analog and SIP on premises. Being in the sticks, the POTS service is abysmal for quality, especially in the rain. Recently, cable has become available with VOIP phone. The cost savings are attractive as it can replace several independent services for TV and internet (currently satellite). But, I cannot get much out of them, regarding

Please chime in if you've ever wished for digium to make a 4-port daughter board with a combination of 2FXO AND 2FXS ports on the same card. When using the 800 series cards, one must either choose 4-port permutations of FXS/FXO, OR one must give up 2 valuable ports. In other words, when you add ONE 100-series daughter board, you give up TWO of your physical ports. Is there a technical

Hello All, I have a customer site that is using * for ACD. In comming calls are eventually routed to a support rep via a queue. For new accounts the agent needs to be able to send a flash to the PSTN trunk (a POTS line with 3-way calling enabled), dial the number of an authentication center and then connect all three parties together. The trick is that both the agent and the customer need to be

Hello! Attached to the bottom of this e-mail is an edited version of an e-mail I originally wrote to Digium tech support regarding Ouch and Power alarm errors I have been receiving on my TDM400. It contains a great deal of detail regarding my setup. In the end, I have found that one of the 5 phones I'm trying to make work with Asterisk is contributing to the generation of these errors.

Hello All, I have the latest Asterisk and am using the TDM400 card with 4 FXO ports. One problem I have is that I want one of my Queues to go to a cell phone if all agents for the queue don't answer. The problem is that once the TDM400 goes to a POTS line it does not time out after 15 seconds like I have configured (and like the other agents), it will ring to voicemail which doesn't

Hi all - Been using Asterisk installed on Debian and love it. But it's time to rearrange some lines and looking for a few features I didn't enable or have in the dial plan the first time around and wondering if you would recommend doing it through configs again or if one of the prepackaged solutions would more easily support these needs. One that caught my eye was TrixBox but I'd be

Thanks for you comments. I have the one port card now. I plan on purchasing the TDM400. My only question is whether or not the Dell optiplex has pci 2.1 (I think) I have one system running with the single port card. Today I received 2 sipura 2000 modules. Configured the first one tonight... Works great. Pat -----Original Message----- From: asterisk-users-bounces@lists.digium.com

I've tried to ask this before but didn't want to waste a lot of bandwidth. However, I have to give the params in order to make the exact question understood. We have 2 ZAP FXO connected to two phone lines(2xX100P) 3 phones connected to a TDM400P with FXS modules Usually a caller will be on ZAP/2 and the first outbound line in the g1 group is on ZAP/1. There are two (main) call waiting

I am running Asterisk 1.6.1.6 and DAHDI/DAHDI tools 2.2.0.2 compiled from source on a Debian Lenny box, also running FreePBX 2.5.2. I also have an OpenVOX TDM400 card installed with an FXO port on port 1, and an FXS port on port 2. I have a POTS line installed and working on the FXO port. However I've encountered a weird problem that I can't seem to figure out. I have incoming POTS

Hi list, would like to know if some of you have tested asterisk connected to an EADS 6550 analogique PBX (also know as Nexpan50). Our set up is a Dell Optiplex with 1 TDM400 4 FXO, 1 TDM400 4 FXS, (no other card, each of them have their own IRQ) all ports connected to the EADS. We have GS ATA286 as EP. Calling from ATA to ATA give a perfect sound. Calling to PSTN numbers or reverse side,

I've been asked to assess the cost of implementing Asterisk with a single T1 line in one of our offices. I've had plenty of experience w/ TDM400 cards, but T1 is new for me so a couple of questions: 1) Will I need a digital or analogue interface card? I expect digital is the answer, but the Digium web site said something about analogue cards being able to support "provider T1

I have been having a strange issue with my Asterisk 1.2.1 server. I have a TDM400 for the three POTS lines I have and I can receive calls without any problems. But sometimes (not everytime, but 70%) when I dial out of those lines it drops a number and of course gives me the telco error message. How do I know that it's dropping? Because I thought it was a line problem and the tech came out

You have no dial-peer telling the router what to do with the outbound call. http://www.tape.net/~gerry/asterisk/cisco26x0.html At 12:50 PM 9/26/2003, you wrote: >Like Gerry wrote for callerid you need VIC-2FXO-M1 card. > >Right now I am stuck on making outgoing call. > >Could soembody help me with the configuration. > >On cisco I have soemthing like that: > >dial-peer

I have one POTS line going into a TDM400P. Here in Atlanta, we have 10 digit local dialing. I launch a call "Zap/1/7705551212" and it goes thru just fine. The next time I try it, without any modifications, I get a Bell recording telling me that I must dial the area code and seven digit number when placing a local call. It's like Asterisk may be starting the dial before the line

Ok so this is officially driving me crazy. I have an Asterisk 1.6.1.6 install with DAHDI/DAHDI tools (latest) and an OpenVOX TDM400 card with 1 FXO port and 1 FXS port. I have a POTS line from my phone company attached to the POTS line. I have asked for "disconnect supervision" to be provisioned on my line and they claim to have added it. However, my scenario is as follows: I receive a

Hello all, I'm working on a little project right now and have ran into a snag. Was hoping someone would be kind enough to give me a few pointers to help me get past the current issue... I have an AudioCodes MediaPack MP-114 (2FXS and 2FXO... SIP firmware...) that I'm trying to get to play nice with Asterisk 1.4. I've got it to the point where the AudioCodes box picks up

I've been following the list for months, and I have a working Asterisk setup, but it'd be *really* useful to me at this point if someone could summarize when Asterisk has echo problems and when it doesn't. For instance, I usually hear a far-end echo when talking on my 7940, but not when using a POTS phone plugged into a TDM400 FXO port. It doesn't seem to matter if the call