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: Asterisk PC with one TE410P 4 port T1 card. One port goes to the PSTN and has my incoming calls. One port goes to the Portmaster which has perhaps as many as 24 "modems" in it. Two ports are free. A user would dial in, get the auto attendant, dial in an extension (for the modem), and be transfered to the Portmaster T1. The Portmaster would establish a 56K modem connection with the caller. Some questions for the gurus out there: 1. Will Asterisk route from one T1 to another "perfectly"? That is, the bits that arrive on the Portmaster would need to be the exact bits sent on the PSTN T1. Seem obvious that this should be so. 2. Would you predict any trouble interfacing a Portmaster to the Digium card? Can it both "sink" a T1 (from the PSTN) and "source" a T1 (to the Portmaster)? 3. Has anyone out there done this? Any suggestions? Which modem banks to buy/avoid? Places to buy used/surplus/cheap ones? 4. Will the above plan actually achieve the "56K" modem connection when routed through the asterisk box? That is, will there be issues of latency/bandwidth in handling the 64 kbps streams? Thanks for everyone's help. -- Mike Ciholas (812) 476-2721 voice CIHOLAS Enterprises (812) 476-2881 fax 2626 Kotter Ave, Unit D mikec@ciholas.com Evansville, IN 47715 http://www.ciholas.com
> 1. Will Asterisk route from one T1 to another "perfectly"? That > is, the bits that arrive on the Portmaster would need to be the > exact bits sent on the PSTN T1. Seem obvious that this should be > so.As of this weekend it does.> 2. Would you predict any trouble interfacing a Portmaster to the > Digium card? Can it both "sink" a T1 (from the PSTN) and > "source" a T1 (to the Portmaster)?Yes. This should be fine. Might consider turning off echo cancellation to be sure.> 4. Will the above plan actually achieve the "56K" modem > connection when routed through the asterisk box? That is, will > there be issues of latency/bandwidth in handling the 64 kbps > streams?In principle it should be able to pass even ISDN calls through. Mark
> > 1. Will Asterisk route from one T1 to another "perfectly"? That > > is, the bits that arrive on the Portmaster would need to be the > > exact bits sent on the PSTN T1. Seem obvious that this should be > > so. > > As of this weekend it does.Can you DACS with it or is it just a passthrough type thing?
> Can you DACS with it or is it just a passthrough type thing?Currently just pass through but drop/insert could be added (configured, presumably, through ztcfg) with relative ease. Mark
> How does the portmaster distinguish between an incoming ISDN call > and incoming analog call? I know this can be done, my local ISP > can handle ISDN and analog calls on the same phone number and it > must know when the call comes in.You could look for HDLC framing...> Whatever method the portmaster uses to tell those apart should be > applicable to * to disable the echo cancel. This might be as > simple as "voice" versus "data" call (is that info provided by > the PSTN?). Is the echo cancel needed on voice ISDN calls? I > can live with no support for voice ISDN calls (can imagine why I > would ever get one).You could also try to look at voice vs. data, yah. Mark
Hi all, I'm not yet an * user, but I'm planning to become one soon. Here is a random question for you * experts: I'm often dialed into a teleconference using an external service (dial toll free number, enter PIN, etc). Sometimes I have an incoming call or want to call someone else and then return to the conference. I can't put the conference on hold because MOH will be injected to the conference (this is extremely rude when it happens because the entire conference is shut down until the holding party comes back!). So I'm stuck. What I want, in addition to MOH, is QOH (quiet on hold). Then I can put the conference on hold, no sound will disturb the other participants, and I can return later. Obviously I can make all holds be quiet (no music), but I would prefer to retain MOH as the basic hold function. The QOH would be an additional "feature". I'm likely to be using Cisco phones if that matters. So, can * do this, and if so, how? Can MOH be selectively enabled/disabled by extension? Are there other ways to solve this problem besides QOH? Thanks for everyone's help. -- Mike Ciholas (812) 476-2721 voice CIHOLAS Enterprises (812) 476-2881 fax 2626 Kotter Ave, Unit D mikec@ciholas.com Evansville, IN 47715 http://www.ciholas.com