similar to: Support or 12 Bogen HS201C analog phones

Hi, I've got a general question about analog gateways (Xorcom, Audiocodes, Patton, ...) . Is it usual for analog gateways to detect when an analog phone is plugged in or out ? If positive, would it be then useful to send "qualify" queries for each connect phone (I'm implying here that an analog gateway would then reply appropriately for qualify query. Regards --------------

I would recommend that you look at the Pika Technologies Daytona MM board. It has onboard DSP and onboard analog bridging taking up much less horsepower. Please contact me off-list if you would like more information. Bill Hunt Stroudwater Contact Point 207 347 8080 x219 877 870 1234 Toll Free www.stroudwater.com "Realize the Value of Customer Contact!"TM This e-mail is intended

> -----Original Message----- > From: C F [mailto:shmaltz@gmail.com] > Sent: Tuesday, January 11, 2005 4:38 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] internal caller id on analog phones > connected tozap > > How are the analog phones connected to * ? this is where the setting > should be. They're connected to

Hi, We've got IAX softphones, GrandStream VOIP phones and zaptel connected analog phones. Caller id, internally, works just fine (as long as I use numeric only callerids) for IAX and grandstream. Is there a way to have the analog phones' LCD display show the caller id? These are plain old regular analog phone, that if I had callerid from my telco would show on the screen. thanks

I have a TDM20B and asterisk compiled fine. The drivers have been loaded (wcfxs and run ztcfg -vv, I see the drivers loaded with lsmod). Asterisk starts up fine. I am using the default configuration files that are made when you do a "make samples". I was wondering if someone had a link or website that stepped someone through this kind of setup. What I want to do right now, is use a

Hi, There seems to be no solution for call progress on analog lines and using outgoing spool call files . My wave file starts playing before the person has answered the phone so the first part of the message is missed. Can the backgrounddetect app be used for this. I have tried but the message still plays before I answer. I generated 60 seconds wave file. [callprogress] exten =>

hello, I am planning a fairly large hotel VoIP system, using analog phones. It will consist of about 100 analog phones, that must have access to a VoIP server. I am considering an option to use a couple of asterisk boxes, bundled with a total of four TDM2460E cards, and one TDM2451E card. Has anyone on this list done something similar? It would be great to hear some comments regarding a smilar

Hello! Now that I finally have my TDM board working, I want to move forward with using PBX functions. However, it seems cumbersome to use standard POTS telephones with Asterisk. I know that there are many of you installing even large systems based on channel banks and analog telephones. What phones are you using? How do you simulate phone system features on a phone that doesn't have

Googling the archives there is some debate about what are good analog phones to use with *. Aastra seems popular, but they are somewhat pricey and the proprietary seems like it can be a headache. Can someone weigh in on what would be good analog phones for a small office (8 lines and 20 phones) to use with *. So far I'm most impressed with Smartalk primarily bc. they're don't use

I wanted to get everyone's opinion on an issue I am having. I am currently using linksys PAP2NA ATA adapters to terminate analog calls from my auto dialer to the voip termination co. The problem I have is when I call the PSTN everything goes fine until the person being called hangs up the phone. Once they hang up on the PSTN side it takes almost 15-20 Seconds for the ata to see the

Does anyone know the best approach to take for handling 100 analog phones? It seems to me that a chassis like Carrier Access or Adtran would work. The chassis would do much of the hard work of converting the analog sound to data. Any recommendations on hardware for the chassis? ...Jeff

Yo! I changed callprogress to no, and in wcfxo.c source around line 334 i changed the value 32000 and -32000 to 10000 and -10000 because it had something to do with the DC voltage when it was ringing. I found reference here (http://www.voipuser.org/forum_topic_1791.html) with an interesting diagram of wiring that was incorrect for sending voltage to a phone or something like that. So put it

Working A@H 1.3 two 4 port TDM100 WildCards, 3 port FXS, 4 port FXO. I've been able to work the FXO ports out and been able to make and receive calls using softtel PC phones. I'm having difficulty with configuring 4 line non-PBX analogs to function on the FXS side tho.. I've tried using ZAP protocols as some techs have suggested, but all I get are slow busy signals. If someone

A very good day to you all, We can't get the phones to pick up on an incoming call on analog trunks. We're using the digium products in the box, with snom phones internally. This is the output from the asterisk console: linux*CLI> zap show channels Chan Extension Context Language MusicOnHold pseudo pstn-incoming en default 1 pstn-incoming

I am trying to find out the best way to replace one of our hardware PBXs. It currently has 200+ analog phones connected to it. The idea is to take advantage of the already installed phone cables (big building) so I'm trying to avoid the use of ethernet adapters (if possible). However, I'm realizing that it's an expensive setup and will definitely require two or more cooperating

I've got a run of Shielded Twisted Pair (4 conductors) which used to be a Token Ring Network drop and I'm not using it anymore. Would it be decent to replace the ends with normal analog phone connectors and use it for a phone extension, or is STP unsuitable for that? Thanks Steve

I am currently a new asterisk user and new to telephony in general. I have been looking around to implement a solution with asterisk that has many of the nice features of a proprietary PBX for a small office. The features that I am looking for that I haven't been able to find any information on are: - status light indicators for which incoming line in ringing - status light indicators for

Asterisk gurus, To have analog call progress (which as far as I know asterisk does not have right now) does it not come down to 4 states to detect - and I hope that some asterisk gurus can implement those 4 states easily. I see this as 4 states: Busy - a 50% duty cycle Ringing - a % on duty and MORE of a bigger % off duty cycle - or basically not BUSY and not TALK detect Talk - something

Hi, I have a TDM400P (31B) in a PIV 2.8, 512Mb ram, CentOS 4.3, zaptel 1.2.5 and Asterisk 1.2.7.1 and a couple of standard analog phones with a flash button. A hook flash works fine for setting up a 3way call. But pressing the flash button doesn't do anything. The zapata config is below. Anyone have an idea what I'm doing wrong? [channels] context=local usercallerid=yes hidecallerid=no

FC2, Asterisk 1.0.0, Zaptel 1.0.0 TDM400P Port 1 FXS Port 4 FXO Standard analogue handset plugged in with pstn line. Problem: I have 2 analog phones that I use, when plugged directly into pstn line both phones work perfectly, dialing no issues. When I plug the handsets into the TDM400P, one works perfectly the other drops random numbers. Its like the tone is slightly different on the second