similar to: Problem with digits blending on inbound pulsed digits?

To answer my own question for the record: The relevant timing parameters in zaptel.h are #define ZT_MINPULSETIME (15 * 8) /* 15 ms minimum */ #define ZT_MAXPULSETIME (100 * 8) /* 150 ms maximum default, lowered to 100ms */ #define ZT_PULSETIMEOUT ((ZT_MAXPULSETIME / 8) + 50) And the pulse detecion loop that consumes these parameters begins at line 4866 of zaptel.c The

I have a set of analog DID lines coming into my Asterisk box, via a channel bank. The numbers in the DID bank route to various places, including voice lines of various staff. I am using the fax detection engine to intercept faxes accidentially sent to numbers on the DID bank and reroute them to a physical fax set up in the office. I would now like to preserve the received digits and pass them

In the 'me too' vein, I also have an untested 3624 here on the shelf and am waiting on a shipment of T100 cards to play with. Documentation is very hard to come by. Alcatel are certainly the owners of the Mainstreet product line but, without a support contract, any documentation they may have is essentially unavailable as their per-incident fees for support cost more than most of the

Hi I just came accross this Newbridge Mainstreet 3624 but the Alctel site appears to have zip for reference/user manuals Anyone by chance have 1 of these or a url for the docs ? thx

I have an older Newbridge Mainstreet 3624 upon which I'm terminating some analog DID lines. They are effectively loop-start trunks with battery supplied by me (ie. FXO) and consumed by the serving central office. One major part of DID is the requirement for providing Answer Supervision in the form of battery polarity reversal on the analog trunks. Without it wierd things start happening, like

I've got a Newbridge CB hanging on the wall not being used right now and I'd like to hear opinions on using it with Asterisk. If anyone has a manual for it I'd like to get a copy of it. I tried the googling approach but turned up nothing much except a Tech manual if I want to change out control boards. Thanks David Cox Director of Information Technology Ramtex, Inc.

Hi all, I have posted a copy of the 3624 manual on the web. It's 11MB and over 650 pages, so not exactly light reading! You can grab it at http://www.caeveo.com/files/newbridge3624.pdf. Please be kind and save it to your local machine instead of reading it from the web! Thanks! Sean -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type:

Shoval, Interesting Mention. I agree, most people don't have CO exp. And I wish daily I had enough. Understand that what I mean by my e-mail is consumer side FXS ports, in broader terms, I mean, customer picks up a phone line, it signals a channel bank which signals *. 24 of those channels. Not channels equipped to Send Signal to the CO that a loop has been made.. meaning FXO. 24

Hello, I've been experimenting with Asterisk 1.0.6 and a Sipura SPA-3000, and I've run into a couple of questions I haven't yet found clear answers to: It appears that the SPA-3000 has no call progress on it's FXO interface? Asterisk considers a dial() as answered when the SPA-3000 has dialed the number on the PSTN line, not when someone has answered a phone on the

Thanks for your suggestion. I have compiled according to http://www.voip-info.org/wiki/index.php?page=Asterisk+zaptel+pulse +dialing dialing at 10 pps works fine with Asterisk with the newly compiled wctdm. but when I dial at 20 pps, the pulses cannot be decoded correctly. I tried changing the make/break ratio but dialing at 20 pps still has the decoding problem. Does anyone have any

Hi. I have a SPA3102 updated with with Software Version: 5.1.7(GW). I have this settings on Voice/Regional: Interdigit Long Timer: 10 Interdigit Short Timer: 3 Anyway, when hooking up (without dialing anything), the timeout starts after 3 seconds. It's like the Long Timer is unused. After dialing, the Short Timer is also used to timeout. Is that normal? Am I missing something? Thanks. --

Hello, I installed 4.8-RELEASE a few weeks ago, and since I let the effort sit stagnant for a while I decided to do the cvsup/buildworld/buildkernel/installkernel/installworld/mergemaster/MAKEDEV steps to get current. Went fine, rebooted, then noticed that just one CPU was recognized: FreeBSD 4.8-STABLE #0: Thu Jun 19 17:05:20 PDT 2003

# export USB_DEBUG=5 # /usr/local/ups/bin/usbhid-ups -a CP550SLG -DDDDD Network UPS Tools - Generic HID driver 0.34 (2.4.3) USB communication driver 0.31 0.000000 debug level is '5' 0.000426 upsdrv_initups... usb_set_debug: Setting debugging level to 5 (on) usb_os_init: Found USB VFS at /dev/bus/usb usb_os_find_busses: Found 001 usb_os_find_busses: Found 002

Hi list, What paremeter can I change to control interdigit timing? Because my PSTN provider aren't receiving all the digits I dialed on Zap/g1. My Zap/g1 are an E1 (E400P) using E&M immediate sigalling. thanks in advance Eduardo

Hello list, does anyone know how to change the "interdigit timeout" when using Cisco IP Phone 7940/7960 with SIP-Firmware and Asterisk? it's default value is 15 sec. but i have nothing found to set this in tftp-config file etc. Thanks in advance, Roland

I'm having problems with unclean hangups (being read as a flash instead of a hangup?). Can I turn off hook flash recognition in asterisk, but still have the flash button on the analog phone operational? Could I use these settings in zapata.conf to fix my problem? *prewink*: Sets the pre-wink timing. *preflash*: Sets the pre-flash timing. *wink*: Sets the wink timing. *rxwink*: Sets the

I get the following message on a Centos 5 system (really a Trixbox 2.4 build on Centos 5): Jan 14 00:12:28 sip2 kernel: hub 1-0:1.0: connect-debounce failed, port 1 disabled What does this mean? This message occurs about 30 times/sec for about 45 sec. Then my Bluetooth token starts up. Jan 14 00:12:28 sip2 kernel: hub 1-0:1.0: connect-debounce failed, port 1 disabled Jan 14 00:13:00 sip2

Hi, I'm getting a situation where 2 of my trunks (loopstart pots) are occationally bridged together (3624 Newbridge channel bank - asterisk signalling=fxs_ks for trunks) and are staying off hook until I do a 'soft hangup' on one of them. When I listen on a butt set each of the lines are silent (or at least by the time I find them). I am guessing one of the end users isn't properly

Hi, I had previously posted about connecting an AS5300 to * via SIP/H323. I got it to work via SIP, but only 1 call at a time would work, and if a user from the * side hung up, the cisco would'nt catch the hangup. I an now trying to hook up to the cisco via E1, with a Sangoma A101 card in my * box. I would like it such that I call from * via E1/PRI to the cisco, and call out via R2 to

Everyone - thought I would pass on a useful piece of information. Finally got a solution to my phantom ringing problem. Problem - the zhone is triggered into detecting ringing by the Automatic Line Insulation Tests (ALIT or LIT) run nightly automatically by the telco. Here it is twice between 8pm and 9pm on my particular lines. My first approach before I knew specifically the buzzword for