similar to: ATA186 & X100P - detect hangup

Howdy y'all, I am using an ATA-186, through Vonage, connected to my X100P. It works well - except, hanging up... The hangup doesn't register, most of the time, with Asterisk. Are there any known tweaks out there that I should look into? I've noted some have altered dsp.c, and rebuilt. Just curious to see what y'all think. Below is my zapata.conf file:: [channels]

After over a month (well, ok, no more than an hour a day :) of planning, getting hardware, tinkering and testing, I'm about to my Ultimate Home Phone System (tm) online. Connectivity to the outside world is provided by: A. 1 POTS phone line connected through an X100P ($11/month, needed to carry DSL) B. 1 Vonage ATA186 connected through an X100P (needed for the rate center :( ) C. 3 Broadvoice

Hi, I have an AsteriskAtHome installation with two X100p clones. Everything has been apparently fine for 5 weeks of use or so, but today, I decided to do some tweaking of my echo cancel parameters, and I realized that all along, one of my cards has been unavailable for outbound calls for some reason. Most of my calls go out one card, which is connected to a Vonage ATA. That is working fine.

Hello, I'm planning to convert my phone system to Asterisk, as I've outgrown my TalkSwitch system. I have a few questions for experienced * users, most of which can be answered yes/no. Current Setup: - Talkswitch 48NLS (4CO/8Ext) phone system. - One CO line, two Vonage lines, one Voicepulse line connected to phone system - A third Vonage line directly connected to a fax machine - A

Hi All, I've got an * PBX up with couple of stations and now I'd like to integrate my Vonage service for outgoing PSTN calls. Is this possible if I have an account with them that uses their hardware box (ATA186) or do I need a 'softphone' account? thanks...

Hi, I have the following basic configuration: - one X100P card - two IP Phones (connected to an ATA186) When an incomming call on X100P, both phones connected to an ATA186 rings (configured like that in extensions.conf). When a call is established between the two ATA phones and a call occurs from the PSTN line, the callerid is correctly displayed on both phones (as internal callwaiting

I've had an spa3k in service here at the house for a while now. After some initial wrangling, it's been working okay. I've had to reboot it a couple times and have noticed something rather annoying though. My setup is pretty simple and, dare I say, common. I have the SPA-3000 "inline" between my incoming POTS line and the internal house phone. It's setup to deliver

I finally got round to installing a second X100P tonight in my asterisk system. Hangup detection, which was already pretty dicey in that it only seemed to work when the other side sends a busy, now doesn't work at all. I'm running the CVS-03/28/03-00:03:08 version. I have tried every combination of callprogress and busydetect and nothing makes a difference. I wonder if others in

I get echo on my X100P for about 15 seconds then it disappears. Believe me, when I get a chance I'm going to tweak it since my wife nags me about it. I converted my house over to *. My setup: Compaq Deskpro EN SFF P3 500 Dual NIC for real IP and internal IP X100P Card for PSTN. Cisco ATA186 (two cordless phones for the old lady) Cisco 7960 Phone Cisco 2621 w/ 2 FXO's and 2 FXS (Fax is

Just for the record and to possibly help with others who get BudgeTone phones. My asterisk box is behind NAT, and I use Vonage, NuFone, and iconnecthere for my "POTS backhaul." On the front end I have an ATA186, a Digium TDM20, and now a BudgeTone 102. The BudgeTone definitely has issues wrt the RTP stream and NATting, although unfortunately I haven't yet been able to dig

Hi.. I've got an asterisk setup with an X100P card installed.. I'm noticing that upon hangup, it takes a good 3 to 5 seconds before asterisk realizes the line has been hung up and drops the call.. this causes my SIP phone to continue ringing, and occassional phantom voice mail messages to be left.. I'm located in good old standard North America, with a regular Verizon residential

I am trying to do a "factory reset" of an ATA186 using the widely-available instructions (basically dialing "FACTRESET#" on the keypad while at the menu prompt). I have done this a number of times before with success, but on this unit the lady spells out "P A S S W D" when I finish up the entry. Does anyone know what to do next? Hitting the star key (which is

Hello, I just bought a X100P from digitnetworks. It is supposed to be a FXO card, but there are 2 rj-11 plug on the card. One is labelled "phone" and the other "pstn". When i plug the "pstn" on the wall and the "phone" on my analog phone, everything (incoming and outgoing calls) works like before (without asterisk). AFAIU, i should have an FXS card in my

I am still haveing problems (echo) with my X100P but I'm thinking it has more to do with the server it is in which is not a negotiable item at this time. My question then is to the use of SPA3000's as a replacement from the FXO standpoint. 1. Can you setup the FXO port to recognize distinctinve ring and call a different context like you can do with Zap channels? Being able to call a

-----BEGIN PGP SIGNED MESSAGE----- Greetings, I've just about got Asterisk up and running and am wondering the following. Currently, I subscribe to both Vonage and Broadvoice and as such, I've got a Sipura and Cisco ATA186. Although I'm sure this is expressly prohibited somewhere in my service agreements, can I reprogram these devices to access my own asterisk server rather than

Hi I'm having a problem with the X100P, where it doesn't detect the caller hanging up before it has answered. The result is that if the caller hangs up just before asterisk answers the line, it still picks up the line and records dialtone in my voice mail. Is there any way to fix this? I'm in the UK, and I use opermode=1 when loading wcfxo. Cheers Rob

Hi, I'm trying to get Asterisk to connect to Vonage (softphone acct) to allow me to place and receive calls. I have successfully configured Asterisk to route inbound calls and send them to the correct extension, but I can't get outbound calls to work. I have Asterisk successfully registering with Vonage, but when an INVITE is sent out, I get a "404 Not Found" back from Vonage

Specs: Fedora Core 3. Dual P3 600 (Dell PEdge 1300) SCSI Disks 1x X100P (channel 1) 1x TDM20 (channels 2+3) 1x Knockoff X100P (channel 4) I am looking to have all local and all toll free calls go outbound through the Copper line, and all long-distance and international to go out through the Vonage line. This way I can eliminate LD on my home line, and pay minimal LD charges through

Hello: I have problems sending dtmf signal to an ATA186 my configuration is: ATA186 --> asterisk --> ATA186 --> FXS to FXO Converter --> PSTN The ATA186 are set to send dtmf RFC2833, but it seems that the ATA186 can't generate dtmf so I can dial to a PSTN number. Is there a setting that can fix my problem, inband dtmf does not work because I'm using G729 codec Thanks

If Asterisk is configured as a simple answering machine replacement with the X100P connected to PSTN line. No FXS ports in the Asterisk machine. Standard phones are connect in parallel with the X100P like you would a regular answering machine. Can Asterisk detect that a phone has been picked up and cancel the outgoing message and/or voice recording? What about if the phones are connected to