similar to: Cisco ATA 186 / FXO card problem

I've done something similar at home, but made my dialplan such that I can dial either 10 or 11 digits locally. I don't use a "throw away" digit at all. Any 7, 10, or 11 digit call will be appropriately mangled and sent out the PSTN / VoIP provider. ________________________________ From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On

i have been trying to get a newly liberated (from vonage) cisco ata-186 (sip ios v3.1) working properly with asterisk. my client is behind a linksys wrt-54g, which up to this point hasn't proven to be a problem (i have several sipura spa-2000's and polycom phones working just fine behind them). (i'm running cvs-head from yesterday). after looking at the various suggestions,

Calling from a Cisco FXO port to an ATA-186 (SIP 3.1 image) via * (either CVS-HEAD-06/28/04-11:43:41 or CVS-HEAD-07/12/04-15:49:58) I didn't hear any ringing sound & get the following on the console: -- Called 5503 -- SIP/5503-f6b5 is ringing WARNING[-1323201616]: channel.c:1375 ast_indicate: Unable to handle indication 3 for 'SIP/10.10.2.250-9903' -- SIP/5503-f6b5 answered

I am currently running 2.16. Is there good reason to get the update to 3.1? Anything significant? Otherwise I am happy how it is, i just don't want to miss out on anything.

For anyone who is interested, I have a working tftpd (modified wvtftpd) capable of serving configuration, dialplans, and ringtones to Cisco 7960/7940 and ATA-186 devices that are located behind NAT firewalls. As TFTP is not a very firewall/NAT friendly protocol, I had to break some rules to get it to work with these cisco devices. It might cause problems for other TFTP clients, but it works with

In the process of investigating a Cisco ATA 186 that was locked by Vonage, I found that you can still unlock the device yourself. But there's a catch. The device's design has a great plus: a DIP32 *socketed* SST28SF040A flash chip. I found an 8 digit unlock code at 0x03FA71-0x03FA78. I do not know if that is a standard location. If you have the equipment, you're in luck. But

If anyone out there has an ATA-186 that they purchased but cannot use with Asterisk due to it's being locked by Vonage, please contact me off-list. JT

I have a ATA 186 with SIP firmware 3.1 when I changed the configurations to use the g.726 codec I received many erros and the calls doesn't work. I changed the fields: - LBRCodec: 6 <- the code for g.726 - TXCodec: 6 - RxCodec: 6 The errors: Jul 9 13:15:37 NOTICE[1192491824]: rtp.c:500 ast_rtp_read: Unable to calculate samples for format G726 Jul 9 13:15:37 NOTICE[1192491824]:

Does anyone have disconnect supervision working on their TDM400P or X101P cards (to/from the telco)? Googling "telco disconnect supervision," I found the following description: http://mirror.lcs.mit.edu/telecom-archives/telecom-archives/TELECOM_Digest_Online/1559.html [cut] Re: How Do I Get "Kewlstart" From my Phone Company? Paul A Lee Tue, 24 Aug 2004 16:58:16 -0400 ... It

I setup a vonage account last year, and cancelled it last night when I put my asterisk box together and signed up for a Broadvoice account to use with it. Now I would like to use my Linksys router as an MTA, but realize it is still programmed with all of vonage's proprietary information and I do not know how to clear it. I understand that just pushing the reset button will not do it.

Just installed the new 4-port FXO card and moved two pstn lines from existing x100p cards to ports on the FXO card. All zapata.conf entries that were functional on the x100p's were copied to the new FXO channels (including callprogress=no). Observations thus far: 1. asterisk will spontanously decide a pstn call has arrived, and ring the sip phone designated in the dailplan. Verified

Is anyone using Vonage on an FXO port in Asterisk? How well does it work? Specifically, any echo/delay problems? Second part, I am assuming it is possible to separate fxo ports for least cost routing correct? In other words, I would like the routing to be such that any local or 1-800# dials fxo ports 1-4 which is a normal 4 line analog PSTN connection. Any long distance call will try to dial

So Asterisk can't send VOIP calls to Vonage -- but it is still possible to use Vonage for flat-rate long distance by connecting the Vonage AT-196 to an * FXS port, right? The price is an extra D/A <--> A/D conversion. Jim Shilliday IT Director Equal Justice Center 1315 Walnut St. Suite 400 Philadelphia PA 19107 215-238-6970 -----Original Message----- From: Tim Petlock

Hi, I have a legacy Norstar system that I'm looking into integrating with my Asterisk setup. My first attempts have worked, which involves a Wildcard X100P FXO card in the * box connected to the Internal ATA (FXS port) on the Norstar system. Calling from SIP -> Norstar works fine, since the SIP caller initiated the call and generally will be sane enough to hangup the phone when

> -----Original Message----- > From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users- > admin@lists.digium.com] On Behalf Of Robert Withrow > Sent: Tuesday, June 15, 2004 12:32 PM > To: Asterisk-users > Subject: RE: [Asterisk-Users] making * more like a normal pbx (ciscoata- > 186) > > On Mon, 2004-06-14 at 19:34, Reid A. Forrest wrote: > > I've

Any details yet? -Dan

A thought occurred to me to on how to further quantify the impact of "glare" on a properly dimensioned trunk group and debunk the ground start glare concern. A cursory traffic analysis clarifies: 1. Assume you have a two-way trunk group, dimensioned for average busy hour, average busy season for P01 grade of service, Erlang B or Poison distribution. 2. Further assume that the

was breezing over http://voxilla.com/ Looks like a new ATA from the founder of Komodo Technology (aka the Cisco 186) Sipura SPA 2000 http://www.sipura.com/products/spa2000.htm to join the others Cisco ATA 186/188 http://www.cisco.com/warp/public/cc/pd/as/180/186/ 8x8 DTA-310 http://www.8x8.com/products/home_office/dta-310/index.asp.html Grandstream HandyTone 286

I have an adit 600 with one fxo card connected to a Digium single span T1 card. CallerID, disconnect supervision work perfect, however the users complain that they have some sound quality issues, after testing it I realized that whenever one is in a phone call they get like silence between the sounds coming from the other party, almost like a cell phone, in other words if there is no sound coming

Hi all, do you know any firmware release which fixes that issue for cisco ATA186? ATA 186 3.x.x Cisco ATA 186 v3.* CANCEL requests can be sent with a completely bogus URI, making it impossible to cancel a call. bug no workarounds This is from: http://www.communigate.com/SIP/SIPProblems.html. In other words, you make a call, other side doesn't pickup, then you hang up,