similar to: my asterisk experience (long)

Here is my system: POWER SUPPLY 450W ALLIED ATX450P4 R(41) MB NFORCE2 A7N8X DELUXE ASUS RTL(Standard) CPU AMD|2500/333 ATHLON XP BARTON R(Standard) DDRAM 256M|DDR333 PC-2700 -K %(Standard) HD 40GB|WD 7200RPM 8MB WD400JB%(70) VGA ASUS|V8170MAGICII/T 64M MX440SE(58) CD ROM 56X|AOPEN CD-956 RTL(22) Besides the above parts, i have 2 X100P FXO PCI cards, and 1 TDM40B I bought 2 phones from radio

Is there anyway to turn off the call waiting beep in the grandstream and/or cisco ata186? I have a dial statement in my extensions.conf that rings 5 phones at the same time by combining them with the & in the dial statement. i.e.) exten => blah,blah,Dial(SIP/GS1&SIP/GS2&SIP/GS3&SIP/ata186a&SIP/ata186b,25,t) If one of those lines is being used, then the user gets a really

Today class we are going to be talking about the wonderful line of GrandStream products. Or should I say BarbieTone phones? Who else is having MAJOR issues with the grandstream products? How many times have you been told upgrade upgrade upgrade? How many of you have paperweights, granted the phone is light as a feather and couldn't weight papers down in the first place? How about that

I'm having a horrible experience getting a Cisco ATA-186 to work with *. I can make calls from the ATA with no problems. However, incoming calls make the ATA ring once, and then the call is disconnected. I have no problems with my Sipura 2000 or my Grandstream phones. I am running 2.16.1 sip code on the ATA 186. Neither * nor the ATA is behind a NAT. They are both on public IP addresses

If the 1/2 ATA186 device supported IAX I think I'd have found nirvana :-)(especially at $49.95 ;-) Lenny -----Original Message----- From: John Todd [mailto:jtodd@loligo.com] Sent: Monday, April 07, 2003 7:50 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] IP hard phone devices (grandstream) The device at VON would have various parts of it's LCD screen go into

I have an asterisk server running on Redhat 8.0 with a Digium TDM400P w/4 FXO modules (TDM04P) There are 2 lines going into the Digium card. One line is a Vonage digital line, and the other line is a Comcast voice line. I have a SIP Grandstream 100 phone connected to the Asterisk server. I also have IAX configured with FWD. The problem is that on occasionally, after talking for about 20

I have upgraded my grandstream phone from firmware 1.0.3.78 to 10.0.4.30 and now I am having problems with early dial. On the older firmware earlydial worked fine with my asterisk server, but now as soon as I have dialed the number I get a congested tone, and the number 4 flashes up on the LCD screen. Has anyone had this problem, and if so, how do I fix it? -------------- next part

Hi All, I'm looking at getting some Cisco VoIP hardware to play with in combination with a Asterisk server. I've heard that there is beta software available to do SIP on the 7905G. So, I'm thinking of either getting a 7905G or a ATA186. My dillema is, which one to buy? I can get both for about the same price, has anyone had any experience with using a 7905G with Asterisk? On

Hi, I'm away at a conference in Amsterdam. My home is in Cambridge in the UK. On a whim, I tossed an ATA186 and a phone into my bags before leaving home. I was able to plug my ATA186 into a LAN here at the conference and was connected to my home Asterisk in a few seconds. Total time from unzipping my bag to talking to home no more than 15 seconds. OK, so the kit could be more portable,

Hi, I've tried the latest CVS Zaptel and Asterisk after following the Echo fix threads. But echo is the same if not worst. Has anyone managed to alleviate their echo from these latest changes? --------------------------------- ALL-NEW Yahoo! Messenger - sooooo many all-new ways to express yourself -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi I have a brand new ATA186 with the following firmware: Version: v2.15.ms ata186 (Build 020919a) I have been through the archives about how to configure it, but my colorful configuration web page does not have the same fields that people say I need to adjust. Even the examples on Cisco's web site don;t match. For example, I don't have the GtkOrProxy field, which is an important

Hello Everyone - Well, I think I'm getting closer with the asterisk connection. This is my setup and I keep getting this error below in ,my /var/log/asterisk/messages file. I have opened 5060 port on the firewall box. I would this is Warning which I can ignore! But I see the connetcion coming but NO DIAL TONE on mt ata186 box sitting in my 192.168.200.x site! I'm using ATA186(cisco

I've configured Asterisk 0.7.2 to work together with Cisco ATA186 (SIP,G.711. RFC2833) and OpenPhone (H.323, G.711). But there is an issue while calling from ATA186 to OpenPhone via Astrisk - when I press any key on analogue phone connected to ATA, Asterisk shows following message: -- Executing Dial("SIP/519-3781", "OH323/62.213.36.100|20|Tt") in new stack --

I dont have a cisco acount yet can some bady hel me with the ata18x-v2-16-030401a-1.zip file. thanks in advance Rodney Acosta Coya. Dpto. Tecnologia de la Informacion. racosta@moanickel.com.cu Tel:(53)(24) 62 611 -----Mensaje original----- De: Paul Rodan [mailto:asterisk@glitch.cc] Enviado el: Martes, 23 de Noviembre de 2004 11:24 a.m. Para: 'Asterisk Users Mailing List - Non-Commercial

Hi all, Just a little note for the records and archives. We see many small glitches / troubles in the mailing-list but rarely success stories ... Here's one : Asterisk is running perfectly fine in our setup : Debian 3.0 stable / Athlon 1.8, 256 MB Ram / Digium E-100P / Swisscom PRI isdn We have 6 companies (more to come) sharing the system in the building with a total of about 20

I've been trying to use net2phone's sip service at sip.net2phone.com with * but keep getting SIP/2.0 401 Unauthorized. Do you know if this should be possible? So far: I can use an ata186 to connected directly to n2p through sip.net2phone.com without any special settings. I can connect from * to iconnecthere, but, whatever I try from * to n2p produces "SIP/2.0 401 Unauthorized"

Hi Is posible to make a call from site A to Site C, and my question is, the rtp data is from A to C or is from A to B to C Site A Site B Site C ata186<-------->FW<--------->Asterisk<--------->FW<----------->ata186 Thanks -------------- next part -------------- An HTML attachment was scrubbed...

-----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 trouble getting caller*id to appear on my phone connected to an ATA186, and being called from Asterisk. Does anyone out there successfully see callerid on their ata186-connected phone? The "From:" header in the INVITE to the ATA seems to have the "right stuff" - eg From: "Study phone" <sip:6002@195.217.255.45:5062>;tag=as412db061 But

Hi, I saw your posting about the cisco ata186 I2 vs I1 and the simple vs complex impedance. I ordered a cisco ata186 i2 for use in Canada by mistake, didn't know that I needed the I1 version. Will the I2 version work in Canada with regular anlog phones, or will I need to change it. Thanks for your answer. Samy -------------- next part -------------- An HTML attachment was scrubbed... URL: