similar to: SIP and special functions - do they work?

Someone recently posted the following list as functions built into * *0# sends flash *8# remote call pickup (pickup phone in your group) *67# disable caller id *70# no call waiting *78# do not disturb on *79# do not disturb off *72# enable call forwarding *73# disable call forwarding *82# enable callerid I'm running a CVS from a couple of weeks ago with multiple C7960's, snom 200,

-----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 All... I have been having a recurring problem with * for several weeks now. I am using it with two ATA186 devices configured for SIP. There is a USB zaptel device plugged in but to the PC but no phone is plugged into it. I have two X100P cards, each with a phone line. About once each day, * will just decide to stop answering calls from the phone lines or from an extension. The *

I am using asterisk at home with a Cisco ATA186 and a clone X100P card. My inbound telco line is plugged into the X100P card. My telco line is also plugged into other phones in the house for now so someone else can answer the phone without asterisk being involved. What I would like to do is if someone has answered the call on a normal phone in the house I would like to be able to join the call

I just saw the S100-FX v2.0 on eBay. I was wondering if anyone has tried it out and what their opinion of it was. ---- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060126/3da2e4e5/attachment.htm

I thought I'd post my experiences for the benefit of anyone else who may be at the point I was when I first started with asterisk. I have 2 incoming analog lines (north eastern U.S., Verizon) where one is set to ring if the first is busy. I bought a bare-bones system from abs-pc with the following components: POWER SUPPLY 450W ALLIED ATX450P4 R(41) MB NFORCE2 A7N8X DELUXE ASUS RTL(Standard)

Op 2013-07-13 om 19:58 schreef Michael Szerencsits: > Hi, > > I tested the following config: > > DEFAULT AutoInstall > PROMPT 0 > LABEL AutoInstall > KERNEL http://server/vmlinuz64 > APPEND initrd=http://server/initrd64.img > root=live:http://server/LiveOS/squashfs.img quiet lang=en > keymap=de-latin1 systemd.unit=rescue.target > system.unit=rescue.target

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

I've got the following setup: IAXy -> Dect Base Station. When you dial from a SIP phone (cisco 7960), the rings with very little delay. However, if you hangup it takes 3-4 rings after hanging up before the dect base station phone stops ringing. The same applies when an incoming call is directed from PSTN FXO -> Dect Base. Is there a fix to this I've looked about on voip-info but

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 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

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

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

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...

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:

I've installed Asterisk and configured an ATA-186 as described at this link: http://www.djernes.org/~shawn/ata186.htm Unfortunately this guide abruptly ends before it explains how to deal with the sip.conf and extensions.conf files. So I left extensions.conf alone and my sip.conf looks like this: [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0