similar to: linux sip or iax phone that will autoanswer and route to console

Hi there, bristuff comes with these two applications - and too little info to understand what they are for. Anyone has a clue and is willing to share it? Thanks, Philipp -= Info about application 'Autoanswer' =- [Synopsis]: Autoanswer a call [Description]: Autoanswer(exten):Used to autoanswer a call for an extension. -= Info about application 'AutoanswerLogin' =-

My colleague left our company, then I have to manage all our phones lines and asterisk: please, apologize me because I'm 'absolute beginner' about voip/asterisk!! Well... all seems work fine; we have some queues and some agents; the "music on hold" works fine when the agent press the hold button on the phone (thomson); the agents have the 'autoanwser' flag

Hi There, i think all VOIP clients for Linux are unusable! i got testet: Linphone + Linphonec all in version 12.2 Kphone gophone and other... the only programm that is usable is gnomemeeting... does anybody knew some other tools? Best Regards, Mark

continuing discussions of 79xx issues. i've seen referenced and am experiencing difficulty getting a 7970 to work behind NAT to a public asterisk server. i am successful with 7960s. 1. SIP load is 70.8-3-3SR2S 2. config works fine if 7970 is connecting to an asterisk server a local LAN (same subnet) 3. when debugging it in a NAT'd environment I see the register and

Hello-- What I'd like to do: Use IAX softphones running on computers, in Auto-answer mode, with sound going to speakers, as a sort of public announcement system. What isn't working: Well, my first experiment was to set up the MeetMe system described on the Wiki... This works fine for voice announcements. You pick up a phone, dial the right extension, and an agi is fired up to put files

In console.conf.sample it says run the command "console list available" CLI command. It does not seem to be present: console list available No such command 'console list available' These are the only console commands I see: console answer Answer an incoming console call console autoanswer Sets/displays autoanswer console dial

A friend pointed out this url http://www.cisco.com/univercd/cc/td/doc/pcat/clmn32.htm where it lists intercom/auto-answer as being a feature in Cisco Call Manager (which as I understand it, uses SIP predominately for handsets). I've come across comment somewhere that intercom isn't supported in the SIP spec. Does anyone know if the apparent capability of Intercom being available in SIP

On a Cisco 7960 Auto Answer is only configurable using the phone (not via TFTP), does anybody know if it is possible using sip notify or any other way but walking over to the phone?

Hello; To let the Phone answer automatically, this can be configured from asterisk (at the sip.conf for the phone)? Or it has to be from the IP Phone? Because, some phones does not support auto answer, also we do not need to do it for each Phone. Regards Bilal -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I have tried to install this phone for hours now and I can't get it working. Maybe someone can help me :) I have searched for more info from everywhere but there isn't much about 7910 :( >From the CLI I get this: NAME ADDRESS MAC Reg. State ================ =============== ================ ========== telefon --

Hello I am trying to connect linphone 0.12.2 to an * 0.9.1 box over a LAN using the console version of linphone. both boxs are using the latest alsa drivers on a LFS kernal 2.4. I am running into errors with codec compatability between linphone and *. A point to note is that I am able to connect to asterisk using other sip phones noteably sjphone however linephone is giving me

Hello friends, I'm trying to make a simple call from asterisk CLI, but is quite confuse i followed the information here: http://www.voip-info.org/wiki/view/Asterisk+CLI+dial and changed my extensions.conf like this: alsa.conf [general] autoanswer=no context=consolecontext extension=100 By the way, how do i know if my console is using the channel driver ALSA or OSS? then, in

Hi. I've got a working dial plan on my home system but there are problems with it and I was hoping someone more comfortable with dial plans might be able to help. In a nutshell here's what I'm currently doing on an incoming outside phone call [default] Set(TIMEOUT(digit)=3 Set(TIMEOUT(response)=60 exten => s,1,NoOp(Answering in default context) exten =>

Hi, When I dial from my sip device to the extension 1234 which is linked to the ALSA console driver the call fails with the message "No channel type registered for 'ALSA'" (see below). I would like to have the console autoanswer for paging. However when I call from the console to the sip device the call completes fine. I alias alsa device hw:1,0 to card1 in /etc/asound.conf

Hi, I have been using Cisco 7960's with Asterisk for years. I am trying get a 7961 working and have a problem. In my configuration, not all of my line appearances register to the same Asterisk SIP server. I have an Asterisk server at home and another at work. My Line 1 button registers to the home server and my Line 2 button registers to the work server. This has worked for years

Hi, I have a RH9 system with an onboard VIA sound chip. According to the archives, VIA won't work for asterisk. So, I disabled the VIA and I purchased a Creative Labs "Soundblaster PCI 128-Voice" soundcard ($13). This card is on the "approved" RedHat list. However, the documentation inside the package says "VIBRA 128". Anyhow, kudzu doesn't see then

Hi Sorry if this comes twice; i sendt first version from non-member address. I'm learning use Asterisk but cannot solve following problem: i have Asterisk v1.0.7 (DEbian) and Linphonec v1.2 (Debian). Every time i try to register within LAN i got 'Forbidden' message from Linphonec. Where to start searching for reason for this failure, is there more debuggin options

nautiluz wrote: >I am starting project to develop open implementation of some (naturally open >codec) for simply designed embedded devices which can be used by small to >big VoIP operators or hobbyists which wants to build their own small and >low cost VoIP phone. >Speex seems to be great choice. And greater will be if there should be >some guys who want to help ;-). Sorry

It would be cool to get one of these and see if it can be hacked and loaded with your favorite SIP or IAX softphone. Looking at the pic, it looks like the dongle is both a soundcard and memory stick. Heck, I would be glad to have it if I could get the soundcard to work. Might as well since it is free after rebate. http://www.circuitcity.com/ssm/Accessories-for-Vonage-V-Phone-VPHONE/sem

I am using a extension to dial the console which has autoanswer enabled. I am getting a strange warning, has anyone seen this before? Nothing on Google, or Voip-Info [Jan 9 13:50:05] WARNING[5009]: chan_oss.c:1048 oss_request: oss_request ty <console> data 0x0xb7851e00 <dsp> << Call to device 'dsp' dnid '(null)' rdnis '(null)' on console from