search for: autoanswers

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' =-

Is there a linux sip or iax phone that will autoanswer and connect to the console or soundcard? I found linphonec but it does not autoanswer from what I can tell. Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050916/8bc76bbb/attachment.htm

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

Matteo Brancaleoni wrote: >Hi. > >I'm experiencing a strange issue with *. >I have a dev kit, aka a T100P + a zhone cb. > >Sometimes, on certains phones (on the fxo ports >of the cb) , when the phone rings, * detect >it as answered after the first ring, even >if no one is at the phone! > >The result is that on the other party (which >called the phone) hears

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?

My teeth are on edge after this one. A couple of perfectly good hours of my life, and I still don't know what's going on. . . . The extensions.conf.sample that comes with the current SVN trunk has this line, in an example that shows how to use ChanIsAvail: exten => s,n,GoToIf([${AVAILSTATUS} = "1"]?autoanswer:fail) I couldn't get this to work unless I surrounded the

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:

Ladies, Gentlemen We are experiencing an unusual problem in our asterisk 1.4.34.. We are attempting to determine if channels are in use before paging to them. This works correctly, as in it pages the phone.. however, we see the error message below on the console... after googling, we discovered limited information regarding the issue... -- Executing [NPANXX7298 at from-pstn:1]

Hello As far as ive understood, you can just write Exten => s,n,GotoIf([${AVAILSTATUS} = 1]?autoanswer:fail) ${AVAILSTATUS} would return 1, and "${AVAILSTATUS}" would return "1" Jon -----Oprindelig meddelelse----- Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne af Brian Capouch Sendt: 27. juni 2006 09:10 Til:

an external program, which at this stage, is not customizable ... I don't know if alternatives (LiMO, Android, ...) would be more open to this customization but for Symbian, not only Nokia SIP client wouldn't let you autoanswer to SIP calls, but any other SIP client complying to Symbian design wouldn't support autoanswer. PS: Please, note that I'm far from being an expert in GSM

Since 1.4 ALERT_INFO variable has been deprecated. I used to send this via AMI: Action: Originate Channel: Sip/1234 Application: AgentLogin Data: 1234 Variable: _ALERT_INFO=info=alert-autoanswer Callerid: AutoLogin[1234] In order to send an autologin and autoanswer call to the agent 1234 on an Aastra phone at extension 1234. (just for example). Now in * 1.4 with ALERT_INFO deprecated I

I'm afraid I already know the answer because I've done a lot of searching, but does anyone know of a softphone that supports a central phone book and paging (like the sip autoanswer option of some hardphones) Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200 david at safedatausa.com

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 =>

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

Hi list: I'm having the next problem, it appear that the application ChanIsAvail is not working on Asterisk 1.4.5 always return me 0 in AVAILSTATUS. I add my dialplan and the output to the cli. THanks. In the example i'm dialing from extension SIP/112 My DialPlan Secction: [macro-callonlyiffree] exten => s,1,ChanIsAvail(${ARG1}|s) exten => s,n,NoOp(${AVAILCHAN}) exten

I'm replacing a Merlin for a client and they have a PagePal Intercom that I would like to reuse. Here is what I know about it: It has a screw-down wires that goto rj-11 (This was told to me over the phone) that went into one of the Merlin ports. I tried bring it up with fxo_ks and fxo_ls (assuming it was analog and autoanswered) but no luck. I would be happy to replace if anyone knows of

But where do can you get this later firmware from? I'm still on 1.0.0.78 on my 480i. Regards Lee -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Peter Passchier Sent: 05 August 2005 00:04 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] AASTRA 480i Firmware 1.2.0.162 SIP ALERT_INFOproblems

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

I have been testing * with some Cisco 7912G's, in hope to trash our Nortel system. One feature our Nortel system has that I will need to fiqure out on the * system is paging. Is it possible to page a group of phones (all phones) with announcements? We are a k-12 school and we use our current phone system to make announcements on the phones monitor speaker. Any direction I can be pointed in