search for: autoansw

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' =- [Synopsis]: Log in for autoanswer [Description]: AutoanswerLogin(exten):Used to login to the autoanswer application for...

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

...on't hear 'anything' (no music, nothing at all) until the agent press (of course) again the button (but usually the caller hang up since he don't hear anything) there is a way to send the 'music on hold' to the caller even with the asterisk send the call to the phone (autoanswer on) but the 'hold' button is already pressed? I have to search/manage the asterisk config or the phone one? We are using asterisk 1.2.1 with Thomson ST2030. this is the asterisk log: (...) -- Executing Queue("CAPI/ISDN4/********-ce", "coda_azienda|t| 3600")...

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?

...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([${AVAILSTATUS} = "1"]?autoanswer:fail) I couldn't get this to work unless I surrounded the first part of the test with quotes, too, like this: exten => s,n,GoToIf(["${AVAILSTATUS}" = "1"]?autoanswer:fail) Leaving aside the completely separate madness of trying to determine just what values mean wh...

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:

...-- Executing [s at macro-page:1] ChanIsAvail("Local/7299 at page-9da0,2", "SIP/7299|js") in new stack [Aug 9 08:10:27] WARNING[29209]: ast_expr2.y:901 op_div: non-numeric argument -- Executing [s at macro-page:2] GotoIf("Local/7299 at page-9da0,2", "0?fail:autoanswer") in new stack -- Goto (macro-page,s,3) -- Executing [s at macro-page:3] Set("Local/7299 at page-9da0,2", "_ALERT_INFO="RA"") in new stack -- Executing [s at macro-page:4] SIPAddHeader("Local/7299 at page-9da0,2", "Call-Info: <sip:XX...

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: Asterisk Us...

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 handsets but this is the conclusion I reached after reading Symbian forum and documentation. > > > -S > > > _______...

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 don't have that option, rather than calling a Local/ channel to SetSIPheaders() and Dial(). I don...

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 =>

...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 the call spool to call the autoanswer extensions, simultaneously as it were, and all are entered into the same conference. The caller is the admin. You speak, they hear. It works fine. I changed the "kicked" gsm to a beep, as the conference is terminated by kicking everyone off, and it is kinda comical to end an announceme...

...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 => s,n,NoOp(${AVAILORIGCHAN}) exten => s,n,NoOp(${AVAILSTATUS}) exten => s,n,GoToIf($[${AVAILSTATUS} < 1]?autoanswer:fail) exten => s,n,NoOp() exten => s,n(autoanswer),Dial(${ARG1}||) exten => s,102(fail),Hangup [pruebas] exten => *99,1,Dial(Local/111 at inpuerta&Local/112 at inpuerta||r) [inpuerta] exten => _1XX,1,Macro(callonlyiffree,SIP/${EXTEN}) The Log: -- Executing [*99 at home...

...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 an analog phone to page system, but of course I would like to reuse what is there. Thanks for any advice or pointers, Jeb Campbell jebc@c4solutions.net

...2005 00:04 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] AASTRA 480i Firmware 1.2.0.162 SIP ALERT_INFOproblems The ALERT_INFO variable works for 480i firmwares 1.2.1.207 and up (like the 1.2.5-series). Set it like in the example below: exten=_*55XXX,1,SetVar(ALERT_INFO=info=alert-autoanswer) exten=_*55XXX,2,Dial(SIP/${EXTEN:3},12,Ttr) (*55 will be the prefix for the normal phone number; if a single digit is used --or anything of a different length-- adjust the slicing of the ${EXTEN}, like ${EXTEN:1} for a single digit) The 'info=alert-autoanswer' is the only value that se...

...ubscribeExpires> <timerSubscribeDelta>5</timerSubscribeDelta> <timerT1>500</timerT1> <timerT2>4000</timerT2> <maxRedirects>70</maxRedirects> <remotePartyID>true</remotePartyID> <userInfo>None</userInfo> </sipStack> <autoAnswerTimer>1</autoAnswerTimer> <autoAnswerAltBehavior>false</autoAnswerAltBehavior> <autoAnswerOverride>true</autoAnswerOverride> <transferOnhookEnabled>false</transferOnhookEnabled> <enableVad>false</enableVad> <preferredCodec>none</pre...

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