similar to: agents and music on hold with autoanswer..

I'm trying to send an sms: smsq --motx-channel=CAPI/g1/0809101000 0607396666 "X" It seems to try to do something, but FT aren't happy: -- Attempting call on CAPI/g1/0809101000 for application SMS(0) (Retry 1) == ISDN4#02: Setting up DTMF detector (PLCI=0x104, flag=1) [Sep 13 15:45:50] WARNING[23584]: pbx.c:5142 ast_pbx_outgoing_app2: CAPI/ISDN4#02/0809101000-1 already

Hi! This weekend we had a problem with our Asterisk Box which ran flawlessly for nearly 4 weeks. The Asterisk server sits between the PSTN and a Siemens PBX and bridges 2 BRI lines. No calls, not incoming, not outgoing. The admin rebooted the Dell Box and then everything worked fine again. Now, I'm analyzing log files to find the cause. During the Asterisk outage the logfiles only show

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 it possible to specify a 'wrap-up' time in a queue so agents will have a specified amount of time to complete tasks between calls unless they hit a key on the phone? As it is they can recieve a call moments after they hang up with no 'down time'. Thanks Jim Friedeck

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

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:

Hi! I tried this in features.conf testfeature => *9,callee,Dial,CAPI/ISDN4/my_phone_number/b,60,T and it works... but... I would be able to transfer a call to any phone number, so I tried to use this line: testfeature => _*9.,callee,Dial,CAPI/ISDN4/${EXTEN:2}/b,60,T but... Asterisk crash! (it doesn't want even to reload configuration) Any idea about how to do so? Thanks a lot!

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 =>

I've a big problem with SIP forwarding back into 'ringing groups' creating what can only be described as call storms :-( I have a 'ringing groups' of SIP phones with an effective dialplan (much simplified) like so: ; Purchase ledger [ptsn_inbound] exten => _846061,1,Dial(Local/6061 at groups) .... [groups] exten =>

? Sat, 20 May 2023 10:59:18 +0000 akshay kulkarni <akshay_e4 at hotmail.com> ?????: > By "holding a lock", you mean a bug in the process right Well... one person's bug ("your threaded program breaks if I fork() the process") is another person's documented behaviour ("of course it does, it says 'please do not fork()' right in the manual").

Hi, I'm experiencing difficulties to get a B410P running with Asterisk 10.3.1 and DAHDI 2.6.1. Am I supposed to use DAHDI for this card and ISDN BRI for my country (Belgium)? thx, BC

Folks, please, take a look at this asterisk log message: [Oct 2 08:55:13] WARNING[10290] app_queue.c: Announcement file 'atcert' is unavailable, continuing anyway... [Oct 2 08:55:13] WARNING[10290] app_queue.c: Agent on Agent/1001002 hungup on the customer. but: -bash-3.1$ whoami asterisk -bash-3.1$ ls -ls $HOME/sounds/atcert* 12 -rw-r--r-- 1 asterisk asterisk 10956 Oct 2 07:00

Dear Ivan, REgrets to reply this late... By "holding a lock", you mean a bug in the process right (I am not a computer science guy, excuse my naivete)? THanking you, Yours sincerely, AKSHAY M KULKARNI ________________________________ From: Ivan Krylov <krylov.r00t at gmail.com> Sent: Thursday, May 18, 2023 1:08 PM To: akshay kulkarni <akshay_e4 at

https://bugzilla.netfilter.org/show_bug.cgi?id=847 Summary: Owner matching fails on listening socket Product: netfilter/iptables Version: unspecified Platform: x86_64 OS/Version: Debian GNU/Linux Status: NEW Severity: enhancement Priority: P5 Component: ip_tables (kernel) AssignedTo:

Am 13.06.2020 um 22:09 schrieb Antony Stone: Hi Antony > You are *assuming* that it's the codec causing the difference. Well, I really don't know what I can think, now... > We don't know that. > > Let me get this clear, to make sure I understand (differences emphasised): > > 1. You use *a VoIP softphone app* on your mobile, which is registered by SIP, > to

Hi, I''m fairly new to Rails and v. new to RSpec. Think it looks pretty useful and so I''m trying it out for the first time. I have installed the rspec-rails gem and created a simple plain text my_story file and my_story.rb file along the lines described here - http://www.tomtenthij.co.uk/2008/1/25/rspec-plain-text-story-runner-on-a-fresh-rails-app. When I run

Am 13.06.2020 um 22:56 schrieb Antony Stone: Hi Antony, > I would like to see a much simpler one-for-one comparison: only change one > thing at a time, and see what the difference is. > > So: I suggest you try *two* independent *pairs* of tests: OK > 1a. Using your Android phone, connect using your home wireless network (I > assume you have a wireless network, if not then

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