similar to: How do I match a "D"? (Was: RE: In-band disc onn ect problem (legacy PBX) - asterisk doesn't hear the touchtone?)

> -----Original Message----- > From: Paul Rodan [mailto:asterisk@glitch.cc] > I'll ask a stupid question, how does a user hit an alpha > letter from his touchtone? > > I know that the Cisco 7960's support entering alpha letters, > and it could > potentially do it (maybe), but how does the average end user > enter an a b c or d from their touchtone phone?

> -----Original Message----- > From: Gilad Ben-Yossef [mailto:gilad@codefidence.com] > I'm prbably stupid, but wont this do what you want? > > > exten => 1,1,Goto(bye,s,1) No, because I wanted to match on "D", not "1". Anyway, I figured it out. The extension was working, but Background() ignores the tones A through D by default. I didn't

> -----Original Message----- > From: Kevin P. Fleming [mailto:kpfleming@starnetworks.us] > David Brodbeck wrote: > > > Anyway, I figured it out. The extension was working, but > Background() > > ignores the tones A through D by default. I didn't realize > this because I > > wasn't waiting for message playback to finish. > > Please enter a

> -----Original Message----- > From: David Brodbeck [mailto:DavidB@mail.interclean.com] > Okay, the problem appears to be that I'm tone deaf. ;) > > I finally thought to turn on debugging on the channel. The > PBX is sending > "D", not "*". The programmer of the previous voice mail system (whose > configuration I was cribbing from) seems to have

The legacy PBX I'm working with does in-band disconnect notification -- it sends a * touchtone when the line is hung up. I've been trying to get this to work with Asterisk. I added a * extension to my menu context that plays "Goodbye" and hangs up. This works fine if I manually press *, but it never triggers when I hang up and the PBX sends it. I've plugged in an analog

> -----Original Message----- > From: Andrew Kohlsmith [mailto:akohlsmith-asterisk@benshaw.com] > Is the channel physically being hung up before the * tone is heard? Good question. If it is, Asterisk doesn't detect it -- the PBX doesn't support Kewlstart-style disconnect notification. The sequence I hear on the extension, when I plug in an analog phone, is the click of the

I recently posted a message about an interesting dilemma which I could not figure out. After doing a whole lot more digging I think I have figured out part of what is going on. The parking app is implemented as an app, ie can be called from extensions.conf) but if you look really closely, it is also implemented in each channels driver (handler, whatever you want to call it) and those

Has anyone else noticed this? I use 'set verbose 25' (insane, but I want to see *everything* right now) and would like to do the same for 'set debug', but as you see, set debug has a bad impact on my CLI output. asterisk*CLI> -- Executing Answer("SIP/824-b4ff", "") in new stack -- Executing Wait("SIP/824-b4ff", "0.5") in new

Hi everyone, I've just finished the 'SWait' app for *. SWait = Super Wait :) Syntax: SWait([timeout][cim]) 'timeout' is the number of seconds to wait. Defaults to 'ResponseTimeout'. 'c' is for continue. This changes the default behavior of the app from performing a 'Goto(t,1)' on timeout, to a 'Goto(pri+1)' on timeout, ie. continue to

Hi All, Using Cisco 7940 in SIP mode, is there a reason why a blind transfer to an extension that calls the Park app would be any different than a blind transfer to extension 700 (the default parking extension)? They both produce the same effect, a call being parked. But the * manager output is different for the two different methods. calling Park results in 'From' and

Hi all, With the recent 's' and 'n' priorities, as well as the advantage of labels, dialplan management has become *much* simpler IMHO. However, I have one suggestion for possible improvement. In any of the Goto[If|IfTime] statements, the ability to do 's' + a number or label + a number would be _nice_. Example extensions.conf: exten => 1,1,NoOp(Start) exten

I know this is on the wiki, I just want to confirm so I don't blow up my cisco phones. I've got several cisco 7940's all running using cisco power cubes. However, my boss wants me to switch just a few over to poe, but doesn't want to fork out the dough for a nice cisco poe switch, or anybody else's poe switch for that matter. So my question is, what is the '99.999%

Hi, I have created an agent node from a master node using below command. puppet node_aws create --image ami-cc5af9a5 --keyname icos-client --type ti.micro Now as i am trying to install puppet on it using below command puppet node install \ --install-script=puppet- enterprise \ --installer-payload=/usr/local/puppet/puppet-2.7.0.tar.gz \ --installer-answers=/usr/local/puppet/agent.txt \

When someone calls into our asterisk server over a PSTN line, dials an extension and then hangs up, the SIP phone related to the given extension will ring about 4 or 5 times before asterisk shows that the channel has been hung up in the console. This isn't such a big deal on its own, but what's happening now is that if a user calls in from a PSTN line, gets voicemail on the extension, and

Hi, since version 1.4.12 the libpri package supports ETSI Explicit Call Transfer feature: http://downloads.asterisk.org/pub/telephony/libpri/ChangeLog-1.4.12 Does anyone know, how to use this feature in the dialplan? I can not find any hints in the asterisk doc. Best regards, -Thorsten-

https://bugzilla.netfilter.org/show_bug.cgi?id=928 Summary: ECN: --ecn-tcp-ece and --ecn-ip-ect is not supported Product: nftables Version: unspecified Platform: x86_64 OS/Version: Debian GNU/Linux Status: NEW Severity: normal Priority: P5 Component: nft AssignedTo: pablo at netfilter.org

Hey Guys Im trying to find out how to transfer a call with ECT (explicit call transfer) ? Im currently transferring a call as following: exten=>2,1,Dial(capi/720****:078888****,18) exten => 2,2,Goto(2-${DIALSTATUS},1) exten => 2-NOANSWER,1,Dial(capi/720****:07979****) exten => 2-CHANUNAVAIL,1,Goto(1,1) exten => 2-BUSY,1,Dial(capi/720****:07979****) If I wanna transfer a call with

Hi, I'm am currently in the process of trying to integrate an * box with an NEC Electra Elite IPK. Currently, we have 7 POTS lines coming into our building. These lines are plugged into our NEC using the appropriate analog line interface card from NEC. The NEC effectively has NO configuration done to it, other than to make all the internal phones ring when a call comes in. We also

Hello, I've been real busy latley and have been unable to spend on time on vorbis [or IRC for that matter :)] I'll be getting some spare time soon so I will be back in action. Anyhow, I found this /excellent/ website of audio testing for soundcards, Pro DAC, I/O Video cards ect. The operator has this GREAT suite of tools. http://www.pcavtech.com Second, I was browsing over the

Hi, does anyone have the setup for go2call ? I have digium boards and quicknet linejacks and phonejacks. The cards work fine in asterisk without the g729 or g723.1 for the phonejack. I will like to do SIP origination using the codec in the phonejack and linejack g729 or g723 and send the calls to go2call. Anyone has the setup for this ? Or similar setup to a SIP provider using g729 or g723