similar to: 'set debug' problems

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

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 people, question one i see that asterisk is now in 1.x release. having tried it in the past i want to know if i can use a voice modem as an outgoing line. i know in the past that was not possible/supported so im just asking in case the option is now available. question two im planing to use asterisk as a pure voip solution with sip phones and h323 phones no need for digium/dialogic hardware

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

Is there a way to ring selective line on multi-line phone. For example if I'm on the phone talking internally on line 1 and the calls comes-in the line 2 will automatically ring. The phone P104 allow extension to be assign each line. Is there a way to call certain line (example line 3) on multi-line phone instead of line 1 when the phone is not busy? For example the Sip phone P104 has

Hi, now I've got caller id working on my BT line in the UK, I'd like to play a different message to those pesky sort who with hold their outgoing number. How can I do this in my extensions.conf for my [incoming-analog] context? I realise some people may call who are unable to change the way that their system withholds the outbound number, so I'll give them chance to leave a voice

Sorry if this is the wrong list... I need a toll-free number to be delivered to me on IAX. (This is NOT an existing number need to buy the whole service.) Anyone know of a service provider offering this? -Mark 707-735-1038

If I have different opening hours on different days, can I accomodate that in a single gotoiftime, or will I need to filter them out one by one ? For example, our hours are Mon-Fri 9:00-17:00 and Sat 09:00-13:00 can this be done something like GotoIfTime([9:00-17:00|mon-fri][9:00-13:00|sat]|*|*?open,s,1) or something like that, or do I have to do:

I'm getting these "calls" at 16 and 46 minutes after every hour. The SIP phone rings, and if we pick up, we get a dial tone. If we don't pick up, we get the dial tone in a voicemail message. An analog phone connected to the incoming POTS line doesn't ring (whether or not * remains connected to the line). It's like the horror movie where the babysitter is getting

I've been banging my head against a brick wall for the last hour and I'm sure this is one of those easy to solve things - just that I can't see the wood for the trees. I'm trying to do: ----------- [some-context] Exten => s,1,Macro(dodial,'SIP/201,15,tT',123456,MOHClass) [macro-dodial] Exten => s,1,SetCallerID(${ARG2}) Exten => s,2,SetMusicOnHold(${ARG3}) Exten

Hi there, I've seen people using SER with Asterisk. I took a look at SER website, and I didn't see the point in using it, since Asterisk already handles SIP very well (apparently, at least). But, as I'm starting, and some of you (more experienced) use it, I know that there's something there... So I would like to know why to use SER. Is it because of scalability, performance,

Hi: Just saw your post while trying to solve a similar asterisk problem. Did not see any responses. Was your problem solved and what was the solution? Carey

> -----Original Message----- > From: Chris Wade [mailto:clwade@sparco.com] > Brian Roy wrote: > > I think that my PBX does this too. Is there any way I can get the > > Zaptel drivers to disconnect on that tone too? I would love > to replace > > my existing voicemail with * but I can't get my PBX to signal a > > disconnect properly. I have to use

I have set up an auto attendant and all is working but I am bothered by a long pause when switching between menus. This pause is between 5 and 7 seconds and is quite annoying. Is there anyway to address this. One other thing I find interesting is that when I move from the main menu to the sub menu the delay is there but when I move from the sub menu to the main menu the delay is not there.

I am currently writing a prototype agent monitoring system, which (as most others in question) simply monitors the output from the event system, and displays the relevant information. I would hope to donate this back to the community once it works properly :) However, I feel that it would be more useful for the manager output to be tab delimited, one record per line: i.e. instead of Event:

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

Hi all, Has anyone gotten custom ring tones to work using ALERT_INFO with the Cisco 7940 SIP phone? I've read the wiki, but just can't get this to work. I'm currently using the 7.2 SIP image. Thanks, Chris