similar to: Changes to manager outputs - A discussion

Displaying 20 results from an estimated 200 matches similar to: "Changes to manager outputs - A discussion"

2004 Jul 29
2
Aastra 480e phone ADSI config
There isn't much documentation on adsi, but I called NETXUSA (the vendor of my 480e) and they helped me along. My experience: 1. I really had no experience with ADSI so I had (probably still have) some misconceptions on how the configuration is loaded onto the phone. 2. I set the following in my /etc/asterisk/asterisk.adsi (most of this is the stock asterisk.adsi script): ;
2004 Jul 16
6
Asterisk + NEC Electra Elite IPK Integration
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
2004 Dec 06
5
two questions
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
2004 Dec 01
1
[OT] [slightly] app lever vs driver level implementation...
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
2004 Dec 07
1
Ringing multiline phone
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
2004 Dec 07
1
Inoming caller id withheld, move to new context, possible?
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
2004 Dec 13
2
Incoming Toll-Free
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
2005 Jan 14
1
gotoiftime - different hours
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:
2004 Oct 05
1
Phantom calls on FXO
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
2004 Nov 29
2
Variable substitution - How can I do Dial(${DIALSTRING}) where ${DIALSTRING} is 'SIP/201, 15, tT'?
I've been banging my head against a brick wall for the last hour and I'm sure this is one of those easy to solve things - just that I can't see the wood for the trees. I'm trying to do: ----------- [some-context] Exten => s,1,Macro(dodial,'SIP/201,15,tT',123456,MOHClass) [macro-dodial] Exten => s,1,SetCallerID(${ARG2}) Exten => s,2,SetMusicOnHold(${ARG3}) Exten
2004 Sep 20
6
SER + Asterisk
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,
2004 Aug 31
3
Cisco 79XX SIP Ring Tones
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
2004 Nov 11
6
cisco poe
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%
2005 Jan 03
9
Just saw your [Asterisk] xJack Segfault in Asterisk
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
2004 Oct 05
2
Long pause between menus
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.
2004 Oct 07
1
'set debug' problems
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
2004 Aug 23
2
Cisco 7940 Question
Hi all, I know this is a stupid question, but it is one I've been trying answer for quite some time. Exactly how many simultaneous calls can the Cisco 7940 have, considering you can be talking to one, and have XXX others on hold? Using SIP, is XXX only 1? I've found documents in various places indicating different values in regard to the max number of calls the phone can handle.
2004 Sep 23
3
app_valetparking / parking in general
Does anyone have Music-On-Hold and valet parking, or regular parking working together? No matter how I configure it, I cannot get moh to continue to play after I park a call using either valet parking or regular parking. The only thing I can think of is that I might need to use # transfer instead of sip native transfer? Shouldn't this just work? If needed I can post the config for one
2005 Aug 15
7
Switch between FXS ports
Hello, I have two FXS port on my TDM card. channel 4 is attached with a telco line that I use frequently. And channel 3 have another telco line. but I dont publish that number to my friends. If I receive a call through channel 4, how can I handover that call to channel 3 ..so that I can keep channel 4 open for incoming call. Thanks,
2016 Sep 29
2
[cfe-dev] improving test-suite`s FP subtests to be able to compare both exact-match outputs and more-optimized builds that may have different outputs due to FP optimizations
Dear all, I would like some help, please, with implementing Hal`s excellent suggestion, which I have reworded as below. Hal has confirmed a previous version of my rewording as a correct interpretation. [I made minor changes since then, e.g. for grammar.] [Abe wrote:] >> I think you [Hal] are suggesting something like this: >> 1) compile the program with FP fusion off,