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| 00:41.35 | solitude88 | Do I need a PRI to run a asterisk system? |
| 00:42.52 | pabelanger | no, what makes you think that? |
| 00:45.13 | solitude88 | Im not sure pabelanger I was reading an article thats what I took away from it. |
| 00:45.36 | solitude88 | So essentially all I need is internet connection and a few DID's |
| 00:45.38 | solitude88 | ? |
| 00:45.53 | pabelanger | solitude88: well, you need to tell us what you want to do |
| 00:45.59 | pabelanger | ~book |
| 00:45.59 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
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| 00:46.10 | pabelanger | solitude88: ^ is a good place to start |
| 00:46.27 | solitude88 | huge understaking |
| 00:46.29 | solitude88 | lol |
| 00:46.44 | [TK]D-Fender | Not really |
| 00:47.11 | solitude88 | Time I have, is the learning curve hight? |
| 00:47.14 | solitude88 | high* |
| 00:47.27 | [TK]D-Fender | solitude88: Depends how far you need to go and how fast |
| 00:49.46 | solitude88 | [TK]D-Fender I'm not sure aobut how far but I need pretty fast |
| 00:50.16 | solitude88 | I need to setup a switchvox but not sure which way to go about it |
| 00:50.18 | [TK]D-Fender | solitude88: Describe the end scenario you need to accomplish then. |
| 00:50.31 | [TK]D-Fender | solitude88: First this isn't the place to talk about Switchvox at all. |
| 00:50.46 | solitude88 | switchvox is no good? |
| 00:50.59 | [TK]D-Fender | solitude88: It's a closed commercial package that has its own support resources |
| 00:51.31 | [TK]D-Fender | solitude88: Have you already purchased your Switchvox system? |
| 00:51.37 | solitude88 | not yet |
| 00:52.03 | solitude88 | is asterisk and switchvox the same thing? |
| 00:52.07 | [TK]D-Fender | solitude88: While it may be a suitable solution for you, there may be others as well. Describe the setup you are planning to implement |
| 00:52.58 | [TK]D-Fender | solitude88: No, Asterisk is the engine (and/or pieces thereof). Switchvox is a complete software solution that is the OS, with Asterisk, and proprietary GUI for configuring *, etc all bundlesd into one. Also typically sold with their hardware. |
| 00:53.09 | solitude88 | I would like to be able to allow a system to take calls from about 20 different phone numbers. There's always two agents on at all time, at most 5 consecutive calls going |
| 00:53.38 | [TK]D-Fender | solitude88: Ok, what kind of call processing? |
| 00:54.07 | solitude88 | Paying for a nice system Im ok with but what I don't want to pay alot for is the internet service, that's why I mentioned the PRI not knowing |
| 00:54.16 | solitude88 | just regular phone calls |
| 00:55.07 | [TK]D-Fender | solitude88: 5 calls at ULAW quality = 85kbps per direction or 425kbps. What is your current internet connection? |
| 00:55.27 | [TK]D-Fender | solitude88: Because there is no way a PRI could possibly be cost-effective vs 5 channels worth of ITSP connectivity. |
| 00:56.08 | solitude88 | [TK]D-Fender I believe its 5Mbps |
| 00:56.27 | [TK]D-Fender | solitude88: Compare that to the < 0.5 Mbps your calls will require |
| 00:56.40 | [TK]D-Fender | solitude88: 10:1 |
| 00:56.56 | solitude88 | so the internet is ok |
| 00:56.56 | *** join/#asterisk jetlag (~jetlag@pool-71-168-205-104.cmdnnj.east.verizon.net) |
| 00:56.57 | [TK]D-Fender | solitude88: So this should not be any kind of worry |
| 00:57.19 | [TK]D-Fender | solitude88: So that settles the "usability" of PSTN connectivity options at least. |
| 00:57.19 | solitude88 | expet internet going down from time to time? |
| 00:57.29 | [TK]D-Fender | solitude88: Back to the PBX... what kind of call processing? |
| 00:57.38 | solitude88 | call processing? |
| 00:57.41 | [TK]D-Fender | solitude88: Well if your internet is flakey, then that is something else... |
| 00:57.44 | solitude88 | like fax? |
| 00:58.09 | [TK]D-Fender | solitude88: How do you call out? Are you an inbound call processing center? What kind of handling for your calls? |
| 00:58.33 | solitude88 | oh inbound and outbound but all calls start with inbound calls |
| 00:58.46 | solitude88 | and one fax line |
| 00:59.05 | [TK]D-Fender | Ok, so basically all user-dialed calls by hand for outbound. |
| 00:59.13 | solitude88 | yes |
| 00:59.23 | [TK]D-Fender | solitude88: Anything else special? |
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| 00:59.49 | solitude88 | different phone numbers |
| 00:59.53 | solitude88 | 20 |
| 00:59.59 | [TK]D-Fender | solitude88: I get that |
| 01:00.21 | [TK]D-Fender | solitude88: Anything else unusual? |
| 01:00.25 | solitude88 | nothing that I can think of except maybe wanting to allow agents to answer phone in other cities |
| 01:00.39 | solitude88 | without the quality being bad |
| 01:00.50 | [TK]D-Fender | solitude88: Well so far yours is as simple a setup as anyone's, perhaps even moreso. |
| 01:01.14 | solitude88 | how much would it cost to implement something like this hardware wise? |
| 01:01.28 | [TK]D-Fender | solitude88: your setup implies nothing special for which Switchvox addressed in a way that is unique to any other solution, or in any realistic way "superior" |
| 01:01.43 | [TK]D-Fender | solitude88: My watch could do this.... and it's ANALOG |
| 01:01.54 | solitude88 | whats my watch? |
| 01:02.15 | [TK]D-Fender | solitude88: Don't know .... never seen your watch. Mine is a Timex. |
| 01:02.22 | solitude88 | lol |
| 01:02.36 | solitude88 | I dont follow, are you saying not to use switchvox for htis? |
| 01:02.38 | solitude88 | to much |
| 01:02.53 | solitude88 | I could use someting thats cheaper |
| 01:03.17 | [TK]D-Fender | solitude88: There is nothing that makes Switchvox a better choice than anything else. It also has a price-tag and other options don't |
| 01:04.02 | solitude88 | [TK]D-Fender could you possibly point me a direction you would take? |
| 01:04.19 | [TK]D-Fender | solitude88: Your description was something I'd jsut as easily do in Asterisk from scratch, or maybe install FreePBX's ISO which includes their OSS GUI and everything in a "ready-to-go" disc |
| 01:04.58 | solitude88 | Do you do setups? |
| 01:05.00 | [TK]D-Fender | solitude88: http://schmoozecom.com/distro-download.php <- I'd recomend the latest ISO for your PC archetecture |
| 01:05.28 | solitude88 | These are all ubuntu? |
| 01:05.29 | [TK]D-Fender | solitude88: I do, but for your own experience it would be good for you to see what's out there for you to get your hands on right away. |
| 01:05.39 | [TK]D-Fender | solitude88: Any cheap junk P4 system is enough for this. |
| 01:05.48 | [TK]D-Fender | solitude88: No, all CentOS-based |
| 01:05.54 | solitude88 | could I put this on the cloud? |
| 01:05.57 | solitude88 | AWS? |
| 01:06.06 | [TK]D-Fender | solitude88: and as I said, a FULL distribution. You dump it on a comuter and it sets everything up |
| 01:06.13 | [TK]D-Fender | Probably |
| 01:06.28 | [TK]D-Fender | You could also do it piecemeal by hand too... |
| 01:06.37 | [TK]D-Fender | That ISO is just a 1-stop-shop |
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| 01:07.19 | solitude88 | This looks pretty cool I've never seen it |
| 01:07.30 | solitude88 | This is free? |
| 01:07.35 | DarwinSurvivor | I'm looking to "upgrade" our current POTS-only system with asterisk. I want to add SIP functionality without breaking the ability for the older phones to make simple calls |
| 01:08.05 | [TK]D-Fender | solitude88: Yes |
| 01:08.09 | pabelanger | DarwinSurvivor: how are you planning on connecting the 'older phones' |
| 01:08.16 | DarwinSurvivor | I'm looking at an SPA3000 ($30) or SPA3102 ($55) and was wondering if they would work well for this setup |
| 01:08.26 | solitude88 | I dont know yet, thats what I thought the pri was for |
| 01:08.36 | solitude88 | transferring the analog phones to the pri |
| 01:08.46 | [TK]D-Fender | DarwinSurvivor: Ironic nick. You are trying to extend the life of your system which apparently deserves to be extinct :) |
| 01:09.00 | WIMPy | analog is evil |
| 01:09.11 | solitude88 | lol |
| 01:09.21 | DarwinSurvivor | [TK]D-Fender: sort of, my father isn't very technical so he'll need a "transition", but my mother runs an online store so it would be nice to have a mailbox system and stuff |
| 01:09.51 | [TK]D-Fender | DarwinSurvivor: Using * as a mailbox system is a real PITA |
| 01:10.06 | [TK]D-Fender | DarwinSurvivor: the trick is dealing with the hand-off, etc |
| 01:10.12 | [TK]D-Fender | DarwinSurvivor: What do they have? |
| 01:10.13 | DarwinSurvivor | All the phones (except an old emergency rotary) are on 1 base station, so should work on the FXS port |
| 01:10.36 | solitude88 | whats the best way to prevent the system becoming unreachable because of internet? |
| 01:10.42 | [TK]D-Fender | DarwinSurvivor: Doesn't sound like they HAVE aa "system", just a single phone with multiple handsets.... |
| 01:10.46 | solitude88 | I think all ISP's go down here and there |
| 01:10.50 | [TK]D-Fender | solitude88: Get better internet |
| 01:10.55 | DarwinSurvivor | right now we have 1 base station with 2 handsets, I don't care if the handsets get treated as 1 extension |
| 01:11.02 | DarwinSurvivor | pretty much |
| 01:11.05 | WIMPy | DarwinSurvivor: Get a sip base. |
| 01:11.06 | solitude88 | but even good internet goes down once in a while rihgt? |
| 01:11.19 | [TK]D-Fender | solitude88: Also .... upgrade your "think" to "has obtained realistic market research data for his area" |
| 01:11.25 | DarwinSurvivor | which is what the old "emergency" rotary is for :) |
| 01:11.51 | WIMPy | solitude88: More often than phone lines, but it's all a question of what you'r ewilling to pay for, usually. |
| 01:11.56 | [TK]D-Fender | DarwinSurvivor: well it's still jsut 1 party-line between them... what is this "SIP functionality" you want to add? |
| 01:12.05 | DarwinSurvivor | Eventually I'd like to get some sip phones, but for now I'm just going to put some virtual ones on the family's cell phones for use over LAN |
| 01:12.07 | solitude88 | WIMPy service wise? |
| 01:12.14 | solitude88 | as in internet not pbx? |
| 01:12.36 | WIMPy | Err, what? |
| 01:12.41 | DarwinSurvivor | I'd like to add a quick incoming menu if nobody picks up so that people can leave voicemails for family/store/volunteer stuff |
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| 01:13.04 | [TK]D-Fender | solitude88: PRI's sometimes go down.... evrything dies at some point. |
| 01:13.14 | DarwinSurvivor | having the messages on a server (I run FreeBSD on mine) so that they can be accessed from the computers would also be useful |
| 01:13.20 | [TK]D-Fender | solitude88: as I said, go get REAL stats on your providers failure rates |
| 01:13.27 | [TK]D-Fender | solitude88: And consider backup options. |
| 01:14.16 | [TK]D-Fender | DarwinSurvivor: So far basically all you're really doing is having Asterisk as a VM box, and maybe adding SIP phones and still only sharing a single analog line. That about right? |
| 01:14.20 | DarwinSurvivor | I'm basically looking for 2 features right now. 1) Multiple voicemails instead of 1 answering machine 2) Ability for us to forward to our cell phones over SIP when we're home |
| 01:14.31 | DarwinSurvivor | yes, only 1 incoming phone number |
| 01:15.02 | [TK]D-Fender | DarwinSurvivoryes, only 1 incoming phone number <- wrong approach to the answer |
| 01:15.02 | DarwinSurvivor | It's an online from-our-house business, we don't even have employees yet |
| 01:15.20 | [TK]D-Fender | DarwinSurvivor: When I say "1 analog line" even a boring analog line can have more than 1 "number" via distinctive ringing, etc |
| 01:15.43 | [TK]D-Fender | DarwinSurvivor: So are you still looking to use that line? |
| 01:15.55 | [TK]D-Fender | darAnd keep in mind so far that means it'll support 1 call period. |
| 01:16.01 | DarwinSurvivor | right now we only have 1 number on the line, we may look into more lines in the future but we don't have enough volume to worry about that yet |
| 01:16.04 | *** join/#asterisk saint_ (~saint@c-68-38-56-184.hsd1.nj.comcast.net) |
| 01:16.24 | DarwinSurvivor | of course, multiple calls == multiple numbers/lines |
| 01:16.55 | [TK]D-Fender | DarwinSurvivor: Then this : http://www.telephonydepot.com/Catalog/Cisco-Analog-Adapters/Linksys-SPA3102 |
| 01:16.57 | saint_ | Knowing that Set(CALLER_SETINFO(number,${CALLERID(num)},eta) expects a date under the format YYYY-MM-DD HH:MM:SS |
| 01:17.04 | DarwinSurvivor | basically we want multiple VM's and ability to use our cell phones as home phones |
| 01:17.07 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 01:17.13 | saint_ | Is it OK to do that : Set(CALLER_SETINFO(number,${CALLERID(num)},eta)=${STRFTIME($[${STRFTIME(,,%s)} + $[${CALLER_INFO(defaulteta,number,${CALLERID(num)})} * 60]]),,%Y-%m-%d %X}) |
| 01:17.24 | [TK]D-Fender | DarwinSurvivor: Unsing he Cellphones how? |
| 01:17.32 | [TK]D-Fender | DarwinSurvivor: Using he cell-phones how? |
| 01:17.34 | DarwinSurvivor | cello phones as home phones == cell phone -> asterisk over sip -> POTS |
| 01:17.37 | saint_ | never mind .. |
| 01:17.52 | DarwinSurvivor | cell phone with SIP softphone software installed (I have that part working already) |
| 01:17.59 | [TK]D-Fender | DarwinSurvivor: So your cells will have SIP clients on them? |
| 01:18.08 | [TK]D-Fender | DarwinSurvivor: Sure, go for it./ |
| 01:18.11 | DarwinSurvivor | yes, mine already does and I'll be working on the others soon |
| 01:18.40 | DarwinSurvivor | I just wanted to check if that setup would work with the hardware I've found and that the hardware is "decent" (ie: no known sound issues, etc) |
| 01:18.56 | [TK]D-Fender | DarwinSurvivor: So I'v just linked you what you'll need to account for your analog line and phones. |
| 01:19.44 | DarwinSurvivor | linked? |
| 01:20.36 | [TK]D-Fender | [20:16][TK]D-FenderDarwinSurvivor: Then this : http://www.telephonydepot.com/Catalog/Cisco-Analog-Adapters/Linksys-SPA3102 |
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| 01:21.07 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 01:21.21 | DarwinSurvivor | that's one of the one's I found (ebay) for $55 |
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| 01:22.01 | DarwinSurvivor | I also found an SPA3000 for $30, is there any advantage to the 3102? I noticed it has a second ethernet port (LAN + WAN), but can't see why i'd need both |
| 01:22.22 | leifmadsen | you could use it as a router |
| 01:22.26 | leifmadsen | if you don't need that, then not necessary |
| 01:22.54 | DarwinSurvivor | so the SPA3000 would be just fine for my use (if there's anything I'm short on, it's NOT routers :P)? |
| 01:22.55 | [TK]D-Fender | DarwinSurvivor: the SPA-3000 is the older revision. First e-bay = no warranty, and on an older model. |
| 01:23.09 | [TK]D-Fender | DarwinSurvivor: Is it worth getting burned? |
| 01:23.32 | DarwinSurvivor | for 1/3 the price I could get 2 spares for the cost of a new one |
| 01:23.45 | saint_ | Can I add 2 dates throught STRFTIME to get another date , if they are under the same format ? |
| 01:24.09 | DarwinSurvivor | telephonydepot appears to be out of stock anyways |
| 01:24.25 | saint_ | or do I need to switch to unix time, then add them , then print the result in the correct format ? |
| 01:24.39 | WIMPy | saint_: You can convert them to epoch and add something to that. |
| 01:24.50 | DarwinSurvivor | thank you [TK]D-Fender you've been a great help :) |
| 01:24.55 | solitude88 | sorry, daughter |
| 01:24.56 | solitude88 | :) |
| 01:25.19 | saint_ | WIMPy: what's epoch ? I saw that in the doc. this is the time i am passing to the functino ? |
| 01:26.12 | WIMPy | epoch= unix time |
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| 01:27.36 | saint_ | WIMPy: i m going nuts with this : Set(CALLER_SETINFO(number,${CALLERID(num)},eta) = ${STRFTIME( $[${STRFTIME(,,%s)} + $[${CALLER_INFO(defaulteta,number,${CALLERID(num)})} * 60]]),,%Y-%m-%d %X}) |
| 01:28.17 | saint_ | so I'm supposed to return a date under the format YYYY-MM-DD HH:MM:SS |
| 01:28.25 | saint_ | on the right of the = sign |
| 01:28.39 | [TK]D-Fender | DarwinSurvivor: Well if you're willing to waste a variable amount of time and failure of service for this.... go for it. |
| 01:29.04 | saint_ | ${CALLER_INFO(defaulteta,number,${CALLERID(num)})} returns a time in minutes, from 1 to 10 |
| 01:29.39 | WIMPy | saint_: That's a long one, but it's at least the right direction. What kind of function is that, BTW? |
| 01:29.42 | saint_ | so that part, Ihave to translate it in unix time first, then add ${CALLER_INFO(defaulteta,number,${CALLERID(num)})} * 60 , then translate the whole thing into yyyy-mm-dd hh:mm:ss then, right ? |
| 01:30.00 | [TK]D-Fender | saint_WIMPy: i m going nuts with this : Set(CALLER_SETINFO(number,${CALLERID(num)},eta) = ${STRFTIME( <- BAD |
| 01:30.03 | WIMPy | yes |
| 01:30.06 | *** join/#asterisk Wiretap (~wiretap@unaffiliated/wiretap) |
| 01:30.11 | [TK]D-Fender | saint_: Never EVER put spaces around the equals sign. |
| 01:30.39 | saint_ | [TK]D-Fender: oh that is only for re-lecture so i can try to find out where I have it wrong. once i think i have it right, i remove all spaces then try it |
| 01:30.41 | solitude88 | http://www.expressinterconnect.com/246/asterisk-ip-pbx-installation-on-amazon-ec2-web-service-part-1-of-4/ |
| 01:30.51 | igcewieling | at least in a Set. In conditions (various IF like stuff) it is OK |
| 01:31.07 | solitude88 | basically this article is saying you can setup a complete pbx solution for free |
| 01:31.09 | solitude88 | wow |
| 01:31.40 | igcewieling | saint_: you may very well be setting the value with a space at the beginning. |
| 01:31.50 | DarwinSurvivor | [TK]D-Fender: Well right now we're just in the testing phase, we don't want to spend a bunch of money and then find out we need a different setup that the hardware won't work on |
| 01:32.06 | saint_ | igcewieling: that s only for me to read correctly since i broke it. |
| 01:32.11 | DarwinSurvivor | [TK]D-Fender: Once we get it working nicely, we'll look into more professional hardware |
| 01:32.13 | saint_ | igcewieling: before i try, i remove all spaces |
| 01:32.26 | DarwinSurvivor | [TK]D-Fender: But I'll definitely keep your advice in mind |
| 01:33.50 | [TK]D-Fender | DarwinSurvivor[TK]D-Fender: Well right now we're just in the testing phase, we don't want to spend a bunch of money and then find out we need a different setup that the hardware won't work on <- like what? |
| 01:34.35 | saint_ | WIMPy: so can i use strftime and give in epoch a date like "yyyy-mm-dd" and have it output as unixtime ? |
| 01:35.23 | saint_ | WIMPy: the function I had before that was working was ${STRFTIME($[${STRFTIME(,,%s)} + $[${ETA} * 60]],,%Y-%m-%d %X)} |
| 01:35.31 | WIMPy | That's two times STRFTIME, but I guess you already got that. |
| 01:35.43 | DarwinSurvivor | [TK]D-Fender: don't know, I'm just starting in this stuff. |
| 01:35.49 | [TK]D-Fender | saint_: Your nested epxression are also a mess and unnecessary |
| 01:36.23 | [TK]D-Fender | DarwinSurvivor: Well you seem to be looking for the emergency exit right at the start while making decisions that are risking failure right out of the gate. |
| 01:36.30 | saint_ | [TK]D-Fender: well remember i'm barely 1 month into asterisk .. i'm right in the learning curve.. |
| 01:37.28 | [TK]D-Fender | saint_: slow down for the corners |
| 01:37.46 | DarwinSurvivor | [TK]D-Fender: One of the things we need to test is whether everyone in house will even be able to understand how to use the system. It took a while just to do the Windows->Linux conversion |
| 01:37.54 | saint_ | WIMPy: the goal is I have to return a date under yyyy-mm-dd hh:mm:ss from (current_date + ETA) , current date being from strftime() , and ETA being a number of minutes from 1 to 10 |
| 01:38.14 | [TK]D-Fender | DarwinSurvivor: If they can't handle voicemail then you have mor serious problems. |
| 01:38.32 | [TK]D-Fender | darBecause so far the only variable that changes is that. |
| 01:38.47 | saint_ | WIMPy: so what i was trying to do is convert the current time into unix time, then add ETA * 60sec , then convert back to yyyy-mm-dd hh:mm:ss |
| 01:39.04 | [TK]D-Fender | DarwinSurvivor: What they dial does not need to change. What they call out on does not need to change. the # people call in on doesn't change. If the power goes out those phones will all righ anyway. |
| 01:39.07 | saint_ | WIMPy: unles if I can directly add minutes to a date which is not in unix time ? |
| 01:39.25 | [TK]D-Fender | DarwinSurvivor: So far NOTHING has changed except using * as an ANSWERING MACHINE |
| 01:39.27 | WIMPy | saint_: That's the way to do it. But it might be easier not to try to do it in one line. |
| 01:39.50 | DarwinSurvivor | [TK]D-Fender: and tying in the various cell phones as extra house phones |
| 01:40.09 | saint_ | WIMPy: okay. I'll try for another 30 minutes and if i can t have it to work, i'll break it down. if it works, i'll be glade to share it lol |
| 01:40.20 | DarwinSurvivor | eventually we may look into call forwarding and stuff as well, but that's mid-long term |
| 01:40.39 | [TK]D-Fender | DarwinSurvivor: Well you already have a cell set up. If they can't answer a call you make to it or place a call with it... then that's beyond sad. And it's something yuo can test right NOW. You have the cell ready. |
| 01:45.20 | DarwinSurvivor | ok, I guess I have some more research to do first then before picking out the hardware |
| 01:45.46 | DarwinSurvivor | are there any other places you'd recommend for buying the POTS->SIP adapters (either external or as cards)? |
| 01:50.18 | [TK]D-Fender | DarwinSurvivor: tehre are dozens of vendors for this stuff |
| 01:50.23 | [TK]D-Fender | Linksys = commodity gear |
| 01:50.47 | [TK]D-Fender | More like hundreds. |
| 01:50.50 | [TK]D-Fender | perhaps thousands |
| 01:50.57 | DarwinSurvivor | commodity in features, quality or both? |
| 01:51.19 | DarwinSurvivor | I don't think I need a lot of features (that's what * is for right?) |
| 01:52.17 | [TK]D-Fender | DarwinSurvivor: it's an ATA... almost all will do the same things. Then there is the FXO aspect which it ALSO does |
| 01:52.36 | [TK]D-Fender | As an FXO there is much less to be said for it |
| 01:52.54 | *** join/#asterisk DarthExpeditor (~IceChat9@107.41.225.224) |
| 01:54.20 | DarwinSurvivor | I've found that pretty much anything with an FXO port tends to have an FXS port as well |
| 01:58.53 | saint_ | WIMPy: got it to work: |
| 01:58.55 | saint_ | Set(CALLER_SETINFO(number,${CALLERID(num)},eta)=${STRFTIME($[${STRFTIME(,,%s)}+$[${CALLER_INFO(defaulteta,number,${CALLERID(num)})}*60]],,%Y-%m-%d %X)}) |
| 02:02.25 | *** join/#asterisk bkolden (~bkolden@68.83.217.247) |
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| 02:15.12 | solitude88 | Can anyone recommend a good SIP provider? |
| 02:16.04 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 02:19.26 | [TK]D-Fender | solitude88: voip.ms = if you don't intend of fax over the internet. Vitelity if you do. les.net also good |
| 02:20.32 | solitude88 | [TK]D-Fender will check them out thank you. Have you heard anything about anveo.com or nexvortex.com? |
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| 02:36.32 | [TK]D-Fender | solitude88: Almost no mention of them here. Not a great start |
| 02:36.47 | [TK]D-Fender | solitude88: They might be OK ... but little word-of-mouth for them |
| 03:03.25 | igcewieling | I like Vitelity. |
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| 03:22.43 | ChannelZ | I use em at home |
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| 03:44.46 | igcewieling | I've used them for at least 10 years, but very low volume most of the time. |
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| 04:45.16 | saint_ | solitude88: google voice |
| 04:45.23 | solitude88 | ? |
| 04:46.13 | saint_ | talking about google voice, is gtalk encrypted ? |
| 04:48.40 | solitude88 | oh sorry saint_ I don't know that answer to that |
| 04:48.50 | solitude88 | I would think so |
| 04:49.04 | solitude88 | skype is and other voice services |
| 04:49.14 | solitude88 | I would think gvoice would follow suit |
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| 07:48.10 | fling | Hello! Is g722 the best quality codec available for iax2? |
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| 07:51.35 | Skavin | Anyone up for helping with a directrtpsetup issue? |
| 07:53.30 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:54.02 | Skavin | should directrtpsetup work create an outbound invite with direct media if SIP call has a dial(Local/ to another extension then a dial(SIP/ |
| 07:55.28 | Skavin | doing the Dial(SIP/ from directly from the extension works but I need to create a sim ring with different A formats |
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| 08:48.40 | PbxMan | morning |
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| 09:05.43 | skirmisha | guys |
| 09:05.56 | skirmisha | why ast does not delete entires from its db |
| 09:06.07 | skirmisha | it shows deleting but nothing is removed |
| 09:06.11 | skirmisha | how to debug that? |
| 09:06.54 | bulkorok | where are the spanish digits files in sounds!? I downloaded from http://downloads.asterisk.org/pub/telephony/sounds/asterisk-core-sounds-es-wav-current.tar.gz |
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| 09:30.08 | ChannelZ | the digits/ dir is not what you are looking for? |
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| 10:01.39 | bulkorok | found it... |
| 10:01.43 | bulkorok | but strange thing: |
| 10:02.28 | bulkorok | (asterisk 1.4.23.2) if setting AGI Rx << SET VARIABLE CHANNEL(language) "es" |
| 10:02.42 | bulkorok | digits are played from en and not from es |
| 10:02.56 | bulkorok | but cli shows: Playing 'digits/4' (language 'es') |
| 10:03.41 | bulkorok | Playing 'vm-theperson' (language 'es') is correct spanish... |
| 10:10.11 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 10:10.41 | angryuser | Hello, can someone point me to an explanation of this Warning ?[Feb 4 09:52:32] WARNING[20185]: func_lock.c:324 unlock_read: No datastore for dialplan locks. Nothing was ever locked! |
| 10:10.46 | angryuser | Should i worry ? |
| 10:12.12 | bulkorok | solved: every language must be in ../sounds/ referenced |
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| 10:33.27 | wdoekes | angryuser: UNLOCK() was attempted on a channel that hadn't done any locking |
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| 10:35.36 | wdoekes | that probably means that you're not handling your locks correctly. and that probably means that you should worry, because you're using locks for a good reason (I hope) |
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| 11:49.31 | mathi | hi |
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| 12:13.50 | Greenlight | How well does FastAGI scale? Will it behave okay with say 20 - 50 calls per second firing through it? |
| 12:15.50 | Greenlight | I'm planning on using FastAGI to determine which Agent/ to pass the call to |
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| 13:31.13 | leifmadsen | Greenlight: the answer is "possibly" |
| 13:31.29 | leifmadsen | you should use sipp in a test environment to load test your agi and see how it performs in your environment |
| 13:33.00 | Greenlight | Okay cool - the alternative is to dump the calls into a Wait(10) loop, and deal with them using the AMI, but that seemed a lot more "hacky". As long as there's no known issues with AGI under that kinda load, I'll give it a test and see |
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| 13:41.06 | Tim_Toady | can jiter buffer be enabled only for specific peers in sip.conf? |
| 13:41.14 | Tim_Toady | or its ageneral setting |
| 13:41.21 | Tim_Toady | a general* |
| 13:45.15 | leifmadsen | general |
| 13:45.50 | Tim_Toady | thx leifmadsen |
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| 13:50.44 | mathi | hi |
| 13:50.51 | mathi | [TK]D-Fender, are you here? |
| 13:51.39 | [TK]D-Fender | looks around |
| 13:51.43 | [TK]D-Fender | ummm .... no? |
| 13:53.58 | Greenlight | What does Asterisk do if it can't connect to a FastAGI server? Does it just continue in the dialplan? |
| 13:54.15 | [TK]D-Fender | should |
| 13:55.34 | *** join/#asterisk serafie (~erin@nat/digium/x-xwgbwpajpzrlwhom) |
| 13:56.03 | mathi | [TK]D-Fender, do you remember about my bad SIP calls quality ? I found the solution, but I don't understand it. Remember one could call me through FXO or through SIP (another number). And through SIP the audio prompts (wav files) waere terrible. I just rebooted the server and now it is excellent |
| 13:56.23 | mathi | do you have any idea why through FXO it was OK and through SIP it was terrible quality until I reboot ? |
| 13:56.54 | Greenlight | Sounds timing related |
| 13:57.19 | mathi | timing ? |
| 13:57.50 | Greenlight | Yea, it needs accurate timing when playing audio files |
| 13:58.49 | [TK]D-Fender | Greenlight, And TDM doesn't? |
| 13:59.07 | leifmadsen | timing is not part of audio playback |
| 13:59.21 | leifmadsen | that is only used in scenarios such as iax2 trunking and conferencing |
| 13:59.37 | Greenlight | Oh my bad - I thought it used timing when playing back audio as well |
| 14:00.02 | leifmadsen | incorrect |
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| 14:01.47 | Greenlight | So, "internal_timing=yes" has no effect at all on audio playback? |
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| 15:09.40 | jeffspeff | i have the setting in asterisk.conf set to deny calls if the system gets below 2gb ram (i have a total of 16gb) and it happened this morning or over the weekend. is there a way to look back in the log files and see what was using all of the memory? i had the admin on duty restart the server as a quick fix since i wasn't in the office. |
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| 15:47.00 | curfont | Why does asterisk "miss" some logged calls? |
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| 15:47.50 | curfont | I have a bunch of calls in log (cdr/Master.csv) that are 400+ seconds long but I can't find a sound log for them (I record via exten => _X.,n,Monitor(wav,,b)) |
| 15:48.02 | curfont | Does Monitor not always work? |
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| 15:49.46 | KermitTheFragger | hi all |
| 15:50.01 | KermitTheFragger | quick question; are there any Asterisk 11 CentOS RPMS available yet? |
| 15:50.10 | KermitTheFragger | (in any repo that is) |
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| 15:57.02 | igcewieling | curfont: look at the problem calls in asterisk logs |
| 16:13.34 | *** join/#asterisk aberrios (~aberrios@mail.solutiontelecom.co.uk) |
| 16:17.06 | aberrios | any polycom guru's about? struggling making the polycom accept ilbc codec from asterisk. I've set voice.codecPref.iLBC.13_33kbps="5" voice.codecPref.iLBC.15_2kbps="4" in sip.cfg for the phones and made sure Asterisk is set to only allow iblc in the device's sip.conf section but still get a Warning: 488 SDP "NotAcceptableHere" when sending a call to phone. Definately a phone config issue than anything to do with * but I thought i'd find m |
| 16:17.06 | aberrios | aybe some Polycom peeps hanging in IRC in #asterisk.... |
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| 16:36.06 | [TK]D-Fender | aberrios, pastebin your complete call attempt with SIP debug enabled |
| 16:36.07 | [TK]D-Fender | ~pb |
| 16:36.07 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 16:36.08 | [TK]D-Fender | ^^^ |
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| 16:47.12 | *** join/#asterisk Invader (~Invader@unaffiliated/invader) |
| 16:54.00 | *** join/#asterisk pzn (~pzn@pdpc/supporter/active/pzn) |
| 16:55.19 | pzn | I need to configure something in extensions.conf but I don't know how... suppose I do Dial(sip/201&sip/202)... then 201 is busy and 202 rings... while 202 is ringing, the person at 201 finishes the last conversation. then 201 does not ring (I'd like it to ring)... any hints? |
| 16:56.13 | aberrios | http://pastebin.com/Z5wf5KxB |
| 16:58.05 | igcewieling | pzn: no. Dial(sip/201&sip/202) will dial both devices at the same time. |
| 16:58.49 | pzn | igcewieling, yes, I know that. but if the call has not been answered and 201 is now available, how can I make some extension rule that will ring it? |
| 16:59.52 | aberrios | pzn, a timeout with a loop? |
| 17:00.41 | pzn | aberrios, interesting idea... thinking about it... |
| 17:01.13 | [TK]D-Fender | aberrios, And from the phone to *? |
| 17:02.39 | [TK]D-Fender | aberrios, And yes, that is the approach to pzn's question... |
| 17:03.33 | igcewieling | pzn: you will have to do that in your dialplan. |
| 17:03.37 | aberrios | [TK]D-Fender, That's odd the sip packets aren't showing up. Works fine when I allow alaw. |
| 17:04.29 | [TK]D-Fender | aberrios, they have to be... |
| 17:04.42 | [TK]D-Fender | abbYou've picked the qwrong target. In fact you shouldn't isolate at all |
| 17:10.40 | [TK]D-Fender | BRb |
| 17:12.20 | *** join/#asterisk Mon|A|rch (~SBean@72.29.180.35) |
| 17:12.41 | Mon|A|rch | So, I've noticed that I have a context within an IVR setup |
| 17:13.09 | Mon|A|rch | that doesn't appear to be able to access variables set up in contexts before it |
| 17:13.13 | *** join/#asterisk cmendes0101 (~cmendes01@wtnl.corp.tierra.net) |
| 17:13.17 | Mon|A|rch | is this common? do i need to use globals? |
| 17:13.34 | aberrios | Mon|A|rch, you could use channel variables |
| 17:13.37 | Mon|A|rch | for some reason i thought variables kept their scope even when you 'goto'd somewhere else |
| 17:13.43 | Mon|A|rch | huh |
| 17:13.52 | WIMPy | They do |
| 17:13.57 | Mon|A|rch | Set(var=othervar) right? |
| 17:14.16 | WIMPy | var=string |
| 17:14.24 | Mon|A|rch | k |
| 17:14.29 | Mon|A|rch | yeah |
| 17:14.31 | WIMPy | wouldn;t make much sense otherwise, would it? |
| 17:14.37 | Mon|A|rch | it's true |
| 17:15.03 | Mon|A|rch | hm, it looks like their accessible in other contexts in the IVR, it's just this one |
| 17:15.12 | Mon|A|rch | maybe I'm overwriting them or doing something dumb |
| 17:15.17 | Mon|A|rch | thanks for your help WIMPy |
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| 18:05.39 | *** join/#asterisk DoSJustin (~justin@vpn.bctconsulting.com) |
| 18:10.41 | igcewieling | ROFL. customer claims until last week they could dial *98 from their cell phone and check their Asterisk voicemail. |
| 18:14.01 | Qwell | Your customer is an idiot. ;/ |
| 18:14.37 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 18:18.21 | *** join/#asterisk LiuYan (~LiuYan@211.154.128.171) |
| 18:19.49 | jmetro | ive had a customer break a phone a month. dont ask me what he does. theyre polycoms too |
| 18:26.03 | *** join/#asterisk DoSJustin (~justin@vpn.bctconsulting.com) |
| 18:37.38 | *** join/#asterisk ruben231 (~OpenDial@112.198.79.97) |
| 18:38.19 | ruben231 | hi guys |
| 18:38.36 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 18:38.36 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 18:39.55 | ruben231 | hi guys how do i install ldap dependency..? or development..? |
| 18:40.15 | *** join/#asterisk cmendes0101 (~cmendes01@wtnl.corp.tierra.net) |
| 18:42.33 | leifmadsen | ruben231: asteriskdocs.org |
| 18:42.39 | leifmadsen | already documented |
| 18:42.48 | *** join/#asterisk jkroon (~jkroon@kerberos.uls.co.za) |
| 18:44.04 | *** join/#asterisk rue_work (~rue_mohr@24-207-105-65.eastlink.ca) |
| 18:44.29 | rue_work | asterisk 1.8.4.1 also does not support multiple mgcp devices |
| 18:44.45 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 18:44.45 | rue_work | could someone have simply not said that in the begginning? |
| 18:46.19 | chuckf | rue_work: if someone had, they would have deprived you of a learning oppurtunity |
| 18:48.04 | ruben231 | leifmadsen: ----->http://pastebin.com/8tawPEjb |
| 18:48.42 | leifmadsen | chuckf: +1 |
| 18:49.02 | leifmadsen | ruben231: yes, you're missing the dependencies |
| 18:49.13 | leifmadsen | like I said, this is documented, so you should follow the instructions |
| 18:49.20 | ruben231 | leifmadsen: how do i install i have ubutu-server |
| 18:49.44 | leifmadsen | ruben231: http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/ExternalServices_id291590.html |
| 18:49.49 | leifmadsen | again, I've documented already |
| 18:50.32 | jmetro | Leif is good at documenting things. |
| 18:50.53 | leifmadsen | I've had some experience |
| 18:52.32 | ruben231 | <PROTECTED> |
| 18:53.36 | leifmadsen | perhaps your version of ubuntu doesn't match the one we documented. You should use apt-cache search to find the right pacakge |
| 18:53.51 | rue_work | chuckf, they would also have not deprived me of 3 weeks of asking me for pointless system details |
| 18:53.56 | rue_work | glares at [TK]D-Fender |
| 18:55.07 | leifmadsen | glares at rue_work |
| 18:55.19 | rue_work | I'd like to see the newest chan_mgcp.c without downloading the whole source package, is there an browable copy? |
| 18:55.38 | leifmadsen | svn.asterisk.org |
| 18:55.54 | leifmadsen | with a /svn/ on the end |
| 18:55.55 | rue_work | forbidden |
| 18:55.57 | rue_work | ah |
| 18:56.00 | Qwell | rue_work: If it doesn't support more than 1 in 1.8, it won't support more than 1 in anything else. |
| 18:56.10 | leifmadsen | that's because no one uses mgcp |
| 18:56.20 | rue_work | qwell interesting |
| 18:56.35 | Qwell | Why? Nobody cares about MGCP. |
| 18:56.43 | rue_work | I think I made a comment that the source has never changed to support more than one, and a bunch of people told me I would be wrong |
| 18:56.55 | rue_work | <--- nobody does care |
| 18:57.14 | rue_work | /nick nobody |
| 18:57.38 | Qwell | I stand by my comment. |
| 18:57.46 | *** join/#asterisk jsjc (~Adium@124.Red-83-55-40.dynamicIP.rima-tde.net) |
| 18:58.03 | rue_work | http://svn.asterisk.org/svn/asterisk/trunk/channels/chan_mgcp.c |
| 18:58.14 | jmetro | seems like mgcp shouldnt be that disregarded |
| 18:58.24 | igcewieling | nobody uses it. |
| 18:58.28 | Qwell | ^^^^ |
| 18:58.30 | rue_work | as you can clearly see, all the mgcp variables are done as globals instead of being put in a struct |
| 18:58.31 | Qwell | because...nobody cares. |
| 18:58.46 | rue_work | so I get to be the person to make it work properly? cool |
| 18:58.57 | Qwell | Correct. |
| 18:58.58 | leifmadsen | yep |
| 18:59.00 | Qwell | Welcome to Open Source. |
| 18:59.04 | leifmadsen | welcome to open source |
| 18:59.05 | leifmadsen | Qwell: +1 |
| 18:59.13 | rue_work | I'm fine with that |
| 18:59.22 | igcewieling | seems like it would be a lot cheaper to throw the MGCP gateways off the roof and buy something which supports SIP. |
| 18:59.26 | [TK]D-Fender | tries to count how many times rue_work has been told to put the issue up on the tracker & mailing lists. |
| 18:59.28 | rue_work | I'm not fine with being given a runaround for 3 weeks |
| 18:59.41 | Qwell | [TK]D-Fender: It'll be ignored, as nobody cares. |
| 18:59.46 | [TK]D-Fender | overflows & recodes the counter as a LONGINT |
| 18:59.53 | rue_work | I'm sure the lists would have given me the same process that tk has for 3 weeks now |
| 18:59.56 | Qwell | rue_work: Go away. You aren't entitled to a damn thing. |
| 18:59.59 | igcewieling | rue_work: nobody uses MGCP so nobody can help you with MGCP. |
| 19:00.16 | Qwell | Feel free to take that nonsense elsewhere. |
| 19:00.17 | pabelanger | Qwell: harsh |
| 19:00.33 | rue_work | dont you want to make asterisk _the_ pbx system that rules? |
| 19:00.43 | Qwell | It's MGCP. _NOBODY CARES_ |
| 19:00.49 | rue_work | isn't that why your here? to be a part of the best system support there is? |
| 19:00.51 | [TK]D-Fender | rue_house, It does rule. MGCP is an exception |
| 19:00.53 | seanbright | you're* |
| 19:00.56 | kikohnl | I showed him yesterday that 11.2.1 DOES support multiple gateways |
| 19:01.06 | rue_work | does freepbx do mgcp? |
| 19:01.13 | jmetro | im hungry, lets go get tacos. |
| 19:01.15 | Qwell | rue_work: Go ask them. |
| 19:01.22 | rue_work | kikohnl, http://svn.asterisk.org/svn/asterisk/trunk/channels/chan_mgcp.c |
| 19:01.32 | [TK]D-Fender | rue_work, And we take exception to your sense of entitlement and complete lack of being part of the solution as you bitch at all of us |
| 19:01.44 | Qwell | drmessano: facebook, really? |
| 19:01.47 | rue_work | that file, clearly by the code, it may show muliple gateways, but everything is global, so it wont work |
| 19:01.57 | [TK]D-Fender | kikohnl, I did note that.... |
| 19:02.26 | seanbright | so make it support multiple gateways |
| 19:02.26 | seanbright | what am i missing? |
| 19:02.26 | [TK]D-Fender | rue_work, No, it doesn't |
| 19:03.13 | kaldemar | ruben231: libldap2-dev worked for you 5 days ago. what's the problem with it now? |
| 19:03.57 | kikohnl | I still don't get the fascination with MGCP |
| 19:04.34 | WIMPy | Anything is better than SIP, so why not MGCP? |
| 19:05.41 | [TK]D-Fender | kikohnl, One neurotic twit. |
| 19:09.25 | kikohnl | rue_work the sure looks like 11.2.1 does handle multiple gateways: http://pastebin.com/UxHHc0TE |
| 19:10.17 | [TK]D-Fender | <rue_work> asterisk 1.8.4.1 also does not support multiple mgcp devices <--- The latest in that branch is 1.8.20.1 which is at least 16 releases newer. |
| 19:10.57 | [TK]D-Fender | rue_work, Your comparison is therefor of little interest |
| 19:12.16 | kikohnl | off to the gym, PT for my knee. |
| 19:12.21 | rue_work | I think I made a comment about you guys rejecting probelms from people using anything older than the last hourly release |
| 19:12.36 | Qwell | ~upgrade asterisk |
| 19:12.36 | infobot | Before requesting assistance, you should be running the latest version of a supported release branch. See the channel topic for the latest versions available in currently supported branches. |
| 19:12.39 | rue_work | kikohnl, the problem is when it tries to initialize the devices |
| 19:13.24 | WIMPy | That's the way it is. If you're using old versions, you have to live with all the bugs or missing features that have been dealt with alerady. |
| 19:13.30 | rue_work | and that is the ultimate disclaimer, technically, you dont have to answer anything |
| 19:13.46 | Qwell | have to? |
| 19:13.57 | Qwell | Who said we had to answer even if you were? |
| 19:14.05 | Qwell | You are not entitled to anything. |
| 19:14.10 | rue_work | I meant to fullfill your role here |
| 19:14.31 | rue_work | your reason you are here |
| 19:14.32 | Qwell | What, exactly, do you think my role here is? |
| 19:14.34 | WIMPy | What do you think, this channel is? |
| 19:14.39 | [TK]D-Fender | rue_work, And yours was 3 YEARS old and you provided no real debug. Again, null and void. |
| 19:14.53 | rue_work | I think you back the asterisk project and you want to help people reep its rewards |
| 19:15.11 | Qwell | I'll help people that I see as benefiting the project, sure. |
| 19:15.11 | rue_work | to show them how great it is |
| 19:15.27 | rue_work | but you also help people use asterisk |
| 19:15.41 | rue_work | newbs who need directions on writing extensions.conf |
| 19:15.49 | Qwell | What exactly is your point? |
| 19:16.02 | [TK]D-Fender | Qwell, Some people take along time to make their pointless |
| 19:16.40 | rue_work | sometimes it helps people more to say that you dont know an answer than to ask them a lot of questions about what they are doing |
| 19:17.13 | Qwell | Why am I even bothering responding? |
| 19:17.14 | rue_work | all I was every told in school was "that question is beyond the scope of this course" what kinda answer is that? |
| 19:17.26 | rue_work | you want to prove your right, and you are |
| 19:19.54 | [TK]D-Fender | rue_house, First course is to prove what's actually happening. You didn't, and immediately and consistently refused to provide backup. You are the one required to prove your situation first. |
| 19:20.32 | [TK]D-Fender | I SAY IT DOESN'T WORK, YOU GO PROVE IT TO ME! |
| 19:20.44 | [TK]D-Fender | AND DON'T GIVE ME ANY SHIT! (c) drmessano |
| 19:22.07 | rue_work | if I had the time and the url, I would go thru the irc logs to show you I was never asked for a completel network dump |
| 19:22.25 | Qwell | ~upgrade asterisk |
| 19:22.25 | infobot | Before requesting assistance, you should be running the latest version of a supported release branch. See the channel topic for the latest versions available in currently supported branches. |
| 19:22.58 | rue_work | yes, if its more than an hour old, you wont asnwer questions about it |
| 19:23.05 | Qwell | That's right. |
| 19:31.31 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 19:33.55 | [TK]D-Fender | <[TK]D-Fender> rue_work, And yours was 3 YEARS old and you provided no real debug. Again, null and void. |
| 19:34.48 | [TK]D-Fender | rue_work, And your latest "comparison" you somehow felt we care at all about another branch at a point 16+ releases behind. |
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| 20:14.39 | srl295 | hi all… trying out incrediblePi (incrediblepbx on the pi) based on recommendations here. It seems to run well ,but be a bit… heavy handed, as far as a un*x box goes.. special scripts running on login such as /etc/profile.d/helloworld.sh - I appreciate that it's a 1-click solution, but am I going to run into problems if I disable some of these scripts and then try to update? |
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| 20:18.09 | [TK]D-Fender | srl295, this isn't a support channel for their distro.... |
| 20:18.54 | srl295 | srl295, I know - but it was suggested here. My question is whether it is still a good distro to use or whether I should try to roll my own |
| 20:19.55 | navaismo | srl295, use your own |
| 20:20.32 | *** join/#asterisk ChkDigit (~u388mw@74.3.144.66) |
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| 20:30.32 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
| 20:31.05 | gavimobile | anyone know why my asterisk won't connect properly with my gmail calendar? res_calendar_caldav.c:157 caldav_request: Unknown response to CalDAV calendar myGoogleCal, request REPORT to /calendar/dav/gavimobile@gmail.com/events/: Could not read status line: connection was closed by server |
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| 20:37.57 | pzn | [TK]D-Fender, aberrios, igcewieling: had to go out... now I'm back. the approach to have many Dial() with timeout worked as expected. thanks!!! |
| 20:40.22 | [TK]D-Fender | pzn, Lets see how you did yours |
| 20:41.23 | *** join/#asterisk xSmurf (~MrSmurf@unaffiliated/mrsmurf) |
| 20:42.20 | *** join/#asterisk genii-around (~quassel@ubuntu/member/genii) |
| 20:42.26 | pzn | [TK]D-Fender, I did this: 10 lines of Dial(sip/801&sip/802,3) in the extensions.conf, then the total time is 30 seconds for a call to "really timeout" |
| 20:44.01 | [TK]D-Fender | ~pb |
| 20:44.02 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
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| 20:45.06 | *** join/#asterisk zhando (~user@c-67-161-122-77.hsd1.wa.comcast.net) |
| 20:45.36 | gavimobile | [TK]D-Fender: if you got a dollar for all the help you contributed you'd be a millionair |
| 20:45.47 | gavimobile | [TK]D-Fender: is the man! |
| 20:47.36 | zhando | hi.. I'm running 1.4.17 and trying to upgrade to 11.. I blacklist and want to know if there's a link to an update of this: http://www.jackenhack.com/adding-blacklist-to-an-asteriskhome-pbx-voip-server |
| 20:48.00 | zhando | does anyone know of a link rather... |
| 20:49.25 | gavimobile | jmls: you around? |
| 20:49.51 | jmls | no |
| 20:49.53 | jmls | ;) |
| 20:50.38 | [TK]D-Fender | pzn, kinda messy... jsut think of the missed calls you'll log if it takes 8 tries to get an answer.. |
| 20:50.55 | gavimobile | im in your mercy.. look what you wrote here http://ibot.rikers.org/%23asterisk-dev/20111029.html.gz at 12:11.58 |
| 20:51.27 | [TK]D-Fender | zhando, that is referring to Asterisk@Home which became FreePBX at least half a decade ago. |
| 20:51.36 | gavimobile | im on day 4 trying to get my calendar to reconnect |
| 20:51.37 | [TK]D-Fender | zhando, Consider it completely outdated. |
| 20:51.44 | pzn | [TK]D-Fender, yes... really messy... but I couldn't manage how to fix this soon... I'll think about a better solution tomorrow |
| 20:51.50 | gavimobile | jmls: ^ |
| 20:52.04 | [TK]D-Fender | pzn, Ok, keep at it and I'll help you with your final then as well |
| 20:52.27 | pzn | [TK]D-Fender, when the receptionst had 2 phones, that was ok... but 1 of her phones broke today, so I had to find a quick solution |
| 20:52.32 | WIMPy | Set the no missed calls flag? |
| 20:52.50 | [TK]D-Fender | zhando, Your system is all bust certainly far too outdated to "upgrade" live. You would be far better served to reinstall from scratch given that scenario |
| 20:53.13 | gavimobile | im starting to think that google make some changes |
| 20:53.14 | genii-around | Would anyone near Toronto be interested in some old telco stuff for free if you pick it up? Norstar ( Meridian) NT5B20 wallmount with DR5 module.. Bayly OMNIflex 19" rackmount with T1 Quad card 051539, one PairGain HRU-402 HDSL endpoint card in single card enclosure box, one ADC H2TU-R-402 HDSL endpoint card in double card enclosure box. |
| 20:53.15 | [TK]D-Fender | WIMPy, Still fugly at best and assumes the phoe supports it. |
| 20:53.26 | [TK]D-Fender | WIMPy, And the the dead-air answer risks.... |
| 20:53.29 | jmls | oh. crap. |
| 20:53.44 | gavimobile | jmls: what was I supose to do? there's nothing else on google |
| 20:53.56 | WIMPy | Yes, it sounds messy, but I don;t know the story behind it. |
| 20:54.09 | zhando | [TK]D-Fender: pretty much what I'm doing.. |
| 20:54.34 | [TK]D-Fender | WIMPy, call X people. if someone is busy and 4/5 are ringing and they then become availab, begin ringing them too (cath-up) |
| 20:54.58 | WIMPy | Ah. THE standard issue. |
| 20:55.46 | WIMPy | I'd do it with local channels that retry the devices. |
| 20:56.00 | jmls | yah. my memory is not that good. |
| 20:56.06 | jmls | at a guess |
| 20:56.09 | jmls | a) firewall |
| 20:56.17 | jmls | service iptables stop |
| 20:56.24 | gavimobile | jmls: tried that |
| 20:56.25 | WIMPy | Would be nice to retain order, however. |
| 20:56.29 | gavimobile | :-( |
| 20:56.30 | jmls | b) invalid credentials |
| 20:57.03 | gavimobile | jmls: blasphamy |
| 20:57.15 | gavimobile | this is maddness |
| 20:57.33 | gavimobile | you were talking about something with the c file |
| 20:57.50 | gavimobile | can you see events? |
| 20:57.53 | [TK]D-Fender | WIMPy, Exactlt as I'd do it |
| 20:58.03 | jmls | nah - I didn't commit the changes back, so I didn't change c code |
| 20:58.13 | gavimobile | jmls: also what version of neon are you using? and do you have both packages? |
| 20:58.22 | gavimobile | neon && neon-devel |
| 20:58.51 | gavimobile | another thing I found on google was talking about neon being an issue than recompiling |
| 20:58.53 | jmls | definitely config, or credentials, or calendar name wrong or firewall or calendar not set up to allow access from api |
| 20:59.08 | jmls | will try to remember |
| 20:59.45 | gavimobile | jmls: how do I allow access from api? also can you tell me which neon* packages you have installed and what versions are they? |
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| 21:04.10 | jmls | gavimobile : as mentioned, I haven't used res_calendar for some time, so I don't have neon installed |
| 21:04.25 | jmls | I build several asterisk boxes per month, each different ;) |
| 21:04.35 | jmls | then scrap them and build more |
| 21:04.35 | gavimobile | jmls: alright man, thanks for your help! |
| 21:05.00 | jmls | access from api is done from settings in the main admin page on google |
| 21:09.46 | gavimobile | jmls: ill have a look |
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| 21:16.23 | zhando | my ancient blacklist application uses asterisk "extra-sounds".. Is it ok to untar these sounds from http://downloads.asterisk.org/pub/telephony/sounds/releases/asterisk-extra-sounds-en-gsm-1.4.13.tar.gz to /var/lib/asterisk/sounds/en ? |
| 21:17.59 | zhando | oops found more "recent" versions of those sounds... |
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| 21:47.32 | solitude88 | can someone please go here and tell me if you can get the inbound DID number part to work https://store.freepbx.com/store.aspx |
| 21:48.05 | solitude88 | How do they expect to make money if they have design that requires users to guess whats available. |
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| 21:53.25 | srl295 | solitude88, freepbx works fine for me, been using their service for a couple of years |
| 21:55.09 | solitude88 | I'm excited to try it out I was speaking about the UI, you can't start using their sipstation service without DID's and their website makes it virtually impossible to do this but I just recevied an email from their support, remarkable fast, and they're going to provision for me. |
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| 21:56.06 | srl295 | solitude88, that is difficult. looks like it's missing suggestions of other possibilities. (I think the web page was a bit more guided when I signed up) |
| 21:56.48 | solitude88 | srl295 I was just told that a new UI will be out next week. |
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| 22:05.26 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:06.55 | rue_work | can anyone propose a suggestion as to why (without me posting network dumps, config files, and log files) that Milliwatt() would not be heard on a sip set even tho it works on an analog set? |
| 22:08.20 | [TK]D-Fender | rue_work: Suggestion : packets aren't even reaching your phone. Now pastebin the call aattempt with SIP debug enabled. |
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| 22:09.09 | n8ideas | Is it possible to disable the "confirm" announcement on FollowMe()? |
| 22:17.11 | *** join/#asterisk cmendes0101 (~cmendes01@wtnl.corp.tierra.net) |
| 22:18.08 | [TK]D-Fender | n8ideas: Nope |
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| 22:24.25 | ChannelZ | rue_house: can you give me an estimate to repair my roof without me telling you how messed up it is, how big it is, what kind of material it is? |
| 22:28.17 | *** join/#asterisk din3sh (~din3sh@41.136.83.114) |
| 22:28.25 | din3sh | hello all |
| 22:29.53 | ChannelZ | holla |
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| 22:30.55 | *** part/#asterisk klnikita (~klnikita@unaffiliated/klnikita) |
| 22:31.43 | [TK]D-Fender | ChannelZ: https://www.youtube.com/watch?v=TFO2_pgWz9w |
| 22:32.40 | *** join/#asterisk Uthark (~Uthark@190.0.58.186) |
| 22:33.08 | rue_work | [TK]D-Fender, I know they are, I can place a call to it just fine, the only thing not working is milliwatt |
| 22:34.07 | rue_work | tho... I didn't answer() |
| 22:34.21 | [TK]D-Fender | HMMM |
| 22:34.33 | [TK]D-Fender | I wonder if that might impact audio not being set up... |
| 22:34.39 | [TK]D-Fender | Sorta like ... for Echo() |
| 22:34.40 | ChannelZ | That was going to be my suggestion. |
| 22:34.44 | ChannelZ | But I didn't want to show my work. |
| 22:34.55 | ChannelZ | :P |
| 22:34.59 | rue_work | haha! I'm right! adding answer() fixed it! |
| 22:35.23 | rue_work | I thought almost everything automatically answer()'d |
| 22:35.42 | rue_work | (so you were right too) |
| 22:35.44 | [TK]D-Fender | Assume != think |
| 22:35.52 | WIMPy | No, and looking at your phone you should have been able to see that. |
| 22:37.06 | *** part/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 22:39.58 | rue_work | exten => 20[0-9],1,NoOp(); wont match 200 thru 209 will it? |
| 22:40.12 | [TK]D-Fender | rue_work: nope |
| 22:40.21 | rue_work | I wonder where I got the idea it would |
| 22:41.06 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 22:41.07 | rue_work | _20X |
| 22:41.39 | navaismo | _20[0-9] |
| 22:41.55 | rue_work | :) |
| 22:42.50 | [TK]D-Fender | It's nice to see that all of the most basic bits of using * have been forgotton.... |
| 22:43.49 | *** part/#asterisk genii-around (~quassel@ubuntu/member/genii) |
| 22:44.37 | rue_work | hey, I'v not done anything for 2 years now :) |
| 22:45.08 | rue_work | put asterisk on a machine with raid and sit it in a corner, it just goes forever |
| 22:45.49 | rue_work | and if some nut wants to run 5 companies off the same phone number, so be it |
| 22:46.02 | rue_work | and its cheaper than a panasonic system |
| 22:46.46 | solitude88 | was just just told I cant use voip with my dial up service |
| 22:47.12 | [TK]D-Fender | solitude88: What happened to you 5/5/? |
| 22:47.27 | solitude88 | :) Im joking |
| 22:47.54 | solitude88 | but my call quality is as if I am on dial up :) |
| 22:48.11 | [TK]D-Fender | solitude88: And you could get away with 1 channel on dialup.... and nothing else... |
| 22:48.23 | solitude88 | lol |
| 22:48.36 | solitude88 | I'm trying wireless right now |
| 22:48.40 | solitude88 | but not going so well |
| 22:48.43 | solitude88 | it works though |
| 23:01.13 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 23:02.51 | SeRi | so on the new polycom software callpark is licensed? |
| 23:08.53 | *** join/#asterisk saint_ (~saint@c-68-38-56-184.hsd1.nj.comcast.net) |
| 23:08.55 | saint_ | hi all |
| 23:09.13 | WIMPy | lo you |
| 23:09.35 | *** join/#asterisk Mon|A|rch (~SBean@72.29.180.35) |
| 23:10.06 | Mon|A|rch | so I'm looking into hangup handlers, but all the documentation is for < 1.2 or so |
| 23:10.13 | Mon|A|rch | do they still work essentially the same as they did then? |
| 23:10.23 | din3sh | when load balancing, does both server have to be of same exact specification? |
| 23:10.28 | saint_ | I'm reading the paging part in the documentation. I just want to try it on my small installation. Is there any thing to do on Digium phones for them to work with underchin paging ? |
| 23:12.08 | Mon|A|rch | nvm, found recent docs |
| 23:12.37 | din3sh | WIMPy: when load balancing, do both server have to be of same exact specification? :) |
| 23:13.28 | pabelanger | Mon|A|rch: hangup handlers are asterisk 11+ |
| 23:13.32 | pabelanger | if we are talking about the same thing |
| 23:13.44 | Uthark | din3sh: If you expect to handle the same load, yes. |
| 23:14.07 | Mon|A|rch | pabelanger, basically, i was wrong, is what that amounted to |
| 23:15.38 | din3sh | Uthark: even if one server will be main? |
| 23:15.45 | WIMPy | din3sh: Load balancing WHAT? |
| 23:16.58 | [TK]D-Fender | din3sh: How do you expect "balance" if they aren't the same spec. |
| 23:17.06 | Mon|A|rch | pabelanger, is there anything similar to hangup handler behavior in 1.8? |
| 23:17.15 | [TK]D-Fender | Go hom Dictionary .. YOU'RE DRUNK |
| 23:17.25 | WIMPy | Wheighted balance? |
| 23:17.26 | Mon|A|rch | can i do it on a context by context basis? |
| 23:17.34 | din3sh | ermmm |
| 23:17.42 | [TK]D-Fender | Mon|A|rch: Please clarify exactly which "handler" youer referring to... |
| 23:17.54 | pabelanger | Mon|A|rch: aside from the default exten => h, no |
| 23:18.11 | pabelanger | https://wiki.asterisk.org/wiki/display/AST/Hangup+Handlers |
| 23:18.14 | Mon|A|rch | haven't heard of exten => h |
| 23:18.16 | Mon|A|rch | thanks |
| 23:18.24 | din3sh | 240 simultaneous calls to run on high availability, does this require load balancing? |
| 23:18.35 | Uthark | din3sh: You can use an active-passive approach |
| 23:19.11 | din3sh | active-passive? when one goes down, the other takes over? |
| 23:19.13 | Uthark | din3sh: With a more powerful server as main/master, and a backup server with less specs as backup |
| 23:19.37 | pabelanger | din3sh: load balancing just splits the load |
| 23:19.42 | pabelanger | what are you trying to do |
| 23:19.46 | pabelanger | have redundancy? |
| 23:19.55 | din3sh | yes redundancy |
| 23:20.09 | din3sh | what is the best solution to go with? |
| 23:20.27 | din3sh | minimum downtime, or automatic failover |
| 23:21.12 | Skavin | should directrtpsetup create an outbound invite with direct media if the inbound SIP call has a dial(Local/ to another extension then a dial(SIP/? |
| 23:21.23 | pabelanger | You tell us, what do you want to achieve |
| 23:22.51 | *** join/#asterisk zerohalo (~zerohalo@74.61.196.236) |
| 23:23.41 | Skavin | from my testing having a dial(Local/ seems to cause directrtpsetup not to work. is this a limitation or a bug? |
| 23:23.56 | din3sh | call centre with 240 simultaneous inbound E1 calls (quad E1 card) to run on high availability |
| 23:24.49 | WIMPy | First of all you need some hardwar to switch over the lines. |
| 23:24.53 | Uthark | din3sh: Do you have a backup server with that same card? |
| 23:25.18 | din3sh | yes I could have same specs on a 2nd server |
| 23:25.18 | Mon|A|rch | pabelanger, so, if i use the h extension in a context, and at some point I travel through a pile of other contexts, will asterisk recursively go through those contexts looking for h? or will it only execute the h extension in the context that the hangup occurred in? |
| 23:25.29 | Mon|A|rch | ignore the run-on |
| 23:26.20 | pabelanger | asterisk will look for the exten => h in the current context |
| 23:26.27 | Mon|A|rch | okay |
| 23:26.34 | Mon|A|rch | good deal |
| 23:26.36 | Mon|A|rch | thanks |
| 23:26.42 | Skavin | but will call the h of a dialed context on answer |
| 23:26.47 | din3sh | with redundancy in mind, i wanted to connect 2 E1s on 1st server, and 2 E1s on the 2nd server, both having quad cards |
| 23:27.20 | WIMPy | And then do a manual fail over? |
| 23:27.35 | din3sh | yes, in case of 1 server being down |
| 23:28.03 | Mon|A|rch | Skavin, what? |
| 23:28.06 | Mon|A|rch | why would it do that |
| 23:28.08 | [TK]D-Fender | din3sh: That still takes down 1/2 your E1's. |
| 23:28.17 | [TK]D-Fender | din3sh: So for not goo. |
| 23:28.20 | [TK]D-Fender | good* |
| 23:28.21 | Mon|A|rch | i might be confused on what you mean |
| 23:28.24 | [TK]D-Fender | far* |
| 23:28.42 | din3sh | [TK]D-Fender: what would be the ideal solution? |
| 23:28.46 | pabelanger | well, loss of power to his site takes down 100%, so there is always a single point of failure |
| 23:29.06 | WIMPy | UPS? |
| 23:29.17 | pabelanger | last only so long |
| 23:29.21 | Skavin | as the channel is bridged it will destroy the intermediate channels and cal their h |
| 23:29.35 | WIMPy | din3sh: Look at fail-over switches. I think Digium has such a thing as well. |
| 23:29.37 | pabelanger | so, figure out how long you want an outage to be, and work back from that |
| 23:29.52 | pabelanger | Assuming din3sh also doesn't have all his E1 from the same provider |
| 23:29.54 | Skavin | I found this strangeness last week in 8.20.1 |
| 23:30.25 | WIMPy | If they aren't from the same provider he will have some trouble with the usual cards. |
| 23:30.51 | Mon|A|rch | Skavin, you figure out a way to work around it? |
| 23:30.52 | WIMPy | SOne of the nice things noone warns you about until you find out yourself. |
| 23:31.02 | Mon|A|rch | does the hangup cause variable get set? |
| 23:31.25 | din3sh | all the E1s are from same provider lol |
| 23:31.46 | pabelanger | din3sh: so, what is the plan if the CO goes down? |
| 23:31.47 | *** join/#asterisk radic (~radic@dslb-094-216-237-145.pools.arcor-ip.net) |
| 23:32.00 | din3sh | ouch! |
| 23:32.10 | Skavin | example sip -> 1234@inboundcontext -dial local-> abc@outbound context -dial sip-> 12345678@peer |
| 23:32.14 | din3sh | back here we only have one E1 provider anyway |
| 23:32.16 | WIMPy | Good for the usual hardware used with Asterisk, but bad for availability. |
| 23:33.10 | Skavin | when the OK comes in from 12345678@peer the h in outbound context will fire. then the bye comes in from 12345678@peer the h for inbound context will fire |
| 23:33.17 | Mon|A|rch | so this behavior doesn't happen on a goto, it happens when you're dialing between contexts |
| 23:33.22 | Mon|A|rch | that's bizarre |
| 23:33.25 | *** join/#asterisk jmetro (~nickserv@75-150-221-198-Illinois.hfc.comcastbusiness.net) |
| 23:33.38 | Skavin | yea its dial magic |
| 23:34.04 | din3sh | WIMPy: i practically already have the hardware, 2 HP DL380 G7 servers, 2 quad cards |
| 23:34.04 | Mon|A|rch | well, this particular system won't quite be doing that] |
| 23:34.09 | Mon|A|rch | but i appreciate the warning |
| 23:34.14 | Mon|A|rch | I'll keep it in mind for later |
| 23:34.28 | din3sh | how to build a high availabilty solution with what I already have? |
| 23:34.47 | WIMPy | >>First of all you need some hardware to switch over the lines. |
| 23:34.54 | Skavin | but I need to have different A party format depending on my outbound peer and I can sim ring and have calls going to both peers so need that seperation |
| 23:34.56 | pabelanger | din3sh: setup server 1, duplicate it for server 2, ???, profit |
| 23:35.19 | navaismo | stupid question that i cant answer with google's help: Which tool is used to create a asterisknow, elastix ,trixbox ISO CD? |
| 23:35.22 | pabelanger | serious though, setup the two servers, then once everthing is working... kill the power to one and see what happens |
| 23:35.53 | pabelanger | navaismo: not sure what they use, but dd will do it |
| 23:35.56 | pabelanger | man dd |
| 23:36.44 | Skavin | local dial also stops directrtpsetup from creating the call with direct media :( |
| 23:37.18 | Uthark | din3sh: Have you signed some kind of SLA with your E1 provider? |
| 23:37.47 | navaismo | dd to create an ISO cd, ok thats new to me, i have used DD to create images or clone disk but not to create a installable ISO |
| 23:38.33 | din3sh | Uthark: nope, 1 single provider out here, monopoly rules |
| 23:39.19 | rue_work | if FOO=42 then ${FOO} is 42 I cant $66{FOO} can I? |
| 23:39.25 | rue_work | to get 6642 |
| 23:39.38 | [TK]D-Fender | ... |
| 23:39.47 | leifmadsen | sure you can |
| 23:39.50 | leifmadsen | but not like that |
| 23:39.53 | [TK]D-Fender | 66${FOO} |
| 23:39.54 | leifmadsen | 66${FOO} |
| 23:39.59 | rue_work | ah |
| 23:40.03 | rue_work | thankyou |
| 23:40.04 | *** join/#asterisk danfromuk (IceChat77@unaffiliated/danfromuk) |
| 23:41.28 | danfromuk | When the Beep is played followed by the Record application, the beep doesnt properly clear the line and the beep tone continues in a distorted way. Has anyone experienced this and worked out how to resolve it? |
| 23:42.20 | Uthark | din3sh: OK, so your single point of failure is your E1 provider. If their infrastructure fails, there is nothing you can do |
| 23:42.43 | din3sh | yes i guess so |
| 23:42.49 | Uthark | din3sh: So you can set up those HP servers in a active-passive setup |
| 23:43.39 | din3sh | 1 running with 4 E1s, and the other sitting idle as backup? |
| 23:44.04 | WIMPy | That's by far the easiest way. |
| 23:44.57 | din3sh | if the load is split physically (2 x 2E1s on each server), how to make agents register on 1 same single ip? |
| 23:45.00 | Uthark | Yes, or you can set up a third server that acts as a "load balancer" or "call distributor": receives calls from the provider and distribute them to the 2 HP's |
| 23:46.20 | WIMPy | What's the point of doing load balancing with only two boxes, if each of them has to be ablte to take the full load individually anyway. |
| 23:46.23 | WIMPy | ? |
| 23:46.23 | din3sh | manual switch over seems simplest |
| 23:46.25 | rue_work | danfromuk, beep like playtones? |
| 23:46.47 | *** join/#asterisk radic (~radic@dslb-094-216-241-227.pools.arcor-ip.net) |
| 23:46.52 | danfromuk | No. I think its beep.gsm. Whatever the voicemail application uses |
| 23:47.07 | rue_work | exten => 264,7,Playback(beep) |
| 23:47.07 | rue_work | ? |
| 23:47.14 | Uthark | If your call center can tolerate a few seconds of downtime, I think you're OK: go for manual switch |
| 23:47.15 | din3sh | point being i guess if server1# is down, server2# still has 120 calls incoming (at least)? |
| 23:47.37 | danfromuk | rue_work: yes. playback(beep) |
| 23:47.46 | Uthark | din3sh: yes, and those calls aren't affected by the other server's being down |
| 23:48.24 | din3sh | but how to have single IP for agents to log on to |
| 23:48.27 | din3sh | ? |
| 23:49.03 | WIMPy | The same way you use the same E1 lines. Change the cable over to the other. |
| 23:49.20 | *** part/#asterisk cusco (~tralala@2001:41d0:1:6cef::8899) |
| 23:49.34 | Uthark | There is a tool on Linux, can't remember it's name, which is installed in the slave server, and pings every nth second to the master |
| 23:49.50 | pabelanger | din3sh: just do an active/passive setup. It will be the simplest for you |
| 23:50.01 | WIMPy | There are more elegant ways, but if you have to change 4 cables anyway, a 5th doesn't matter. |
| 23:50.04 | Uthark | What pabelanger said. |
| 23:50.42 | pabelanger | otherwise you need something like pacemaker |
| 23:52.10 | pabelanger | but this is not asterisk specific, you need to read up on Linux HA in general |
| 23:53.08 | din3sh | reading...active passive seems simpler |
| 23:54.20 | pabelanger | yes |
| 23:54.33 | din3sh | HA requires both servers to be exact same??.. |
| 23:54.34 | Uthark | Please consider that by going active-passive, all your calls will be interrupted |
| 23:54.36 | pabelanger | High availability takes a lot of work, and testing |
| 23:54.46 | Uthark | After the failure |
| 23:55.44 | WIMPy | Having both servers load balancing will still kill 3/4 of your calls. |
| 23:57.35 | Uthark | WIMPy: 3/4? Why? |
| 23:57.43 | din3sh | 1/2? |
| 23:58.15 | WIMPy | Because half of the phones will be on the wrong server as well? |
| 23:59.18 | din3sh | thats why wanted to know how to have a single sip IP |
| 23:59.23 | din3sh | for both servers |
| 23:59.24 | saint_ | I'm reading the HINT part at http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/DeviceStates_id265377.html in the book. |
| 23:59.32 | din3sh | is 3rd server needed? |
| 23:59.54 | saint_ | I do not understand it. Is it a function / application that has to be in for every exten => ..? |