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| 01:00.35 | edong23 | right |
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| 01:22.16 | joobie | hey guys, my phone boots up and gets stuck on the HTC boot screen.. how do i resolve it? I've flashed the recovery image and unlocked the bootloader.. |
| 01:22.20 | joobie | i have a htc one x |
| 01:24.17 | eirirs | #android , I assume? |
| 01:25.22 | joobie | woops thanks eirirs |
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| 02:36.05 | mnathani | How do I get multiple SIP phones from behind a NAT to register to the same Asterisk server (on a public IP) |
| 03:03.07 | igcewieling | mnathani: that should work by default, assuming you didn't do something silly like port forward 5060/udp. |
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| 03:58.09 | mnathani | igcewieling: Is STUN required? |
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| 04:32.41 | igcewieling | mnathani: almost never |
| 04:33.15 | igcewieling | if you enable asterisk's support for remote natted endpoints (with nat=yes or some varient) and any OTHER nat traversal things you do may break things. |
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| 06:24.17 | Bradada | Hi, I tried to add a custom module(with external library) to asterisk by following the wiki but without any luck. Is there any more well-documented article? Wiki is way to simple. |
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| 06:38.55 | ChannelZ | was it compiled already? |
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| 06:53.28 | schmidts | good morning |
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| 06:55.39 | ChannelZ | puts his hands in the air |
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| 07:03.05 | v0lZy | Morning |
| 07:03.44 | v0lZy | Can anyone offer me some advice on using asterisk to call skype names? |
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| 07:08.36 | Bradada | ChannelZ: Yes, I'd compile it. |
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| 07:14.12 | kaldemar | ~ask |
| 07:14.12 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 07:14.15 | kaldemar | Bradada: ^ |
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| 07:21.03 | v0lZy | Hi kaldemar |
| 07:22.06 | kaldemar | hello |
| 07:23.00 | v0lZy | kaldemar, whats the best way to get asterisk to make calls to skype users? Right now im thinking some kind of gateway... |
| 07:23.19 | v0lZy | Where one side of the gateway behaves like a skype client and the other side like a pbx or something |
| 07:23.28 | v0lZy | SIPtoSIS looks extremely ugly though. |
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| 07:27.48 | schmidts | v01zy there is a solution which runs on windows, where they simulate a lot of clients in virtual machines, iirc the company who build this was from italy but i didnt remeber the name |
| 07:29.01 | v0lZy | schmidts: sounds exactly the same as SIPtoSIS |
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| 07:29.35 | v0lZy | The way i understood SIPtoSIS is that they require multiple user with multiple skype profiles (basically each user has their own home directory with their own profile) |
| 07:29.44 | kaldemar | v0lZy: since skype for asterisk got the boot, i wouldn't have high hopes getting it to work in a decent manner. there used to be a SIP connection to skype but i have a feeling that it has been discontinued also. |
| 07:30.06 | v0lZy | then all these users spawn their own X sessions, running their own skype client, and SIPtoSIS uses those clients to make/receive calls... FUGLY. |
| 07:30.29 | v0lZy | kaldemar: Yeah, I've come accross that while googling |
| 07:30.53 | v0lZy | Microsoft is really viewing anyone who develops software is its competition apparently. |
| 07:31.04 | v0lZy | Was all good before they bought Skype the way i understood it |
| 07:31.43 | v0lZy | Then support for linux became dodgy at best, and they pulled out from the digium contract... |
| 07:32.57 | v0lZy | schmidts: simulating a lot of clients in virtual machines... thats one damn ugly way to do it, but i guess short of running a windows terminal server, theres no other way to do it.. |
| 07:39.06 | schmidts | v01Zy it is atleast a windows application/solution they offer, so it might be a terminal server |
| 07:41.40 | v0lZy | thats an L btw :D |
| 07:42.07 | v0lZy | schmidts: all these terminal solutions sound inheritly bad |
| 07:42.29 | v0lZy | Sounds like I dont know, a bash script using sendkeys |
| 07:42.58 | v0lZy | or actually placing a fixed sized window somewhere then controling then scripting the mouse cursor to control the application |
| 07:43.24 | schmidts | :D ok voLzy |
| 07:43.28 | v0lZy | I can accept that kind of mechanics if it works reliably, but this doesnt sound so :D |
| 07:43.40 | v0lZy | oh, no, the second one is a zero though ... |
| 07:43.45 | v0lZy | :D |
| 07:45.24 | v0lZy | And the fact that there isnt a command line skype utility u could just suid or something.. |
| 07:45.39 | v0lZy | that it *absolutely* needs a graphical environment |
| 07:45.49 | v0lZy | all this rings alarms. |
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| 08:24.55 | nunne | Anyone have any experience with asterisk sip -> microsoft lync? I have problems with getting no audio in conference.. Dialin/out + redirect to cell phones works as it should. But conferecing does not. I don't think anything is wrong with asterisk.. since everything else is working. but anyone here have experience with lync? |
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| 11:26.12 | Ice_Strike | Is Pentium Dual-Core CPU E5300 2.Ghz and 3GB RAM powerful enough for approx 6 concurrent channels? |
| 11:26.22 | Ice_Strike | Via SIP and ULAW codec |
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| 11:44.50 | schmidts | Ice_Strike simply yes. atleast even an asterisk running on a rasberry pi can do more calls than 6 concurrent |
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| 11:46.56 | Ice_Strike | lol |
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| 11:47.12 | Ice_Strike | rasberry pi is slow as hell on xmbc |
| 11:47.15 | Ice_Strike | i give up |
| 11:49.50 | ghost75 | raspberry is pretty expensive if you ask me |
| 11:51.40 | Ice_Strike | you meaning buying all the parts? |
| 11:51.50 | Ice_Strike | like power adapters and such |
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| 12:01.08 | gavimobile | how can I record a playback? |
| 12:01.27 | gavimobile | for example I wanted to use google tts and save that audio file |
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| 12:03.45 | ghost75 | Ice_Strike: yes, if you compare to atom or similar which costs the same |
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| 12:05.22 | kaldemar | gavimobile: core show application Record |
| 12:07.55 | gavimobile | :-( |
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| 12:34.17 | Ice_Strike | ghost75 I have two rasberry pi and I dont really use it. |
| 12:34.19 | Ice_Strike | Two reason |
| 12:34.20 | Ice_Strike | Too Slow |
| 12:34.31 | Ice_Strike | and too messy with a lot of cables |
| 12:38.00 | ghost75 | and 2 raspberry are not faster than 1 atom |
| 12:38.54 | nunne | rasps are excellent devices for trinkets and stuff like this.. since they have lots of gpio outputs/inputs.. but for raw processing power they are not very good at all. |
| 12:41.54 | eirirs | they works for simple home-network tasks like home-pbx, router, ... |
| 12:42.21 | ghost75 | good thing is energy usage |
| 12:42.28 | eirirs | I'm just waiting for them to spit out a PoE-powered Pi which I can just slap into a PoE switch and go |
| 12:43.04 | eirirs | THAT would make a excellent router for my home :) |
| 12:45.17 | ghost75 | router with 1 nic? |
| 12:45.33 | eirirs | usb-nic |
| 12:45.40 | eirirs | or hope for dual-nic on next model with PoE |
| 12:45.48 | ghost75 | i use openwrt |
| 12:45.49 | eirirs | you don't need killer-nic for router |
| 12:46.45 | ghost75 | i wish there were more routers with larger flash |
| 12:46.58 | [TK]D-Fender | Wonder how much IPsec & OpenVPN it could handle... |
| 12:47.25 | ghost75 | and how much l7 |
| 12:53.40 | [TK]D-Fender | Anyone here now know of a good low power/noise/cost x86 appliance I could make a router out of. |
| 12:54.21 | [TK]D-Fender | So far I've got one supermicro system that is using a chipset that DOESN'T suppoedly have the issues that most of theirs do on their NICS |
| 12:54.27 | [TK]D-Fender | And it's about 500$ |
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| 12:54.37 | plundra | eirirs: I very much doubt they will make such a rPi. |
| 12:54.46 | [TK]D-Fender | I need the equivalent to a cheap-ass netbook + 1 more NIC |
| 12:54.47 | plundra | eirirs: If you want that kind of setup now, maybe use a PoE splitter? |
| 12:55.07 | Ice_Strike | eirirs You setup PI as a router? |
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| 12:55.37 | plundra | eirirs: http://global.level1.com/POS-1002/p-2022.htm Or something similar. Doesn't seem very bulky. Use a rubber band around it and your rpi, or whatever :P |
| 12:57.41 | ghost75 | [TK]D-Fender: how about Intel DQ77MK + G1610 |
| 12:59.14 | StaRetji | what would be exten => to allow calls from whitelist.txt and drop all other? |
| 12:59.17 | StaRetji | thx |
| 13:02.34 | StaRetji | I want to block all callerids except from the list I have |
| 13:04.15 | _methods | [TK]D-Fender: you should hit up ##pfsense or the monowall guys |
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| 13:19.23 | ghost75 | anyone has mk-sip.jar for cisco? |
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| 13:20.04 | eirirs | Ice_Strike: No, I currently use a Cisco 1841, but would love something smaller. |
| 13:20.57 | eirirs | plundra: lol, now that was new for me |
| 13:21.19 | ghost75 | i have this ufo http://www.tp-link.com.au/resources/images/products/Large/TL-WR842ND-01.jpg |
| 13:21.45 | eirirs | is it flying? |
| 13:21.54 | ghost75 | not yet, can try lol |
| 13:22.02 | carrar | *Y*A*W*N* |
| 13:22.36 | ghost75 | asterisk,postfix etc are on atom and storage is hp microserver |
| 13:23.47 | carrar | CentOS 6.4 Released!! |
| 13:24.29 | ghost75 | wants wheezy |
| 13:24.52 | OldSmurf | I am trying to setup directmedia over SIP between two asterisks. The caller has been trough a IVR menu before redirecting the call to the other *. Could this be the reason why I can't get remote bridging to work? |
| 13:25.27 | StaRetji | database put whiltelist 12345 1 |
| 13:25.35 | StaRetji | is next one 45678 2 |
| 13:25.36 | StaRetji | ? |
| 13:25.43 | StaRetji | or is always 1 at the end? |
| 13:25.55 | ghost75 | thats a name |
| 13:25.58 | StaRetji | ah |
| 13:26.02 | StaRetji | got it :) |
| 13:26.07 | StaRetji | so, I can put john |
| 13:26.11 | StaRetji | no need to be number |
| 13:26.25 | ghost75 | you can also put obama |
| 13:26.49 | StaRetji | lo9l |
| 13:27.33 | StaRetji | can I leave it empty? |
| 13:27.34 | [TK]D-Fender | <StaRetji> what would be exten => to allow calls from whitelist.txt and drop all other? <- "core show function SHELL" |
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| 13:28.42 | [TK]D-Fender | <ghost75> [TK]D-Fender: how about Intel DQ77MK + G1610 <- tad large & noisy.... |
| 13:28.57 | StaRetji | thx [TK]D-Fender dude, meanwhile I googled and found db example |
| 13:29.03 | ghost75 | noisy not rlly but how about itx |
| 13:29.12 | StaRetji | just wonder if I can just do database put whitelist 123454 |
| 13:29.17 | StaRetji | and so, without names |
| 13:29.38 | [TK]D-Fender | ghost75, ITX / FlexATX. Something I can get in a short-depth 1U |
| 13:29.50 | [TK]D-Fender | StaRetji, You could |
| 13:29.52 | ghost75 | there is thin itx |
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| 13:30.15 | StaRetji | hm, it seems not |
| 13:30.31 | OldSmurf | StaRetji, you can't, you need a value |
| 13:30.32 | StaRetji | can I call them all john? |
| 13:30.42 | ghost75 | http://www.intel.com/p/de_DE/support/highlights/dsktpboards/db-dq77kb |
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| 13:31.51 | coppice | [TK]D-Fender: that might be large, but there's no reason for it to be noisy |
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| 13:32.18 | ghost75 | http://www.gigabyte.com/products/product-page.aspx?pid=4419 cheap and has 2 nic |
| 13:32.22 | [TK]D-Fender | coppice, CPU fan on top.... would have to go LP... That's why Atom is kinda the ideal base for me... |
| 13:32.37 | [TK]D-Fender | coppice, Need for small * server, and another for firewall appliance |
| 13:32.52 | ghost75 | celeron b847 is much faster than atom |
| 13:32.56 | coppice | [TK]D-Fender: even the stock intel fans are pretty quiet these days |
| 13:33.18 | [TK]D-Fender | trick is getting ons suitable fora 1U case as well... |
| 13:33.29 | dpilon | remember..1u would not have much space fir a fan on top |
| 13:33.51 | [TK]D-Fender | exactly |
| 13:33.53 | coppice | oh, 1Us are usually noisy because nobody seems to bother trying to make them quiet |
| 13:34.01 | dpilon | most have it in from...and those fans are normally loud |
| 13:34.06 | dpilon | in front * |
| 13:34.10 | dpilon | too early |
| 13:34.28 | ghost75 | yeah 1u kinda sucks in cooling |
| 13:34.41 | [TK]D-Fender | Hence passive on Atom..... |
| 13:34.43 | dpilon | which is why atom is the better solution |
| 13:35.09 | ghost75 | but dont get atom (i have one) |
| 13:35.17 | [TK]D-Fender | just trying to avoid the NIC issues with Intel 82574L NIC's which supermicro's boards tend to use |
| 13:35.29 | dpilon | i have am old baracuda 1u here...i can here the sucker 3 offices down with door closed |
| 13:35.58 | coppice | there is an assumption in the industry that rack mount == used in another room |
| 13:37.08 | dpilon | now....if you could get liquid cool in a 1u that would be nice :) |
| 13:37.10 | [TK]D-Fender | They have a new one that looks like it might do. A tad pricier than I'd have hoped but worth it I think : http://www.supermicro.com/products/system/1U/5017/SYS-5017A-EF.cfm |
| 13:37.47 | [TK]D-Fender | Suppose I shouldn't think too bad of it @ $500 |
| 13:38.14 | coppice | I bet that one is noisy. look at the tiny fan in the PSU |
| 13:38.43 | dpilon | that is normal size |
| 13:38.44 | [TK]D-Fender | coppice, Reports say it's quite a bit more quiet than it's predecessors |
| 13:39.31 | coppice | dplion: and normal is noisy in a 1U. if you want a quiet 1U you use a large horizontal impeller |
| 13:39.37 | dpilon | :) |
| 13:40.06 | [TK]D-Fender | Is that like a propeller that gets you nowhere? ;) |
| 13:40.54 | coppice | its the merging of impale and propel |
| 13:41.26 | *** join/#asterisk tuxx- (tuxx@2a02:2308::216:3eff:feac:73b6) |
| 13:41.44 | [TK]D-Fender | So not quite "impaler"? |
| 13:41.58 | tuxx- | Hiya, does asterisk have a feature where the /var/spool/voicemail/ folder is deleted when the voicemailbox doesnt exist in the realtime database any more? :) |
| 13:42.00 | eirirs | Vlad the Impaler? |
| 13:42.28 | [TK]D-Fender | eirirs, It does suck :p |
| 13:42.51 | eirirs | what |
| 13:43.05 | [TK]D-Fender | tuxx-, If you're using a realtime storage it should never touch files at all IIRC |
| 13:43.30 | eirirs | realtime storage that never touch files? |
| 13:43.46 | [TK]D-Fender | eirirs, "Vlad the Impaler" -> "Dracula" -> Vampire -> Sucker |
| 13:43.48 | eirirs | can you elaborate? |
| 13:43.52 | tuxx- | hm |
| 13:43.53 | eirirs | haha |
| 13:44.09 | [TK]D-Fender | eirirs, Realtime -DB -> not FS storage |
| 13:44.22 | tuxx- | well, the voicemails are still saved on the disk, not in the DB. |
| 13:44.44 | tuxx- | but it would make it alot easier if they were in the DB |
| 13:44.45 | [TK]D-Fender | tuxx-, Shouldn't be.... |
| 13:45.12 | tuxx- | that enough information for me, thanks! :) |
| 13:45.17 | tuxx- | thats* |
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| 14:14.41 | StaRetji | folks, is this looks ok? |
| 14:14.43 | StaRetji | exten => s,1,GotoIf(${DB_EXISTS(whitelist/${CALLERID(num)})}?:hangupcontext,s,1) |
| 14:14.43 | StaRetji | exten => _.,1,Goto(a2billing,${EXTEN},1) |
| 14:15.26 | StaRetji | line 1 is the only thing I added to normal setup, and I've added hangupcontext |
| 14:15.53 | [TK]D-Fender | StaRetji, _. = HORRIBLE pattern |
| 14:16.57 | *** join/#asterisk studybot_ (~studybot_@gateway/tor-sasl/studybot/x-68286794) |
| 14:17.01 | StaRetji | got it |
| 14:17.47 | StaRetji | besides that, first line, is it okay? will this allow numbers from whitelist to pass to a2billing else, go to hangupcontext? |
| 14:17.58 | [TK]D-Fender | StaRetji, Also your usage seems to say that what you are using isn't a white-list |
| 14:18.20 | [TK]D-Fender | StaRetji, Perhaps you should test it.... |
| 14:19.01 | StaRetji | I am looking at it now, tbh, testing it on live traffic lol |
| 14:19.28 | *** join/#asterisk |Physis| (c813bc0b@gateway/web/freenode/ip.200.19.188.11) |
| 14:19.31 | StaRetji | but seems non of the calls are passing |
| 14:20.56 | StaRetji | seems like it is oposite |
| 14:21.41 | [TK]D-Fender | Well so far... that first pattern is worthless |
| 14:22.23 | kaldemar | StaRetji: also GotoIf syntax is wrong. you're missing $[] |
| 14:23.04 | StaRetji | I was following this one http://muchtall.com/2012/05/23/whitelisting-incoming-calls-on-asterisk/ |
| 14:23.11 | StaRetji | thx kaldemar |
| 14:23.15 | StaRetji | will check it again |
| 14:23.43 | [TK]D-Fender | kaldemar, Not required |
| 14:24.01 | StaRetji | yes, just checked, seems asterisk doesn't complain |
| 14:24.32 | [TK]D-Fender | StaRetji, Your first esten is setill a dead-endpatterns as shown are still worthless for |
| 14:24.50 | StaRetji | :( |
| 14:25.16 | StaRetji | I don't know what to do, I've made whitelist, I thought my nightmare is over lol |
| 14:25.59 | StaRetji | I thought it would be simple as that, chect whitelist, if not inside, hangup, else continue |
| 14:26.35 | [TK]D-Fender | StaRetji, You aren't looking at the STEPS or patterns. Those 2 line have no relation to each other. |
| 14:27.08 | [TK]D-Fender | StaRetji, "s" either jumps to your "hangup" destination.... or just DIES |
| 14:27.14 | [TK]D-Fender | Either way = fail. |
| 14:27.27 | [TK]D-Fender | and the next line is a different pattern entirely that checks nothing at all |
| 14:27.37 | StaRetji | aaah, ,1 ,2 |
| 14:27.40 | StaRetji | let me try now |
| 14:28.05 | [TK]D-Fender | StaRetji, the is no point to having "s" and "." as patterns there. This is severly broken logic |
| 14:28.05 | kaldemar | [TK]D-Fender: i stand corrected. binary return value from a function is enough. |
| 14:28.09 | StaRetji | before I do, calls where passing, so werid |
| 14:28.16 | [TK]D-Fender | kaldemar, Shortcuts++ |
| 14:28.32 | [TK]D-Fender | StaRetji, prioirties are wrong. patterns are wrong. |
| 14:28.47 | [TK]D-Fender | ~book |
| 14:28.47 | 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 |
| 14:28.48 | [TK]D-Fender | ^^^ |
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| 14:33.30 | Greenlight | If I run asterik |
| 14:33.39 | [TK]D-Fender | +s |
| 14:33.54 | [TK]D-Fender | +"rest-of-sentence" |
| 14:34.08 | Greenlight | asterisk virtualied, will applications like say Playback suffer with timing issues, or will remote jitter buffer take care of most of that ? |
| 14:34.31 | [TK]D-Fender | Greenlight, Jitter buffers are on the RECEIVING end |
| 14:34.36 | [TK]D-Fender | Playback is something you SEND |
| 14:34.37 | Greenlight | Yea, exactly |
| 14:34.49 | Greenlight | I said "remote" jitter buffer |
| 14:34.53 | Greenlight | It received what I send ... |
| 14:34.57 | [TK]D-Fender | So if your client has one it might compensate |
| 14:34.57 | Greenlight | *receives |
| 14:35.37 | Greenlight | I'm considering splitting one large box into multiple virtual boxes, and that's my main concern |
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| 14:36.11 | Greenlight | If I understand correctly, most endpoints should have a jitter buffer, and so that should compensate. As long as I'm not needing say conferencing, I'll be mostly okay ? |
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| 14:37.27 | Greenlight | Or would timing issues, skew the rtp timestamps and cause issues anyway ? |
| 14:39.16 | [TK]D-Fender | Don't know all the finer points unfortunately |
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| 14:40.05 | Greenlight | No probs, thanks anyways :) Guess I can build it and see... |
| 14:40.39 | Greenlight | It would just give me a real neat way to seperate out customers, especially smaller ones, who don't merit their own dedicated box |
| 14:41.08 | lorsungcu | Greenlight: i've seen it go both ways, i have ~350 endpoints in a huge vmware cluster, and it works great, and tried a 10 person office in proxmox and it was a disaster |
| 14:41.15 | lorsungcu | so yeah, build it |
| 14:41.16 | lorsungcu | :p |
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| 14:41.56 | Greenlight | Fingers crossed then! It's ESXi that I'll be running, so nice and low level virtulisation |
| 14:42.26 | lorsungcu | yeah also had somewhat good luck with that. |
| 14:42.37 | Greenlight | That's promising then |
| 14:43.12 | Greenlight | Most calls will be p2p bridged once connected, it's just for any IVR type stuff, or announcements that I feared |
| 14:44.09 | lorsungcu | yeah, and that's where i did see issues when I had them, |
| 14:45.19 | Greenlight | It'll be an learning experience if nothing else I guess |
| 14:46.37 | BludSuckingFiend | I've got an IVR and a Voicemail box on an ESX cluster... it works relatively well, not a lot of load though |
| 14:47.11 | BludSuckingFiend | I am very leary about using VMWare for anything timing-sensitive though. |
| 14:48.30 | Greenlight | Yea, that was my fear too. We generally try to always under provision the host box, and if it seems to work well I certianly dont mind spending £5k or so and getting another host dedicated to just running the Asterisk guests |
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| 15:16.21 | [Outcast] | does anyone know of any solutions to automate mos scoring |
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| 16:05.25 | _zoom_ | fellas, what does it mean "span 1 got hangup request, cause 31" when place outgoing call in pri |
| 16:09.37 | igcewieling | _zoom_: it means "call ended normally" |
| 16:09.54 | igcewieling | 31 is sometimes used instead of 16 for reasons I never understood |
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| 16:21.27 | Salman | is there any tutorial on extensions.conf? |
| 16:25.39 | igcewieling | Salman: too complicated for a tutorial, have you read The Book? |
| 16:26.04 | Salman | yes but thats also complicated |
| 16:26.16 | Salman | i was looking for with an example where all dial plan can be explained |
| 16:27.25 | [TK]D-Fender | There is no such this as "all dialplan". The basic concepts are all out there, and tons of specific little samples |
| 16:27.38 | [TK]D-Fender | Salman, What point in particular are you having trouble with? |
| 16:28.53 | Salman | actually I just trying that all extensions should dial each other without specifying in externsions.conf |
| 16:30.56 | [TK]D-Fender | Salman, You can't |
| 16:31.00 | [TK]D-Fender | that's the poitn of extensions.conf |
| 16:31.16 | [TK]D-Fender | to define what happens when your devices dial |
| 16:31.21 | [TK]D-Fender | There is no "just assume this number relates to something else" |
| 16:31.36 | [TK]D-Fender | You make your patterns and tell it what actions to take |
| 16:33.17 | BludSuckingFiend | unless you're using users.conf -_- |
| 16:33.29 | BludSuckingFiend | Then it has an annoying habit of breaking convention |
| 16:33.34 | [TK]D-Fender | not entirely.. and EWWWW |
| 16:34.10 | [TK]D-Fender | Salman, the dialplan is 95% of configuring * |
| 16:34.18 | BludSuckingFiend | I am surprised they chose to do that. The document that explains best practices extolls the virtues of separating devices from the extension #s |
| 16:34.35 | [TK]D-Fender | Salman, A phone is just a phone... no one model is really that much different than every other |
| 16:34.35 | BludSuckingFiend | and then if you use users.conf it automatically associates the device to a # |
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| 16:35.22 | [TK]D-Fender | Salman, The dialplan defines what happens when * gets a call. |
| 16:35.32 | [TK]D-Fender | Salman, So that is where all the real work of setting up * is |
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| 16:36.16 | _zoom_ | igcewieling, call never goes out |
| 16:37.58 | Salman | sorry was away |
| 16:39.13 | Salman | i understand all that |
| 16:40.25 | [TK]D-Fender | Salman, So go start a nice new dialplan, make a context for your phones and a simple exten to dial each phone to start |
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| 16:42.30 | [TK]D-Fender | eek |
| 16:43.12 | Salman | what about freepbx? it makes itself or we have to define manually.. |
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| 16:43.40 | Salman | actually that thing is very complicated. I know dial plan is key of telephony but I'm not too much handy with asterisk |
| 16:44.33 | [TK]D-Fender | Salman, What do you actually want to do? |
| 16:45.37 | Salman | i just want to communicate two extension dial each other and two peers |
| 16:45.55 | [TK]D-Fender | Salman, basic * is tiny for that... could be a handful of lines |
| 16:46.26 | Salman | Visual Dial Plan could be helpful? |
| 16:47.11 | [TK]D-Fender | No |
| 16:47.17 | [TK]D-Fender | waste of time. |
| 16:47.47 | Salman | ok refer me to a guide |
| 16:47.53 | Salman | let me learn |
| 16:47.59 | [TK]D-Fender | If you're determined to fight for a shortcut, then go the full-GUI route. but it's overkill and will pretty much suck up a whoe computer |
| 16:48.07 | [TK]D-Fender | the book is the place to start. |
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| 16:49.12 | Salman | right |
| 16:49.33 | [TK]D-Fender | 5 phones = 6 lines of extensions.conf |
| 16:49.42 | [TK]D-Fender | Want voicemail for each? 11. |
| 16:50.09 | [TK]D-Fender | Simple dial-out to a provider? Probably .... ONE more line |
| 16:50.19 | [TK]D-Fender | We haven't even broken 20 here |
| 16:50.33 | [TK]D-Fender | Then if you want to get fancier... well... build as you go. |
| 16:50.42 | [TK]D-Fender | Inbound handling? Maybe another couple of lines |
| 16:50.52 | [TK]D-Fender | This really isn't Raw-Cat Sigh Hence |
| 16:51.36 | Salman | heh |
| 16:51.39 | Salman | you are good |
| 16:51.51 | BludSuckingFiend | My dialout to our provider has about 80 lines |
| 16:52.02 | BludSuckingFiend | but that's just because of the callerID stuff .. |
| 16:52.21 | [TK]D-Fender | BludSuckingFiend, Bet I could trim that for you :) |
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| 16:53.03 | Salman | suppose we have 2 phones and one a2z trunk for calling outside. Enabling voicemail feature |
| 16:53.09 | MrMeek | hi all |
| 16:53.21 | navaismo | o/ |
| 16:53.41 | BludSuckingFiend | Basically our provider requires callerID set to a 10 digit DID we own. Most of our employees have their own DIDs... so I've got to set their outbound callerID to the 10 digit on dialout |
| 16:53.43 | [TK]D-Fender | Salman, < 20 |
| 16:53.52 | [TK]D-Fender | Salman, maybe < 10 |
| 16:53.59 | BludSuckingFiend | which is different from their normal internal CallerID between stations. I've thought about using a DB |
| 16:54.06 | [TK]D-Fender | Salman, Dial() <- calls devices (your phones, your provider, etc) |
| 16:54.10 | [TK]D-Fender | Salman, All basic stuff |
| 16:54.34 | [TK]D-Fender | BludSuckingFiend, SetVar <- for each of your sip peers |
| 16:54.50 | BludSuckingFiend | aah, actually a really good idea |
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| 16:55.59 | MrMeek | need some advice on festival... docs say that if intkeys arg is provided the key press is 'returned', but does not say how/where it is returned. I can't seem to get it to do anything except dial the extension of the DTMF pressed ... |
| 16:56.23 | BludSuckingFiend | thanks for that idea [TK]D-Fender |
| 16:56.24 | MrMeek | hoping to capture it like w/ cepstral: swift(tts text,5000,1) |
| 16:56.53 | MrMeek | GotoIf( ${SWIFT_DTMF}=1 ? context) |
| 16:56.53 | MrMeek | etc |
| 16:57.14 | MrMeek | would rather use cepstral but not avail on this server atm :/ |
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| 17:00.02 | Salman | Hello [TK]D-Fender |
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| 17:01.31 | navaismo | MrMeek, have you considered to create the audio and then use within background |
| 17:01.52 | MrMeek | I have- it seems straight forward to call out to system() and cache the audio |
| 17:02.04 | MrMeek | Do you think that may be the best solution considering the apparent constraints of festival? |
| 17:02.59 | MrMeek | ofo app_festival, i should say :) |
| 17:07.39 | MrMeek | hrm actually i don't think you can capture the DTMF into a variable from Background() app either? |
| 17:08.02 | MrMeek | Looks like the functionality i get from Swift() is closest to what is provided by Read() |
| 17:08.06 | MrMeek | thoughts? |
| 17:14.41 | *** join/#asterisk CunningPike (~CunningPi@d206-116-73-41.bchsia.telus.net) |
| 17:19.33 | FLeiXiuS | Is it possible to have a SIP client setup to use both TLS-SIP and unsecured SIP |
| 17:20.34 | *** part/#asterisk kontinuity (~kontinuit@122.167.100.176) |
| 17:22.03 | *** join/#asterisk kontinuity (~kontinuit@122.167.100.176) |
| 17:22.27 | *** part/#asterisk kontinuity (~kontinuit@122.167.100.176) |
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| 17:27.47 | igcewieling | FLeiXiuS: check the docs for your SIP client. |
| 17:29.13 | MrMeek | navaismo, thx |
| 17:29.32 | MrMeek | shame on me for not immediately relizing the obvious :P shell out to text2wav and playback with Read() working fine |
| 17:29.56 | *** join/#asterisk drdru (~drdru@76.77.182.56) |
| 17:30.00 | navaismo | good |
| 17:30.31 | drdru | is there any other way than using ChanSpy to inject audio into a channel? |
| 17:30.51 | igcewieling | FLeiXiuS: You could set up 2 accounts on your SIP phone and set up two peers/friends in asterisk, one for TLS, one for no-TLS. |
| 17:30.56 | drdru | I'm planning to do some custom module development, but I'd like to get as close as possible before I start |
| 17:31.17 | drdru | I want to capture audio from each side of the call separately (so I'm using Monitor) |
| 17:31.20 | igcewieling | drdru: check with #asterisk-dev if you are writing asterisk modules |
| 17:31.49 | drdru | and I want to inject audio into each side of the call independently, but I don't want to have an extra extension involved, so I don't want to use ChanSpy |
| 17:32.41 | FLeiXiuS | igcewieling, Alright, thats what I figured needed to be done. I was trying to avoid having to create 2 sip accounts. |
| 17:33.34 | igcewieling | if you are talking about running chanspy, it doesn't sound very "module-like" |
| 17:33.57 | igcewieling | drdru: you're not doing something silly like asking how to write a FreePBX module on #asterisk, are you? |
| 17:34.42 | drdru | I'm not using FreePBX |
| 17:34.52 | drdru | and truth be told, I'm an asterisk n00b |
| 17:35.09 | drdru | just looking for some pointers on how to setup my architecture before I have to write my custom code |
| 17:35.19 | drdru | not sure what the difference is between module and application, etc |
| 17:35.43 | drdru | I setup a very simple dialplan which allows me to monitor both sides of a call to 2 separate wav files |
| 17:35.57 | drdru | so that already gets me 50% of what I want |
| 17:35.59 | *** join/#asterisk brookshi1e (mbrooks@hijacked.us) |
| 17:36.10 | drdru | now I want to inject audio back into each side independently |
| 17:36.24 | drdru | and I know ChanSpy can do it, but I don't want to have a third extension involved in the call |
| 17:37.05 | igcewieling | drdru: are you fluent in C? |
| 17:37.48 | drdru | yeah |
| 17:38.47 | igcewieling | Then there is hope. 8-) You should be asking on #asterisk-dev channel or asterisk-dev mailing list. |
| 17:39.14 | *** join/#asterisk SuperLag (~akulbe@unaffiliated/superlag) |
| 17:39.37 | drdru | ok, I'm asking there in parallel, but haven't heard any response or activity thus far |
| 17:40.41 | drdru | igcewieling: what about injecting audio into an existing call? do I have to use ChanSpy or is there another way? |
| 17:41.16 | drdru | and is it possible to inject a file, rather than what is coming from the whispering extension? |
| 17:41.19 | igcewieling | drdru: I'm sure there is, but since this is a dev question I don't know for sure. |
| 17:41.43 | drdru | it's not dev yet - I haven't edited any C files yet |
| 17:41.58 | drdru | I'm trying to get as close to my goal as possible before I start coding |
| 17:42.03 | igcewieling | drdru: you are asking about asterisk's internal API (can a module inject audio) |
| 17:42.15 | drdru | ChanSpy() can inject audio |
| 17:42.24 | drdru | I'm asking if any other functions can do it |
| 17:42.42 | igcewieling | drdru: right, but if you are writing an asterisk module in C you will want to use the Asterisk API, not dialplan applications |
| 17:42.46 | drdru | ChanSpy() has a whisper mode |
| 17:43.25 | igcewieling | drdru: then go read the code for chanspy and see |
| 17:43.36 | navaismo | not sure if bridge can do that |
| 17:43.50 | drdru | ok |
| 17:46.25 | drdru | I'm having trouble finding bridge() examples |
| 17:48.29 | igcewieling | drdru: try audiohook or something like that |
| 17:48.50 | igcewieling | remember all this changes between major asterisk versions |
| 17:48.56 | drdru | yes, that's what file told me in #asterisk-dev |
| 17:49.07 | drmessano | Bridbe |
| 17:49.10 | drmessano | grrr |
| 17:49.37 | drmessano | I found a few examples of Bridge() with a quick google search |
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| 18:29.26 | MrMeek | Is there any advantage of using Playback(silence/1) over Wait(1) ? |
| 18:29.42 | MrMeek | the previous seems to be the preferred but i can't see any practical difference |
| 18:29.53 | leifmadsen | it answers the line |
| 18:30.02 | leifmadsen | Playback() |
| 18:30.15 | MrMeek | ah, so if Answer() has already been called, there actually is no difference ? |
| 18:30.16 | leifmadsen | so you end up not getting the little bit of audio cut off when you play a prompt |
| 18:31.08 | leifmadsen | I just find usage of Playback(silence/1) a better method if you're going to call anything else that might call audio |
| 18:31.20 | leifmadsen | if not, there is no reason to not use Wait(1) |
| 18:31.42 | MrMeek | awesome ty for advice |
| 18:31.58 | leifmadsen | np, it's likely my fault you're seeing the Playback(silence/1) method :) |
| 18:31.59 | MrMeek | oh and, leif of asteriskdocs fame? :) |
| 18:32.02 | *** part/#asterisk glaz (strke@hiro.glaciuz.com) |
| 18:32.07 | leifmadsen | heh, fame? sure :) |
| 18:32.18 | MrMeek | lol, amonst us dial plan hackers anyway :) |
| 18:32.26 | MrMeek | appreciate all your effort on that book, especially the latest editions |
| 18:32.26 | MrMeek | <3 |
| 18:33.31 | leifmadsen | np :) |
| 18:34.34 | igcewieling | I bet he even has groipues. |
| 18:34.40 | igcewieling | groupies, even |
| 18:34.40 | leifmadsen | so many groupies |
| 18:35.08 | MrMeek | the ladies love asterisk, lol ;) |
| 18:36.05 | leifmadsen | wait wat? |
| 18:36.06 | leifmadsen | :) |
| 18:36.21 | leifmadsen | my wife just calls me a nerd |
| 18:36.31 | igcewieling | leifmadsen: Asterisk Rock Star |
| 18:36.42 | leifmadsen | it's all about the sun glasses |
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| 18:58.28 | Weezey | is there an easy way to playback music and a message on hold or mix the two? I have 5 messages I want to play in random order with music as the background |
| 18:59.02 | [TK]D-Fender | Weezey, use a streaming source you an do this to. There is nothing in * to allow it |
| 18:59.32 | Weezey | [TK]D-Fender: thanks, I'll just have to mix it myself then. |
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| 19:54.25 | joobie | hey guys, running 1.4 and having issues where users are complaining of poor quality calls (muffled). They have polycom 321 phones and ive switched them over to ulaw to asterisk and then DAHDI ulaw going out |
| 19:55.09 | joobie | what should i look at to try improve this? I was thinking the actual phone itself, looking for something with a better speaker |
| 19:55.12 | joobie | but what do u guys think? |
| 19:55.38 | Qwell | ~upgrade asterisk |
| 19:55.38 | 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. |
| 19:58.38 | joobie | Qwell, will that make a difference to sip call quality? |
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| 19:59.29 | igcewieling | joobie: are you at least running the latest 1.4.x? |
| 20:01.50 | joobie | igcewieling, yes |
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| 20:08.25 | leifmadsen | a muffled sounds seems like an issue with a handset or someone not using the phone propertly |
| 20:08.28 | leifmadsen | properly* |
| 20:08.36 | leifmadsen | I don't see any way asterisk could cause that issue |
| 20:09.43 | [TK]D-Fender | joobie, So calling direct from 1 phone to another sounds bad? |
| 20:10.30 | gavimobile | hey folks! |
| 20:10.37 | lorsungcu | sup yo |
| 20:11.08 | gavimobile | iight homie |
| 20:11.10 | gavimobile | lol |
| 20:11.22 | leifmadsen | homie don't play that |
| 20:11.26 | gavimobile | is it possible to record a playback to a file? |
| 20:11.35 | leifmadsen | huh? |
| 20:11.50 | gavimobile | im using googletts, so I want to Record the output |
| 20:11.56 | WIMPy | Hi again |
| 20:11.58 | leifmadsen | just use Monitor() then |
| 20:12.15 | gavimobile | let me read up on Monitor |
| 20:12.20 | navaismo | gavimobile, the output of that script is stored in the /tmp directory as sln format |
| 20:12.29 | leifmadsen | navaismo: that's an even better method :) |
| 20:13.13 | gavimobile | leifmadsen: navaismo: sweet! that's exactly what I need then |
| 20:13.21 | gavimobile | I'll play with it tommorow time for me to bed |
| 20:13.26 | gavimobile | to go to bed |
| 20:13.31 | gavimobile | gnight folks! |
| 20:13.35 | navaismo | well the name of the file is a something like a hash so... |
| 20:13.40 | gavimobile | Hangup() |
| 20:13.51 | gavimobile | navaismo: ill clear it first before I run the app |
| 20:14.01 | gavimobile | thall narrow down the results |
| 20:14.09 | *** join/#asterisk sipman (~slane@108.77.51.89) |
| 20:14.11 | joobie | [TK]D-Fender, ill test, but i suspect no |
| 20:14.22 | joobie | i have to actually run to work now, sorry will jump on again later |
| 20:14.24 | joobie | thanks |
| 20:14.25 | gavimobile | gnight folks |
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| 20:56.39 | linlin | so it looks like asterisk@home was replaced by trixbox...which now appears to be dead. any other free turn-key options out there for asterisk like AAH used to be ? |
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| 21:00.05 | navaismo | freepbx-distro or asterisknow+freepbx |
| 21:01.03 | WIMPy | There are quite a few to choose from, but I doubt anyone is going to recommenda anything. |
| 21:01.07 | WIMPy | -a |
| 21:03.54 | linlin | it looks like asterisknow come with freepbx |
| 21:04.15 | leifmadsen | people seem to use PBX In A Flash, but the community that surrounds it is vitriol |
| 21:04.21 | linlin | just looking for something simple i can use at home, maybe drop my comcast voip eventuially |
| 21:04.27 | leifmadsen | ya, asterisknow uses freepbx |
| 21:05.22 | Qwell | I hear the maintainer of AsteriskNOW is awesome. |
| 21:05.34 | WIMPy | If you want something simple, I guess the best place to look would be the antiques store. |
| 21:06.08 | linlin | what |
| 21:06.46 | *** join/#asterisk gusto (~gusto@ppp-93-104-85-119.dynamic.mnet-online.de) |
| 21:06.48 | WIMPy | Simple doesn't exist eany more. |
| 21:11.13 | linlin | oh...too bad |
| 21:11.33 | linlin | i guess ill give asterisknow a shot... |
| 21:11.42 | leifmadsen | Qwell: I heard from tdotzilla that guy is an asshole |
| 21:12.08 | Qwell | He totally is. |
| 21:13.54 | linlin | i have a feeling Qwell = asterisknow maintainer? |
| 21:14.16 | Qwell | Whaaaaaaaat? No way! |
| 21:14.28 | linlin | cant slip anything past me |
| 21:15.29 | linlin | think last time i played with asterisk was around 2006....holy hell have these interface cards gotten cheaper |
| 21:17.16 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:19.54 | leifmadsen | the SIP ATA's are creazy cheap now too |
| 21:24.25 | linlin | yeah i see that, i remember buying sipura boxes that were pretty expensive |
| 21:25.03 | linlin | iaxy still expensive though |
| 21:26.55 | leifmadsen | no one uses an iaxy any more |
| 21:27.12 | Qwell | We haven't sold them in years. |
| 21:31.02 | malcolmd | mattf uses one |
| 21:31.11 | malcolmd | the original model, not the fancy-pants 101 model |
| 21:31.39 | Qwell | wonders if he would still get in trouble if the word "super" came out right now. |
| 21:31.40 | linlin | i thought they were nice |
| 21:34.44 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 21:34.53 | file | Qwell, the paperwork has begun! |
| 21:35.07 | Qwell | que? |
| 21:35.29 | file | super paperwork. |
| 21:35.32 | Qwell | EEP! |
| 21:35.51 | Qwell | I don't know what that means. |
| 21:35.58 | file | hi |
| 21:36.02 | Qwell | nope |
| 21:37.34 | linlin | i still dont get it |
| 21:37.39 | Qwell | exactly |
| 21:39.18 | linlin | extra special digium inside joke i assume |
| 21:39.29 | Qwell | just file being file |
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| 22:59.38 | *** mode/#asterisk [+o pabelanger] by ChanServ |
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| 23:30.46 | MrMeek | i pop in from time to time but i'm still quite new here |
| 23:30.55 | MrMeek | i wonder if 'mattf' is who i think it may be |
| 23:45.53 | *** join/#asterisk saint_ (~saint@68.38.56.184) |
| 23:56.49 | saint_ | hi all |
| 23:57.05 | WIMPy | lo you |
| 23:57.09 | saint_ | damn. fire call. gotta go. |
| 23:57.34 | WIMPy | wonders if he should script that. |
| 23:58.26 | igcewieling | MrMeek: PRI wizard extraordinaire? |
| 23:59.03 | igcewieling | WIMPy: April 1 is coming up, maybe an RFC for Reality Scripting Language? |
