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| 03:52.14 | BoRiS | Hi guys |
| 03:53.21 | BoRiS | When building asterisk 11 from svn, It seems to always failed when building libasteriskssl with multiple definition of `CRYPTO_set_locking_callback', and multiple definition of `ERR_free_strings' |
| 03:53.32 | BoRiS | This doesnt happen with asterisk 10. |
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| 04:12.18 | igcewieling | file a bug report on Jira |
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| 07:58.35 | bulkorok | hi |
| 08:09.10 | atan | Howdy |
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| 08:31.10 | pietro | hello |
| 08:39.32 | ectospasm | ~ask |
| 08:39.32 | 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. |
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| 08:56.33 | _zoom_ | hello |
| 08:58.50 | ectospasm | ~ask |
| 08:58.50 | 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. |
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| 09:28.59 | linocisco | hi all, users.conf vs sip.conf ? |
| 09:30.43 | wdoekes | sip.conf |
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| 09:34.09 | kaldemar | ~users.conf |
| 09:34.09 | infobot | [~users.conf] users.conf is a flaming pile of sh1t that takes the fine control of several perfectly usable asterisk config files and reduces them to the lowest common denominator and makes your system behave like a "toaster grade" PBX system. |
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| 09:36.51 | bulkorok | rofl... |
| 09:41.24 | linocisco | bulkorok, ~rofl |
| 09:41.32 | linocisco | ~rofl |
| 09:41.32 | infobot | well, rofl is rolling on the floor laughing |
| 09:42.25 | linocisco | kaldemar, so if GUI put entries in users.conf, should I not touch sip.conf anymore? |
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| 09:46.04 | linocisco | what is the meaning of "exten = o,1,"? |
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| 09:49.13 | bombev | is there any Asterisk application. I want to be able somehow to check given number if has some digits and redirect the call to different trunk |
| 09:50.31 | _zoom_ | am not able to detect sipgate dtmf with asterisk 1.6 any idea? |
| 09:50.59 | ectospasm | Asterisk 1.6.* is EOL and will not be supported here. |
| 09:51.02 | ectospasm | _zoom_: |
| 09:51.03 | bombev | can i use ExecIf |
| 09:52.29 | linocisco | ectospasm, lol. i m with asterisk 1.4 |
| 09:52.40 | linocisco | ectospasm, lol. i m with asterisk 1.4. i have no choice |
| 09:53.33 | ectospasm | you always have a choice. |
| 09:57.04 | kaldemar | linocisco: if a GUI puts conflicting entries in users.conf and sip.conf, you should not touch sip.conf. |
| 09:57.39 | kaldemar | linocisco: https://wiki.asterisk.org/wiki/display/AST/Handling+Special+Extensions |
| 09:57.52 | _zoom_ | ectospasm: have u tested it with head version of asterisk? |
| 09:58.16 | kaldemar | bombev: ExecIf, GotoIf, IF... |
| 09:58.29 | ectospasm | You mean trunk? No, and you shouldn't either unless you're testing Asterisk. |
| 09:58.41 | bombev | kaldemar ok |
| 09:58.50 | ectospasm | _zoom_: ^ |
| 10:02.36 | aruntomar | which asterisks web frontend is mostly used? and what to consider while choosing one? |
| 10:03.56 | linocisco | kaldemar, if GUI defines users in users.conf , should I only touch users.conf and sip.conf should be untouched? what about extensions.conf? |
| 10:05.08 | kaldemar | linocisco: you should not touch anything if a GUI writes your configs, unless you know precisely what you're doing. |
| 10:05.31 | linocisco | kaldemar, but with GUI, dialplan are not ok |
| 10:10.05 | kaldemar | linocisco: i don't understand what you mean by that. |
| 10:10.53 | linocisco | kaldemar, it is weird device. called QNAP. asterisk -r doesn't show |
| 10:11.43 | ectospasm | linocisco: is this a NAS, or some other device? |
| 10:12.57 | linocisco | ectospasm, NAS with many optional features like asterisk, glpi can be installed |
| 10:13.05 | linocisco | ectospasm, it is busy box based linux box |
| 10:13.26 | ectospasm | linocisco: there's a QNAP TS-559 Pro+ sitting next to me (-; |
| 10:13.30 | ectospasm | But I don't use Asterisk on it. |
| 10:13.32 | aruntomar | can i save my asterisk call recordings directly in ogg format rather than in wav and then converting them to ogg? |
| 10:14.26 | linocisco | ectospasm, to reduce power of a normal PC, management want me to use existing device QNAP |
| 10:15.24 | ectospasm | linocisco: why not have a VM host/hypervisor and run Asterisk on a guest? It's what I do. |
| 10:16.25 | linocisco | ectospasm, but any way i need a pc which should be powered all night a day. |
| 10:16.49 | ectospasm | what's wrong with keeping a PC in that state? I do that at work, home... |
| 10:17.21 | ectospasm | Unless a PC is pegging the CPU all the time, they don't use that much power. |
| 10:17.36 | ectospasm | HVAC systems typically consume far more power than anything else. |
| 10:17.52 | ectospasm | ...unless you're running a datacenter |
| 10:18.34 | linocisco | ectospasm, thannsk bro |
| 10:18.45 | linocisco | ectospasm, i need to go to catch ferry |
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| 10:25.03 | bombev | guys |
| 10:25.15 | bombev | No such label 'BG' in extension 's' in context 'macro-fastvat' |
| 10:25.23 | bombev | how can I assign label in extension |
| 10:25.49 | kaldemar | exten => exten,n(labelgoeshere),Noop() |
| 10:26.44 | bombev | perfect |
| 10:26.47 | bombev | thanks kaldemar |
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| 11:47.21 | bombev | kaldemar if I have 5 condition which application is better |
| 11:47.36 | bombev | conditions |
| 11:47.37 | bombev | ExecIf, GotoIf, IF... |
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| 12:11.51 | f843d0 | Hi all, any hint on how limiting the number of occupied lines on phone calls directed to an internal extension? How would you handle this? |
| 12:12.21 | *** join/#asterisk [TK]D-Fender (~Joe@216-191-106-165.dedicated.allstream.net) |
| 12:12.26 | WIMPy | Taie a look at the GROUP* functions. |
| 12:13.00 | f843d0 | WIMPy: I will, thanks a lot for the suggestion |
| 12:16.34 | f843d0 | Once, that was achievable through the call-limit parameter; isn't that gonna work anymore? |
| 12:32.44 | kaldemar | f843d0: call-limit has been deprecated for ages. |
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| 12:34.09 | f843d0 | kaldemar: I see; busylevel as well? |
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| 12:37.09 | kaldemar | f843d0: no. |
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| 12:42.34 | bombev | [TK]D-Fender Hi |
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| 12:43.58 | bombev | <PROTECTED> |
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| 12:44.26 | bombev | GotoIf does not work the way I want |
| 12:44.47 | bombev | May be i should use ExecIf |
| 12:45.01 | bombev | what do you think |
| 12:45.34 | bombev | or there is other asterisk application |
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| 12:48.12 | [TK]D-Fender | bombev, Gotoif works fine |
| 12:49.03 | [TK]D-Fender | bombev, Execif is only good for minimal 1-2 line conditionals where you hve a common "continue" point (in-line). After that it get real ugly |
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| 12:50.30 | bombev | but [TK]D-Fender but I want to check 5 numbers with GotoIf |
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| 12:51.20 | [TK]D-Fender | bombev, Show us what you've done |
| 12:51.23 | bombev | http://pastebin.com/34F8Y6z0 |
| 12:51.48 | *** join/#asterisk CheeseMouse (~cheddar@cheesy.simonics.com) |
| 12:52.40 | bombev | and the call always goes thru first GotoIf s,n,GotoIf($["${MACRO_EXTEN:0:5}" = "00359"]?BG:FREE) |
| 12:53.19 | bombev | and when I dial 0048 number again it goes thru first gotoif |
| 12:53.53 | WIMPy | Yes. It can never reach the 2nd GotoIF. You've got two targets on the 1st one. |
| 12:54.04 | [TK]D-Fender | bombev, pyou shouldn't be usnig the ELSE condition <- |
| 12:54.28 | CheeseMouse | I'm looking for numbers on codec processing on a CPU (not dedicated transcoding hardware) - specifically, increase in CPU usage over SLIN or G.711, for example |
| 12:54.37 | bombev | so if I dont have else condition it will goes to the 2nd gotoif |
| 12:54.38 | bombev | correct? |
| 12:54.47 | [TK]D-Fender | bombev, yes |
| 12:54.54 | bombev | perfect I will try now |
| 12:54.55 | bombev | thanks |
| 12:54.57 | bombev | guys |
| 12:55.08 | CheeseMouse | I'm not able to find this kind of data. core show translation tells me how much time it will take to transcode, but not how much additional CPU |
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| 12:55.49 | [TK]D-Fender | CheeseMouse, there is no "CPU%" number to get for this |
| 12:56.00 | WIMPy | CheeseMouse: What "additional CPU"? |
| 12:56.25 | [TK]D-Fender | CheeseMouse, the time is the relative CPU load to each other, but is not relative to a load % on your processor(s) |
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| 12:57.01 | CheeseMouse | Understood |
| 12:57.27 | [TK]D-Fender | bombev, You have also done secveral other serious dialplan errors |
| 12:57.39 | CheeseMouse | If I go out and read about codecs I see vague phrases like g.729 is "heavy" on CPU and gsm is "lighter" than that... but no one seems to be able to really quantify it |
| 12:57.49 | [TK]D-Fender | bombev, You only put a label name on ONE priority |
| 12:57.57 | bombev | [TK]D-Fender for example |
| 12:58.07 | [TK]D-Fender | bombev, You don't put "BG" on multiple lines. it is the TARGET of your GotoIf |
| 12:58.24 | WIMPy | CheeseMouse: It's always about transcoding. No transcoding, no CPU. |
| 12:58.45 | bombev | so I must have it only on the first line |
| 12:59.05 | leifmadsen | CheeseMouse: core show translation |
| 12:59.33 | leifmadsen | that'll give you a relative idea which is more or less heavy on transcoding |
| 12:59.33 | CheeseMouse | WIMPy, right. So, what is the load put on the CPU to transcode from G.711u to GSM-FR? |
| 12:59.47 | leifmadsen | that's not really the correct question |
| 12:59.56 | leifmadsen | the load is relative to the CPU |
| 13:00.11 | [TK]D-Fender | bombev, Yes, and at the end of the priorities related to that "block" you want to process you need to make sure you don't continue into ANOTHER block |
| 13:00.17 | WIMPy | That's what 'core show translation' tells you. |
| 13:00.24 | CheeseMouse | I'm unsure how to translate # of ms into CPU loading |
| 13:00.32 | leifmadsen | by testing :) |
| 13:00.34 | WIMPy | To be exact it telly you how long YOUR CPU takes to do it. |
| 13:00.35 | [TK]D-Fender | bombev, At the end of your "BG" .... you just continue on to your PL lines. |
| 13:01.32 | bombev | [TK]D-Fender is it better this way: http://pastebin.com/TkyJXf86 |
| 13:01.59 | CheeseMouse | ok, now that makes sense |
| 13:03.46 | [TK]D-Fender | bombev, Still has a roll to the next "section" |
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| 13:04.35 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 13:04.48 | bombev | [TK]D-Fender hm.. how to avoid this? |
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| 13:07.04 | [TK]D-Fender | bombev, Stop continuing on. |
| 13:09.27 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
| 13:09.44 | bombev | ok but where should I put PL label? |
| 13:09.57 | bombev | could you show me what do you mean |
| 13:10.50 | [TK]D-Fender | bombev, http://pastebin.com/TkyJXf86 <--- |
| 13:11.26 | [TK]D-Fender | bombev, When line 9 there fails for whatever reason ... it will continue on to the NEXT line there which "logically" speaking is for "PL" |
| 13:12.10 | jmetro | so add a hangup() or other such routing , whatever you want. |
| 13:12.32 | bombev | [TK]D-Fender ok the question is where to put this PL |
| 13:12.34 | bombev | label |
| 13:12.40 | [TK]D-Fender | bombev, that isn't the problem. |
| 13:12.57 | [TK]D-Fender | bombev, its that at the end of the first "block" ... YOU AREN'T ENDING THE CALL |
| 13:13.58 | [TK]D-Fender | bombev, Stop letting it just go to that next line. End the call. Go somewhere else. DO SOMETHING |
| 13:15.37 | bombev | [TK]D-Fender is it better now: http://pastebin.com/8F6pL78L |
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| 13:19.36 | hilio | gday folks |
| 13:20.14 | hilio | anyone know of a channel on freenode that specializes in media streaming? i'm looking for people into Wowza |
| 13:20.47 | hexanol | I have a question about the LOCK dialplan function |
| 13:20.50 | *** join/#asterisk igcewieling (~igcewieli@ip98-183-25-31.pn.at.cox.net) |
| 13:20.56 | [TK]D-Fender | bombev, And what does that do? |
| 13:21.20 | hexanol | I have this simple dialplan |
| 13:21.26 | hexanol | exten = 130,1,Answer() |
| 13:21.27 | hexanol | same = n,Set(LOCKED=${LOCK(foobar)}) |
| 13:21.27 | hexanol | same = n,SayDigits(${LOCKED}) |
| 13:21.31 | hexanol | same = n,GotoIf(${LOCKED}?:end) |
| 13:21.32 | hexanol | same = n,Wait(15) |
| 13:21.32 | hexanol | same = n(end),Hangup() |
| 13:21.40 | *** join/#asterisk leedm777 (~leedm777@nat/digium/x-qkmyhwejxtovrxdo) |
| 13:21.47 | jmetro | hexanol: :( |
| 13:21.51 | hexanol | from what I understand of the LOCK function documentation, I would expect |
| 13:22.06 | hexanol | that if I call the 130 extension from two different phone |
| 13:22.11 | hexanol | one would say 1 and wait 15 seconds |
| 13:22.28 | bombev | [TK]D-Fender well goes to given ${DIALSTATUS} |
| 13:22.29 | hexanol | the other would wait the other channel to close and also say 1 |
| 13:22.45 | hexanol | problem is, the second channel only waits 3 seconds and then says 0 |
| 13:22.47 | bombev | if it is BUSY goes to s-BUSY,1,Noop(Dial failed due to trunk reporting BUSY - giving up) |
| 13:22.57 | bombev | s-BUSY,n,Playback(custom1/s-busy) |
| 13:23.37 | [TK]D-Fender | bombev, And what is there for it to land on? |
| 13:24.44 | hexanol | anyone knows if it's a bug in the LOCK function or in the LOCK function documentation ? |
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| 13:28.02 | leifmadsen | hexanol: could be a bug in the function... it should probably only try for 3 seconds then return 1 |
| 13:28.21 | leifmadsen | hexanol: likely you'll need to either provide a patch, or file an issue and wait a while |
| 13:28.25 | leifmadsen | or hire a consultant to fix it |
| 13:30.04 | hexanol | leifmadsen: I would expect it to wait until the lock is released (in the case the channel "owns" no other lock), and then return 1 |
| 13:30.17 | leifmadsen | that's not what the documentation says |
| 13:30.17 | igcewieling | you can also hack something up using System or an AGI and the linux "flock" command |
| 13:30.28 | leifmadsen | hexanol: it will only try and wait 3 seconds |
| 13:30.38 | leifmadsen | if it was still locked, I would expect it to return 1, not 0 |
| 13:30.42 | leifmadsen | otherwise, it works as the docs state |
| 13:30.59 | igcewieling | That is what we did in older Asterisk boxes without the LOCK function |
| 13:31.08 | hexanol | well, the docs says "LOCK will wait for the lock to become available." |
| 13:31.09 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 13:31.15 | hexanol | and then there's the note below |
| 13:31.27 | leifmadsen | right |
| 13:31.31 | leifmadsen | which is part of the documentation |
| 13:31.41 | hexanol | which says that "if the channel already has another lock", it will only try to optain the lock for 3 seconds |
| 13:31.51 | hexanol | but in my case, I don't see how the channel has already another lock |
| 13:31.59 | danfromuk | Hi, when I do cmd RINGING, is asterisk providing the ringing sound, or is that generated by the endpoint? |
| 13:32.02 | hexanol | i.e. I'm just calling the 130 extension from two different phones |
| 13:32.10 | hexanol | (see above) |
| 13:32.32 | leifmadsen | hexanol: I'd read that as, "if a channel has a lock" |
| 13:32.55 | leifmadsen | but that's just me |
| 13:32.57 | hexanol | but that channel never called the LOCK function before, how could it have a lock ? |
| 13:33.03 | hexanol | well it might be me also :) |
| 13:33.08 | hexanol | as I am a non native english speaker |
| 13:33.10 | leifmadsen | i would expect LOCK() to work across channels |
| 13:33.17 | leifmadsen | otherwise I don't necessarily see the point of it |
| 13:33.22 | hexanol | indeed |
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| 13:33.37 | hexanol | oh ok, if "a channel" |
| 13:33.52 | hexanol | yeah, well I guess the documentation can be a bit confusing then |
| 13:34.00 | hexanol | i.e. the usage of "a" vs "the" |
| 13:34.07 | leifmadsen | right |
| 13:34.16 | leifmadsen | likely a typo |
| 13:34.24 | hexanol | but then |
| 13:34.27 | hexanol | it talks about "another lock" |
| 13:34.36 | leifmadsen | at this point, you'll have to look at the code |
| 13:34.38 | hexanol | in my case, I'm talking about the same lock |
| 13:34.45 | leifmadsen | we can nitpick on the docs :) |
| 13:35.00 | leifmadsen | I'd look at it, but don't have time, but the function is likely very simple |
| 13:35.02 | hexanol | yeah I did look a bit, got confused too |
| 13:35.35 | hexanol | leifmadsen: ok |
| 13:35.45 | hexanol | thanks for your help |
| 13:36.09 | [TK]D-Fender | <danfromuk> Hi, when I do cmd RINGING, is asterisk providing the ringing sound, or is that generated by the endpoint? <- depends on the "answer" status of the channel |
| 13:36.46 | WIMPy | .. and thec channel and its configuration. |
| 13:36.53 | danfromuk | [TK]D-Fender: If the call has not been answered yet and the endpoint is a sip phone. |
| 13:37.11 | *** join/#asterisk vlad_sta_ (~vlad_star@109.188.126.78) |
| 13:38.09 | [TK]D-Fender | danfromuk, then it should be the phone ringing. I'm figuring this is an "indications" issue, correct? |
| 13:39.00 | igcewieling | In my experience Asterisk always generates the ringing sound as early media on unanswered SIP. This is not how it is usually documented, but I've been unable to make it not do that. Plays hell with g729 licenses. |
| 13:39.04 | danfromuk | The SIP Phone is dialling a pstn number but doesnt hear any ringing until the call is actually answered. |
| 13:39.18 | igcewieling | danfromuk: ISDN is a special case. |
| 13:39.45 | danfromuk | No isdn involved. Only sip |
| 13:39.51 | igcewieling | danfromuk: is the dest port analog or ISDN or SIP? |
| 13:39.52 | danfromuk | and a wholesale sip provider |
| 13:40.15 | danfromuk | Sip phone -> our hosted asterisk boxes -> wholesale sip provider |
| 13:40.47 | igcewieling | danfromuk: what codec? |
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| 13:41.06 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:41.28 | PickledParrot | Has anyone experienced any issues with RRMEMORY queue strategy and wrapup time before? Had a google but can't seem to find much on it under reasonable traffic (I'm using 20 concurrent calls for testing) wrapup time will be ignored and the call will be pushed to an agent as soon as they end the previous call (Ignoring the wrap up time) |
| 13:41.54 | danfromuk | igcewieling: alaw on both sides |
| 13:42.23 | [TK]D-Fender | <danfromuk> The SIP Phone is dialling a pstn number but doesnt hear any ringing until the call is actually answered. <- then they aren't passing indication back. Feel free to fake your own. |
| 13:42.35 | [TK]D-Fender | danfromuk, Like the DIAL app already offers you |
| 13:43.38 | igcewieling | danfromuk: try putting a Progress() before your Dial. If that doesn't work use the dreaded "r" option of dial. Everytime you use the "r" option to dial, god kills a kitten, but sometimes kittens must be sacrificed for the greater good. |
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| 13:55.06 | danfromuk | Just checked the SIP debug and it does send a 180 Ringing to the endpoint but the endpoint doesnt produce ringing. |
| 13:55.35 | danfromuk | Sip peer -> asterisk -> sip peer does produce ringing. |
| 13:56.11 | igcewieling | danfromuk: do either of my suggestions solve the issue? |
| 13:59.37 | *** join/#asterisk PickledParrot (~aj@79.135.102.10) |
| 13:59.52 | danfromuk | I've not had much success with Progress and ringing. For example, ringing followed by an engaged tone if the destination is engaged. |
| 14:01.40 | igcewieling | danfromuk: there are a couple of sip.conf options related to ringing / early media / progress |
| 14:04.57 | jmetro | I dont think I ever use Answer() or Progress() or NoOp() for that matter |
| 14:05.24 | danfromuk | I should mention that only one client is experiencing this issue. The other clients on the same server with the same dialplan are working fine. So i'm not too keen on changing the conf files. |
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| 14:10.34 | navaismo | add a fake ring |
| 14:10.36 | danfromuk | Doing a test now to compare the sip debug for external calls and internal calls to see why the endpoint produces ringing for internal calls but not external calls. |
| 14:12.17 | danfromuk | I dont want to do a fake ring if i can help it since mobiles can take a few seconds to actually start ringing and sometimes the line is engaged. |
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| 14:18.44 | navaismo | hmmm So you dont like to edit at all sip or dialplan, so tell me more how you want to resolv the issue |
| 14:19.03 | danfromuk | I think it might be a firewall issue. |
| 14:19.11 | danfromuk | On the sip peers side. |
| 14:21.14 | danfromuk | Apparently, they hear a 'blip' of ringing but then its stops until the call is answered. |
| 14:24.22 | *** join/#asterisk jeffspeff (~jeff.clay@12.49.160.131) |
| 14:26.07 | jeffspeff | what is the "secure" method of deploying phone configuration files over the public internet? i'm using tftp right now, which works great, but anybody can browse the tftp server and read the SIP usernames and passwords of all my phones. ftp and http would also have this same issue |
| 14:27.51 | jmetro | http |
| 14:27.51 | hexanol | jeffspeff: this is not an asterisk issue, but some phones supports configuration file encryption |
| 14:27.58 | hexanol | but this is painful to put in place |
| 14:28.08 | hexanol | and some systems (like the one used by Aastra) are broken |
| 14:28.10 | danfromuk | jeffspeff: polycom can provide a config file encryption program. but i think its pointless since you can simply download the config file and use it in an encrypted state. |
| 14:28.21 | danfromuk | Safest is to set up a vpn |
| 14:28.30 | jmetro | http. |
| 14:28.31 | igcewieling | jeffspeff: no, people cannot browse a tftp server. The solution is to not use the factory default username/password when using FTP. |
| 14:28.56 | danfromuk | igcewieling: they could packet sniffer for the ftp login details. |
| 14:29.25 | jmetro | danfromuk: by that logic, no system is secure ever because anything is breakable except [at the moment] diffe hellman transactions |
| 14:29.30 | [TK]D-Fender | jeffspeff, How about using something that IS encrypted ... and passworded instead? |
| 14:29.47 | igcewieling | danfromuk: indeed, but they would need access to the stream, which eliminates "random hacking" from the internet |
| 14:30.17 | igcewieling | danfromuk: I believe the polycom enctyption is tied to the mac of the phone so the config file would not work on a different polycom phone |
| 14:31.05 | danfromuk | As far as i'm aware, the mac is simply in the file name and not in the file. Unless they use the mac to salt the encryption. |
| 14:31.07 | igcewieling | I've not had a single system exploited via FTP since we stopped using the Polycom factory default user/pass about 5 years ago. |
| 14:31.38 | igcewieling | danfromuk: if either of us cared enough we could read the polycom docs to be sure. 8-) |
| 14:31.57 | jmetro | use http, configs are only visible if you know the mac address, and only if you configure them to be visible |
| 14:32.02 | danfromuk | igcewieling: we made things a little more secure and each client has its own ftp account so clients can't sniff their own phones and gain access to other user login details. |
| 14:32.11 | igcewieling | we've had systems exploited in plenty of OTHER ways, but not via FTP |
| 14:32.19 | danfromuk | with http, you can't receive log files. |
| 14:32.20 | igcewieling | danfromuk: same here |
| 14:32.24 | WIMPy | jmetro: It's not that hard to guess the MAC address. |
| 14:35.29 | jmetro | you can also read the mac and make sure it matches the mac youre trying to look at |
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| 14:37.43 | jeffspeff | thanks for the info everybody. very informative from all angles. |
| 14:38.40 | jmetro | though i suppose at any point you can sniff traffic, spoof a mac, and request their config |
| 14:39.00 | [TK]D-Fender | jeffspeff, Run something for IDS, use something encrypted and passworded. |
| 14:39.35 | danfromuk | Sorted that ringing issue. |
| 14:39.40 | danfromuk | It was a config on the yealink phone. |
| 14:39.56 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 14:39.58 | WIMPy | https with username/password should be ok. |
| 14:40.04 | jeffspeff | fail2ban + polycom encrypted config + sftp should do the trick then |
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| 14:41.17 | jmetro | jeffspeff: vpn might be a better use of time, especially since you can QOS it. |
| 14:42.06 | jeffspeff | many users are at home with handset phones (polcom ip 335) |
| 14:46.10 | igcewieling | jmetro: huh? VPN often makes QoS harder to do. |
| 14:46.59 | igcewieling | real QoS is impossible over "the internet" anyway. |
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| 14:50.17 | igcewieling | When you put data over a VPN the QoS fields of the packets are encrypted so are not seen by routers. Some VPN software (Cisco IPSec is one) allows you to copy the ToS/DSCP tags to the wrapper packet header, but that does expose the QoS info. |
| 14:50.48 | igcewieling | But none of this matters if the data is going over the general internet |
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| 15:08.04 | MLNoah | http://pastebin.com/7QC7jUfy - I have a customer's Asterisk 1.6.2.7 box that is using this sip.conf to connect a SIP trunk to my Asterisk 11.3.0 call routing box at my office. The settings on the central office box are essentially the same. With from-user turned on, calls route, but the CID is getting overwritten. With from-user turned off, the 1.6.2.7 box rejects calls, with this error: |
| 15:08.10 | MLNoah | [Apr 29 10:57:43] WARNING[3761]: chan_sip.c:12671 check_auth: username mismatch, have <ssb-trunk>, digest has <ssb-sherwood> |
| 15:08.11 | MLNoah | [Apr 29 10:57:43] NOTICE[3761]: chan_sip.c:19990 handle_request_invite: Failed to authenticate device <sip:4195530049@209.173.175.174>;tag=as06c1ba0 |
| 15:08.51 | leifmadsen | s/from-user/fromuser |
| 15:08.53 | MLNoah | anyone have any suggestions I may have overlooked? I've re-entered the sip.conf file from scratch with no symptom changes, and it's driving me nuts. I have another 1.6.2.7 box with the same customer (different branch) with effectively the same settings and it works. |
| 15:09.13 | leifmadsen | you need the fromuser, yes, and the CID should be separate of that, but you need trustrpid and sendrpid |
| 15:10.34 | MLNoah | thanks, that got it |
| 15:10.51 | MLNoah | so, the box that's working without fromuser (with just defaultuser) is not "working correctly" and I should adjust accordingly? |
| 15:11.02 | leifmadsen | I will not answer that question :) |
| 15:11.25 | *** join/#asterisk butthurtface (~Butthurtf@38.122.108.2) |
| 15:11.31 | [TK]D-Fender | MLNoah, both should set fromuser, both should use rpid for CID |
| 15:11.47 | [TK]D-Fender | MLNoah, This is "doing it right". Don't make that a selective thing.... |
| 15:11.50 | MLNoah | ok. I'll adjust the other box then. Thanks a lot |
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| 16:51.58 | *** join/#asterisk StaRetji (~LittleAll@91.142.129.1) |
| 16:52.04 | StaRetji | Howdy folks |
| 16:52.15 | *** join/#asterisk cmendes0101 (~cmendes01@72.1.46.254) |
| 16:52.23 | StaRetji | can someone help me out, I need to redirect to extension, based on rdnis |
| 16:52.54 | igcewieling | StaRetji: OK. What is the actual issue? |
| 16:52.55 | *** join/#asterisk moos3 (~textual@cpe-72-224-215-87.maine.res.rr.com) |
| 16:52.57 | StaRetji | basically, I have mobile phone number, redirecting to land line and landline is did on my asterisk |
| 16:53.05 | StaRetji | thx igcewieling for reply |
| 16:53.07 | moos3 | question anyone using hylafax with rhel 6 ? |
| 16:53.33 | WIMPy | Goto(${CALLERID(whatever)},1)? |
| 16:53.34 | StaRetji | so, currently, I am forwading landline to asterisk extension, as did, works well |
| 16:53.36 | igcewieling | StaRetji: it is unlikely you will receive any redirecting information on your landline. |
| 16:53.45 | StaRetji | I am |
| 16:53.49 | StaRetji | provider is giving me that |
| 16:54.04 | WIMPy | Or if it's not in CALLERID, it might be in REDIRECTING. |
| 16:54.07 | StaRetji | I see rdnis number as mobilenumber@ipdrdess |
| 16:54.37 | StaRetji | thx WIMPy, if this is meant for me lol |
| 16:54.42 | [TK]D-Fender | StaRetji, and RDNIS is in the CALLERID() function |
| 16:54.57 | igcewieling | StaRetji: you need to extract the number from the redirecting info, using CUT function may work well for that |
| 16:55.04 | StaRetji | ehm |
| 16:55.27 | StaRetji | bitchsplaps himself |
| 16:55.42 | StaRetji | I am trying to understand it |
| 16:56.40 | WIMPy | I have no idea if the parts in CALLERID and REDIECTING are the same or can be different. |
| 16:56.40 | StaRetji | I guess it is something like ${CALLERID(rdnis)} |
| 16:57.16 | igcewieling | StaRetji: "core show function CALLERID" and also run the DumpChan application when the call comes in so you can see all the channel variables which are set. |
| 16:57.43 | igcewieling | Though this doesn't sound much like a "landline" it sounds like a "sip line" |
| 16:58.47 | StaRetji | well, it is mobilephone number of operator, 10 numbers attached to 1 landline |
| 16:58.56 | WIMPy | wonders how many ITSPs would supply that information. |
| 16:59.08 | StaRetji | so, users are calling 10 mobile numbers, 1 land line is attached to my asterisk as DID |
| 16:59.27 | StaRetji | now, if you call landline directly, it will ring my 2001 extension |
| 16:59.45 | StaRetji | but I need to make it mobile number x2001 ring extension 2001 |
| 16:59.56 | StaRetji | and so on untul x2010 ring 2010 |
| 17:00.12 | StaRetji | all coming from one x1000 landline |
| 17:00.16 | [TK]D-Fender | StaRetji, nothiing attaches "as a DID". DID is a just a NUMBER... how it is delivered to you is another matter |
| 17:00.20 | aruntomar | i would like to control all the settings via the config files, but also need a web frontend as client needs it. there are couple of choices like freepbx or asteriskgui etc. which one is suggested, that will not get in the way of the admins. |
| 17:00.22 | igcewieling | sounds like ISDN. |
| 17:00.39 | [TK]D-Fender | aruntomar, Neither. |
| 17:00.53 | igcewieling | StaRetji: what PSTN interface do you have for your landline? FXO, ISDN BRI, ISDN PRI? |
| 17:00.59 | [TK]D-Fender | aruntomar, AsteriskGUI is effectively dead, and FreePBX doesn't let you do much custom work... |
| 17:01.06 | StaRetji | its coming via SIP |
| 17:01.15 | StaRetji | I am connected to telco operator via SIP |
| 17:01.23 | igcewieling | StaRetji: that stop calling it a landline |
| 17:01.29 | igcewieling | s/that/then |
| 17:01.34 | [TK]D-Fender | StaRetji, RDNIS is in the CALLERID functions. Go LOOK at it's value and processes your call based on it however you want |
| 17:01.41 | StaRetji | yes sorry, I calling did number, ok? |
| 17:02.14 | igcewieling | StaRetji: You already have all the information required to solve this. "(12:57:16 PM) igcewieling: StaRetji: "core show function CALLERID" and also run the DumpChan application when the call comes in so you can see all the channel variables which are set." |
| 17:02.23 | StaRetji | so, instead CALLERID=x2001 I can do RDNIS=x2001 |
| 17:02.24 | StaRetji | ? |
| 17:02.30 | jmetro | aruntomar: Like [TK]D-Fender all the pre-built GUI's will blow up your configs. do you need a GUI like an operator panel or a GUI to actualyl modify dialplan, add users to the system, etc. |
| 17:02.33 | StaRetji | ok folks, thx |
| 17:03.11 | aruntomar | jmetro: add user, create conference bridge etc. |
| 17:03.44 | [TK]D-Fender | StaRetji, CALLERID ...... FUNCTION |
| 17:03.50 | [TK]D-Fender | StaRetji, "core show function CALLERID" |
| 17:03.52 | jmetro | aruntomar: so you need to enter a phones mac in the system, assign its settings, add an extension, etc... from your gui? your CLIENTS are doing this? |
| 17:04.45 | jmetro | aruntomar: cause there are operator panels that will let you have a GUI for making calls, parks, conference bridges, voicemail... user-end stuff. |
| 17:05.12 | aruntomar | jmetro: yes, they want to maintain it. if i had to maintain it, i would maintain everything from command line and config files. |
| 17:05.26 | StaRetji | [TK]D-Fender: I did it, thx, I kind of understand it lol |
| 17:05.48 | [TK]D-Fender | aruntomar, There is no suitable product for your expectations to config by hand and still have them use a GUI for their end |
| 17:06.04 | jmetro | aruntomar: the easiest way for the client would be freepbx, but youre going to hate yourself if you try to do anything command line. |
| 17:06.04 | jmetro | Make one yourself, in other words. |
| 17:06.05 | aruntomar | jmetro: but the client don't want a continuous support or maintenance. they'll come running only when there is a fire and when they'll screw up things and can't fix it ;) |
| 17:06.09 | StaRetji | I still don't understand the logic how to match it, but I guess you guys already help me |
| 17:06.11 | StaRetji | thx |
| 17:06.15 | igcewieling | You would have better luck finding a rainbow unicorn |
| 17:06.52 | jmetro | aruntomar: if thats the case, make them do it command line as well so they can understand why they need to pay you 20$ a month for low level support |
| 17:07.37 | aruntomar | [TK]D-Fender: ok. good to know that. |
| 17:08.22 | *** part/#asterisk hexanol (~bibi@modemcable094.94-70-69.static.videotron.ca) |
| 17:08.34 | [TK]D-Fender | aruntomar, Which leaves pretty much FreePBX as the solution you'll both have to learn |
| 17:08.41 | aruntomar | jmetro: from what i've read, freepbx will maintain all the config in it's own database or settings, and that it would be frustrating for users/sysadmins used make things work just by modifying the config files. |
| 17:09.03 | aruntomar | [TK]D-Fender: looks that way. |
| 17:09.05 | [TK]D-Fender | aruntomar, because you simply DON'T. |
| 17:09.34 | [TK]D-Fender | aruntomar, Users just use what the GUI offers them and there is little reason to ever to9uch a config file manually. |
| 17:09.38 | jmetro | aruntomar: You can TRY to modify the configs command line but youll probably wind up having to junk your changes because everything will break |
| 17:11.35 | BoRiS | Hi everyone, having an issue building Asterisk 11 from source. Ever time when it builds, it keeps erroring out at multiple definition of `CRYPTO_set_locking_callback', multiple definition of `ERR_free_strings'. This is not a problem with asterisk 10. I even tried the latest asterisk 11 from svn. ( http://www.pastebin.ca/2369566 ). Any ideas? |
| 17:11.36 | aruntomar | [TK]D-Fender: jmetro: thanks for the heads up guys. that's y i love the community. i might have come to this conclusion after banging my head for may be a week or 2, but then you saved me from all that pain and showed the right direction :) |
| 17:12.21 | jmetro | aruntomar: If you feel like becoming an engineering genius you could design your own that only interacts with dialplan and allows for phone add/changes |
| 17:12.53 | jmetro | gives them options to change things within the flow like dial length, sound files, but doesnt let them change the routing itself [easier on you] |
| 17:13.20 | [TK]D-Fender | Doesn't take genius... just some very specific planning on the scope you intend to offer. |
| 17:14.28 | [TK]D-Fender | And as to how much you want to be able to modularize and make easy to mod |
| 17:14.46 | *** join/#asterisk Alagar (~helpdesk@vsusm15.vernalissystems.com) |
| 17:15.39 | aruntomar | may be this calls for another opensource webui project for asterisk, just saying ;) |
| 17:16.55 | jmetro | it would be pretty easy to make a dialplan out of variables stored in a DB and let them edit the variables but have to ask you for routing changes |
| 17:17.36 | igcewieling | aruntomar: if it was easy someone would have done it already. |
| 17:17.45 | aruntomar | another query, can i record the calls directly in ogg rather than in wav and then convert them to ogg. |
| 17:17.53 | igcewieling | BoRiS: try using a released verison of Asterisk 11 instead of SVN |
| 17:17.57 | jmetro | trying to automate coding is hard |
| 17:18.01 | [TK]D-Fender | Each piece is easy. Actually... from a certain point of view, it's ALL easy... just time consuming. |
| 17:19.00 | igcewieling | jmetro: if you limit your requirements it isn't all that hard. the problem is people don't want to limit their requirements and releasing something public requires you do not limit your requirements. |
| 17:19.57 | *** join/#asterisk anonymouz666 (~anonymouz@189-105-205-226.user.veloxzone.com.br) |
| 17:20.35 | anonymouz666 | anyone already configured the audiocodes mp118 gateway? |
| 17:22.51 | WIMPy | As it's from 2006 (or earlier) it's likely that someone hase configured one by now. |
| 17:24.40 | igcewieling | I wonder if there is information on the audiocodes mp118 in the mailing list archives.... |
| 17:25.01 | anonymouz666 | it's hard to find any documentation about this gateway |
| 17:25.02 | WIMPy | Google did help. |
| 17:25.17 | *** join/#asterisk italoros_ (~italoross@187.60.66.11) |
| 17:25.42 | anonymouz666 | i dialed from gw01 to gw02. I don't know why, the second gateway answers and provides a "dial tone" instead forward the SIP number received |
| 17:26.47 | anonymouz666 | like two stages dialing |
| 17:27.37 | *** join/#asterisk cmendes0101 (~cmendes01@72.1.46.254) |
| 17:28.00 | *** join/#asterisk HolTech (~IceChat77@pool-74-107-104-2.bltmmd.fios.verizon.net) |
| 17:35.30 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 17:35.40 | Kobaz | okay so, i have a realtime logger that's going to be writing to an unlogged table for speed... and then i'm going to have a background daemon, routinely sweep the logs into a permenant table. What's the best way to sweep them over without causing excessive locking/etc? |
| 17:36.11 | Kobaz | whoops wrong channel |
| 17:36.13 | drmessano | Use a broom with a smaller handle |
| 17:38.39 | *** join/#asterisk cmendes0101 (~cmendes01@wtnl.corp.tierra.net) |
| 17:40.46 | *** join/#asterisk mmlj4 (1000@ip68-11-55-215.no.no.cox.net) |
| 17:41.20 | *** join/#asterisk cmendes0101| (~cmendes01@wtnl.corp.tierra.net) |
| 17:44.39 | *** join/#asterisk cmendes0101 (~cmendes01@72.1.46.254) |
| 17:51.21 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 17:54.01 | igcewieling | or a leaf blower! Not the electric ones, the big loud gas powered ones belching toxic fumes and making so much noise it is hard to think. </rant> |
| 18:02.57 | *** join/#asterisk ledoktre (~ledoktre@199-102-208-169.static.osage.net) |
| 18:03.32 | *** part/#asterisk aruntomar (~Thunderbi@49.248.155.66) |
| 18:03.40 | jmetro | igcewieling: I wonder if ozone is more dangerous than gas exhaust |
| 18:03.48 | StaRetji | ok folks, I am willing to pay for help, need someone to make me extension on Asterisk 1.8 to read rdnis and upon rdnis value to forward to specific extension |
| 18:03.49 | StaRetji | thx |
| 18:03.59 | Qwell | jmetro: gas exhaust eats ozone for breakfast. |
| 18:04.31 | WIMPy | StaRetji: don't you find the information or can't you get your dialplan to work? |
| 18:04.40 | StaRetji | when I do, sip set debug on, I clearly see rdnis value |
| 18:04.52 | StaRetji | I am failing to understand it :/ |
| 18:04.59 | WIMPy | But not in any function? |
| 18:05.00 | jmetro | is there a variable that contains rdnis? |
| 18:05.11 | [TK]D-Fender | StaRetji, Never use the term "forward". |
| 18:05.26 | [TK]D-Fender | StaRetji, Now when you call our your ITSP then you get the CALLERID that THEY give you |
| 18:05.37 | [TK]D-Fender | StaRetji, Unless your ITSP ALLOWS you to change the callerID |
| 18:05.45 | StaRetji | ok [TK]D-Fender, and thx WIMPy. |
| 18:05.55 | StaRetji | I have contract with them to give me that |
| 18:05.57 | *** join/#asterisk darksk1ez (~mhb@fsf/member/darkskiez) |
| 18:06.03 | [TK]D-Fender | StaRetji, So first go prove that your provider even allows you to set the calleriD> |
| 18:06.04 | StaRetji | it is clearly visible |
| 18:06.15 | StaRetji | yes, this if for incoming calls |
| 18:06.17 | [TK]D-Fender | Then... before you dial out.... CHANGE IT YORUSELF to prove that you can. |
| 18:06.31 | WIMPy | [TK]D-Fender: Wrong direction |
| 18:06.57 | [TK]D-Fender | WIMPy, He want's the outgoing to reflect something... it does matter. |
| 18:07.07 | WIMPy | StaRetji: If both CALLERID and REDIRECTING fail, you can still use SIP_HEADER. |
| 18:07.08 | StaRetji | I am receiving call toward 1234, but rdnis is 3344 |
| 18:07.13 | [TK]D-Fender | WIMPy, Who cares what the inbound looks like if the outbound do9esn't allow it |
| 18:07.23 | StaRetji | based on rdniss value I need to send to correct extension |
| 18:07.34 | WIMPy | It's only about inbound. |
| 18:07.46 | [TK]D-Fender | StaRetji, So you are getting a call IN and just need to ring a DEVICE on your server? |
| 18:07.54 | [TK]D-Fender | StaRetji, Also stop using the word "extension" like that |
| 18:08.04 | [TK]D-Fender | StaRetji, Be very specirfic about where your calls are supposed to be going |
| 18:08.30 | StaRetji | yes |
| 18:08.31 | [TK]D-Fender | StaRetji, Go show us this inbound call also dumping the CALLERID forrdnis for it |
| 18:08.34 | [TK]D-Fender | ~pb |
| 18:08.34 | 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. |
| 18:08.36 | [TK]D-Fender | ^^^^^ |
| 18:08.38 | StaRetji | it is like this |
| 18:08.55 | StaRetji | 1234 for calls 4567, I need to ring 600 echo lol |
| 18:09.06 | StaRetji | if rdnis is 1234 |
| 18:09.17 | [TK]D-Fender | StaRetji, Fine, if it's all internal functioally then go show us your code & call |
| 18:09.17 | StaRetji | if rdnis is 2345 I need to ring 1111 |
| 18:09.19 | jmetro | Qwell: This same high oxidizing potential, however, causes ozone to damage mucus and respiratory tissues in animals, and also tissues in plants, above concentrations of about 100 parts per billion. |
| 18:09.31 | jmetro | very toxic apparently. |
| 18:12.00 | igcewieling | StaRetji: what variable/function contains the rdnis information on YOUR asterisk? |
| 18:13.03 | *** join/#asterisk kikohnl (~keith@udp278022uds.hawaiiantel.net) |
| 18:13.28 | StaRetji | igcewieling: I don't have idea |
| 18:13.31 | igcewieling | GotoIf($["${RDNIS}" = "1234"]?600,1) assuming on YOUR asterisk the RDNIS variable contains the information you are looking for. This isn't rocket science. |
| 18:13.49 | igcewieling | StaRetji: then apparently you did not do a Dumpchan or look at the callerid function |
| 18:13.53 | StaRetji | well, yes, for Armstrong is not ;) |
| 18:14.06 | StaRetji | I watched sip debug |
| 18:14.55 | igcewieling | that is pretty useless once you verify the info is in SIP. Now do a dumpchan when the call comes in and stop wasting everyone's time by not following suggestions. |
| 18:15.03 | WIMPy | Use Verbose to look at the contents of CALLERID or REDIRECTING functions. |
| 18:15.08 | HolTech | does anyone know if a message waiting stutter dial tone can work with an analog phone and * ? |
| 18:15.27 | WIMPy | HolTech: Connected how? |
| 18:15.29 | igcewieling | HolTech: Assuming you mean on an analog PORT, yes. |
| 18:15.58 | HolTech | an analog phone connected to * through an ATA ( ie sipura 3000 ) |
| 18:16.20 | igcewieling | HolTech: that is a SIP device and it is up to the sip device to provide the dialtone and stutter dialtone |
| 18:16.27 | WIMPy | Then you have to look at the ATAs manual. |
| 18:16.45 | HolTech | gotcha... thanks |
| 18:18.23 | *** join/#asterisk krotos (~root@devapp.as48500.net) |
| 18:18.25 | krotos | hi all guy |
| 18:19.27 | krotos | there is a way to catch the operator that answer to a call inside a queue? The scenario is: incoming call is sent to queue, the operator answer so i need to know wich operator has answered |
| 18:20.04 | StaRetji | igcewieling: I did DumpChan |
| 18:20.33 | StaRetji | I see RDNIS = 1234 |
| 18:20.49 | WIMPy | Great. |
| 18:20.56 | WIMPy | So what more do you need? |
| 18:21.07 | igcewieling | StaRetji: now you can use my example I posted as a starting point |
| 18:21.21 | StaRetji | ehm, I new rdnis is 1234 since the beginning . |
| 18:21.37 | igcewieling | StaRetji: but you didn't know what dialplan variable it was contained in |
| 18:21.44 | WIMPy | Goto({CALLERID(rdnis)},1) |
| 18:21.56 | igcewieling | now you do and the rest should be obvious from my example |
| 18:22.03 | WIMPy | oops |
| 18:22.08 | WIMPy | Goto(${CALLERID(rdnis)},1) |
| 18:22.40 | StaRetji | well, should be, but I need to try several hours until I will hit it. Blind chicken can also eat a work once in a while :D |
| 18:22.47 | StaRetji | worm* |
| 18:23.46 | igcewieling | StaRetji: site your source? In my experience blind chickens get eaten. |
| 18:24.02 | jmetro | wikipedia is my source |
| 18:24.03 | WIMPy | Not only the blind ones. |
| 18:24.07 | StaRetji | lol |
| 18:31.38 | HolTech | ok so, Received SIP subscribe for peer without mailbox , is complaining because the peer is not setup in user.conf ? voicemail is recording otherwise. |
| 18:32.43 | *** join/#asterisk blee (~blee@67.8.206.215) |
| 18:33.02 | WIMPy | Is that going to be a question? |
| 18:34.19 | HolTech | poorly stated i guess... thought i was on jeopardy for a second... trying to decipher what message means. Anyone know if it the user has to be setup in users.conf, or if voicemail.conf is sufficient |
| 18:34.48 | Qwell | Did you tell chan_sip which mailbox to use for that device? |
| 18:35.42 | HolTech | yes, in my sip.conf the extenstion has the line voicemail = ... and the dialplan works, recording the call when no answer |
| 18:35.56 | Qwell | Where did you see voicemail= as a valid option? |
| 18:35.59 | WIMPy | HolTech: It means the peer doesn't have a mailbox= set in sip.conf. |
| 18:36.35 | HolTech | ah, mailbox... not voicemail... dumb mistake.. thanks |
| 18:37.29 | HolTech | Aha, and now the stutter dial tone works... cheers ! thanks for your help ( again ! ) guys |
| 18:39.30 | StaRetji | igcewieling: exten => 2222,1,GotoIf($["${RDNIS}" = "1234"]?444444,1) |
| 18:39.41 | StaRetji | I am very close I guess |
| 18:39.59 | StaRetji | it seems it tries to call 0?444444 |
| 18:40.30 | igcewieling | StaRetji: pastebin the output of the failed call. |
| 18:40.50 | WIMPy | Use ${CALLERID(rdnis)} |
| 18:40.52 | igcewieling | Is the exten line you pasted EXACTLY what is in your extensions.conf? Also what version of Asterisk. |
| 18:41.04 | StaRetji | 1.8.10 |
| 18:41.15 | StaRetji | I will paste in a minute, it is on another pc |
| 18:41.51 | igcewieling | StaRetji: put a Noop(RDNIS is '${RDNIS}) as the first priority, then the exten line as the next one |
| 18:42.05 | igcewieling | sorry Noop(RDNIS is '${RDNIS}') |
| 18:43.22 | *** join/#asterisk Captain_Proton (~quassel@173.162.32.1) |
| 18:45.21 | StaRetji | igcewieling: thx, here is the paste http://pastebin.com/rfnHGWdF |
| 18:45.52 | igcewieling | StaRetji: now show us a pastebin with the Noop |
| 18:46.08 | StaRetji | ok, moment |
| 18:46.15 | igcewieling | (2:41:50 PM) igcewieling: StaRetji: put a Noop(RDNIS is '${RDNIS}) as the first priority, then the exten line as the next one |
| 18:46.37 | igcewieling | but put the trailing ' |
| 18:47.35 | igcewieling | and fix the extra ) you have |
| 18:48.32 | StaRetji | yes, trailing is there, CLI says: Executing [5111@default:1] NoOp("SIP/SmsbranaUfon-00000037", "RDNIS is ''") in new stack |
| 18:48.48 | StaRetji | other 2 lines identical |
| 18:48.53 | StaRetji | as previous |
| 18:49.46 | StaRetji | I can paste DumpChan to show that RDNIS = 1234 |
| 18:50.02 | WIMPy | Use ${CALLERID(rdnis)} |
| 18:51.00 | igcewieling | FINE! Use ${CALLERID(rdnis)} instead of ${RDNIS} ! |
| 18:51.23 | StaRetji | Wimpy, thx man, I simply don't follow where to put it? My brain understands GotoIf |
| 18:51.33 | igcewieling | there is no functional difference between the two. |
| 18:51.41 | WIMPy | Instead of ${RDNIS} |
| 18:52.19 | igcewieling | StaRetji: 1) put DumpChan as the first priority. Replace the ${RDNIS} with ${CALLERID(rdnis)} in both the noop line and the goto if line, pastebin the result when yo have a failed call. |
| 18:52.50 | igcewieling | StaRetji: so you will have three exten lines, the Dumpchan, the Noop, and the GotoIf |
| 18:53.01 | StaRetji | ha! |
| 18:53.16 | StaRetji | it seems it works |
| 18:53.18 | StaRetji | moment |
| 18:53.41 | *** join/#asterisk kresp0 (~kresp0@81.61.199.176.dyn.user.ono.com) |
| 18:53.47 | Captain_Proton | ? may boss ask me if we get a different phone system avayna, cisco VOIP whatever will that fix the small echo problem we have. I told them regerdless of the what VOIP systems we get there will be echo we can manage but t will be there. Am I right? |
| 18:54.15 | igcewieling | Captain_Proton: that depends on many things. EXACTLY how are you interfacing with the PSTN? |
| 18:54.55 | Captain_Proton | right now we have digiam card w/ hard echo |
| 18:55.21 | [TK]D-Fender | Captain_Proton, What model exactly and have you contacted Digium support for it? |
| 18:55.23 | igcewieling | Captain_Proton: contact Digium support for help with the hardware echo cancel not working correctly. |
| 18:55.29 | [TK]D-Fender | Captain_Proton, Also, what are you using for endpoints? |
| 18:55.42 | Captain_Proton | Polycom 650 |
| 18:57.25 | igcewieling | Captain_Proton: The vast majority of correctly designed and configured VoIP systems with working EC will not experience echo. |
| 18:58.19 | Captain_Proton | Will regardless am I correct in saying there will be echo but we can manage it? When talking to the outside world. |
| 18:58.45 | WIMPy | That's what echo cancelling is there for. |
| 18:58.47 | [TK]D-Fender | Captain_Proton, Your card is the issue |
| 18:59.01 | [TK]D-Fender | Captain_Proton, Check with Digium support immediately and directly for it. |
| 18:59.14 | WIMPy | But sometimes the echo is just too bad for the echo canceller to filter. |
| 18:59.22 | igcewieling | Captain_Proton: doing anything other than contacting Digium support is a TOTAL WASTE OF YOUR TIME |
| 18:59.26 | [TK]D-Fender | Captain_Proton, next steps beyond that are removing the HWEC module and testing with SWEC if the problem was not resolved earlier |
| 18:59.54 | Captain_Proton | ok i will give them a call |
| 19:03.00 | StaRetji | igcewieling: and WIMPy: thx so much guys, it seems rdnis is working, but not still sure http://pastebin.com/cUTxGtYV |
| 19:03.32 | igcewieling | StaRetji: What is your objection to Dumpchan and the noop. |
| 19:03.35 | igcewieling | leave them in there. forever |
| 19:03.41 | StaRetji | as I use a2billing, I don't have 4454808028 anywhere in extensions.conf |
| 19:03.49 | igcewieling | when you have problems in the future you will thank me. |
| 19:03.50 | StaRetji | well, I need to have 1000 numbers like this |
| 19:04.00 | StaRetji | got it, will do it right away |
| 19:04.27 | WIMPy | StaRetji: The just use a straight Goto as I suggested multiple times. |
| 19:04.28 | StaRetji | i thought to keep it small as possible, but hey, I wont argue, you help me dude |
| 19:05.07 | igcewieling | StaRetji: extra dialplan lines cause little extra overhead. |
| 19:05.19 | [TK]D-Fender | StaRetji, You aren't showing us your actual dialplan as requested |
| 19:05.43 | igcewieling | you are just slowing down the solution to your problem by leaving out the criticially important debugging lines I suggested you add. |
| 19:05.59 | [TK]D-Fender | StaRetji, Complete call. complete dialplan |
| 19:06.03 | igcewieling | If I have to ask again I will stop helping. |
| 19:07.35 | StaRetji | ok, I will paste all in a moment, I have hard time, because I need to keep it private, so need to change few names and numbers |
| 19:07.52 | StaRetji | and thx all for helping so far, it is highly appreciated |
| 19:08.18 | igcewieling | We don't care about your TNs. If making a TN public is a security issue then you have no hope of having a secure system. |
| 19:11.46 | WIMPy | Well, that's a normal issue if you have premium rate numbers. |
| 19:13.32 | *** join/#asterisk blee (~blee@67.8.206.215) |
| 19:16.22 | StaRetji | igcewieling WIMPy [TK]D-Fender extensions.conf: http://pastebin.com/hhP2LhGW Asterisk CLI: http://pastebin.com/83cCtHwv |
| 19:16.33 | StaRetji | thx folks :) |
| 19:17.06 | igcewieling | StaRetji: as you can see from the DumpChan RDNIS is not set so it won't go to the number |
| 19:17.48 | [TK]D-Fender | StaRetji, exten => 5111,3,GotoIf($["${CALLERID(rdnis)}" = "1234"]?4454808028,1) <-- there is no "4454808028,1" in default just like it says |
| 19:17.54 | [TK]D-Fender | StaRetji, LOOK AT THAT |
| 19:18.04 | [TK]D-Fender | StaRetji, You are telling it to go somewhere that does not exist? |
| 19:18.07 | [TK]D-Fender | StaRetji, why? |
| 19:18.17 | [TK]D-Fender | StaRetji, Where should it ACTUALLY be ging? |
| 19:18.23 | StaRetji | but it works if I do without rdnis |
| 19:18.26 | WIMPy | The Noop doesn;t use CALLERID! |
| 19:18.44 | StaRetji | I have a2billing which handles sip peers |
| 19:18.53 | [TK]D-Fender | StaRetji, LOOK at what you made there |
| 19:19.02 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 19:19.07 | [TK]D-Fender | StaRetji,YOU told it to jump somewhere that clearly doesn't have a match |
| 19:19.20 | [TK]D-Fender | StaRetji, there is no match in [default] for that |
| 19:19.36 | WIMPy | StaRetji: Do a new context containing all possible redirecting numbers and Goto tehere without any conditionals. |
| 19:19.43 | jmetro | I think igcewieling is about to DC |
| 19:19.46 | WIMPy | Then define the real targets there. |
| 19:19.53 | WIMPy | DC? |
| 19:19.59 | jmetro | disconnect |
| 19:20.07 | StaRetji | if I do this, it works: exten => 5111,1,Dial(SIP/4454808028,15) |
| 19:20.10 | WIMPy | Death! |
| 19:20.18 | [TK]D-Fender | StaRetji, that has NOTHING to do with the GOTO you put there |
| 19:20.30 | [TK]D-Fender | StaRetji, You don't seem to have a single clue where you are SENDING it to |
| 19:20.39 | StaRetji | well, sort of lol |
| 19:20.40 | igcewieling | jmetro: nope. He is so far beyond my help we might as well be in different galaxies. Sad, since it really is a simple problem. |
| 19:20.58 | StaRetji | problem is, a2billng handles peers, so I thought it will do in this case too |
| 19:21.14 | WIMPy | That doesn't matter. |
| 19:21.18 | jmetro | igcewieling: I thought the mention of a2billing would just make your IRC client uninstall itself |
| 19:21.20 | StaRetji | I mean, can I do SIP/4454808028,15 ? |
| 19:21.35 | StaRetji | well, I have to bill somehow jmetro :) |
| 19:21.37 | jeffspeff | lol @jmetro |
| 19:21.58 | [TK]D-Fender | StaRetji, You don't seem to understand GOTO |
| 19:22.05 | [TK]D-Fender | StaRetji, GOT is NOT a DIAL |
| 19:22.32 | StaRetji | ok, is there Dialif then? lol |
| 19:22.33 | StaRetji | :D |
| 19:22.34 | WIMPy | And I doubt A2Billig will work if you Dial a SIP peer directly, but I hve no idea about it. |
| 19:22.58 | WIMPy | There is execif. |
| 19:23.00 | StaRetji | ok, so I will have to make context for each client in a2b? |
| 19:23.08 | [TK]D-Fender | WIMPy, He has no clue about the dialplan whatsoever |
| 19:23.10 | StaRetji | and put Dial there |
| 19:23.31 | StaRetji | well, I have some clue |
| 19:23.38 | WIMPy | And I told you at least three times how to use a goto and make extensions for the redirecting numbers. |
| 19:23.45 | [TK]D-Fender | StaRetji, Nothing we've seen evidence for... |
| 19:23.59 | igcewieling | [TK]D-Fender: give up, it isn't worth the stress. |
| 19:24.06 | [TK]D-Fender | StaRetji, exten => 5111,3,GotoIf($["${CALLERID(rdnis)}" = "1234"]?a2billing,4454808028,1) <---- SPECIFY THE CONTEXT |
| 19:24.08 | StaRetji | c'mon folks, I try here |
| 19:24.27 | jeffspeff | StaRetji, is your name Jared by chance? |
| 19:24.35 | StaRetji | nope |
| 19:24.39 | StaRetji | why> |
| 19:24.40 | StaRetji | ? |
| 19:24.43 | WIMPy | StaRetji: You really do give the impression of having no clue about dialplans. |
| 19:24.51 | jeffspeff | are you with a tech company in Houston? |
| 19:25.12 | [TK]D-Fender | StaRetji, Not really... I am recalling the long history of your coming in here with no clue and trying to do as little as humanly possible to run a system you don't understand. |
| 19:26.38 | WIMPy | And using the default context for incomming calls is not a good idea, either, BTW. |
| 19:26.45 | StaRetji | well, I try, really, it is not my choice :/ |
| 19:26.45 | [TK]D-Fender | StaRetji, Go fix the line I gave you |
| 19:26.52 | StaRetji | I did |
| 19:26.59 | [TK]D-Fender | StaRetji, And you are showing us nothing |
| 19:27.15 | jeffspeff | now go to command prompt as root user and type rm -rf /* |
| 19:27.38 | WIMPy | What's the * good for? |
| 19:27.40 | jmetro | jeffspeff: :| |
| 19:28.04 | navaismo | if you are specifying the rm -rf / you dont need the * |
| 19:28.05 | jeffspeff | WIMPy, * is pattern match for anything... not really necessary, just force of habbit |
| 19:28.11 | navaismo | oooh |
| 19:28.12 | navaismo | ok |
| 19:28.37 | jmetro | jeffspeff: you know * is called * because * is the wildcard that can match anything right? |
| 19:28.52 | [TK]D-Fender | UUOC <- :) |
| 19:29.00 | WIMPy | Actually /* matches less than /. |
| 19:29.11 | jeffspeff | jmetro, never thought about it, but makes sense |
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| 19:29.30 | jmetro | jeffspeff: perhaps you should read the book :3 |
| 19:29.55 | jeffspeff | jmetro, i've referenced quite a bit from the book, but never read it from cover to cover |
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| 19:31.23 | navaismo | heresy |
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| 19:37.49 | igcewieling | The key to troubleshooting customer reported problems is to know that whatever the customer reports is guaranteed NOT to be what is happening. |
| 19:43.34 | *** join/#asterisk [SySteM] (~antoine@85.69.246.241) |
| 19:43.36 | [SySteM] | Hello |
| 19:43.44 | [SySteM] | Anyone use Ices with asterisk ? |
| 19:44.01 | igcewieling | jmetro: now we get the DC |
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| 19:49.21 | StaRetji | I've made it. I guess I needed to be slapped by you guys, specially by [TK]D-Fender. I am sorry, it is not that I am ignorant, but it is hard for me to understand and I am not administering it because I like it. I have idiot boss which doesn't want to hire expert for this and is jumping on my head from time to time with tasks that are hard for me. |
| 19:49.30 | StaRetji | Thx one more time for all help and patiance |
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| 19:50.52 | jmetro | <PROTECTED> |
| 19:52.37 | ledoktre | hey guys, quick question. I have two POTS lines, and was recommended by someone to use two SPA3102's to interface the lines with Asterisk. Its working, but I can't seem to chase the echo out. Anyone have any advice on either how to chase it out or if I need to replace those two boxes with what? |
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| 19:53.29 | jmetro | StaRetji: You just need to get more familiar with programming really, and it all becomes simple to figure out |
| 19:54.13 | haryv | Do you have fx0/fxs cars in the pci slots? are they old? |
| 19:54.36 | haryv | jmetro, not if they are first generation digium cards. |
| 19:54.56 | jmetro | haryv: huh |
| 19:55.05 | haryv | no mater what changed made in volvume or echo using he tools would only temperarly help |
| 19:55.37 | haryv | ledoktre, are you using a sip account ? |
| 19:57.13 | navaismo | [SySteM], just ask specific question |
| 19:57.50 | [SySteM] | Impossible to running Asterisk/Ices with Liquidsoap/Icecast |
| 19:58.02 | [SySteM] | I dont know where is the problem |
| 19:58.11 | [SySteM] | so i search anyone who ever play with that |
| 19:58.17 | ledoktre | haryv: I'm using asterisk, 5 ip550 (Polycom) phones, and the two spa3102s for incoming and outgoing local calls. Any calls greater than 1 (inbound or outbound) are routed through my voip carrier (Vitelity). |
| 19:58.38 | ledoktre | Using busy call forward for inbound calls and dial plan logic for outbound calls |
| 19:59.48 | ledoktre | All calls on Vitelity are problem free. Crystal clear, no echos. Just the local stuff. |
| 20:00.24 | jmetro | ledoktre: why are you using a gateway? |
| 20:00.44 | haryv | nver mind my original question. have mount and kernel panic issues. not looking good. |
| 20:00.47 | [TK]D-Fender | ledoktre, SPA's for PSTN use is iffy... using 2 is just plain bad |
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| 20:04.31 | ledoktre | jmetro: I had been advised to use these two boxes by a supposed "asterisk expert / consultant". I'm not happy with the results. |
| 20:04.55 | ledoktre | [TK]D-Fender: I am beginning to agree. Whats your advice? Throw them away and.... ? |
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| 20:05.44 | JustinAiken | is there a way to destroy a ConfBridge room |
| 20:05.45 | JustinAiken | ? |
| 20:05.53 | haryv | I think at this point, it would be a good idea to switch to hosting my connection then running it on the server |
| 20:08.35 | jmetro | ledoktre: who was the expert? =p |
| 20:08.41 | jmetro | haryv: whats the issue? |
| 20:09.08 | [TK]D-Fender | ledoktre, I don't recommend SPA's for PSTN use execpt for personaly / failover use, never as primary. |
| 20:09.27 | ledoktre | jmetro: an independent old school telephony guy working for a ITSP |
| 20:09.46 | jmetro | well theres the issue <.< |
| 20:10.23 | jmetro | i had an "old school" citrix guy tell me there was no way to increase the virtual memory on his server..until i showed him how and it worked. |
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| 20:11.49 | jmetro | ledoktre: What is the supposed purpose of the gateways? |
| 20:12.38 | ledoktre | [TK]D-Fender: Whats your thought then? How do you interface with the POTS lines? |
| 20:13.20 | ddickenson | Hello there, I'm trying to recreate a "hotline" or ringdown phone in dahdi. From what I've been able to read I need to make an entry in chan_dahdi that uses immediate=yes. I've done that just for one specific channel, however when I go off hook on any channel now it doesn't give dialtone to anything. All analog lines act as a hotline all of a sudden. When I comment out the lines It goes back to normal. Ideas? http://past |
| 20:13.21 | ddickenson | ebin.com/qK8KgDYk |
| 20:14.01 | [TK]D-Fender | ledoktre, My preference is Sangoma TDM cards with HWEC |
| 20:14.40 | ledoktre | jmetro: My customer required to keep their existing main two lines. Need the calls routed through their new phone system (which is working flawlessly with voip only). Any calls over 1 would get busy call forwarded to a local voip DID. Any outgoing local calls would use the local line, unless its in use, in which case the call is routed out through voip |
| 20:14.56 | ledoktre | [TK]D-Fender: Are you going to make me smile and tell me you never have any echo? :) |
| 20:14.58 | ddickenson | The desired result is to go off hook on one specific set and have it dial another phone or initate some action in the dialplan rather than just dialtone. |
| 20:15.32 | [TK]D-Fender | ddickenson, because after channel => 33 .... it will CONTINUE to affect all of channel => 49-72 because you never overrode it back |
| 20:15.48 | jmetro | ddickenson: editing chan_dahdi sounds like youre changing the whole channel to be immediate. does that sound logical to you? |
| 20:16.23 | ddickenson | [TK]D-Fender: excellent so if I just put immediate =no on the following group it should fix it? |
| 20:16.59 | ddickenson | jmetro: I do want one specific channel on this channel bank, which is one set, to be immediate |
| 20:17.05 | [TK]D-Fender | ddickenson, yes |
| 20:17.09 | ddickenson | thanks |
| 20:17.39 | jmetro | ledoktre: d-fender suggests the sangomas to everyone who walks in the door, so i'd say yes |
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| 20:29.19 | ledoktre | jmetro: Thanks. I will google them and try to figure out what they are. |
| 20:34.58 | ledoktre | [TK]D-Fender : I am reading on the sangoma site. They've got a 2 port USBFXO U100 model with HWEC. Have you ever played with this one before? Reason I'm asking is because this rig has two pci express slots and I'd have to move some stuff around if we put an actual card in. |
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| 21:35.34 | lvlinux | anybody real familiar w snom 3xx phones? I'm getting a 401 on the intial invite and I don't know why. |
| 21:36.22 | jmetro | unauthorized. |
| 21:36.28 | lvlinux | phone registers fine, and receives calls fine |
| 21:36.53 | navaismo | lvlinux, 401 is a normal message |
| 21:36.54 | [SySteM] | Impossible to running Asterisk/Ices with Liquidsoap/Icecast, anyone one make this? |
| 21:37.18 | lvlinux | when I place a call, the initial INVITE gets a 401 unauth from *, then sends another invite and it seems to go through after about 10 seconds |
| 21:37.35 | igcewieling | lvlinux: the initial invite is always not authorized, the client then sends a 2nd invite using the nonce obtained from the rejection |
| 21:37.47 | igcewieling | you have some OTHER issue |
| 21:37.58 | lvlinux | ok I didn't realize that |
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| 21:38.31 | lvlinux | when I answer the called phone, it acts like everything is ok and starts the call, but the Snom phone keeps ringing for 10 seconds. |
| 21:38.37 | lvlinux | then it gets audio |
| 21:39.34 | igcewieling | lvlinux: sounds like a NAT or direct media issue |
| 21:39.51 | igcewieling | set directmedia=no and see if that helps |
| 21:40.25 | lvlinux | I don't have NAT in this setup, but I'll try the directmedia=no. What does that actually do? |
| 21:41.17 | igcewieling | lvlinux: makes audio go through Asterisk instead of direct between the phones. direct audio sounds good, but it often isn't. |
| 21:42.39 | lvlinux | does that do the same thing as canreinvite=no? |
| 21:43.59 | [TK]D-Fender | lvlinux: directmedia replaced canreinvite in 1.6+ |
| 21:45.13 | lvlinux | oh, ok well now I get no audio at all on either phone |
| 21:46.08 | lvlinux | i got to go in a sec so I guess I'll get back on this tomorrow. |
| 21:46.19 | lvlinux | thanks for the quick help. |
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| 21:58.23 | ChannelZ | snickers |
| 22:07.01 | igcewieling | <rant>*censored* clients who insist on using their own *censored* internet service, switches, and phones. *censored* idiots blame us when someone *censorted* breaks something</rant> |
| 22:13.03 | nightrid3r | igcewieling: you have grasped the essence of customer sdirvice, grasshopper |
| 22:13.13 | nightrid3r | service |
| 22:13.56 | igcewieling | nightrid3r: the vast majority of our customers are on our internet service, with our router and for hosted or asterisk service also have our switches. |
| 22:14.26 | nightrid3r | those are the smart ones |
| 22:20.00 | navaismo | Any hints to resolve this--->http://pastebin.com/dLgkxaCd |
| 22:20.21 | navaismo | configure runs as expected but the make command fail with those errors |
| 22:20.34 | navaismo | wehn trying to compile asterisk 11.3.0 |
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| 22:20.59 | jmetro | navaismo: i think youre missing some dependencies. |
| 22:21.01 | jmetro | thats my input. |
| 22:22.20 | navaismo | if so why configure ran fine? |
| 22:23.29 | jmetro | im not sure, but it looks like there is a check going on that it is missing, thats just my instinct telling me youre missing dependencies |
| 22:24.04 | navaismo | looking for libavcodec |
| 22:24.20 | jmetro | http://www.lavrsen.dk/foswiki/bin/view/Motion/BugReport2005x11x08x140111 |
| 22:24.53 | jmetro | anyway i have to go eat sleep and wake up and come back to work! |
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| 22:26.16 | navaismo | GN |
| 22:27.12 | cHarNe2 | hi guys, i got instructions to run a couple of commands this night, and asteriskt aint my general field of expertise. 'asterisk -rx "stop now"' => no such command |
| 22:27.40 | navaismo | core stop now<---that will stop your asterisk |
| 22:28.14 | cHarNe2 | thanks! |
| 22:31.07 | navaismo | jmetro, removed ffmpeg-devel and seems its working...gosh I Mega-F***ed my system |
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