| 00:00.15 | Kobaz | I don't know what astlinux sets up, so if you delete them and your asterisk is broken then, well... apparently you needed them |
| 00:00.30 | Kobaz | safest thing is just mv them somewhere |
| 00:00.37 | sktyau | mv extensions.ael extensions.ael.bak :) |
| 00:00.40 | lvlinux | well that brings up a related question--- do I need the ael-dundi-e164-blah-blah contexts and such? |
| 00:00.48 | lvlinux | yes that is what I would have done - just rename. |
| 00:00.52 | Kobaz | lvlinux: depends what you are doing |
| 00:01.42 | lvlinux | I'm trying to keep my dialplan simple, just the stuff I know what it is and what it does. I have one PSTN line, an IAX trunk, and a couple SIP trunks. |
| 00:02.40 | Kobaz | in regards to core asterisk the only things really required is asterisk.conf and modules.conf |
| 00:02.53 | Kobaz | everything else is ancillary |
| 00:03.13 | lvlinux | really? that little? cool. I didn't know you could even get it that small (not that I would want to). |
| 00:03.40 | lvlinux | So is it safe to assume that if I don't know what all those special contexts are doing, I don't need them? |
| 00:03.46 | Kobaz | yeah |
| 00:04.00 | Kobaz | if you wrote all the stuff you need in extensions.conf |
| 00:04.05 | Kobaz | and that's the only thing you're using |
| 00:04.09 | Kobaz | then yeah everything else can go away |
| 00:04.24 | Kobaz | i would highly recommend taking a look at ael though |
| 00:04.28 | lvlinux | k good - that's what I figured but just wanted to make sure. |
| 00:04.30 | lvlinux | why? |
| 00:04.32 | Kobaz | it will make your dialplan writing much more sane |
| 00:05.04 | lvlinux | how so? I did look at it - didn't look hard to understand. |
| 00:05.06 | Kobaz | it's like programming in oldschool BASIC versus python |
| 00:05.15 | lvlinux | oh ok that makes sense |
| 00:05.20 | Kobaz | extensions.conf is flat and unstructured |
| 00:05.33 | lvlinux | with GOTO :-) |
| 00:05.35 | Kobaz | but you should understand how extensions.conf works before you really dig into ael |
| 00:05.37 | Kobaz | hah yeah exactly |
| 00:05.42 | sktyau | Horses for courses. Give me extensions.conf + PHPAGI over AEL anyway. |
| 00:05.50 | Kobaz | well yeah |
| 00:06.05 | sktyau | *anyday |
| 00:06.08 | Kobaz | all your 'business logic' type stuff could be done in FastAGI with the language of your choice |
| 00:06.23 | Kobaz | but you do need dialplan to tie things together |
| 00:06.26 | Kobaz | so why not make it pretty |
| 00:06.43 | lvlinux | So is there anything that ael can do that the standard extensions.conf cannot, or is it just a matter of style/cleanness? |
| 00:06.51 | Kobaz | style |
| 00:06.58 | sktyau | Because it keeps the Developers in the office out of the Voice boxes :-) |
| 00:07.05 | Kobaz | ael is processed into extensions.conf |
| 00:07.07 | Kobaz | and then executed |
| 00:07.13 | Kobaz | it's a transpiler basically |
| 00:07.35 | keyy1 | anyone have experience with sms handling? |
| 00:07.38 | lvlinux | you mean asterisk translates ael into the extensions.conf format before actually loading the dialplan? |
| 00:07.43 | Kobaz | yeah |
| 00:08.09 | Kobaz | which is nice in a way, it doesn't have to parse the code when it's running |
| 00:08.20 | Kobaz | just on dialplan reload |
| 00:08.59 | lvlinux | So extensions.conf is the actual way asterisk understands the dialplan? do you know of a docs site or tutorials for ael that I might could check out? (off hand--otherwise I'll google :-) |
| 00:09.54 | Kobaz | voip-info wiki has some stuff |
| 00:09.57 | Kobaz | look at ael2 |
| 00:10.08 | Kobaz | basically the stuff on the wiki is about all you need |
| 00:10.27 | Kobaz | there's some oddities here and there but they should be obvious when you run into them |
| 00:10.49 | Kobaz | trying to think of one offhand but i'm blanking |
| 00:11.13 | lvlinux | ok thanks for the info. I'll appreciate it - trying to keep this box as streamlined as possible so it'll be worth me looking into i think. Thanks |
| 00:11.28 | Kobaz | i find i can write ael like 10x faster than extensions.conf |
| 00:11.30 | lvlinux | I mean I do appreciate it not "I'll appreciate it" lol |
| 00:11.54 | Kobaz | and i'm sure someone is going to step in here and say ael sucks blah blah |
| 00:11.54 | Kobaz | but |
| 00:12.03 | Kobaz | i see no reason not to use it |
| 00:13.07 | Kobaz | like you can do "cool" stuff like: foo=1+2+3; |
| 00:13.12 | Kobaz | where in dialplan you have to do |
| 00:13.20 | Kobaz | Set(foo=$[1+2+3]); |
| 00:13.23 | Kobaz | in order to do math |
| 00:13.30 | lvlinux | what about extensions.lua? ever use that one? |
| 00:13.34 | Kobaz | lua is eh |
| 00:13.45 | Kobaz | it doesn't have as tight of an integration into dialplan as i would have hoped |
| 00:13.55 | Kobaz | and, it runs lua entirely in its own space |
| 00:14.07 | Kobaz | so you can't use the built in console verbosity to see what's running |
| 00:14.15 | lvlinux | ah, ok that doesn't interest me then... |
| 00:14.16 | Kobaz | might as well use AGI/FastAGI |
| 00:14.28 | Kobaz | lua as a language is nice |
| 00:14.36 | Kobaz | but it's not tightly bound into the dialplan |
| 00:14.43 | Kobaz | it's just like, hand off control to lua, kthanks |
| 00:15.22 | lvlinux | nope - that sounds like the opposite of what i want lol |
| 00:17.39 | Kobaz | http://www.propublica.org/article/why-it-may-be-illegal-to-instagram-your-ballot |
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| 00:55.47 | leifmadsen | ya, basically FastAGI is the future :D |
| 00:56.12 | leifmadsen | if you want tight code integration, AGI is the method. Otherwise, it's all dialplan |
| 00:56.18 | leifmadsen | everything else is just a dialplan abstraction layer |
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| 01:10.58 | sktyau | I don't know whats giving me more 404's, Asterisk or Up to date documentation on peer matching hurrrrr |
| 01:12.15 | tzanger | heh |
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| 02:15.04 | SeRi | quite around here |
| 02:18.59 | unicron | i voted |
| 02:19.39 | jpsharp | Im house hunting.\ |
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| 02:37.31 | carrar | jpsharp, which country? |
| 02:37.46 | jpsharp | US. |
| 02:38.31 | carrar | America is the single greatest nation that God ever gave man on this earth. |
| 02:38.35 | carrar | heh |
| 02:38.56 | carrar | I stole that from hannity |
| 02:42.14 | pabelanger | Canada, America's hat |
| 02:42.34 | carrar | And South America? and Mexcia? |
| 02:42.49 | pabelanger | No, just America |
| 02:51.09 | dijib | SeRi: ka pasa? |
| 02:51.27 | dijib | dijib: <-party animal right here |
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| 05:34.19 | deo | hello everyone... just want to ask your advice.. is it advisable to have an all in package of asterisk like trixbox or elastix? or are you going to suggest install asterisk from source??? |
| 05:35.15 | deo | and configure purely on the terminal.. not in the gui of those packages like trixbox or elastix... |
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| 05:37.08 | WIMPy | GUIs may help you with your first steps but will restrict your possibilities later on. |
| 05:37.43 | WIMPy | And because of the rather intransparent way they work, they are not supported here. |
| 05:39.10 | deo | yup.. i know.. just want to ask if it is advisable to use those gui packages... |
| 05:39.17 | deo | thanks WIMPy |
| 05:40.07 | WIMPy | If you just want some quick results, yes. If you want more, most probably not. |
| 05:40.25 | deo | thanks... |
| 05:41.40 | deo | im planning to to install asterisk servers on 3 locations.. i probably will install that gui package on the main site.. the install asterisk on source on remaining two locations.. |
| 05:41.50 | deo | *then |
| 05:42.16 | deo | hopefully i can connect them both... |
| 05:42.22 | WIMPy | At least you wil know the difference then. |
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| 06:55.08 | wdoekes | WIMPy: then you were probably seeing 481s from your peer, not 408 |
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| 07:41.46 | WIMPy | wdoekes: I can't remember, but I certainly saw the the message about the non-existing call leg. |
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| 07:48.06 | wdoekes | quite possible.. if it was a 408 in-dialog, it'd probably be worthy of a notice/warning too.. but a different one |
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| 08:05.02 | Rico29 | hi all |
| 08:05.14 | Rico29 | http://pastebin.com/dZc33abT |
| 08:05.23 | Rico29 | still have my problem with saynumber() |
| 08:05.42 | roswell | hi everyone. assuming i have 2 ip addresses on one eth interface (primary and alias) on my asterisk box, is it possible to point asterisk to route outgoing traffic accordingly from each address it has been received on, instead from primary? |
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| 08:16.55 | Rico29 | hi leifmadsen |
| 08:17.31 | Rico29 | if you have time to spend, can you take a look at this please ? http://pastebin.com/dZc33abT |
| 08:17.43 | Rico29 | trying to use saynumber but doesn't work as expected |
| 08:18.09 | wdoekes | Rico29: you left an impression yesterday |
| 08:18.52 | ectospasm | Rico29: how do you expect it to work? And exactly how is it working (or not working, as it were)? |
| 08:18.54 | wdoekes | 18:28 < [TK]D-Fender> Rico29, You are showing things in bits and pieces. and only rarely at all |
| 08:18.58 | wdoekes | 18:30 < Rico29> good afternoon all ! |
| 08:19.00 | wdoekes | 18:30 < [TK]D-Fender> ... |
| 08:19.03 | wdoekes | 18:32 < beardy> Always interesting how they expect people helping them for free on their free time if they go home from a paying job in the middle of it eh. |
| 08:20.17 | Rico29 | wdoekes> do you want me to pastebin all the conversation, because pieces are missing here... |
| 08:20.21 | kaldemar | Rico29: also you should probably tell what you expect. |
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| 08:21.00 | Rico29 | <Rico29> ok, will pastebin it |
| 08:21.00 | oyugik | Hey guys how can I reduce the noise on asterisk |
| 08:21.00 | Rico29 | <Rico29> ... tomorrow |
| 08:21.00 | Rico29 | <Rico29> time to go home (in france) |
| 08:21.13 | kaldemar | turns channel EC on and shuts up |
| 08:21.43 | kaldemar | oyugik: what noise? |
| 08:22.03 | Rico29 | kaldemar> ectospasm >when I use "saynumber(0123456789)", it tells me 123 millions etc... and I want it to say "zero one twenty-three fourty-five..." |
| 08:22.21 | ectospasm | Rico29: saydigits() then |
| 08:22.34 | oyugik | @kaldemar there is a shhhhhh sound that I hear whever I make a call |
| 08:22.38 | ectospasm | or saydigit() |
| 08:23.13 | Rico29 | ectospasm> saidigits will tell me : zero one two three four five ... |
| 08:23.23 | Rico29 | that's not what I want |
| 08:23.36 | ectospasm | Rico29: so: SayDigit(01), SayNumber(23), SayNumber(45)... |
| 08:23.38 | Rico29 | what I want is what I wrote in say.conf : _pho[n]e:0[1-9]XXXXXXXX => num:${SAY:0:1}, num:${SAY:1:1}, num:${SAY:2:2}, num:${SAY:4:2}, num:${SAY:6:2}, num:${SAY:8:2} |
| 08:23.44 | kaldemar | oyugik: what kind of a call? what devices are you using? what comes up in CLI when you make a call? when exactly do you hear the sound? |
| 08:25.12 | Rico29 | ectospasm> I don't want to do that. I want to pass the entiere number. (I'm trying to debug why callerid num is not announced as expected in voicemail) |
| 08:25.14 | oyugik | when I make an outbound call there is some noise. Much like when the sound you hear when you are changing channels on you radio ... |
| 08:25.40 | Rico29 | so I just want to know why my number is not understood by say.conf |
| 08:25.51 | Rico29 | and "played" as it's mritten in |
| 08:25.55 | ectospasm | Rico29: then SayDigit should be good enough |
| 08:25.58 | Rico29 | s/mritten/written/ |
| 08:26.41 | kaldemar | oyugik: you did not answer any of the questions i asked you. |
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| 08:31.18 | Rico29 | ectospasm> tried yesterday. As I sayed sooner : saydigits will tell me "zero one two three four five..." |
| 08:31.19 | Rico29 | http://pastebin.com/B1yDMm8z |
| 08:31.54 | ectospasm | Rico29: what's the difference between zero one two three and zero one twenty-three, forty-five, etc...? |
| 08:31.54 | Rico29 | I want : "zero one twenty-three fourty-five sixty..." |
| 08:32.09 | Rico29 | ectospasm> try to say it and you will understand |
| 08:32.18 | ectospasm | Rico29: why? For your purposes that sounds like adding extra complexity unnecessarily |
| 08:32.33 | Rico29 | no, because in rance, number are sayed like this |
| 08:32.36 | Rico29 | France |
| 08:32.52 | oyugik | @kaldemar here is the cli output http://pastebin.com/fCVYGbff |
| 08:33.29 | oyugik | I am using a TDM400P |
| 08:33.35 | ectospasm | Rico29: is it just you that's consuming this spoken digit string? |
| 08:33.46 | oyugik | I get the noise whenever I make an outbound call |
| 08:34.09 | oyugik | @kaldemar have I answered your question? |
| 08:34.27 | kaldemar | oyugik: when _exactly_ do you hear the noise? |
| 08:34.52 | Rico29 | ectospasm> did not understood your question, sorry |
| 08:35.05 | ectospasm | Rico29: who is this spoken digit string for? |
| 08:35.34 | Rico29 | for custommer. Actually my voicemail system is not announcing callerid number correctly |
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| 08:36.10 | Rico29 | I'm just trying to understand why my say.conf file is not understood |
| 08:36.20 | Rico29 | and used |
| 08:36.28 | oyugik | @kaldemar when I make a call from a sip phone in my case (linphone) - when a channel is engaged and the number am calling is ringing I hear alot of noise. |
| 08:36.35 | ectospasm | Rico29: you're better off splitting the number into the chunks you want, and running SayNumber on the ones you want (like "twenty-three", "forty-five", etc.) |
| 08:37.00 | Rico29 | ectospasm> can't do that in voicemail |
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| 08:37.50 | Rico29 | in voicemail I just have "saycid=yes" or "saycid="now" |
| 08:38.01 | ectospasm | Rico29: I still don't see why saying the entire string of single digits isn't good enough |
| 08:38.33 | ectospasm | since breaking it into "vingt-et-un" just sounds sillly |
| 08:38.36 | ectospasm | s/lll/ll/ |
| 08:38.52 | Rico29 | ectospasm> because in france, "zero one two three four five six seven eight nine" is not understood as a phone number |
| 08:39.26 | Rico29 | and "ero one, twenty-three ,fourty-five, sixty-seven, eighty-nine" is |
| 08:39.35 | Rico29 | understood as a phone number |
| 08:39.46 | oyugik | @kaldemar have you seen the pastebin? |
| 08:39.58 | kaldemar | oyugik: does the call sound fine otherwise? |
| 08:40.04 | Rico29 | isn't say.conf file made for things like i'm trying to do ? |
| 08:40.19 | ectospasm | Rico29: yeah, I don't think the voicemail app will do it that way |
| 08:40.43 | Rico29 | ok, will see that later and report a bug if needed |
| 08:40.54 | oyugik | @kaldemar the call has a background noise |
| 08:41.05 | Rico29 | but for now, I just want to make my saynumber morking as expected (with say.conf schemes) |
| 08:41.13 | Rico29 | s/morling/working/ |
| 08:41.20 | Rico29 | sh$t |
| 08:41.26 | Rico29 | s/morking/working/ |
| 08:41.36 | ectospasm | too late |
| 08:41.40 | Rico29 | :) |
| 08:42.15 | Rico29 | so do you have any idea about "why is it not working ?" ectospasm ? |
| 08:42.37 | ectospasm | I don't deal with say.conf much at all |
| 08:42.50 | Rico29 | ok |
| 08:42.59 | Rico29 | does anybody else do ? |
| 08:43.15 | oyugik | kaldemar: what config should I modify to reduce the noise |
| 08:44.25 | kaldemar | oyugik: well there is not an option anywhere to reduce unwanted noise. you'd have to make sure where it comes from. does the linphone-asterisk leg work as expected? try app echo to see if it is noise free. |
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| 08:52.56 | oyugik | @kaldemar: when it comes to the software echo cancellation which is the best one to use? kb1, mg2, hpec, sec, sec2? |
| 08:54.15 | ectospasm | oyugik: it depends on the situation. One may not necessarily work better in all situations... you've got to try each and see which one works best for you |
| 08:54.23 | ectospasm | ...and keep in mind that may change |
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| 08:54.54 | kaldemar | oyugik: according digium, hpec is superior, but you can be the judge of that. |
| 08:56.18 | ectospasm | I wouldn't even say HPEC is superior... just uses the same algorithm as the Digium hardware echo cans... and I'm not even sure if that's necessarily true anymore |
| 08:59.41 | oyugik | @ectospasm : what would be the best to use then? |
| 09:00.13 | ectospasm | oyugik: did you not understand what I said? You'll need to test each one and use the one that best works for you |
| 09:00.49 | ectospasm | There is no "best" necessarily, because some situations will find e.g. mg2 that works best, others kb1, still others HPEC... |
| 09:01.02 | ectospasm | Granted, HPEC will require a license from Digium |
| 09:01.20 | ectospasm | if you have a registered Digium analog card, you can get a free HPEC license |
| 09:02.52 | x2s | Hi. When I'm in the hangup handler in the dialplan and I do some stuff, what can I call, if I wanna stop in it? Normally you call Hangup(), but this feels a bit odd in the hangup handler... |
| 09:04.11 | ectospasm | x2s: not wrong... if you call Hangup() processing in the h exten will stop |
| 09:04.25 | x2s | then it just looks odd and is totally ok. Thanks :) |
| 09:04.34 | ectospasm | ...Asterisk is smart enough not to get caught in a loop |
| 09:04.48 | oyugik | @ectospasm : what would be major cause of noise when making outbound calls? Is it an internal network issue or is there a tweak on the settings I am not getting right |
| 09:05.05 | ectospasm | not a tweak of settings |
| 09:05.22 | ectospasm | oyugik: do inbound calls over the same interface give you the same noise? |
| 09:09.38 | oyugik | ectospasm : inbound calls are abit clearer, though the noise still exists |
| 09:10.09 | ectospasm | oyugik: analog trunks? |
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| 09:18.26 | kaldemar | Rico29: btw, your issue is that the leading zero gets stripped from the number before the number gets to the say matching. pbx_builtin_saynumber uses atoi() on the given number, which does it. |
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| 09:20.23 | Rico29 | kaldemar> thanks ! is there a way to avoid the stripping of the leading zero ? |
| 09:20.54 | oyugik | @ectospasm : what is the the best way to go about it |
| 09:21.16 | ectospasm | oyugik: you did not answer the question... is this an analog trunk? |
| 09:21.59 | *** join/#asterisk BorjaGVO (d51beb92@gateway/web/freenode/ip.213.27.235.146) |
| 09:22.28 | Rico29 | kaldemar> you've find my problem source |
| 09:22.30 | Rico29 | thanks a lot |
| 09:23.00 | kaldemar | Rico29: leading zeroes are not considered as a part of a number and atoi is not a part of asterisk, so... |
| 09:23.38 | kaldemar | you'd have to modify sources to go around that. |
| 09:24.04 | kaldemar | maybe another say app could be used. |
| 09:24.21 | Rico29 | kaldemar> it's in voicemail, so... |
| 09:26.21 | *** join/#asterisk oyugik (~oyugik@41.212.110.90) |
| 09:30.04 | Rico29 | kaldemar> maybe UI've foudn a workaround |
| 09:30.16 | Rico29 | <PROTECTED> |
| 09:31.00 | ectospasm | oyugik: you did not answer the question... is this an analog trunk? |
| 09:35.06 | kaldemar | Rico29: the zero never gets to that point. |
| 09:35.07 | Rico29 | now that saynomber is working well, and voicemail callerid nos announced correctly, I think I can open a case on digium bugtracker |
| 09:35.30 | Rico29 | kaldemar> so why is it working ? |
| 09:35.43 | kaldemar | Rico29: what is working? |
| 09:36.07 | kaldemar | <PROTECTED> |
| 09:36.40 | Rico29 | my saynumber(0123456709) is saying "zero one twenty-three fourty-five sixty-seven zero nine" |
| 09:40.14 | Rico29 | kaldemar> http://pastebin.com/J9ytPhMV |
| 09:42.51 | kaldemar | Rico29: that says a leading zero for every 9-digit number, whether there is one or not. |
| 09:43.58 | Rico29 | yes |
| 09:44.24 | Rico29 | hope I will not have to say too many 9-digits numbers :p |
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| 09:45.57 | *** join/#asterisk oyugik (~oyugik@41.212.110.90) |
| 09:47.59 | bombev | hi |
| 09:48.04 | bombev | Asked to transmit frame type ulaw, while native formats is 0x100 (g729) read/write = 0x4 (ulaw)/0x4 (ulaw) |
| 09:58.57 | Rico29 | kaldemar> another interesting thing : exten => _99.,n,Playback(phone:${EXTEN:2},say) plays the number as expected |
| 09:59.12 | Rico29 | with line : _pho[n]e:0XXXXXXXXX => digits/0, num:${SAY:1:1}, num:${SAY:2:2}, num:${SAY:4:2}, num:${SAY:6:2}, num:${SAY:8:2} |
| 09:59.15 | Rico29 | in say.conf |
| 09:59.37 | Rico29 | so with playback, leading 0 is not stripped ? |
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| 10:07.51 | kaldemar | Rico29: SayPhonetic != SayNumber |
| 10:08.48 | Rico29 | ok, so playback uses sayphonetic ? |
| 10:09.01 | kaldemar | if you tell it to, like you did. |
| 10:09.59 | Rico29 | and is there a way to tell voicemail to use sayphonetic ? |
| 10:10.11 | kaldemar | dig in the sources and see. |
| 10:10.19 | Rico29 | ok |
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| 10:23.58 | Rico29 | kaldemar> I just saw an option in voicemail.conf called "cidinternalcontexts=<context>" |
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| 10:24.49 | Rico29 | will try with this |
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| 10:30.14 | Rico29 | kaldemar> posted a new improvement request : ASTERISK-20657 |
| 10:30.15 | mcolombo | hi all! |
| 10:30.43 | mcolombo | anybody say how to add +39 in the from header? |
| 10:31.30 | Rico29 | set(${CALLERDI(num)}=+39${CALLERID(num)}, something like this ? |
| 10:35.55 | mcolombo | i have try with this : Set(CALLERID(num)=+39${CALLERID(num)} |
| 10:35.57 | mcolombo | but does not work |
| 10:36.32 | kaldemar | mcolombo: show what you have and what you see. |
| 10:38.40 | Rico29 | quick test : exten => 998898,1,Set(CALLERID(num)=+33${CALLERID(num)}) works for me |
| 10:39.40 | mcolombo | i have this exten => _X.,n,Set(CALLERID(num)=+39${CALLERID(num)}) |
| 10:40.12 | mcolombo | but in the sip invite, the from header is withoud +39 |
| 10:40.25 | kaldemar | mcolombo: pastebin CLI output with sip debug, not a single line from dialplan. |
| 10:41.47 | mcolombo | this is the sip debug |
| 10:41.48 | mcolombo | http://pastebin.com/wnr76Psc |
| 10:42.36 | kaldemar | that's not sip debug. sip debug is enabled with "sip set debug on". |
| 10:44.34 | mcolombo | ok, one second |
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| 10:48.06 | mcolombo | kaldemar, thanks i resolv my problem! |
| 10:48.07 | mcolombo | thanks |
| 10:50.01 | Rico29 | and what was the cause ? |
| 10:51.17 | mcolombo | Rico29, thanks, i insert an error in the string |
| 10:52.13 | Rico29 | ok |
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| 10:58.29 | *** join/#asterisk Core-NET (~Core@82.178.185.122) |
| 10:59.00 | Core-NET | hey.. |
| 10:59.21 | Core-NET | anyone tested Asterisk with i7 CPUs |
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| 11:13.07 | madduck | is it possible to host virtual domains with asterisk? e.g. calls to sip:foo@example.org should go to foo in the example-org context and sip:foo@example.com should look up foo in the example-com context? |
| 11:17.34 | kaldemar | madduck: there is no destination-based matching on the chan_sip side, you'd have to do that in dialplan. |
| 11:19.40 | *** join/#asterisk mirela666 (~Thunderbi@212.200.146.253) |
| 11:19.54 | kaldemar | madduck: i'll take that back. see "SIP DOMAIN SUPPORT" in the sample sip.conf. |
| 11:20.26 | *** join/#asterisk mathi (Matthew@ip-62-235-244-174.dsl.scarlet.be) |
| 11:20.27 | mathi | hi |
| 11:22.20 | mathi | the number of the logs displayed on the console is limited, I need to see a log of one hour ago, but I can't scroll up until there |
| 11:22.31 | mathi | any workaround?? |
| 11:22.43 | madduck | kaldemar: gosh, I can't believe I missed that… sorry, but thanks! |
| 11:25.24 | bulkorok | mathi: check /var/log/asterisk/messages |
| 11:25.57 | mathi | bulkorok, I have no directory messages |
| 11:26.15 | bulkorok | it's a file |
| 11:27.08 | mathi | no file messages |
| 11:27.33 | kaldemar | you can't see what does not exist. |
| 11:28.13 | bulkorok | the I have no idea how you can check your CLI |
| 11:28.17 | madduck | am I correct that if I have two separate asterisks, I can still let them peer with each other and be able to transfer calls between them? |
| 11:28.27 | mathi | there is only 3 directories, cdr-csv, cdr-custom cel-custom |
| 11:28.33 | mathi | bulkorok, ^ |
| 11:28.34 | *** join/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
| 11:28.38 | AliRezaTaleghani | hi all |
| 11:29.11 | AliRezaTaleghani | I am looking to authenticate my SIP clients with theire ActiveDirectory cridential.... |
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| 11:29.34 | mathi | bulkorok, I launched asterisk with sudo asterisk -rv |
| 11:29.47 | bulkorok | mathi: well in those directories are the CDR datas... but in the directory /var/log/asterisk should be "messages" |
| 11:30.03 | mathi | bulkorok, I don't have it:( |
| 11:30.20 | mathi | I have asterisk 11 by the way |
| 11:30.57 | bulkorok | mmh... I don't use 11 yte... but in logger.conf should be enabled messages => notice,warning,error or some more if you need |
| 11:31.06 | bulkorok | s/yte/yet |
| 11:32.30 | mathi | bulkorok, I am not using logger.conf |
| 11:32.43 | bulkorok | ah... thats why no messages :-) |
| 11:32.44 | mathi | but I do have the logs on the console, and I want to access earlier logs |
| 11:32.59 | bulkorok | shift + page up usually |
| 11:34.26 | bulkorok | if you use putty you have to change settings in window => lines of scrollback, but that works only for future lines |
| 11:34.53 | mathi | bulkorok, it seems that those logs are definitely lost? :'-( |
| 11:35.02 | mathi | I just need to go up 100 lines :/ |
| 11:35.21 | bulkorok | if it's not there it's not there :-/ |
| 11:36.14 | bulkorok | so... maybe you check logger.conf |
| 11:36.21 | bulkorok | for the future |
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| 11:47.14 | oyugik | How can one reduce noise on active calls? |
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| 12:09.59 | kjs | argh, asterisk vociemails are full... can i just delete them out of /var/spool ?\ |
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| 12:21.16 | Scottyob | Howdy. Wondering if there's a free plugin for speech recognition anyone knows of? |
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| 12:39.35 | madduck | what is this trying to tell me? |
| 12:39.39 | madduck | "No application 'VoicemailMain' for extension (lehel-martin, 870, 1)" |
| 12:40.19 | madduck | I do have "exten => _87X.,1,VoicemailMain(${EXTEN:2}@mycontext)" in that dialplan context… |
| 12:40.28 | madduck | and mycontext is a voicemail context… |
| 12:40.56 | mathi | bulkorok, is it possible to create different files messages1, messages2, etc. ? |
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| 12:59.38 | x2s | Hi. I'm registering my asterisk server at another sip pbx. I've set the extension fixed for incoming calls, now I'm looking for something to set the context. Is there a bit of documentation how to do that? |
| 13:00.04 | x2s | the register setting has sadly just an option for the extension... |
| 13:01.23 | kaldemar | ~book |
| 13:01.23 | 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 |
| 13:01.59 | kaldemar | you need a matching peer whose context parameter defines it. |
| 13:04.33 | x2s | Found my error then. Somehow I typed type=user instead of type=peer |
| 13:04.52 | Rico29 | <PROTECTED> |
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| 14:02.50 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 14:03.03 | _zoom_ | how to config my asterisk as trunk provider? |
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| 14:19.37 | [TK]D-Fender | _zoom_, As in? Almost every install could count as that depending on your point of view |
| 14:20.02 | [TK]D-Fender | _zoom_, Call comes in, call goes out. That's every PBX out there. |
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| 14:45.27 | jmetro | call goes in, call gets held up with old friends, call stays a while. |
| 14:45.33 | *** join/#asterisk fredericve (~fes@host-212-68-194-46.brutele.be) |
| 14:45.59 | fredericve | what's everyone's opinion on astcanary? use it or not? |
| 14:56.07 | Maliuta | fredericve: depends? are you going down into and astmine? ;) |
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| 15:03.43 | *** join/#asterisk markit (~marco@151.78.74.112) |
| 15:04.50 | markit | hi, is there a program I can run from the shell I can use to detect the kind of NAT I'm behind? |
| 15:05.31 | Maliuta | kind of NAT? I don't know that there is more than one kind on NAT |
| 15:06.11 | Maliuta | There are plenty of sites that can tell you the IP of your browser vs. the IP that presents itself to the webserver. |
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| 15:07.51 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 15:10.39 | markit | Maliuta: I mean Full-cone, restricted cone, etc. and would love to know from command line |
| 15:11.03 | markit | I don't need public IP, just the type of nat |
| 15:11.22 | markit | (or of nat traversal) |
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| 15:16.40 | *** join/#asterisk madduck (~madduck@debian/developer/madduck) |
| 15:18.48 | madduck | what file provides the VoicemailMain application? |
| 15:19.37 | glaz | voicemail.conf ? |
| 15:19.57 | madduck | no, i mean module |
| 15:20.06 | madduck | /usr/lib/asterisk/modules/app_minivm.so ? |
| 15:20.15 | *** join/#asterisk AdvoWork (~AdvoWork@unaffiliated/advowork) |
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| 15:22.14 | [TK]D-Fender | app_voicemail.so |
| 15:22.39 | [TK]D-Fender | Or the equivalent version for DB/IMAP/etc depending on which you're running |
| 15:23.16 | madduck | oh, I should have read README.Debian. :( |
| 15:23.44 | SeRi | another day at work. |
| 15:27.35 | AdvoWork | Hi there, im trying to perform an outgoing sip test, and it needs to be anonymous. from a test, we are sending sip:Anonymous@anonymous@invalid, but in RFC3325 they are all defined as sip:anonymous@anonymous@invalid - note the capital A. Can I change this somehow? |
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| 15:29.30 | [TK]D-Fender | AdvoWork, You should probably be showing us exactly what you are doing... |
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| 15:32.35 | kaldemar | AdvoWork: the rfc defines it as anonymous@anonymous.invalid, not anonymous@anonymous@invalid. |
| 15:33.10 | AdvoWork | kaldemar, ahh yeah, sorry thats my typo too, so I need to change @invalid to .invalid as well as A to a |
| 15:34.08 | kaldemar | the Anonymous comes from CALLERID_UNKNOWN channels/sip/include/sip.h. |
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| 15:37.00 | kaldemar | in 11 the whole string was changed from "Anonymous" <Anonymous@anonymous.invalid> to "Anonymous" <anonymous@anonymous.invalid> |
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| 15:38.21 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 15:41.15 | AdvoWork | kaldemar, ive got a few sip.h files and ive checked all but dont see anything relating to Anonymous |
| 15:45.56 | kaldemar | i guess you're using a pre 1.8 version then. |
| 15:46.51 | *** join/#asterisk navaismo (~navaismo@189.241.62.150) |
| 15:47.30 | kaldemar | in those it is defined in chan_sip.c. the real reason for the capital A in Anonymous@anonymous.invalid is in chan_sip.c anyway. |
| 15:51.41 | kaldemar | quite odd to change it with a hardcoded value though, when the rest is defined in the header. |
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| 15:57.34 | AdvoWork | kaldemar, i can only see one chan_sip.c but it shows: #define CALLERID_UNKNOWN "Anonymous" #define FROMDOMAIN_INVALID @anonymous.invalid" ie commented out anyway? |
| 15:58.14 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.9.207) |
| 15:58.34 | kaldemar | AdvoWork: look how CALLERID_UNKNOWN is used. |
| 15:59.20 | kaldemar | AdvoWork: and # is not a comment character in C. |
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| 16:04.37 | AdvoWork | kaldemar, ahh ok, how do you mean how its used, in that file? its set as: #define CALLERID_UNKNOWN "Anonymous" which I assume would be anonymous, and then change: #define FROMDOMAIN_INVALID "anonymous.invalid" to anonymous@invalid. would that suffice? |
| 16:06.18 | kaldemar | AdvoWork: not at all. don't touch those. search for CALLERID_UNKNOWN elsewhere in the file. |
| 16:08.41 | kaldemar | proper fix would be to add #define FROMUSER_INVALID "anonymous" and then change l = FROMUSER_INVALID and n = CALLERID_UNKNOWN later in the file when those definitions are actually used. |
| 16:15.18 | AdvoWork | kaldemar, yeah further down it says: l = CALLERID_UNKNOWN; and then: n = l; and then:d = FROMDOMAIN_INVALID; but if i were to defined FROMUSER_INVALID "anonymous" and then set l = FROMUSER_INVALID; and then n = CALLERID_UNKNOWN; would that fix it then, and a restart of asterisk or? |
| 16:15.59 | kaldemar | you'd have to recompile and reinstall to get those changes in to the module. |
| 16:16.07 | AdvoWork | oh |
| 16:16.14 | AdvoWork | is there any way that doesnt involve that? |
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| 16:16.52 | leifmadsen | changes to code require a recompile |
| 16:16.56 | leifmadsen | for them to be active |
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| 16:17.28 | AdvoWork | hmm, ive seen Remote party identity which may work, but im not completey sure |
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| 16:24.12 | fredericve | what's everyone's opinion on astcanary? use it or not? |
| 16:26.43 | wdoekes | never tried it.. I have separate scripts that check sip responsiveness |
| 16:29.00 | wdoekes | I see it serves a purpose when running asterisk with realtime prio, which I don't anyway |
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| 16:32.58 | *** join/#asterisk TimeRider (~steve@188-220-34-144.dsl.cnl.uk.net) |
| 16:38.52 | fredericve | another question. I have users that speak different languages on the same box. When they receive voicemail, asterisk sends out and e-mail. Any idea how I can send that mail in a specific language depending on the channel language? |
| 16:43.39 | *** join/#asterisk tonikasch (~tonikasch@unaffiliated/tonikasch) |
| 16:46.11 | *** join/#asterisk Conker (~Conker@static-207-54-111-107.ptr.terago.net) |
| 16:48.59 | Conker | I've just setup asterisk with some minimal config and I can see 'sip show peers' shows status ok, the ip, etc but whenever i try to make a call i receive a 'security check failed' message |
| 16:51.52 | *** join/#asterisk vince_ (~quassel@2620:8b:0:f065:533:feb2:94e9:9c67) |
| 16:54.01 | Conker | Any ideas? |
| 16:54.07 | [TK]D-Fender | ~pb |
| 16:54.08 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 16:54.18 | [TK]D-Fender | Conker, ^^ show us the call with SIP debug enabled |
| 16:55.04 | Conker | [TK]D-Fender: just tried enabling that, logger set level DEBUG ? |
| 16:55.10 | vince_ | anyone have experience with chan_sccp? |
| 16:55.57 | Conker | [TK]D-Fender: sorry nvm |
| 17:00.44 | *** join/#asterisk cyford (~allen@c-76-97-235-69.hsd1.ga.comcast.net) |
| 17:03.13 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-aqydqcgljziadzow) |
| 17:03.13 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 17:04.20 | *** join/#asterisk ujjain (~ujjain@unaffiliated/ujjain) |
| 17:04.49 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 17:09.28 | jmetro | dial_exec_full..wat |
| 17:11.48 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200::42:4) |
| 17:11.48 | gusto | so |
| 17:12.01 | *** part/#asterisk tonikasch (~tonikasch@unaffiliated/tonikasch) |
| 17:13.00 | *** join/#asterisk Conker (~Conker@static-207-54-111-107.ptr.terago.net) |
| 17:13.23 | Conker | ok, lets try that again and hope ekiga doesn't go defunct |
| 17:15.37 | jmetro | so is there a fix for "dial exec full - unable to create channel of type sip - cause 20 unknown" or are there too many variables |
| 17:16.26 | WIMPy | jmetro: That's just the name of the function. There is nothing full. |
| 17:16.49 | jmetro | well, yeah. but its stopping a call from completing |
| 17:17.02 | jmetro | i meant the error. like what should I look around through to troubleshoot |
| 17:17.23 | WIMPy | Probably a peer that's unreachable. |
| 17:17.35 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 17:17.57 | jmetro | so odd though. everyone else in the office is reachable except that one phone [its a dial exten&exten&exten ] and only that one phone doesnt ring |
| 17:18.09 | jmetro | made a direct dial to it, same error. |
| 17:18.17 | jmetro | can transfer to the phone but not dial it. |
| 17:19.39 | [TK]D-Fender | jmetro, Clarify that |
| 17:19.57 | [TK]D-Fender | jmetro, what is "transfer to" vs "dial it"? |
| 17:21.19 | jmetro | Dialing into the office executes a Dial(Sip/exten&Sip/exten&Sip/exten) that rings all the phones up front. 101 doesnt ring, so 102 picks me up, parks me, and 101 picks it up. |
| 17:22.23 | WIMPy | Ok, so no transfer. |
| 17:22.34 | jmetro | The workgroup ring above puts up a Dial_exec_full cannot complete etc etc.. |
| 17:22.35 | WIMPy | Looks like the phone isn't registered. |
| 17:23.04 | jmetro | sip show peers shows registered though. |
| 17:23.43 | [TK]D-Fender | that is not a transfer |
| 17:23.54 | [TK]D-Fender | that is the phone being able to PLACE calls. |
| 17:23.56 | [TK]D-Fender | jmetro, And the error means that the peer either failed to qualify, or is not even registered. |
| 17:24.01 | [TK]D-Fender | ^ |
| 17:24.16 | [TK]D-Fender | "sip show peer X" |
| 17:24.58 | jmetro | hm... |
| 17:25.50 | jmetro | What would I be looking for in that list? |
| 17:26.17 | [TK]D-Fender | ~pb |
| 17:26.17 | 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. |
| 17:27.39 | Conker | [TK]D-Fender, now it says call completed and then closes like the call never went thru |
| 17:28.07 | [TK]D-Fender | Conker, And I still see nothing. |
| 17:28.34 | Conker | [TK]D-Fender, sorry you wanted the output of 'sip set debug on' ? |
| 17:28.56 | [TK]D-Fender | Conker, If you want a diagnosis you have to actually show the problem. |
| 17:29.09 | jmetro | http://pastebin.com/UgqRvjdg |
| 17:29.34 | jmetro | i was more hoping for a hint than a solve |
| 17:29.57 | Conker | [TK]D-Fender, i agree, and i appologize as i'm unfortunately still really new to this |
| 17:30.09 | [TK]D-Fender | <PROTECTED> |
| 17:30.16 | jmetro | right |
| 17:30.23 | jmetro | weird that it is in my sip show peers though |
| 17:30.43 | [TK]D-Fender | jmetro, Show that too.. |
| 17:30.46 | jmetro | actually it keeps dropping in and out of my sip show peers - i can see the registrations. |
| 17:31.57 | jmetro | Hm..i'm thinking it sthe phone - just had it randomly drop its configged buttons earlier in the week too. |
| 17:33.19 | jmetro | I should be good from here - thanks. |
| 17:36.59 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 17:37.17 | *** join/#asterisk johoja (~sakhter@24-246-1-22.cable.teksavvy.com) |
| 17:37.29 | johoja | hey guys quick question under what circumstance would i get this error ' app_dial.c:2218 dial_exec_full: Unable to create channel of type 'SIP' ' ? |
| 17:37.43 | johoja | it seems to happen after a few hours, and restarting asterisk fixes it. |
| 17:39.28 | [TK]D-Fender | johoja, <[TK]D-Fender> jmetro, And the error means that the peer either failed to qualify, or is not even registered. |
| 17:40.40 | johoja | [TK]D-Fender: the peer being the outgoing trunk i would assume ? |
| 17:41.36 | [TK]D-Fender | johoja, yes. |
| 17:42.37 | [TK]D-Fender | johoja, pastebin "sip show peer THEPEER", and your complete call attempt @ * CLI "sip set debug on", "core set verbose 10" |
| 17:43.04 | johoja | i'll have to wait for it tohappen again |
| 17:43.24 | johoja | another question , is there a way to turn on sip debug, but not for update/registartion messages |
| 17:43.36 | johoja | I jus twant to see the invite call flow sometimes. |
| 17:45.36 | *** join/#asterisk cklimos (~Claude@209.5.121.227) |
| 17:47.00 | *** join/#asterisk Pusher (~Pusher@95.235.5.222) |
| 17:47.04 | Pusher | hy to all ! |
| 17:47.17 | Pusher | someone speeks italian? |
| 17:49.21 | Pusher | i'm modifing a dialplan for ivr, i only need a simple help for a simple function: the logic is all ok i can enter in a specific routine and i tested id with a simple hangup and it's ok, ineed to add previsius of hangup the call to a specific voicemail |
| 17:49.49 | Pusher | i used: exten => s,1000,Voicemail(200) |
| 17:49.49 | Pusher | exten => s,1001,Hangup() |
| 17:49.57 | Pusher | but it doesn't work |
| 17:50.08 | Pusher | 200 is my extension's voicemail |
| 17:50.21 | Pusher | <PROTECTED> |
| 17:50.29 | Pusher | thanks previsious |
| 17:50.38 | *** join/#asterisk caveat- (hoax@gateway/shell/bshellz.net/x-bdcdppmfqlgmhhhv) |
| 17:51.21 | navaismo | priority 1000 O_o thats a big dialplan |
| 17:51.36 | navaismo | s/1000/1001/ |
| 17:51.41 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.111) |
| 17:52.09 | cklimos | is your voicemail.conf ok? |
| 17:52.38 | cklimos | what happens when you make a call? |
| 17:53.40 | Pusher | yes, i'm using askozia |
| 17:53.50 | Pusher | the voicemail normally is ok |
| 17:54.27 | [TK]D-Fender | Pusher, Show us the actual failed call |
| 17:54.35 | [TK]D-Fender | ~pb |
| 17:54.36 | 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. |
| 17:54.37 | [TK]D-Fender | ^^^ |
| 17:54.59 | Pusher | i want this function ( is alredy accordig with other) rdirect to voicemail, not else, the 200 (extension) voicemail |
| 17:55.07 | Pusher | one moment |
| 17:55.20 | Pusher | i use pastebin for function |
| 17:55.43 | cklimos | make a call and paste the CLI output to pastebin |
| 17:57.26 | jmetro | priority 1000 makes me think of the torture AA |
| 17:57.49 | Pusher | http://pastebin.com/GiqDNkGm |
| 17:57.55 | Pusher | the problem is at row 149 |
| 17:58.20 | Pusher | i used 1000 for a jump at the end |
| 17:58.34 | jmetro | 149 is a commented line |
| 17:58.57 | Pusher | 150 ! |
| 17:59.14 | Pusher | my edit at ivr starts at 149 |
| 17:59.33 | [TK]D-Fender | Pusher, I said show the FAILURE. Show us the actual CALL fail at * CLI |
| 17:59.41 | cklimos | nevermind all that. show a trace of a call |
| 17:59.41 | Pusher | are 2, between "inserisci" and "fine inserisci" |
| 18:01.21 | Pusher | i'm not expert... how for the trace? i use askozia |
| 18:01.27 | Pusher | with web gui |
| 18:01.33 | Pusher | but i can log in console |
| 18:02.36 | Pusher | i'm on the cli |
| 18:02.44 | *** join/#asterisk TheCompWiz (~TheCompWi@198.211.95.6) |
| 18:02.55 | jmetro | now on the cli type "core set verbose 1000" and "core set debug 1000" |
| 18:03.12 | jmetro | then call in and get to the part where its failing |
| 18:03.27 | jmetro | and post everything from how it fails. |
| 18:03.42 | Pusher | ok |
| 18:03.44 | cklimos | post everything from the beginning of your call |
| 18:03.46 | Pusher | now? start call ?= |
| 18:03.51 | Pusher | ok |
| 18:04.18 | *** join/#asterisk anonymouz666 (~anonymouz@189-25-53-120.user.veloxzone.com.br) |
| 18:04.38 | Pusher | ok |
| 18:04.45 | Pusher | in paste bin or here? |
| 18:04.52 | Pusher | are about 10 rows |
| 18:04.57 | Pusher | (the last= |
| 18:04.58 | Pusher | ) |
| 18:05.25 | Qwell | ~pb |
| 18:05.25 | 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. |
| 18:05.50 | Pusher | http://pastebin.com/ifqCBrLU |
| 18:06.23 | jmetro | there is no error there? |
| 18:06.50 | Pusher | this is the last log.... |
| 18:06.59 | Pusher | FUUCK..... |
| 18:06.59 | cklimos | ${200}?????? |
| 18:07.00 | Pusher | ONE MOMENT |
| 18:07.10 | Pusher | THEW EXTENSION IS WRONG!!!! |
| 18:07.11 | Pusher | DHO !!! |
| 18:07.17 | Pusher | IS 201 NOT 200 .... |
| 18:07.20 | Pusher | I CHANGE AND TRY ! |
| 18:07.21 | [TK]D-Fender | exten => s,1000,Voicemail(s${200}) |
| 18:07.25 | Pusher | EXCUSE ME !! |
| 18:07.30 | [TK]D-Fender | 200 is NOT a valid VARIABLE NAME |
| 18:07.44 | cklimos | don't replace 200 with 201 |
| 18:07.51 | cklimos | i won;t work better |
| 18:08.23 | cklimos | it won't work better |
| 18:08.45 | *** join/#asterisk TimeRider (~steve@188-220-34-144.dsl.cnl.uk.net) |
| 18:09.24 | Pusher | ok... in fact !!! |
| 18:09.52 | Pusher | the result is the same, but the wanted estension's voicemail is the one for 201 not 200 |
| 18:10.02 | Pusher | how can i solve? |
| 18:10.28 | Pusher | when it comes in this routine mustinli call the 201's voicemail |
| 18:10.30 | Pusher | and stop ! |
| 18:10.38 | Pusher | *must only |
| 18:10.40 | cklimos | try removing the ${} and simply put 201 |
| 18:10.45 | Pusher | ok |
| 18:11.18 | Pusher | i have exten => s,1000,Voicemail(s${201}) |
| 18:11.25 | [TK]D-Fender | NO |
| 18:11.27 | Pusher | change in exten => s,1000,Voicemail(200) |
| 18:11.30 | Pusher | ok? |
| 18:11.35 | cklimos | no ${} |
| 18:11.41 | Pusher | change in exten => s,1000,Voicemail(200) |
| 18:11.43 | Pusher | ? |
| 18:11.46 | [TK]D-Fender | ${} <--- this is for VARIABLES |
| 18:11.54 | [TK]D-Fender | you are NOT trying to put a variable there. |
| 18:11.55 | [TK]D-Fender | just the NUMBEr |
| 18:12.10 | cklimos | exten => s,1000,Voicemail(201) |
| 18:12.15 | [TK]D-Fender | yes |
| 18:12.45 | Pusher | whowwwwwwwww |
| 18:12.48 | Pusher | thanks a lot !! |
| 18:13.00 | Pusher | it work !! |
| 18:13.02 | Pusher | :))) |
| 18:13.11 | cklimos | i think you're trying to go too fast before properly understanding the basics |
| 18:13.41 | cklimos | it is nice that it works but you should understand why it was not working and why it is fixed |
| 18:13.43 | Pusher | yes... i'm beginner with asterisk, i discovere a fantastic world ! |
| 18:13.51 | Pusher | yes yes of course |
| 18:13.56 | [TK]D-Fender | ~book |
| 18:13.56 | 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 |
| 18:13.57 | [TK]D-Fender | ^^^ |
| 18:14.02 | Pusher | i must study the sintax |
| 18:14.36 | Pusher | asteriskdocs |
| 18:14.50 | Pusher | good to know... i'll read it !! |
| 18:15.00 | cklimos | you promise? |
| 18:15.02 | jmetro | dont forget you can always find the commands on teh wiki |
| 18:15.04 | cklimos | hehe |
| 18:15.10 | Pusher | oh yes |
| 18:15.19 | jmetro | like whenever I am questioning my pattern matching, i just google "asterisk dial patterns" |
| 18:15.26 | Pusher | this error in fact was a really basic error.... |
| 18:16.13 | Pusher | some one knows askozia? |
| 18:16.36 | jmetro | never heard of it |
| 18:16.56 | Pusher | it's a very good project |
| 18:17.05 | Pusher | witht a great web interface |
| 18:17.08 | Pusher | very verysolid |
| 18:17.38 | jmetro | I have vanilla asterisk =p |
| 18:17.48 | cklimos | asme here |
| 18:17.51 | cklimos | same |
| 18:17.55 | Pusher | i use it at home from some months, no problem, and now i start """building""" (MODIFING) some diaplan |
| 18:18.23 | jmetro | the dialplan is the easiest part for me, i do lots of coding in many languages. |
| 18:18.27 | Pusher | but i must study...because asterisk is a bomb !!! it's reat ! |
| 18:18.50 | *** part/#asterisk maetrik (maetrik@2a02:2308::c61:c0ca:c01a) |
| 18:19.14 | Pusher | i read thath now... with html5 is possible to build web application wit a sip client integrated for web call |
| 18:19.19 | Pusher | great! |
| 18:19.35 | jmetro | that and XML make it all easy |
| 18:19.44 | Pusher | yes! |
| 18:20.28 | Pusher | now i'have this release (askozia) in alix board... a grandstream for pstn line a a pap2t |
| 18:20.39 | Pusher | alla funcioning... also fax |
| 18:21.32 | Pusher | next step is connect the hylafax server for centralize office fax |
| 18:22.20 | Pusher | taskozia also have the fax to mail.... but a share with fax, or a graphic client (java??) was good |
| 18:22.47 | Pusher | i have all that in hylafax and now must couple askozia and hylafax ! |
| 18:23.20 | Pusher | (hylafax have email to fax and fax to email... of course) |
| 18:23.36 | Pusher | many office need it for mobility |
| 18:33.05 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 18:33.06 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 18:44.24 | *** join/#asterisk JasonL (~jason@216.223.114.3) |
| 18:45.35 | Pusher | an other question: i have exten => s,n,Set(DBTEST_DN=${DB_EXISTS(status/dn)}) |
| 18:45.36 | Pusher | exten => s,n,GotoIf($[${DB_RESULT}=2]?1000) |
| 18:45.55 | Pusher | ho can i show in the cli the valure of db_result? |
| 18:46.13 | JasonL | Hi all... has anyone had an issue with multiple peers from one IP? I have one peer set to register and the other as static. Anytime a call comes in on the static peer, it fails because Asterisk is matching the IP with the other peer. |
| 18:46.51 | JasonL | Pusher: s,n,NoOp(${DB_RESULT}) |
| 18:48.37 | [TK]D-Fender | Peer matches by IP. That's how it works. Make a "user" so it can match by username instead |
| 18:49.04 | JasonL | [TK]D-Fender: Thanks! let me try |
| 18:51.44 | *** join/#asterisk ageis (kevin@67.222.146.23) |
| 18:52.06 | ageis | i wanna change my dialplan so callers are put on hold straight away while phones are rung-- easy to do? |
| 18:53.16 | ageis | heres an existing extension http://pastebin.com/PaCkAPh2 |
| 18:53.44 | ageis | basically want to change so it will ring forever and callers will never get the unavailable message |
| 18:54.40 | [TK]D-Fender | exten => 1,n,Dial(SIP/salescomp&SIP/salescomp2&SIP/salescomp3&SIP/office, 25) <-- 25 != forever |
| 18:55.28 | ageis | [TK]D-Fender: thanks, what about having it play Hold music instead of ring? |
| 18:55.44 | TheCompWiz | one ringy ringy... ah ah ah.... two ringy ringy... ah ah ah... |
| 18:57.53 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 18:59.20 | navaismo | aegis, use option m |
| 19:01.57 | ageis | is that a dial option? BTW, can I merely remove the comma and 25 to remove timeout? |
| 19:08.16 | *** join/#asterisk Galen (~Galen@rrcs-24-43-17-237.west.biz.rr.com) |
| 19:10.27 | navaismo | yes that is a dial option more info: core show application dial |
| 19:11.39 | ageis | thanks sir |
| 19:20.13 | *** join/#asterisk tjfontaine (~tjfontain@unaffiliated/tjfontaine) |
| 19:32.20 | *** join/#asterisk keycruncher (~Adium@c-174-59-233-118.hsd1.pa.comcast.net) |
| 19:33.24 | *** join/#asterisk wonderworld (~w@dsdf-4d0a096f.pool.mediaWays.net) |
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| 20:01.41 | paulc | ~book |
| 20:01.42 | 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 |
| 20:24.43 | *** join/#asterisk dtcrshr (~datacrush@unaffiliated/datacrusher) |
| 20:39.15 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 20:39.15 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 20:56.21 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
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| 21:02.59 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 21:20.54 | *** join/#asterisk epaphus (~user1@108.174.50.29) |
| 21:22.22 | epaphus | Hello. I have a asterisk server set up with a couple of extensions, and a DID.... what I want to do is when the DID rings.. send it to a prerecorded greetings and then ring the extension. |
| 21:22.29 | epaphus | How is that possible, and do i need a queue? |
| 21:24.49 | paulc | ephaphus: you can use Playback to play the announcements, then Dial to send it to one or more phones |
| 21:25.17 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 21:25.28 | paulc | You don't need Queue unless you want to distribute calls between phones or queue them up (like "what happens if a call comes in and all phones are busy?") |
| 21:27.36 | [TK]D-Fender | epaphus, No. Answer. Playback. Dial. The End. |
| 21:27.46 | *** join/#asterisk sktyau (~nick@monitor.ic.ipera.net.au) |
| 21:29.25 | *** join/#asterisk TimeRider (~steve@188-220-34-144.dsl.cnl.uk.net) |
| 21:29.42 | epaphus | <[TK]D-Fender> where can i find a howto , in order to implement that |
| 21:30.11 | leifmadsen | asteriskdocs.org |
| 21:30.49 | Qwell | to implement playing a file? |
| 21:32.26 | epaphus | yes |
| 21:32.27 | epaphus | thanks |
| 21:43.22 | dijib | [Nov 7 16:30:58] WARNING[20986]: dsp.c:1403 ast_dsp_silence_noise_with_energy: Can only calculate silence on signed-linear, alaw or ulaw frames :( |
| 21:43.25 | dijib | [Nov 7 16:30:58] WARNING[20986]: channel.c:5043 ast_write: Codec mismatch on channel SIP/voipms-00000092 setting write format to unknown from ulaw native formats (ulaw) |
| 21:43.35 | dijib | anybody know ths error from a RecieveFAX() |
| 22:00.30 | *** join/#asterisk whtsup (~whtsup@WimaxUser38142-36.wateen.net) |
| 22:01.09 | whtsup | <PROTECTED> |
| 22:01.09 | whtsup | failed to extend from 1024 to 1318 |
| 22:01.09 | whtsup | failed to extend from 1024 to 1321 |
| 22:01.09 | whtsup | failed to extend from 1024 to 1323 |
| 22:01.09 | whtsup | failed to extend from 1024 to 1334 |
| 22:01.09 | whtsup | failed to extend from 1024 to 1312 |
| 22:01.10 | whtsup | failed to extend from 1024 to 1306 |
| 22:01.14 | whtsup | wht is dis ? |
| 22:01.31 | whtsup | when i make calls asterisk cli show this i m new to asterisk |
| 22:01.37 | whtsup | i m using asterisk 10 |
| 22:01.41 | *** join/#asterisk sawgood (~sawgood@173-13-158-29-sfba.hfc.comcastbusiness.net) |
| 22:01.53 | TheCompWiz | check your timing. |
| 22:02.02 | whtsup | which timing ? |
| 22:02.10 | TheCompWiz | what version of asterisk |
| 22:02.34 | whtsup | <PROTECTED> |
| 22:02.58 | TheCompWiz | in CLI run "timing test" |
| 22:03.37 | whtsup | Attempting to test a timer with 50 ticks per second. |
| 22:03.37 | whtsup | Using the 'timerfd' timing module for this test. |
| 22:03.37 | whtsup | It has been 1000 milliseconds, and we got 50 timer ticks |
| 22:03.42 | whtsup | showing dis |
| 22:04.08 | *** join/#asterisk fskrotzki (~fskrotzki@cpe-67-253-245-174.rochester.res.rr.com) |
| 22:04.52 | TheCompWiz | drop to cmd prompt and do "dahdi_test" and give it a few runs... and then paste teh best/worst/average |
| 22:05.29 | *** join/#asterisk sonstwo (~garland@unaffiliated/ffs) |
| 22:05.40 | whtsup | Best: 99.999% -- Worst: 99.904% -- Average: 99.991121% |
| 22:05.41 | whtsup | Cummulative Accuracy (not per pass): 99.996 |
| 22:06.11 | TheCompWiz | ... then I'm out of guesses. sorry. |
| 22:06.53 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:06.54 | *** join/#asterisk navaismo (~navaismo@189.241.62.150) |
| 22:07.48 | whtsup | ;s |
| 22:17.54 | *** join/#asterisk Iamnach0 (~Iamnacho@ip174-70-132-58.ks.ks.cox.net) |
| 22:26.50 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200::42:4) |
| 22:27.44 | *** join/#asterisk BenC[UK] (~bcummins@cpc10-lock3-2-0-cust14.6-1.cable.virginmedia.com) |
| 22:41.27 | epaphus | Hey guys. I have seen queues.conf ... i see most of it has good defauls. At the end iam going to put a agent as a static member for simplicity. Once i do that.. how do I actually go on testing the queue works |
| 22:46.10 | *** join/#asterisk sertaconay (~sertacona@unaffiliated/sertaconay) |
| 22:48.46 | [TK]D-Fender | epaphus: Use it. |
| 22:51.57 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 22:56.39 | epaphus | i dont recall configurating anything to enable the queue to listen on something.. so i dont know how to actually enter the queue |
| 23:01.13 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 23:02.06 | [TK]D-Fender | Queue's don't "listen" You dump your caller there. That is all |
| 23:04.43 | Micc_ | Does asterisk voicemail have a way to fast forward during playback of a message? |
| 23:04.53 | TheCompWiz | yes. |
| 23:05.00 | Micc_ | what is the key for that? |
| 23:05.20 | Micc_ | I couldn't find it while googling. |
| 23:05.27 | TheCompWiz | I belive 4/6 (back/forwards) |
| 23:05.38 | Micc_ | that jumps to the next message I think. |
| 23:06.46 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-aqydqcgljziadzow) |
| 23:07.24 | [TK]D-Fender | Micc_: http://www.voip-info.org/wiki/view/Asterisk+cmd+VoiceMailMain |
| 23:07.34 | TheCompWiz | Micc_: it has a config option... skipms This setting defines an interval in milliseconds to use when skipping forward or reverse while a voicemail message is being played. The value entered here should be a positive integer. The default value for this setting is 3000 (3 seconds). |
| 23:08.31 | TheCompWiz | # Exit; during msg playback: Skip forward |
| 23:08.37 | TheCompWiz | * Help; during msg playback: Rewind |
| 23:08.43 | Micc_ | ok, yeah I see it. |
| 23:08.49 | Micc_ | thanks |
| 23:09.29 | Micc_ | problem is if its outside the playing of the message it hangs up. |
| 23:10.01 | TheCompWiz | I fail to see the problem. |
| 23:10.36 | Micc_ | its not bad unless you press it after the end of the message on accident. |
| 23:10.42 | Micc_ | but thats unlikely I think. |
| 23:11.02 | Micc_ | especially when I tried to forward past the end it wouldn't go more than a few seconds from the end. |
| 23:11.04 | Micc_ | so thats good. |
| 23:11.17 | TheCompWiz | unfortunately... it's hard-coded... so if you want something else... time to dig through src code and modify. |
| 23:11.33 | Micc_ | I think it will be fine. |
| 23:11.53 | Micc_ | I'm sure it would be easy to disabled that # key at that point if I needed to though. |
| 23:12.35 | SeRi | dijib: you in? |
| 23:15.09 | *** join/#asterisk sertaconay (~sertacona@unaffiliated/sertaconay) |
| 23:19.29 | *** join/#asterisk Neptu (~Neptu@213-67-244-97-no100.tbcn.telia.com) |
| 23:28.12 | epaphus | can I feed a .wav file to the PlayBack command? |
| 23:36.18 | [TK]D-Fender | yes |
| 23:36.44 | [TK]D-Fender | Though you never specify the file-type extension in Playback, background, etc. |