| 00:05.04 | *** join/#asterisk qbx (186bbde4@gateway/web/freenode/ip.24.107.189.228) |
| 00:21.55 | aster1sk | Back |
| 00:21.58 | aster1sk | You around. |
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| 00:22.20 | aster1sk | woops, forgot to ctrl+x |
| 00:31.22 | lorsungcu | wat do |
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| 02:02.23 | Platinum | Anyone have experience with asterisk replacing audix? |
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| 02:50.06 | Maxus2 | Hi People, Does anyone know if there is a module to allow cdr logging via curl? and if so where is it? |
| 02:51.31 | ectospasm | cdr_curl was a project I was working on with Corydon76-home, don't know if it ever got anywhere |
| 02:51.42 | Maxus2 | ahh |
| 02:51.49 | ectospasm | ...back when he taught us an intro to programming for Asterisk |
| 02:51.51 | Maxus2 | i have seen irc logs mentioning it |
| 02:52.14 | *** join/#asterisk TeknoJuce (~TeknoJuce@xbmc/staff/TeknoJuce) |
| 02:53.17 | Maxus2 | Do you happen to knwo if it got finished or if its hard to write from scratch |
| 02:53.28 | Maxus2 | arrg got to go a meeting bbs |
| 02:53.39 | TeknoJuce | anyone know of an open sourced website plugin to allow you to make phone calls with your astrisk server from the website? |
| 02:54.06 | jpsharp | Click to call? Or a web-based softphone? |
| 02:54.26 | TeknoJuce | web-based softphone kinda like this http://www.mizu-voip.com/Products/WebPhone.aspx |
| 02:56.36 | jpsharp | There's a couple of them. i don't remember the SIP based ones off the top of my head, but I use a java-based IAX one called jiax. |
| 02:58.18 | TeknoJuce | I use a google voice account would that matter for jiax |
| 02:59.08 | TeknoJuce | Ideally I just want it on a website so it can page a room in the building the server exists in |
| 03:00.09 | TeknoJuce | so the web-based sip phone would just be calling another Page Ext. |
| 03:00.40 | TeknoJuce | -minus the sip in that sentence |
| 03:01.00 | Maxus2 | meeting cancelled, win! |
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| 03:01.21 | jpsharp | Yes, the web based phone would be calling anyother extension. |
| 03:01.32 | jpsharp | Maxus2: Those are the best kinds of meetings. |
| 03:01.39 | Maxus2 | agreed :) |
| 03:01.55 | TeknoJuce | so if I was just doing that the jiax one should work for my scenario correct? |
| 03:03.01 | TeknoJuce | if you think of the other ones jpsharp let me know, I will look up jiax |
| 03:03.50 | Maxus2 | ectospasm, do you have any examples of the code, was it hard to put together? |
| 03:04.07 | jpsharp | it should do what you want. I used it when I was developing something like blogtalkradio.com |
| 03:05.35 | jpsharp | Maxus2: You should be able to take the basic CDR module that's already there and modify it. |
| 03:05.46 | jpsharp | That's what I did when I wrote cdr_mysql ages ago. |
| 03:06.07 | ectospasm | Maxus2: a quick poking around my system reveals that I can't find it, sorry |
| 03:06.23 | TeknoJuce | cool jpsharp thanks for the nfo and the site looks good |
| 03:06.27 | Maxus2 | hard to write do you think? |
| 03:07.18 | ectospasm | it wasn't terribly difficult, IIRC |
| 03:07.48 | Maxus2 | cool, im surprised it doesn't exist already |
| 03:08.16 | jpsharp | it seems kinda pointless :) |
| 03:08.47 | Maxus2 | why is that? |
| 03:09.00 | Maxus2 | is there anthoer way to send cdr to a web endpoint? |
| 03:09.56 | jpsharp | Not to a web page. Most people log directly to a database or to a texas file. |
| 03:09.59 | jpsharp | er, text file. |
| 03:10.14 | Maxus2 | yeah we aren't allowed to talk directly to the database for security reasons. |
| 03:10.38 | Maxus2 | and a text file means the data needs to be harvested and some point and wont allow realtime data gathering |
| 03:11.08 | jpsharp | Ah. Well, then you'll either need to write it yourself or contract it out to be written. |
| 03:11.21 | Maxus2 | yeah, i suspect i can write it myself |
| 03:11.41 | Maxus2 | haven't done a huge amount of C, but done a fair chunk of c++ |
| 03:11.48 | Maxus2 | and i have other modules to use as examples |
| 03:12.01 | Maxus2 | just worried about threading |
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| 03:15.02 | jpsharp | I don't believe you have to worry about thread locking in the CDR modules. |
| 03:17.14 | Maxus2 | oh okay |
| 03:17.22 | Maxus2 | thats good news |
| 03:17.36 | Maxus2 | I suspect once i get this working i sill submit it back to asterisk |
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| 03:18.23 | jpsharp | Just write it, test it, submit it, and make sure you fill out the appropriate disclaimer paperwork. |
| 03:21.09 | Maxus2 | cool will do. |
| 03:21.34 | Maxus2 | is it okay to ask development questions in this chat, or is there a specific chat for that? |
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| 03:52.21 | jpsharp | Maxus2: there's #asterisk-dev |
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| 04:47.38 | dijib | hello asterisk |
| 04:47.42 | dijib | how is everyone this evening |
| 04:54.26 | ectospasm | trying not to fall asleep... |
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| 05:30.39 | dijib | hows that going for you? |
| 05:34.45 | ectospasm | so far so good... |
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| 05:50.57 | gg608f | ok |
| 05:50.58 | gg608f | ... |
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| 06:39.15 | racho | how can it Goto to a context with a _X. extension? somehow i feel it won't be Goto(mycontext,_X.,1) |
| 06:42.57 | bulkorok | goto(mycontext,123,1) should work... |
| 06:43.16 | bulkorok | but to me it looks a bit "undefined"... |
| 06:43.58 | bulkorok | a bit better would be goto(mycontext${EXTEB},1) |
| 06:44.11 | bulkorok | mycontext,${EXTEN} |
| 06:44.19 | bulkorok | needs a new keyboard... |
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| 06:48.27 | schmidts | good morning |
| 06:48.32 | bulkorok | hi |
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| 07:54.51 | kontinuity | hi all |
| 07:55.15 | kontinuity | is there a way to disable a peer from sip.conf? I don't want to remove the configuration or comment the whole block out |
| 07:55.32 | ectospasm | kontinuity: change it's secret |
| 07:58.31 | kontinuity | after changing secret in sip.conf can I just do a reload to make it work? |
| 08:02.18 | wdoekes | sip reload |
| 08:02.36 | kontinuity | wdoekes: ectospasm: thanks |
| 08:13.40 | ectospasm | no problem |
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| 10:33.13 | schmidts | ectospasm remember what we have talked about yesterday, my problem with dtmf on iax. enabling jitterbuffer didnt helped |
| 10:33.38 | ectospasm | schmidts: no, the suggestion was to explicitly disable it |
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| 10:35.08 | schmidts | ok it was not enabled till yesterday, so at all this doesnt help either ;) |
| 10:36.28 | schmidts | something about trunkfreq, trunktimestamps? bandwith, tos? i dont think one of these could help |
| 10:37.52 | ectospasm | only suggestion I found on voip-info was to disable the frame buffer |
| 10:37.54 | ectospasm | er, |
| 10:37.57 | ectospasm | jitter buffer |
| 10:38.06 | ectospasm | I work with frame buffers too much |
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| 10:41.14 | schmidts | :D |
| 10:44.12 | schmidts | btw it also depends on the caller, for example iphone or nokia cell phones it works just fine but all android phones have troubles which sounds not very typical for a problem on the trunk overall |
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| 10:51.31 | WIMPy | hasn't heard of non working DTMF on IAX so far. |
| 10:51.48 | WIMPy | Are you sure it's working before being converted to IAX? |
| 10:52.29 | WIMPy | I assume when talking about mobiles, you're not using IAX clients on them, are you? |
| 10:59.52 | schmidts | WIMPy it has worked before we have changed our carrier uplink. it was E1s before but on the same gateway server and now its only sip |
| 11:00.02 | schmidts | and all other clients doesnt have these problems |
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| 11:30.05 | tompaw | Morning |
| 11:30.23 | tompaw | Is there a way to debug WHY postgresql connection fails? Even with -vvvvv it only tells me "it failed". |
| 11:30.41 | ectospasm | tompaw: maybe debug messages will tell you |
| 11:31.21 | ectospasm | make sure logger.conf contains the line: console => notice,warning,error,debug,verbose,dtmf,fax (which is everything) |
| 11:31.26 | ectospasm | then logger reload |
| 11:31.32 | ectospasm | then core set debug 10 |
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| 11:31.38 | ectospasm | and try again |
| 11:31.54 | tompaw | ok |
| 11:34.49 | tompaw | argh... it might actually be a problem with the psql |
| 11:34.52 | tompaw | ectospasm: thanks |
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| 12:09.34 | OldMonk | hi |
| 12:10.21 | OldMonk | any reason why setting blindxfer=>#5 in features.conf wouldn't work, while blindxfer=># does? asterisk 1.8 |
| 12:11.56 | OldMonk | 1.8.8.0 to be precise |
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| 12:14.54 | ectospasm | OldMonk: that should work... did you restart Asterisk to change it? |
| 12:15.03 | ectospasm | ...unless #5 is taken by something else |
| 12:15.04 | OldMonk | ectospasm: yes, many times |
| 12:15.17 | OldMonk | well, i looked in features.conf and couldn't find it anywhere |
| 12:15.33 | OldMonk | any other place i should look? |
| 12:15.34 | as001 | Hello I have Asterisk device with 1 fxo and 1 fxs port. When I get call I can see callerid on Asterisk CLI but later in dialplan when I dial dahdi 1 (telephone) I can just see Incoming instead of callerid. I catch call with s extension in default context. |
| 12:15.48 | ectospasm | OldMonk: see what features show shows |
| 12:16.40 | OldMonk | Blind Transfer # #5 |
| 12:17.15 | OldMonk | pickup is *8, disconnect is * and nothing else is defined |
| 12:17.35 | ectospasm | OldMonk: so turn on dtmf debug and see if Asterisk ever processes the #5 event |
| 12:17.53 | OldMonk | wait, if it processes # wouldn't it process #5 too? |
| 12:18.15 | ectospasm | not necessarily |
| 12:18.28 | OldMonk | currently some 200 calls on that server so i can't experiment, will have to wait till late at night |
| 12:18.45 | ectospasm | maintenance windows ftw! |
| 12:19.06 | OldMonk | nah this one's ok, just calls between 09:30 and 19:00 our time |
| 12:19.26 | OldMonk | problem is, the server is remote, and the IT chaps also leave by that time :) |
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| 12:24.22 | OldMonk | thanks ectospasm, i'll be back </arnie> |
| 12:25.14 | ectospasm | OldMonk: you can at least turn on dtmf debug in the console... edit logger.conf, add "dtmf" to the console line... logger reload in the CLI, and try it again |
| 12:26.12 | OldMonk | ectospasm: will do that once calling stops... client gets soggy and hard to light if you reload any feature while calling is on |
| 12:26.55 | ectospasm | logger reloading should be innocuous |
| 12:28.11 | OldMonk | i agree, but there are wheels within wheels... users don't like asterisk (since it takes control and abuse out of their hands) and are just waiting to jump down IT's throat at the sign of the smallest problem |
| 12:28.16 | OldMonk | better to play it safe |
| 12:28.47 | OldMonk | IAC, no point enabling dtmf debug right now, the console is scrolling 10000 lines per minute |
| 12:30.24 | OldMonk | and while i bow to no one in my grep skills, every point of load average is significant |
| 12:31.35 | OldMonk | bb in a few hours |
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| 12:33.18 | mcolombo | hi all |
| 12:33.41 | mcolombo | can anybody help me with asterisk and call forwarding? |
| 12:33.46 | mcolombo | thanks |
| 12:36.13 | [TK]D-Fender | mcolombo, Ask a more specific question, get a more specific answer..... |
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| 12:46.01 | *** join/#asterisk keycruncher (~keycrunch@pa-67-234-13-5.dhcp.embarqhsd.net) |
| 12:46.42 | Alex_Bkash | i've configured IAX2 between two asterisk. but bw is not minimized as i see in asteriskgure calsulator |
| 12:47.48 | [TK]D-Fender | Alex_Bkash, Pastebin and show us |
| 12:48.23 | *** join/#asterisk aross42 (~aross@CPE009400809a9c-CMb89bc9d2e1a5.cpe.net.cable.rogers.com) |
| 12:49.07 | leifmadsen | Alex_Bkash: did you enable trunking? |
| 12:49.22 | leifmadsen | note that a single call will still be the same with or without trunking |
| 12:49.28 | *** part/#asterisk keycruncher (~keycrunch@pa-67-234-13-5.dhcp.embarqhsd.net) |
| 12:49.31 | Alex_Bkash | ok |
| 12:49.40 | Alex_Bkash | how can i do that? |
| 12:49.40 | leifmadsen | bandwidth is only minimized between end points when several simultaneous calls are pushed through |
| 12:49.49 | leifmadsen | look at the sample file and search for "trunk" |
| 12:49.59 | *** join/#asterisk nanoha-sama (~nanoha-sa@nanoha-sama.freenode.bouncers.smb.curriegrad2004.ca) |
| 12:50.04 | Alex_Bkash | i want to through multiple calls |
| 12:50.13 | Alex_Bkash | okk |
| 12:50.27 | *** part/#asterisk as001 (~uros@82.117.198.142) |
| 12:51.25 | Alex_Bkash | will it hamper call quality? |
| 12:52.08 | Alex_Bkash | which is most low bandwidth codec with good quality? |
| 12:53.20 | mcolombo | thanks, [TK]D-Fender |
| 12:53.26 | mcolombo | this is my problem |
| 12:53.51 | mcolombo | i have already insert in my dialplan this line, to do Call forwarding on Busy |
| 12:53.55 | [TK]D-Fender | Alex_Bkash, Where are the configs? |
| 12:54.16 | mcolombo | <PROTECTED> |
| 12:54.16 | mcolombo | <PROTECTED> |
| 12:54.16 | mcolombo | <PROTECTED> |
| 12:54.16 | mcolombo | <PROTECTED> |
| 12:54.18 | [TK]D-Fender | mcPASTEBIN <- |
| 12:54.19 | [TK]D-Fender | ~pb |
| 12:54.19 | 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. |
| 12:54.26 | [TK]D-Fender | mcolombo, PASTEBIN <- |
| 12:54.57 | Alex_Bkash | ok |
| 12:54.59 | mcolombo | ok, one second |
| 12:55.04 | [TK]D-Fender | mcolombo, And those lines by themselves don't necessarily mean anything. You are simply setting AstDB values. You have to actually LOOK at them at some point an make a decision based on them. |
| 12:55.19 | leifmadsen | Alex_Bkash: no trunking only affects the signalling, not the codec at all |
| 12:55.21 | Alex_Bkash | i cant access server right now. my office blocks ssh |
| 12:55.45 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 12:56.29 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-qbnwxcphrbibdjpg) |
| 12:56.29 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 12:56.30 | [TK]D-Fender | Alex_Bkash, Do you ask your mechanic what's wrong with your car without bringing it to him? |
| 12:56.53 | Alex_Bkash | i can give u id and pass of my server |
| 12:57.16 | [TK]D-Fender | Alex_Bkash, What good is that if YOU can't get in? |
| 12:57.28 | [TK]D-Fender | Alex_Bkash, You just told us it's blocked |
| 12:57.42 | Alex_Bkash | now im in office. cant access from office |
| 12:57.58 | leifmadsen | o.O |
| 12:58.01 | Alex_Bkash | i can give u the configs with 1.5 hrs |
| 12:58.08 | [TK]D-Fender | Alex_Bkash, Ask again when you've got something to show us |
| 12:58.18 | mcolombo | this is the pastbin link : http://pastebin.com/h2ZdD3sx |
| 12:58.37 | Alex_Bkash | ok |
| 12:58.40 | [TK]D-Fender | mcolombo, And read what I told you above... |
| 12:59.13 | mcolombo | yes of course, i already read |
| 12:59.18 | WIMPy | mcolombo: And where do you make use of the values you set? |
| 12:59.45 | [TK]D-Fender | mcolombo, exten => #21#,1,DBdel(CFIM/${CALLERIDNUM}) <---- and in 2 places we see you referencing the old variable for callerid instead of the the function that replaced it in * 1.2 over 5 years ago... |
| 13:00.14 | *** join/#asterisk Rac-on (jasper@bambi.rac-on.nl) |
| 13:01.02 | WIMPy | And can you use :4 within Replace()? I doubt so. |
| 13:01.29 | mcolombo | i'm following this link : http://www.voip-info.org/wiki/view/Asterisk+call+forwarding |
| 13:01.35 | mcolombo | to set call forwarding |
| 13:01.41 | [TK]D-Fender | mcolombo, That link doesn't prove what you created. |
| 13:02.00 | *** join/#asterisk awk (awk@trancehouse.co.za) |
| 13:02.02 | [TK]D-Fender | mcolombo, This is your dialplan. It does what you tell it to. Show us where you actually make use of any of that... |
| 13:02.04 | leifmadsen | pukes all over priority numbers |
| 13:02.25 | awk | Hi, when working with manager, and I turn off events how can I set only 1 event (Newstate) |
| 13:02.29 | *** join/#asterisk nanoha-sama (~nanoha-sa@nanoha-sama.freenode.bouncers.smb.curriegrad2004.ca) |
| 13:02.32 | Chainsaw | leifmadsen: But it was the 1.2 way! |
| 13:02.47 | leifmadsen | Chainsaw: true story ;) |
| 13:02.56 | *** join/#asterisk serafie (~erin@nat/digium/x-vwbcukjxciseelsn) |
| 13:06.37 | mcolombo | ok... I should have understood where is the problem |
| 13:06.48 | mcolombo | now l'im looking |
| 13:06.56 | mcolombo | I will let you know |
| 13:06.58 | mcolombo | thanks for all |
| 13:07.23 | Katty | morning |
| 13:08.11 | Katty | what's the word. |
| 13:08.17 | Katty | that you heard. which has groove and meaning. |
| 13:09.45 | [TK]D-Fender | Katty, Haven't you heard? |
| 13:09.59 | *** join/#asterisk TriJetScud (~TriJetScu@2001:470:e97f:1000::1) |
| 13:10.17 | Katty | no i'm deaf. |
| 13:10.56 | [TK]D-Fender | Katty, That the BIRD bird BIRD the BIRD is THE word! I SAID the BIRD bird BIRD the BIRD is THE word! I SAID the BIRD bird BIRD the BIRD is THE word! I SAID the BIRD bird BIRD the BIRD is THE word! I SAID the BIRD bird BIRD the BIRD is THE word! I SAID the BIRD bird BIRD the BIRD is THE word! I SAID the BIRD bird BIRD the BIRD is THE word! I SAID the BIRD bird BIRD the BIRD is THE word! I SAID the BIRD bird BIRD the BIRD is THE word! I SA |
| 13:10.57 | [TK]D-Fender | ID the BIRD bird BIRD the BIRD is THE word! I SAID the |
| 13:11.09 | Katty | really now. |
| 13:11.13 | [TK]D-Fender | INDEED |
| 13:11.18 | Katty | mkay. |
| 13:11.29 | *** join/#asterisk nanoha-sama (~nanoha-sa@nanoha-sama.freenode.bouncers.smb.curriegrad2004.ca) |
| 13:12.30 | *** join/#asterisk OldMonk (~raju@120.56.169.94) |
| 13:16.47 | OldMonk | ok, when trying to blind xfer with #5 instead of just #, this is the result: http://www.pastebin.ca/2210572 |
| 13:17.05 | OldMonk | it times out within 4 seconds |
| 13:17.20 | OldMonk | it does not provide the "dial extension to transfer to" message |
| 13:17.38 | leifmadsen | show your features.conf and 'features show' output |
| 13:18.24 | OldMonk | features show: http://www.pastebin.ca/2210573 |
| 13:19.00 | [TK]D-Fender | OldMonk, and the call... |
| 13:19.10 | OldMonk | features.conf: http://www.pastebin.ca/2210574 |
| 13:19.21 | [TK]D-Fender | 8 = disconnect call? That doesn't look hazardous at all... |
| 13:19.23 | [TK]D-Fender | * |
| 13:19.46 | leifmadsen | are you dialing the dtmf fast enough? |
| 13:19.52 | OldMonk | will it affect the blindxfer problem? |
| 13:20.00 | OldMonk | leifmadsen: yes |
| 13:20.13 | leifmadsen | then I don't know -- usually works for me |
| 13:20.26 | OldMonk | if you set blindxfer to just #, it prompts and waits paitiently for the extension to be dialled |
| 13:20.50 | OldMonk | where can i set that timeout? |
| 13:21.00 | leifmadsen | in features.conf |
| 13:21.00 | OldMonk | asterisk 1.8.8 |
| 13:21.15 | Katty | mixes vinegar and baking soda together near Qwell |
| 13:21.22 | leifmadsen | Katty: nice moves |
| 13:21.44 | Katty | acetic acid and sodium bicarbonate = plant air. |
| 13:21.52 | Katty | also, carbon dioxide. |
| 13:22.13 | OldMonk | ;featuredigittimeout = 1000 (default), which seems ample |
| 13:23.02 | OldMonk | ok, will increase that to 5 seconds and see if the prompt comes |
| 13:23.31 | *** join/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 13:23.58 | mcolombo | i found the problem, but when i make a call i receive an error like this : http://pastebin.com/FMeRuW2U |
| 13:24.07 | mcolombo | this is my dialplan : http://pastebin.com/KxGJ1Hrm |
| 13:24.35 | [TK]D-Fender | mcolombo, exten => _X.,n,Set(CALLERID(num)=${CALLERID(num)}) <- waste of time |
| 13:25.38 | mcolombo | ok, i delete it |
| 13:25.39 | [TK]D-Fender | mcolombo, And look at the pattern of the line that you think has that label, and then look at all the OTHER lines |
| 13:25.43 | [TK]D-Fender | ^ |
| 13:25.45 | OldMonk | great, that works |
| 13:25.48 | OldMonk | thanks folks |
| 13:28.29 | mcolombo | sorry but i don't find any big error xd |
| 13:28.48 | [TK]D-Fender | mcolombo, I jsut told you something EXTREMELY specific to look at. |
| 13:29.17 | mcolombo | it's ok |
| 13:29.21 | mcolombo | i'm find the error |
| 13:29.23 | mcolombo | goood! |
| 13:29.26 | tompaw | python + asterisk call manager + asterisk realtime + res_pgsql = <# |
| 13:29.28 | tompaw | <3 even |
| 13:29.29 | tompaw | :> |
| 13:29.30 | mcolombo | thanks for all D-Fender |
| 13:30.10 | tompaw | if only there was a decent java/flash web phone... |
| 13:30.30 | [TK]D-Fender | mcolombo, If you think the label is there... look very close to make sure that it's on the exact same pattern and that every single character, comma, and space matches exactly as it should. |
| 13:31.38 | *** join/#asterisk bchia (~Adium@nat/digium/x-bqmaxwuntlzhevwe) |
| 13:31.59 | tompaw | ekhm... IF ONLY THERE WAS A DECENT JAVA/FLASH WEB SIP CLIENT |
| 13:32.07 | mcolombo | true, I will pay more attention |
| 13:44.52 | leifmadsen | tomaw: pretty sure a few exist... |
| 13:45.09 | leifmadsen | http://flashphoner.com/ |
| 13:45.12 | leifmadsen | per a google search |
| 13:45.15 | tompaw | leifmadsen: do you happen to know any that works on the client side (i.e. doesn't require server proxy) |
| 13:45.28 | leifmadsen | never used one, so I can't answer that |
| 13:45.46 | tompaw | I know flashphoner, but unfortunately it doesn't work like that, |
| 13:46.00 | tompaw | And Zoiper for web doesn't have OSX version :\ |
| 13:46.12 | leifmadsen | tomaw: ya I thik it uses activeX |
| 13:46.15 | leifmadsen | so no linux either |
| 13:46.42 | awk | hmm, anyone good with AMI here, as I don't want to write detailed info if nobody can assist? |
| 13:46.52 | tompaw | awk: what exactly do you have in mind? |
| 13:47.39 | tompaw | awk: I'm using AMI a lot, but until I know your question I won't be able to tell if I can help :> |
| 13:47.44 | wdoekes | awk, if you're going to be too lazy to write a question.. |
| 13:48.27 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 13:48.33 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 13:48.33 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 13:48.59 | *** join/#asterisk TriJetScud (~TriJetScu@2001:470:e97f:1000::1) |
| 13:51.16 | Katty | yarn bombs leifmadsen |
| 13:51.36 | leifmadsen | NOOOOOOOOOOOOOO...wat? |
| 13:51.41 | tompaw | Since AMI was mentioned - is there any other way to automatically add and remove agents from the queue? |
| 13:51.49 | leifmadsen | tomaw: yes |
| 13:52.17 | leifmadsen | pretty sure you can just dynamically add and remove queue members with Add/RemoveQueueMember |
| 13:52.17 | Katty | leifmadsen: http://smallstudio.typepad.com/.a/6a00d83452186e69e201630580eb93970d-800wi <- yarn bombed trees. |
| 13:52.20 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 13:52.23 | leifmadsen | if not that way, then just through an Exec |
| 13:52.32 | tompaw | At the moment I'm doing it like this: http://pastebin.com/hBr6wipP |
| 13:52.43 | leifmadsen | Katty: wtf, did someone knit enough to cover all those trees?! |
| 13:52.59 | tompaw | leifmadsen: that's what I'm using now, but I wonder if it's not an overkill to use AMI for such a trivial task. |
| 13:53.17 | Katty | leifmadsen: yes. |
| 13:53.19 | leifmadsen | not really if you're doing it from an interface or some other app |
| 13:53.29 | Katty | leifmadsen: likely a group of someones tho |
| 13:53.30 | leifmadsen | AMI is usually the preferred method than via some hack like asterisk -rx or something :) |
| 13:53.44 | leifmadsen | tompaw: usually if doing via AMI, it's not the wrong way |
| 13:53.47 | Katty | leifmadsen: knitting isn't exactly something that is quick. |
| 13:53.52 | leifmadsen | Katty: this I know |
| 13:53.53 | tompaw | ah, I thought maybe there is something like queue=ABC in users.conf |
| 13:54.01 | tompaw | would be neat if * could handle that automagically |
| 13:54.05 | leifmadsen | I avoid users.conf like the plague |
| 13:54.15 | Katty | speaking of knitting, i should really finish up the tardis laptop sleeve for my friend /before/ christmas. |
| 13:54.47 | *** join/#asterisk Johnny- (~John@23-24-48-117-static.hfc.comcastbusiness.net) |
| 13:55.27 | tompaw | leifmadsen: when I say users.conf I really mean extconfig.conf, but you know what I had in mind. |
| 13:55.42 | *** join/#asterisk autofsckk (~que@unaffiliated/autofsckk) |
| 13:55.49 | tompaw | somehow letting * know that we want that particular user in ABC queue as long as they're registered |
| 13:56.26 | bchia | tompaw - a permanent queue member in queues.conf wouldn't accomplish that? |
| 13:56.36 | leifmadsen | I see... I think the only real way to do that is via monitoring of a login in AMI |
| 13:56.51 | tompaw | bchia: will that work with res_pgsql users? |
| 13:56.52 | leifmadsen | bchia: ya could do that too -- since when not reg'd would show as offline/unavailable |
| 13:58.17 | bchia | I'm not sure why it wouldn't |
| 13:58.56 | mirela666 | tompaw: You can do it with RealTime table too |
| 13:59.54 | tompaw | mirela666: got it, I would have to duplicate the agent data in my app though so it's in both sipusers and queues tables |
| 14:00.18 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:00.18 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:00.30 | *** join/#asterisk nanoha-sama (~nanoha-sa@nanoha-sama.freenode.bouncers.smb.curriegrad2004.ca) |
| 14:01.15 | tompaw | You know what would be cool? res_json.conf :-) |
| 14:01.29 | tompaw | I wonder if * folks are planning something like that in the near future. |
| 14:02.16 | tompaw | We wouldn't be bound by the database structure and in some cases wouldn't have to duplicate the business data. |
| 14:04.28 | *** join/#asterisk Tom_123 (~tmccurdy@147.177.61.191) |
| 14:04.36 | Tom_123 | are there any good tools to simulate sip traffic |
| 14:04.42 | Tom_123 | or calls |
| 14:04.45 | Tom_123 | etc |
| 14:06.03 | wdoekes | Tom_123: sipp |
| 14:07.14 | *** part/#asterisk aster1sk (~aster1sk@24-212-138-13.cable.teksavvy.com) |
| 14:08.46 | Tom_123 | wdoekes, is this the defacto tool to use for this type of testing |
| 14:08.57 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 14:10.32 | mjordan | Tom_123: we use it extensively in the Asterisk Test Suite. |
| 14:11.23 | Tom_123 | how does that compare to any type of expensive hardware simulators? |
| 14:12.02 | tompaw | Tom_123: my business offers infrastructure tester that can simulate live traffic quite well and monitor the differences on the end |
| 14:12.46 | tompaw | If you're interested, come to private chat. |
| 14:12.51 | mjordan | Tom_123: its a free way to create SIP messages and send them to some other endpoint. Its a tool for a job - whether or not it works for what you want to go use it for is up to you to decide. |
| 14:14.06 | Tom_123 | Alright, thank you for the input, I will take a look at sipp |
| 14:15.27 | carrar | sipsak is nice too |
| 14:16.43 | Katty | you put some sip and some sak and some sip and sak it's a ssipsakstack a sipsakstack |
| 14:16.49 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 14:16.49 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 14:17.17 | carrar | You put the SIP in the SAK can and what do ya got! |
| 14:17.34 | Katty | a sip sak stack!!! |
| 14:17.42 | carrar | OMFG |
| 14:18.57 | carrar | tompaw, what software does your company make? |
| 14:20.16 | carrar | looks at Katty |
| 14:20.16 | drmessano | Asterisk4Win |
| 14:20.22 | carrar | haha |
| 14:20.27 | Katty | carrar: yes dear? |
| 14:20.35 | carrar | Hi |
| 14:20.45 | Katty | herro |
| 14:20.46 | carrar | was waiting for tompaw to respond |
| 14:20.52 | carrar | but |
| 14:20.57 | carrar | his sales pitch is a fail |
| 14:21.15 | carrar | drmessano FTW |
| 14:23.24 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 14:25.20 | tompaw | carrar: http://www.tesserakt.eu/lcr |
| 14:25.56 | tompaw | It's a wholesale carrier toolbox, not sure if I should be talkign about it here tho. |
| 14:26.00 | carrar | thanks! |
| 14:26.18 | carrar | if someone ask you a question I think it's ok |
| 14:26.47 | WIMPy | Again something called LCR... |
| 14:27.37 | carrar | I need LCR for my car |
| 14:27.49 | WIMPy | At least you can be sure noone can find you on google. |
| 14:28.18 | tompaw | WIMPy: I started working on it in 2006, and originally it was supposed to be LCR engine only :) |
| 14:31.22 | AkkerKid | heya everyone! If my app_queue.so module won't load, where could I look for a reason? I was messing with my conf files but even after reverting, it's still not loading |
| 14:32.40 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 14:32.41 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 14:32.56 | WIMPy | AkkerKid: Did you try to increase verbose and possibly debug and load it manually? |
| 14:33.13 | *** join/#asterisk rbowles (~bb@75-147-63-201-NewEngland.hfc.comcastbusiness.net) |
| 14:33.27 | AkkerKid | i'll try that now |
| 14:36.19 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 14:36.20 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 14:36.40 | AkkerKid | <PROTECTED> |
| 14:36.41 | AkkerKid | Unable to load module app_queue.so |
| 14:36.41 | AkkerKid | Command 'module load app_queue.so' failed. |
| 14:36.41 | AkkerKid | <PROTECTED> |
| 14:36.41 | AkkerKid | <PROTECTED> |
| 14:36.47 | AkkerKid | and that's it. |
| 14:37.11 | *** kick/#asterisk [AkkerKid!~pabelange@asterisk/contributor-and-bug-marshal/pabelanger] by pabelanger (pastebin) |
| 14:38.17 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 14:38.17 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:39.52 | *** join/#asterisk aross42 (~aross@CPE009400809a9c-CMb89bc9d2e1a5.cpe.net.cable.rogers.com) |
| 14:41.22 | *** join/#asterisk AkkerKid (~AkkerKid@50-200-18-202-static.hfc.comcastbusiness.net) |
| 14:41.49 | AkkerKid | wow someone's got their dick in a knot about pastebin. Who's making money on that? |
| 14:41.58 | tompaw | pastebin |
| 14:41.59 | AkkerKid | Anyways... |
| 14:42.19 | pabelanger | not flooding is the solution |
| 14:42.34 | AkkerKid | i figure 5 lines would kill anyone |
| 14:42.39 | AkkerKid | but so be it. |
| 14:42.47 | AkkerKid | http://pastebin.com/piVLFgq6 |
| 14:43.01 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 14:43.02 | AkkerKid | that's all i get. |
| 14:43.25 | pabelanger | what version of asterisk? |
| 14:43.39 | AkkerKid | Asterisk 1.6.2.10 |
| 14:43.52 | pabelanger | upgrade to 1.8 and retry |
| 14:43.53 | AkkerKid | sitting under FreePBX and Elastix. |
| 14:43.56 | pabelanger | looks like a bug |
| 14:44.00 | AkkerKid | can't upgrade this box. |
| 14:44.02 | AkkerKid | it's live |
| 14:44.11 | pabelanger | then you need to live with the problem |
| 14:44.17 | pabelanger | 1.6.2 no longer supported |
| 14:44.19 | AkkerKid | it was working fine until it rebooted this morning |
| 14:44.22 | carrar | upgrade to 10! |
| 14:44.32 | pabelanger | or look at 1.8 and see if it is fixed |
| 14:45.52 | AkkerKid | I'm going to have a hard time beliving that I can't undo whatever I did in a few conf files and get it working again. |
| 14:46.11 | pabelanger | so what did you do? |
| 14:46.13 | eduzimrs | Hi, is it possible to hear clearly a call using MixMonitor with SRTP media stream? |
| 14:47.03 | AkkerKid | I'm trying to make some queues without using the Elastix interface. |
| 14:47.18 | AkkerKid | I'm putting them directly into the conf files and having asterisk reload them |
| 14:47.24 | AkkerKid | I wasn't successful yesterday |
| 14:47.35 | drmessano | Remove what you added |
| 14:47.40 | AkkerKid | and when my server rebooted overnight, the queues wouldn't load at all |
| 14:47.57 | AkkerKid | even the queues that we put in by FreePBX/Elastix son't work. |
| 14:48.12 | pabelanger | #elastix or #freepbx then |
| 14:48.17 | AkkerKid | and I though I removed all of my manual stuff but it's still not loading the app_queue.so module |
| 14:48.55 | pabelanger | eduzimrs: I don't see why not |
| 14:49.01 | AkkerKid | is there not a more detailed log file regarding module loading errors for asterisk? |
| 14:49.23 | pabelanger | ~collectdebug |
| 14:49.23 | infobot | extra, extra, read all about it, collectdebug is a method of collecting logs allowing others help troubleshoot an issue. Refer to https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 14:51.10 | AkkerKid | infobot should have used pastebin. :) |
| 14:52.18 | drmessano | I guess infobot was trying to be more like you |
| 14:52.36 | AkkerKid | flattering |
| 14:54.34 | *** join/#asterisk StaRetji (~LittleAll@178.79.11.166) |
| 14:54.58 | StaRetji | folks, is there a online list of known spam sip or fraud sim servers? |
| 14:55.22 | StaRetji | I received two calls to my sip test number, directly from unknown@ipaddress sip number |
| 14:55.36 | StaRetji | weird, specially if I know that my sip number is long and random |
| 14:55.45 | StaRetji | and it is the only 1 on the server |
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| 14:56.45 | StaRetji | how is this possible, I thought noone can see my sipnumber, specially if I never used this number on any server except mine, local usage |
| 14:57.15 | StaRetji | in other words, never connected to another sip server, so I am puzzled and worried :/ |
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| 14:58.06 | ke-esc | morning all. I'm trying to set up odbc storage for voicemail as in the definitive guide. i've got odbc pointing to a mysql db which is also being used for realtime (which is working for sippeers and voicemail).. when I record a voicemail message though, nothing is put into the table defined in voicemail.conf |
| 14:59.00 | ke-esc | i'm using Asterisk 1.8.11.1-1digium1~squeeze |
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| 15:05.38 | StaRetji | anyone folks? should I be worried? |
| 15:06.39 | pabelanger | sudo apt-get install install voicemail-odbcstorage |
| 15:06.40 | pabelanger | I think |
| 15:06.59 | pabelanger | err |
| 15:07.03 | pabelanger | sudo apt-get install install asterisk-voicemail-odbcstorage |
| 15:09.37 | StaRetji | can someone easy scan for asterisk extensions? |
| 15:10.02 | ke-esc | pabelanger, aha! silly package managers.... that did the trick |
| 15:10.13 | pabelanger | yes, look at sipvicious |
| 15:10.26 | pabelanger | ke-esc: it is a hack to get around the asterisk build environment |
| 15:10.41 | pabelanger | since voicemail can be compile 3 different ways |
| 15:11.40 | *** join/#asterisk chris_n (~Chris@nc-63-162-197-228.sta.embarqhsd.net) |
| 15:18.01 | tompaw | erm |
| 15:18.09 | tompaw | what does status = 4 in AMI's QueueStatus mean? |
| 15:18.19 | tompaw | here's what the docs say: http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+QueueStatus :D |
| 15:20.19 | tompaw | oh, it means "Invalid" |
| 15:21.55 | tompaw | That's very offensive. |
| 15:27.48 | tompaw | ah, I know. |
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| 18:53.58 | Katty | khan university is pretty sweet. |
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| 18:54.52 | Qwell | university? |
| 18:56.47 | Katty | it's a website. |
| 18:56.55 | Katty | run it through google, poke around |
| 18:56.57 | Qwell | academy |
| 18:57.21 | Katty | ah yes, so it is. |
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| 19:06.05 | newtonr | Katty: you should do an asterisk class on there |
| 19:08.01 | drmessano | In a wet t-shirt |
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| 19:11.04 | Deeewayne | does anyone know if carriers handling ISDN traffic would typically convert ISDN UUI data to SIP headers? |
| 19:14.29 | lvlolvlo | Deeewayne: I think you're looking for ISUP translations (http://tools.ietf.org/html/rfc3398) |
| 19:15.21 | lvlolvlo | and, yes -- they should be |
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| 19:18.08 | Deeewayne | lvlolvlo, thanks, I'll check that out |
| 19:18.21 | lvlolvlo | np |
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| 19:36.51 | Hive | I'm having an issue when I call AT&T conference numbers. Seemingly for no reason my call will be terminated. There doesnt seem to be any error in the asterisk logs, it just appears that the server recieves some sort of signal to hang up because there is no error displayed, the call just goes to the h context. Time doesnt seem to be the issue, one dropped at 23 minutes, and a couple have dropped at 4-6 minutes. Does anyon |
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| 19:43.25 | lvlolvlo | Hive: are you on mute? Is this a toll-free number that you're calling? |
| 19:43.57 | Hive | lvlolvlo: I am not on mute. This is a toll-free number that I am calling. It is an AT&T conference number. |
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| 19:44.46 | lvlolvlo | that's odd, most of the time i see people have this issue when they're on mute with toll-free numbers. |
| 19:45.01 | lvlolvlo | did you look at a SIP trace to see where the d/c is coming from? |
| 19:45.39 | Hive | Actually, let me double check the mute idea. |
| 19:45.53 | Hive | Just curious, why would being on mute cause a disconnect? |
| 19:46.14 | leifmadsen | possible asterisk isn't receiving audio |
| 19:46.18 | leifmadsen | then is dropping the call |
| 19:46.30 | leifmadsen | option in sip.conf to modify that is available |
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| 19:48.27 | Hive | leifmadsen: I am listening to the recordings and there is audio coming through from the other end so I don't think that is the issue. |
| 19:48.31 | lvlolvlo | leifmadsen: what is the option? is it a comfort noise option? |
| 19:48.45 | leifmadsen | lvlolvlo: no, it's has to do with rtp |
| 19:48.54 | leifmadsen | you'll have to read the sip.conf.sample as I don't remember offhand |
| 19:48.54 | Hive | lvlolvlo: I've just been told that it was, in fact, on mute |
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| 19:49.39 | lvlolvlo | Hive: it is likely then the d/c is coming from the TF provider as they don't have any RTP ingress to the TF gateway |
| 19:49.46 | lvlolvlo | so they think it is a "phantom call" |
| 19:49.53 | lvlolvlo | and disconnect it |
| 19:50.17 | lvlolvlo | remember toll-free calls are paid for by the owner, so they don't want to be charged for a call that isn't there |
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| 19:51.17 | Hive | Hmm, there must be a way to send data to keep the connection alive. |
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| 19:51.58 | Hive | I guess one solution would be for me to instruct the person dialing to speak up more :P |
| 19:52.20 | lvlolvlo | comfort noise is one way |
| 19:52.35 | lvlolvlo | RFC3389 |
| 19:53.17 | lvlolvlo | sometimes phones themselves have this option if you can't do it on your softswitch |
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| 19:56.04 | Hive | Ok I'm looking into comfort noise to see if this is a possible solution |
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| 20:05.44 | Hive | We found an option in our aastra.conf called "silence suppress" which was on by default. We're going to try turning it off and see if this fixes things. |
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| 20:06.26 | Katty | Qwell: Mad Scientist Party. |
| 20:06.33 | Katty | Qwell: Next month. Bring mentos. |
| 20:07.23 | Qwell | next month? |
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| 20:08.00 | Katty | yes, this month is nearly over. |
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| 20:08.29 | Katty | Qwell: see the line above it. |
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| 21:06.34 | kontinuity | hi all |
| 21:07.05 | kontinuity | is it possible to setup a redundant Asterisk server setup in EC2? any pointers to documents will be helpful |
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| 21:17.46 | libryder | i'm having a problem with app_swift, described here: https://gist.github.com/83545859d3c7bfb57b4e |
| 21:18.43 | libryder | for some reason swift_generator_running returns true past the time the file stops playing |
| 21:20.28 | gusto | kontinuity: well |
| 21:20.41 | gusto | i do not see any reason for running asterisk on a EC2 |
| 21:21.07 | gusto | there are vservers out there for little money and you can configure it with only the needed modules, so that he does not take too much RAM |
| 21:21.34 | gusto | because an asterisk is useless for a time and when you want it to be running 24/7 you are better off with an vserver |
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| 21:29.46 | kontinuity | gusto: we have our entire system in EC2 so its only logical we move our Asterisk servers there too |
| 21:29.59 | kontinuity | gusto: any good vserver providers that have redundancy? |
| 21:30.07 | *** part/#asterisk libryder (~libryder@libryder.com) |
| 21:35.00 | qbx | kon, just wondering, you guys have no local servers? like file, exchange, etc? |
| 21:39.17 | kontinuity | qbx: my apologies, we are a pass through VOIP service, with feature like call recording that we provide to our customers |
| 21:39.36 | kontinuity | qbx: I think I confused some folks here with my requirements |
| 21:39.47 | qbx | Oh, I have no real input.. lol. I was just curious |
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| 21:47.01 | kontinuity | so yeah its not an inhouse setup and we need to ensure 100% uptime |
| 21:50.26 | navaismo | kontinuity: not sure but i guess you can try to create a pool of servers in the cloud running the heartbeat daemon |
| 21:51.28 | kontinuity | that would solve the failover, how do I also ensure load balancing? |
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| 21:54.57 | navaismo | no idea, never installed a cluster of asterisk servers |
| 21:55.53 | kontinuity | navaismo: interesting, so how do people manage capacity issues? they simply scale vertically? |
| 21:55.59 | lvlolvlo | kontinuity: for ingress traffic with your upstreams you can use SRV records |
| 21:58.10 | navaismo | kontinuity: in my case I know the initial amount of simultaneous calls and add 20 more and buy hardware that support that, when need more pstn/e1/t1 lines just add cards |
| 21:59.59 | kontinuity | navaismo: but the CPU and other resources? |
| 22:03.14 | navaismo | thats why Im need to know maximum simultaneous calls first, usually people dont grwo from 5 to 200 in a week |
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| 22:05.47 | navaismo | s/Im/I/ |
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| 23:44.15 | ChannelZ | BROED |
| 23:44.17 | ChannelZ | oop |
| 23:51.13 | WIMPy | Go, fix some bugs. |
| 23:52.45 | jpsharp | I just squished some bugs. |
| 23:55.27 | jpsharp | damn florida palmetto bugs. |