| 00:04.57 | *** join/#asterisk NickRios05 (n=unicall2@static-200-105-140-238.acelerate.net) |
| 00:05.48 | *** join/#asterisk cesar_CR (n=cesar@201.201.41.242) |
| 00:06.09 | NickRios05 | Hello am having a little problem with my incoming calls through SIP they used to work before but now am getting No compatible codec http://pastebin.com/m2ed06a0a help plz?? |
| 00:06.22 | jblack | looking |
| 00:06.57 | jblack | It looks like you've disabled all of your protocols but g729 |
| 00:07.02 | NickRios05 | yes |
| 00:07.12 | NickRios05 | i do have g729 licences i bought from digium |
| 00:07.25 | jblack | I would check to see if the licenses are working right. |
| 00:07.36 | NickRios05 | they are, i can make calls out |
| 00:07.49 | jblack | I'd suggest you keep a backup protocol going, either ulaw or gsm, |
| 00:08.01 | jblack | well, I see "trunk1". |
| 00:08.15 | jblack | are calls going in, or our trunk1 ? |
| 00:08.18 | NickRios05 | there is another one i didnt paste trunkg729 |
| 00:08.44 | NickRios05 | and I tried with uncommenting ulaw and gsm |
| 00:08.57 | *** join/#asterisk uluatu (n=uluatu@187.58.238.14) |
| 00:09.07 | jblack | so it's possble that calls going work one way because it's using a route with gsm, and not work another way, because the route only allows g729, which may not be configured right. |
| 00:09.24 | jblack | I'm not saying that's the case; merely that it's a possibility. |
| 00:10.17 | NickRios05 | so you are saying that there is a chance that my g729 codecs are not configured right?? where can i check that?? is that i followed all the steps digium says, |
| 00:10.25 | *** join/#asterisk TSM2 (n=the_soft@87-194-32-212.bethere.co.uk) |
| 00:10.27 | jblack | I don't know. I don't support g729 |
| 00:10.58 | jblack | I would turn on verbose, debug and sip debug, and trace a call attempt |
| 00:11.16 | NickRios05 | ok let me do that and give you the pastebin |
| 00:11.16 | jblack | the routing will be intersting, as well the protocol negotiation. |
| 00:12.01 | *** join/#asterisk lost_soul (i=shawn@cpe-74-71-234-100.twcny.res.rr.com) |
| 00:12.03 | jblack | ok. I could go away at any moment without notice |
| 00:12.33 | jblack | well, I guess I just gave notice.. but I could still go away at any moment |
| 00:12.56 | *** join/#asterisk Zokeber (n=Zokeber@unaffiliated/zokeber) |
| 00:15.48 | NickRios05 | ok http://pastebin.com/m27dc001c |
| 00:19.26 | *** join/#asterisk Failrar (n=Failrar@2001:470:1f15:316:2a0:d1ff:fe4e:e802) |
| 00:20.11 | Gugge | peer only support ulaw |
| 00:20.32 | Gugge | 34.Capabilities: us - 0x100 (g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x0 (nothing) |
| 00:21.04 | NickRios05 | according to my softswitch am sending the call through ulaw |
| 00:22.08 | Gugge | try again? |
| 00:23.08 | loather-work | NickRios05: core show translation recalc 10 |
| 00:23.29 | loather-work | if you see numbers in the g729 column, you know g729 transcode is working OK |
| 00:23.33 | Gugge | the call is coming in through trunkg729, where you only allow g729 ... but the peer really only support ulaw |
| 00:23.37 | *** join/#asterisk manxpower (n=ewieling@24.42.221.26) |
| 00:23.47 | NickRios05 | ahhhh ok i see the issue |
| 00:23.53 | NickRios05 | let me try something |
| 00:25.43 | *** join/#asterisk dkirker (n=dkirker@gateway0.openmobl.com) |
| 00:30.20 | NickRios05 | ok thnx guys i solved the coded issue |
| 00:31.23 | NickRios05 | but now am getting this error am not sure http://pastebin.com/m45ed589 Address Incomplete |
| 00:32.12 | Gugge | 35.Looking for 12345 in default (domain 192.168.1.6) |
| 00:32.19 | Gugge | thats a hard one |
| 00:32.47 | Gugge | maybe try adding 12345 in your default context ? |
| 00:33.14 | NickRios05 | but isnt supposed to go to extension s if it doesnt match any other ? |
| 00:33.17 | Gugge | no |
| 00:33.46 | Gugge | The "s" extension is used when there is no known called number in the context used. |
| 00:33.51 | Gugge | and the called number is known (12345) |
| 00:34.01 | Gugge | its non existing ... but its known |
| 00:34.15 | NickRios05 | ahh ok |
| 00:35.01 | NickRios05 | thank you so much Gugge, am really new to asterisk but thnx for ur patience |
| 00:36.38 | Gugge | dont worry ... you are not the first one to think the s extension is a default one :) |
| 00:37.45 | *** join/#asterisk lost_soul (i=shawn@cpe-74-71-234-100.twcny.res.rr.com) |
| 00:38.40 | NickRios05 | hehe thnx, well thnx to all of you guys you always help ppl like me, have a good one see you some other time |
| 00:46.53 | *** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110) |
| 00:58.09 | Katty | ahhh. i'm feeling better. |
| 00:59.08 | *** join/#asterisk Iamnacho (n=Iamnacho@ip98-186-180-143.ks.ks.cox.net) |
| 00:59.14 | Katty | jblack: did you go away? |
| 00:59.22 | jblack | heh |
| 00:59.39 | jblack | I'm kinda here |
| 00:59.41 | Katty | k |
| 01:01.22 | *** join/#asterisk Caplain (i=shayne@2001:470:5:fb:43b:536f:1c09:b92a) |
| 01:01.30 | *** join/#asterisk wonderworld (n=ww@ip-62-143-22-226.unitymediagroup.de) |
| 01:06.35 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 01:06.42 | Katty | :>>> |
| 01:06.45 | Katty | hugs on jaytee |
| 01:07.01 | jaytee | hugs Katty |
| 01:07.45 | Katty | jaytee: i made homemade garlic mashed potatoes (= |
| 01:07.58 | jaytee | mmmmmm |
| 01:08.14 | Katty | they're on the stove, go get yourself some :P |
| 01:08.55 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 01:17.33 | jblack | katty: You're very focused on food. |
| 01:17.57 | Katty | you're just now realizing this? :P |
| 01:18.24 | jblack | I noticed a long while ago that 3/4 of the stuff you talk about is food. Just wondering if you'd noticed. :) |
| 01:18.33 | Katty | hmm. not really. |
| 01:18.44 | Katty | but it is a very passionate hobby of mine |
| 01:18.54 | Katty | i spend a good chunk of my day going through recipe websites and magazines. |
| 01:19.02 | Katty | and youtube videos |
| 01:20.15 | jblack | grins |
| 01:20.53 | *** join/#asterisk Caplain (i=shayne@2001:470:5:fb:43b:536f:1c09:b92a) |
| 01:20.58 | Katty | the other one is animals |
| 01:21.05 | Katty | did you see the birdie i linked today? |
| 01:21.07 | jblack | which are made out of food. |
| 01:21.42 | Katty | infobot: peekaboo? |
| 01:21.43 | infobot | rumour has it, peekaboo is Peeeeeeeeeeeeeeeeeeeeek-aboo!!!! http://www.youtube.com/watch?v=iB52iP2a_MY |
| 01:22.15 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 01:22.38 | jblack | I miss caring about things |
| 01:23.02 | Katty | ^- see 0:55 |
| 01:24.47 | jblack | Ok, I stand corrected. =) |
| 01:25.01 | Katty | about what? |
| 01:25.20 | jblack | I dunno. I didn't scroll back to 0:55, so I figured it was non-food proof. :) |
| 01:25.58 | Katty | it's an Indian Ringneck Parrot (= |
| 01:26.03 | Katty | was featured on cuteoverload a few days ago |
| 01:26.20 | jblack | hmmm. drumsticks. |
| 01:26.33 | jblack | I suppose those would be curry flavored drumsticks? |
| 01:26.39 | Katty | ^_- |
| 01:28.40 | Katty | http://cuteoverload.files.wordpress.com/2009/11/chicken-surprise.jpg <- Mister Featherhead's reaction to jblack's comment. |
| 01:29.17 | jblack | Ohhh. hotwings |
| 01:29.45 | Katty | hehe |
| 01:30.40 | Katty | may i quote you on that? |
| 01:33.04 | *** join/#asterisk Zokeber (n=Zokeber@unaffiliated/zokeber) |
| 01:33.51 | hardwire | <PROTECTED> |
| 01:33.54 | hardwire | 13713 asterisk -11 0 842m 31m 11m S 9999 0.4 4:02.30 asterisk |
| 01:33.56 | hardwire | :P |
| 01:37.51 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 01:42.36 | Katty | hi tony |
| 01:51.10 | hardwire | tony tony so bony banana fana fo fony |
| 01:51.50 | *** join/#asterisk Iamnach0 (n=Iamnacho@ip98-186-180-143.ks.ks.cox.net) |
| 01:52.11 | *** join/#asterisk voipmonk (n=voipmonk@67.204.57.187) |
| 01:52.19 | Katty | mister monk |
| 01:53.21 | voipmonk | hello :) |
| 01:56.06 | *** join/#asterisk Caplain (i=shayne@2001:470:5:fb:43b:536f:1c09:b92a) |
| 01:59.18 | *** join/#asterisk bbt (n=sam@180.189.139.92) |
| 02:04.15 | *** join/#asterisk hobodave_ (n=hobodave@63.149.16.13) |
| 02:04.21 | *** join/#asterisk haps (n=hapoteh@yossman.net) |
| 02:04.25 | haps | howdy |
| 02:04.31 | Katty | herroes. |
| 02:04.51 | haps | just wondering what magic i need to do to get asterisk to change the outging callerid based on area code |
| 02:05.21 | Katty | _1888 |
| 02:05.24 | Katty | _1413 |
| 02:05.27 | Katty | etc |
| 02:05.50 | Katty | asuming you don't have pots lines. |
| 02:05.54 | haps | i have, in my [internal-sip] dialplan, something like exten=>_514NXXXXXX,n,Set(CALLERID(all)=abc <number>) |
| 02:06.40 | haps | with an exten=>_NXXNXXXXXX,n,Set(CALLERID(all)=xyz <number>) |
| 02:06.51 | haps | oh fuck, is it this stupid, is it first match or last match? |
| 02:07.29 | voipmonk | yawns |
| 02:07.33 | russellb | haps: huh? |
| 02:07.37 | haps | fuck me i'm retarded |
| 02:07.39 | haps | never mind |
| 02:07.51 | *** join/#asterisk CcRnp (n=shishir@208.179.165.18) |
| 02:07.54 | Katty | if you have exten => 555,1,Dial(SIP/555) and exten => 555,1,Dial(SIP/557) after it...it will dial 555 |
| 02:07.54 | haps | i had my default callerid set AFTER my specific ones |
| 02:07.58 | haps | i thought it was first match |
| 02:08.06 | russellb | they get sorted. |
| 02:08.16 | russellb | doesn't matter how you put them in your file |
| 02:08.37 | haps | russellb: no it does, because i put my catch-all first, and my specific ones after and now it works right |
| 02:08.40 | haps | it's last match |
| 02:08.42 | haps | not first |
| 02:08.53 | *** join/#asterisk evil_gordita (n=evilgord@ip70-188-50-186.rn.hr.cox.net) [NETSPLIT VICTIM] |
| 02:08.53 | *** join/#asterisk [netman] (n=netman@85.Red-83-32-24.dynamicIP.rima-tde.net) [NETSPLIT VICTIM] |
| 02:08.53 | *** join/#asterisk hobodave (n=hobodave@63.149.16.13) [NETSPLIT VICTIM] |
| 02:08.53 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 02:08.57 | haps | too used to firewall rules |
| 02:09.02 | russellb | i promise you that it sorts them, heh |
| 02:09.30 | *** join/#asterisk chendy (n=chatzill@219.133.69.81) |
| 02:09.46 | haps | russellb: i had exten=> areacode1, set callerid abc; then exten => all area codes, set callerid xyz |
| 02:09.57 | haps | and now i reversed it and it's working as expected |
| 02:09.58 | haps | so... |
| 02:10.09 | haps | russellb: maybe sorting is brand new? |
| 02:10.14 | russellb | has been there for ages |
| 02:10.23 | russellb | do it one way, look at the output of "dialplan show" |
| 02:10.28 | russellb | then change it, reload, and look again |
| 02:10.32 | russellb | it _should_ show you the same thing |
| 02:10.52 | russellb | it outputs the extensions in their sorted matching order |
| 02:10.56 | hardwire | haps: i usually have a context for outbound calls where I set the caller id and then goto the actual outbound context for all NXXXXXX |
| 02:11.09 | hardwire | so if it matches, set the caller id then go to the real context. |
| 02:11.20 | hardwire | saves on dialplan |
| 02:11.41 | Katty | water |
| 02:11.42 | Katty | WATER |
| 02:11.44 | Katty | runs off |
| 02:12.05 | haps | hardwire: good idea |
| 02:12.12 | haps | i'm afraid i'm a bit too simple for that just yet |
| 02:13.47 | p3nguin | hardwire: That is good logic. |
| 02:13.49 | Katty | haps: you'll get there. |
| 02:14.00 | *** join/#asterisk toddejohnson (n=toddejoh@adsl-70-226-237-31.dsl.spfdil.sbcglobal.net) |
| 02:14.22 | haps | Katty: possibly, i'm more of a 'set it and forget it' type though :( |
| 02:14.41 | haps | so unless i have more needs.. i just needed to change my callerid for certain area codes |
| 02:14.49 | haps | makes remote consulting work easier |
| 02:14.55 | haps | ie i look 'local' to them |
| 02:15.04 | Katty | everytime i upgrade, or build a new server, my dialplan becomes more and more organized. |
| 02:17.44 | p3nguin | hardwire: If you do it that way, where is the CallerID being set when it doesn't match one of your extens that sets it? |
| 02:18.07 | haps | russellb: well, my only counter to that is the fact that i swapped 2 lines in my extensions.conf and now my callerid is different... this is a set call though so it might be different? |
| 02:26.28 | Katty | ryan's recording a new song. |
| 02:26.38 | Katty | there's audio equipment /everywhere/ |
| 02:28.51 | russellb | haps: it would be useful to verify it, as it shouldn't change behavior. If it did, it would be a bug, and I'd want to know that as breaking dialplan matching behavior is pretty serios. |
| 02:28.55 | russellb | s/serios/serious/ |
| 02:29.46 | haps | russellb: i don't know if it's a dialplan change though, since it's in one dialplan but only a set call |
| 02:29.59 | haps | it isn't changing anything else |
| 02:30.58 | p3nguin | If you bothered to paste your config before and after, someone else could form an opinion about it. |
| 02:33.19 | haps | it's really basic, but here: http://pastebin.com/m28b64441 |
| 02:33.26 | haps | that is literally the only change |
| 02:33.37 | haps | the other stuff is all _NXXNXXXXXXX stuff |
| 02:33.45 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 02:35.02 | [TK]D-Fender | haps: exten => _514NXXXXXX,n,Set(CALLERID(all)=XYZ <416xxxxxxx>) <-- you shoved in an "n" priorit without a "1" <----------- |
| 02:35.17 | russellb | oh neat ... i'm not even positive what to expect with swapping around n's like that |
| 02:35.25 | [TK]D-Fender | haps: Every pattern needs to have its own "1" or you're going to break stuff |
| 02:35.37 | haps | oooh |
| 02:35.41 | haps | good to know :) |
| 02:35.56 | russellb | at least to have the most predictable behavior :-) |
| 02:36.07 | haps | i thought within a 1, i could split the dependence based on the subtype? |
| 02:36.29 | haps | like A,1 -> A_1,n,doA1, A_2,n,doA2 |
| 02:36.31 | [TK]D-Fender | haps: No, but I'll have some of whatever you're on :) |
| 02:36.35 | haps | heh |
| 02:36.42 | russellb | yeah, I have seen dialplans do it. If you're going to do that, though, use numbered priorities so that it does exactly what you want |
| 02:36.43 | haps | ignorance and bliss? |
| 02:36.48 | Katty | Corydon76-dig: garnier is giving away free samples of eye cream. |
| 02:36.50 | russellb | like, in that case, number them both as '2' instead 'n' |
| 02:36.57 | haps | aha |
| 02:36.59 | haps | ok |
| 02:37.02 | haps | russellb: thanks |
| 02:37.20 | russellb | otherwise i don't even know what it would do, i'd have to look at the dialplan show, heh |
| 02:37.20 | haps | that is making a lot of sense from the 'now that i know' side |
| 02:37.29 | russellb | i'll call it "undefined" |
| 02:37.46 | russellb | yay for now knowing what happened |
| 02:37.51 | haps | garbage in ... for some reason it gave me good output |
| 02:37.57 | haps | so i have the 1, set caller pres |
| 02:38.19 | haps | and 2, 515 with a 2, NXX area code set caller id, then everything else is n |
| 02:38.25 | haps | does that sound more correct? |
| 02:38.46 | russellb | yeah, that should do it. |
| 02:40.08 | haps | seems to work.... thanks |
| 02:40.24 | haps | and here i was thinking i didn't need to paste... pfft |
| 02:40.31 | haps | thanks p3nguin :) |
| 02:42.23 | Katty | http://tinyurl.com/5dt69q <- free coffee sample. |
| 02:43.05 | haps | did you just post a walmart link? |
| 02:43.09 | haps | schade... |
| 02:43.33 | Katty | you expect me to not link free caffeine? ;) |
| 02:44.50 | jblack | <PROTECTED> |
| 02:46.15 | [TK]D-Fender | no, $1000 coffee is grand :) |
| 02:47.30 | russellb | Katty: spammer! |
| 02:48.19 | *** join/#asterisk lost_soul (i=shawn@cpe-74-71-234-100.twcny.res.rr.com) |
| 02:48.21 | haps | it's just walmart is soo.... anti-capitalist |
| 02:52.29 | [TK]D-Fender | BBL |
| 02:53.22 | *** join/#asterisk Caplain (i=shayne@2001:470:5:fb:43b:536f:1c09:b92a) |
| 02:59.38 | *** join/#asterisk acxty (i=x3rus@201.220.136.117) |
| 03:02.55 | Katty | russellb: you know you like free coffee |
| 03:03.16 | russellb | yes. |
| 03:03.23 | russellb | we have free coffee at work <3 |
| 03:03.45 | acxty | Hi guys, I have a problem trying to make a call. I can receive calls using that sip accound but cannot make. Thi is the configuration on sip.conf http://dpaste.com/123180/ |
| 03:05.09 | russellb | your Dial() syntax is not correct |
| 03:05.43 | russellb | not "SIP/5011414-out/5551010", use "SIP/5551010@5011414-out" |
| 03:09.17 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 03:10.47 | acxty | working now, I addef fromuser on sip.conf |
| 03:11.06 | jaytee | wb [TK]D-Fender |
| 03:11.55 | [TK]D-Fender | jaytee: I'm getting ready to clone my Ubuntu partitions to a separate SSD shortly... |
| 03:13.28 | jaytee | what are you using for cloning? |
| 03:16.59 | [TK]D-Fender | jaytee: Thats what I'm working on now... doing a lot of reading. |
| 03:17.34 | [TK]D-Fender | jaytee: looking like a livecd boot and dd+grb+partimage+idunno |
| 03:18.06 | [TK]D-Fender | jaytee: just seeing which bits match the partition resizing I need to do and will let me hand-pick the ones to do |
| 03:22.57 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 03:26.50 | carrar | mmmm sheep cloning |
| 03:27.03 | carrar | THATS HOT++ |
| 03:34.44 | Katty | :< |
| 03:35.42 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) [NETSPLIT VICTIM] |
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| 03:37.45 | [TK]D-Fender | Scotland... where men are men, and sheep are nervous |
| 03:37.55 | [TK]D-Fender | . |
| 03:48.25 | jblack | There's nothing quite as sexy as a sheep |
| 03:48.51 | jblack | "You treat me so baaaaAAAaaAAadd" |
| 03:53.00 | *** join/#asterisk syntaxman (i=wade@kernel-panic/member/syntaxman) |
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| 04:02.11 | *** part/#asterisk horsesgofaster (n=daveC@c-24-0-115-52.hsd1.nj.comcast.net) |
| 04:05.12 | Katty | well. |
| 04:05.15 | Katty | the good news is the dog is clean. |
| 04:08.09 | *** join/#asterisk Iamnacho (n=Iamnacho@ip98-186-180-143.ks.ks.cox.net) |
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| 04:12.09 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
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| 04:19.47 | Katty | http://whathub.com/outreach.html <- more free coffee |
| 04:20.37 | *** join/#asterisk Hunner (i=hunner@firefly.cat.pdx.edu) |
| 04:25.56 | carrar | Katty, drinking that coffee will make someone homo? |
| 04:26.06 | Katty | i wouldn't think so |
| 04:26.27 | carrar | Powerfull stuff |
| 04:26.43 | carrar | Thats why I make my own coffee/espresso |
| 04:26.48 | Katty | personal preference (= |
| 04:26.52 | carrar | heh |
| 04:27.11 | carrar | I use the stnx venti cups at home |
| 04:27.13 | carrar | stbx |
| 04:27.24 | carrar | cost me a $1 per venti cup |
| 04:27.30 | carrar | added it all up one time |
| 04:27.52 | carrar | hazzlenut, soymilk, coffee |
| 04:28.39 | carrar | vs what, they charge like $4.50 or so for that same drink |
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| 05:25.14 | QAChip | Hello everyone, Im newbie to asterisk and have a question |
| 05:26.12 | QAChip | Is it possible to use asterisk the way Skype does (but better if course)? |
| 05:27.15 | QAChip | i think I wrote a mess, one more time: Is it possible tu use Asterisk like Skype (but better)? |
| 05:32.24 | n0cturnal | QAChip you'll need to be a bit more specific.. what features from skype do you want to use? |
| 05:33.09 | ppc | but better? |
| 05:33.10 | n0cturnal | if you mean you want to use it through your computer via a softphone you can - but that's not asterisk. there are several softphones that would work with asterisk, ie xlite |
| 05:33.12 | ppc | how much better? |
| 05:33.19 | QAChip | n0cturnal: Ony VoIP |
| 05:33.32 | ppc | QAChip: just stick with Skype |
| 05:34.45 | jblack | QAChip: Yes. |
| 05:34.47 | QAChip | ppc: I know Skype is the easyest way, but I want to start learning Asterisk, and I "think" this could be a good aproach |
| 05:34.58 | carrar | ~book |
| 05:34.59 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 05:35.04 | carrar | read that |
| 05:35.05 | jblack | QAChip: but you need to understand unix well, and probably spend a few weeks learning asterisk |
| 05:38.21 | QAChip | jblack:, infobot: I already started reading that, but it is huge. Any specific chapter for my need? |
| 05:40.43 | snadge | im moving house soon.. and theres these new "naked" adsl packages that dont come with land lines.. so im looking into running an asterisk server at home, with a VOIP provider.. and an ip phone |
| 05:41.01 | snadge | and maybe link it with my skype just to be "cool" |
| 05:41.39 | snadge | and have an IVR answer my home phone :P |
| 05:41.46 | *** join/#asterisk gmarsh_ (n=gmarsh@c-98-220-15-168.hsd1.in.comcast.net) |
| 05:43.13 | p3nguin | snadge: Might as well. I don't have POTS service at home, but I have a cable modem and a puny little computer running * on it. |
| 05:43.55 | snadge | if you would like to talk with the owner.. please press 1.. your call is important to us.. just keeps looping them around and frustrating the shit out of them |
| 05:44.07 | snadge | meanwhile i tell my friends the secret code to just skip it and get straight through to me ;) |
| 05:44.14 | snadge | and record the results.. for hilarity |
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| 06:48.47 | dan__t | 'evening. |
| 06:49.11 | TJNII | snadge: I'm a fan of the "queue" that isn't a queue. It is a short loop of MOH in a for() loop interspersed with messages like "your call is important to us" and "Are you aware of our priority services" |
| 06:49.27 | TJNII | Got the idea from Briggs and Stratton. |
| 06:49.44 | TJNII | Give away was the MOH changes and someone answered within seconds. |
| 06:54.56 | dan__t | Well. I still can't find a way to "shadow" a channel that's already attached to a bridge or MeetMe. |
| 06:56.07 | dan__t | I want a channel "attached" to another channel, such as what ChanSpy would do, where the spying channel could only listen to that one spied on channel, regardless of what "conference" state (bridge, MeetMe, etc etc) the spied on channel was in. I'd also want the spied on channel not to be able to hear anything that the spying channel was saying. |
| 06:56.20 | dan__t | Been hacking on it for a while and I can't find anything that works. |
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| 07:34.08 | PMantis | Hi guys, trying to figure out why * won't use my dahdi channels. "dahdi show status" shows my TDM400P card, but "dahdi show channels" doesn't show channels - not even pseudo. I must be missing something obvious. |
| 07:34.26 | ChannelZ | Did you configure them in /etc/asterisk/chan_dahdi ? |
| 07:34.29 | ChannelZ | +.conf |
| 07:35.14 | PMantis | chan_dahdi.conf includes dahdi-channels.conf, but I tried removing the include, and pasting it there, too. |
| 07:35.41 | ChannelZ | mphm.. freepbx? |
| 07:35.42 | PMantis | Permissions are right on the conf, as well as /dev/dahdi, recursively. |
| 07:35.54 | PMantis | No |
| 07:36.37 | PMantis | I used dahdi_genconf... trying to get this nailed down so i can script it. |
| 07:36.41 | ChannelZ | well without seeing your config, do a 'module reload chan_dahdi' on the console with the verbosity turned up a little, does it say anything fascinating? |
| 07:37.01 | PMantis | You mean with asterisk -crvvvvvvvvvvvvvvvvvvvvvvv ? |
| 07:37.39 | ChannelZ | well 4 v's or so should be enough to get an idea of whats going on but yes |
| 07:37.50 | PMantis | :-) |
| 07:39.20 | PMantis | Right now... nothing returned from the reload command. |
| 07:39.35 | ChannelZ | absolutely nothing? |
| 07:39.47 | PMantis | Yeah, just drops to the CLI prompt. |
| 07:39.57 | PMantis | Tried a dahdi restart, too. |
| 07:40.25 | PMantis | I used sed to strip all comment lines from chan_dahdi.conf to see what it boils down to... maybe something starting with ; was needed? |
| 07:40.25 | ChannelZ | errr... it should be saying *something* |
| 07:40.38 | ChannelZ | pastebin it |
| 07:40.43 | PMantis | np |
| 07:41.57 | PMantis | http://www.pastebin.ca/1680575 |
| 07:42.41 | ChannelZ | you have no channel definitions that assign the channels to anything |
| 07:42.50 | PMantis | and http://www.pastebin.ca/1680576 |
| 07:43.18 | PMantis | The first one includes the second. |
| 07:43.30 | ChannelZ | ah |
| 07:44.44 | PMantis | I also just moved this card from a working system to this box I'm using to replace it. |
| 07:44.49 | ChannelZ | well I must say I am confused because when you reloaded the dahdi module from the console it should have at LEAST said "Reloading module 'chan_dahdi.so' (DAHDI Telephony)" and something about parsing the config file |
| 07:45.20 | PMantis | I saw that a few times, and I agree. But as of right now, it's not. :-/ |
| 07:45.24 | ChannelZ | did you build the dahdi drivers and * anew? (did it have any dahdi hardware before?) |
| 07:46.15 | PMantis | It's actually from the Ubuntu 9.10 repository. I usually compile them, but thought this might be more maintainable with DKMS. |
| 07:46.36 | PMantis | No other cards were in this box since loading Ubuntu. |
| 07:47.44 | PMantis | I just ran dahdi_cfg again... here's one line from it (avoiding pastebin): Channel 01: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 01) |
| 07:48.40 | ChannelZ | hmm |
| 07:49.50 | PMantis | Interesting. |
| 07:50.11 | PMantis | I just ran module unload chan_dahdi.so, then loaded... |
| 07:50.22 | PMantis | (should be same as reload, right?) |
| 07:50.32 | ChannelZ | eh? |
| 07:50.34 | PMantis | It FOUND the channels! |
| 07:50.39 | PMantis | confused |
| 07:50.52 | PMantis | Got dialtone, too. |
| 07:50.56 | ChannelZ | oh. So that means you don't have dahdi_cfg in your init script or something when the drivers start |
| 07:51.14 | PMantis | Hmmmm |
| 07:51.14 | ChannelZ | or something failed the first time and it just didn't configure the channels |
| 07:51.44 | PMantis | I can't tell you how many times I ran dahdi_cfg and dahdi_genconf, stopped and started asterisk, etc. |
| 07:51.49 | ChannelZ | I dunno how those packages are, but you probably have an /etc/init.d/dahdi or somesuch that loads the proper driver and (should) run dahdi_cfg afterwards |
| 07:52.48 | PMantis | Yeah, and I wrote my own init script that checks for a valid driver and /dev/dahdi/ctl , and if not modprobes dahdi_dummy, then run dahdi_cfg. |
| 07:53.07 | ChannelZ | well you need more than just dahdi_dummy |
| 07:53.25 | PMantis | udev is loading wctdm |
| 07:54.04 | ChannelZ | (and actually with the TDM card you don't even need dahdi_dummy) |
| 07:54.29 | PMantis | Right, I check to see if /dev/dahdi/ctl exists, and if not, then load the dummy. |
| 07:54.32 | ChannelZ | though I'm not sure if it would interfere or not.. |
| 07:54.56 | ChannelZ | I'd just reboot the whole box and see if it comes up proper, otherwise you have some funniness with your scripts |
| 07:55.05 | PMantis | I dunno... it's 3 AM here, maybe I was just tired... but something doesn't make sense. |
| 07:55.50 | PMantis | Still, it looks like I'm understanding the puzzle. I'll hit it again tomorrow. |
| 07:55.52 | PMantis | Thanks! |
| 07:56.11 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) |
| 07:56.20 | ChannelZ | sure have fun |
| 07:56.55 | ChannelZ | jebus.. how many times is tzdata going to get updated? |
| 07:57.11 | PMantis | lol |
| 07:57.25 | ChannelZ | seems like every time I update there's a new one |
| 07:57.51 | PMantis | Yup, I see it on all the machines I maintain. |
| 07:58.05 | PMantis | Debian, Ubuntu, RedHat |
| 07:59.21 | ChannelZ | I need to update my linux box hardware. This computer is sad, it only has USB1. It's actually faster to fire up another box in the other room and run backups over the network to an external drive than to hook the drive up to the actual server |
| 08:00.49 | PMantis | LOL |
| 08:00.59 | PMantis | NFS vs USB1 |
| 08:01.00 | PMantis | Hmmmmm |
| 08:01.21 | PMantis | You can always get a USB 2.0 PCI card |
| 08:01.55 | *** part/#asterisk icyValk77 (n=icyValk7@host81-155-31-214.range81-155.btcentralplus.com) |
| 08:02.41 | PMantis | Oh, anyone know how to get rid of the dozen "doing dnsmgr_lookup for..." lines scrolling by in the CLI evert 20 seconds? It's REALLY annoying, and appears even without verbosity. |
| 08:03.35 | ChannelZ | yeah |
| 08:03.54 | ChannelZ | (to the USB card, not the dns thing) |
| 08:04.23 | PMantis | Most of the time, I just don't *care*. I want to see *call* activity only unless I ask for more with set debug on commands. |
| 08:05.17 | ChannelZ | do you use dnsmgr? if not just move/rename the config |
| 08:05.47 | PMantis | I've never played with it, so does it default to on? What's the advantage? |
| 08:06.15 | ChannelZ | (or do a noload in the modules.conf for it) |
| 08:07.31 | PMantis | module show like dns returned 0. |
| 08:07.43 | ChannelZ | it's some sort of DNS lookup cache but I imagine unless you are doing a bunch of random VoIP to random places it's not useful |
| 08:07.46 | PMantis | BTW, look at this: http://www.newegg.com/Product/Product.aspx?Item=N82E16815123010&cm_re=usb_pci-_-15-123-010-_-Product |
| 08:08.12 | *** part/#asterisk dgoner (n=david@c-71-201-22-50.hsd1.il.comcast.net) |
| 08:08.19 | PMantis | $6 + (probably) modest shipping is not bad for faster backups. :) |
| 08:08.59 | ChannelZ | maybe it's not a module.. just move the config |
| 08:09.49 | PMantis | Hmmm, I don't have a dnsmgr.conf file. LOL |
| 08:09.57 | ChannelZ | haha newegg wants to sell me an extended warranty for $6.99 - a dollar more than the card is worth |
| 08:10.04 | PMantis | ROFL |
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| 08:10.49 | PMantis | Need firewire? |
| 08:11.19 | PMantis | Oh, here's 5 ports for $8 with free shipping: http://www.ledshoppe.com/Product/com/CA4010.htm |
| 08:11.48 | ChannelZ | nope |
| 08:12.14 | ChannelZ | yah there's a mess of them on amazon for ~$7 and I get free 2-day there |
| 08:12.26 | PMantis | Nice |
| 08:13.12 | PMantis | ok, well, definitely time for bed... have a good night. |
| 08:13.48 | ChannelZ | nighty |
| 08:13.50 | PMantis | Hope a couple of those links were helpful. |
| 08:13.56 | PMantis | cya |
| 08:13.57 | *** part/#asterisk PMantis (n=sswitzer@cpe-67-244-157-0.rochester.res.rr.com) |
| 08:13.58 | ChannelZ | yah thanks |
| 08:16.27 | *** part/#asterisk geoffmcc (n=geoffmcc@cpe-76-180-149-84.buffalo.res.rr.com) |
| 08:23.38 | *** join/#asterisk DelphiWorld (n=Miranda@196.20.95.121) |
| 08:23.40 | DelphiWorld | hi |
| 08:23.42 | DelphiWorld | dear all |
| 08:24.02 | DelphiWorld | asterisk is using h323+/openh323/... to build the chanH323 or is using there own implementation? |
| 08:33.37 | *** part/#asterisk DelphiWorld (n=Miranda@196.20.95.121) |
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| 10:11.32 | Godfather_ | how can i disable musiconhold after X seconds? cause in 1.4 i dont have MusicOnHold(class[,duration]) |
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| 11:08.25 | Godfather_ | how can i disable musiconhold after X seconds? cause in 1.4 i dont have MusicOnHold(class[,duration]) |
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| 11:42.25 | vally | i have a dial in my dialplan that connects to a channel streaming sound do an external application. now i want simultaneously use streamFile (or any other method that plays sound to a connected phone) while that Dial occurs. is there any way to accomplish this? |
| 11:43.13 | vally | at the moment i have an agi that uses streamFile. but that agi stays open all over the call and so the method blocks. and it never reaches the dial in the dialplan. |
| 11:43.25 | vally | same thing vice versa if i place the dial and then agi. |
| 11:43.27 | vally | :-( |
| 11:47.13 | TJNII | Stream the stream through MOH and have dial play moh while dialing. |
| 11:49.44 | vally | the files should be played in order to a user interaction in an external interface. so thats not a static setup. can this still be accomplished using moh? |
| 11:50.27 | TJNII | So this stream is interactive? |
| 11:51.30 | vally | a user on a website can click on several links that interact with a specific sound file that should be streamed to the phone then. |
| 11:52.07 | TJNII | So * is just playing the stream, and not providing the method of interaction? |
| 11:52.14 | vally | yes |
| 11:52.31 | TJNII | Yea, you can do that through moh. |
| 11:52.38 | vally | i need just an asynchronous non-blocking way to play a file |
| 11:53.19 | TJNII | Yea, the best way I can think of to make it non-blocking during a dial is to play the stream through moh. |
| 11:54.18 | vally | okay, i will have a look at it. although its a pity because i have both parts ready now. but they dont work together ;) |
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| 11:58.48 | EugenA | hi, if i create "user" from asterisk gui, should i see new user in sip.conf? |
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| 12:36.17 | dandate2 | <dandate2> anyone know what this is about? |
| 12:36.17 | dandate2 | <dandate2> file convert /var/lib/asterisk/mohmp3/fpm-calm-river.wav /var/lib/asterisk/mohmp3/fpm-calm-river.g729 |
| 12:36.17 | dandate2 | <dandate2> Failed to convert /var/lib/asterisk/mohmp3/fpm-calm-river.wav to /var/lib/asterisk/moh/fpm-calm-river.g729! |
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| 13:40.24 | kaldemar | dandate2: do you have a g.729 codec? |
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| 13:57.50 | tzafrir | Any of the ops here around? |
| 14:08.40 | Corydon76-dig | Nope |
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| 14:32.37 | *** mode/#asterisk [-b *!*@*unaffiliated/mchou] by Corydon76-dig |
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| 15:50.53 | nicola_pav | hello. I use ForkCDR() with option "a" |
| 15:51.07 | nicola_pav | this yields to two entries in Master.csv |
| 15:51.21 | nicola_pav | the first one has duration and billsec "0" |
| 15:51.31 | nicola_pav | the second entry will have correct values |
| 15:51.36 | nicola_pav | is this right? |
| 15:54.58 | *** join/#asterisk Xetrov` (n=xetrov@unaffiliated/xetrov/x-827361) |
| 15:57.16 | andrew` | this is a bit embarassing, but my asterisk system was hijacked and used for a credit card phishing scam across multiple states of the US. Does anyone know what I should do? |
| 15:58.26 | voipmonk | shut it down |
| 15:58.31 | voipmonk | save the drive |
| 15:58.33 | voipmonk | and start over |
| 15:58.42 | voipmonk | eventually the authorities will contact you |
| 15:59.09 | voipmonk | start over with a new drive, rather - save the old one or ones |
| 15:59.11 | andrew` | local police came by to give me a case number, but said their 2 person fraud department wouldn't know what to do |
| 15:59.28 | voipmonk | the feds will |
| 16:00.03 | voipmonk | dont be afraid just save the drive and start over |
| 16:00.28 | andrew` | i'm not afraid, i'm just figuring i should try to get the evidence to the reight person |
| 16:00.36 | coppice | andrew: be very very careful in your dealings over this. a lot of innocent partied get badly screwed in these situations |
| 16:02.02 | andrew` | how so? |
| 16:02.45 | voipmonk | $15k per dnc list number for starters :) |
| 16:03.03 | coppice | because if the authorities decide to prosecute you most lawyers will tell you to plea bargain and settle. juries never understand these cases, and the outcome of a trial is too uncertain |
| 16:03.05 | voipmonk | kidding |
| 16:03.27 | voipmonk | http://www.munciefreepress.com/node/21423 |
| 16:03.33 | andrew` | if it came to that, i couldn't plea bargain or I'd likely be deported |
| 16:04.11 | *** join/#asterisk moos3 (n=rgenthne@cpe-76-179-253-210.maine.res.rr.com) |
| 16:04.24 | moos3 | can anyone recommend sip provider for home service? |
| 16:04.30 | coppice | they you could be the perfect fall guy the authorities are looking for to improve their clear up figures |
| 16:04.57 | andrew` | I will not be anyone's fall guy :P |
| 16:06.39 | Corydon76-dig | The FBI generally isn't looking for a fall guy. As long as they think that you've done your best to preserve evidence and aren't trying to hide anything, you won't become a focal point of the investigation |
| 16:06.56 | andrew` | someone's calling me from texas now...returning the missed call from the scam |
| 16:07.02 | andrew` | i had around 200 people do that yesterday |
| 16:07.15 | andrew` | (they used my cell phone caller ID, which was default in my asterisk outbound macro) |
| 16:08.02 | Corydon76-dig | Now if they think you're hiding something -ANYTHING-, you're screwed |
| 16:10.07 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 16:10.35 | andrew` | yeah, i don't have anything to hide |
| 16:10.45 | andrew` | i actually had already called the FBI before I found out my computer was involved |
| 16:10.47 | *** join/#asterisk Arsenick (n=rpurcell@modemcable022.82-21-96.mc.videotron.ca) |
| 16:10.57 | andrew` | i thought the scammers had randomly chosen my # as caller ID |
| 16:11.17 | andrew` | when I got home I went to use my asterisk system to record the voicemails i've received and saw the console messages that it was dialing the same area codes that were calling me |
| 16:11.33 | andrew` | that's when i turned it off and called the police |
| 16:12.30 | drmessano | Was it a trixbox? |
| 16:12.37 | andrew` | no |
| 16:12.47 | andrew` | OpenBSD machine running asterisk |
| 16:13.32 | andrew` | what's strange is when i ran last -10, it appeared nobody had logged in to the machine...is there some vulnerability in asterisk that they could get in? i haven't upgraded in a while |
| 16:13.38 | andrew` | 1.4.something |
| 16:13.55 | andrew` | (I figured who would try to hack my little system) |
| 16:14.19 | drmessano | Theres always vulnerabilities in everything... asterisk is no more or less susceptible |
| 16:14.30 | andrew` | yup |
| 16:14.48 | drmessano | Which is why you always keep up, especially if it's exposed to the outside |
| 16:15.04 | Amorsen | Either that or they guessed a password or got in through allowguest=yes |
| 16:15.32 | andrew` | can they modify the dial plan without gaining shell access? |
| 16:15.46 | hardwire | only if you let them. |
| 16:15.50 | andrew` | i've been using asterisk for years, but never more than playing around |
| 16:16.25 | hardwire | asterisk is like a web server that people can use to access other web servers :) So you need good input validation. |
| 16:16.56 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 16:17.19 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 16:17.31 | Amorsen | Fortunately Asterisk comes with really good tools for input validation, such as CUT and ... err... that's basically it |
| 16:18.26 | [TK]D-Fender | Scissors... the only tool you need for string manipulation! |
| 16:18.57 | hardwire | heh |
| 16:19.04 | Amorsen | Well all you really need is 3 stones and an infinite roll of paper |
| 16:19.05 | drmessano | Asterisk also comes with something called a dialplan which is good for directing calls exactly as you want them inside the system |
| 16:19.17 | hardwire | yay matching! |
| 16:19.32 | drmessano | If you got pwned by a guest account, you need to reread the book, starting with the reciept |
| 16:20.08 | Amorsen | drmessano: True, I was being a bit facetious. It's just that if you try to handle say untrusted X-whatever SIP headers, it's difficult to do it safely |
| 16:20.35 | hardwire | firewall yarrr |
| 16:20.39 | hardwire | allow/deny yarrrr |
| 16:21.03 | hardwire | goes off pirating. |
| 16:21.36 | [TK]D-Fender | Amorsen: No, I agree, *'s lack of full regex sucks. The dialplan pattern system sucks, etc. However I haven;t contributed code to do anything about it so my ability to complain is limited |
| 16:22.02 | drmessano | Yeah, I know.. but it annoys me when someone leaves a web server exposed as shit, gets hacked, and someone comes along and complains that apache doesn't validate gopher requests properly.. What about leaving the machine wide open.. did we miss that? |
| 16:22.39 | [TK]D-Fender | Amorsen: add to that list :no typed vars / everything = dumb text / escaping stuff doesn't always cut it, etc |
| 16:23.19 | Amorsen | Indeed |
| 16:23.33 | andrew` | yes i think i left the asterisk ports wide open |
| 16:23.47 | drmessano | andrew`: As do many of us.. |
| 16:23.57 | Amorsen | andrew`: allowguest? And default context? |
| 16:24.13 | drmessano | andrew`: Having ports open isn't a problem.. shitty dialplan and old code are more likely |
| 16:24.16 | Amorsen | DISA is a killer too |
| 16:24.17 | andrew` | i doubt i used allowguest |
| 16:24.28 | Amorsen | andrew`: It's the default config |
| 16:24.37 | drmessano | weak passwords? |
| 16:24.43 | andrew` | i know i turned it off at some point...not sure if on this time around |
| 16:24.48 | drmessano | a peer named [] ? |
| 16:24.50 | Amorsen | There was a crispy flame war about it recently on the mailing list |
| 16:25.16 | andrew` | maybe i should duplicate the disk drive and then do some investigation on the copy |
| 16:25.18 | [TK]D-Fender | [11:24]<Amorsen>andrew`: It's the default config <_ DEFAULT? There is no "default" |
| 16:25.38 | Amorsen | [TK]D-Fender: If you don't specify allowguest, you get allowguest=yes |
| 16:25.40 | Amorsen | That's default. |
| 16:26.06 | [TK]D-Fender | Well if you leave [general] pointing to a context that has outbound access then you are a twit |
| 16:26.54 | Amorsen | True |
| 16:27.16 | [TK]D-Fender | YouAlso if you even have a context named [default] you should also be shot... |
| 16:27.59 | [TK]D-Fender | People gang-pile their crap all over the place because of a lack of understanding on the importance of contexts for security |
| 16:28.49 | [TK]D-Fender | Letting generic IVR's do things they're not supposed to, letting caller's transfer THEMSELVES,e tc |
| 16:29.11 | moos3 | i need a good sip provider for my house |
| 16:29.11 | moos3 | recommendations |
| 16:30.15 | Amorsen | "Register with the address of the SIP provider" is also a funny little trick, for people who don't use acl's |
| 16:30.36 | Amorsen | At least that one requires authentication so you can twap the luser when you catch him |
| 16:33.53 | *** join/#asterisk elliot98 (n=elliot@unaffiliated/elliot98) |
| 16:34.53 | florz | [TK]D-Fender: Well, if you write software where special values in some general namespace do have special meaning with potentially catastrophic security implications ... guess what should be done to you ... |
| 16:35.17 | *** join/#asterisk Arsenick (n=rpurcell@modemcable022.82-21-96.mc.videotron.ca) |
| 16:37.31 | [TK]D-Fender | florz: So if I leave a special setting like "locked" off my front door I don't deserve what's coming? |
| 16:38.42 | [TK]D-Fender | florz: and that doesn't forgive to context blunder. If your idea is that in "general" things have access to services that can bill you.... then you are straight-up dumb |
| 16:40.10 | florz | [TK]D-Fender: you are implying that the name "default" would only ever be chosen for your concept of "general" for which it wouldn't matter |
| 16:40.24 | florz | that's basically exactly the fault in that construction |
| 16:40.51 | [TK]D-Fender | florz: No.... [deafult] is where * jumps on all sorts of failures.. including not specifying a proper context in other places |
| 16:41.16 | [TK]D-Fender | default* |
| 16:41.17 | florz | yeah, right, and that's asterisk's fault, really |
| 16:41.35 | elliot98 | what is the new bridging feature in 1.6? |
| 16:42.13 | florz | "we don't know what to do, the config seems to be incomplete/buggy, so let's do the least expected thing possible" |
| 16:42.58 | [TK]D-Fender | florz: No.. the magic context "default" will save me! |
| 16:44.23 | florz | erm, yeah, obviously =:-) |
| 16:44.55 | [TK]D-Fender | florz: "I just want it to work!". The word just is often followed by the word "desserts" when a lazy twit who hasn't worked with *, payed attention, and read the docs does as little as humanly possible to lay out their system |
| 16:45.28 | [TK]D-Fender | "I'll learn the rest later".... |
| 16:50.43 | *** join/#asterisk Ad-Hoc (n=nimbus@62.1.164.133.dsl.dyn.forthnet.gr) |
| 16:52.42 | *** join/#asterisk jasonwert (n=jason@97-83-97-13.dhcp.trcy.mi.charter.com) |
| 16:54.28 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 16:54.49 | manxpower | If you don't know enough to secure your PBX then you should not be managing a PBX |
| 16:55.31 | florz | well, by that measure most people in this channel probably shouldn't be managing a PBX ... |
| 16:55.39 | manxpower | florz: Exactly. |
| 16:55.54 | florz | no, more than that ;-) |
| 16:56.13 | manxpower | The only thing that is going to make most people secure their PBX is when their PBX gets "hacked". |
| 16:56.37 | manxpower | I put "hacked" in quotes, because I don't think the term is really correct when applied to something that's not secure. |
| 16:57.38 | manxpower | voice "open relays" are being exploited more and more every day. |
| 16:59.34 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:00.16 | manxpower | At my last job about once a week we got a panicked call from someone with a "hacked" pbx. |
| 17:00.23 | manxpower | Asterisk PBX |
| 17:04.08 | manxpower | It's not THAT hard to secure you PBX. Assuming SIP and/or Zap: context=INVALID in sip.conf [general], use decent passwords for your SIP accounts. Make sure your IVRs don't allow outside calling, make sure your T/t/W/w/whatever don't let calls from the outside to do DTMF transfers, don't allow outside dialing from voicemail. [TK]D-Fender anything else you can think of? |
| 17:05.28 | [TK]D-Fender | manxpower: And make sure your usernames are alpha numeric. nd run fail2ban. And don't run GUI's on publicly open ports. And don't leave stupid PWs for it. |
| 17:06.03 | elliot98 | if you use ssh port to log in |
| 17:06.06 | elliot98 | use an alternate port |
| 17:06.31 | elliot98 | monitor the logs for bogus log ins |
| 17:06.39 | elliot98 | and block those ip addresses |
| 17:06.42 | manxpower | elliot98: the thing is it does not appear that most Asterisk "hacks" happen via SSH. |
| 17:07.29 | elliot98 | true...but if a system was hacked, it makes we wonder |
| 17:07.33 | p3nguin | If I had to remember every port that some dipshit moved sshd to, I'd be in trouble. |
| 17:07.56 | manxpower | elliot98: I mean "hacked" as in being able to route SIP calls for random people, not the actual OS hacked. |
| 17:09.48 | jaytee | none of my asterisk server ports are public and SIP is blocked at our firewall. we use PRI's to get to the rest of the world via PSTN. my default context is set to something other than default and that only allows guest calls from my Exchange server by 4 digit extension. no DISA in my dialplan |
| 17:11.13 | manxpower | [TK]D-Fender: but if you use alphanumeric usernames you dialplan can't be just one line long! Dial(SIP/${EXTEN})! |
| 17:11.19 | florz | Well, I remember when nobody in here was able to tell me how to limit calls to specific PSTN number ranges, given variable number length ... that's what I mean by "most". |
| 17:11.22 | *** join/#asterisk datacompboy (n=datacomp@l64-93-216.static.cn.ru) |
| 17:11.33 | manxpower | Classic n00b mistake is thinking you can have a simple dialplan on a PBX. 8-| |
| 17:12.30 | [TK]D-Fender | manxpower: Sure it can... you mean not everyone is dialing the entire alphabet on soft-phone? |
| 17:12.51 | [TK]D-Fender | manxpower: How do I plug a PS/2 keyboard into my GrandStream phone again? |
| 17:14.13 | jaytee | all my sip peers are in "class of service" contexts that give or remove call priviledges based on including or excluding other "calling" contexts. i.e. local-only, local-toll, long-distance, international, special, manager. |
| 17:14.25 | carrar | [TK]D-Fender, you have to use a soundcard |
| 17:14.41 | florz | [TK]D-Fender: on snoms you can enter pretty much any ASCII character, in particular ampersands |
| 17:15.34 | [TK]D-Fender | ~savemoney |
| 17:15.34 | infobot | <Gremlin> It's okay, but I was hoping to save some money with a soldering iron and a bunch of sound cards. |
| 17:15.37 | [TK]D-Fender | ahhhhhhhhhhhh |
| 17:16.02 | datacompboy | Hi everyone:) Anybody have good knowledge in DTMF troubles? It seems that i get double-detected tones, as in inband recognision, as in rfc. :( caller party is GSM phone, sent to mine server over SIP. To read DTMF I'm combine answers of StreamFile() + GetData() on silence of needed length. And seems that sometimes DTMF get doubled, sometimes works OK. In call record (inband mode) i hear that user tapped only 3 digits, but i get read "111" i |
| 17:17.24 | p3nguin | 111 is three digits. |
| 17:18.09 | manxpower | datacompboy: what codec? |
| 17:18.11 | carrar | depend how you look at it |
| 17:18.21 | datacompboy | manxpower: 711u |
| 17:18.48 | manxpower | datacompboy: do you have something like relaxdtmf= set in sip.conf? |
| 17:19.49 | manxpower | datacompboy: chances are you'll have to enable debug (sip debug and dtmf debug) to find out where the problem is. Many carriers have problems working with Asterisk with regards to DTMF. What version of Asterisk are you using? |
| 17:19.53 | datacompboy | manxpower: relaxdtmf=no in peer configuration |
| 17:20.19 | p3nguin | carrar: 1 three times is three digits. |
| 17:20.35 | datacompboy | manxpower: 1.4.26-1 (debian package) |
| 17:20.39 | carrar | BINARY? |
| 17:20.56 | coppice | manxpower: haven't they sorted out their DTMF detector after all these years? |
| 17:21.12 | manxpower | coppice: 1.4 seems to do much better. |
| 17:21.55 | manxpower | Most of the issues seemed to be with RFC2833 DTMF interop with carriers. |
| 17:22.02 | coppice | manxpower: the original DTMF detector can just be left in relaxed mode all the time |
| 17:22.30 | manxpower | coppice: not on any system I worked with. relaxdtmf always caused doubled digits |
| 17:22.53 | coppice | the one in asterisk does, but my original does not |
| 17:23.07 | manxpower | coppice: Most of your stuff works well. 8-| |
| 17:23.19 | datacompboy | manxpower: btw, opposite party is Nortel - CS2K ISN09U |
| 17:25.03 | datacompboy | manxpower: so, what better -- relaxdtmf=yes or =no ? in global or in peer config? |
| 17:26.04 | manxpower | datacompboy: in my experience removing the option or setting it to know is best |
| 17:26.38 | manxpower | But I tend to deal with PRI and Polycom phones. I don't usually deal with random SIP "carriers" |
| 17:26.45 | *** join/#asterisk pirulo (n=andres@12.236.109.2) |
| 17:27.39 | datacompboy | manxpower: it not random, it one of russia's huges telecom... well, wil enable debug and try to monitor problem further. thanks for help :) |
| 17:27.50 | [TK]D-Fender | manxpower: I ran into a SIP carrier once... boy am I glad I'm immunized... |
| 17:28.29 | manxpower | datacompboy: The largest DID carrier in the USA has DTMF issues with Asterisk (Level3) |
| 17:29.05 | datacompboy | manxpower: heh. in RFC mode? btw -- will "rfc2833compensate" helps there? |
| 17:30.42 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 17:30.54 | datacompboy | and does relaxdtmf have any corellation with dtmfmode = rfc2833 ? |
| 17:31.43 | manxpower | relaxdtmf only applies to inband DTMF |
| 17:32.02 | *** join/#asterisk Malkor (n=marco@hlle-d9ba03e9.pool.mediaWays.net) |
| 17:32.44 | coppice | manxpower: what are you referring to with level 3? TDM or SIP and them detecting or * detecting? |
| 17:33.51 | datacompboy | manxpower: "dtmf debug" -- what cmd to enable it? |
| 17:34.47 | manxpower | coppice: most of the reports I see is someone calling a Level3 DID that gets send to Asterisk. Caller can't navigate IVRs. |
| 17:35.40 | lesouvage | Perhaps a little bit of topic but does any of you succeed in running SipDroid softphone with proper nat traversal. Sipdroid - asterisk - other local extension is working fine. Calling out over wifi or g3 doens't give any sound. SIgnaling is working fine. |
| 17:35.44 | manxpower | Asterisk -> PSTN TDM DTMF problems are usually pretty easy to fix with toneduration= settings |
| 17:35.45 | coppice | manxpower: several things could cause that, like the octasic echo cancellers |
| 17:36.51 | datacompboy | manxpower: agh, added "dtmf" to console in logger, and logger reload -- is that "dtmf debug" ? |
| 17:37.18 | manxpower | datacompboy: it's different for different versions of Asterisk. "help" in the CLI should be informative |
| 17:40.46 | datacompboy | manxpower: http://pastebin.ca/1681107. |
| 17:41.00 | datacompboy | manxpower: why there "duration" if there rfc?! |
| 17:42.35 | datacompboy | manxpower: sip show peer shows "DTMFmode : rfc2833" |
| 17:44.17 | manxpower | datacompboy: "duration" should be part of whatever RFC talks about variable length DTMF. |
| 17:45.27 | manxpower | Asterisk was the first version to support VL DTMF |
| 17:45.38 | manxpower | ..er...Asterisk 1.4 was the first... |
| 17:45.43 | manxpower | Maybe I need more coffee. |
| 17:46.46 | datacompboy | manxpower: ok, many-many thanks! will leave sip/dtmf/agi debugs enabled and rfc mode, and monitor result. hope, after get more info will be easier found problem corner. |
| 17:51.46 | *** join/#asterisk errotan (n=errotan@81.0.115.3) |
| 17:51.46 | datacompboy | manxpower: btw! about simple dialplan -- you wrong, i have very easy dialplan: [pstn-in] _X. => 1,AGI(); [dialout] _X. => 1,Dial($EXTEN@provider) == and it really works! |
| 17:55.05 | [TK]D-Fender | datacompboy: And that shoddy recreation is broken and you've simply attempted to shift the wrok to a full-on programming language. |
| 17:56.48 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 17:58.07 | datacompboy | [TK]D-Fender: but dialplan is easy :) make things, currently done in external server, via extensions.conf its advanced aerobatics... |
| 18:06.13 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 18:09.25 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:09.26 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 18:11.43 | jaytee | waves |
| 18:14.39 | leifmadsen | waves back |
| 18:19.46 | *** join/#asterisk dandate2 (n=mangy@121.1.37.147) |
| 18:20.08 | dandate2 | anyone know why iax clients timeout but can connect with SIP regardless of router |
| 18:24.08 | *** join/#asterisk verywiseman (n=verywise@unaffiliated/verywiseman) |
| 18:24.45 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 18:25.37 | *** join/#asterisk verywiseman (n=verywise@unaffiliated/verywiseman) |
| 18:26.26 | *** join/#asterisk |Cybex| (n=John@atwork-20.r-212.178.82.atwork.nl) |
| 18:28.04 | jblack | looks at katty and tries to decide something |
| 18:28.31 | Katty | go with option b. |
| 18:28.41 | jblack | Plan B? |
| 18:28.44 | Katty | sure. |
| 18:29.03 | jblack | I know. Thanks =) |
| 18:29.57 | dandate2 | anyone know why iax clients timeout but can connect with SIP regardless of router |
| 18:30.38 | jblack | dandate2: iax2 uses different ports than sip. Perhaps a firewall.. |
| 18:30.43 | drmessano | lol |
| 18:31.45 | Katty | jblack: what did i just help you decide on |
| 18:32.19 | jblack | What I wanted to know about you. :P |
| 18:32.30 | Katty | i don't get it :< |
| 18:32.34 | jblack | I'm playing 21 questions on fb. |
| 18:33.44 | jblack | it asked me what I wish I knew about you. It then next asked whether or not I'd sleep with Oscar for a million bucks. |
| 18:34.26 | manxpower | Framebuffers can talk? |
| 18:34.41 | jblack | Wrong fb. ;P |
| 18:36.59 | jblack | rofl. Now it wants to know where I would kiss drmessano, if I had to |
| 18:37.22 | jblack | That settles it. Facebook thinks I'm gay |
| 18:38.40 | drmessano | Maybe facebook was trying to get me a date |
| 18:38.47 | drmessano | I cant fault it for trying |
| 18:38.59 | jblack | Are you that hard up, that it's pairing you up with 37 year old fat guys? |
| 18:39.45 | dandate2 | theres no firewall though, unless its in my pbx |
| 18:39.52 | jblack | Fuck me. Now it want's my ex-wife's best features |
| 18:40.27 | drmessano | jblack: Her ability to walk through the door before it slammed on her sorry ass? |
| 18:40.43 | [TK]D-Fender | "leaves". worst feature "doesn't stay gone" |
| 18:40.52 | drmessano | lol |
| 18:41.01 | jblack | I gotta be nice. I need her help from time to time |
| 18:41.37 | dandate2 | in the pbx do i need to go "setup" and add 4569 under "other ports" ? |
| 18:41.44 | drmessano | jblack: You should get married a few more times.. Then you'll have a pool of ex-bitches to work with |
| 18:42.08 | drmessano | Setup? |
| 18:42.14 | drmessano | Setup.exe ? |
| 18:42.14 | jblack | Yeah, it sounds like a set up. :) |
| 18:42.34 | [TK]D-Fender | gets ready to knock'em down |
| 18:42.57 | drmessano | Sounds like a 7-12 split to me |
| 18:43.04 | dandate2 | no no in the command line you type setup |
| 18:43.07 | dandate2 | and get fierwall configuration |
| 18:43.18 | [TK]D-Fender | hey its not every bowler who can get a strike in a neighbouring lane! |
| 18:43.29 | drmessano | Its important to have ports open, dandate2 |
| 18:43.31 | jaytee | lol |
| 18:43.45 | dandate2 | lol i know |
| 18:43.51 | dandate2 | just that my machine isnt behind a router |
| 18:44.00 | drmessano | dandate2: So? |
| 18:44.01 | dandate2 | must be the pbx internal firewall messing with it right? |
| 18:44.21 | drmessano | Let do the math.. shall we? |
| 18:44.29 | drmessano | PBX has a firewall = important info |
| 18:44.43 | drmessano | PBX isn't behind an external firewall = Important info |
| 18:45.06 | drmessano | PBX has only the ports open I have needed until now, which doesn't include IAX2 = important info |
| 18:45.14 | dandate2 | ok |
| 18:45.17 | dandate2 | i understand |
| 18:45.21 | drmessano | IAX2 clients timeout on connect to the PBX =Important info |
| 18:45.24 | dandate2 | is the best way to open that in the setup ? |
| 18:45.31 | drmessano | Im gonna go out on a limb here |
| 18:45.43 | drmessano | I think your firewall is blocking the ports you havent opened |
| 18:45.43 | jaytee | hope it's a strong oak |
| 18:45.50 | dandate2 | yeah i think so too |
| 18:45.55 | drmessano | jaytee: Fat joke, eh? |
| 18:46.07 | drmessano | :( |
| 18:46.11 | jaytee | no :-) |
| 18:46.17 | dandate2 | so in the commandline i just go to setup then firewall, then under other ports put 4569 and should be good no? |
| 18:46.23 | jaytee | I'm just risk averse |
| 18:46.34 | drmessano | dandate2: I believe the command is /j ##linux |
| 18:47.38 | dandate2 | cant find any info on that command heh |
| 18:48.03 | drmessano | jblack: Is also happy to help you.. Please send him your paypal info via PM and arrange a remote session |
| 18:48.23 | [TK]D-Fender | drmessano: sounds kinky |
| 18:49.37 | drmessano | lol |
| 18:49.58 | jblack | [TK]D-Fender: Money _is_ tight |
| 18:52.12 | dandate2 | damnit |
| 18:52.18 | jaytee | http://www.youtube.com/watch?v=NYWUXDTuLYk |
| 18:52.28 | dandate2 | i opened 4569 in the "setup" option in commandline and it still times out |
| 18:52.35 | dandate2 | wtf! |
| 18:54.21 | drmessano | Is IAX2 even running? |
| 18:55.10 | drmessano | asterisk -r then iax2 show peers or something |
| 18:56.34 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 18:56.40 | jblack | dandate2: pastebin "lsof -i -n | grep asterisk" please. |
| 18:57.19 | Godfather_ | can obtain the caller id number if it comes via spa3102? |
| 18:57.19 | jblack | I haven't heard simply red in forever |
| 18:57.59 | [TK]D-Fender | jblack: they were nearly respectable till they started ripping off Hall & Oates |
| 18:58.19 | jblack | jaytee: Know another great redhead? http://www.youtube.com/watch?v=Yu_moia-oVI |
| 18:58.21 | [TK]D-Fender | I COULD GO FOR THAT! |
| 18:58.40 | dandate2 | http://pastebin.ca/1681205 |
| 18:59.37 | Godfather_ | http://pastebin.com/m39824788 |
| 18:59.43 | jblack | uhh. |
| 19:00.08 | *** join/#asterisk hobodave (n=hobodave@adsl-99-92-176-167.dsl.chcgil.sbcglobal.net) |
| 19:00.11 | hobodave | hey guys |
| 19:00.18 | jblack | why is your webserver running as user asterisk? |
| 19:00.25 | drmessano | Freepbx |
| 19:00.27 | dandate2 | that is the default for piaf |
| 19:00.41 | hobodave | any ideas why installing the asterisk16 RPM on CentOS doesn't put anything in /etc/asterisk/ ? |
| 19:00.45 | hobodave | not a single conf file |
| 19:00.47 | jblack | There's your problem. You're running freepbx in the wrong irc channel. :) |
| 19:01.03 | dandate2 | haha |
| 19:01.05 | hobodave | I followed these instructions: http://www.asterisk.org/downloads/yum |
| 19:01.12 | hobodave | and I did this yesterday, and it worked |
| 19:01.13 | jblack | Well, we know he hasn't done something silly like unload the iax module, or change what port iax is on |
| 19:01.15 | hobodave | but today, it doesn't :( |
| 19:01.32 | hobodave | nevermind |
| 19:01.35 | hobodave | asterisk16-config |
| 19:01.36 | hobodave | fail |
| 19:01.41 | dandate2 | yeah but this is really killing me, how do i add 4569 to that list? |
| 19:01.51 | drmessano | netcat? |
| 19:01.53 | jblack | dandate2: The one you pasted? |
| 19:01.57 | dandate2 | yeah |
| 19:02.21 | jblack | dandate2: grep iax /etc/services |
| 19:02.34 | jblack | # |
| 19:02.35 | jblack | asterisk 3904 asterisk 10u IPv4 9979 UDP *:iax |
| 19:02.43 | jblack | iax port _is_ 4569 |
| 19:03.51 | dandate2 | viax 4569/tcp # Inter-Asterisk eXchange |
| 19:03.51 | dandate2 | iax 4569/udp # Inter-Asterisk eXchange |
| 19:04.04 | dandate2 | so its already setup properly??? |
| 19:04.14 | drmessano | jblack: the 3 things I like about Katty: Her love of tech, her love of cooking, the fact she hasn't married me |
| 19:04.33 | jblack | That's three things I love about her too. |
| 19:04.45 | jblack | her love of tech, her love of cooking, the fact she hasn't married you. |
| 19:04.49 | drmessano | dandate2: Asterisk is listening on the IAX2 port, doesnt mean SHIT otherwise |
| 19:04.52 | drmessano | lol |
| 19:04.53 | jaytee | and she loves fuzzy critters |
| 19:05.00 | jblack | as lunch! |
| 19:05.12 | drmessano | I hear she cooks a mean bald eagle |
| 19:05.20 | dandate2 | damn what do i do |
| 19:05.30 | drmessano | dandate2: Hire a consultant |
| 19:05.42 | jblack | dandate2: what's the problem again? |
| 19:05.52 | drmessano | dandate2: Maybe one came inside the box with that lenovo |
| 19:05.55 | dandate2 | iax clients time out connecting to server but work with SIP |
| 19:05.59 | jblack | Other than the fact you're not asking #asterisk for help with somethign that's not asterisk ? |
| 19:06.46 | jblack | Look over your firewall, and make sure udp 4569 is open. |
| 19:06.49 | drmessano | rm -Rf.. oh forget it.. Already did that this week |
| 19:07.08 | dandate2 | in commandline i typed setup and went to firewall and added 4569 under "other ports" |
| 19:07.21 | dandate2 | and when i returned to it it put iax:tcp under other ports |
| 19:07.26 | dandate2 | should i change that tcp to udp? |
| 19:08.31 | *** join/#asterisk voipmonk (n=voipmonk@67.204.57.187) |
| 19:08.36 | jblack | When I said "make sure udp 4569 is open", did you think i meant "tcp port 80"? |
| 19:09.07 | dandate2 | well, i turned off security settings and it still didnt work heh |
| 19:11.23 | jblack | decides to go kill zombies |
| 19:11.42 | jblack | I'm going to pretend that they're all named dandate2 |
| 19:13.40 | *** join/#asterisk riddlebox (n=user@173-113-126-222.pools.spcsdns.net) |
| 19:15.38 | dandate2 | does asterisk need to be restarted?? |
| 19:18.00 | tzafrir | dandate2, is the server behind NAT? |
| 19:18.20 | hobodave | wtf, I hate DSL |
| 19:20.00 | dandate2 | no its set to NAT=no |
| 19:20.27 | dandate2 | its got its own connection in a datacenter, no router |
| 19:20.37 | tzafrir | good |
| 19:20.51 | dandate2 | but cant get iax clients working for the life of me |
| 19:21.00 | tzafrir | now, is asterisk listening on UDP port 4569? netstat -lnup | grep 4569 |
| 19:21.56 | *** join/#asterisk hobodave_ (n=hobodave@adsl-99-92-176-167.dsl.chcgil.sbcglobal.net) |
| 19:22.57 | *** join/#asterisk Tim_Toady (n=moi@212.251.125.215.dsl.dyn.forthnet.gr) |
| 19:24.01 | dandate2 | root@pbx:~ $ netstat -lnup | grep 4569 |
| 19:24.02 | dandate2 | udp 0 0 0.0.0.0:4569 0.0.0.0:* 3904/asterisk |
| 19:24.51 | [TK]D-Fender | dandate2: trash your firewall |
| 19:25.04 | dandate2 | i already disabled it |
| 19:25.10 | dandate2 | still dont work! |
| 19:25.14 | [TK]D-Fender | dandShow me |
| 19:25.23 | dandate2 | sec |
| 19:28.15 | dandate2 | http://tinypic.com/view.php?pic=1z3vywj&s=6 |
| 19:32.38 | [TK]D-Fender | danPic means precisely jack shit to me. |
| 19:32.55 | [TK]D-Fender | dandate2: iptables --list |
| 19:33.19 | dandate2 | ok |
| 19:33.26 | drmessano | ROFL |
| 19:33.33 | drmessano | SELINUX is NOT your firewall |
| 19:33.40 | dandate2 | security settings i disabled |
| 19:33.44 | dandate2 | lemme check iptables |
| 19:33.53 | drmessano | ok |
| 19:33.54 | voipmonk | wakes up |
| 19:34.19 | dandate2 | my iptables are all blank |
| 19:35.02 | [TK]D-Fender | dandate2: . |
| 19:35.07 | [TK]D-Fender | dandate2: where is it.... |
| 19:35.12 | dandate2 | http://pastebin.ca/1681259 |
| 19:39.16 | [TK]D-Fender | dandate2: Better. Now it looks like you're running * in a VM |
| 19:39.26 | dandate2 | lol no |
| 19:39.53 | [TK]D-Fender | dandate2: Well I'm seeing VMWare and 2003 server in there... |
| 19:40.27 | [TK]D-Fender | dandate2: Also, who says I trust the firewall on your WORKSTATION? |
| 19:40.36 | dandate2 | no hthats putty |
| 19:40.55 | [TK]D-Fender | dandate2: I still se the other reference to VMWare... |
| 19:41.10 | dandate2 | running mirc in vmware basically |
| 19:41.33 | [TK]D-Fender | dandate2: And again I don't trust the PC you're running Zoiper on either |
| 19:44.41 | [TK]D-Fender | BBIAB |
| 19:47.52 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 19:48.08 | tzafrir | dandate2, next step: tcpdump -n 'udp port 4569' |
| 19:48.20 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 19:48.26 | tzafrir | try to connect, and see if any packets actually flow in |
| 19:52.18 | *** join/#asterisk moy (n=moy@CPE001cdfec4cee-CM00080dab8485.cpe.net.cable.rogers.com) |
| 19:59.11 | *** join/#asterisk hobodave (n=hobodave@adsl-99-92-176-167.dsl.chcgil.sbcglobal.net) |
| 20:05.51 | *** join/#asterisk doubletoker (n=doubleto@adsl-218-151-35.jax.bellsouth.net) |
| 20:06.35 | *** join/#asterisk sun28 (n=light@95.129.165.106) |
| 20:07.37 | *** join/#asterisk GameGamer43 (n=GameGame@CPE-65-27-76-78.new.res.rr.com) |
| 20:09.49 | doubletoker | is there a way with dialplan to load a script for the duration of a call |
| 20:10.04 | doubletoker | but continue with the dialplan |
| 20:10.46 | *** join/#asterisk dandate3 (n=mangy@112.202.217.16) |
| 20:11.06 | dandate3 | alright i fixed the iax, had to call the datacenter and found out im behind a router there |
| 20:11.40 | dandate3 | so i am renting from 2 datacenters, one has me behind a router and is using XO/level 3, and the other does not have me behind a router and just say they are on fiber optic (he.net) to be exact. which should i go with? |
| 20:12.58 | *** join/#asterisk TheDavidFactor-H (n=chatzill@nc-71-0-16-133.dhcp.embarqhsd.net) |
| 20:13.30 | jblack | doubletoker: Look at agi. |
| 20:13.54 | [TK]D-Fender | dandate3: Everything is behind a router. Just depends what its doing |
| 20:15.05 | *** join/#asterisk SirThomas (n=tomc@mail.kendeco.com) |
| 20:15.08 | dandate3 | really cuz the he.net told me no router in play |
| 20:15.47 | [TK]D-Fender | dandthat is a retarded generalization |
| 20:17.29 | Gugge | as soon as you have to send traffic between different subnets there has to be a router in play :) |
| 20:18.21 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
| 20:18.34 | doubletoker | ok how about this question then |
| 20:18.52 | dandate3 | ok |
| 20:19.45 | doubletoker | is there a way that say I got ext 30 with 40 priorities last one being hangup() |
| 20:19.53 | *** join/#asterisk jmacz (n=jmacz@190.25.115.67) |
| 20:20.05 | [TK]D-Fender | doubletoker: huh? |
| 20:20.44 | doubletoker | basicly I'm wonding, is there a way in dialplan to call an agi script for example when someone hangups |
| 20:21.05 | doubletoker | not me calling hangup() in the dialplan |
| 20:21.07 | voipmonk | sure you can |
| 20:21.17 | [TK]D-Fender | doubletoker: "h" Asterisk Standard Extension |
| 20:21.35 | doubletoker | oh ok |
| 20:22.31 | doubletoker | I feel stupid, and this might be a stupid question |
| 20:22.42 | doubletoker | but if I'm in ext h |
| 20:22.53 | doubletoker | can you only call deadagi? |
| 20:23.29 | [TK]D-Fender | doubletoker: The call is DEAD. When someone hangs up you executing more dialplan in your current exten anymore |
| 20:25.21 | doubletoker | basicly I want to do, db stuff when they call and at different times while moving through out the system, would like to clean up the db when they hangup |
| 20:25.55 | dandate3 | does each iax client need its own port or is that only if there are multiple clients behind the same router? |
| 20:27.28 | doubletoker | ok ty |
| 20:28.01 | drmessano | Didnt we go through this earlier? |
| 20:28.15 | drmessano | ONE port |
| 20:28.36 | drmessano | If the clients are behind a router, NAT takes care of everything.. you open NO PORTS |
| 20:30.52 | *** join/#asterisk AndyHarris2 (n=AndyHarr@dsl-217-155-202-52.zen.co.uk) |
| 20:31.20 | AndyHarris2 | Hi --- is there any way to absolutely clean out asterisk before a re-install ? |
| 20:31.31 | drmessano | Yep |
| 20:32.05 | *** join/#asterisk Malkor (n=marco@hlle-d9ba03e9.pool.mediaWays.net) |
| 20:32.16 | [TK]D-Fender | drmessano: .... you've already done that this week ;) |
| 20:32.33 | AndyHarris2 | I know --- and I'm still stuck ... |
| 20:32.38 | drmessano | rm -R.... oh right |
| 20:32.40 | AndyHarris2 | once I have a nearly working system, |
| 20:32.58 | AndyHarris2 | now not even calling another extension works ! |
| 20:33.20 | AndyHarris2 | I've deleted files from /etc/asterisk and rebuilt ,, |
| 20:33.28 | [TK]D-Fender | andAnd your description of your debugging attempts is GLORIOUS |
| 20:33.29 | AndyHarris2 | database deltree'd everything |
| 20:33.39 | drmessano | You given NO debug, no cli, nothing |
| 20:33.47 | AndyHarris2 | nothing if not dogged ! |
| 20:33.52 | drmessano | 10000 view of the problem |
| 20:33.59 | drmessano | 10000ft view of the problem |
| 20:34.11 | drmessano | "I clear out stuff, reinstall, and it breaks" |
| 20:34.14 | drmessano | Tells me nothing |
| 20:34.20 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 20:35.01 | AndyHarris2 | K -- I'm not clearing out everything, something else is holding some config data .. |
| 20:35.10 | AndyHarris2 | I can't figure out what ... |
| 20:35.16 | drmessano | Nor can we |
| 20:35.23 | *** join/#asterisk puzzled (n=patrick@535335AA.cable.casema.nl) |
| 20:35.25 | *** join/#asterisk ruben23 (n=RPL@122.55.48.243) |
| 20:35.35 | ruben23 | hi |
| 20:35.40 | drmessano | Do you want help or did you just come to tweet about it? |
| 20:36.53 | AndyHarris2 | Guidance ... there must be several places to clear ... |
| 20:37.59 | drmessano | How about pastebin a failed call.. if you cleared the configs, there's nothing else "holding config data" |
| 20:38.12 | [TK]D-Fender | AndyHarris2: You haven't shown us a single symptom with any debug to back it up |
| 20:38.36 | AndyHarris2 | Will do |
| 20:41.44 | ruben23 | anyone familier with this warning: --->http://pastebin.com/m5bd372ba |
| 20:42.42 | drmessano | Since you didnt put a space between the arrow and the h, I can't open it without copy/paste, so no |
| 20:45.00 | *** join/#asterisk Alagar (n=Administ@122.164.40.206) |
| 20:46.00 | ruben23 | http://pastebin.com/m5bd372ba |
| 20:46.02 | *** join/#asterisk farkus_ (i=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 20:48.42 | jblack | his error is WARNING[15309]: rtp.c:891 ast_rtcp_read: RTCP Read too short |
| 20:48.57 | jblack | That reads to me like the ports defined in /etc/rtp.conf aren't being forwarded by the firewall |
| 20:50.36 | toresbe | Hello, I'm struggling a bit with my SPA2102 |
| 20:50.45 | toresbe | It sends me this register message, with From: "Tore Sinding Bekkedal" <sip:21019883@sip.phonzo.com> |
| 20:50.55 | toresbe | But I get [Nov 21 21:47:43] NOTICE[5085]: chan_sip.c:21006 handle_request_register: Registration from '"Tore Sinding Bekkedal" <sip:21019883@sip.phonzo.com>' failed for '10.0.0.188' - No matching peer found |
| 20:51.06 | toresbe | I've been experimenting but haven't quite found a working syntax |
| 20:51.14 | jblack | Your sip.conf isn't right. |
| 20:51.42 | jblack | register=> username:password@server |
| 20:51.57 | jblack | some servers expect a /extension appended too |
| 20:52.55 | toresbe | Uhm.... I'm not trying to register to my outgoing VoIP, my VoIP box that I got from my VoIP company is trying to register with me (since I've spoofed the DNS for the SIP server :)) |
| 20:56.08 | doubletoker | is there a rxgain txgain for sip? |
| 20:56.47 | toresbe | Anyone? |
| 20:58.10 | chuckf | why are you spoofing dns? why not just register with the provider you got the spa from? |
| 20:58.59 | toresbe | chuckf: I do that; My * regs with the provider, and my SPA regs with my * |
| 21:03.33 | *** join/#asterisk tamiel (n=tamiel@ip-1.net-81-220-19.versailles.rev.numericable.fr) |
| 21:04.40 | chuckf | I dont' get why you're spoofing DNS |
| 21:06.14 | jblack | doubletoker: I don't know of one |
| 21:06.32 | manxpower | you do not have a [21019883] in your sip.conf |
| 21:06.54 | manxpower | doubletoker: much of the time audio isn't even going thru Asterisk |
| 21:07.20 | toresbe | chuckf: because my SPA2102 is provisioned |
| 21:07.25 | doubletoker | oh |
| 21:07.32 | doubletoker | alright thanks |
| 21:07.38 | toresbe | manxpower: I have a [tastafon] (â¦) user=21019883 |
| 21:07.51 | toresbe | isn't that good enough? |
| 21:07.51 | manxpower | toresbe: I can't help you with users.conf |
| 21:07.53 | manxpower | ~users.conf |
| 21:07.54 | infobot | [~users.conf] users.conf is a flaming pile of sh1t that takes the fine control of several perfectly usable * config files and reduces them to the lowest common denominator and makes your system behave like a "toaster grade" PBX system. |
| 21:08.26 | toresbe | manxpower: I'm using sip.conf |
| 21:08.43 | manxpower | toresbe: the stuff in [here] is your sip userid. |
| 21:09.03 | toresbe | manxpower: yes, but it isn't possible to override_ |
| 21:09.33 | manxpower | toresbe: change [tastafon] to [21019883] and your device should be able to register |
| 21:09.46 | manxpower | to change the OUTGOING username from asterisk use fromuser= |
| 21:09.50 | toresbe | Nope, it didn't... |
| 21:10.14 | toresbe | I want to change the _incoming_ user, not the _outgoing_ |
| 21:10.28 | manxpower | the incoming user is [inthere] |
| 21:10.44 | manxpower | take my advice or don't take my advice. I don't care. |
| 21:10.49 | toresbe | But can I not also specify that with user=⦠in the clause itself? |
| 21:11.25 | manxpower | obviously not or the device would be able to register to your Asterisk server and not generate that error message |
| 21:11.52 | toresbe | manxpower: It's doing the exact same thing when I am following your advice. |
| 21:12.08 | toresbe | oh, wait, now I'm getting somewhere. |
| 21:12.15 | manxpower | You get the exact same error message? |
| 21:12.39 | toresbe | I tried it again, since I think I may have left out a sip reload :-) |
| 21:20.19 | *** join/#asterisk geneticx (n=geneticx@adsl-10-104-182.mia.bellsouth.net) |
| 21:30.47 | *** join/#asterisk Zokeber (n=Zokeber@unaffiliated/zokeber) |
| 21:35.18 | *** join/#asterisk memph (n=memph@sd1438.sivit.org) |
| 21:37.14 | *** join/#asterisk MmixX (i=mixed@61.14.191.133) |
| 21:41.52 | *** join/#asterisk dkirker (n=dkirker@gateway0.openmobl.com) |
| 21:44.08 | Katty | stretches |
| 21:46.49 | Katty | morning |
| 21:48.02 | jblack | there ya are |
| 21:51.31 | Katty | yes. |
| 21:56.51 | *** join/#asterisk GGD (i=48c4f168@gateway/web/freenode/x-xigahvefkvpuwszp) |
| 21:59.49 | jblack | AHA! j_wb is available in twitter! |
| 22:00.00 | Katty | dragged off to lowes. |
| 22:01.55 | *** join/#asterisk ChannelZ (i=channelz@burner.com) |
| 22:12.03 | *** join/#asterisk uluatu (n=uluatu@187.58.238.14) |
| 22:15.21 | GGD | <PROTECTED> |
| 22:16.43 | lesouvage | I have sipdroid (a sip phone for adroid mobile phones) registered on my natted asterisk box. Over 3g and over wifi "sip show peers" still shows UNREACHABLE while I can make outbound phonecalls. Inbound calls aren't working. Is someting wrong in the sip entry? (see http://www.pastebin.be/22077) |
| 22:16.46 | manxpower | GGD: Compile from source. |
| 22:17.16 | GGD | oh? |
| 22:17.22 | manxpower | lesouvage: does the NAT IP or the public IP show in sip show peers. |
| 22:18.11 | manxpower | GGD: 1) if you install from a package you won't get much support from people here 2) Packagers always seem to screw something up. 3) packages are seldom up to date for software like Asterisk that has frequent releases. |
| 22:18.34 | manxpower | I highly recommend packages for most software, but not Asterisk (or clamav, for that matter) |
| 22:19.06 | GGD | i don't mean to sound like a noob here but how do i compile it from the source? |
| 22:19.18 | manxpower | lesouvage: also remember that some SIP phones (especially softphones) don't support the OPTIONS request that qualify=yes requires |
| 22:19.21 | moos3 | does anyone know a good SIP handset? |
| 22:19.28 | moos3 | cisco or some one else |
| 22:19.30 | lesouvage | manxpower: the nat ip being the inside of the the router |
| 22:20.04 | manxpower | GGD: There is a README file. But if you are not comfortable compiling from source you should reconsider using Asterisk. Asterisk requires significant skills in Asterisk, telecom, Linux, networking, NAT, SIP, and RTP. |
| 22:20.11 | manxpower | lesouvage: yes |
| 22:20.16 | manxpower | ~phones |
| 22:20.17 | infobot | While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. |
| 22:20.25 | manxpower | moos3: ~phones is for you |
| 22:20.36 | lesouvage | manxpower: that was actually an answer :-) |
| 22:20.51 | moos3 | yeah |
| 22:21.10 | manxpower | lesouvage: if your Asterisk server is set correctly then the public IP of the NAT'd SIP phone should show in sip show peers, not the private IP behind the remote NAT router. |
| 22:21.46 | manxpower | Make sure you don't have any NAT features enabled on your SIP phone (except for NAT Keep-Alive, that's usually OK) |
| 22:22.16 | *** join/#asterisk Buklov (n=buklov@213.138.71.254) |
| 22:22.34 | manxpower | lesouvage: I am assuming Asterisk is on a public IP and the SIP device is behind NAT. |
| 22:24.03 | moos3 | manxpower: so what is the answer |
| 22:24.32 | manxpower | moos3: My answer is Polycom Soundpoint IP 550 |
| 22:25.37 | manxpower | That is what I have on MY desk. |
| 22:25.48 | lesouvage | manxpower: I have the impression that sipdroid is using NAT features without asking. I couldn't get it working on a server with a public IP number. Asterisk is behind NAT and the softphone is using 3g. |
| 22:26.10 | lesouvage | With an iphone the 3g was working fine sing SIAX. |
| 22:26.26 | jblack | sipdroid is a disaster. |
| 22:26.51 | jblack | anything that always reports it's address as 127.0.0.1 ... |
| 22:27.10 | ChannelZ | call me! |
| 22:27.27 | lesouvage | jblack: Are you sure, I was thinking that I was messing up . |
| 22:28.25 | jblack | maybe you'll have more luck than I |
| 22:28.36 | lesouvage | jblack: Is there an other softphone available for android. It is hard to believe that for this open system there is no decent softphone available while there is one for the iPhone (SIAX) that is working easy. |
| 22:28.44 | jblack | that's the only one I know of |
| 22:29.23 | [TK]D-Fender | moos3: How many are you looking to buy, and for what kind of use? |
| 22:30.08 | [TK]D-Fender | lesouvage: iPhone is older, and you listed 1. 1 vs 0 is not a crushing lead. |
| 22:30.17 | lesouvage | jblack: You gave up? |
| 22:30.31 | toresbe | Oh, my. dtmf detection settings in sip.conf will _not change_ from a sip reload, but need a full reload |
| 22:30.34 | toresbe | that one kept me a while. |
| 22:30.37 | lesouvage | [TK]D-Fender: you just need one, if it is working and reliable. |
| 22:31.00 | lesouvage | [TK]D-Fender: and yes, I think I have to be a little bit more patience. |
| 22:31.24 | jblack | I did |
| 22:31.28 | manxpower | lesouvage: Did you set up Asterisk to be behind NAT? (localnet, externip, forward ports, etc)? |
| 22:31.42 | lesouvage | jblack: I join you :-( |
| 22:32.14 | lesouvage | manxpower: yes, the whole set of adjustments . |
| 22:32.34 | manxpower | localnet/externip tells Asterisk IGNORE what the remote client tells you about it's IP address and pull the information from the packet header. |
| 22:33.10 | [TK]D-Fender | manxpower: Umm.... no |
| 22:33.11 | manxpower | Also, the NAT features of every phone I've heard of screws up the way Asterisk handles NAT |
| 22:34.27 | jblack | I would make things work as long as it was on the local wireless network. |
| 22:34.36 | lesouvage | manxpower: That was what surprised about the SIAX softphone for the iPhone. It simply works, without any hassle. |
| 22:35.27 | [TK]D-Fender | localnet tells Asterisk when to report the externip in SIP communications to the device. it has nothing to do with trusting the IP the device claims its sending on. "nat=yes" alone for the peer does that |
| 22:35.29 | moos3 | [TK]D-Fender: like 3 at most |
| 22:35.45 | [TK]D-Fender | moos3: What kind of use? |
| 22:35.50 | moos3 | and I need to find sip provider that does out bound and inbound |
| 22:35.57 | moos3 | home use |
| 22:36.20 | manxpower | [TK]D-Fender: I believe you are correct. I had it backwards. |
| 22:36.51 | lesouvage | jblack: that is working for me too but I wat to make honecalls ovr my 3g connection, an extra internal phone is not what I'm looking for. |
| 22:36.55 | [TK]D-Fender | manxpower: caffeinate <- |
| 22:37.15 | [TK]D-Fender | moos3: Do you have network jacks where you would place these? |
| 22:37.34 | manxpower | [TK]D-Fender: perhaps I'm over caffeinated. |
| 22:37.48 | [TK]D-Fender | manxpower: UNPOSSIBLE |
| 22:38.04 | [TK]D-Fender | is NASA's first certified caffeine-based life-form |
| 22:38.37 | moos3 | yeah I have jacks where they need too |
| 22:38.37 | manxpower | [TK]D-Fender: You are the Promised One for the Church of Scientology! |
| 22:38.45 | lesouvage | Thanks for the info and the help. I will wait another couple of month and look into the sip softphone for Android issue again. |
| 22:38.52 | toresbe | I know several JPLers who abstain from the coffee |
| 22:39.19 | jblack | lesouvage: I had trouble with 3g. I made it work once. |
| 22:39.57 | jblack | actually, a few times. But when the connection was too weak, sipdroid would just drop packets, rather than log out, and that would cause asterisk to get "stuck"... |
| 22:40.02 | jblack | as I remember it.. it's been well over a month |
| 22:42.21 | AndyHarris2 | One extension just rang another .... at last ! |
| 22:43.33 | moos3 | [TK]D-Fender: I would like a phone that I can setup for my house and my office SIP server |
| 22:44.36 | lesouvage | jblack: 3g is working for outbound calls but it is registring and registring over and over again, draining the battery. Making phonecalls work well. So it is something. |
| 22:46.41 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 22:47.40 | lesouvage | In short time I will do my first research with OpenBTS (mini private GSM network) combined with Asterisk. |
| 22:48.09 | [TK]D-Fender | manxpower: I promise not to jump up & down on Oprah's couch professing my love to Dawsons Creek leftovers... |
| 22:53.36 | *** join/#asterisk joako (n=ston3d@opensuse/member/joak0) |
| 23:01.56 | jblack | So, now that oprah is quitting... that means there's going to be a whole shitload of pissed off women with nothing to do. |
| 23:02.11 | ChannelZ | hides his guns |
| 23:04.10 | Katty | frowns |
| 23:04.16 | Katty | the clerk at lowes asked us if we were building a bomb |
| 23:04.36 | ChannelZ | Did you say "Yes, but it's only to kill terrorists." |
| 23:04.42 | Katty | no |
| 23:05.20 | Katty | in other sad news, i can't make dinner cause i'm out of parmesan :< |
| 23:05.37 | Katty | and heavy whipping cream too. |
| 23:05.40 | Katty | tis a sad day indeed. |
| 23:06.12 | Katty | runs to schnucks |
| 23:08.35 | [TK]D-Fender | [18:04]<Katty>the clerk at lowes asked us if we were building a bomb <- how dare he insult your cooking! |
| 23:09.58 | *** join/#asterisk mariobalibrera (n=mario@c-98-234-89-16.hsd1.ca.comcast.net) |
| 23:12.10 | toresbe | Correct me if I'm wrong |
| 23:12.57 | toresbe | but the right way to have a "press 1 for â¦" etc. thing is to just run off the end of one extension and then have subsequent ones, right? |
| 23:13.42 | toresbe | http://pastebin.com/m77f402ea as in this one? |
| 23:15.30 | ChannelZ | well you'd just play your menu and then have extension 1, 2, 3, etc that does whatever is supposed to happen when the person presses 1, 2, 3 etc |
| 23:15.51 | toresbe | so the pastebin'd file is correct? |
| 23:16.19 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 23:16.35 | toresbe | I get a status like "" -- Auto fallthrough, channel 'SIP/1005-b761eb48' status is 'UNKNOWN' |
| 23:17.05 | ChannelZ | Well yeah if it's instructions are "Dial 1005 for so-and-so, or 0XXXXXXX blah blah, or 1XXX..." |
| 23:17.29 | toresbe | well, no, not yet, I'm just testing; it's meant for my use only |
| 23:17.49 | toresbe | when I get there I'd like to be able to dial 0 to make an external call, or 1 to call an internal phone |
| 23:19.48 | [TK]D-Fender | Torrieri: add autofallthrough=no under [general] |
| 23:21.30 | ChannelZ | or don't, it's not really an error per se |
| 23:21.46 | toresbe | You understand what I'm trying to accomplish, right? |
| 23:22.06 | ChannelZ | yes |
| 23:26.24 | toresbe | Do you know what I'm doing wrong? |
| 23:27.34 | mchou | [TK]D-Fender: hey you around? |
| 23:28.11 | ChannelZ | toresbe: no, because you haven't said what isn't working |
| 23:29.17 | toresbe | ChannelZ: immediately after the Background(), it drops the connection |
| 23:29.21 | [TK]D-Fender | toresbe: I have already handed you the answer. Now dit it |
| 23:29.24 | [TK]D-Fender | do* |
| 23:29.49 | toresbe | [TK]D-Fender: I have. |
| 23:30.09 | [TK]D-Fender | toresbe: Show me, along with your attempt |
| 23:31.37 | ChannelZ | you can also use WaitExten(10) or something to wait for them to dial something for 10 seconds before timing out |
| 23:38.42 | toresbe | [TK]D-Fender: I have - it worked :) |
| 23:38.43 | toresbe | Thanks. |
| 23:38.54 | toresbe | I'm just scratching my head now over some weirdness with my provider. |
| 23:39.35 | toresbe | Now, when I dial out, I get "This account number is not valid", which is odd |
| 23:42.33 | toresbe | I can register, but it won't let me ring out anymore. |
| 23:43.30 | ChannelZ | is this "phonzo"? |
| 23:44.28 | toresbe | yeah |
| 23:44.35 | toresbe | I'm consufed now. |
| 23:44.38 | *** join/#asterisk xpot (n=xpot@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 23:45.07 | ChannelZ | you have exten => 0XXXXXXXX,1,Dial(SIP/${EXTEN:8}@phonzo) which means if you dial 012345678 it's only dialing the last digit |
| 23:45.37 | ChannelZ | (so SIP/9@phonzo) |
| 23:46.11 | toresbe | I fixed that too, sorry. I'm getting a bit tired. :) |
| 23:46.20 | toresbe | To: <sip:91859508@80.232.37.178> |
| 23:46.26 | toresbe | I think that's the problem. :) |
| 23:48.32 | *** join/#asterisk Zokeber (n=Zokeber@unaffiliated/zokeber) |
| 23:53.44 | mchou | [TK]D-Fender: mind looking at pm? |
