| 00:00.00 | JT | i don't get that at all if my server is on the lan |
| 00:00.04 | perd | analog works fine |
| 00:00.04 | nDuff | fujin: if you transfer the file to your workstation and play it back locally, do you have the same issue? |
| 00:00.21 | fujin | no |
| 00:00.24 | fujin | it's fine |
| 00:00.42 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 00:01.14 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
| 00:01.15 | fujin | impossible to diagnose what's causing it |
| 00:01.15 | fujin | lol |
| 00:01.25 | fujin | and it only happens some times |
| 00:01.27 | fujin | like packet loss |
| 00:01.38 | perd | you're the only other person i know who has had this issue |
| 00:02.22 | *** join/#asterisk poin-dexter (n=jmjonese@cpe-024-167-187-217.triad.res.rr.com) |
| 00:03.54 | fujin | here's the nipper |
| 00:03.57 | fujin | what phones do you use? |
| 00:04.08 | fujin | I haven't got around to testing softp[hones yet |
| 00:04.18 | perd | i get it with x-lite (or any other sip client) and cisco phones (7960, 7902) |
| 00:04.32 | *** join/#asterisk denon (n=denon@208.122.43.201) |
| 00:04.32 | *** mode/#asterisk [+o denon] by ChanServ |
| 00:04.36 | perd | when i use the analog phones, it doesnt happen |
| 00:04.45 | perd | everything is crystal clear |
| 00:04.45 | fujin | heh |
| 00:04.54 | perd | i've tried it on multiple servers |
| 00:05.00 | fujin | it's a shame cause my system is flawless otherwise |
| 00:05.05 | perd | with directly connected phones, using switches, hubs, etc |
| 00:05.13 | JT | hubs, lol |
| 00:05.17 | perd | haha dude i tried everythin |
| 00:05.40 | JT | wet string? |
| 00:05.53 | perd | even with a completely fresh install, ulaw or slin audio files, gsm, whatever |
| 00:06.01 | perd | it still does that shit. |
| 00:06.24 | perd | no i didnt try the wet string, i did attempt a fishing line connection though |
| 00:06.48 | WilliamK | hey perd, couple ideas... where's your box located? offsite at another hosting facility, etc? |
| 00:07.00 | perd | williamk i tested it locally |
| 00:07.05 | fujin | my box is local |
| 00:07.07 | fujin | across a lan |
| 00:07.14 | perd | with new installs, the only thing set up on it is 1 sip client and a voicemail box |
| 00:07.18 | WilliamK | any routers involved? |
| 00:07.20 | perd | on its own network |
| 00:07.27 | perd | with or without routers, same damn thing! |
| 00:07.33 | fujin | the router is where the vlan is defined here, but that's it |
| 00:07.42 | perd | i've already gone through hours of troubleshooting though, i dont care at this point. i cant take it any more :) |
| 00:08.06 | WilliamK | perd, fine with me, only tossing out ideas |
| 00:08.09 | fujin | perd: do you occasionally get ridiculous call distortion? |
| 00:08.14 | fujin | clicks and pops etc |
| 00:08.32 | perd | not pops or clicks so much as a weird oscillation in the voice, almost makes the voicemail chick sound like a robot for a second |
| 00:08.40 | perd | then it's fine for a while, then it will happen again randomly |
| 00:08.49 | perd | i can record a wav file if you want |
| 00:09.04 | fujin | nah it's cool |
| 00:09.08 | fujin | sounds exactly like mine |
| 00:09.53 | perd | williamk yeah man, i appreciate the ideas :) but at the moment i cant handle troubleshooting that damn problem :P |
| 00:10.08 | JT | only within the voicemail subsystem, or any voice prompts? |
| 00:10.29 | perd | i've only really noticed it in voicemail |
| 00:10.46 | perd | let me try some other menus and see |
| 00:11.02 | perd | of course the only asterisk setup i have in house at the moment is asterisknow |
| 00:12.27 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-4c843a032f125bb6) |
| 00:14.10 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 00:26.16 | perd | yeah it appears to happen with any menu |
| 00:26.20 | perd | easiest to duplicate with vm though |
| 00:26.32 | perd | and it happens much less frequently with gsm files as opposed to ulaw |
| 00:29.44 | perd | and yeah now that i listen to the issue again, fujin, there is a lot of popping |
| 00:29.52 | fujin | ;\ |
| 00:29.59 | fujin | I dunno what causes it |
| 00:30.02 | fujin | what distro are you using? |
| 00:30.04 | perd | popping and oscillation and weird choppiness |
| 00:30.13 | perd | i have tried it with debian and centos |
| 00:30.18 | perd | and now asterisknow |
| 00:30.19 | JT | what's the network setup? |
| 00:30.21 | perd | whatever distro that's based on |
| 00:30.34 | perd | this network setup is server -> switch -> sip client |
| 00:30.45 | perd | i can set it up with a direct connection to the server too |
| 00:30.49 | JT | sip client being? |
| 00:30.50 | perd | let me make a crossover cable |
| 00:30.52 | perd | x-lite |
| 00:31.00 | perd | it also does it with cisco phones though |
| 00:31.00 | JT | try something that sucks less |
| 00:31.02 | JT | like a polycom |
| 00:31.08 | perd | heh |
| 00:31.23 | perd | i've tried with a bunch of clients, unfortunately the only hardphones i have are cisco 7960 and 7902 |
| 00:31.37 | perd | and yes, they do such hugely |
| 00:31.41 | perd | err, suck |
| 00:32.11 | JT | it's still a very strange problem, and maybe the crossover cable will help |
| 00:32.18 | JT | what ethernet adapter does the server have? |
| 00:32.35 | *** part/#asterisk poin-dexter (n=jmjonese@cpe-024-167-187-217.triad.res.rr.com) |
| 00:32.40 | perd | this server has an onboard intel 82801BA |
| 00:32.46 | perd | other servers have had netgears or 3coms |
| 00:32.50 | perd | and experienced the same issue |
| 00:33.18 | perd | this particular box im trying it on is running asterisknow, i've also duplicated the problem using 1.4 and 1.2 on centos and debian distros with a fresh install |
| 00:33.25 | JT | with the same swtich? |
| 00:33.32 | perd | the same switch and different switches |
| 00:33.45 | perd | directly connected, with its own switch and even a hub |
| 00:33.52 | perd | the problem haunts me still |
| 00:33.52 | fujin | it's impossible to work out what's causing it |
| 00:34.18 | perd | if i use gsm audio files though the problem happens rarely.. i just like the clear audio from ulaw |
| 00:34.28 | perd | i've also tried alaw and slin and have the same issues |
| 00:35.00 | JT | ok, really looking at root causes here, but is your power feed clean? |
| 00:35.11 | perd | that i cant say for sure |
| 00:35.17 | perd | i have it on an ups... but who knows |
| 00:35.17 | fujin | mine definitely is |
| 00:35.23 | perd | i dont have any power filter |
| 00:35.43 | JT | what type of ups? |
| 00:35.55 | perd | some crappy netlite 2000 |
| 00:36.04 | perd | it's not a nice apc or anything like that |
| 00:36.11 | JT | heh ok |
| 00:36.22 | perd | could the power really cause that much of an issue? |
| 00:36.25 | JT | i'm guessing it's not a double conversion online ups then |
| 00:36.25 | perd | that's crazy... |
| 00:36.30 | perd | no it definitely is not |
| 00:36.31 | JT | doubtful |
| 00:36.45 | perd | these are just little pos ups' |
| 00:36.47 | JT | but you never know, seeing you said you've tried totally different servers |
| 00:36.54 | perd | the office im in now is small, doesnt require much |
| 00:37.01 | JT | did you try different ethernet cables? |
| 00:37.14 | perd | yeah this is the fourth system i've had asterisk on, with different dists and whatnot, even in different areas of this building |
| 00:37.21 | perd | all get the same problem. yeah different ethernet cables also |
| 00:37.33 | perd | it's bizzare. |
| 00:37.35 | JT | maybe you have a variable time diallation field in your office |
| 00:37.38 | perd | haha |
| 00:37.46 | perd | that is the only thing that makes sense to me |
| 00:37.49 | fujin | lol |
| 00:37.52 | fujin | I've got one here then, too |
| 00:37.52 | perd | where is data when you need him |
| 00:38.04 | perd | Commander Data, that is. |
| 00:38.05 | fujin | JT: NOT having ztdummy wouldn't cause this, would it? |
| 00:38.14 | fujin | I haven't built ztdummy at all, doing a pure sip, no meet-me setup |
| 00:38.20 | fujin | perd: do you have ztdummy? |
| 00:38.31 | perd | i have in older servers |
| 00:38.36 | perd | the one i'm using now has zaptel devices in it |
| 00:38.37 | JT | hrm |
| 00:38.43 | fujin | ah |
| 00:38.43 | perd | a tdm2400 |
| 00:38.47 | fujin | can't be that then. |
| 00:39.10 | JT | zap timing needs show up in funny places |
| 00:39.16 | fujin | I thought it may have been a latency issue, so I set up expedited forwarding on all of my devices |
| 00:39.18 | JT | but doesn't sound like a zap timing issue |
| 00:39.31 | fujin | nah, if we were missing a timing source one would expect that somethign wouldn't work entirely |
| 00:39.36 | fujin | like, wouldn't be able to get any sip up at all |
| 00:39.51 | JT | new idea, run asterisk vm prompts from ramdisk |
| 00:40.18 | fujin | not sure if I have the spare ram for that |
| 00:40.21 | perd | ok |
| 00:40.24 | fujin | gimme a sec |
| 00:40.33 | JT | heh, nothing urgent |
| 00:40.47 | fujin | oh, it's only 6.5m |
| 00:40.48 | JT | but it seems like nothing has worked (or will work...) |
| 00:40.48 | fujin | heh. |
| 00:40.58 | perd | it's something i havent tried yet |
| 00:41.00 | perd | worth a shot |
| 00:41.29 | JT | it's possible asterisk is borked and you're the only people noticing |
| 00:42.42 | perd | nah it still does it |
| 00:42.44 | perd | :/ |
| 00:42.50 | fujin | I had noticed a bit of chop when my queue call delivery macro executes |
| 00:42.52 | fujin | cause it hammers the cpu |
| 00:42.54 | fujin | instant spike |
| 00:43.06 | fujin | I just tried it on a tmpfs and it didn't help :\ |
| 00:43.07 | perd | ./dev/ram0 16M 7.6M 8.0M 49% /var/lib/asterisk/sounds |
| 00:43.11 | perd | no go. |
| 00:43.25 | fujin | mount -t tmpfs tmpfs /var/lib/asterisk/sounds |
| 00:43.31 | fujin | tmpfs 506M 6.5M 500M 2% /var/lib/asterisk/sounds |
| 00:52.48 | JT | well |
| 00:52.57 | JT | if your cpu spikes, maybe that's it |
| 00:53.08 | perd | mine doesnt spike :( |
| 00:53.28 | perd | gonna try a crossover cable again though |
| 00:53.31 | perd | see what happens |
| 00:53.38 | perd | server to client, direct connect ooh yeah |
| 00:54.36 | fujin | that's not ideal |
| 00:54.50 | perd | well i dont have any switches around atm |
| 00:54.54 | perd | best i can do |
| 00:55.14 | fujin | i initially had all my staff reporting when ti happened |
| 00:55.19 | fujin | just came too hard to monitor |
| 00:55.33 | perd | if you swap to GSM it happens a lot less |
| 00:55.41 | perd | but you lose that nice crystal clear sexy voice |
| 00:57.22 | fujin | hardly ideal either - I should be able to have everything transcoded to it's natural format |
| 00:57.30 | fujin | hell, I did that to overcome transcoder overhead |
| 00:57.37 | fujin | all of my MOH is the same |
| 00:57.42 | perd | yeah |
| 01:08.19 | perd | yeah so crossover cable.. same deal |
| 01:08.23 | perd | like you expected anything else! |
| 01:11.51 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 01:16.22 | *** join/#asterisk CVirus (n=GoD@82.201.174.191) |
| 01:29.12 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 01:30.46 | *** join/#asterisk rpm (n=russell@75.153.43.151) |
| 01:34.01 | *** join/#asterisk axscode (n=axscode@203.213.217.123) |
| 01:37.24 | axscode | is Tiger Jet Network Inc the TDM400?> |
| 01:42.21 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 01:43.58 | *** join/#asterisk p-d (n=blah@mmds-216-19-46-146.sqpk.az.commspeed.net) |
| 01:46.00 | *** join/#asterisk obnauticus (n=obnautic@c-76-115-29-47.hsd1.wa.comcast.net) |
| 01:54.17 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 01:59.44 | *** join/#asterisk slakware (n=slak@201.53.76.85) |
| 02:00.26 | *** join/#asterisk Delvar (n=Delvar@77.240.56.18) |
| 02:00.33 | slakware | I am trying to use voicemail odbc storage. however i cant run make menuconfig (sshed in). is there any particular config setting that i can apply to set odbc_storage? |
| 02:00.56 | *** join/#asterisk saftsack (n=saftsack@pD9E05C35.dip.t-dialin.net) |
| 02:07.21 | *** join/#asterisk wunderkin (n=wunderki@ip72-223-86-126.ph.ph.cox.net) |
| 02:11.59 | *** join/#asterisk ez` (n=ezw@c142.169.166-68.clta.globetrotter.net) |
| 02:12.03 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 02:13.09 | Ryushin | I'm tired. Can someone help me understand this message that I'm getting when I have a call incoming from Teliax: Rejected connect attempt from 63.211.239.2, request '8177171820@default' does not exist |
| 02:13.22 | Ryushin | I set up my iax.conf file for it. |
| 02:13.33 | Ryushin | My context is from-pstn |
| 02:15.03 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 02:17.39 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 02:19.09 | hmmhesays | its looking for that in default |
| 02:19.51 | axscode | what happend if asterisk is installed then zaptel.. |
| 02:20.04 | axscode | or it must be.. zaptel then asterisk? |
| 02:20.32 | JT | if you want asterisk to use zaptel, zaptel must be installed first |
| 02:21.33 | *** join/#asterisk ironhead_webby (n=webby@CPE-121-217-112-178.nsw.bigpond.net.au) |
| 02:21.40 | axscode | what if.. asterisk, zaptel, asterisk ? is that fine? |
| 02:22.28 | outtolunc | what if, chicken, egg, chicken? <G> |
| 02:22.43 | axscode | outtolunc: hmmm irrelevant.. |
| 02:23.06 | outtolunc | he just told you zaptel MUST be done before asterisk IF you expect asterisk to use zaptel |
| 02:23.10 | axscode | JT: coz i already have my asterisk installed.. then i bought my TDM card... then i guess i have to install zaptel, then asterisk again. |
| 02:23.10 | Ryushin | hmmhesays: That's what I thought. I told it though that the context is from-pstn. It just doesn't seem to be picking it up. |
| 02:23.15 | flenders | axscode: asterisk > zaptel > asterisk will work |
| 02:23.21 | flenders | you're just reinstalling asterisk |
| 02:23.22 | axscode | flenders: thanks. :) |
| 02:23.25 | flenders | waste of time |
| 02:23.28 | flenders | that's all |
| 02:24.55 | JT | axscode: irrelevant? he was just having a chuckle at the question |
| 02:25.01 | axscode | flenders: do i have to make clean with asterisk? or can i just make install? |
| 02:25.13 | flenders | make clean is always good |
| 02:25.33 | axscode | ok thank you... :) |
| 02:25.36 | JT | if it's 1.4, you should do make menuconfig too i believe |
| 02:25.57 | axscode | so make clean; make menuconfig; and make; make install; |
| 02:26.00 | outtolunc | and if he had 1.2 before a new install of 1.4 he should do 'make distclean' <G> |
| 02:26.07 | axscode | ok got ya all. thanks.. im having 1.4 |
| 02:28.01 | hmmhesays | JT: only if you want to unselect somethings |
| 02:28.13 | axscode | ok.. which comes first? ./configure or make menuconfig ? |
| 02:28.17 | hmmhesays | configure |
| 02:28.25 | hmmhesays | and its make menuselect I believe |
| 02:28.41 | JT | hmmhesays: no, if you want to make absolutely sure than chan_zap is being compiled |
| 02:28.55 | hmmhesays | JT you can see it in the configure mang |
| 02:29.05 | JT | mang? |
| 02:29.16 | hmmhesays | haha sorry doing a little tony montana there |
| 02:29.49 | JT | heh, not sure who that is, but okay :) |
| 02:30.03 | axscode | oh.. so what comes first again? |
| 02:30.07 | *** part/#asterisk ez` (n=ezw@c142.169.166-68.clta.globetrotter.net) |
| 02:30.51 | flenders | tony montana... haha |
| 02:31.08 | flenders | that sounded very cuban to me |
| 02:31.13 | outtolunc | whos on second? |
| 02:31.19 | axscode | i have installed zaptel, all in menuconfig of zaptel is selected.. and there is no XXX on it.. so i guess its good. |
| 02:31.41 | axscode | then.. i made make clean and make menuconfig on asterisk.. the zap_ is XXX... |
| 02:31.47 | axscode | and it says.. it need zaptel |
| 02:32.08 | axscode | i mean the description says.. dependencies: zaptel |
| 02:32.27 | flenders | axscode: you done ./configure ; make ; make install on zaptel yet? |
| 02:32.41 | axscode | i did flenders... |
| 02:32.46 | axscode | but im doing it again to make sure.. |
| 02:33.37 | axscode | do i have to modprobe already before installing asterisk? |
| 02:34.16 | *** join/#asterisk mordaunt (n=mordaunt@unaffiliated/mordaunt) |
| 02:35.34 | flenders | no need to modprobe before installing asterisk |
| 02:36.14 | axscode | menuselect shows zapchan already. thanks flenders |
| 02:36.33 | flenders | I never do menuselect |
| 02:36.44 | flenders | I just ./configure ; make ; make install |
| 02:37.00 | flenders | I guess on yours make clean ; ./configure ; make ; make install |
| 02:38.03 | axscode | yups works well too i guess. i believe menuselect is for the ./configure for the Makefile for whatever you select in the menu |
| 02:38.25 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 02:38.32 | axscode | but all in all, great support.. thank you.. :) |
| 02:40.58 | dijungal | cya fellas |
| 02:40.59 | dijungal | next day |
| 02:41.02 | *** part/#asterisk dijungal (n=kdaniel@208.0.231.108) |
| 02:41.44 | *** join/#asterisk teknoprep (n=Chris@unaffiliated/teknoprep) |
| 02:41.46 | teknoprep | hey all |
| 03:04.11 | hmmhesays | hello |
| 03:05.54 | teknoprep | man am i having a problem |
| 03:06.18 | teknoprep | i have a sip trunk and it seems to be working but it keeps throwing me to the "this number is not in service" |
| 03:06.29 | teknoprep | too bad i use asterisk with freepbx tho |
| 03:06.46 | fujin | die in a fire |
| 03:06.49 | fujin | you don't have a sip trunk |
| 03:06.53 | fujin | you have a sip connection |
| 03:07.04 | fujin | well, really, you have the definition for a sip connection |
| 03:07.09 | fujin | being that it's stateful |
| 03:18.43 | *** join/#asterisk denon (n=denon@208.122.43.201) |
| 03:18.43 | *** mode/#asterisk [+o denon] by ChanServ |
| 03:19.30 | *** part/#asterisk drfreeze (n=Jim@www.freeze.org) |
| 03:19.51 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 03:24.26 | teknoprep | ty for you symantecs |
| 03:25.00 | hypa7ia | semantics |
| 03:25.01 | hypa7ia | :) |
| 03:29.27 | *** join/#asterisk Lucky7 (n=Adam@cpe-70-122-42-73.austin.res.rr.com) |
| 03:29.35 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 03:29.53 | Lucky7 | hm |
| 03:29.58 | Lucky7 | I'm trying to find some good choices for CDR |
| 03:30.06 | Lucky7 | I've installed MySQL CDR in my box |
| 03:30.16 | fujin | mysql cdr seems to be great |
| 03:30.21 | Lucky7 | Are there any pre-written PHP CDR tools that will hook to that? |
| 03:30.22 | fujin | after you turn on the uniqueid field |
| 03:30.27 | fujin | I haven't found any |
| 03:30.30 | fujin | :\ |
| 03:30.40 | Lucky7 | so i've gotta write my own? suck |
| 03:30.43 | fujin | should be relatively to write it, though |
| 03:30.54 | fujin | the date time fields are all unix epoch so you can strtime them |
| 03:31.00 | fujin | i believe |
| 03:31.00 | Lucky7 | Yea, i just dont wanna fux with gd to make them all pretty and shit for my managers |
| 03:31.05 | fujin | ah, yes |
| 03:31.12 | fujin | gdinating it would be suck |
| 03:31.18 | Lucky7 | maybe i can steal trixbox's |
| 03:31.50 | Lucky7 | for as much as i dont like trixbox, they did a pretty good job with CDR |
| 03:32.33 | fujin | I haven't seen it |
| 03:32.37 | fujin | they probably stole it from somewhere else |
| 03:32.40 | Lucky7 | lol |
| 03:32.44 | Lucky7 | most likely |
| 03:33.05 | fujin | I must take a look at that, possibly steal it myself |
| 03:33.14 | Lucky7 | Ha |
| 03:33.20 | Lucky7 | Found what they stole ^.^ |
| 03:33.23 | Lucky7 | http://areski.net/asterisk-stat-v2/about.php |
| 03:33.47 | Lucky7 | i guess its not technically stolen, but ... borrowed without siting. |
| 03:34.03 | fujin | well, there we go |
| 03:34.06 | fujin | easy enough |
| 03:34.49 | Lucky7 | its a pretty slick lil tool |
| 03:37.22 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:40.40 | *** join/#asterisk monkeyb (n=nihm@mail.phonecontrol.com.au) |
| 03:47.29 | fujin | Lucky7: heh, that's pretty odd, it needs mysql_pconnect |
| 03:48.49 | Mavvie | is there a way to get a return value from an AGI script and use that? |
| 03:49.50 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 03:50.03 | SplasPood | anyone played with the new polycom firmware? |
| 03:50.14 | SplasPood | seems to drop registration after... a bit.. |
| 03:52.41 | *** join/#asterisk bmg505 (n=leon@196.209.182.75) |
| 04:00.33 | *** join/#asterisk Lucky7 (n=Adam@rrcs-67-78-114-237.sw.biz.rr.com) |
| 04:10.23 | [TK]D-Fender | SplasPood, not that I've seen |
| 04:10.34 | *** join/#asterisk PepOSX (n=pepOSX@190.72.148.155) |
| 04:13.49 | rpm | anyone here a sip protocol guru? i have a single question to ask because i don't want to read the rfc again. |
| 04:14.02 | fujin | learn to ctrl+f |
| 04:16.22 | osiris | ill take a stab |
| 04:16.39 | osiris | im no guru, but i work for a provider |
| 04:16.41 | rpm | How is the Digest response calculated when being a 407 is being ACK'd? Is it urp = MD5(username:realm:password), response = MD5(urp, nonce) ? |
| 04:17.25 | osiris | from what i understand, it is standard digest. i dont know if that helps |
| 04:18.19 | SplasPood | [TK]D-Fender: hrm.. Happened to my personal 500, and an associate who tried it reports the same behavior |
| 04:18.25 | rpm | i'll just open up the rfc and take a look.. rtfm to me. |
| 04:18.41 | SplasPood | [TK]D-Fender: are you setting any type of nat keepalive? |
| 04:18.53 | [TK]D-Fender | SplasPood, the basics said the 300 & 500 are supposed to be dropped with 2.2.0 |
| 04:18.56 | SplasPood | [TK]D-Fender: I haven't bothered to debug it yet, just noticed it earlier |
| 04:18.58 | [TK]D-Fender | SplasPood, no NAT |
| 04:19.13 | SplasPood | Maybe that has something to do with it then |
| 04:19.26 | osiris | im trying to figure why/how i can get the inbound proxy to deliver the calls without having to force inbound proxy on incoming |
| 04:19.31 | SplasPood | Yea, I don't have any 300s or 500s to contend with |
| 04:19.57 | SplasPood | In this case it's happened to both a 501 and 601, both behind NAT |
| 04:20.11 | osiris | for some reason, this versioon of broadsoft wont let you force inbund proxy on a registering device |
| 04:20.23 | [TK]D-Fender | SplasPood, Ok, doesn't occur in my setups yet... |
| 04:20.43 | [TK]D-Fender | SplasPood, just threw me off because you said 500.... |
| 04:21.00 | osiris | anyone have trunks through speakeasy ? |
| 04:21.02 | rpm | osiris, r13? |
| 04:21.10 | osiris | yeah |
| 04:21.35 | osiris | 13 has been givin me fits with outbound/inbound proxies |
| 04:21.38 | SplasPood | [TK]D-Fender: ahh didn't even notice, misfire |
| 04:21.55 | rpm | i manage a distributed broadsoft r13 platform |
| 04:22.12 | osiris | rpm, then your my guy |
| 04:22.13 | fujin | I wish we had have went to broadsoft |
| 04:22.14 | fujin | it looks awesome |
| 04:22.19 | osiris | mind if i pm for sec ? |
| 04:22.22 | rpm | sure |
| 04:40.31 | *** join/#asterisk famicon (i=redz@c51447ddc.cable.wanadoo.nl) |
| 04:40.37 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 04:44.19 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.174.112) |
| 04:44.38 | *** join/#asterisk zeeesh (i=zeeesh@202.166.161.45) |
| 04:47.13 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:57.27 | *** part/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 05:14.21 | *** join/#asterisk sjobeck (n=sjobeck@208-151-246-203.dq1sn.easystreet.com) |
| 05:15.13 | *** join/#asterisk sjobeck (n=sjobeck@208-151-246-203.dq1sn.easystreet.com) |
| 05:17.00 | *** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 05:46.55 | *** join/#asterisk dlynes_laptop (n=dlynes@d154-20-9-152.bchsia.telus.net) |
| 05:51.25 | *** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr) |
| 05:52.41 | *** join/#asterisk epaulin (n=epaulin@219.136.238.127) |
| 06:05.19 | *** join/#asterisk booray (n=ray@150.118.ultimate-int.uia.net) |
| 06:06.55 | *** join/#asterisk Corydon76-home (i=six@pdpc/supporter/sustaining/Corydon76-home) |
| 06:06.55 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 06:11.46 | *** part/#asterisk p-d (n=blah@mmds-216-19-46-146.sqpk.az.commspeed.net) |
| 06:20.40 | *** join/#asterisk McDouglas (n=mcd@mmcomp.adsl.datanet.hu) |
| 06:25.14 | *** join/#asterisk af_ (n=getsmart@81-174-47-128.dynamic.ngi.it) |
| 06:36.37 | *** join/#asterisk McDouglas (n=mcd@mmcomp.adsl.datanet.hu) |
| 06:36.48 | *** part/#asterisk CaT[tm] (n=cat@nessie.weebeastie.net) |
| 06:43.06 | *** join/#asterisk Delvar (n=Delvar@77.240.56.18) |
| 06:55.11 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53) |
| 07:00.24 | *** join/#asterisk Aurs (n=Aurs@1ult2p8.ip.hipercom.no) |
| 07:02.50 | *** join/#asterisk booray (n=ray@150.118.ultimate-int.uia.net) |
| 07:03.48 | axscode | Changing signalling on channel 1 from Unused to FXS Kewlstart |
| 07:03.57 | axscode | meaning channel 1 is for PSTN or Phone? |
| 07:04.16 | Strom_C | FXS signaling is used on FXO ports |
| 07:04.28 | McDouglas | [Sep 18 08:58:30] WARNING[20500]: channel.c:2612 ast_indicate_data: Unable to handle indication 5 for 'mISDN/3-u39' |
| 07:04.34 | McDouglas | what does this message mean? |
| 07:04.47 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 07:04.52 | axscode | Strom_C: meaning..? my channel1 is for? |
| 07:05.00 | axscode | PSTN or PHONE? |
| 07:05.02 | Strom_C | ~fxofxs |
| 07:05.03 | jbot | fxofxs is probably An FXO port expects to receive dialtone and receive ring voltage, you plug a phone LINE into this. An FXS port expects to provide dialtone and provide ring voltage, you plug a PHONE into this. |
| 07:05.22 | axscode | so FXS signalling is for PHONE! |
| 07:05.31 | Strom_C | you didnt read what I said |
| 07:05.38 | Strom_C | the FXS signaling is for the FXO port |
| 07:05.48 | Strom_C | so the phone line plugs into the port |
| 07:06.31 | axscode | My CHANNEL1 = Phone Line (POTS/PSTN) ? |
| 07:06.50 | Strom_C | i dont know; what hardware do you have? |
| 07:07.04 | axscode | i have TDM400P |
| 07:07.23 | Strom_C | which modules do you have on the card? |
| 07:07.47 | axscode | two RED and two green |
| 07:07.57 | axscode | Channel 01: FXS Kewlstart (Default) (Slaves: 01) <-- this is what ztcfg said |
| 07:08.02 | Strom_C | the red ones are for your phone lines |
| 07:08.10 | Strom_C | the green ones are for your telephone sets |
| 07:08.23 | axscode | ok.. problem is.. i dont know where it is |
| 07:08.29 | axscode | its inside the CPU already |
| 07:08.42 | Strom_C | well, open it back up and take a look |
| 07:08.50 | grimsy | or lots of trial and error ;) |
| 07:09.16 | axscode | how about the ztcfg ? |
| 07:09.28 | booray | Quick question for any interested; If I'm getting "Crypto support not loaded!" when attempting to receive an IAX2 call and a subsequent failure to find a key (it's there, I promise), does this mean a recompile? Google isn't much help here. Asterisk 1.4.11 |
| 07:09.31 | Corydon76-dig | only problem is if you connect an FXS to a trunk, you could end up destroying your card |
| 07:09.59 | zeeesh | how to compile "asterisk-addons-1.4.2 " i did .configure what shud be the second and third step .. make and then make clean and then make install ? |
| 07:10.22 | Corydon76-dig | booray: load res_crypto.so |
| 07:10.23 | booray | Corydon76-vcch: only if someone calls you, right? :-P |
| 07:10.31 | axscode | its |RED RED GREEN GREEN |
| 07:11.08 | Corydon76-dig | zeeesh: you should not 'make clean' between 'make' and 'make install' |
| 07:11.20 | Corydon76-dig | otherwise you're just undoing your 'make' |
| 07:11.38 | axscode | ok.. its |RED RED GREEN GREEN |
| 07:11.54 | zeeesh | <Corydon76-dig>: then just ./configure and then make and then make install ? |
| 07:12.06 | Corydon76-dig | zeeesh: correct |
| 07:12.08 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 07:12.28 | Corydon76-dig | zeeesh: you could add a 'make menuselect' after ./configure, if you want to eliminate certain modules from the build |
| 07:12.59 | JT | axscode: the CPU is the processor, not the computer case |
| 07:13.05 | booray | Corydon76-vcch: okay.. I see res_crypto.c in my source tree, but no res_crypto.so anywhere on the system. I'll see if I can figure out why and then compile it individually. Or.. maybe a make menuselect and redo the whole thing? hmm |
| 07:13.34 | booray | aha, make menuselect. of course |
| 07:13.49 | *** join/#asterisk MaliutaWrk (n=nikolai@fw.hitwise.com) |
| 07:14.55 | zeeesh | <Corydon76-dig>: "The existing menuselect.makeopts file did not specify that 'app_addon_sql_mysql' should not be included. However, either some dependencies for this module were not found or a conflict exists."? |
| 07:15.29 | booray | I should not be working on this at this hour |
| 07:16.46 | snk00sj | what is the correct way to pause between 2 sound files when using Background(file1&file2) ? |
| 07:16.55 | booray | dude, I just called time, and it said "Good Morning"... |
| 07:16.58 | booray | damnit |
| 07:17.20 | Corydon76-dig | zeeesh: make menuselect |
| 07:18.19 | *** join/#asterisk Delvar (n=Delvar@77.240.56.18) |
| 07:18.28 | booray | Corydon76-vcch: thank you for the help |
| 07:18.43 | booray | i meant corydon76-dig, but bitchx thought differently |
| 07:18.56 | Corydon76-dig | same person, different location |
| 07:27.09 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 07:31.37 | zeeesh | getting error by installing asterisk-addons-1.4.2 pls help" as http://sial.org/pbot/27543"? |
| 07:41.43 | *** join/#asterisk gremzoid (n=dan@d58-111-174-8.rdl5.qld.optusnet.com.au) |
| 07:48.08 | zeeesh | <Corydon76-dig>: i did make menuselect .. and then press F8 after this do i need to "make install" or what next ? |
| 07:51.21 | *** join/#asterisk |NexT| (n=next@netcache1.altecom.net) |
| 07:57.11 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 08:00.14 | mvanbaak | heya |
| 08:00.28 | mvanbaak | the TE110P rev B from 2004, is that card any good |
| 08:02.38 | mvanbaak | we have echo issues and disconnected calls |
| 08:02.52 | *** join/#asterisk MikHell (n=michel@203.116.19.240) |
| 08:02.59 | MikHell | Hi |
| 08:03.07 | McDouglas | why is that if i call an extension and the called party picks it up, he has to wait for 1-2 sec before he can actually talk ? |
| 08:04.12 | MikHell | Anybody knows a VoIP provider with good quality and good rates that allows more than one simultaneous call out? |
| 08:05.58 | *** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu) |
| 08:08.51 | *** part/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-4c843a032f125bb6) |
| 08:10.49 | slima | wrong channel. |
| 08:13.38 | *** join/#asterisk dlynes_ (n=dlynes@d154-20-9-152.bchsia.telus.net) |
| 08:18.56 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:20.32 | penguinFunk | zeeesh, there is a small bug in two of the files |
| 08:20.53 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 08:21.49 | penguinFunk | zeeesh, The 'make install' |
| 08:21.49 | penguinFunk | is looking for .libs/libchan_h323.so.1.0.1, but the compile |
| 08:21.49 | penguinFunk | produced .libs/libchan_h323.1.0.1 |
| 08:21.49 | penguinFunk | You can copy the file manually and it will work fine: |
| 08:21.49 | penguinFunk | cp .libs/libchan_h323.1.0.1 /usr/lib/asterisk/modules/chan_ooh323.so |
| 08:25.08 | penguinFunk | http://bugs.digium.com/view.php?id=9643 |
| 08:29.52 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 08:30.01 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 08:35.21 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 08:35.34 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 08:36.11 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 08:36.57 | *** join/#asterisk nohup- (i=hmmmph@203.81.239.64) |
| 08:37.14 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 08:38.21 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 08:38.24 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 08:39.09 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 08:40.51 | DarKnesS_WolF | mmm how to defive the operator exntesion in context ? |
| 08:41.10 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 08:42.32 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-7af49de505ff13fc) |
| 08:46.13 | *** join/#asterisk darviria (n=darviria@194-105-181-29.ifb.co.uk) |
| 08:47.52 | kaldemar | DarKnesS_WolF: what is "the operator extension"? |
| 08:54.53 | *** join/#asterisk axai (n=axai@217.150.124.10) |
| 08:56.07 | axai | Hey, quick question for you guys. I need to forward my internal extention out the PRI |
| 08:56.31 | axai | aka 9500 -> 2509500 instead 2509572 (default) |
| 08:57.00 | *** join/#asterisk tuzhila (i=tuzhila@84.47.128.99) |
| 08:57.14 | tuzhila | hi all |
| 08:57.33 | tuzhila | what fxo gates is better for use? |
| 08:59.36 | kaldemar | axai: what is your question? |
| 08:59.51 | tuzhila | kaldemar: what fxo gates is better for use? |
| 09:00.07 | axai | Basically, when dialing from an internal extension i want my number to be sent "out" |
| 09:00.10 | tuzhila | for sip |
| 09:00.18 | axai | so people can dial it back |
| 09:00.19 | tuzhila | what fxo gates is better for use for sip? |
| 09:00.32 | *** join/#asterisk xnoudas (n=chatzill@ppp171-76.adsl.forthnet.gr) |
| 09:01.25 | kaldemar | axai: so you want to change the CID. use function CALLERID for that. |
| 09:01.46 | tuzhila | what fxo gates is better for use for sip? |
| 09:01.49 | axai | could you give me an example, i've tried it already |
| 09:01.52 | axai | tuzhila: try a linksys |
| 09:01.57 | *** join/#asterisk hermuli (n=Eladamri@xdsl-83-145-207-63.nebulazone.fi) |
| 09:02.03 | tuzhila | axai: linksys |
| 09:02.03 | tuzhila | & |
| 09:02.03 | Strom_C | tuzhila: your question makes little sense |
| 09:02.04 | tuzhila | ? |
| 09:02.22 | kaldemar | the whole person makes no sense. |
| 09:03.46 | axai | hehe |
| 09:03.57 | kaldemar | axai: exten => 9500,1,Set(${CALLERID(num)}=250${EXTEN}) |
| 09:04.19 | axai | Btw, this is coming out of the PRI line |
| 09:04.29 | axai | so, i want people on the PSTN to see the whole number "2509500" |
| 09:04.37 | axai | 250 = what my operater has given us |
| 09:04.41 | axai | 500 = |
| 09:04.50 | axai | and the 500 -572 is our range |
| 09:04.59 | kaldemar | axai: oops, disregard the ${} around the function. |
| 09:05.04 | axai | oh okay |
| 09:09.13 | axscode | Unable to create channel of type 'Zap' (cause 0 - Unknown) <-- any help with this please? |
| 09:09.32 | axai | kaldemar: im afraid ti doesnt work, i want to set the caller ID on a Zap channel |
| 09:11.42 | tzafrir | axscode, this is a very generic error message |
| 09:11.54 | kaldemar | axai: how does it not work? is it not setting the callerid number? is the number not showing right in the callee's phone? |
| 09:12.20 | axscode | tzafrir: i dont know how to set-up the TDM |
| 09:12.44 | tzafrir | What Dial command have you used? |
| 09:12.48 | axscode | Channel 01: FXS Kewlstart (Default) (Slaves: 01) |
| 09:12.52 | axscode | 4 channels configured. |
| 09:12.55 | DarKnesS_WolF | kaldemar: o extension |
| 09:12.56 | DarKnesS_WolF | i'll try |
| 09:13.08 | DarKnesS_WolF | tzafrir: wow dude long time no seen :-) how are u ? |
| 09:13.13 | axscode | i have 01-04... with RED RED GREEN GREEN in the TDM |
| 09:13.24 | axai | kaldemar: Sorry, let me explain my situation more. I'm in india. We have the 9500-9572 extension range in sip.conf. We have been given the "2509500-2509572" block by our telco. Whenever we make a call from an internal SIP phone to the outside line, the call always appears to be from 2509572. I want it too appear to be from the extension that was dialed |
| 09:13.44 | JT | tuzhila: what don't you understand about "your question does not make sense"? : |
| 09:13.48 | axscode | tzafrir: how will i know that the device is ready to use? |
| 09:13.48 | tzafrir | axscode, in your dialplan, what was the Dial() command that ended up with that error? |
| 09:13.58 | tzafrir | DarKnesS_WolF, hi :-) |
| 09:14.19 | nohup- | i finally managed to run my tdm400p 4 fxo card ! |
| 09:14.47 | axscode | <PROTECTED> |
| 09:14.49 | JT | axscode: Set(CALLERID(num)=<something>) |
| 09:14.55 | JT | axai: even |
| 09:14.56 | tzafrir | axscode, zap show channels |
| 09:15.13 | axscode | <PROTECTED> |
| 09:15.14 | axscode | <PROTECTED> |
| 09:15.15 | tzafrir | caller ID shouldn't matter |
| 09:15.19 | axscode | only two lines. |
| 09:15.23 | JT | axscode: that is NOT the dial command |
| 09:15.33 | JT | extensions.conf, the dial command in question |
| 09:15.34 | kaldemar | axai: ask your telco what the number should be like. for example here in finland they want the area code to be included in the id but without the leading zero in it. |
| 09:16.04 | axscode | JT: tzafrir is asking about zap show channels. |
| 09:16.13 | nohup- | tzafrir: try doin modeprobe wcfxo |
| 09:16.27 | nohup- | then do genzaptelconf |
| 09:16.43 | axai | i see >< |
| 09:16.52 | JT | <PROTECTED> |
| 09:16.52 | JT | <PROTECTED> |
| 09:17.00 | JT | axscode: pay more attention |
| 09:17.10 | kaldemar | axai: if i'd be in 09 area, having a block 123 200-300, i'd have to set my number to 9123200 for outgoing for extension 200, or the telco changes it. |
| 09:17.53 | axscode | i already pasted it with tzafrir.. 4 lines.. im sorry |
| 09:18.10 | axai | kaldemar: Okay, ill try that |
| 09:18.18 | axai | why does asterisk default to "572"? |
| 09:19.16 | axai | ah ha! |
| 09:19.25 | axai | kaldemar: it works! thankyou |
| 09:19.45 | axscode | maybe i have to edit may zapata.conf.. |
| 09:20.05 | axai | kaldemar: One last thing, how do i get the extension of the person making the call? |
| 09:21.34 | kaldemar | axai: it's not asterisk that defaults it, it's the telco. |
| 09:21.47 | axai | kaldemar: Oh i see.. :) |
| 09:22.14 | kaldemar | if the number is something they don't allow you to send, they'll just overwrite it. |
| 09:23.10 | kaldemar | axai: by the person, do you mean your local extensions? |
| 09:23.17 | axai | kaldemar: Yes :) |
| 09:23.48 | kaldemar | ${CALLERID(num)} will show you the callerid number of the calling extension. |
| 09:24.22 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 09:24.27 | axai | thanks |
| 09:24.33 | *** join/#asterisk davixx (n=davixx@85.69.178.138) |
| 09:25.30 | JT | axai: i already told you how to set outbound callerid |
| 09:26.04 | axai | JT: I didnt want to set it |
| 09:26.16 | axai | JT: I wanted to get it this time ;) |
| 09:26.23 | *** join/#asterisk qdk (n=qdk@213.237.44.34) |
| 09:26.29 | axai | JT and kaldemar: all works now, thankyou all very much! |
| 09:27.12 | JT | axai: so what did you do? |
| 09:28.27 | axai | exten => _9X.,1,Set(CALLERID(num)=8040649${CALLERID(num)}) |
| 09:28.27 | axai | exten => _9X.,2,Dial(Zap/g1/${EXTEN:1},190,o) |
| 09:29.02 | JT | right |
| 09:29.46 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
| 09:30.49 | awk | hmm, anyone know what could be cause of this |
| 09:30.49 | awk | Sep 18 11:28:09 WARNING[11899]: app.c:1232 ast_lock_path: Failed to lock path '/var/spool/asterisk/voicemail/default/9999/Old': File exists |
| 09:31.00 | awk | everytime i try login to voicemail it kicks me out |
| 09:31.48 | *** part/#asterisk xnoudas (n=chatzill@ppp171-76.adsl.forthnet.gr) |
| 09:36.20 | yang | Which is a good supported billing solution which works with asterisk 1.2.13 and uses mysql ? |
| 09:36.49 | yang | awk: remove that file then |
| 09:37.12 | yang | and restart asterisk |
| 09:37.17 | awk | so I should remove Old |
| 09:37.19 | awk | I cant restart asterisk |
| 09:37.24 | awk | have about 50 concurrent calls |
| 09:38.36 | axscode | tzafrir: can you help please. |
| 09:39.15 | tzafrir | axscode, so you have no zaptel channels |
| 09:39.29 | tzafrir | aparantly, zapata.conf is misconfigured |
| 09:40.04 | yang | hm no searches found for "asterisk billing" in debian :( |
| 09:41.51 | awk | tng is a very nice billing solution |
| 09:42.00 | awk | we got them to port it to linux |
| 09:42.53 | yang | url |
| 09:45.21 | axscode | tzafrir: i beleive tzafrir.. ill show you in private the zapata.conf |
| 09:45.46 | tzafrir | ~pb |
| 09:45.47 | jbot | A Pastebin is a place to paste your stuff without flooding 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://www.rafb.net/paste |
| 09:46.30 | *** join/#asterisk hank (n=hank@netwichtig.de) |
| 09:46.34 | hank | hi |
| 09:47.31 | hank | i need a little help with dialplan logic. i have this line: exten => *1126,1,GotoIf( ${DB_EXISTS(/Agents/${CALLERID(num)})} )?*1100,1:*1000,1 ) but it always goes to *1100. is there anything wrong with my syntax? |
| 09:48.21 | *** join/#asterisk yannj_fr (n=yannj@chilli.esiee.fr) |
| 09:49.08 | *** join/#asterisk michael-i (n=michael-@Ldb80.l.pppool.de) |
| 09:49.09 | axscode | tzafrir: http://pastebin.com/m293d36c9 |
| 09:50.01 | tzafrir | axscode, looks OK. maybe you need to restart asterisk ? |
| 09:50.06 | tzafrir | or simpler: |
| 09:50.16 | tzafrir | zap restart |
| 09:51.20 | kaldemar | hank: try GotoIf($[${DB_EXISTS(Agents/${CALLERID(num)})}]?*1100,1:*1000,1) |
| 09:51.36 | axscode | tzafrir: http://pastebin.com/m5cbe79e9 |
| 09:52.38 | tzafrir | The error there is that you used fxols=3 in zaptel.conf |
| 09:52.43 | hank | kaldemar: nice, it works :) thanks a lot |
| 09:52.58 | tzafrir | if so, use: signalling=fxo_ls |
| 09:53.02 | tzafrir | in zapata.conf |
| 09:53.09 | axscode | ok |
| 09:59.47 | axscode | tzafrir: http://pastebin.com/m7f5fa408 |
| 10:00.25 | tzafrir | try running it again |
| 10:01.15 | axscode | zap restart? |
| 10:03.34 | *** part/#asterisk hank (n=hank@netwichtig.de) |
| 10:03.50 | axscode | i did. |
| 10:03.58 | axscode | zap show cahnnels |
| 10:04.02 | axscode | i already have 4 channels |
| 10:04.24 | axscode | tzafrir: how to dialplan that my analog telephone will ring? |
| 10:04.32 | axscode | when i dial the number? |
| 10:05.06 | tzafrir | so it should work now |
| 10:05.12 | tzafrir | your dialplan looks OK |
| 10:06.58 | *** join/#asterisk jacq (n=jal@61.12.17.165) |
| 10:08.50 | *** join/#asterisk _mgf (i=mgf@ATuileries-152-1-55-21.w82-123.abo.wanadoo.fr) |
| 10:08.56 | _mgf | hi |
| 10:09.31 | _mgf | i'm looking for a french asterisk developper, is there any here ? |
| 10:10.55 | JT | doubtful |
| 10:13.03 | michael-i | _mgf, I would send a mail to asterisk-biz |
| 10:13.12 | _mgf | ok thanks, i will |
| 10:24.21 | *** join/#asterisk HaMYaI (n=LAMER@ppp-58.8.213.191.revip2.asianet.co.th) |
| 10:24.57 | HaMYaI | Hi, anyone using PAP2T and knows how I can transfer the call to another extension? |
| 10:27.07 | *** join/#asterisk Delvar (n=Delvar@77.240.56.18) |
| 10:34.19 | *** part/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-7af49de505ff13fc) |
| 10:45.08 | yang | How do I enable recording calls for all users in asterisk? |
| 10:45.24 | yang | i was looking at |
| 10:45.29 | yang | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Monitor |
| 10:45.53 | yang | But do I have to enable exten - for every extension? |
| 10:46.59 | McDouglas | any idea what does this mean? http://pastebin.com/d2842b556 |
| 10:48.08 | *** join/#asterisk BockBilbo (n=BockBilb@eu85-84-62-227.clientes.euskaltel.es) |
| 10:54.16 | *** join/#asterisk Daejeo1 (n=chatzill@211.177.189.60) |
| 10:54.25 | Daejeo1 | hello guys |
| 10:57.40 | Daejeo1 | anyone knows how to obtain FACTORY FRESH GPP_K value / vonage adapter |
| 11:10.56 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 11:19.10 | the_lalelu | did someone knows the differents between the digium TE212P and digium TE220B PRI Cards? Specialy the difference between the DSP Chips? |
| 11:22.43 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 11:23.18 | Zeeek | hay |
| 11:26.53 | *** join/#asterisk ironhead_webby (n=webby@202-154-113-132.people.net.au) |
| 11:28.11 | *** join/#asterisk Delvar (n=Delvar@77.240.56.18) |
| 11:29.34 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
| 11:38.44 | *** join/#asterisk ming_zym (n=ming_zym@124.254.55.179) |
| 11:42.33 | Zeeek | now |
| 11:42.45 | *** join/#asterisk MindTheGap (n=iote@c9505ffe.bhz.virtua.com.br) |
| 11:45.08 | *** join/#asterisk saftsack (n=saftsack@pD9E07EE3.dip.t-dialin.net) |
| 11:49.04 | *** join/#asterisk MindTheGap (n=iote@c9505ffe.bhz.virtua.com.br) |
| 11:49.08 | styelz | brown cow |
| 11:49.13 | *** join/#asterisk appelza (n=d@dsl-240-189-01.telkomadsl.co.za) |
| 11:50.30 | appelza | is there a way for asterisk to ignore (pass through) calls to certain numbers? |
| 11:52.35 | JT | ...what? |
| 11:54.06 | styelz | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+LookupBlacklist |
| 11:54.21 | appelza | asterisk is rejecting a number without a destination route, but i dont want to give it a route in asterisk |
| 11:54.25 | styelz | appelza: like that ? |
| 11:54.54 | appelza | think so |
| 11:55.56 | JT | still don't know what you mean by "pass through" |
| 11:58.17 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 11:59.22 | styelz | blacklist is to ignore calls from certain numbers though |
| 11:59.33 | styelz | you want .. to |
| 12:00.07 | styelz | and which way? outgoing calls or incomming |
| 12:03.48 | appelza | incoming |
| 12:05.35 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
| 12:05.46 | thewiizle | hey probably the wrong channel but has anyone set 'astcc' up? |
| 12:06.55 | *** join/#asterisk lsodi (n=lsodi@ts200.wavecom.ee) |
| 12:07.22 | McDouglas | i'm having some trouble with attended call transfer: if i transfer a call to an extension which is busy, i get no notification or anything, but insted i get connected back to the caller on hold |
| 12:07.24 | McDouglas | is that normal? |
| 12:08.35 | lsodi | greetings, Is there way to ignore/block sip user who has tryed to register 10 times with server? registerattempts=10 doesnt help |
| 12:12.23 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:13.31 | JT | styelz: not ignore, process in a different manner |
| 12:13.37 | JT | asterisk doesn't "ignore" calls |
| 12:13.51 | *** join/#asterisk af_ (n=getsmart@81-174-47-128.dynamic.ngi.it) |
| 12:15.04 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 12:15.37 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 12:17.49 | Zeeek | It's up to the user to ignore them |
| 12:19.13 | thewiizle | he's not trying to ignore calls |
| 12:19.24 | thewiizle | he's trying to ban an IP after it has attempted registration >10 |
| 12:19.46 | thewiizle | lsodi, i think registerattempts is for SIP Trunks only |
| 12:21.38 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 12:21.42 | *** join/#asterisk BBHoss (n=hoss@146.229.183.160) |
| 12:21.56 | JT | no such thing as a sip trunk ;) |
| 12:22.03 | JT | and not talking about lsodi's issue |
| 12:22.28 | JT | talking about appelza's issue |
| 12:22.31 | thewiizle | ahh i see |
| 12:22.32 | thewiizle | just saw |
| 12:22.40 | thewiizle | s/trunk/channel |
| 12:23.25 | *** join/#asterisk Delvar (n=Delvar@77.240.56.18) |
| 12:25.40 | *** join/#asterisk Delvar (n=Delvar@77.240.56.18) |
| 12:27.07 | *** join/#asterisk Delvar (n=Delvar@77.240.56.18) |
| 12:27.08 | *** join/#asterisk coppice (n=chatzill@26.162.17.210.dyn.pacific.net.hk) |
| 12:27.56 | lirakis | morning |
| 12:29.13 | *** join/#asterisk yannj_fr (n=yannj@chilli.esiee.fr) |
| 12:31.05 | Zeeek | what is the desierd result of an "ignored" call? asterisk does nothing? |
| 12:31.24 | Zeeek | s/desierd/desired/ |
| 12:31.34 | *** join/#asterisk HarryR (n=harryr@77.240.56.18) |
| 12:31.42 | styelz | like > /dev/null i guess |
| 12:32.04 | styelz | Hangup() |
| 12:32.22 | Zeeek | That result can be had by GoTo to a long wait() |
| 12:32.29 | styelz | or Wait(60) |
| 12:32.34 | Zeeek | right |
| 12:32.44 | styelz | yea |
| 12:33.53 | styelz | sorry lag |
| 12:34.17 | Zeeek | Lager |
| 12:34.26 | thewiizle | shh |
| 12:34.33 | Zeeek | LAGER |
| 12:34.34 | thewiizle | tmi |
| 12:34.40 | Zeeek | noch ein, bitte! |
| 12:34.40 | styelz | BEER!!! |
| 12:34.46 | styelz | larger? |
| 12:35.05 | styelz | lolz |
| 12:37.24 | Zeeek | zvi |
| 12:37.33 | Zeeek | zwei |
| 12:39.27 | styelz | why would eveything work |
| 12:39.32 | axscode | <tzafrir> your dialplan looks OK <-but the fone is not ringing. |
| 12:39.48 | styelz | except for music on hold, for incomming sip calls |
| 12:40.21 | axscode | tzafrir: http://pastebin.com/m56d83819 |
| 12:40.22 | styelz | they get music on hold . if i place the call on hold after picking up the call |
| 12:40.23 | tzafrir | axscode, what do you expect to happen, and what happens? |
| 12:40.37 | *** join/#asterisk xarmiex (n=Armand@static-69-95-184-178.har.choiceone.net) |
| 12:40.41 | axscode | i expect that the analog telephone will ring. |
| 12:40.47 | styelz | but if i send them direct to MOH or to a queue. i get no MOH |
| 12:40.57 | axscode | not untill i pick-it-up |
| 12:41.03 | tzafrir | Zap/1 is an FXO trunk (connects to a telco), right? |
| 12:41.39 | styelz | could it be a bridge issue ? |
| 12:42.05 | axscode | woo.. i thought its for telephone. |
| 12:42.26 | *** join/#asterisk Delvar (n=Delvar@77.240.56.18) |
| 12:42.42 | axscode | thanks.. i got the wrong port.. :) |
| 12:44.11 | axscode | tzafrir: whats wrong if, im calling to the outside (telco), then it continues ringing even if the end user pick-up the phone already.. and keeps on ringing. |
| 12:46.04 | tzafrir | What number are you calling? You dialed an empty number |
| 12:46.15 | *** join/#asterisk creativx (n=creadure@226.62-97-205.bkkb.no) |
| 12:46.25 | tzafrir | To dian 123456 through Zap/1 , use: Dial(Zap/1/123456) |
| 12:47.23 | axscode | tzafrir: ok thanks, how about, im using the zap/3 telephone... im going to dial to a connected sip user. is that automatically? |
| 12:47.29 | axscode | they are at the same context |
| 12:47.45 | [TK]D-Fender | axscode: ..... |
| 12:47.47 | [TK]D-Fender | ~book |
| 12:47.48 | jbot | [book] Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 or Temporarily at http://www.aocomputing.net/AsteriskTFOT.pdf |
| 12:47.49 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^ |
| 12:48.00 | [TK]D-Fender | axscode: and NO, there is no "automatic" |
| 12:48.13 | axscode | ok thanks. :) |
| 12:48.16 | [TK]D-Fender | axscode: Go read up on how to use the dialplan. |
| 12:48.34 | rob0 | Book 'em, Danno. |
| 12:48.38 | [TK]D-Fender | tzafrirIt will be officially released in PDF next week. |
| 12:48.53 | [TK]D-Fender | tzafrir : Following Astricon |
| 12:49.16 | tzafrir | hardcopies will be available in Astricon? |
| 12:49.24 | thewiizle | has anyone got a Linksys 941 to remotely provision? |
| 12:49.29 | Zeeek | Zeeek 2007 will be available in 1st quarter of 2008 |
| 12:49.55 | [TK]D-Fender | tzafrir : To my awareness they're available NOW. dCAP Montreal gave away several during the end meeting. |
| 12:50.37 | [TK]D-Fender | Zeeek: You're ahead of your time, but behind on distribution :p |
| 12:51.36 | Zeeek | Closed beta took longer than I expected! |
| 12:53.07 | xarmiex | does anyone have experience with the linear patch for app_queue ? |
| 12:54.13 | JunK-Y | tzafrir: jsmith told me they will be available at astricon yeah. |
| 12:55.22 | Zeeek | Jared is pretty good about telling the truth |
| 12:57.28 | kaje | is trixbox a debian based distro or red hat based? |
| 12:57.49 | Zyl0ne | centos |
| 12:57.59 | Zeeek | =redhat |
| 12:58.02 | Zyl0ne | which I think is based off redhat |
| 12:58.12 | Zeeek | redhat without the logo |
| 12:58.24 | Zyl0ne | hehe |
| 12:58.25 | Zyl0ne | there ya go |
| 12:58.27 | Zyl0ne | I hate redhat |
| 12:58.29 | Zyl0ne | so bloated |
| 12:58.42 | Zeeek | me too, after lunch |
| 12:58.47 | kaje | hehe, thanks, what about asterisknow? deb or rpm? |
| 12:59.10 | pHnz | asterisknow ? it's not the place for sorry. |
| 12:59.11 | Zeeek | AsteriskNOW will be discussed this week on VOIP USers COnference |
| 12:59.57 | [TK]D-Fender | kaje: rPath. |
| 13:00.35 | Zeeek | http://voipusersconference.org/topics.php |
| 13:00.59 | Zeeek | <PROTECTED> |
| 13:07.32 | *** join/#asterisk chris_1 (n=chris@ng1.kurtkrenn.com) |
| 13:08.07 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
| 13:12.59 | *** join/#asterisk fjean5 (n=fjean5@atelka.info) |
| 13:13.12 | fjean5 | good morning community |
| 13:14.16 | fjean5 | i was wondering if there is any known limit to the number of users defined in iax.conf ? |
| 13:15.39 | yannj_fr | sure there is one? |
| 13:18.35 | JunK-Y | fjean5: not that I know. |
| 13:20.14 | fjean5 | ok, i am asking because we are receiving cores on iax reloads sometimes and we have quite a few in there |
| 13:20.48 | tzafrir | kaje, asterisknow uses rpath packages (conary) |
| 13:21.47 | JunK-Y | fjean5: which * version? |
| 13:21.53 | JunK-Y | how many do u have ? |
| 13:23.40 | Nivex | Qwell: you know you're supposed to take the spam out of the can before you eat it. |
| 13:24.05 | Qwell | Nivex: eh, the can tasted better |
| 13:24.36 | fjean5 | Junk-Y: about 7300 on 1.2.14 |
| 13:25.39 | chris_1 | hi!there seems to be a problem with agentcallbacklogin / queue. suddenly * stopps. any hints? |
| 13:26.18 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 13:26.18 | *** mode/#asterisk [+o mog] by ChanServ |
| 13:31.43 | [TK]D-Fender | chris_1: With the glorious detail you have provided... NO |
| 13:31.58 | chris_1 | :-) |
| 13:38.20 | hypa7ia | Qwell: you fot that too eh :p |
| 13:38.26 | hypa7ia | s/fot/got |
| 13:38.35 | chris_1 | before implemented the queue / agents all went well. now * suddenly stopps (from 2h - 1week); the agent phones are snom360, asterisk v.1.2.13; nothing in the logfiles. |
| 13:39.24 | [TK]D-Fender | chris_1: Well so far you have no evidence to follow up on your problem with and that version is VERY old. |
| 13:39.52 | [TK]D-Fender | chris_1: Go upgrade who knows if whatever problem you're actually facing has been fixed along the way.... |
| 13:40.13 | [TK]D-Fender | chris_1: And 1.2 isn't getting any more bug fixes..... |
| 13:43.09 | *** join/#asterisk ELBunce (n=erik@kde/developer/bunce) |
| 13:44.15 | *** join/#asterisk |NexT| (n=next@netcache1.altecom.net) |
| 13:44.34 | *** join/#asterisk damjan|work (n=damjan@legolas.on.net.mk) |
| 13:46.15 | thewiizle | yo |
| 13:46.40 | thewiizle | im trying to get my dialplan remotely provisioned to my phone however my dialstring replacement is going haywire when i use <0:0044>xxxxxxxxxx. |
| 13:46.45 | thewiizle | does anyone know of a replacement for < |
| 13:46.48 | thewiizle | and > |
| 13:46.53 | damjan|work | can anybody tell me if Asterisk will use multiple CPU's? |
| 13:46.54 | thewiizle | or how to escape them |
| 13:46.59 | coppice | ^ and v? |
| 13:47.13 | damjan|work | or better said, if I get a maxed cpu on my Astersik instance would a dual-core cpu help? |
| 13:47.39 | *** join/#asterisk apardo (n=apardo@9.37.221.87.dynamic.jazztel.es) |
| 13:48.08 | tzafrir | damjan|work, asterisk is multi-threaded, and thus will easily use multiple CPUs. At least for multiple=2 |
| 13:48.37 | damjan|work | tzafrir: thanks, does this apply to 1.2 too? |
| 13:48.47 | tzafrir | damjan|work, yes |
| 13:48.56 | *** part/#asterisk fjean5 (n=fjean5@atelka.info) |
| 13:49.24 | thewiizle | woohoo |
| 13:49.29 | thewiizle | < |
| 13:49.31 | thewiizle | > |
| 13:49.36 | damjan|work | tzafrir: thanks, I've checked with 'ps -axm' but obviously that's different than 'ps axm' |
| 13:52.18 | *** join/#asterisk blinky42 (n=me@67.106.175.130.ptr.us.xo.net) |
| 13:54.37 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 13:55.11 | Uatec | hey, has anyone here used andrews and arnold as a VOIP provider in the uk? |
| 14:03.05 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 14:04.15 | *** join/#asterisk anonymouz666 (n=anonymou@189.25.37.205) |
| 14:04.16 | mattboll | hi |
| 14:05.01 | mattboll | does anyone know something about asterisk when it isn't on the local network ? I can connect to it and call but not talk |
| 14:05.14 | mattboll | and I don't know where I should search (and what) |
| 14:05.41 | mattboll | my be it an ekiga problem ? |
| 14:06.23 | [TK]D-Fender | mattboll: .... |
| 14:06.26 | [TK]D-Fender | ~sipnat |
| 14:06.27 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 14:06.28 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 14:06.41 | datachomper | Asterisk is throwing "status NOANSWER" on calls to a specific number and dropping the call, yet my cell phone connects through. What would cause this? |
| 14:06.49 | *** join/#asterisk funxion (n=nunya@63.214.236.169) |
| 14:07.07 | mattboll | thx :) |
| 14:07.21 | [TK]D-Fender | datachomper: PASTEBIN is your friend..... |
| 14:07.23 | [TK]D-Fender | ~pb |
| 14:07.23 | jbot | A Pastebin is a place to paste your stuff without flooding 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://www.rafb.net/paste |
| 14:07.25 | [TK]D-Fender | ^^^^^^^^^^^^^^ |
| 14:07.25 | datachomper | We are dialing every other number just fine... |
| 14:08.57 | k31th | Afternoon guys. |
| 14:09.18 | tzafrir | someone in the proper time zone :-) |
| 14:09.21 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 14:12.26 | *** join/#asterisk jsmith (n=jsmith@000-181-995.area3.spcsdns.net) |
| 14:12.26 | *** mode/#asterisk [+o jsmith] by ChanServ |
| 14:12.32 | *** join/#asterisk Blackthorn (i=blacktho@76.77.160.10) |
| 14:13.50 | *** part/#asterisk goupil (n=goupil@62.147.224.49) |
| 14:14.55 | *** join/#asterisk ming_zy1 (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 14:15.39 | McDouglas | [TK]D-Fender: that first link about sip+nat says i have to forward the rtp port range to asterisk. Will it cause any problem if i shrink that range to .. say 100 ports inrtp.conf? |
| 14:16.03 | [TK]D-Fender | McDouglas: only if you need more that you set for |
| 14:16.20 | McDouglas | how can i calcualte? one for each sip conversation? |
| 14:16.32 | [TK]D-Fender | McDouglas: Exactly |
| 14:17.06 | RypPn | I was led to believe it was 4 |
| 14:17.14 | lirakis | does any one here actually have a gxp-2020 ? |
| 14:17.32 | jsmith | RypPn: It depends on whether you get RTCP messages |
| 14:17.36 | [TK]D-Fender | lirakis: There must be some sort of chump out there who thinks its a Csico ;) |
| 14:17.39 | [TK]D-Fender | Cisco* |
| 14:17.52 | lirakis | [TK]D-Fender: sure you dont have one hidden away in your closet ?? lol |
| 14:18.06 | RypPn | jsmith: so 4 would be a safer assumption? |
| 14:18.35 | jsmith | RypPn: Yeah, that's a fairly safe assumption |
| 14:18.37 | [TK]D-Fender | jsmith: Still on the road? |
| 14:18.45 | lirakis | [TK]D-Fender: heh heh |
| 14:18.51 | jsmith | [TK]D-Fender: Nope... back home for a week, before I head off to AstriCon next week |
| 14:19.02 | Qwell | w00t, astricon |
| 14:19.05 | datachomper | would zapateller on the destination asterisk box, cause my calls to be dropped? |
| 14:19.07 | [TK]D-Fender | jsmith: Enjoy the breather.. the party never ends :) |
| 14:19.11 | jsmith | [TK]D-Fender: You know, no rest for the wicked... |
| 14:19.30 | jsmith | Yeah, I'm looking forward to AstriCon... it's always my favorite time of the year |
| 14:19.34 | xheliox | Strange question -- If both Asterisk and Zaptel are not started.. but the phone lines are plugged into a TDM400P, should the lines ring busy? |
| 14:19.36 | [TK]D-Fender | jsmith: Insomniacs Anonymous... I know thee well |
| 14:19.38 | Qwell | jsmith: agreed |
| 14:20.09 | *** join/#asterisk Yourname`` (n=IM@unaffiliated/yourname/x-837320) |
| 14:20.13 | jsmith | xheliox: No, they just won't be answered |
| 14:20.19 | xheliox | Yeah.. weird. |
| 14:20.21 | jsmith | [TK]D-Fender: Exactly... |
| 14:20.22 | Qwell | [TK]D-Fender: You should go. You still have like 6 days to make arrangements |
| 14:20.30 | Yourname`` | WARNING[24112]: chan_sip.c:12224 handle_response_register: Got 200 OK on REGISTER that isn't a register |
| 14:20.31 | jsmith | [TK]D-Fender: You're not going? |
| 14:20.31 | xheliox | jsmith: That's not what's happening.. |
| 14:20.32 | Yourname`` | Error! |
| 14:20.55 | xheliox | When I unplug the lines, it rings.. |
| 14:21.16 | jsmith | xheliox: Well, I guess I'm wrong then :-) |
| 14:21.27 | xheliox | jsmith: Are you? Or is something wrong? :p |
| 14:21.33 | [TK]D-Fender | jsmith: I'm not a real coder, am not running a full-on VoIP business, and its a big trip and I don't have a passport..... |
| 14:21.34 | xheliox | Because the card isn't working properly... that's why I ask. |
| 14:21.36 | jsmith | xheliox: Naw, I"m probably just wrong. |
| 14:21.47 | Qwell | [TK]D-Fender: it's not just for devs |
| 14:21.49 | [TK]D-Fender | jsmith: So regrettably no. No chance of my day-job sending me ANYWHERE |
| 14:21.50 | jsmith | [TK]D-Fender: So, so, and oh... |
| 14:21.54 | xheliox | anyone else want to confirm jsmith's wrongness? |
| 14:22.02 | xheliox | :) |
| 14:22.04 | jsmith | [TK]D-Fender: The first two were lame excuses, but the third might be a problem |
| 14:22.16 | Qwell | bbl |
| 14:22.48 | [TK]D-Fender | jsmith: Hish-cost, low-return, and the mandatory DHS invasive cavity search is on my no-no list :p |
| 14:22.53 | [TK]D-Fender | high* |
| 14:23.28 | jsmith | [TK]D-Fender: Hey, if blitzrage can get across, anyone can... |
| 14:23.51 | [TK]D-Fender | jsmith: Yes, but he LIKES the DHS ICS! |
| 14:24.04 | hypa7ia | [TK]D-Fender: at least we don't need to get fingerprinted yet :( |
| 14:24.24 | [TK]D-Fender | hypa7ia: YET. |
| 14:24.35 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 14:24.47 | hypa7ia | [TK]D-Fender: when they do, no more defcon for me :( |
| 14:25.56 | [TK]D-Fender | hypa7ia: Terrist!!!!!!!!!!! |
| 14:26.19 | kaje | once I install asterisk how do I configure it? the link on asterisk's webpage is broken... |
| 14:26.41 | hypa7ia | [TK]D-Fender: i just don't want anyone having my prints, is all :) |
| 14:27.20 | hypa7ia | kaje: start by doing make samples, and have a look at what that puts in /etc/asterisk |
| 14:27.28 | hypa7ia | kaje: there's a bit of a learning curve |
| 14:27.47 | Sci_05 | hypa7ia: just do what the guy in the movie 7 did, cut them off ;-) |
| 14:27.59 | kaje | did that... well, I'm trying to get it to work with asterfax and asterfax asked for a username and password from manager.conf... but there isn't one in there |
| 14:28.32 | hypa7ia | Sci_05: ouch |
| 14:28.50 | hypa7ia | kaje: have you looked up asterfax on voip-info? |
| 14:28.59 | *** join/#asterisk gardo (n=gardo@121.97.177.138) |
| 14:29.09 | *** join/#asterisk Boones (i=Boones@port-212-202-170-97.dynamic.qsc.de) |
| 14:29.22 | jsmith | kaje: I'm working on the AsteriskDocs website... in the meantime, you can download the PDF from a mirror... I'll find you a link |
| 14:29.33 | [TK]D-Fender | kaje: ... |
| 14:29.35 | [TK]D-Fender | ~book |
| 14:29.35 | jbot | [book] Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 or Temporarily at http://www.aocomputing.net/AsteriskTFOT.pdf |
| 14:29.37 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 14:30.18 | [TK]D-Fender | there |
| 14:30.51 | jsmith | Ah, thankis [TK]D-Fender |
| 14:31.08 | kaje | nice, that book is in safari books online, thanks guys!! |
| 14:31.19 | jsmith | kaje: Yup :-) |
| 14:31.38 | jsmith | kaje: The second edition is in Safari now, and will be available as a free PDF next week |
| 14:31.53 | kaje | yeah, I saw that =) |
| 14:34.03 | *** join/#asterisk Victor_Yure (n=aaaa@esp5.deibotoch.com.br) |
| 14:34.43 | Yourname`` | Yeah, right now there isn't any index. |
| 14:34.48 | Yourname`` | Good luck, jsmith. |
| 14:34.53 | Yourname`` | May the httpd force be with you. |
| 14:35.09 | Yourname`` | WARNING[24112]: chan_sip.c:12224 handle_response_register: Got 200 OK on REGISTER that isn't a register <= Why am I getting this? |
| 14:37.08 | jsmith | Yourname``: Can you pastebin the SIP messaging... it seems that someone sent you a "200 OK" response that's fishy |
| 14:37.46 | Yourname`` | ok |
| 14:39.36 | *** join/#asterisk Lucky7 (n=Adam@207.200.28.175) |
| 14:45.00 | [TK]D-Fender | jsmith: Don't forget to advise me the moment I'm cleared to mirror TFOT 2nd Ed :) |
| 14:45.53 | *** join/#asterisk bintut (n=bintut@cm47.gamma178.maxonline.com.sg) |
| 14:45.55 | jsmith | [TK]D-Fender: For the second edition, O'Reilly is going to handle the mirroring, so that we can get more effective statistics |
| 14:46.03 | thewiizle | fucking yes |
| 14:46.10 | thewiizle | remote provisioning is COOL as fuck |
| 14:46.17 | *** join/#asterisk pzn (i=foobar@201-26-168-225.dial-up.telesp.net.br) |
| 14:46.30 | [TK]D-Fender | jsmith: load chan_expletivedelete.so |
| 14:46.35 | jsmith | [TK]D-Fender: One of the things Tim was unhappy about with the first edition is that he didn't really know how many times it had been downloaded |
| 14:47.05 | jsmith | [TK]D-Fender: And since this is somewhat of an experiment with them, we figured we'd humor them and give them a chance... |
| 14:47.17 | [TK]D-Fender | jsmith: Don't think that I'm offering up my bandwodth like its nothing, this was a NECESSARY evil.... |
| 14:47.25 | *** join/#asterisk nfi|ermes (n=ErMeS@217.220.121.62) |
| 14:47.26 | pzn | Hi! Can you recommend a client for voip for using with asterisk (a free software version). I need a simple client, it can even be command line; however it must be good and stable... thanks! |
| 14:47.28 | jsmith | [TK]D-Fender: Please explain |
| 14:47.44 | jsmith | pzn: For which operating system? |
| 14:47.45 | pzn | I mean a client for using speaker+mic of the PC |
| 14:47.52 | pzn | jsmith, for linux |
| 14:48.27 | [TK]D-Fender | jsmith: Astriskdocs was down and people come in here looking for help and resources, both of which I provide in large quantities. As sson as they lost access to the book I ensured that no user would have to sit and wait. |
| 14:48.54 | [TK]D-Fender | jsmith: Just doin my part.... |
| 14:49.29 | *** join/#asterisk Buhntz (n=bytewalk@port-212-202-170-97.dynamic.qsc.de) |
| 14:49.34 | *** join/#asterisk cuco (n=diego@62.90.10.53) |
| 14:49.37 | [TK]D-Fender | pzn: Ekiga, Twinkle, Zoiper, etc... |
| 14:49.59 | cuco | bkruse: ping |
| 14:50.39 | pzn | [TK]D-Fender, I'll check this clients. thanks! |
| 14:51.13 | [TK]D-Fender | pzn: Ekiga is the most featured of them, supporting video, transfers, conference, etc. |
| 14:52.10 | *** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 14:52.33 | jsmith | [TK]D-Fender: I'm fine with you turning up a mirror if O'Reilly's goes down... no problems with that whatsoever. We're trying to get it off of asteriskdocs.org as well (as the site has been unstable) |
| 14:52.58 | kippi | is there any reason why when using ChanSpy, why i can't hear the other person, only when the person at my end is talking |
| 14:53.21 | pzn | [TK]D-Fender, what I really need is a sample code for building my own client (it will be command line). I just need simple call with G711u codec. do you have other suggestions? |
| 14:53.54 | [TK]D-Fender | pzn: Google up SIP libraries, there are a number of them out there. |
| 14:54.48 | coppice | good clients aren't simple |
| 14:55.31 | jsmith | kippi: You can also try the iaxclient library, if you prefer IAX instead of SIP |
| 14:55.37 | jsmith | coppice: Amen to that! |
| 14:55.58 | Uatec | xlite has a crap front end |
| 14:56.03 | Uatec | but the back end is published i believe |
| 14:56.09 | *** join/#asterisk denon (n=denon@208.122.43.201) |
| 14:56.09 | *** mode/#asterisk [+o denon] by ChanServ |
| 14:56.18 | *** join/#asterisk Delvar (n=Delvar@77.240.56.18) |
| 14:57.08 | *** join/#asterisk mog (i=mog@nat/digium/x-054a1723a9e9839c) |
| 14:57.08 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:57.27 | pzn | [TK]D-Fender, nice, google pointed me to http://www.gnu.org/software/osip/ it seems to solve my problem. thanks! |
| 14:58.13 | coppice | well, it might solve the SIP part. what are you going to do about the rest? |
| 14:58.31 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 14:59.30 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:02.16 | coppice | osip is not very good. sofia is the best SIP library out there |
| 15:02.28 | Uatec | what kind of device would i use to connect my asterisk box to an analogue POTS line? |
| 15:02.58 | jsmith | Uatec: Something like a Digium TDM400P card (with at least one FXO port), or an ATA that has an FXO port |
| 15:03.05 | *** join/#asterisk lsodi (n=lsodi@80-235-55-96-dsl.kjj.estpak.ee) |
| 15:03.10 | *** join/#asterisk blackgecko (n=blackgec@200.36.96.215) |
| 15:03.15 | Strom_M | Uatec: a digium TDM card |
| 15:03.26 | *** join/#asterisk beek (n=klinebl@64.9.22.203) |
| 15:03.27 | funxion | has anyone in here ever gotten a quote from digium before for custom development? |
| 15:04.03 | jsmith | funxion: No, but I've worked with them on a couple of projects... why? |
| 15:04.25 | blackgecko | does anyone knows a way to restrict who can call a specific extension ?? |
| 15:04.47 | funxion | I aked for a quote for some custom development so I can get an approval and the salesman aked me to "buy an hour of time |
| 15:04.57 | funxion | " so he can derive a quote |
| 15:05.02 | jsmith | blackgecko: There are lots of ways... you could use the Authenticate() dialplan application to force them to enter a PIN number |
| 15:05.42 | funxion | what kind of idiot would I look like to my employer to have a quote as an expense for a project |
| 15:06.16 | blackgecko | i dont want to authenticate |
| 15:06.43 | *** join/#asterisk Delvar (n=Delvar@77.240.56.18) |
| 15:06.47 | *** join/#asterisk nephfl (n=none@wsip-68-110-130-57.ga.at.cox.net) |
| 15:06.50 | blackgecko | i tried doing somethign like exten => 500/100,1,dial(Sip/500) |
| 15:06.52 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 15:06.52 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:07.03 | elixer | funxion: is that a rhetorical question? ;-) |
| 15:07.15 | funxion | lol |
| 15:07.19 | nephfl | im having trouble with a system using a digium analog card...when it tries to dial aparently the first digit isnt getting accepted sometimes |
| 15:07.30 | nephfl | anyone know how i can fix that? |
| 15:07.38 | funxion | Im just wondering if the guy had a bad day or if that is SOP for digium |
| 15:07.40 | blackgecko | nephf1: put a pause before the dial |
| 15:07.46 | jsmith | nephfl: You can add a "w" to the front of the number, which will tell Asterisk to wait a half-second before dialing |
| 15:08.00 | nephfl | i see |
| 15:08.27 | jsmith | funxion: Oh, the guy from Digium told you to buy an hour so that they could then generate the quote? |
| 15:08.32 | funxion | yeah |
| 15:08.40 | blackgecko | @jsmith: any other way to restrict the callers ? |
| 15:08.40 | jsmith | funxion: That's interesting... |
| 15:08.48 | funxion | I thought so too |
| 15:09.04 | funxion | I'm a bit disapointed |
| 15:09.07 | jsmith | blackgecko: Sure, I can think of a bunch... give me more details on exactly what you're trying to do, and I can help more. |
| 15:09.08 | outtolunc | was it a complex quote (big job) |
| 15:09.11 | *** join/#asterisk hfb (n=hfb@pool-71-118-252-254.lsanca.dsl-w.verizon.net) |
| 15:09.14 | [TK]D-Fender | blackgecko: Yeah, put your users in contexts that don't even OFFER the extens you don't want them to dial in the first place. |
| 15:09.15 | funxion | no |
| 15:09.21 | outtolunc | weird |
| 15:09.23 | lsodi | greetings, somwhere is one misconfigured sip device witch tryes to register to asterisk, how can I ban that device for a while, or reregistering after failure is configured from device? |
| 15:09.44 | blackgecko | @jsmith: i have a voicing system hooked to a grandstream configured to auto answer |
| 15:09.49 | funxion | I asked for a patch that is currently in existance and working on * open source to be ported to business edition |
| 15:09.54 | funxion | how hard is that |
| 15:10.09 | Qwell | funxion: is it a feature? |
| 15:10.16 | funxion | probably werx as is but I didnt get teh source with business edition so I dont knwo |
| 15:10.17 | outtolunc | depending on the feature it could be very hard <G> |
| 15:10.21 | *** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-76a70b6438e4bc37) |
| 15:10.27 | blackgecko | @jsmith: but dont want that everyone can use without control |
| 15:10.28 | funxion | its a mod to chan_sip |
| 15:10.40 | funxion | to make an operator panel for thomson phones werk |
| 15:11.00 | jsmith | funxion: Asterisk Business Edition is currently derived from the Asterisk 1.2 source. The next version (coming soon!) will be based on Asterisk 1.4. |
| 15:11.16 | funxion | i know but what rev |
| 15:11.19 | [TK]D-Fender | jsmith: Likely jsut after the 1.6 release party! ;) |
| 15:11.23 | outtolunc | maybe it is due to it being the business version and they have to do all the retesting to cert it .. whatever |
| 15:11.27 | jsmith | blackgecko: OK, what determines who can and can't call it? |
| 15:11.42 | blackgecko | @jsmith: instead i want to be able to say wich extensions can dial the voicing one based on the CID |
| 15:11.54 | jsmith | funxion: I don't know, to be honest. |
| 15:11.57 | blackgecko | @jsmith: my self jejeje |
| 15:12.08 | outtolunc | funxion: call again <G> |
| 15:12.18 | jsmith | blackgecko: GotoIf($[${CALLERID(num) = 101]?voicing,s,1) |
| 15:12.22 | jsmith | blackgecko: GotoIf($[${CALLERID(num) = 102]?voicing,s,1) |
| 15:12.22 | funxion | I'm considering trying to compile with a couple different rev and trying to replace the chan_sip module |
| 15:12.39 | [TK]D-Fender | blackgecko: Then add a pile of GotoIf's to check for CID's or a bunch of exten =>100/XXX,1, etc |
| 15:12.43 | *** join/#asterisk masus (n=tet@88.248.73.2) |
| 15:12.51 | blackgecko | @jsmith: and a one like that for each cid i need to be able to call ? |
| 15:12.58 | masus | hi all, is it possible to authenticate an agent with voice speaking |
| 15:13.11 | outtolunc | funxion: that sounds extreme if you are gonna replace the whole module |
| 15:13.12 | jsmith | blackgecko: Yup, that's one way |
| 15:13.15 | blackgecko | @jsmith: i tried the second one without success |
| 15:13.21 | funxion | outtolunc apparently the guy I spoke with is the head of custom dev sales |
| 15:13.34 | funxion | outtolunc its like 4 lines of code addded to chan_sip.c |
| 15:13.38 | jsmith | masus: No, not really... we don't have any kind of voice fingerprinting in Asterisk |
| 15:13.40 | outtolunc | hehe <G> |
| 15:13.42 | funxion | just needs to be recompiled |
| 15:13.53 | masus | jsmith: ok thanks |
| 15:14.41 | blackgecko | @jsmith: ill check with the gotos, thanks man |
| 15:14.52 | [TK]D-Fender | blackgecko: PASTEBIN your coding attempts... |
| 15:14.54 | [TK]D-Fender | ~pb |
| 15:14.54 | jbot | A Pastebin is a place to paste your stuff without flooding 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://www.rafb.net/paste |
| 15:14.55 | [TK]D-Fender | ^^^^^^^^^^^^^^ |
| 15:14.57 | yang | can someone tell me how to add proper monitor lines to this output (4 lines) http://pastebin.ca/701981 |
| 15:15.05 | [TK]D-Fender | blackgecko: And we'll tell you where the errors are |
| 15:15.31 | [TK]D-Fender | yang: Monitor has to be set BEFORE your dial |
| 15:15.37 | outtolunc | funxion: i thought the business version you got the source code with it |
| 15:16.15 | funxion | outtolunc nope |
| 15:16.45 | blackgecko | [TK]D-Fender: the problem was that only the first declaration wokrs |
| 15:17.10 | outtolunc | hmm it reads 'source code for the drivers are included' |
| 15:17.11 | blackgecko | [TK]D-Fender: the second one gave me fast busy on the phone before getting to asterisk |
| 15:17.28 | outtolunc | chan_sip is a 'channel driver' |
| 15:17.38 | [TK]D-Fender | yang: http://pastebin.ca/701983 <- keep in mind I left your Macro based vars in there. |
| 15:18.05 | [TK]D-Fender | blackgecko: PASTEBIN it so we can see what you're doing. |
| 15:18.07 | blackgecko | [TK]D-Fender:but ill check it again, maybe y typed something wrong, one las question, all the priorities has to be number 1 ? |
| 15:18.28 | [TK]D-Fender | blackgecko: Depends who's idae you are following. |
| 15:18.42 | [TK]D-Fender | blackgecko: PASTEBIN IT. |
| 15:19.14 | *** join/#asterisk Delvar (n=Delvar@77.240.56.18) |
| 15:19.21 | yang | [TK]D-Fender: thank you ARG2=EXTEN and ARG1=CALLER ID....I dont really distiguish between these 2....my username is the same as callerid, so i keep both values the same? |
| 15:20.59 | yang | [TK]D-Fender: so the first line comes as exten => 200,1,Set(MONITOR_FILENAME=${REC_DIR}/${TIMESTAMP}-${200}-${200}-out) |
| 15:21.13 | jsmith | [TK]D-Fender: Don't forget, we've got asterisk.pastebin.ca now too :-) |
| 15:21.17 | [TK]D-Fender | yang: Well the way I did your sample was as a fixed exten where ARGX doesn't exist. |
| 15:21.41 | yang | [TK]D-Fender: yeah i am wondering if i can replace ARG-1 and ARG-2 with number? |
| 15:21.45 | [TK]D-Fender | yang: and last I checked ${200} wasn't a valid var name... |
| 15:21.52 | funxion | outtolunc I searched the server and the install cd's for it to no avail |
| 15:22.13 | ai-a | we have 6 fax machines on our network connected to Asterisk via LinkSys Sipura SPA-1001 devices, With also about 40+ voice phones on the system. 2 of the fax machines just refuse to work. Get CID but transmission fails. bandwidth usage over the network is low. Connected a analogue phone shows 2 way audio works. However faxing to/from these 2 fax machines to other fax machines on the PBX or external fail most of them time. Any |
| 15:22.18 | [TK]D-Fender | jsmith: pastebin.aocomputing.net <- Got my OWN |
| 15:22.20 | blackgecko | [TK]D-Fender: http://pastebin.com/m3a4d2473 |
| 15:22.27 | yang | [TK]D-Fender: Or do I put it like this |
| 15:22.28 | outtolunc | funxion: well if they got the code, you are at their mercy |
| 15:22.33 | *** join/#asterisk Tili (n=tili@cm48.gamma244.maxonline.com.sg) |
| 15:22.44 | yang | [TK]D-Fender: so the first line comes as exten => 200,1,Set(MONITOR_FILENAME=${REC_DIR}/${TIMESTAMP}-${CALLERID}-${EXTEN}-out) |
| 15:24.22 | *** join/#asterisk mog (i=mog@nat/digium/x-1f2831cb2fa46107) |
| 15:24.22 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:28.17 | [TK]D-Fender | jsmith: I've got my support bases covered..... everything except a mirror for the WIKI :p |
| 15:28.36 | JerJer | the wiki is evil |
| 15:28.43 | [TK]D-Fender | blackgecko: taht should allow only 3 callers to dial 790 |
| 15:29.04 | [TK]D-Fender | ..... telnet |
| 15:29.04 | Nugget | telnet is eeeeeeevil! |
| 15:29.18 | [TK]D-Fender | More "e"'s FTW! |
| 15:29.37 | jsmith | No, what's more evil is writing a perl script with Net::Telnet to provision phones :-) |
| 15:30.47 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 15:31.24 | Blackthorn | what does extensions.ael do? |
| 15:31.26 | defswork | what was the command line to get an aastra phone to reload ? |
| 15:32.27 | yang | [TK]D-Fender: well, i did extensions reload ... but i dont get any loggin in /var/spool/asterisk/monitor...? |
| 15:33.16 | *** join/#asterisk thx2000 (n=evan@netblock-208-127-150-56.dslextreme.com) |
| 15:33.37 | thx2000 | Does anyone know if it's possible to have at&t forward on busy to a voip DID? |
| 15:33.47 | jsmith | Blackthorn: You can write your dialplan in AEL (the Alternative Extension Language) if you prefer it to the regular dialplan language |
| 15:34.09 | jsmith | defsdoor: Probably "sip notify [peer]" |
| 15:34.51 | *** join/#asterisk admin0 (n=admin@bb121-6-233-92.singnet.com.sg) |
| 15:34.57 | [TK]D-Fender | yang: pastebin a call attempt at verbose 10 |
| 15:35.17 | *** join/#asterisk Norm (n=Norm@normmac.net.wm.edu) |
| 15:35.35 | Norm | has there been any experience with avaya handsets on an asterisk server? i don't see them listed in the wiki |
| 15:36.08 | admin0 | Hi. I am setting up a asterisk box .. when I dial, even though I have disallow=all and allow=g723.1 , it always go out via as gsm . my linksys pap2 is sending the calls as g723.1 .. shouldn't it be doing a passthru ? |
| 15:36.30 | admin0 | i tried to put allow=first and disallow=all in line below, but that too does not help |
| 15:36.46 | Blackthorn | I get alot of context not found or context is empty int he log file messsages. should that be of any concern? |
| 15:36.49 | Strom_M | asterisk calls it g723, not g723.1 IIRC |
| 15:37.07 | jsmith | Blackthorn: Yah, probably |
| 15:37.24 | coppice | well, if you want to name it properly its G.723.1 |
| 15:37.28 | yang | [TK]D-Fender: http://pastebin.ca/702007 |
| 15:37.53 | Blackthorn | jsmith: does that mean yes it should be a concern or no it should not be? |
| 15:38.29 | *** join/#asterisk Delvar (n=Delvar@77.240.56.18) |
| 15:38.31 | [TK]D-Fender | admin0: If you outbound channel is GSM then coming from G.723.1 is NOT passthrough, you need to transcode it, and there is no legal codec available for * for that except the TC100 transocder card |
| 15:39.25 | admin0 | my outbound gateway supports g723.1 as well as g729 |
| 15:39.33 | [TK]D-Fender | yang: All I see is a DIAL, no call to monitor BEFORE dialing... |
| 15:40.01 | yang | [TK]D-Fender: extensions reload is enough or do i need to restart asterisk |
| 15:40.30 | admin0 | so put disallow=all and allow=G.723.1 in the outbound peer ? |
| 15:40.43 | jsmith | Blackthorn: Yes, that should be a concern |
| 15:40.54 | [TK]D-Fender | admin0: Should be "allow=g723.1", no uppercase |
| 15:41.11 | *** join/#asterisk AndrewGearhart (n=chatzill@h1.39.213.151.ip.alltel.net) |
| 15:41.16 | [TK]D-Fender | yang: Yes, extensions reload is enough... |
| 15:41.20 | admin0 | ditto for inbound user also ? |
| 15:41.34 | Blackthorn | jsmith: i actually removed everything int he file and let it be blank but that too created all sorts of messages. |
| 15:41.51 | [TK]D-Fender | yang: Of course I just gave you an EXAMPLE, I have no idea how you implemented it, what context you put it into, etc, so there are a million things you could have done wrong... |
| 15:41.57 | jsmith | Blackthorn: Oh, from the extensions.ael... just delete it! |
| 15:42.09 | AndrewGearhart | good morning folks. Anybody have suggestions on a PC or motherboard/processor combination to use for an Asterisk PBX? |
| 15:42.13 | russellb | http://www.vote756.com/marcecko/ |
| 15:42.20 | yang | [TK]D-Fender: and I dont know why it says SIP-200 If my username is 600... |
| 15:42.20 | russellb | everyone vote to brand it with an asterisk! |
| 15:42.42 | [TK]D-Fender | Norm: Few people use Avaya handsets with *. It is ill-advised unless you're already stuck with them. |
| 15:42.58 | Norm | [TK] - thanks |
| 15:43.49 | [TK]D-Fender | AndrewGearhart: Just about anything I guess.... maybe a nice supermicro Zeon rackmount server... |
| 15:44.13 | [TK]D-Fender | admin0: if it doesn't match it will try to transcode and fail. |
| 15:44.38 | AndrewGearhart | [TK]D-Fender: not sure a supermicro Xeon rackmount server is in the budget. ;-) |
| 15:44.46 | admin0 | thanks all |
| 15:44.47 | admin0 | its working now |
| 15:45.04 | [TK]D-Fender | AndrewGearhart: Then go with "whatever". |
| 15:45.18 | AndrewGearhart | [TK]D-Fender: plus, will your sangoma 200d cards fit in a supermicro rackmount? ;-) |
| 15:45.49 | [TK]D-Fender | AndrewGearhart: Yup. They come in LP bracket sizes, and many of their servers have PCI risers as well |
| 15:47.23 | the_lalelu | Can someone tell me the differents between the digium TE212P and digium TE220B PRI Cards? Especialy the difference between the DSP Crap? |
| 15:48.54 | yang | [TK]D-Fender: basically i only wanted a MONITOR to record my calls, my friend gave me those 2 strings, but they dont seem working |
| 15:49.42 | *** join/#asterisk teknoprep (n=teknopre@unaffiliated/teknoprep) |
| 15:49.42 | teknoprep | yo |
| 15:49.43 | [TK]D-Fender | yang: Show me what you've done because clearly the last bit of CLI you showed me wasn't even CALLING monitor. So you dialplan or devices settings are screwed up |
| 15:49.50 | teknoprep | how do i add a + to all outbound calls? |
| 15:50.02 | teknoprep | i have NXXNXXXXXXX |
| 15:50.16 | teknoprep | well NXXNXXXXXX |
| 15:50.19 | Strom_M | the_lalelu: the 212 is for a 5v PCI slot, while the 220 is for a PCI Express slot |
| 15:50.24 | teknoprep | i want to add a + to the front of that... |
| 15:50.31 | teknoprep | for outbound calls |
| 15:50.47 | *** join/#asterisk Teeli (n=tili@cm48.gamma244.maxonline.com.sg) |
| 15:50.51 | yang | [TK]D-Fender: shall i paste you the whole extension.conf? |
| 15:51.16 | [TK]D-Fender | teknoprep: +${EXTEN} |
| 15:52.13 | [TK]D-Fender | yang: just look at from-local-users, 0038641710598, 1 yourself. Why is step 1 DIAL? |
| 15:52.13 | jsmith | teknoprep: Find the extension that actually calls Dial(), and add the + before the ${EXTEN} |
| 15:53.03 | yang | [TK]D-Fender: http://www.pastebin.ca/702019 |
| 15:54.19 | [TK]D-Fender | yang: you are only monitoring your INTERNAL extension, not what you use to dial out your ITSP <----------- |
| 15:55.00 | [TK]D-Fender | yang: and in a lot of places you are still calling monitor AFTER YOUR DIAL. |
| 15:55.24 | admin0 | guys |
| 15:55.26 | admin0 | it helped |
| 15:55.38 | yang | [TK]D-Fender: yes, but i am testing just 600 extension now |
| 15:55.49 | yang | [TK]D-Fender: do i need to change all the lines? |
| 15:55.56 | the_lalelu | Strom_M: ok - except that the 212 is for a 3,3V pci slot. but why is the TE220 (even if bundled to TE220B) cheaper? is this because of an older dsp chip? for me it looks like the TE212P and the TE220B are almost similar ... |
| 15:56.08 | *** join/#asterisk arekm (i=arekm@pld-linux/arekm) |
| 15:56.43 | [TK]D-Fender | yang: First of all, your pastbin was for a call placed by SIP/200-b5d10858 . and you were calling SIP/0038641710598@e1|60|t. Wake up and realize what you're doing here.... |
| 15:57.26 | arekm | what could be the reason? |
| 15:57.27 | the_lalelu | Strom_M: i mean from the point of features almost similar. |
| 15:58.10 | [TK]D-Fender | arekm: because the exten you showed is 36677 and its looking for 3667741 <---------- |
| 15:58.19 | [TK]D-Fender | arekm: 36677 != 3667741 |
| 15:59.06 | arekm | [TK]D-Fender: ok but wasn't there some "as you dial" thing? |
| 15:59.37 | [TK]D-Fender | the_lalelu: 3.3V PCI is for servers and has lower production QTY's, thus the card would cost more to produce |
| 15:59.40 | *** join/#asterisk eldon (i=eldon@nat/digium/x-8e44654f0ba1e680) |
| 15:59.40 | arekm | [TK]D-Fender: I should ask directly what I'm trying to archieve. the goal is to strip 36677 from beginnig and use short 2 digi numbers in dial plan |
| 15:59.51 | Blackthorn | is it common to log messages such as "[Sep 18 11:56:18] WARNING[4450] chan_iax2.c: Call rejected by 64.61.93.90: No such context/extension |
| 16:00.09 | [TK]D-Fender | arekm: * is trying to find a match for the incoming call and there ISN'T ANY. |
| 16:00.14 | yang | [TK]D-Fender: that was another user coming from 200 i think...mine was 600....well sorry to disturb you so much, i modified my extensions.conf now tell me if it looks proper...http://www.pastebin.ca/702025 |
| 16:00.15 | Blackthorn | when that ip is known to not be a user of your * box? |
| 16:00.22 | the_lalelu | [TK]D-Fender: aaaah, ok. that could be an answer. thx. |
| 16:00.33 | [TK]D-Fender | arekm: Once you create a match for the incoming call pattern you can THEN mangle it up any way you feel like. |
| 16:00.36 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 16:00.54 | arekm | [TK]D-Fender: how to strip it then? |
| 16:01.08 | blackgecko | does anynone know of some issue between a te410 and a dell poweredge 1900, im having kernel panics since we installed this card |
| 16:01.20 | arekm | I know how to strip in Dial() only ;( |
| 16:01.32 | alrs | blackgecko: Dell motherboards and Digium cards have known issues |
| 16:02.02 | alrs | blackgecko: which ones exactly I dunno, but I've read an article on the subject on voip-info.org |
| 16:02.11 | *** join/#asterisk Delvar (n=Delvar@77.240.56.18) |
| 16:02.33 | blackgecko | yeah thats what i tougth, do you think that changing it for a sangoma sould solve the problem ? |
| 16:02.55 | yang | [TK]D-Fender: if its related I am trying to call outside numbers through a SIP trunk |
| 16:03.40 | *** join/#asterisk modu (n=modu@rue92-6-82-237-172-115.fbx.proxad.net) |
| 16:03.44 | modu | hello |
| 16:03.56 | Uatec | does the switch directive only work with IAX? |
| 16:04.36 | modu | I've a question for asterisk administrators |
| 16:05.02 | modu | All the docs does not seems to help me ... |
| 16:05.21 | modu | when you want to connect 100 phones on a asterisk |
| 16:05.33 | modu | you need to create account in sip.conf |
| 16:06.03 | modu | but in the extension.conf did everyone put one line for each phones ? |
| 16:06.19 | Buhntz | not for outgoing |
| 16:06.27 | Nugget | if the extensions match a particular pattern, and you've chosen your sip peer names properly, it's not necessary. |
| 16:06.37 | Uatec | if your sip accounts are just numbers 001, 002, 003, etc |
| 16:06.44 | Uatec | and are the same as your extensions |
| 16:06.46 | Uatec | you could just have |
| 16:06.50 | Nugget | you can do a single line like "exten => 7XXX,1,Dial(SIP/${EXTEN}) |
| 16:07.05 | Uatec | exten => XXX,1,Dial(SIP/${EXTEN}) |
| 16:07.06 | Uatec | :) |
| 16:07.16 | modu | ok, seems logic for me but never see that .. |
| 16:07.26 | jsmith | Uatec: Yes, it only works for IAX2 |
| 16:07.30 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 16:07.38 | Uatec | ARGH! |
| 16:07.41 | Uatec | that's a shame |
| 16:07.48 | Uatec | ty for a definitive answer |
| 16:07.49 | Buhntz | or internal calls |
| 16:08.02 | jsmith | Uatec: There's no way to do it across SIP because the SIP protocol doesn't allow for querying a remote dialplan |
| 16:08.16 | Buhntz | but you can do exten => <numberhere><extensionhere>,1,Dial... etc. |
| 16:08.58 | lsodi | I have one device somwhere in network witch tryes to register with sip server with false username and password, and asterisk prints out "... Device does not match ACL" notice. Can I block/ignore that device fore some time? or is there option to set how meny times user can try to register? |
| 16:09.04 | *** join/#asterisk tsurko (n=tsurko@78.90.100.214) |
| 16:10.06 | [TK]D-Fender | yang go look at what you're doing.... |
| 16:10.21 | [TK]D-Fender | arekm: You need to match the number that comes in, what you do AFTER is up to you. |
| 16:10.28 | *** join/#asterisk superpop02 (n=ozverenm@se167-1-82-242-148-65.fbx.proxad.net) |
| 16:10.34 | superpop02 | hello all |
| 16:11.06 | Buhntz | lsodi: you can block the ip with iptables, so you're asterisk won't bothered anymore |
| 16:11.22 | superpop02 | I have a question about asterisk internal: How asterisk handle rate adaptation between per example a ISDN and a analog call ? |
| 16:11.38 | [TK]D-Fender | superpop02: ....huh? |
| 16:11.55 | admin0 | guys .. thanks .. |
| 16:12.01 | arekm | [TK]D-Fender: I have a match now but don't know *how* to strip leading digits so next rule will check only 2 last digits in dialplan |
| 16:12.04 | superpop02 | I know V.110 is a ITU recommandation about rate adaptation on ISDN |
| 16:12.04 | lsodi | Buhntz: there is no way to control it with asterisk? |
| 16:12.13 | superpop02 | Does asterisk support V.110 ? |
| 16:12.39 | admin0 | one question.. if the incoming gateway is g723.1 the default IVR does not work ... where can I find the info to load the ivr in g723? |
| 16:12.51 | [TK]D-Fender | arekm: what "next rule"? You have already made your match... trying showing me exactly what you want to do... |
| 16:13.02 | Buhntz | lsodi i don't know anyone you can't delete extensions out of extensions.conf per automatic |
| 16:13.19 | [TK]D-Fender | admin0: If your prompts aren't in G.723.1 then you're DOA |
| 16:13.28 | admin0 | DOA = ? |
| 16:13.32 | [TK]D-Fender | admin0: And there is no legal codec for it in *. |
| 16:13.38 | Buhntz | superpop02 afaik yes |
| 16:13.39 | [TK]D-Fender | admin0: Dead On Arrival <- |
| 16:13.53 | yang | [TK]D-Fender: well, I dont know what I am doing wrong, that is why I am here |
| 16:14.06 | ai-a | we have 6 fax machines on our network connected to Asterisk via LinkSys Sipura SPA-1001 devices, With also about 40+ voice phones on the system. 2 of the fax machines just refuse to work. Get CID but transmission fails. bandwidth usage over the network is low. Connected a analogue phone shows 2 way audio works. However faxing to/from these 2 fax machines to other fax machines on the PBX or external fail most of them time. Any |
| 16:14.14 | [TK]D-Fender | yang: Look at the exten that is getting dialed. Are you calling monitor FIRST? |
| 16:14.26 | Buhntz | superpop02 there is a protocol handler |
| 16:14.28 | admin0 | isn't there any software that will allow me to make the prompts in g723.1 ? |
| 16:14.33 | yang | [TK]D-Fender: well it doesnt enable MONITOR at all... |
| 16:14.33 | superpop02 | buhntz ? some precisions , |
| 16:14.34 | admin0 | whats the codecs that all use ? |
| 16:14.35 | superpop02 | ? |
| 16:14.36 | admin0 | g729 ? |
| 16:14.44 | ai-a | admin0: sox convert them. |
| 16:14.51 | [TK]D-Fender | ai-a: because the tiniest little flaw will KILL a fax, and faxing analog over SIP is suicidal. |
| 16:15.06 | superpop02 | because in code I don't find any v.110 reference |
| 16:15.06 | [TK]D-Fender | admin0: G.729 & G.711 are the most popular. |
| 16:15.15 | ai-a | [TK]D-Fender: yes, i agree with this. however 4 fax machines seem to work perfectly. |
| 16:15.17 | [TK]D-Fender | admin0: G.711 is free, G.729 is not. |
| 16:15.33 | admin0 | where do I find the prompts for the g729 ? |
| 16:15.34 | [TK]D-Fender | ai-a: Tried swapping ATA's to see of those 2 suck? |
| 16:15.47 | [TK]D-Fender | admin0: You don't FIND them, you MAKE PROMPTS. |
| 16:15.47 | ai-a | also, what alternatives are their to fax over ip? What is the point of IP Phone network if we need analogue phone lines anyway ? |
| 16:16.04 | *** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu) |
| 16:16.11 | admin0 | mp3 -> g729 :D |
| 16:16.21 | admin0 | ok .. how do people make prompts ? |
| 16:16.32 | [TK]D-Fender | ai-a: IP for PHONES is great because you don't need special telephony equipemnt to add 100 new phones to your office. Jut the phones, and a SWITCH. |
| 16:16.38 | ai-a | admin0: microphone, computer, audio software. |
| 16:16.43 | lsodi | Buhntz: I can control with asterisk how many times server tryes to register before giving up and I cant control users/devices who are trying to register, only with iptables? |
| 16:16.51 | admin0 | ai-a, which software :) |
| 16:16.52 | [TK]D-Fender | admin0: Record them on a PC, record them with * using "show application record", etc. |
| 16:16.58 | admin0 | OK |
| 16:17.13 | arekm | [TK]D-Fender: something like http://pastebin.com/m6d164894 |
| 16:17.25 | Buhntz | lsodi its a client setting |
| 16:17.48 | yang | [TK]D-Fender: it refuses to start MONITOR http://www.pastebin.ca/702045 |
| 16:17.53 | Buhntz | lsodi you can handle how often your browser tries to reach www.google.de but google can't control how often YOU try |
| 16:17.53 | AndrewGearhart | [TK]D-Fender: so, returning to the hardware issue, we're talking 10 lines, VoIP phones (internally) and analog lines to the POTS. Other than the requirements of the PCI slot for the sangoma card, any hardware recommendations? |
| 16:17.56 | lsodi | Buhntz: ok. thank you! |
| 16:18.08 | [TK]D-Fender | arekm: exten => _36677XX,1,Goto(local,${EXTEN:5},1) |
| 16:18.11 | admin0 | i can directly connect and record in this asterisk box itself :) ? |
| 16:18.46 | arekm | [TK]D-Fender: thanks! |
| 16:18.54 | [TK]D-Fender | AndrewGearhart: 1 GIG RAM should be plenty comfortable, and enough HD for your OS, logging, redcording, etc. Basically, my ANALOG WATCH could do this.... |
| 16:19.24 | AndrewGearhart | [TK]D-Fender: lol. the catch is finding the right hardware to build your analog watch! ;-) |
| 16:19.26 | modu | if I use the "exten => XXX,1,Dial(SIP/${EXTEN})" macro, Can I use the ChanIsAvail() to see if the (internal) account exist, or is there another way ? |
| 16:19.57 | [TK]D-Fender | modu: There is no easy way.... patterns like that are messy and should be avoided.... |
| 16:20.31 | modu | [TK]D-Fender: pattern like exten... ? |
| 16:20.36 | yang | [TK]D-Fender: as you said, it will only monitor internal connections...this worked now....but how do i make it that it monitors the outside ones too? |
| 16:20.40 | *** join/#asterisk Delvar (n=Delvar@77.240.56.18) |
| 16:20.45 | modu | eh like {EXTEN} |
| 16:21.04 | *** join/#asterisk ManxPower (n=manxpowe@139.sub-75-202-162.myvzw.com) |
| 16:21.20 | [TK]D-Fender | modu: like _XXX for a pile of SIP phones. jsut hard-code them and do the job right |
| 16:21.37 | [TK]D-Fender | yang: ..... put a MONITOR IN FRONT. |
| 16:21.46 | [TK]D-Fender | yang: A call is a call is a call. |
| 16:22.04 | [TK]D-Fender | yang: You want it to record? Shove the monitor app in front! |
| 16:23.44 | yang | [TK]D-Fender: I am not following you |
| 16:23.54 | yang | In front where? |
| 16:24.40 | yang | exten => 100,1,Set(MONITOR_FILENAME=${REC_DIR}/${TIMESTAMP}-${EXTEN}-${CALLERID}-out) |
| 16:24.42 | modu | [TK]D-Fender: yes but if I have 100+ phones it should be more expensive for asterisk that a regexp like XXX |
| 16:24.43 | yang | exten => 100,2,Monitor(wav,${MONITOR_FILENAME},mb) |
| 16:24.52 | yang | the Monitor line before the SET line,m changins order? |
| 16:25.01 | [TK]D-Fender | yang: That. Isn't. Where. You. Dial. OUT |
| 16:25.15 | [TK]D-Fender | *sigh* |
| 16:25.34 | yang | :( |
| 16:25.47 | [TK]D-Fender | modu: You are going to waste a lot of processing on EVERY CALL trying to code up something to see if what you want to do is legitimate |
| 16:26.40 | [TK]D-Fender | modu: And assumes you want to treat every SIP device in the exact same way. |
| 16:26.40 | [TK]D-Fender | yang : go pastebin another failed attempt and your dialpln. |
| 16:27.22 | yang | http://www.pastebin.ca/702053 it worked, but i just dont see any files in /var/spool/asterisk/monitor |
| 16:28.10 | modu | Pasting sames lines for each phone isn't really relable, there is no way to do something like that ? a clean way |
| 16:28.12 | yang | But I think that you are trying to tell me to put the Monitor line , before the Set line |
| 16:29.03 | yang | [TK]D-Fender: And I am not quite following you |
| 16:29.26 | *** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu) |
| 16:29.28 | *** join/#asterisk ramindia (n=ramindia@202.63.96.9) |
| 16:29.43 | [TK]D-Fender | yang: Look at your call to monitor. you have a "/" in front. thats an ABSOLUTE PATH.... |
| 16:29.59 | ramindia | can some one tell me.. how can unlock sunrocket SPA-2102-R, any iirc channel for this |
| 16:30.07 | [TK]D-Fender | modu: its as reliable as the person doing the job. |
| 16:30.37 | [TK]D-Fender | ramindia: Nope. Go check out www.voxilla.com 's forums and say a prayer |
| 16:30.51 | yang | [TK]D-Fender: I cannot follow you well, becouse I am not a coder...please keep it simple |
| 16:31.17 | yang | [TK]D-Fender: Which line do I have to change and to what? |
| 16:31.26 | *** part/#asterisk ramindia (n=ramindia@202.63.96.9) |
| 16:31.49 | [TK]D-Fender | yang: Executing Monitor("SIP/600-b5d063f8", "wav|/20070918-181943-100-"Jan Prunk" <600>-out|mb") in new stack <- you put a "/" in front of 20070918........ it isn't GOING into /var/spool..... and so on because you gave monitor an ABSOLUTE PATH for where to put the recording. |
| 16:32.07 | yang | ah |
| 16:32.09 | yang | damn |
| 16:32.19 | [TK]D-Fender | yang: Go caffeinate. Now. |
| 16:33.20 | Buhntz | hehe |
| 16:33.59 | hmmhesays | grand idea |
| 16:34.10 | *** join/#asterisk [T]ank (n=ckwall@206.71.78.172) |
| 16:34.16 | yang | [TK]D-Fender: I appologise about my non-coding skills - So I need to change exten => 600,2,Monitor(wav,${MONITOR_FILENAME},mb) to exten => 600,2,Monitor(wav,${/var/spool/asterisk/monitor/MONITOR_FILENAME},mb) |
| 16:34.26 | *** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu) |
| 16:34.34 | [T]ank | what does this mean when I am dialing an 800 number from a polycom 501 phone? PROGRESS with cause code 127 received |
| 16:34.45 | *** join/#asterisk Delvar (n=Delvar@77.240.56.18) |
| 16:35.08 | [T]ank | the linksys phones get it as well, but the call still goes through. When dialing from the polycom it hangs up |
| 16:36.02 | [TK]D-Fender | yang if you jsut removed your leading "/" it would go into the normal folder ..... |
| 16:37.06 | yang | [TK]D-Fender: but there is no / set in extensions.conf , why are you joking ? |
| 16:37.49 | yang | or did you mean this line ${REC_DIR}/ |
| 16:38.02 | [TK]D-Fender | [TK]D-Fender>yang: Executing Monitor("SIP/600-b5d063f8", "wav|/20070918-181943-100-"Jan Prunk" <600>-out|mb") in new stack <- you put a "/" in front of 20070918........ it isn't GOING into /var/spool..... and so on because you gave monitor an ABSOLUTE PATH for where to put the recording. |
| 16:38.03 | yang | to replace with ${REC_DIR} |
| 16:38.06 | dlynes | I'm just curious....I've got a sip peer set to using a certain dialplan context, and looking at sip debug, it seems to be looking in that context, but then it dials out on a totally different context |
| 16:38.12 | [TK]D-Fender | yang: See this? This is the CLI output of your call attempt |
| 16:38.13 | dlynes | What could be causing this? |
| 16:38.23 | dlynes | A corrupted dialplan, or something? |
| 16:38.31 | [TK]D-Fender | yang: see the "wav|/2007"? thats YOU! |
| 16:38.39 | yang | [TK]D-Fender: yeah but something places it to / directory instead of default |
| 16:38.51 | [TK]D-Fender | yang: * did not invent that "/" and shove it in there. |
| 16:39.06 | [TK]D-Fender | yang: And that something is YOUR extension.conf. |
| 16:39.14 | [TK]D-Fender | yang: Go fix your typos! |
| 16:39.27 | [TK]D-Fender | ok, lunch time, back in a few... |
| 16:39.39 | yang | [TK]D-Fender: well, ok, it was my friends line...what do I know about the coding...:( |
| 16:42.46 | yang | [TK]D-Fender: well I give up .... I removed the / from extensions.conf and its still getting there |
| 16:43.11 | yang | exten => 600,1,Set(MONITOR_FILENAME=${REC_DIR}${TIMESTAMP}-${EXTEN}-${CALLERID}-out) |
| 16:43.14 | yang | exten => 600,2,Monitor(wav,${MONITOR_FILENAME},mb) |
| 16:44.16 | yang | I am probably the hardest case that appeared on this channel |
| 16:45.51 | thewiizle | $calleridnum |
| 16:46.01 | thewiizle | doesnt work in 1.4 is that correct? |
| 16:46.16 | *** join/#asterisk Delvar (n=Delvar@77.240.56.18) |
| 16:47.03 | *** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu) |
| 16:48.30 | ManxPower | thewiizle: no, it is not. |
| 16:48.34 | *** join/#asterisk saftsack (n=saftsack@pD9E07EE3.dip.t-dialin.net) |
| 16:48.41 | ManxPower | thewiizle: you did not read the upgrade.txt for 1.2 and 1.4, did you. |
| 16:48.48 | *** part/#asterisk thx2000 (n=evan@netblock-208-127-150-56.dslextreme.com) |
| 16:48.59 | thewiizle | yeh i did |
| 16:49.05 | thewiizle | and i didnt upgrade |
| 16:49.07 | thewiizle | fresh install |
| 16:49.15 | ManxPower | then you saw the info about CALLERIDNUM being removed and what it is replaced with. |
| 16:49.18 | thewiizle | i can see they are depreciated, i just cant get my head around the replacement |
| 16:49.25 | |NexT| | ${CALLERID(num)} |
| 16:49.35 | ManxPower | thewiizle: the extensions.conf.sample was not helpful? |
| 16:49.59 | ManxPower | |NexT|: Build a man a fire and keep him warm for the night, set a man on fire and keep him warm the rest of his life. |
| 16:50.12 | ManxPower | We give free support -- at least make them work for it. |
| 16:50.12 | thewiizle | no |
| 16:50.52 | |NexT| | ManxPower, I need help, my problem is this: |
| 16:51.06 | |NexT| | WARNING[1806]: rtp.c:2157 ast_rtp_senddigit_begin: Don't know how to represent 'f' |
| 16:51.28 | |NexT| | Google does not helpme ;-( |
| 16:51.37 | *** join/#asterisk SgtDitt (n=SgtDitt@63.251.157.172) |
| 16:51.56 | *** part/#asterisk SgtDitt (n=SgtDitt@63.251.157.172) |
| 16:51.57 | |NexT| | I use Asterisk 1.4, zaptel an libpri form SVN |
| 16:52.04 | *** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu) |
| 16:52.30 | *** join/#asterisk mocker (n=user@198.247.173.227) |
| 16:52.31 | [TK]D-Fender | yang : thats 600!! exten => 600,1,Set(MONITOR_FILENAME=${REC_DIR}${TIMESTAMP}-${EXTEN}-${CALLERID}-out) |
| 16:52.31 | |NexT| | all dtmf are in auto |
| 16:52.38 | ManxPower | |NexT|: your sip device is sending the DTMF digit "f" (there is no such DTMF digit) and asterisk does not know how to deal with it. |
| 16:53.09 | [TK]D-Fender | yang your problem is wth the monito that gets called for Spawn extension (from-local-users, 100, 3) |
| 16:53.21 | mocker | Guh, /me sends a book long message to asterisk-users |
| 16:53.22 | |NexT| | yes, but this problem is in tdm in call |
| 16:54.02 | |NexT| | I recive a call, aand speak normally, but in aleatory moment, asterisk show this message |
| 16:54.51 | ManxPower | |NexT|: I hate to break it to you, but TDM calls don't use RTP. One leg of that call MUST be SIP for you to get that message. |
| 16:54.57 | thewiizle | riiiight i get it i get it |
| 16:54.59 | arekm | [TK]D-Fender: one more problem - http://pastebin.com/m614ca3e3 |
| 16:55.17 | ManxPower | |NexT|: now, what actual SIP device are you using? |
| 16:55.25 | yang | Brrr |
| 16:55.48 | thewiizle | set($CALLERID(num)=$EXTEN) |
| 16:55.50 | thewiizle | would that work |
| 16:55.58 | |NexT| | ManxPower: I Know, but only the problem is reproduced when i recive the call form TDM |
| 16:56.09 | ManxPower | thewiizle: that would set the callerid number to be the same as the currently executing exten => line |
| 16:56.28 | thewiizle | Which is correct in theory |
| 16:56.38 | thewiizle | eg, i call from 101 my CallerId = 101 |
| 16:56.45 | ManxPower | |NexT|: chances are if you are not using TDM the RTP may be going direct between the two end points. |
| 16:56.45 | |NexT| | PAP2 --> Asterisk 1.4 --> Asterisk 1.4 with TE420 --> Normal Phone |
| 16:56.54 | ManxPower | put canreinvite=no in each device section of sip.conf |
| 16:57.19 | ManxPower | thewiizle: that is not correct. |
| 16:57.21 | *** join/#asterisk fugitivo (n=ajf@201-212-144-95.cab.prima.net.ar) |
| 16:57.24 | |NexT| | but the problem is in the middle of the conversation, no in hte negotiation |
| 16:57.28 | fugitivo | hello |
| 16:57.45 | yang | [TK]D-Fender: So do I have to remove context=from-local-user line from sip.conf for user 600 ? |
| 16:57.50 | *** join/#asterisk tripps (n=ss@66.60.235.100) |
| 16:57.50 | fugitivo | is any way to "destroy" a global variable in 1.2? |
| 16:57.51 | ManxPower | thewiizle: ${EXTEN} is the DIALED number. So it will set your callerid to be the same as the currently executing exten => line, which is the number you dialed. |
| 16:58.02 | fugitivo | setting the variable to "" doesn't destroy it |
| 16:58.22 | ManxPower | fugitivo: how do you know that? |
| 16:58.38 | [TK]D-Fender | arekm: watch out for INCLUDE prioritization. You have put 2 possible matches into one area. And you wonder why they are getting mixed up? |
| 16:58.42 | ManxPower | well, technically setting a variable to "" sets the variable to "", set it to |
| 16:59.03 | [TK]D-Fender | yang: Make a brand new call attempt and pastebin it. Then pastebin your whole dialplan again. |
| 16:59.22 | ManxPower | i.e. Set(FNORD=) rather than Set(FNORD="") since the second one sets the variable FNORD to contain two double quotes |
| 16:59.28 | yang | [TK]D-Fender: i see this line == Spawn extension (from-local-users, 100, 3) exited non-zero on 'SIP/600-b5d063f8'....but I dont know the meaning of it.. |
| 16:59.31 | fugitivo | ManxPower: I'm using AgentMonitorOutgoing(c) which looks for a variable called AGENTBYCALLERID_xxx |
| 16:59.31 | arekm | [TK]D-Fender: so order of entries doesn't matter at all? |
| 16:59.56 | thewiizle | ah shite |
| 16:59.59 | yang | [TK]D-Fender: I dont know where does that 100, 3 come from for user 600 |
| 17:00.01 | ManxPower | arekm: order matters very, very little. |
| 17:00.03 | fugitivo | ManxPower: If I set that variable to nothing, AgentMonitorOutgoing keeps finding it |
| 17:00.09 | [TK]D-Fender | yang: Means you keep talking about EXTEN => 600......and youare not even DIALING that exten. |
| 17:00.17 | arekm | ManxPower: ok, rewriting |
| 17:00.20 | [TK]D-Fender | yang: You were in 100 <--------- |
| 17:00.22 | ManxPower | yang: that is extension 100, priority 3 in context from-local-users |
| 17:00.30 | tripps | in my old asterisk box, my sip trunk peers showed the latency figure in the status column during sip show peers command. in my new box it simply displays "Unmonitored" - what setting must I change to get the latency number back? |
| 17:00.35 | fugitivo | ManxPower: I know that "" sets the variable to "", but I don't find a function to destroy the variable |
| 17:00.45 | ManxPower | fugitivo: You can't test the existance for a variable. |
| 17:00.49 | jwh | tripps: qualify=yes as per sample configs |
| 17:01.04 | tripps | jwh: thanks - I'll check that out |
| 17:01.18 | jwh | np |
| 17:01.22 | yang | [TK]D-Fender: No I am calling with username 600 ! |
| 17:01.40 | ManxPower | Set(AGENTBYCALLERID_xxx=) |
| 17:01.53 | [TK]D-Fender | yang it doesnt' matter where you're calling FROM, it matters where you're calling TO. |
| 17:01.56 | ManxPower | yang: username and extension are two totally different things |
| 17:02.17 | [TK]D-Fender | yang: You are dialing an EXTENSION. That extension is 100. Thet is what's oging to get executed. |
| 17:02.38 | ManxPower | [TK]D-Fender: do people seem unusually dense today? |
| 17:02.48 | *** join/#asterisk Delvar (n=Delvar@77.240.56.18) |
| 17:03.06 | [TK]D-Fender | yang: exten = 600,1,...... is not a set of rules applied when your SIP/600 places ANY CALL, its when a call call comes in DIALING that number! |
| 17:03.14 | yang | That is correct I was calling 100 |
| 17:03.15 | yang | from 600 |
| 17:03.53 | [TK]D-Fender | yang: Forget "from 600". That doesn't mean ANYTHING. You keep talking about an EXTENSION numbered "600" which is IRRELEVANT. |
| 17:03.57 | [TK]D-Fender | ManxPower: Agreed |
| 17:04.50 | yang | [TK]D-Fender: Ok I will make another call |
| 17:04.54 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 17:04.54 | yang | and paste the log |
| 17:05.05 | fugitivo | ManxPower: setting the variable to nothing doesn't work, AgentMonitorOutgoing still find the variable |
| 17:06.01 | yang | [TK]D-Fender: If I call any other number than 100 I dont get my monitor started at all |
| 17:06.23 | [TK]D-Fender | yang: I'm waiting for your pastebin's...... |
| 17:06.25 | *** join/#asterisk errr_ (n=errr@fedora/errr) |
| 17:06.29 | fugitivo | ManxPower: I'm wondering why using AgentCallbackLogin from manager doesn't set the global variable AGENTBYCALLERID_xxx like the cmd does |
| 17:06.51 | lsodi | anyone here using Elion broadband internet connection? |
| 17:07.19 | [TK]D-Fender | fugitivo: Because as its coming from the manager... THERE IS NO CALLERID! |
| 17:07.43 | [TK]D-Fender | Whee!!! |
| 17:08.20 | yang | http://pastebin.ca/702102 |
| 17:08.42 | fugitivo | [TK]D-Fender: shouldn't the variable be set with the Exten parameter? |
| 17:09.22 | [TK]D-Fender | yang: First you still have a stupid "/" in front of your filename, and 2nd you call is never getting ANSWERED, so there is nothing to record! |
| 17:09.49 | yang | [TK]D-Fender: I dont know how to get rid of the stupid / string in front... |
| 17:09.59 | [TK]D-Fender | fugitivo: By callerID. As in "gets its from callerid", as in "If you don't HAVE a callerid, WTF are you expecting?" |
| 17:10.19 | [TK]D-Fender | yang: -- Executing Set("SIP/600-b5d063f8", "MONITOR_FILENAME=/20070918-190721-100-"Jan Prunk" <600>-out") in new stack <- look at your set......... |
| 17:10.53 | yang | [TK]D-Fender: And why is it trying to record only internal calls, not international? you said earlier that a call is a call...but seems that asterisk knows which are internal |
| 17:11.05 | fugitivo | [TK]D-Fender: i believe the variable is called like that for later use, i think the name doesn't means from where it gets its value from at login time |
| 17:11.07 | *** join/#asterisk magic_hat (n=geoffdou@h-74-2-87-16.chcgilgm.covad.net) |
| 17:11.30 | magic_hat | hey all. anyone know if it's possible to set up conference calling without using meetme? |
| 17:11.31 | [TK]D-Fender | yang: Get your head out of your ass and focus on your obvious bugs. |
| 17:11.47 | thewiizle | anyone got astcc going on 1.4? |
| 17:11.47 | [TK]D-Fender | fugitivo: I dunno.... |
| 17:12.00 | *** join/#asterisk t3rror (n=t3rror@gateway.sscgp.com) |
| 17:12.02 | [TK]D-Fender | magic_hat: app_conference . Go look it up |
| 17:12.05 | fugitivo | [TK]D-Fender: that's nonsense, you can't login an agent from manager and then use AgentMonitorOutgoing because you don't have that variable set |
| 17:12.08 | [TK]D-Fender | yang: apstebin yuor dialplan. |
| 17:12.27 | [TK]D-Fender | fugitivo: I may have missed something in there... |
| 17:12.30 | *** part/#asterisk [T]ank (n=ckwall@206.71.78.172) |
| 17:12.39 | *** join/#asterisk teknoprep (n=teknopre@unaffiliated/teknoprep) |
| 17:12.43 | *** join/#asterisk |omni| (n=rob@c-67-185-70-220.hsd1.wa.comcast.net) |
| 17:12.44 | [TK]D-Fender | blarg... complete left-right desync in my typing today... |
| 17:12.55 | teknoprep | how do i setup keepalive for a sip connection to my VoIP provider? |
| 17:13.04 | teknoprep | i have sip working fine behind nat.. but only for about 10 min |
| 17:13.07 | teknoprep | then it goes dead |
| 17:13.15 | yang | [TK]D-Fender: in which file is the dialplan, you mean sip.conf....damnit you got me all confused with these strings |
| 17:13.17 | fugitivo | well, if it's a bug i suppose nobody is going to fix it for 1.2 :) |
| 17:13.23 | teknoprep | for inbound calls only |
| 17:13.32 | [TK]D-Fender | yang: extensions.conf <------------- |
| 17:13.58 | [TK]D-Fender | teknoprep: You don't need a keep alive to your provider........ |
| 17:14.03 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 17:14.16 | [TK]D-Fender | teknoprep: you need your routings setup properly. |
| 17:14.17 | [TK]D-Fender | ~sinap |
| 17:14.27 | [TK]D-Fender | ~sipnat |
| 17:14.28 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 17:14.31 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^ |
| 17:14.41 | magic_hat | TKD: eay to install? |
| 17:14.42 | [hC] | any of you guys done any iax2 packet analysis with wireshark (ethereal)? Im getting audio frames that sound choppy, and wireshark comes back claiming incorrect checksum on a bunch of them |
| 17:14.45 | magic_hat | easy? |
| 17:14.51 | [hC] | not sure what would be screwing up checksums, though. |
| 17:15.54 | *** join/#asterisk Ebola (i=ebola@goatse.co.uk) |
| 17:16.08 | yang | [TK]D-Fender: if you help me solve this, I will send you some paypal money for the beer :) http://pastebin.ca/702110 |
| 17:16.21 | [TK]D-Fender | magic_hat: Never tried it personally... MeetMe fworks fine for me. |
| 17:17.05 | [TK]D-Fender | yang: exten => 100,1,Set(MONITOR_FILENAME=${REC_DIR}/${TIMESTAMP}-${EXTEN}-${CALLERID}-out) |
| 17:17.12 | yang | [TK]D-Fender: I removed / from the first line in 600 exten |
| 17:17.15 | fugitivo | well, if anyone knows how to destroy a variable so AgentMonitorOutgoing can't find it anymore, that'll be enough for me |
| 17:17.52 | [TK]D-Fender | yang: I see it. You are referencing what SHOULD be a CONSTANT set under [globals] for ${REC_DIR} that you DID NOT SET |
| 17:18.03 | ManxPower | fugitivo: AgentMonitorOutgoing is an Asterisk app? |
| 17:18.14 | [TK]D-Fender | yang: because that value is null, it CONTINUES with the "/" before the timestamp. |
| 17:18.30 | yang | [TK]D-Fender: so tell me which line do I got to change, please |
| 17:18.35 | [TK]D-Fender | yang: You tried yanking someone elses code in pieces at it was NOT adapted to your dialplan. |
| 17:18.49 | [TK]D-Fender | yang: Where do you want them going? |
| 17:19.03 | yang | [TK]D-Fender: yeah into /var/spool/asterisk/monitor |
| 17:19.09 | fugitivo | ManxPower: yes, it's a command, i suppose it's deprecated in 1.4 but i still need to use it http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+AgentMonitorOutgoing |
| 17:19.29 | ManxPower | fugitivo: Asterisk will happily work if a variable does not exist, for example Noop(HAPPYVAR is ${HAPPYVAR}) and HAPPYVAR does not exist, asterisk will just continue on and act like the variable exists, but is empty. |
| 17:19.40 | yang | [TK]D-Fender: it was a string made by a freind of mine, but it only gave me troubles |
| 17:20.45 | *** join/#asterisk Delvar (n=Delvar@77.240.56.18) |
| 17:20.49 | yang | [TK]D-Fender: its hard to understand these strings if you arent a born coder |
| 17:21.25 | ManxPower | [TK]D-Fender: Asterisk really should complain if you try to reference a non-existent variable. |
| 17:25.46 | fugitivo | ManxPower: the problem AgentMonitorOutgoing is that it checks the existance of the variable and not it's content, I need to destroy the variable like when logging out an agent with AgentCallbackLogin |
| 17:25.46 | fugitivo | so AgentMonitorOutgoing will return false |
| 17:25.46 | ManxPower | fugitivo: I don't think you can. |
| 17:25.46 | *** join/#asterisk holiday_42 (n=no@spike.wcta.net) |
| 17:25.46 | fugitivo | oh that sucks |
| 17:25.46 | ManxPower | fugitivo: In asterisk there is not supposed to be any difference in operation between a variable that does not exist and a variable that has no value. |
| 17:25.47 | ManxPower | fugitivo: show me your Set line where you try to clear the contents of the variable |
| 17:25.47 | Corydon76-dig | fugitivo: you cannot destroy a variable from the dialplan. You can only set it to blank |
| 17:25.47 | Corydon76-dig | If you want something like that, use the ASTDB |
| 17:25.54 | [TK]D-Fender | ManxPower: And * dialplans shold really be done in a FULL OOP programming language with explicit datat types, syntax checking, and a built in thesaurus! |
| 17:26.04 | ManxPower | Corydon76-dig: he is having a problem with AgentMonitorOutgoing picking up old variables |
| 17:26.14 | *** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu) |
| 17:26.24 | Corydon76-dig | Then set them to blank |
| 17:26.30 | ManxPower | [TK]D-Fender: yeah, but that is much more complicated than 4 lines of code to throw an error when a variable does not exist. |
| 17:26.36 | ManxPower | Corydon76-dig: that is what I've been telling him. |
| 17:26.37 | yang | [TK]D-Fender: have you figoured the right extensions.conf strings? |
| 17:26.43 | Corydon76-dig | There is no difference from the dialplan between a nonexistant and a blank variable |
| 17:26.56 | ManxPower | then fugitivo must be using Set wrong. |
| 17:27.01 | [TK]D-Fender | yang: Unfortunately for you * was made so that you can do whatever you want with it and is susceptable to user error all over the place. There are things you need to learn to do things right or you'll be looking for help constantly. |
| 17:27.05 | Corydon76-dig | Set(foo=) |
| 17:27.10 | ManxPower | fugitivo: SHOW us the Set line where you try to clear the contents of the variable. |
| 17:27.15 | [TK]D-Fender | yang which usually leads to : |
| 17:27.18 | [TK]D-Fender | ~hafc |
| 17:27.19 | jbot | well, hafc is hire a freaking consultant. Look, if you're having difficulty understanding what you're doing and need a solution soon, you will be far better off hiring a competent consultant than continuing to pull your hair out. |
| 17:27.24 | [TK]D-Fender | :) |
| 17:28.23 | fugitivo | q |
| 17:28.25 | fugitivo | sorry |
| 17:28.30 | fugitivo | ManxPower: exten => 2552,1,SetVar(AGENTBYCALLERID_305=) |
| 17:28.46 | yang | [TK]D-Fender: instead of my friend helping me with original astrerisk config files, he uploaded some of his you know, and we let the things run |
| 17:28.56 | ManxPower | fugitivo: that should cause the AgentOutgoingMonitoring to not see the variable. |
| 17:28.58 | [TK]D-Fender | fugitivo: SetVar = deprecated in 1.2, gone in 1.4 |
| 17:29.10 | ManxPower | and he is using 1.4, I think. |
| 17:29.10 | yang | [TK]D-Fender: so is there a way that you could addopt extensions.conf to the right plan? |
| 17:29.15 | fugitivo | i'm using 1.2 |
| 17:29.30 | ManxPower | so much for that idea. |
| 17:29.32 | [TK]D-Fender | yang: Thats like taking the transmission out of your 57' Chevy and dropping it into a Hummer.... don't get your hopes up... |
| 17:29.53 | fugitivo | AgentMonitorOutgoing keeps seeing the variable |
| 17:29.55 | [TK]D-Fender | yang: Yes, of course I could completely build your entire setup for you. Thats what consulting is for :) |
| 17:29.58 | lsodi | nat and asterisk, in work I'm sitting behind nat, sip client connects to asterisk and I can make calls and recive calls, in home I'm behind nat I can make calls always and sometimes recive calls |
| 17:30.09 | [TK]D-Fender | lsodi: .... |
| 17:30.11 | [TK]D-Fender | ~sipnat |
| 17:30.12 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 17:30.13 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 17:30.16 | yang | [TK]D-Fender: you dont work for beer I suppose :)? |
| 17:30.24 | lsodi | asterisk cli prints out -- Called ..and ..nr |
| 17:30.24 | [TK]D-Fender | yang: Nop, don't drink! |
| 17:30.40 | lsodi | I have looked those but no help |
| 17:30.48 | yang | [TK]D-Fender: Beer or coffe or Soda if you like that better |
| 17:31.04 | *** join/#asterisk Tili (n=tili@cm48.gamma244.maxonline.com.sg) |
| 17:31.09 | [TK]D-Fender | lsodi: Pastebin your CLI output with SIP debug enabled as well as your sip.conf masking only passwords |
| 17:31.29 | yang | [TK]D-Fender: For me it would be much easier to navigate through a working configuration, so I could have a good look at things |
| 17:31.46 | ManxPower | yang: you would think so, but that is not normally true. |
| 17:31.57 | fugitivo | Corydon76-dig: my problem is that using Agent login from manager doesn't set the global variable AGENTBYCALLERID, then I need to set it and destroy it manually |
| 17:31.58 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 17:32.15 | fugitivo | in 1.2 |
| 17:32.25 | [TK]D-Fender | yang: Thats the catch.... see since * is incedibly personal, the odds of you being able to find something you can 100% rip out... in PIECES is pretty low. It takes some understandaing about whats actually going on. |
| 17:33.42 | yang | [TK]D-Fender: so you are saying that extensions.conf should be totally re-made |
| 17:34.54 | [TK]D-Fender | yang: I'm saying you should learn how * variables, constants, the dialplan and everythihng else works... or pay someone to set it up for you. |
| 17:35.13 | [TK]D-Fender | yang: you know there is even a ... |
| 17:35.15 | [TK]D-Fender | ~book |
| 17:35.15 | jbot | [book] Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 or Temporarily at http://www.aocomputing.net/AsteriskTFOT.pdf |
| 17:35.17 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^ |
| 17:35.25 | [TK]D-Fender | yang : and more :p |
| 17:35.46 | lsodi | http://pastebin.com/d68282677 output from CLI with sip debug |
| 17:35.53 | arekm | ManxPower, [TK]D-Fender: I'm still in a forest. Now I have: http://pastebin.com/ma8bcf3d but with this dialling 90 will never ge met to _XX,2,Dial(${TRUNK_ALC}/7${EXTEN}) |
| 17:36.12 | arekm | some Goto(2) is needed or other idea? |
| 17:36.45 | [TK]D-Fender | arekm: Go read up on how contexts included get prioritized on the WIKI.... |
| 17:37.04 | *** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu) |
| 17:37.04 | [TK]D-Fender | ~wikis |
| 17:37.05 | jbot | from memory, wikis is http://www.voip-info.org |
| 17:37.33 | ManxPower | arekm: you never want to split priorities and pattern matches across contexts. |
| 17:37.34 | *** join/#asterisk Delvar (n=Delvar@77.240.56.18) [NETSPLIT VICTIM] |
| 17:37.34 | *** join/#asterisk Ebola (i=ebola@goatse.co.uk) [NETSPLIT VICTIM] |
| 17:37.34 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 17:37.34 | *** join/#asterisk Buhntz (n=bytewalk@port-212-202-170-97.dynamic.qsc.de) [NETSPLIT VICTIM] |
| 17:37.35 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) [NETSPLIT VICTIM] |
| 17:37.35 | *** join/#asterisk famicon (i=redz@c51447ddc.cable.wanadoo.nl) [NETSPLIT VICTIM] |
| 17:37.35 | *** join/#asterisk droops (n=droops@adsl-074-245-001-031.sip.jan.bellsouth.net) [NETSPLIT VICTIM] |
| 17:37.35 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 17:37.35 | *** join/#asterisk alexpe (n=alex@cev75-1-81-57-14-91.fbx.proxad.net) [NETSPLIT VICTIM] |
| 17:37.35 | *** join/#asterisk arguile (n=arguile@KTNRON06-1242488957.sdsl.bell.ca) [NETSPLIT VICTIM] |
| 17:37.35 | *** join/#asterisk FlyboySR22 (n=rsears@206.251.251.2) [NETSPLIT VICTIM] |
| 17:37.35 | *** join/#asterisk enioh (n=enioreh@core.kahmm.net) [NETSPLIT VICTIM] |
| 17:37.35 | *** join/#asterisk the_lalelu (n=lalelu@geek-at-work.org) [NETSPLIT VICTIM] |
| 17:37.35 | *** join/#asterisk dennis- (n=dennis@cipherpunk.org) [NETSPLIT VICTIM] |
| 17:37.36 | *** join/#asterisk ltd (n=z@nox.amused.net) [NETSPLIT VICTIM] |
| 17:37.36 | *** join/#asterisk CCFL_Man2 (i=5f2893e9@pool-71-241-87-104.scr.east.verizon.net) [NETSPLIT VICTIM] |
| 17:37.36 | *** join/#asterisk ido (n=ido@unaffiliated/ido) |
| 17:38.32 | ManxPower | arekm: http://pastebin.com/m22c2d7c5 |
| 17:39.46 | ManxPower | arekm: also you are trying to send just 2 digits out when you are Dialing |
| 17:40.05 | arekm | ManxPower: the case is that any XX is valid via TRUNK_ALC but some are intercepted and redirected to sip or zap/33 |
| 17:40.26 | arekm | ManxPower: 7 + 2 digits, yes, it's local alcatel PBX |
| 17:40.36 | arekm | I've placed asterisk between telco and alcatel |
| 17:40.55 | ManxPower | just making sure you understand what you are dialing. BTW, there is no 7 digits anywhere in that pastebin |
| 17:40.57 | *** join/#asterisk Ebola (i=ebola@goatse.co.uk) |
| 17:41.05 | [TK]D-Fender | arekm: Why are you mixing those fixed extens with a pattern match that can ALSO match the same #'s? |
| 17:41.06 | ManxPower | sorry, you meant a literal 7 |
| 17:41.51 | arekm | [TK]D-Fender: all 2 digit numbers are correct and should go via TRUNK_ALC but I'm trying to intercept some and redirect into other place like SIP |
| 17:42.00 | ManxPower | [TK]D-Fender: I think he is making the classic newbie mistake of thinking that a dialplan can be simple and elegant. Poor thing will be crushed if he ever realizes that all dialplans are big, bulky, and very, very ugly. |
| 17:42.13 | [TK]D-Fender | arekm: pastebini your entire dialplan again. |
| 17:42.31 | [TK]D-Fender | ManxPower: His can be truncated very easily from what I saw. |
| 17:42.38 | arekm | http://pastebin.com/m5b489d9d |
| 17:42.54 | ManxPower | [TK]D-Fender: maybe so, but something in production is never simple |
| 17:42.58 | arekm | from-wold is the incoming context and I'm calling 3667790 for example |
| 17:43.13 | [TK]D-Fender | ManxPower: This spicific thing... sure it is. |
| 17:43.23 | [TK]D-Fender | arekm: Ok, describe where calls are coming FROM, and TO |
| 17:43.30 | [TK]D-Fender | specific* |
| 17:43.35 | ManxPower | notice how he tries to avoid duplicate exten lines |
| 17:44.22 | arekm | FROM from-wold TO 3667790 and the call should go via Dial(${TRUNK_ALC}/7${EXTEN}) |
| 17:44.36 | lsodi | and I have sip accounts in mysql database. short print from mysql http://pastebin.com/d3cc29c52 |
| 17:44.44 | [TK]D-Fender | arekm: Ok, I think I've figured it out. |
| 17:44.47 | arekm | or FROM from-world TO 3667720 and the call should go via Dial(Zap/33,30) |
| 17:46.37 | *** join/#asterisk hmmhesays (n=Neg@24-119-176-74.cpe.cableone.net) |
| 17:49.37 | [TK]D-Fender | arekm: http://pastebin.com/m1d498656 |
| 17:50.04 | [TK]D-Fender | arekm: if the incoming call exists in that other context it'll go there, otherwsie it'll pipe out the other end. |
| 17:50.15 | arekm | [TK]D-Fender: oh, thanks again |
| 17:50.50 | *** join/#asterisk Delvar (n=Delvar@77.240.56.18) |
| 17:51.03 | [TK]D-Fender | lsodi: if your remote phone is behind NAT you should have qualify=yes. |
| 17:51.15 | ManxPower | Maybe I should call verizon wireless customer service and transfer THEM to an IVR. Bastards. |
| 17:51.34 | ManxPower | [TK]D-Fender: I'm sure he already has that since he did read the docs. |
| 17:52.00 | [TK]D-Fender | ManxPower: http://pastebin.com/d3cc29c52 <- NOPE. |
| 17:52.06 | [TK]D-Fender | lsodi: ...... |
| 17:52.13 | [TK]D-Fender | ~osmosis |
| 17:52.14 | jbot | i heard osmosis is the act of beating yourself on the head repeatedly with THE BOOK, until some measure of absorption has occured ... or at least until your unconsciousness restores peace to the channel ... |
| 17:52.18 | ManxPower | [TK]D-Fender: then he is a moron since he said he read the docs |
| 17:52.29 | [TK]D-Fender | :D |
| 17:52.43 | [TK]D-Fender | I haven't used that little jewel in far too long! |
| 17:52.50 | lsodi | even with qualify it doesnot work, it too random, |
| 17:53.13 | [TK]D-Fender | lsodi: is your * behind NAT as well? |
| 17:53.34 | lsodi | no |
| 17:53.57 | [TK]D-Fender | lsodi: describe your full networking path and any forwarding, etc thats going on. |
| 17:57.28 | lsodi | asterisk listens on poet 5060 and all unprivileged ports are open, no nat, server has FQDN. in home I have dsl modem with dynamic IP, modem is nat/router, no port forwarding to client |
| 17:59.19 | [TK]D-Fender | lsodi: what SIP device? |
| 17:59.31 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 17:59.35 | lsodi | X-lite |
| 17:59.47 | *** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu) |
| 18:00.06 | [TK]D-Fender | lsodi: Ok, there is another setting you forgot in the guide, read it again . |
| 18:00.08 | [TK]D-Fender | ~sipnat |
| 18:00.09 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 18:00.19 | [TK]D-Fender | lsodi: I'll be back in a few to see if you've realized what it is. |
| 18:00.48 | hmmhesays | sipnat, funnat |
| 18:01.03 | *** join/#asterisk diclophis-work (n=jbardin@65.203.37.58) |
| 18:01.17 | diclophis-work | hello all |
| 18:01.31 | hmmhesays | theres a few ways you can deal with clients behind nat |
| 18:01.31 | diclophis-work | i am wondering how "feature keys" of the polycom phones work with asterisk |
| 18:01.45 | lsodi | in work I'm behind nat and every thing works fine |
| 18:01.53 | [TK]D-Fender | diclophis-work: Not a clear term this "feature-keys"... clarify |
| 18:01.55 | hmmhesays | xlite has a few features |
| 18:02.04 | hmmhesays | stun, statically setting the public ip |
| 18:02.05 | hmmhesays | etc |
| 18:02.17 | diclophis-work | [TK]D-Fender: for instance, this phone: http://www.polycom.com/usa/en/products/voice/desktop/soundpoint_ip/soundpoint_ip501.html |
| 18:02.17 | *** part/#asterisk fugitivo (n=ajf@201-212-144-95.cab.prima.net.ar) |
| 18:02.25 | diclophis-work | per the description has: http://www.polycom.com/usa/en/products/voice/desktop/soundpoint_ip/soundpoint_ip501.html |
| 18:02.26 | diclophis-work | er |
| 18:02.27 | diclophis-work | damnit |
| 18:02.31 | diclophis-work | Combination of 9 dedicated feature keys and 4 context-sensitive soft keys |
| 18:02.43 | diclophis-work | that says to me, that these keys are "programmable" |
| 18:02.44 | diclophis-work | somehow |
| 18:02.57 | diclophis-work | i am wondering how that is accomplished with aasterisk |
| 18:03.04 | [TK]D-Fender | diclophis-work: well they would be "transfer", "conference", "directories", "Services", etc.. |
| 18:03.08 | diclophis-work | i would imagine some sort of special extension |
| 18:03.15 | [TK]D-Fender | diclophis-work: And they are.. in a sense |
| 18:03.16 | diclophis-work | oh |
| 18:03.19 | diclophis-work | damn |
| 18:03.31 | diclophis-work | so, its not like, customizable keys |
| 18:03.39 | diclophis-work | that i can make do whatever i want |
| 18:03.47 | [TK]D-Fender | diclophis-work: as in you can have certain PHONE functions mapped to overrid the defaults which also correspnd to the key-caps that come on them by default. |
| 18:03.56 | [TK]D-Fender | diclophis-work: Indedd they are not. |
| 18:04.11 | *** join/#asterisk Delvar (n=Delvar@77.240.56.18) |
| 18:04.53 | diclophis-work | damn |
| 18:05.05 | [TK]D-Fender | diclophis-work: What exactly were you looking to do? |
| 18:05.15 | *** join/#asterisk Juggie (n=Juggie@CPE001601df17fb-CM000a73a18a20.cpe.net.cable.rogers.com) |
| 18:05.21 | diclophis-work | i want a button that can change my call routing from going to my desk phone, to going to my cell phone |
| 18:05.24 | diclophis-work | and then change it back |
| 18:05.27 | diclophis-work | with one press |
| 18:09.17 | ManxPower | diclophis-work: and I want a billion dollars |
| 18:09.23 | diclophis-work | haha |
| 18:09.26 | *** join/#asterisk dlynes_ (n=dlynes@d154-20-9-152.bchsia.telus.net) |
| 18:09.41 | admin0 | does anyone here use a2billing ? |
| 18:10.20 | admin0 | i configured it as per the documentation.. when I call in, it says playing the file in the asterisk cli, but in actual, no sound is heard |
| 18:10.39 | *** join/#asterisk elixer (i=elixer@65.207.74.18) |
| 18:12.52 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 18:12.52 | *** join/#asterisk Daejeo1 (n=chatzill@211.177.189.60) |
| 18:12.56 | [TK]D-Fender | diclophis-work: There's a single SOFT-KEY for that already... its call FORWARD <---- |
| 18:13.21 | [TK]D-Fender | diclophis-work: But unfortunately you'll have to pres it TWICE! Oh noes!!!!@!@!!@ |
| 18:13.40 | diclophis-work | forward? |
| 18:13.47 | diclophis-work | and why press it twice? |
| 18:13.48 | [TK]D-Fender | diclophis-work: Yes |
| 18:13.51 | diclophis-work | hmm |
| 18:14.02 | [TK]D-Fender | diclophis-work: Go buy the phone and learn how it works. |
| 18:14.18 | *** join/#asterisk klictel (n=klictel@atelka.info) |
| 18:14.44 | [TK]D-Fender | diclophis-work: First goes into the menu asking you where to forward calls to. This would be filled in which whatever was last filled in there. You would then jsut press it again to accept that value and *poof*, you're done |
| 18:14.54 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 18:14.56 | diclophis-work | nice |
| 18:15.12 | diclophis-work | that seems workable |
| 18:15.22 | diclophis-work | in fact my phone might all ready do that, |
| 18:15.29 | diclophis-work | i have an "autoanswer" button |
| 18:15.33 | diclophis-work | but it doesnt do anything when i push it |
| 18:15.36 | Daejeo1 | i am trying to register pap2 adapter loaded with supra firmware 2.0.9 . |
| 18:16.06 | Daejeo1 | 27673180.355523192.168.0.207203.247.211.227SIPRequest: REGISTER sip:x.x.214.224:5066 |
| 18:16.31 | Daejeo1 | 27723240.388483x.x.211.227192.168.0.207ICMPDestination unreachable (Port unreachable) |
| 18:17.22 | tripps | i'm trying to enable can reinvite feature on my sip cisco 79xx phones. the * and handsets are on a LAN behind a router/fw. nat is enabled on the handsets and in the config files for the 79xx endpoints. on some calls, however the internal party can hear the caller but the caller cannot here them. any ideas? |
| 18:18.07 | *** part/#asterisk Victor_Yure (n=aaaa@esp5.deibotoch.com.br) |
| 18:18.27 | hmmhesays | tripps, look at the o= in the reinvite, I bet it is trying to reinvite to the public ip addy's |
| 18:19.20 | tripps | hmmhesays: i'm using freepbx to config this feature - where would i find the setting? |
| 18:19.21 | *** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu) |
| 18:19.32 | tripps | hmmhesays: or is it something i should find in the logs |
| 18:20.46 | Daejeo1 | oh i got it |
| 18:20.56 | Daejeo1 | : 5066 typo |
| 18:21.07 | Daejeo1 | registered now |
| 18:21.22 | Daejeo1 | :) |
| 18:22.08 | hmmhesays | sip debug on the cli mang |
| 18:22.19 | hmmhesays | or put a hub on your network and user ethereal |
| 18:22.28 | *** join/#asterisk webtech_m33 (i=webtech-@webtech.m33access.com) |
| 18:22.40 | hmmhesays | I carry one in my service bag just for that purpose |
| 18:22.50 | webtech_m33 | asterisk1:/var/log/asterisk# asterisk -r |
| 18:22.50 | webtech_m33 | Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) |
| 18:22.57 | hmmhesays | asterisk is not running |
| 18:23.14 | *** join/#asterisk diclophis-work (n=jbardin@65.203.37.58) |
| 18:23.26 | diclophis-work | thanks for the help |
| 18:23.33 | webtech_m33 | asterisk1:/var/run/asterisk# ps aux | grep asterisk |
| 18:23.39 | webtech_m33 | root 2313 0.0 0.1 3644 700 ? S 10:06 0:00 /bin/sh /usr/sbin/safe_asterisk -U asterisk -G asterisk |
| 18:23.40 | webtech_m33 | it is .. but i moved |
| 18:23.47 | Daejeo1 | [TK]D-Fender: you are not fending anything today |
| 18:23.55 | webtech_m33 | asterisk.conf to use the /var/run/asterisk |
| 18:24.05 | webtech_m33 | i made and change rights to that folder |
| 18:24.13 | webtech_m33 | for asterisk user |
| 18:24.21 | webtech_m33 | but when i do a asterisk -r |
| 18:24.26 | [TK]D-Fender | Daejeo1: I've avoided YOUR questions, haven't I? :p |
| 18:24.32 | webtech_m33 | no worky |
| 18:24.54 | Daejeo1 | i know it was silly |
| 18:25.03 | Daejeo1 | some typo |
| 18:25.10 | Daejeo1 | :) |
| 18:25.15 | [TK]D-Fender | webtech_m33: Thats jsut ehe SCRIPT. that means its probably trying to launch asterisk and it keeps FAILING |
| 18:25.44 | [TK]D-Fender | webtech_m33: Kill safe_asterisk and start * manually to see what the error is. |
| 18:26.20 | tripps | hmmhesays: i have wireshark on a workstation here with a spanned switch port - i can renable that. i enabled sip debug on the internal peer and am looking through the messages. where should i look? i'm doing things like putting the internal peer on hold so i can look for reinvitations after music on hold, etc. |
| 18:26.37 | *** join/#asterisk pruonckk (n=mike@200.212.179.130) |
| 18:26.44 | pruonckk | hello |
| 18:27.28 | hmmhesays | well tripps, you look for the initial invites between the phone, then you look for asterisk to issue another invite to the calling party when the called party picks up the phone, it is in that invite you should look for the ip address it is directing the rtp to |
| 18:27.45 | tripps | hmmhesays: ah found the o= messages |
| 18:28.15 | hmmhesays | tripps should be part if the sip invite |
| 18:28.40 | hmmhesays | if it is reinviting with the rtp on the localnet then it should indicate that |
| 18:28.43 | webtech_m33 | asterisk -U asterisk -G asterisk -cvv |
| 18:28.47 | webtech_m33 | and it works |
| 18:28.57 | webtech_m33 | something in my startup script ? |
| 18:29.08 | hmmhesays | webtech: bash -X asterisk start |
| 18:29.36 | hmmhesays | whoops |
| 18:29.39 | hmmhesays | bash -x asterisk start |
| 18:29.52 | hmmhesays | that'll execute your script and output each line |
| 18:30.01 | hmmhesays | granted you have bash installed |
| 18:30.14 | tripps | hmmhesays: is this what i'm looking for? o=root 757 762 IN IP4 10.1.16.11 |
| 18:30.23 | webtech_m33 | ... /usr/sbin/asterisk: /usr/sbin/asterisk: cannot execute binary file |
| 18:30.28 | hmmhesays | tripps bingo |
| 18:30.30 | [TK]D-Fender | webtech_m33: I didn't say that. I'm currently suspecting that * is bombing out, not a script error. |
| 18:30.42 | [TK]D-Fender | webtech_m33: Usually happens with things like chan_zap fail to load, etc |
| 18:30.45 | hmmhesays | assuming 10.1.16.11 is the phone you are calling to? |
| 18:30.57 | tripps | it does look like internal IP - however on the current call i can hear - it could the router which remembers the previous call |
| 18:31.06 | tripps | that's the * server on the lan |
| 18:31.07 | webtech_m33 | [TK]D-Fender : it's the default config |
| 18:31.19 | [TK]D-Fender | webtech_m33: That means LESS than nothing. |
| 18:31.29 | webtech_m33 | sorry |
| 18:31.40 | [TK]D-Fender | webtech_m33: Start * manually as the appropriate user and see what happens. |
| 18:31.45 | webtech_m33 | just comp it on to a debain box |
| 18:32.04 | webtech_m33 | from source |
| 18:32.15 | tripps | hmmhesays: now i've got o=Cisco-SIPUA 27009 0 IN IP4 10.1.16.116 |
| 18:32.30 | tripps | which is the internal phone |
| 18:32.35 | hmmhesays | tripps you need to know the ip scheme of your phones mang |
| 18:33.06 | hmmhesays | [TK]D-Fender: i'm going to saliva tonight |
| 18:33.22 | *** join/#asterisk pots_line (n=bryan@66-43-34-50.misn.com) |
| 18:33.27 | pots_line | . |
| 18:33.35 | [TK]D-Fender | hmmhesays: You really don't have to give me the play-by-play for your bodily fluid flow you know.... |
| 18:33.42 | [TK]D-Fender | TMI <-------- |
| 18:33.44 | tripps | hmmhesays: i do know the ip scheme of the network - what do you mean exactly? (sorry for being dense; i catch on quicky though :)) |
| 18:33.45 | hmmhesays | LOL |
| 18:34.03 | hmmhesays | tripps explain your problem again |
| 18:35.44 | tripps | when using the canreinvite feature, it appears on calls that the internal party can hear the external party ok, but not the other way aroud. nat is enabled in * and on the phones |
| 18:36.15 | ManxPower | tripps: reinvites are not compatible with NAT |
| 18:36.16 | tripps | but it's sporadic - not every call. very hard to troubleshoot |
| 18:36.24 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 18:36.38 | tripps | ManxPower: i suppose I'm coming to that conclusion :( |
| 18:36.39 | ManxPower | tripps: set canreinvite=no |
| 18:36.46 | ManxPower | or don't use NAT. |
| 18:36.46 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-154-177-39.red.bezeqint.net) |
| 18:37.01 | *** join/#asterisk s34n (n=chatzill@ip-206-159-190-125.mvdsl.com) |
| 18:37.46 | tripps | ManxPower: i was hoping that it would solve some call quality problems we appear to be suffering. how is canreinvite separate or different than native bridging (i.e., setting up rtp media stream between endpoints)? i suppose you need one for the other for musiconhold, etc., to work? |
| 18:37.58 | *** part/#asterisk pots_line (n=bryan@66-43-34-50.misn.com) |
| 18:38.03 | ManxPower | native bridging still goes thru asterisk. |
| 18:38.31 | ManxPower | it just basically shortcuts the path between the two endpoints to run thru as little asterisk code as possible. |
| 18:39.01 | tripps | ManxPower: right i know signalling is still there obviously but i thought the rtp stream could be set up beetween the endpoints directly, kind of like call manager |
| 18:39.05 | ManxPower | reinvites set up a direct RTP stream between the two endpoints, bypassing asterisk for audio (signalling still goes thru Asterisk) |
| 18:39.05 | *** join/#asterisk jmesquita (n=jmesquit@c915230f.virtua.com.br) |
| 18:39.06 | tripps | or cilantro |
| 18:39.09 | *** join/#asterisk HCevan (i=edude@pool-72-77-226-91.tampfl.fios.verizon.net) |
| 18:39.22 | tripps | ManxPower right :) |
| 18:39.30 | ManxPower | tripps: if you are not using NAT you can have the audio go direct between the two endpoints. |
| 18:40.06 | tripps | ManxPower: do you think disabling nat and doing canreinvite and rearchitecting the network would be worthwhile, i.e., it would improve call quality issues and potential problems? |
| 18:40.18 | ManxPower | chances are if you do clever port forwarding on the client router you might be able to make NAT and reinvites work. |
| 18:40.44 | tripps | ManxPower: perhaps setting up dedicated port mappings, etc.? |
| 18:40.54 | ManxPower | tripps: might help, it really depends on many factors. Since you don't actually know WHAT and WHERE the audio quality issue are happening, anything you do will be a shot in the dark. |
| 18:40.54 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-1c04bfa2b599c2c1) |
| 18:41.52 | ManxPower | tripps: when reinvites enabled there is no pre-existing NAT translation in the router for BOTH endpoints so the call will fail. |
| 18:42.32 | tripps | ManxPower: we've been troubleshooting this for weeks, running wireshark and capturing data and streams and the whole nine yards. We're on net to SIP provider layer 2 all the way and only one layer 3 hop to them and the pstn. network is squeaky clean, bandwidth is abundant, * server isn't breathing, everything appears perfect but they still get garbled calls and cutting in and out |
| 18:43.26 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 18:43.27 | AndrewGearhart | anybody here using calling cards via asterisk to reduce the cost of long distance (and if you are, which one)? |
| 18:43.35 | jsmith | tripps: When you do the RTP analysis in Wireshark, do you see dropped packets? Out-of-order packets? Jitter? |
| 18:44.11 | ManxPower | tripps: how are you doing QoS on the link to the provider? |
| 18:44.21 | tripps | there is jitter when we do the analysis - pretty bad in some cases (using default 50ms jitter buffer in wireshark). other times it's perfect, 0 jitter, 0 out of order packets |
| 18:44.34 | ManxPower | tripps: just remember Asterisk 1.2 and earlier did NOT have an RTP jitterbuffer. |
| 18:44.47 | tripps | ManxPower: right - does 1.4 with SIP? |
| 18:45.19 | ManxPower | tripps: SIP is a signalling protocol, it does not care about jitter. Asterisk 1.4 has an RTP jitterbuffer. You may have to explicitly enable it, I don't know, I don't use 1.4 |
| 18:45.42 | ManxPower | tripps: if you are getting jitter on your calls then your QoS is not set up correctly. |
| 18:46.08 | russellb | http://thecomplex.com/photos/iax-lax.jpg |
| 18:46.42 | ManxPower | russellb: that's just twisted. |
| 18:46.45 | [TK]D-Fender | russellb: OMGZ hillarious! |
| 18:46.57 | jsmith | russellb: That's awesome, but you forgot to photoshop the pills themselves |
| 18:47.09 | russellb | heh, i didn't do it |
| 18:47.15 | russellb | someone else in digium pasted the link |
| 18:47.24 | russellb | just passing it on because i found it amusing :) |
| 18:47.32 | bkruse | russellb: hehe |
| 18:47.41 | *** join/#asterisk admin0 (n=admin@bb121-6-233-92.singnet.com.sg) |
| 18:47.56 | ManxPower | bkruse: you mean #asterisk-dev |
| 18:47.57 | admin0 | hi .. how do i find out what codec is used when the demo is played |
| 18:47.58 | tripps | ManxPower: only qos is 802.1p on the internal switch. provider says that with 100 mbps we don't need to enable it on router (managed by isp). what should we have them enable? |
| 18:48.12 | bkruse | ManxPower: It depends on what time of day it is :] |
| 18:48.18 | admin0 | if 1 call the pbx via cisco ata, it does not hear any sound .. if I use via x-lite, it plays the demo |
| 18:48.20 | [TK]D-Fender | admin0: Enable SIP debug and watch the call come in. |
| 18:48.21 | ManxPower | tripps: enable some form of QoS on the router. |
| 18:48.25 | bkruse | ManxPower: and whos talking :P |
| 18:48.40 | [TK]D-Fender | admin0: This the ATA using G.723.1 we were talking about earlier? |
| 18:48.49 | ManxPower | tripps: just show them your jitter stats |
| 18:49.23 | *** join/#asterisk famicon (i=redz@c51447ddc.cable.wanadoo.nl) |
| 18:50.20 | s34n | if 'core show applications' doesn't display Meetme, does that mean it wasn't compiled in? |
| 18:50.27 | admin0 | [TK]D-Fender, it does not actually show me the codec being used |
| 18:50.38 | ManxPower | s34n: that would be a safe assumption. You forgot to install zaptel before installing Asterisk |
| 18:50.51 | tripps | ManxPower: they will install any config on the router we want however. what should we enable? they're using cisco asa5505 routers |
| 18:50.57 | ManxPower | admin0: "sip show channels" will tell you the codec. |
| 18:51.02 | bkruse | s34n: you need a timing source, of course |
| 18:51.04 | s34n | ManxPower: zaptel is installed and running at time of compile |
| 18:51.06 | bkruse | modprobe ztdummy |
| 18:51.10 | tripps | ManxPower: they use basic ios commands, etc., for qos |
| 18:51.11 | ManxPower | tripps: Cisco has like 20 different ways to do QoS. |
| 18:51.12 | bkruse | s34n: lsmod | grep zap |
| 18:51.19 | *** join/#asterisk funxion (n=nunya@63.214.236.169) |
| 18:51.19 | [TK]D-Fender | admin0: you TELL * what codecs it can use. You should already know the answer to this. If not look at your configs, and if yuo want to see for sure, enable SIP debug and watch the call come in. |
| 18:51.22 | s34n | bkruse: it is there |
| 18:51.24 | ManxPower | I would be happy to deisgn a QoS setup for you, but it won't be cheap. |
| 18:51.25 | bkruse | ManxPower: yes, and some of them are RETARDED |
| 18:51.35 | bkruse | s34n: compiled asterisk AFTER zaptel install? |
| 18:51.38 | s34n | yes |
| 18:51.50 | bkruse | ManxPower: not retarded, but rather not friendly |
| 18:51.52 | s34n | just recompiled 5 minutes ago to make sure |
| 18:51.54 | bkruse | s34n: man menuselect |
| 18:51.58 | [TK]D-Fender | s34n: Zaptel has to be ready BEFORE * is compiled. |
| 18:51.59 | bkruse | make menuselect* |
| 18:52.10 | ManxPower | I think there was an issue in 1.4 where if you built asterisk, then built zaptel, asterisk won't see the newly installed zaptel when you try to build it again |
| 18:52.11 | webtech_m33 | I got it to start |
| 18:52.27 | webtech_m33 | some one didn't save the asterisk.conf |
| 18:52.42 | s34n | bkruse: k? |
| 18:53.05 | bkruse | s34n: Go to applications -> meetme |
| 18:53.17 | bkruse | do you see a [XXX] app_meetme or w/e? |
| 18:53.54 | s34n | bkruse: [XXX] |
| 18:54.02 | ManxPower | *grumble* I had to fire a client today. |
| 18:54.04 | s34n | shows a dependency of zaptel |
| 18:54.55 | Qwell | rerun configure |
| 18:55.02 | s34n | I did |
| 18:55.03 | bkruse | s34n: sh configure |
| 18:55.12 | s34n | I did |
| 18:55.17 | Qwell | then zaptel either isn't installed, or you have the wrong version |
| 18:56.11 | tripps | ManxPower: what do you mean by "not cheap"? it may very well be worth it for what we're doing going forward |
| 18:56.54 | s34n | zaptel 1.2.20.1 |
| 18:57.08 | Qwell | s34n: and what version of asterisk? |
| 18:57.16 | s34n | lsmod shows zaptel used by ztdummy |
| 18:57.32 | s34n | asterisk is 1.4.11 |
| 18:57.36 | ManxPower | tripps: QoS is a complex thing to set up. I don't do it for free. |
| 18:57.39 | *** join/#asterisk pots_line (n=bryan@66-43-34-50.misn.com) |
| 18:57.47 | Qwell | you need the same minor version of each... |
| 18:58.48 | s34n | Qwell: that is the latest stable of each. |
| 18:58.53 | Qwell | no it isn't |
| 18:59.17 | s34n | Qwell: uh. *blush* |
| 18:59.28 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 18:59.37 | pots_line | Polycom IP601 . . . rebooting issues . . . |
| 18:59.48 | tripps | ManxPower: that's fine - let me know your rate |
| 18:59.51 | ManxPower | pots_line: doesn't happen with my polycoms |
| 19:00.00 | *** join/#asterisk Trionnis (n=blah@65-117-172-195.dia.static.qwest.net) |
| 19:00.27 | Trionnis | anyone know if there are any known issues with "usereqphone" in 1.4.11 ? |
| 19:00.31 | ManxPower | tripps: $250/hr would be my rate for that project. |
| 19:00.33 | pots_line | 40 lines BW for BLF . . . on 1.4.11 . . . reboots when intercom paging |
| 19:00.39 | *** join/#asterisk elriah (n=null@adsl-070-148-053-190.sip.bhm.bellsouth.net) |
| 19:00.46 | ManxPower | I'm pretty sure you provider can do it faster and cheaper. |
| 19:00.55 | pots_line | the 40 lines are ip 430s |
| 19:01.05 | ManxPower | pots_line: what version of the firmware for the phones? |
| 19:01.27 | elriah | Hi all. We just picked up a customer tha thas over 500 locations, 1 phone each. We're going to migrate them to our hosted solution onthe quick. Will asterisk handle that may registrations easily? (quade core, 4 gb) |
| 19:01.27 | pots_line | 3.1.3.0151 bootrom . . . 2.1.2.0078 sip |
| 19:01.40 | pots_line | docs say they are compat |
| 19:01.44 | elriah | Asterisk 1.2.24 |
| 19:03.09 | tripps | ManxPower: what do you think qos has to do with availability of bandwidth and other factors? i.e., we've got a network with huge amounts of available bandwidth and layer 2 most of the way to pstn with massive core router, etc. |
| 19:03.42 | ManxPower | tripps: I don't care WHAT you have, if you have large amounts of jitter something is seriously wrong. |
| 19:04.07 | ManxPower | jitter w/o jitter buffer = bad call quality |
| 19:04.18 | ManxPower | if you were not getting jitter I would not recommend QoS. |
| 19:04.45 | ManxPower | The first thing you need to do is figure out WHERE jitter is happening. |
| 19:05.22 | ManxPower | I suggest doing a traceroute from the LAN the phones are on to the SIP provider's SIP gateway, then do a "ping -c 100 ip.of.each.hop" |
| 19:05.33 | ManxPower | then you can start to get an idea of where the jitter is happening. |
| 19:05.36 | elriah | mtr |
| 19:05.58 | tripps | ManxPower: what does the latency indicated in the cli on the peers even with the * and the peers on the same LAN indicate? we're getting 100ms on each peer internally. does that indicate anything? |
| 19:05.59 | jsmith | Yeah, mtr is the bomb |
| 19:06.06 | ManxPower | ping is NOT the best tool, but it is a good start. |
| 19:06.12 | jsmith | tripps: Yeah, that's not good. |
| 19:06.19 | ManxPower | tripps: the sip show peers latency is NOT network latency. |
| 19:07.12 | ManxPower | the latency shown my "sip show peers" shows the latency for a response to a SIP OPTIONS packet. Most of the latency you see there is the phone responding, not network latency. |
| 19:07.34 | jsmith | ManxPower: Still, if he's getting 100ms response to a SIP options packet on the same lan, that's awfully slow |
| 19:07.45 | ManxPower | jsmith: it is terrible slow. |
| 19:08.11 | *** join/#asterisk RipeR-81 (n=ircap8@190.53.33.3) |
| 19:08.14 | *** join/#asterisk mitcheloc (n=mitchel@adsl-68-120-230-95.dsl.irvnca.pacbell.net) |
| 19:08.14 | tripps | jsmith & ManxPower: only on 79xx with SIP load, ata and linksys phones get single digits. weird |
| 19:08.17 | ManxPower | personally I think enabling the RTP jitter buffer in 1.4 would be the first step, chances are that will help |
| 19:08.32 | jsmith | tripps: Check the CDP and VLAN settings on your phone. |
| 19:08.33 | s34n | bkruse, Qwell, et al: thx. I feel dummer now, but at least it works. |
| 19:08.49 | Trionnis | irony at its finest |
| 19:08.59 | jsmith | What? CDP causing problems? |
| 19:09.16 | tripps | ManxPower: sip gateway doesn't respond to icmp so I only get all the hops until that point. i'll still try the ping you mentioned |
| 19:09.19 | Trionnis | "I feel dummer now" |
| 19:10.29 | ManxPower | jsmith: at least polycom phones can stop working for a few seconds in some situations where the phone is confgured for CDP, but you are not using CDP on your network. |
| 19:10.51 | ManxPower | If you ARE using CDP on your network it can be totally awesome. |
| 19:10.57 | pots_line | ManxPower: What kinds of things can cause Polycoms to reboot |
| 19:11.16 | ManxPower | we use CDP to make the polycom phones find their voice VLAN |
| 19:11.25 | ManxPower | pots_line: I've never had a polycom reboot. |
| 19:11.31 | pots_line | never? |
| 19:11.31 | jsmith | pots_line: Bad firmware. |
| 19:11.38 | ManxPower | nope., never. |
| 19:11.44 | pots_line | current |
| 19:11.45 | jsmith | pots_line: 9 times out of 10 the phone just needs a newer firmware |
| 19:11.48 | pots_line | what do you recommend |
| 19:11.58 | ManxPower | we use 2.1.mumble, IIRC |
| 19:12.08 | jsmith | If it's current, then call your Polycom rep |
| 19:12.15 | pots_line | 2.1.2 . . . . |
| 19:12.20 | pots_line | is what we are using |
| 19:12.22 | RipeR-81 | anybody now how to use the chanspy option in asterisk ? |
| 19:12.23 | [TK]D-Fender | pots_line: upgrade your firmware |
| 19:12.31 | pots_line | k |
| 19:12.44 | [TK]D-Fender | RipeR-81: "show application chanspy" <--------- |
| 19:12.46 | [TK]D-Fender | ] |
| 19:13.00 | jsmith | pots_line: I think 2.2 is out now |
| 19:13.04 | RipeR-81 | [TK]D-Fender thanks |
| 19:13.06 | pots_line | $$ |
| 19:13.15 | ManxPower | pots_line: we DID have a problem with some phones with the sip.cfg and phone1.cfg being older than the SIP firmware we were using. |
| 19:13.21 | tripps | no CDP here on network and vlan config doesn't appear to have anything either |
| 19:13.24 | [TK]D-Fender | Indeed 2.2.0 is out. |
| 19:13.27 | ManxPower | pots_line: uh, polycom firmware is FREE. |
| 19:13.36 | [TK]D-Fender | HUGE improvements.... |
| 19:13.41 | pots_line | not 2.2 |
| 19:13.43 | ManxPower | tripps: CDP is enabled by default on most polycom phones. |
| 19:14.03 | [TK]D-Fender | pots_line: Yes, the firmware is FREE. Its just the Polycom won't hand it to YOU personally. |
| 19:14.08 | pots_line | ah |
| 19:14.12 | [TK]D-Fender | pots_line: Contact your reseller |
| 19:14.19 | pots_line | looking up reseller phone number now |
| 19:14.50 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 19:15.06 | ManxPower | [TK]D-Fender: you like sip.ld 2.2.x? |
| 19:15.09 | ManxPower | If so, why? |
| 19:15.16 | pots_line | Thnks!! |
| 19:15.32 | pots_line | Quick Q? |
| 19:15.33 | [TK]D-Fender | ManxPower: No need to use the super composite one, but it works... |
| 19:15.36 | pots_line | cdp vlan |
| 19:15.48 | pots_line | how does that work? |
| 19:16.06 | ManxPower | "super composite one"? |
| 19:16.13 | [TK]D-Fender | ManxPower: they broke it up by model... |
| 19:16.17 | *** join/#asterisk Dovid (n=Dovid@bzq-88-153-144-108.red.bezeqint.net) |
| 19:16.19 | ManxPower | pots_line: set it to off or disabled |
| 19:16.26 | pots_line | Setting the vlan via CDP |
| 19:16.29 | ManxPower | [TK]D-Fender: Oh! I guess that is handy |
| 19:16.34 | Dovid | is there any way in a sip trace to see what type of DTMF is being sent to my server ? |
| 19:16.42 | ManxPower | Dovid: "sip debug" |
| 19:16.47 | [TK]D-Fender | ManxPower: Much smaller by piece.... |
| 19:16.53 | tripps | ManxPower: doing what you suggested with pings, interesting thing is it's the first hop (external interface of premise managed router) is highly variable! it cruises along for a while with < 1ms and then 3 pings over 100ms and back down again. i would say that could be the source of our trouble? the rest of the pings along the hops are consistent |
| 19:17.10 | ManxPower | if you see INFO packets, then it is INFO, if you see rtp 101 packets that is rfc2833, if you see nothing it is INBAND |
| 19:17.14 | Dovid | ManxPower: I am doing a SIP debug |
| 19:17.24 | Dovid | i just dont understand most of it and I am trying to now ;) |
| 19:17.45 | pots_line | Nevermind . . . I should read a bit before asking . . . |
| 19:17.45 | ManxPower | tripps: it MAY be significant. Most routers treat ICMP as a very low priority. |
| 19:18.03 | tripps | ManxPower: right - weird though |
| 19:18.50 | yang | [TK]D-Fender: ok, I am learning asterisk commands slowly, but you think tha the whole extensions.conf should be rewritten just for MONITOR to work? |
| 19:19.06 | ManxPower | tripps: it is something to talk to your provider about. |
| 19:19.17 | tripps | ManxPower: shooting off an email as we speak ;) |
| 19:19.18 | *** join/#asterisk blackhole (n=mishu@unaffiliated/blackhole) |
| 19:19.35 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 19:19.36 | ManxPower | tripps: a type of UDP ping would be better. RTP and SIP are both UDP. |
| 19:19.48 | ManxPower | I don't know of a UDP ping program for linux. |
| 19:20.19 | ManxPower | tripps: one other thing to check is to make sure that your switch port and the port in the router are set to the same speed/duplex. What you are seeing COULD be caused by a duplex mismatch |
| 19:20.28 | hmmhesays | can't you specify protocol when you ping? |
| 19:20.34 | denon | ManxPower: like udp ping logger? |
| 19:21.04 | Dovid | ManxPower: Can you have a look at this ? |
| 19:21.05 | Dovid | http://pastebin.ca/702273 |
| 19:21.08 | ManxPower | denon: no idea. Does it do a ping type of thing using UDP instead of ICMP? |
| 19:21.19 | denon | more or less |
| 19:21.23 | denon | ManxPower: http://www.nerdlabs.org/projects/uplog.php |
| 19:21.52 | blackhole | I need to write a script which takes one parameter i.e. SourceNumber and Makes Asterisk Call SourceNumber and provide dial tone to use and use can enter the destination number and there call is connected. As user disconnects the call and presses * or # he again gets dialtone and he can again dial a number. Is that possible can someone guide me the way on how should i be thinking to achieve it |
| 19:21.52 | ManxPower | Dovid: see the one that is working is using rtp 101, that is rfc2833 DTMF |
| 19:22.02 | Dovid | ok |
| 19:22.08 | Dovid | that i understood |
| 19:22.21 | Dovid | and the other one does not have 101 or INFO so it must be sending it to me via inband ? |
| 19:22.23 | ManxPower | you must set both sides to use the same DTMF mode. |
| 19:22.27 | Dovid | nnno |
| 19:22.29 | jsmith | blackhole: Use call files or AMI to make the first call, and drop the call into the dialplan. From the dialplan, call the DISA() application |
| 19:22.45 | ManxPower | Dovid: that would be my assumption and of course INBAND ONLY works on ulaw and alaw |
| 19:22.46 | Dovid | these are calls from two seperate carriers. I was trying to compare the working one to the non working one |
| 19:22.56 | Dovid | ok |
| 19:23.07 | blackhole | jsmith, Will that keep providing Dial Tone again and again if user keep pressing * |
| 19:23.14 | Dovid | and the carrier is trying to send the call over g729 with INBAND |
| 19:23.24 | blackhole | jsmith, And allow user to enter destination number and have call connected ? |
| 19:23.25 | ManxPower | Dovid: that will never work. |
| 19:23.26 | Qwell | inband dtmf over g729? |
| 19:23.34 | jsmith | blackhole: Not 100% sure, but wouldn't be too hard to figure out |
| 19:23.34 | tripps | checking out uplog now |
| 19:23.36 | Qwell | Dovid: switch providers immediately. yours is clearly stupid |
| 19:23.44 | ManxPower | well, maybe 20% of the DTMF will work - but to me that is "not work" |
| 19:23.49 | webtech_m33 | how do i install the genzaptelconf, or is it include in the zaptel? |
| 19:23.49 | blackhole | jsmith, How would you do, do what you said and check? |
| 19:23.58 | jsmith | blackhole: DISA() takes care of allowing the user to enter the destination number and having the call connected |
| 19:24.17 | blackhole | jsmith, Alright but once its disconnected then? |
| 19:24.19 | ManxPower | blackhole: have you even read "show application DISA"? |
| 19:24.21 | Dovid | manxPower: Based on what you see that is what is happening ? |
| 19:24.22 | jsmith | blackhole: Not sure on the * to hangup and try again... you might have to get creative with the dialplan (specifically the 'g' option and Local channels) |
| 19:24.35 | blackhole | Hmm, Okay |
| 19:24.57 | *** join/#asterisk Buhntz (i=Boones@port-212-202-170-97.dynamic.qsc.de) |
| 19:25.01 | s34n | The wiki page for the Page command shows a macro-page. Is paging a command or a macro? |
| 19:25.15 | jsmith | Page() is an application. |
| 19:25.21 | [TK]D-Fender | yang: its not a big fix for basic monitor, but no doubt you'll have a bunch of things to do to get your whole setup working like you want it to. To answer that I'd need a better picture of whatever else you have in mind. |
| 19:25.31 | pots_line | still looking for jitter tools |
| 19:25.42 | s34n | jsmith: does it require or sepend on the macro-page? |
| 19:25.44 | ManxPower | s34n: you should NEVER EVER look at the wiki for application docs. Use "show application X" or "show applications" |
| 19:25.50 | [TK]D-Fender | s34n: The WIKI page uses that word in the macro, the context, EVERYWHERE. Its very poor for your ability to follow... |
| 19:26.10 | yang | [TK]D-Fender: well I would just like the monitor string to be correct for now, I am studying the configurations daily.... |
| 19:26.21 | yang | http://pastebin.ca/702276 |
| 19:26.35 | [TK]D-Fender | s34n: App_page is the application. the sample dialed LOCAL channels for its ability to set device-specific auto-answer headers,e tc. |
| 19:26.40 | ManxPower | yang: expect to totally rewrite your dialplan many times as you learn Asterisk |
| 19:27.12 | yang | ManxPower: I did that 3 times allready in 2 days |
| 19:27.21 | [TK]D-Fender | yang: remove "${REC_DIR}/" from all of your monitor lines, and add a Monitor call before : exten => _X.,1,Dial(SIP/${EXTEN}@e1,60,t) |
| 19:27.23 | ManxPower | yang: that is about average |
| 19:27.38 | yang | [TK]D-Fender: nice |
| 19:27.58 | yang | Its quite difficult to understand all the strings |
| 19:28.11 | blackhole | ManxPower, Yes i read that but i wanted to be sure that DISA would make it work or do i have to do something extra to have call again if user presses * |
| 19:28.12 | tripps | ManxPower: performing udp ping now |
| 19:28.15 | yang | requires some advanced coding skills |
| 19:28.19 | ManxPower | yang: that is why it is a bad idea to use too many variables when learning asterisk |
| 19:28.26 | pots_line | http://wiki.wireshark.org/VoIP_calls might help a bit. |
| 19:28.40 | pots_line | finding jitter |
| 19:28.44 | *** part/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 19:28.44 | ManxPower | blackhole: does "show application DISA" say you can exit out with * ? |
| 19:29.05 | blackhole | ManxPower, No ... |
| 19:29.09 | ManxPower | tripps: why don't you enable Asterisk's jitter buffer before you go thru all that work. |
| 19:29.19 | ManxPower | blackhole: then it prolly does not support it. |
| 19:29.26 | jsmith | ManxPower: He may have to use a Dial(Local/123@foo/n,hH) |
| 19:29.34 | blackhole | ManxPower, But if call ended then pressing * would provide dial tone or not? |
| 19:29.37 | ManxPower | jsmith: *nod* |
| 19:29.47 | ManxPower | blackhole: why would you think that? |
| 19:29.48 | Dovid | ManxPower: once I have you here. A client's server throws this message from time to time |
| 19:29.49 | Dovid | Sep 18 11:05:31 ERROR[10495]: cdr_csv.c:237 csv_log: Unable to re-open master file /var/log/asterisk//cdr-csv//Master.csv : Read-only file system |
| 19:30.03 | Dovid | I thought it was a permission error but it does not happen all the time. It happens on and off |
| 19:30.08 | ManxPower | Dovid: that has NOTHING to do with anything we have been talking about today. |
| 19:30.22 | blackhole | ManxPower, Hmm, I think it wouldn't provide but then how should i go if i want to provide |
| 19:30.35 | blackhole | ManxPower, And allow user to enter another number to be called |
| 19:30.47 | yang | [TK]D-Fender: lines marked with ; can be used as comments in extensions.conf (these arent read)? |
| 19:30.50 | ManxPower | blackhole: you would have to code it in app_disa.c or have someone do that for you. Or you can do the Local/ hack as talked about by jsmith |
| 19:30.50 | Dovid | ManxPower: I know. its a new issue ;) |
| 19:30.59 | Dovid | the old issue is the carriers fault |
| 19:31.07 | blackhole | Hmm, Okay... Thanks ManxPower |
| 19:31.07 | ManxPower | Dovid: it is not an issue I'm interested in helping someone fix. |
| 19:31.41 | Dovid | ManxPower: is it a bug in asterisk ? |
| 19:31.47 | [TK]D-Fender | yang: Correct |
| 19:33.43 | tripps | ManxPower: take a look at that udp page http://www.nerdlabs.org/projects/uplog.php - it says out of sequence udp packets are labeled with a colon. EVERY packet in my UDP ping to the first hop is a colon! |
| 19:34.03 | *** join/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 19:34.14 | tripps | ManxPower: we're using 1.2 so we don't have a jitter buffer (right?) |
| 19:34.17 | styelz | if my asterisk box is behind a NAT box, what should i set the externip to ? The NAT box's external IP or the NAT boxes internal IP. |
| 19:34.34 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 19:34.35 | tripps | it seems to me that there is something seriously awry with the premise router |
| 19:34.41 | [TK]D-Fender | jsmith: Whats the "/n" on the end of that local channel for? Its something I've never used or seem to have needed and never saw documented.... |
| 19:34.47 | Dovid | styelz: The Public IP |
| 19:34.55 | Nugget | styelz: which makes more sense for "externip"? |
| 19:34.56 | styelz | hmm. ok |
| 19:34.57 | jsmith | [TK]D-Fender: It's magic |
| 19:34.58 | [TK]D-Fender | styelz: .... |
| 19:35.00 | [TK]D-Fender | ~sipnat |
| 19:35.01 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 19:35.01 | Dovid | lol |
| 19:35.02 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^ |
| 19:35.20 | [TK]D-Fender | jsmith: how.... informative :p |
| 19:35.36 | jsmith | [TK]D-Fender: Actually, it tells the Local channel not to optimize itself away... ordinarily if you had something like SIP -> Local -> SIP, the Local would try to get out of the way and make it go SIP -> SIP |
| 19:35.51 | styelz | if i set it to the nat external IP, i cant make outgoing sip calls. but i can if i dont set the localnet |
| 19:35.56 | jsmith | [TK]D-Fender: With the /n on the end, it makes the Local channel stay in the middle |
| 19:35.59 | styelz | do i need to set localnet ? |
| 19:36.02 | jsmith | [TK]D-Fender: Which is often a useful thing to do |
| 19:36.08 | [TK]D-Fender | styelz: Yes. Read the guide |
| 19:36.12 | Dovid | styelz: YES |
| 19:36.14 | Dovid | ~RTFM |
| 19:36.15 | jbot | i heard rtfm is Read The F*cking Manual (TM). It is a suggestion to do your homework before posting a question. Sometimes used as RTFM $SPECIFIC_MANUAL to refer to a specific source of information. See also http://en.wikipedia.org/wiki/RTFM |
| 19:36.15 | *** join/#asterisk RoyK (n=roy@35.84-48-13.nextgentel.com) |
| 19:36.18 | jsmith | [TK]D-Fender: (especially for the tThH options to dial) |
| 19:36.20 | RoyK | ~seen royk |
| 19:36.23 | jbot | royk is currently on #asterisk (7s). Has said a total of 1 messages. Is idling for 3s, last said: '~seen royk'. |
| 19:36.23 | [TK]D-Fender | jsmith: Yup... |
| 19:36.24 | RoyK | ~seen xming |
| 19:36.25 | jbot | xming <n=xming@gentoo/user/xming> was last seen on IRC in channel #asterisk, 343d 1h 1m 19s ago, saying: 'sipura2k?'. |
| 19:36.28 | *** part/#asterisk RoyK (n=roy@35.84-48-13.nextgentel.com) |
| 19:36.37 | styelz | yea but it stops working if i set localnet |
| 19:36.40 | [TK]D-Fender | jsmith: I have cheated with Can_local for exactly that purpose before... |
| 19:36.56 | jsmith | [TK]D-Fender: As have I, many a time |
| 19:36.57 | [TK]D-Fender | jsmith: Or mor precisely advised such strategies. |
| 19:37.22 | [TK]D-Fender | styelz: pastebin your sip.conf masking only passwords |
| 19:37.32 | blackhole | jsmith, Can u explain a bit on how dialplan can help? |
| 19:37.54 | styelz | ok |
| 19:38.05 | jsmith | blackhole: Not without you understanding how chan_local works. It's pure speculation on my part that it will actually work, and I don't have time right now to actually try it in the lab |
| 19:38.22 | blackhole | jsmith, Okay.... |
| 19:39.12 | jsmith | blackhole: I wish I could do more, but I'm swamped |
| 19:39.13 | tripps | ManxPower: udp pings to internal hosts are responding correctly, udp pings to external hosts along sip gateway traceroute all respond out of sequence. I'm firing up wireshark to get empirical data |
| 19:39.21 | blackhole | jsmith, No probs |
| 19:39.33 | jsmith | tripps: Out-of-order packets would definitely cause grief. |
| 19:39.39 | blackhole | jsmith, Let me play if i would have any issues i can ask you and you can give your thoughts |
| 19:39.43 | jsmith | tripps: Does "iax2 show netstats" show out-of-order packets? |
| 19:40.01 | arekm | does anyone know where I could get free music (for music-on-hold) ? (not sure if anyone gives such thing for free) |
| 19:40.09 | jsmith | blackhole: I'm always happy to give my thoughts -- I just don't always have time to do a lot with them |
| 19:40.15 | tripps | jsmith: not using iax - using sip |
| 19:40.32 | jsmith | tripps: Oh, that's right... confusing two conversations in two different channels |
| 19:40.43 | Dovid | tripps: Do you mind explaining what Out-of-order packets is ? |
| 19:40.44 | jsmith | tripps: Does the RTP analysis in Wireshark show OOO packets? |
| 19:41.06 | tripps | jsmith: but i think it's definitely a problem. i'm about to find that out now |
| 19:41.07 | Dovid | oops. let me guess. the packets are not coming in, in the correct order ? |
| 19:41.09 | jsmith | Dovid: Packets come along out of order (usually due to jitter or from packets taking different routes) |
| 19:41.20 | Dovid | jsmith: thanks |
| 19:41.28 | yang | [TK]D-Fender: OK, I guess the logging is working now, but only for the Intranet connections...http://pastebin.ca/702295 you wrote to me ==> add a Monitor call before : exten => _X.,1,Dial(SIP/${EXTEN}@e1,60,t) - I add like this exten => _X.,1,Monitor ? |
| 19:41.51 | [TK]D-Fender | yang: Just like any other call...... |
| 19:43.39 | *** join/#asterisk etfonhomey (n=chatzill@mail.advancmed.org) |
| 19:43.57 | *** join/#asterisk twilson (n=twilson@CPE-65-30-30-20.kc.res.rr.com) |
| 19:44.02 | yang | [TK]D-Fender: I guess i am all set for a "extensions reload now" http://pastebin.ca/702304 |
| 19:44.27 | [TK]D-Fender | yang: If at first you don't succeed.... thats why they call it "failure". |
| 19:44.58 | *** join/#asterisk nerdygirl_ellie (n=ellie@209.168.199.178) |
| 19:45.10 | yang | [TK]D-Fender:I wonder how much does asterisk support costs, and who offers it? |
| 19:45.32 | [TK]D-Fender | yang: Depends who you're paying to support it and how much support you need :) |
| 19:46.00 | nerdygirl_ellie | Hi! Is asteriskdocs.org offline permanently or just temporarily? |
| 19:46.05 | yang | Not much support, but I don't dare to touch the running gateways with my skills... |
| 19:46.08 | [TK]D-Fender | nerdygirl_ellie: Temporarily |
| 19:46.33 | [TK]D-Fender | yang: My rates are very accessable :) |
| 19:46.45 | yang | [TK]D-Fender: ok tell me into query |
| 19:47.48 | Dovid | nerdygirl_ellie: have you tried voip-info.org ? |
| 19:47.54 | styelz | [TK]D-Fender: http://pastebin.ca/702308 |
| 19:48.34 | styelz | http://pastebin.ca/702309 |
| 19:48.40 | nerdygirl_ellie | Dovid: Yes, and it's great for "what does this command do" queries, but not so great for I am setting up box number 3 and want to do it "right" this time type stuff. |
| 19:49.07 | [TK]D-Fender | styelz: in reading my guide you forgot "canreinvite=no" which should appear under [general] and pretty much EVERYWHERE. |
| 19:49.11 | Dovid | okeis. |
| 19:49.27 | Dovid | nerdygirl_ellie: then just post all your questions here. TK is real good at answering them ;) |
| 19:49.36 | styelz | ah ok, i thought NAT forced that.. will try |
| 19:49.38 | [TK]D-Fender | styelz: and I smell FreePBX... or at least leftovers.... |
| 19:49.45 | styelz | :P |
| 19:49.56 | [TK]D-Fender | styelz: No, it doesn't and the guide made explicit warning about that... |
| 19:49.57 | styelz | cause problems ? |
| 19:49.58 | jsmith | nerdygirl_ellie: Just temporarily... I"m working on fixing it |
| 19:50.13 | jsmith | nerdygirl_ellie: I'm sitting here staring at the old server as I type this |
| 19:50.19 | nerdygirl_ellie | jsmith: Thanks! |
| 19:50.21 | styelz | ah ok |
| 19:50.38 | nerdygirl_ellie | jsmith: Anything I can do to help? |
| 19:50.41 | jsmith | nerdygirl_ellie: Expect big changes in the next week, including the second edition |
| 19:50.55 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 19:51.01 | jsmith | nerdygirl_ellie: Hmmmn... not that I can think of |
| 19:51.21 | jsmith | nerdygirl_ellie: I've had a new server donated (should show up any day), and in the meantime a friend is letting me use his box |
| 19:51.22 | tripps | jsmith: doing RTP stream analysis now - what am i looking for? |
| 19:51.36 | jsmith | tripps: Jitter and out-of-order packets and dropped packets |
| 19:51.36 | elriah | lol, anyone notice on the new polycom firmware on the 550 at least the phone icon is covering up part of the line label text, kind of like a transparent gif without the transparency turned on... |
| 19:52.05 | Dovid | jsmith: I have a few box's up if u need to host for now or the future |
| 19:52.58 | jsmith | Dovid: Thanks, but that's the problem I ran into last time... I need something *I* control and have physical access to |
| 19:53.06 | jsmith | Dovid: But I appreciate the offer. |
| 19:53.24 | styelz | got a link to this guide ? |
| 19:53.28 | styelz | please |
| 19:55.08 | tripps | jsmith: i saw quite a bit of that with previous captures |
| 19:56.01 | mcab | elriah: you're probably using the wrong config files with the new firmware then |
| 19:56.08 | nerdygirl_ellie | I have about 10 cisco spa942's connecting to an asterisk gateway/switch in a reasonably fast server with a digium 24 port analog card ( 2 4 port daughter boards) and they are griping about echo. The network is switched, and I'm at a bit of a loss to explain the echo. |
| 19:56.17 | elriah | mcab: eh? How would they affect the line label icon? |
| 19:56.29 | elriah | (I'm not btw, but just curious) |
| 19:57.04 | *** join/#asterisk DaveCanoe (n=Dave@H6.C30.B96.tor.eicat.ca) |
| 19:57.15 | drako | analog lines sucks. |
| 19:57.22 | nerdygirl_ellie | nerdygirl_ellie: the current zaptel on the box is 1.2.12, and I'm going to bring that and asterisk up to the current releases. Any other suggestions? |
| 19:57.36 | mcab | elriah: common side effect of using the wrong configs with polycom firmware, and I don't see that issue with my 550 :-) |
| 19:57.51 | elriah | hrm.. I'm using bootrom 4.0, sip.ld 2.2. you? |
| 19:57.57 | mcab | yup |
| 19:58.00 | mcab | same |
| 19:58.13 | elriah | interesting.. maybe i missed something, wouldn't be the first time, lol |
| 19:58.21 | [TK]D-Fender | elriah: and you rebuilt all of your configs based ont he new sample configs right? :) |
| 19:58.41 | elriah | [TK]D-Fender: Uhm, yea... lol, hell no.. I just kind of went through them.. |
| 19:58.42 | elriah | @#$@$ |
| 19:58.45 | elriah | damnit |
| 19:58.56 | *** join/#asterisk el_critter (n=chatzill@190.74.96.121) |
| 19:59.04 | el_critter | hi there |
| 19:59.07 | jsmith | nerdygirl_ellie: You can use fxotune to tune the FXO ports on your Digium card. Or, call Digium support and they'll send you some free licenses for their HPEC software echo cancellation stuff |
| 19:59.13 | [TK]D-Fender | elriah: I'll let you go punish yourself some more... you're far more efficient than I am :) |
| 19:59.14 | jsmith | hello el_critter |
| 19:59.49 | *** join/#asterisk mishkiz (n=lincolnz@189.20.57.154) |
| 20:00.30 | styelz | canreinvite=yes breaks incomming sip calls |
| 20:00.47 | el_critter | I'm going to buy a P3 800Mhz for asterisk, 512MB-RAM, how many concurrent calls can that server handle? |
| 20:00.51 | styelz | i mean =no |
| 20:00.56 | hmmhesays | media or no media? |
| 20:01.11 | mcab | elriah: I'd recommend having a read of this, and seeing if you can incorperate some of the ideas into your configuration :-) http://knowledgebase.polycom.com/kb/search.do?cmd=displayKC&docType=kc&externalId=EndUser-TechAlerts-Audio-whitepaperconfigurationfilemanagementonsoundpointipphonespdf&sliceId=pdfPage_1&dialogID=3860883&stateId=0%200%20372546 |
| 20:01.11 | hmmhesays | if you're not handling media, at least 50 |
| 20:01.33 | el_critter | hmmhesays: are you talking to me? |
| 20:01.37 | hmmhesays | yeah |
| 20:01.54 | hmmhesays | 0 media though |
| 20:02.02 | el_critter | hmmhesays: Oh thanks... no, no media, just voice |
| 20:02.26 | hmmhesays | by no media, I mean have asterisk reinvite the calls so the voice path doesn't go through the box |
| 20:02.33 | elriah | mcab: Thanks. |
| 20:02.47 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 20:02.50 | lirakis | can you set up a sip peer from asterisk to send media and signaling to different ip's? |
| 20:03.22 | jsmith | el_critter: Probably 10 to 12 concurrent calls |
| 20:03.32 | hmmhesays | you can make asterisk reinvite the call so the media goes directly between endpoints |
| 20:03.41 | hmmhesays | yeah handling media would drop that number drastically |
| 20:03.47 | mishkiz | hello all...im using here a asterisk 1.4...I imported a extensions.conf from a 1.2...in console, the asterisk tells me "No such application 'DBget'"...anybody know what is the new name of this application ? |
| 20:03.54 | elriah | mcab: First try... It was sip.cfg |
| 20:04.00 | hmmhesays | thats cause DBget is gone |
| 20:04.19 | elriah | mcab: I have everything broken out so just replacing my old sip.cfg should work great. |
| 20:04.32 | mishkiz | hmmhesays, what replaces it ? |
| 20:04.35 | el_critter | hmmhesays: No, I want all calls through the server |
| 20:04.45 | hmmhesays | the el_critter: probably no more than 20 |
| 20:04.54 | styelz | rtfm .. pfttt |
| 20:05.06 | el_critter | hmmhesays, jsmith: thanks a lot!!! |
| 20:05.36 | hmmhesays | what can you pick a p3 up for these days? |
| 20:05.38 | mcab | elriah: yeah, that's the way I do it - way less of a PITA |
| 20:06.47 | styelz | already read this http://www.voip-info.org/wiki-Asterisk+sip+canreinvite |
| 20:06.54 | *** join/#asterisk elixer (i=elixer@65.207.74.18) |
| 20:09.12 | *** join/#asterisk tsurko (n=tsurko@213.91.216.130) |
| 20:09.29 | el_critter | hmmhesays: Its used, dirt cheap :) |
| 20:09.41 | [TK]D-Fender | hmmhesays: I can get a P3-1000 w/ 256 in a mini desktop for about $90 |
| 20:09.58 | hmmhesays | thats plenty to run a few phones on |
| 20:10.25 | el_critter | brb, thanks again for your help |
| 20:13.02 | [TK]D-Fender | hmmhesays: indeed |
| 20:13.03 | nerdygirl_ellie | styelz: The M is down right now. :D |
| 20:13.14 | [TK]D-Fender | hmmhesays: I'm constantly eyeing it... I like small desktops like those |
| 20:13.43 | styelz | :) |
| 20:14.03 | ManxPower | styelz: so you know that you can't really do reinvites and NAT. |
| 20:14.14 | styelz | yes |
| 20:14.17 | *** join/#asterisk zcionn_ (n=a@58.69.243.203) |
| 20:15.11 | styelz | everything works fine without externip / localnet and nat |
| 20:15.17 | lirakis | <PROTECTED> |
| 20:15.54 | styelz | except.... an external SIP call from outside NAT.. cant hear MOH or RING when i direct it to a Queue |
| 20:15.55 | [TK]D-Fender | lirakis: http://www.dantech.ca/images/photos/ssf.jpg |
| 20:16.19 | [TK]D-Fender | lirakis: Common MicroATX-like size from HP/Compaq/etc |
| 20:16.25 | styelz | but.. it can hear MOH if i answer the call. and place it on hold |
| 20:16.33 | styelz | its got me frigged |
| 20:16.49 | tripps | ManxPower: we've sent off an email to the premise ISP with log files asking what is up . . the problem has to be there somewhere |
| 20:16.56 | lirakis | [TK]D-Fender: hmm.. i had one of these ebox 4800's .. but the cf slot was doa .. had to return it... http://www.wdlsystems.com/ebox/ebox.shtml |
| 20:17.04 | lirakis | [TK]D-Fender: it was so cool too |
| 20:17.21 | *** part/#asterisk nerdygirl_ellie (n=ellie@209.168.199.178) |
| 20:17.36 | styelz | bugger me |
| 20:17.51 | webtech_m33 | ZT_SPANCONFIG failed on span 1: Invalid argument (22) |
| 20:18.00 | [TK]D-Fender | lirakis: Oh no... those are not just SFF's, those are "mini PC's", industrial computers, etc.. |
| 20:18.12 | [TK]D-Fender | lirakis: this chassis takes standard drives, and has slots :) |
| 20:18.13 | jcanfield | ~book |
| 20:18.14 | jbot | [book] Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 or Temporarily at http://www.aocomputing.net/AsteriskTFOT.pdf |
| 20:18.40 | styelz | got that one too |
| 20:19.43 | styelz | meh, suns rising.. bed time |
| 20:20.15 | Blackthorn | you know the book your refering too "the future of telephony" has a lot to be desired since a lot of things have chanced from that book. |
| 20:21.05 | Blackthorn | er. simply put it's outdated inmho |
| 20:21.08 | wishes | anyone here done recording? |
| 20:21.51 | wishes | im getting this http://pastebin.ca/702367 when i try to record |
| 20:22.05 | wishes | (ie answerphone messages etc) |
| 20:23.26 | wishes | actually i think i know what it is - its the camera |
| 20:27.23 | tripps | jsmith, ManxPower: according to ISP pings that jump every 30 seconds are due to BGP scanning routes |
| 20:28.18 | [TK]D-Fender | BBIAB |
| 20:28.25 | wishes | now to figure out how to disable it |
| 20:30.30 | *** join/#asterisk gremzoid (n=gremzoid@122.104.27.157) |
| 20:34.13 | ManxPower | tripps: It really doesn't matter WHAT is causing it, it is adding massive jitter to your voip calls. |
| 20:34.27 | tripps | ManxPower: agreed |
| 20:34.42 | ManxPower | tripps: what happened when you enabled Asterisk's jitter buffer? |
| 20:35.06 | ManxPower | answer fast, as I have to go install a satellite dish |
| 20:35.25 | tripps | ManxPower: remember we're running 1.2 |
| 20:35.33 | ManxPower | tripps: ah. you're screwed. |
| 20:36.15 | tripps | ManxPower: would qos help? or would it be smart to upgrade? the other option is just getting T1 to SIP provider from another ISP |
| 20:37.29 | tripps | ManxPower: also, how do i check cisco 79xx phones (sip load) to disable cdp? can't find the config anywhere |
| 20:38.48 | lirakis | wishes: .. ythere is no translator to write the file in a known audio format |
| 20:39.05 | *** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr) |
| 20:41.07 | pots_line | ManxPower: The Polycom we have that is rebooting is a 601 with three sidecars . . . Monitored the POE switch during an overhead page . . . It asked for more power and the switch gave it . . . But, the phone rebooted |
| 20:41.14 | wishes | <PROTECTED> |
| 20:41.42 | wishes | so it seems anyway :) |
| 20:41.56 | pots_line | Is there a way to tell it to request 12 W by default instead of 11 W and not need to request additional power on page> |
| 20:42.01 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 20:42.57 | *** join/#asterisk marlow (n=marlow@2001:770:119:0:216:d3ff:fe30:973a) |
| 20:43.58 | *** part/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 20:44.45 | *** join/#asterisk marcan (i=1337@host214-134.cvd.fit.edu) |
| 20:45.39 | *** join/#asterisk J4k3- (i=J4k3@wls-a011.intrastar.net) |
| 20:49.16 | *** join/#asterisk VoicePulse (n=contact@unaffiliated/voicepulse) |
| 20:49.51 | pots_line | maxed out the POE settings on the switch to 15.4 W . . . and the reboot problem didn't happen. |
| 20:50.20 | pots_line | CDP auto config POE settings . . . seemed to be causing the problem |
| 20:52.10 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 20:56.13 | *** join/#asterisk zirman (i=zirman@ip194.207.107.216.seg.net) |
| 20:58.02 | DrukenLPY | anyone know is streets and trips 2008 is out yet? |
| 20:58.27 | russellb | ... |
| 21:00.33 | [TK]D-Fender | And now for something completely different! |
| 21:03.25 | DrukenLPY | :) |
| 21:03.44 | DrukenLPY | i figured out of 300+ people, someone would know :) |
| 21:03.50 | rob0 | Google Maps is out, and doesn't require Windows. :) |
| 21:04.29 | DrukenLPY | but does require an internet connection |
| 21:06.29 | rob0 | true |
| 21:08.02 | [TK]D-Fender | DrukenLPY, Blame Rogers & Bell. |
| 21:09.21 | *** join/#asterisk Delvar (n=Delvar@77.240.56.18) |
| 21:12.15 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177583236.dsl.bell.ca) |
| 21:15.10 | *** join/#asterisk kiscokid (n=ron@208.106.35.66) |
| 21:15.45 | Daejeo1 | useragent= anything |
| 21:15.51 | Daejeo1 | will it work? |
| 21:16.21 | Daejeo1 | my provider is blocking call from asterisk |
| 21:18.18 | [TK]D-Fender | Daejeo1, Oh I don't know... have you considered.. umm... TRYING?!?! |
| 21:18.25 | styelz | lol |
| 21:18.54 | Daejeo1 | :) |
| 21:19.03 | Daejeo1 | i am trying |
| 21:19.05 | styelz | its speak like a pirate day today |
| 21:19.09 | styelz | arrrr |
| 21:19.40 | wishes | arrr me hearties :) |
| 21:19.44 | wishes | Ya landlubber whut deserves the black spot! |
| 21:19.50 | wishes | What else ye got? An' be quick about it, I be shippin' out soon! |
| 21:19.57 | styelz | shiver me timbers |
| 21:19.59 | Daejeo1 | why providers hate asterisk? |
| 21:20.00 | wishes | Arrr, so ye be wantin' t' go to sea an' ye don't be wantin' t' end up in Davy Jones' Locker. Then ye best be learnin' t' be talkin' like a buccaneer. |
| 21:20.40 | Qwell | styelz: not yet it isn't |
| 21:20.47 | Qwell | not if you go by GMT anyhow |
| 21:20.56 | styelz | it is if you live in Australia.. with me |
| 21:21.04 | Qwell | Australia doesn't have pirates |
| 21:21.50 | styelz | oh ok |
| 21:22.36 | styelz | anyway.. |
| 21:22.39 | [TK]D-Fender | Daejeo1, because they don't want their asses sued when you blame them for not being able to dial 911 because you screwed your configs up. |
| 21:22.43 | styelz | harrrr!! |
| 21:23.16 | wishes | Qwell: lies! australia was made up of convicts - some of which im sure were pirates :D |
| 21:23.26 | wishes | heck, just look at politictions |
| 21:23.33 | wishes | (sp) |
| 21:23.40 | Qwell | g'darrrgghh! |
| 21:23.43 | Qwell | ... |
| 21:23.46 | Qwell | sorry |
| 21:24.08 | *** join/#asterisk GreggB (n=GreggB@66.206.86.107) |
| 21:25.01 | webtech_m33 | <PROTECTED> |
| 21:25.05 | [TK]D-Fender | tank* |
| 21:25.23 | wishes | [TK]D-Fender: you never told me you were into kinky stuff o_O |
| 21:25.28 | [TK]D-Fender | webtech_m33, by itself a completely MEANINGLESS error. Pastebint he ENTIRE call attempt |
| 21:25.30 | Daejeo1 | >[TK]D-Fender: http://dumbme.voipeye.com.au/trixbox2/trixbox2_without_tears.pdf this guy is telling about useragent spoofing |
| 21:26.17 | [TK]D-Fender | "Trixbox without tears"? PATHETIC. Thats like "Stupidity for Morons". |
| 21:26.23 | webtech_m33 | trying to get my quad T1/pri digiam card to load |
| 21:26.43 | Daejeo1 | >[TK]D-Fender: do you know this guy/ |
| 21:26.50 | [TK]D-Fender | webtech_m33, pastebin a COMPLETE call attempt and your configs if you expect any help. |
| 21:26.52 | [TK]D-Fender | ~pb |
| 21:26.53 | jbot | A Pastebin is a place to paste your stuff without flooding 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://www.rafb.net/paste |
| 21:26.53 | webtech_m33 | that's when i start asterisk like this asterisk -U asterisk -G asterisk -cvv |
| 21:27.06 | [TK]D-Fender | Daejeo1, Maybe, maybe not, but I'm sure I know his KIND. |
| 21:28.36 | webtech_m33 | http://paste.debian.net/37485 |
| 21:29.14 | webtech_m33 | the last few lines is were it dies |
| 21:29.29 | webtech_m33 | i don't think my card drivers are install right |
| 21:30.50 | webtech_m33 | http://paste.debian.net/37486 ztcfg -vvvv |
| 21:31.11 | [TK]D-Fender | webtech_m33,kill your script, and pastebin "cat /proc/interrupts" and "ztcfg -vvvv" |
| 21:31.43 | [TK]D-Fender | webtech_m33, pastebin your zaptel & zapata |
| 21:31.54 | webtech_m33 | http://paste.debian.net/37487 |
| 21:32.51 | webtech_m33 | this has it all http://paste.debian.net/37488 |
| 21:32.54 | [TK]D-Fender | webtech_m33, zaptel & zapata please |
| 21:33.01 | webtech_m33 | i added it |
| 21:33.04 | webtech_m33 | to the http://paste.debian.net/37488 |
| 21:33.39 | [TK]D-Fender | looking... |
| 21:33.47 | webtech_m33 | my card is still T1 lights are still runing |
| 21:34.03 | webtech_m33 | they should be solid |
| 21:34.13 | webtech_m33 | it's at the very bottom |
| 21:34.21 | *** join/#asterisk blinky42 (n=me@c-71-230-47-244.hsd1.pa.comcast.net) |
| 21:34.40 | webtech_m33 | line 320 |
| 21:35.07 | Daejeo1 | [TK]D-Fender: is there wild card dial plan for sipura? |
| 21:35.18 | Daejeo1 | dial anything |
| 21:35.53 | [TK]D-Fender | Daejeo1, sorta |
| 21:36.09 | [TK]D-Fender | webtech_m33, ok, just start * manually, not through the script. |
| 21:36.20 | webtech_m33 | how? |
| 21:36.31 | webtech_m33 | asterisk or the card> |
| 21:36.39 | webtech_m33 | ? |
| 21:37.35 | *** join/#asterisk AJaymn (n=Me@71-82-218-158.dhcp.mdsn.wi.charter.com) |
| 21:38.03 | [TK]D-Fender | webtech_m33, "asterisk -gvvvvvc" as the user you run it as |
| 21:38.15 | AJaymn | when I do a sip show channels im getting 192.168.0.101 (None) a6095f3c-86 00101/17530 unkn No Rx: REGISTER |
| 21:38.29 | AJaymn | and its constitly there.. do i have a phone not registering right? |
| 21:38.40 | [TK]D-Fender | AJaymn, no, thats fine & normal. |
| 21:38.45 | webtech_m33 | [TK]D-Fender same thing as the pastebin |
| 21:39.00 | webtech_m33 | i have to run |
| 21:39.05 | webtech_m33 | i will be back tommorrow |
| 21:39.15 | webtech_m33 | i will hunt google some |
| 21:39.18 | webtech_m33 | take are all |
| 21:39.18 | AJaymn | [TK]D-Fender well i would see it sometimes before.. but this is constitly there everytime i issue the command.. |
| 21:39.48 | [TK]D-Fender | AJaymn, because maybe some phons register more frequently. This is NOT a means of seeing actual channels in use |
| 21:40.34 | AJaymn | ko |
| 21:41.16 | *** join/#asterisk xphat (n=Rhon@65.183.2.101) |
| 21:42.06 | xphat | Im having a problem here with asterisk not connecting calls immediately when calls are answered over zap lines... w |
| 21:42.55 | jsidhu2 | where can I download HUD Server? Is it only available as part of trixbox? |
| 21:44.44 | AJaymn | was looking in the debug ---- chan_sip.c: stale nonce received from .... What does that mean? |
| 21:45.10 | *** join/#asterisk n0n4m3 (n=NoName@noname.rula.net) |
| 21:45.15 | n0n4m3 | evening |
| 21:46.52 | [TK]D-Fender | AJaymn, sounds like a slow response. |
| 21:47.14 | n0n4m3 | any ideas why asterisk wouldn't register to a sip peer? |
| 21:47.31 | n0n4m3 | i have the register => and the [server] in sip.conf |
| 21:47.57 | n0n4m3 | i'm kinda clueless |
| 21:50.04 | [TK]D-Fender | n0n4m3, PASTEEBIN <-------- |
| 21:53.27 | n0n4m3 | [TK]D-Fender http://rula.net/119 |
| 21:53.33 | lsodi | [TK]D-Fender> n0n4m3, PASTEEBIN <-------- |
| 21:53.33 | lsodi | [00:51] *** |omni| quit (Read error: 113 (No route to host)) |
| 21:53.33 | lsodi | [00:51] *** superpop |
| 21:53.40 | lsodi | syrr |
| 21:54.51 | n0n4m3 | lsodi i did paste it |
| 21:55.57 | lsodi | my mistace I copied some text from window and pasted with ctrl+v |
| 21:58.11 | lsodi | [TK]D-Fender: X-lite is acting strangely in home basically after every 60 seconds it registers in sip server, if caller tryes to call me when x-lite has just registered in asterisk, then call goes through |
| 21:59.34 | lsodi | but with Zoiper I dont have such problems |
| 22:00.37 | AJaymn | whats the best Softphone for Asterisk? |
| 22:00.56 | styelz | a fluffy one |
| 22:01.04 | lsodi | every thing works fine, tryed to change settings in X-lite but registring intervall is still after every 60 s |
| 22:03.23 | *** join/#asterisk hi365_m (n=hi365@213.151.63.189) |
| 22:04.09 | hi365_m | can i run a system command and set the results as a variable? |
| 22:05.21 | hi365_m | (from the dialplan) |
| 22:10.18 | *** join/#asterisk diemaco (n=diemaco@unaffiliated/diemaco) |
| 22:44.15 | *** part/#asterisk kiscokid (n=ron@208.106.35.66) |
| 22:49.32 | wishes | AJaymn: wengaphone does video, but it tends to be unstabled as hell :) |
| 22:50.10 | wishes | there used to be one called x10 or something or x ten ? i forget |
| 22:50.58 | n0n4m3 | x-lite? |
| 22:51.55 | wishes | is there a way to disable video for just one macro/routine in the extensions.conf ? |
| 22:52.07 | *** join/#asterisk jedaustin (n=chatzill@austin-j.its.dist.maricopa.edu) |
| 22:52.14 | wishes | we use video for general calls, but i want to record, but only record the sound |
| 23:02.33 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 23:06.28 | *** join/#asterisk thansen|laptop (n=thansen@19.243.sfcn.org) |
| 23:08.27 | *** join/#asterisk Cyon (n=cyon@216.179.31.170) |
| 23:14.59 | *** join/#asterisk jm|laptop (n=jm|home@zen.jamiem.com) |
| 23:17.39 | *** join/#asterisk amarzouk (n=chatzill@217.54.201.107) |
| 23:21.30 | amarzouk | Hi, I am having problems compiling asterisk 1.4.11 on centos, my problem is with the chan_vpb has anybody face similar problems? |
| 23:22.44 | *** join/#asterisk jingles (n=foo@39.183.dowl.anc.borealisbroadband.net) |
| 23:27.27 | CCFL_Man2 | strom_m isn't here yet |
| 23:30.16 | booray | Quick question for anyone bored: I'm getting a T1 installed next week from Verizon, which they claim is "Line formatting:B8ZS/ESF" and loop start signalling. I'm not sure which channel driver to use; pri_cpe or one of the other strange ones? |
| 23:31.15 | *** join/#asterisk elixer (n=seanbrig@c-68-55-114-113.hsd1.md.comcast.net) |
| 23:33.10 | booray | wait, I think I may have answered my own question |
| 23:33.20 | CCFL_Man2 | booray: pri_cpe |
| 23:33.43 | booray | I found the appropriate doc on asteriskguru, I think. |
| 23:33.45 | booray | thanks ccfl |
| 23:33.49 | CCFL_Man2 | booray: i bet vzn is raping your wallet for that T1 |
| 23:33.53 | CCFL_Man2 | np |
| 23:34.01 | mace | question; i've set up call parking, and dialing the park extension correctly parks the call, and everything seems to be working.. except the announce of which park extension: asterisk console thinks it's announcing the number (701) but no audio is heard - any ideas? |
| 23:34.35 | booray | CCFL_Man2: well, it's not _my_ T1 |
| 23:34.37 | booray | :-P |
| 23:35.36 | CCFL_Man2 | booray: no, but vzn is definately overcharging |
| 23:35.46 | CCFL_Man2 | vzn ftl |
| 23:36.32 | booray | I'm sure they are... but some people like the comfort of a large corporation backing their channels |
| 23:37.49 | CCFL_Man2 | thats the thing, vzn probably won't provide the best support |
| 23:38.14 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 23:38.22 | CCFL_Man2 | but maybe they will |
| 23:38.33 | CCFL_Man2 | are they the rboc? |
| 23:39.03 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 23:39.36 | AJaymn | whats a good price for T1 for asterisk? |
| 23:40.24 | CCFL_Man2 | you looking for a pri? |
| 23:40.43 | booray | CCFL_Man2: and by rboc you mean, they own the lines coming to the building? |
| 23:40.48 | booray | in that case, yes |
| 23:41.04 | CCFL_Man2 | booray: regional bell operating company |
| 23:41.08 | booray | other T1 providers (like telepacific) in the area have to run their stuff on verizon's lines |
| 23:41.16 | CCFL_Man2 | ahh |
| 23:41.17 | booray | yes, I googled and started to read the wikipedia article |
| 23:41.22 | AJaymn | CCFL_Man2 pri is what is needed for voice right? |
| 23:41.42 | CCFL_Man2 | AJaymn: it's one option and usually the most common |
| 23:41.42 | booray | but this chunk of the US on the wikipedia graphic shows at&t... |
| 23:42.27 | CCFL_Man2 | booray: unfortunately |
| 23:43.08 | CCFL_Man2 | the Ma'Bell breakup was suppose to end the telco dictatorship |
| 23:43.35 | CCFL_Man2 | and all it really did was bring in more telcos |
| 23:43.37 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 23:43.47 | AJaymn | CCFL_Man2 whats the average charge for T1 pri then? |
| 23:44.07 | CCFL_Man2 | the rules have stayed the same |
| 23:44.20 | booray | CCFL_Man2: I see. I've read about it in the past but didn't know the full history and things |
| 23:44.23 | CCFL_Man2 | AJaymn: i'm honestly not sure, i'm a home user :P |
| 23:44.51 | CCFL_Man2 | booray: overcharging is the main thing that never changed |
| 23:45.12 | booray | here in socal, at&t (formerly here sbc/pacific bell) and verizon (all gte here) kinda have the area cut in half, it seems |
| 23:45.27 | CCFL_Man2 | ahh, yeah |
| 23:46.23 | CCFL_Man2 | what bothers me is that if i don't want pots but my line brought over a T1 PRI, i got to pay thousands per month |
| 23:47.05 | booray | verizon is rolling out fios like mad here, while at&t seems to be just sitting with its head up its ass and not doing anything. |
| 23:47.10 | booray | where are you that it would be that expensive? |
| 23:47.31 | CCFL_Man2 | and since they use hdsl it uses the same pair for pots, why can't they give me a T1 PRI at a decent price |
| 23:47.33 | Nugget | I'm 10 miles too far north to get verizon FIOS, which is a real bummer. |
| 23:47.42 | CCFL_Man2 | booray: i'm a residence |
| 23:48.04 | booray | When I was getting T1 loops for customers through MPower (now telepacific) a data only line was around $350 or $400/month |
| 23:48.10 | CCFL_Man2 | Nugget: which means they don't feel like spending the cash to run the fiber |
| 23:48.12 | booray | ah, gotcha |
| 23:50.11 | CCFL_Man2 | booray: granted, the residence might not know anything about a T1 PRI, but if i want it i should be able to get it for a reasonable price |
| 23:50.36 | booray | FIOS seems to be available everywhere here except where I live. I've been seeing the trucks everywhere running lines over the past few months |
| 23:50.41 | CCFL_Man2 | i'll even supply my own smart jack :P |
| 23:51.15 | booray | CCFL_Man2: It seems as though you'll just have to move into a business park |
| 23:51.35 | CCFL_Man2 | booray: no, just pay the business price |
| 23:52.53 | CCFL_Man2 | booray: with FIOS, voice is brought over ATM |
| 23:54.32 | booray | I figured I would just ditch local phone service at that point; I don't know if they'll let you disclude it from your plan however |
| 23:56.05 | CCFL_Man2 | you need to bitch to make them keep your pots if you install fios |
| 23:56.24 | booray | I thought that wasn't an option, that they would just cut the pots? |
| 23:59.11 | CCFL_Man2 | i hear if you bitch loud enough they'll do it |
