| 00:06.43 | *** join/#asterisk Grnd-Wire (n=grundofw@65.101.128.1) |
| 00:06.49 | Grnd-Wire | ugh, that was really weird.. |
| 00:19.21 | *** join/#asterisk robl^ (n=robl@pdpc/supporter/monthlybyte/robl) |
| 00:27.56 | *** join/#asterisk Op3r (n=edwin@222.127.36.225) |
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| 00:35.39 | j0 | does anyone here successfully run asterisk in a virtual machine? |
| 00:37.12 | niekie | I have runned it in a VM once, though I do not anymore. |
| 00:37.26 | robeph | does it perform ok in a vm? |
| 00:37.27 | j0 | niekie: did it work ok? why not now? |
| 00:37.35 | niekie | Well.. I am running it in a VPS now. |
| 00:37.37 | j0 | i had problems in the past.. but i'm considering using vmware esx server now |
| 00:37.40 | robeph | i can imagine it's processor painful in that |
| 00:37.40 | niekie | So that's kind of the same thing. |
| 00:37.48 | niekie | It's Virtuozzo now. |
| 00:38.05 | j0 | does virtuozzo abstract the cpu too? |
| 00:38.23 | niekie | Not sure what that means, heh. |
| 00:38.44 | robeph | does it use the cpu directly I think he means, while creating a pseudoprocessor that delegates to the true cpu? |
| 00:38.54 | niekie | I don't think it does. |
| 00:38.56 | j0 | yes.. i'm just at swsoft's page.. |
| 00:38.58 | robeph | creating an abstraction layer |
| 00:39.13 | niekie | I know you can't load kernel modules.. |
| 00:39.15 | j0 | i love virtuozo for hosting though :) and plesk is great. |
| 00:39.21 | niekie | So it probably runs off one kernel. |
| 00:39.31 | niekie | Can't even set system time. |
| 00:39.41 | j0 | ok.. thanks |
| 00:40.04 | niekie | Also can't run MeetMe, as it requires ztdummy :-( |
| 00:40.27 | robeph | niekie: can you shove the module into the kernel on the parent os? |
| 00:40.36 | niekie | robeph: *I* can't, hehe. |
| 00:40.40 | robeph | would it be inherited? |
| 00:40.43 | niekie | I don't manage the parent OS. |
| 00:40.46 | robeph | ah ok |
| 00:40.48 | niekie | robeph: possibly.. |
| 00:40.53 | niekie | Though I can't test. |
| 00:40.56 | robeph | yeh |
| 00:41.09 | j0 | when i ran asterisk in vmware gsx server before i had strange timing issues |
| 00:41.25 | niekie | What kind of timing issues? |
| 00:42.07 | j0 | timing or interrupt, i'm not sure.. like my voice would be choppy unless i had cpu intensive programs running in the background |
| 00:42.25 | robeph | wait... IF you had cpu intensive programs it worked right? |
| 00:42.33 | robeph | usually that problem comes from the opposite factor |
| 00:42.37 | niekie | Isn't VMware GSX just "VMware Server" nowadays? |
| 00:43.04 | j0 | yeah.. IF.. it's like it wasn't getting the cpu cycles to run asterisk unless other programs in the asterisk vm were asking for them |
| 00:43.05 | j0 | niekie: yes |
| 00:43.20 | robeph | hrm, could you raise the priority on vmware? |
| 00:43.24 | j0 | tried that too |
| 00:43.31 | robeph | thats rather odd indeed |
| 00:44.03 | robl^ | asterisk really needs realtime acceess. I have never been able to get acceptable performance from a virtual server |
| 00:44.22 | niekie | Anybody know about how much bandwidth a VoIP call of 30 minutes would take (estimated)? |
| 00:44.24 | robeph | yeh I wouldn't assume it'd be too good in a vOs |
| 00:44.41 | j0 | virtuozo would work perfectly though for that |
| 00:44.43 | robeph | niekie: I imagine it'd depend on the codecs involved and bitrates n such? |
| 00:44.46 | robl^ | niekie: depends on codec |
| 00:44.52 | niekie | robeph: yeah. |
| 00:45.06 | j0 | niekie: http://www.packetizer.com/voip/diagnostics/bandcalc.html http://www.asteriskguru.com/tools/bandwidth_calculator.php |
| 00:45.26 | robeph | niekie: I mean it's pretty simple math, bitrate * (30*60) |
| 00:46.39 | robeph | roughly anyhow |
| 00:47.15 | JT | robeph: err, factor in voip overhead |
| 00:47.18 | robeph | yeh |
| 00:47.29 | robeph | why i said roughly ;p |
| 00:47.44 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-9132d7daafe648ae) |
| 00:47.50 | robeph | he said about, not exacting |
| 00:48.20 | *** part/#asterisk JayTee52 (n=jforde05@c-69-137-243-25.hsd1.in.comcast.net) |
| 00:48.26 | robl^ | okie -- I decided I am gonna get an AA50 for my test lab.. the decide if it will work for the install I will need to do in about 2 months |
| 00:48.59 | JT | robeph: well it's significant |
| 00:49.06 | *** join/#asterisk magic_hat (n=geoffdou@h-74-2-87-16.chcgilgm.covad.net) |
| 00:49.29 | robeph | hmm, I never really looked at it as bandwidth hasn't been an issue so far for me |
| 00:50.59 | magic_hat | hey everyone. Soliciting opinions on what the best setup is for an always-on * box... safe mode? checking with monit? Something I haven't thought of? |
| 00:51.30 | j0 | magic_hat: 2 * boxes? |
| 00:52.23 | Grnd-Wire | ok, thanks for the help everybody. I am unsure why, but the configuration just started working.. <ugh> |
| 00:52.39 | magic_hat | well, that may be the next step.... but I want to get the most production reliability I can out of one box before we get another. |
| 00:53.26 | j0 | magic_hat: well the easy things to change is raid1, and maybe dual psus |
| 00:53.36 | niekie | Hrmm.. according to my calculations the worst case scenario for a 30 minute call would be 35 megabytes then.. |
| 00:53.48 | niekie | That about right? |
| 00:53.56 | niekie | Or am I way off? |
| 00:55.17 | niekie | (Excluding SIP overhead) |
| 00:57.30 | JT | niekie: using what codec? |
| 00:58.34 | niekie | JT: g711 64Kibs |
| 00:59.31 | magic_hat | right, but should I be running in safe mode? |
| 01:00.52 | *** join/#asterisk Speedy2 (n=Javier_6@cpe-66-91-247-165.san.res.rr.com) |
| 01:01.40 | niekie | JT: according to my calculation, GSM would take about 13 MB for a 30 minute call. |
| 01:01.55 | niekie | With the http://www.asteriskguru.com/tools/bandwidth_calculator.php |
| 01:02.50 | niekie | Seems I don't have to worry about it, I have a 50GB limit, but rarely call with my system anyway. (Mainly use it for short notifications going out to phones, for e-mail) |
| 01:03.05 | [TK]D-Fender | niekie, 85kbps = 8.5Kbps * 60 * 30 = 15300kbytes = 14.94meg |
| 01:03.16 | [TK]D-Fender | niekie, math failure :) |
| 01:03.35 | [TK]D-Fender | niekie, ... per direction anyways ;) |
| 01:03.41 | niekie | [TK]D-Fender: eh? |
| 01:03.41 | robl^ | its a metric conversion math error ;-) |
| 01:04.07 | [TK]D-Fender | robl^, some are more equal than others :p |
| 01:04.18 | JT | [TK]D-Fender: 85kbps != 8.5kByte/s |
| 01:04.45 | [TK]D-Fender | JT, shhh! |
| 01:04.53 | [TK]D-Fender | JT, its NewMath (th) ;) |
| 01:05.00 | JT | 10.625 |
| 01:05.07 | JT | heh |
| 01:11.43 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
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| 01:35.29 | *** mode/#asterisk [+o file] by ChanServ |
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| 01:48.17 | *** mode/#asterisk [+o file] by ChanServ |
| 01:52.11 | *** part/#asterisk techie (n=techie@adsl-76-214-25-54.dsl.lsan03.sbcglobal.net) |
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| 01:57.08 | *** join/#asterisk smithj (n=smithj@foresight/developer/smithj) |
| 02:09.30 | CrazyTux | Hey guys say I want to disable notices from coming up in rasterisk / console, what would I Do? |
| 02:09.41 | grimsy | set verbose 0 |
| 02:09.52 | CrazyTux | grimsy, I want other errors, but not stupid notices. |
| 02:10.09 | CrazyTux | grimsy, i.e. I could care less about the registration attempts right now |
| 02:10.12 | grimsy | CrazyTux: core set verbose 2 |
| 02:10.16 | grimsy | or something like that |
| 02:10.19 | *** join/#asterisk salviadud (n=dude@131.178.46.207) |
| 02:10.25 | grimsy | it'll take a bit of playing around with to get the right level |
| 02:10.34 | salviadud | hi |
| 02:10.45 | salviadud | I'm trying to do a simple switch |
| 02:10.56 | salviadud | and I just forgot how the hell it works |
| 02:11.38 | salviadud | please tell me if this works |
| 02:12.11 | salviadud | If I try to dial out say... extension => 88,1,Goto(switcher) |
| 02:12.21 | salviadud | and on a context called switcher |
| 02:13.00 | salviadud | i got switch => IAX2/username:password@myserver/${EXTEN}@context |
| 02:14.54 | grimsy | salviadud: take out ${EXTEN} |
| 02:15.13 | grimsy | switch => IAX2/username:password@server/context |
| 02:15.28 | grimsy | switch already knows the exten |
| 02:15.53 | salviadud | doesn't it matter that it doesn't start with s, or 1? |
| 02:16.22 | grimsy | shouldn't. as long as you're catching anything that's unknown |
| 02:16.32 | grimsy | and make sure you're not switching back |
| 02:16.35 | grimsy | cause it will loop |
| 02:16.45 | salviadud | alright, thanx a lot |
| 02:16.49 | grimsy | np |
| 02:17.44 | craigk | is there a way to prove that asterisk is using ztdummy ... or to see what timing source it is using ? |
| 02:21.19 | grimsy | craigk: you could run "zap show status" and see if that tells you what you want |
| 02:21.38 | grimsy | i'm just loading up a vmware image to see what it says when running ztdummy |
| 02:21.44 | craigk | grimsy: legend - thanks |
| 02:21.53 | craigk | that reported: ZTDUMMY/1 (source: HRtimer) 1 |
| 02:22.03 | grimsy | ah, thats your answer then :P |
| 02:22.03 | craigk | so i will take that to mean it is using ztcummy :) |
| 02:22.07 | grimsy | yah |
| 02:22.21 | craigk | that means my choppy music on hold is being caused by something else :( |
| 02:22.22 | grimsy | otherwise it'll say something like Wildcard TDM400P REV I Board 1 |
| 02:24.17 | grimsy | is normal voice clear? |
| 02:24.36 | grimsy | and what format of moh u using? |
| 02:24.57 | craigk | i _think_ so ... i have not had a problem with voice in the past. I am just starting to try and use moh - but it is choppy :/ |
| 02:25.02 | salviadud | I can't get the switcher to work... |
| 02:25.08 | grimsy | :( |
| 02:25.11 | craigk | the format is wav - i am using the ones that came with asterisk at the moment |
| 02:25.13 | salviadud | hehe |
| 02:25.16 | salviadud | yeah i know, crap |
| 02:25.35 | salviadud | grimsy, is it better for me to call the switch another way? |
| 02:25.36 | grimsy | salviadud: do you really need switch? can't just use IAX trunk or dundi? |
| 02:25.46 | grimsy | i haven't really used switches before |
| 02:25.50 | grimsy | kinda been warned off them |
| 02:25.50 | salviadud | i've tried an IAX trunk |
| 02:26.01 | salviadud | but, when I dial, i get errors |
| 02:26.10 | salviadud | you see |
| 02:26.14 | salviadud | i got this asterisk box |
| 02:26.16 | grimsy | craigk: you could try loading various formats of moh. you might have more success |
| 02:26.17 | salviadud | that |
| 02:26.22 | grimsy | just do some tests |
| 02:26.23 | salviadud | when it loads up to 50 sip chans |
| 02:26.33 | salviadud | it sends number 51 to another box |
| 02:26.38 | salviadud | so, should I do a switch? |
| 02:26.39 | grimsy | could be things like problems transcoding etc |
| 02:26.43 | salviadud | or just an iax trunk? |
| 02:26.54 | grimsy | if you know the numbers that are on the other box, then just an iax trunk |
| 02:27.08 | grimsy | switch should only be used if you don't know what numbers are on the other side |
| 02:27.16 | salviadud | so, ian iax trunk would be something like |
| 02:27.17 | grimsy | iax trunk can also be done both ways |
| 02:27.28 | craigk | grimsy: :( that will take some time - but I guess it is where i am up to. I have tested an old trixbox (asterisk 1.2) installation and it works fine - but my 1.4.9 is choppy |
| 02:27.47 | salviadud | exten => 89,1,Dial(IAX2/username:password@server.com@context) ? |
| 02:28.01 | grimsy | craigk: yeah - is it a beefy box? |
| 02:29.06 | craigk | grimsy: 2.8GHz with 1GB RAM ... I am jsut testing on it so there are no other calls in progress |
| 02:29.08 | *** join/#asterisk |omni| (n=rob@c-67-185-144-206.hsd1.wa.comcast.net) |
| 02:29.10 | RypPn | craigk: If you add internal_timer = yes to sip.conf does it help any? |
| 02:29.34 | salviadud | um, guys, is an iax trunk like the stuff i put there? |
| 02:30.11 | grimsy | salviadud: i think that sounds ok... |
| 02:30.17 | grimsy | hang on and i'll look up my conf |
| 02:31.07 | craigk | RypPn: no, that did not help :/ |
| 02:31.36 | RypPn | pity, it worked for me when I used ztdummy |
| 02:31.44 | *** join/#asterisk UserReg_CL (n=COB@pc-248-68-47-190.cm.vtr.net) |
| 02:31.48 | UserReg_CL | mmmm |
| 02:31.51 | UserReg_CL | Buenas Noches |
| 02:32.23 | RypPn | grimsy: did you reload after changing? |
| 02:32.57 | UserReg_CL | helpme please.. |
| 02:33.09 | salviadud | what's up UserReg_CL ? |
| 02:33.10 | grimsy | RypPn: i didn't change anything :P think you meant craigk |
| 02:33.38 | RypPn | oops, yeah |
| 02:33.42 | *** join/#asterisk mordaunt (n=mordaunt@unaffiliated/mordaunt) |
| 02:33.53 | craigk | RypPn: and yes, I restarted asterisk completely after changing it :) |
| 02:34.21 | craigk | i grep'd the code of 1.4.9 and did not see any reference to internal_timer ... checking 1.4.14 now |
| 02:34.27 | grimsy | salviadud: sorry about that - took a while to grab the files |
| 02:34.38 | UserReg_CL | helpme... need make : make una call and answer... and after the 4 minutes finish call ? |
| 02:34.39 | salviadud | np |
| 02:34.45 | UserReg_CL | (sorry by english ) |
| 02:34.47 | grimsy | exten => _0[23478]XXXXXXXX,1,Dial(IAX2/user:pass@server/${EXTEN}) |
| 02:35.02 | salviadud | thanx grimsy |
| 02:35.06 | grimsy | np |
| 02:35.33 | UserReg_CL | please un example the dialplan |
| 02:35.37 | grimsy | and the other server needs to throw that to which ever context, then deal with the number on that end |
| 02:36.14 | grimsy | UserReg_CL: so you want a timer on the call that hangs up after 4 min? |
| 02:36.35 | UserReg_CL | yes grimps |
| 02:38.28 | CrazyTux | Hey guys, say I want to setup a "hunt group" i.e. an extension that automatically rings all users in the group? |
| 02:38.32 | CrazyTux | what would best practice be |
| 02:38.43 | grimsy | CrazyTux: queues |
| 02:38.48 | grimsy | ringall |
| 02:38.56 | UserReg_CL | I need to hang up a call after 4 minutes of talking (need one example) |
| 02:39.11 | grimsy | CrazyTux: http://www.voip-info.org/wiki/view/Asterisk+call+queues |
| 02:39.19 | CrazyTux | grimsy, thankyou |
| 02:39.22 | grimsy | np |
| 02:41.05 | grimsy | UserReg_CL: not sure sorry |
| 02:41.16 | UserReg_CL | helpme please.. |
| 02:41.21 | UserReg_CL | I need to hang up a call after 4 minutes of talking (need one example) |
| 02:41.22 | CrazyTux | grimsy, now context's are global, they are available accross all, files, correct? |
| 02:41.24 | grimsy | i'm sure there's a way to do it, but i haven't used something like that before |
| 02:42.14 | grimsy | CrazyTux: huh? |
| 02:42.31 | CrazyTux | grimsy, say I want to seperate 2 different companies, 1 asterisk box |
| 02:42.41 | *** join/#asterisk Qwell (i=north@pdpc/sponsor/digium/Qwell) |
| 02:42.41 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 02:42.52 | CrazyTux | grimsy, each having its own quesues/extensions, i.e. both Company A and B can have a "literal" extension 1000? |
| 02:43.00 | grimsy | ah |
| 02:43.13 | grimsy | then you want 2 separate contexts |
| 02:43.25 | grimsy | and no, contexts are not global, they have to be explicitly included |
| 02:43.34 | grimsy | you'll actually want more than 2 contexts |
| 02:43.41 | grimsy | but 2 parent contexts |
| 02:43.59 | CrazyTux | grimsy, so [company_a] [company_b] ... and? |
| 02:44.04 | grimsy | yah |
| 02:44.20 | grimsy | then you could have [incoming_a] incoming_b etc |
| 02:44.24 | grimsy | outgoing_a etc |
| 02:44.39 | UserReg_CL | Helpme Please, ... I need to hang up a call after 4 minutes of talking (need one example) |
| 02:44.41 | CrazyTux | grimsy, which incoming_a would map a REAL did to the company context? |
| 02:44.42 | grimsy | just to make it easier to read |
| 02:45.11 | [TK]D-Fender | UserReg_CL, "show function TIMEOUT" |
| 02:45.14 | grimsy | you'd only have one incoming_a |
| 02:45.26 | grimsy | sorry, gotta duck out for a bit |
| 02:45.29 | grimsy | good luck |
| 02:45.38 | CrazyTux | grimsy, really quick |
| 02:46.10 | CrazyTux | grimsy, say I had DID 1111111111, and 1111111112, and I wanted them to both point to extension 1000, I'd have to explicitly say that? |
| 02:46.26 | CrazyTux | exten => 1111111111,1,Goto(1000) ? |
| 02:46.40 | grimsy | you could pattern match it, so you only needed one line |
| 02:47.03 | grimsy | but otherwise you could goto(menu,s,1) or something to send to a common context |
| 02:47.08 | UserReg_CL | Fender: not is TIMEOUT... |
| 02:47.17 | grimsy | i'm sure [TK]D-Fender can help if you need it ;) |
| 02:47.19 | grimsy | cya |
| 02:47.25 | *** join/#asterisk file (n=file@asterisk/developer-and-muffin-lover/file) |
| 02:47.25 | *** mode/#asterisk [+o file] by ChanServ |
| 02:47.29 | CrazyTux | grimsy, thank you. |
| 02:47.38 | *** join/#asterisk UnixDog (n=unixdog@adsl-69-234-183-148.dsl.irvnca.pacbell.net) |
| 02:47.43 | UnixDog | [Nov 26 18:46:17] WARNING[669]: chan_sip.c:1775 __sip_xmit: sip_xmit of 0xbf2327fc (len 431) to 216.55.154.52:5060 returned -1: No buffer space available |
| 02:47.48 | UnixDog | what is this error |
| 02:47.55 | UnixDog | never seen before |
| 02:48.34 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 02:49.04 | craigk | RypPn: thank you ... I set internal_timing in the [options] section in asterisk.conf and it is now working ... thanks for the tip :) |
| 02:49.35 | RypPn | np |
| 02:50.48 | UserReg_CL | Helpme please... "I need the call to hangup after 4 minutes of initialization"... "automatically" |
| 02:52.00 | fujin_ | why would you do that? |
| 02:52.17 | file | Set(TIMEOUT(absolute)=240) will terminate the call after 4 minutes |
| 02:52.38 | RypPn | craigk: you also reminded me to remove mine now :) |
| 02:52.52 | craigk | NOOO - then it will stop working ;) |
| 02:53.21 | file | UnixDog: Google "sendto No buffer space available" |
| 02:53.26 | RypPn | lol, I bought a hardware analog card since |
| 02:53.59 | craigk | RypPn: ah - fair enough. i think my problem was i had a card then moved it to another box. I had to set up ztdummy and set that internal_timing option |
| 02:54.56 | *** join/#asterisk ZX81 (n=matt@202.49.106.158) |
| 02:55.01 | UserReg_CL | file: thank !!! |
| 02:55.17 | ZX81 | hi all - does anyone know if hpen can be used with Digium's b410p and mISDN? |
| 02:55.21 | ZX81 | *hpec |
| 02:55.37 | file | ZX81: I do not believe so |
| 02:55.45 | ZX81 | thought as much |
| 02:55.47 | ZX81 | ty |
| 02:55.50 | UserReg_CL | because: one call is max the 4 minutes the duration :) |
| 03:07.45 | *** join/#asterisk techie (n=techie@adsl-76-214-25-54.dsl.lsan03.sbcglobal.net) |
| 03:07.49 | *** join/#asterisk zenlogix (n=not@modemcable235.73-58-74.mc.videotron.ca) |
| 03:10.31 | zenlogix | Good evening evry one (well it's evening here anyways) - Im trying to play music with an AGI script while it's doing some ASYNC work but guess what! It's blocking until the file finishes to play. Any suggestions on how i could work around this? |
| 03:12.06 | CrazyTux | What do you guys advise, say I want to have two extension '1000's for company A / B, as far as the individual phone extension 1000 as well, so in other words Company A and Company B both have a phone extension 1000, is this possible? As far as phone SIP login credentials / etc, what must be done? |
| 03:13.02 | file | CrazyTux: sure it's possible... just use different contexts for different companies, and don't create SIP login credentials based on extension, use something like the MAC address |
| 03:13.43 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 03:14.44 | *** join/#asterisk tobias (n=tobias@nat1.ppckernel.org) |
| 03:15.14 | CrazyTux | file, how would I authenticate with * via mac? |
| 03:15.32 | CrazyTux | file, like, forget sip credentials now, just open network, for those specific macs? |
| 03:15.37 | file | the username is arbitrary, it can be whatever you want - it doesn't have to be their "extension"... |
| 03:15.40 | file | but you have to configure the phones |
| 03:16.27 | *** join/#asterisk PepOSX (n=pepOSX@190.72.145.179) |
| 03:16.45 | CrazyTux | file, no I mean literally now, authenticating via mac |
| 03:16.52 | CrazyTux | file, is this possible? |
| 03:17.02 | file | no, I didn't mean literally doing that |
| 03:17.11 | file | I just meant using the MAC address of each phone as it's username for SIP authentication |
| 03:17.21 | CrazyTux | file, well, I understood that, I just took it another level. |
| 03:18.16 | BBHoss | CrazyTux: how many users? |
| 03:19.04 | CrazyTux | BBHoss, well, this is a prototype, so a range of 0 - Nth |
| 03:22.27 | BBHoss | well there really isn't anything in SIP that allows for mac-only auth |
| 03:22.41 | BBHoss | now h323 does i think |
| 03:23.56 | BBHoss | the best way to do it would be to make the username the mac address |
| 03:25.18 | zenlogix | i guess i could do my playback music while waiting on ASYNC work thigny using 2 threads and EAGI but we get more complex that i would have liked.... |
| 03:25.30 | JT | BBHoss: that's pretty much what file said |
| 03:25.31 | BBHoss | im not sure if you can do this, but if you could get non-authenticated users to be able to call, then you could use iptables to limit the mac addresses that could connect to port 5060 |
| 03:25.58 | BBHoss | but MAC's are not unique and to use them for any real authentication would be foolish |
| 03:26.22 | JT | that would be a dumb idea if they're not on the same subnet |
| 03:26.26 | JT | segment even |
| 03:26.34 | BBHoss | no |
| 03:26.41 | BBHoss | to do it period would be dumb |
| 03:26.53 | *** join/#asterisk mmlj4 (n=jkelly@ip70-171-94-246.no.no.cox.net) |
| 03:26.55 | BBHoss | you could simply spoof your mac address, and gain auth |
| 03:26.58 | mmlj4 | j #nolug |
| 03:26.59 | JT | anyway, you're the one who suggested it :P |
| 03:27.20 | BBHoss | yeah, for anything other than REAL situations :) |
| 03:27.28 | *** join/#asterisk xxoxx (n=xxoxx@tor/regular/xxoxx) |
| 03:31.37 | BBHoss | best way to do auth is with md5 hashes, or with IAX2 use RSA |
| 03:34.49 | JT | yeah you're answering the question no-one asked |
| 03:39.34 | CrazyTux | when setting up a sip user |
| 03:39.38 | CrazyTux | whats the point of username= |
| 03:39.50 | CrazyTux | when the [username] is the username? or whats the differences. |
| 03:40.13 | file | username is for outgoing authentication, ie you place an outgoing call to a SIP provider and need to authenticate using a username and password |
| 03:50.15 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 03:52.54 | *** join/#asterisk roberted (n=roberted@209.59.110.37) |
| 04:01.28 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:03.23 | *** join/#asterisk bmg505 (n=leon@196.209.179.149) |
| 04:08.43 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 04:16.19 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.140.104) |
| 04:28.14 | *** join/#asterisk bl4q (n=Bl@dslb-088-067-025-155.pools.arcor-ip.net) |
| 04:30.20 | *** join/#asterisk implicit (n=implicit@ip68-105-92-210.sd.sd.cox.net) |
| 04:30.52 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 04:36.54 | *** join/#asterisk truz_`24 (n=truz_`24@74-128-148-184.dhcp.insightbb.com) |
| 04:41.07 | CrazyTux | One other question I have |
| 04:41.19 | CrazyTux | if I edit, sip.conf, queues.conf, extensions.conf, etc. does it all compile into one dialplan? |
| 04:41.41 | CrazyTux | like for example could I place extensions inside of sip.conf, if I wanted to? and could I place queues inside of extensions if I wanted to? |
| 04:41.53 | CrazyTux | or for that matter, in "some_random_included_file' |
| 04:42.47 | file | CrazyTux: ...no, each file configures a specific module of Asterisk, sip.conf does not configure dialplan logic |
| 04:43.12 | CrazyTux | file, so the answer is the same to quesues.conf and extensions.conf? |
| 04:43.25 | file | yes, but you can include other files in using #include |
| 04:43.47 | CrazyTux | file, so, say I have Company A and B, and now each has its own queue, and its own context, and set of extensions, etc. |
| 04:44.02 | CrazyTux | file, what would be the most proper way to generally structure this? |
| 04:46.25 | file | that is up to he who implements it... you could have a directory that contains company specific configs, and do something like #include "companies/extensions_*.conf" and #include "companies/sip_*.conf" |
| 04:46.28 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:46.50 | CrazyTux | file, so I could use wildcard inclusions? |
| 04:46.55 | CrazyTux | file, to include multiple files at once? |
| 04:46.57 | file | sure |
| 04:47.21 | file | context names just can't be the same across them... or else you might run into unexpected behavior |
| 04:47.41 | CrazyTux | file, so heres another question, |
| 04:47.54 | CrazyTux | file, contexts are for extensions, queues, and pretty much everything else? |
| 04:48.24 | file | a context is a term given to the way we structure configuration files |
| 04:48.29 | file | have you read the book? |
| 04:48.56 | CrazyTux | file, I come from a strong programming background, I own the oreilly book, but it seems far to basic. |
| 04:49.28 | CrazyTux | file, I'd rather dabble around, in * than read that book. I understand contexts for the most part, it's essentially like namespaces. |
| 04:49.31 | CrazyTux | file, somewhat |
| 04:49.59 | file | sure, variables/extensions belong to a specific context which separates them from others |
| 04:51.43 | CrazyTux | file, what about recursive inclusion? |
| 04:51.47 | CrazyTux | file, behind directories |
| 04:52.03 | file | example? |
| 04:52.46 | CrazyTux | file, i.e. say I had a structure such as companies/[company_username]/queues.conf, now I want to load all of these queues.conf for each user. |
| 04:53.11 | file | I've never done that, so dunno |
| 04:53.56 | mosty | #include "companies/*/queues.conf" |
| 04:54.05 | mosty | that's untested, but try it |
| 04:54.10 | CrazyTux | mosty, that would work? lol, lets give it a go. |
| 04:54.13 | CrazyTux | alrighty. |
| 04:55.40 | mosty | i use wildcards in #include's, it seems to work for me, but i've only ever used the wildcard to match files, not directories |
| 04:55.51 | CrazyTux | yea, that seemed to work |
| 04:59.15 | CrazyTux | hi |
| 04:59.24 | CrazyTux | i love soraya |
| 04:59.36 | *** join/#asterisk mackes (n=root@cpe-76-180-239-199.buffalo.res.rr.com) |
| 04:59.57 | CrazyTux | lol, apparently my girlfriend wishes to say Hi, to everyone :) and express her love for herslef |
| 05:00.07 | CrazyTux | s/herslef/herself/ |
| 05:00.16 | BBHoss | heh |
| 05:00.39 | mackes | weird |
| 05:01.58 | craigk | when an extension places a call on hold, is there some sort of timeout that i can set so the extension placing it on hold is dialed ? |
| 05:02.19 | craigk | otherwise the person placing the call on hold might forget :/ |
| 05:03.54 | mosty | their phone should have a flashing light or similar |
| 05:03.56 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 05:04.26 | craigk | so there is only a visual indicator ? |
| 05:04.46 | *** join/#asterisk grimsy (n=chatzill@203-206-220-214.perm.iinet.net.au) |
| 05:04.53 | mosty | craigk, that would depend on the phone |
| 05:05.40 | craigk | mosty: by that, do you mean to imply that asterisk does not take any action in this area ? |
| 05:06.04 | mosty | in the most common setup, it's handled by the phone |
| 05:06.52 | craigk | ok, thanks - i will stop looking for it then :) |
| 05:07.46 | mosty | you could craft together a workaround, but i doubt the effort would be worth it |
| 05:08.33 | craigk | maybe i will ..... one day when i do not have better things to do :) |
| 05:10.29 | craigk | i have the impression that there is such a capability with call parking/queuing ... i guess i just assumed it would be there for call hold as well :/ |
| 05:16.33 | [TK]D-Fender | craigk, If you think you might trake a while, you SHOULD park it rather than put it on hold... |
| 05:17.26 | craigk | a fair enough comment ... but my phone does not have a 'park' button, just a 'hold' button so my natural lazy streak means i would just put it on hold :) |
| 05:17.55 | jql | get a better phone. </snark> |
| 05:18.01 | BBHoss | craigk: now if you parked the call, you can set a timeout |
| 05:18.27 | craigk | lol - yes i need a better phone and yes if i park it i get more control :) |
| 05:18.46 | [TK]D-Fender | craigk, the cure for laziness is summary execution :) |
| 05:19.37 | craigk | so - is there a way to break into the default hold processing to park it instead ? :). It seems that hold processing is done outside of the context of extensions.conf - so i can not say 'park it instead of holding it' |
| 05:19.53 | craigk | [TK]D_Fender: or promotion ;) |
| 05:22.03 | CrazyTux | I should be able to, register a user, and simply dial that user by Dial(SIP/user, context) right? |
| 05:22.26 | mosty | you don't put a context in the dial command |
| 05:22.46 | CrazyTux | mosty, so simply SIP/username |
| 05:22.59 | mosty | yes, but probably also a timeout |
| 05:23.23 | CrazyTux | whats the default timeout |
| 05:23.39 | mosty | there probably is no timeout by default |
| 05:24.38 | CrazyTux | mosty, ok, now in console it says "called test-1000" |
| 05:24.43 | CrazyTux | mosty, however the phone never rings. |
| 05:24.47 | CrazyTux | mosty, its definitely registered |
| 05:24.51 | CrazyTux | ok |
| 05:24.53 | CrazyTux | err |
| 05:26.02 | [TK]D-Fender | CrazyTux, ....read the book..... |
| 05:28.00 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 05:32.48 | *** join/#asterisk CCFL_Man2 (i=CCFL_Man@argon.pureshells.com) |
| 05:33.14 | *** join/#asterisk hfb (n=hfb@75.80.37.175) |
| 05:36.08 | iCEBrkr | [Nov 26 00:35:33] WARNING[3403]: chan_sip.c:12132 handle_response_register: Got 200 OK on REGISTER that isn't a register |
| 05:36.13 | iCEBrkr | umm |
| 05:36.43 | iCEBrkr | That's nice, can't you at least tell me a bit more detail, like an IP? |
| 05:36.46 | iCEBrkr | :-\ |
| 05:48.58 | *** join/#asterisk ghento (n=ghento@d216-232-170-111.bchsia.telus.net) |
| 05:56.27 | *** join/#asterisk salviadud (n=root@189.156.170.169) |
| 05:56.52 | salviadud | hi |
| 05:56.56 | salviadud | i'm so screwed |
| 05:57.03 | salviadud | i can't get a simple iax trunk to work |
| 05:57.20 | salviadud | is there an example i can follow besides the obvious one from voip-info.org? |
| 05:57.26 | *** join/#asterisk xoqa (i=46dc9878@gateway/web/cgi-irc/ircatwork.com/x-3e861bd16d4deda4) |
| 05:58.49 | xoqa | ahoy! is anyone aware of a free voip service which allows free pc-to-landline calls? |
| 05:59.03 | JT | only to 1800 numbers |
| 05:59.08 | salviadud | all you guys want is free stuff |
| 05:59.34 | salviadud | i don't blame you |
| 05:59.39 | salviadud | i get dish network for free |
| 06:00.05 | JT | speak for yourself :P |
| 06:00.07 | jql | short of hacking someone's (google's) click-to-dial? naw |
| 06:00.09 | JT | i don't expect shit for free |
| 06:01.15 | jql | everyone I know has to pay something for pstn termination. only a rich fool can eat that for you. *looks at skype* |
| 06:01.34 | JT | heh |
| 06:01.37 | JT | skype isn't free |
| 06:01.42 | jql | it was |
| 06:01.47 | jql | for 6 months post-ebay |
| 06:01.51 | JT | i see |
| 06:02.06 | jql | but, even they couldn't keep it up forever |
| 06:05.07 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 06:10.29 | salviadud | what do I need to add in my iax.conf so i can see it register? |
| 06:10.40 | salviadud | kind of like how SIP does it |
| 06:12.49 | salviadud | is it |
| 06:12.54 | salviadud | qualify=yes ? |
| 06:15.19 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 06:15.34 | grimsy | salviadud: try this tutorial http://dumbme.voipeye.com.au/trixbox/how_to_interconnect_2_boxes.htm |
| 06:15.49 | grimsy | try method 3 first, as it's simple |
| 06:15.51 | JT | trixbox :( |
| 06:15.52 | grimsy | though not secure |
| 06:15.55 | grimsy | i know |
| 06:15.56 | salviadud | thanx grimsy |
| 06:16.05 | grimsy | but tutorial isn't specific |
| 06:39.34 | BadHorsie | wow, DiaglogPalette looks awesome |
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| 07:14.12 | salviadud | exit |
| 07:14.26 | salviadud | stop now |
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| 07:18.27 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 07:23.33 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 07:27.55 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@216-110-94-252.static.twtelecom.net) |
| 07:28.09 | *** join/#asterisk snook3r (n=snook3r@bzq-219-46-202.isdn.bezeqint.net) |
| 07:28.34 | zeeesh | getting problem with voicemail conf .. will anybody help ? |
| 07:30.25 | BadHorsie | .toast |
| 07:30.27 | BadHorsie | ~toast |
| 07:30.28 | jbot | extra, extra, read all about it, toast is a foreign substance entered through one of the bodys many holes that can be used as an energy source. A man once said that toast is great with marmite - but you either love it or you hate it, personally i prefer cheese on my toast, red leicester cheese...... my mouth be watering....... |
| 07:31.17 | CrazyTux[m] | Hi guys, so I asked earlier, about this, I guess the proper term is "ring group". What would be the proper way to ring all extensions associated to a specific extension, should I use queues, or simply DIAL(SIP/usera&SIP/userb&SIP/userc) etc. |
| 07:34.46 | tzafrir | zeeesh, care to be more specific? |
| 07:36.59 | CrazyTux[m] | tzafrir, me? |
| 07:37.21 | CrazyTux[m] | tzafrir, ah nevermind :) |
| 07:40.26 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 07:41.17 | tzafrir | CrazyTux[m], what exactly do you want to do? The dial command you put here will ring all of the SIP users until one of them answers |
| 07:41.59 | CrazyTux[m] | tzafrir, which I'm aware of, however, a call queue will do the same, or? What is the proper way to do this? |
| 07:42.56 | tzafrir | CrazyTux[m], TIMTOWTDT |
| 07:43.12 | CrazyTux[m] | tzafrir, what is that |
| 07:43.35 | tzafrir | gee, you never used perl ;-) |
| 07:45.27 | tzafrir | ~timtowtdt |
| 07:47.05 | CrazyTux[m] | tzafrir, oh. |
| 07:47.10 | CrazyTux[m] | tzafrir, which I understand, as well. |
| 07:47.11 | tzafrir | ~timtowtdt |
| 07:47.12 | jbot | timtowtdt is probably There Is More Than One Way To Do That. Though my way is usually the right one. |
| 07:47.19 | CrazyTux[m] | tzafrir, however I'm saying, which way is the best / most proper. |
| 07:57.38 | *** part/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 07:58.13 | *** join/#asterisk zeeesh (n=zeeesh@202.166.161.45) |
| 08:04.15 | mosty | use queues if you want to use agents, that is phones that can dial a code to join or leave a particular call |
| 08:05.14 | mosty | if you just want a pre-definied group of phones to ring for a particular extension, just use a simple dial command |
| 08:07.33 | mosty | er, agents join/leave a particular queue (not call) |
| 08:08.01 | *** join/#asterisk jarod14 (n=jarod14@ns2.viatelecom.com) |
| 08:21.03 | CrazyTux[m] | mosty, yea I don't want joining or leaving, I want consistent ringing |
| 08:21.07 | CrazyTux[m] | accross all members of group |
| 08:21.38 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 08:22.10 | mosty | how often will the group change? |
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| 08:39.14 | mort_gib | Does anyone know what PCI cards I need in order to connect Asterisk to 3 ISDN2 lines?? |
| 08:39.30 | mort_gib | Digium or other?? |
| 08:41.12 | mosty | get a quad span PRI card, either digium or sangoma. my recommendation is the sangoma card with hardware echo cancellation |
| 08:41.25 | mort_gib | Model number?? |
| 08:41.39 | mosty | just look it up on their websites |
| 08:41.52 | mort_gib | -Sorry I'm kinda new to Asterisk, so I have POTS up and working fine... |
| 08:42.11 | mort_gib | But all their "££*^&*() model number tells me nothing :-) |
| 08:42.25 | mosty | you have to actually read the product description |
| 08:42.38 | mort_gib | This system is to work with BT style ISDN |
| 08:42.50 | mosty | is it PRI or BRI? |
| 08:43.21 | mort_gib | I would make it BRI, as PRI is not sold in a competitive way down here! |
| 08:44.23 | mosty | then you will need to do some research, as BRI is not well supported in asterisk |
| 08:44.36 | mort_gib | :-( I didn't know that! |
| 08:44.53 | mort_gib | I was under the impression that it would work just fine!! |
| 08:45.29 | mort_gib | As in, I have seen configs that claim to be working with BRI interfaces. |
| 08:45.34 | mosty | it will if you buy the right card. unfortunately i doubt the cards that work here (australia) will work with british BRI |
| 08:46.13 | mort_gib | That's why I'm asking rather than just reading on Digium's website. |
| 08:46.45 | mort_gib | Australia! Summer there now! |
| 08:48.37 | mosty | not quite, but getting there |
| 08:48.51 | JT | mosty: they work fine. |
| 08:49.02 | JT | ETSI... ETSI... I fail to see the issue |
| 08:49.34 | mosty | ok well digium only has one BRI card, B410P |
| 08:50.20 | JT | uses terrible misdn only unfortunately |
| 08:51.19 | mosty | i have two BRI/misdn setups, no real issues to speak of. of course i prefer to work with PRI |
| 08:51.46 | JT | i have nothing good to say about misdn apart from "it might work, sometimes" |
| 08:52.04 | Speedy2 | Anyone here using Asterisk on NetBSD? |
| 08:52.44 | mosty | sangoma have a BRI card, but it's not available in australia yet. when that's available it will probably be the best BRI card to use here (assuming it uses wanpipe) |
| 08:53.07 | JT | probably |
| 08:55.04 | mosty | it does use wanpipe, according to their website |
| 08:55.07 | mort_gib | Great guys! I found that card too (B410P) |
| 08:55.22 | JT | i know it uses wanpipe |
| 08:56.51 | mort_gib | I think I might be able to get the Sangoma card -A500 |
| 08:57.06 | mort_gib | www.voipon.co.uk seems to have them in stock! |
| 08:58.21 | mort_gib | <PROTECTED> |
| 09:23.47 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-e347d1c51b197ab6) |
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| 09:38.35 | Bladerunner05 | Hola, I need to user CDR to mysql in addons package. I make addons and then I need to recompile asterisk ? I have to run configure (asterisk) with some parameters? |
| 09:39.16 | mosty | no need to recompile asterisk |
| 09:42.00 | Bladerunner05 | ok |
| 09:42.15 | Speedy2 | Bladerunner05: But you must kill Tyrell |
| 09:42.44 | Bladerunner05 | <Speedy2>: ? what ? |
| 09:43.06 | Speedy2 | Bladerunner05 : You figure ito ut |
| 09:48.18 | *** join/#asterisk scream_ (n=stas@80.81.57.44) |
| 09:49.04 | tzafrir | mort_gib, do you really want to run Frame Relay over BRI? |
| 09:49.24 | mort_gib | -who said I wanted to do that?? |
| 09:49.27 | tzafrir | Anyway, you can run PPP and such over any zaptel digital card, IIRC |
| 09:49.47 | tzafrir | you seem to have quoted that as a feature of wanpipe |
| 09:49.58 | mort_gib | Yes, another user asked! |
| 09:51.05 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 09:51.19 | mort_gib | I have 3X ISDN2 lines deliveerd by the local telco, I would also prerfer PRI, but that is not viable (price) |
| 09:53.35 | *** join/#asterisk Buhntz (n=bytewalk@port-212-202-41-252.dynamic.qsc.de) |
| 09:54.07 | JT | not even for a fractional pri? |
| 09:55.50 | mort_gib | well, for political reasons, they only sell half (15 lines) E1's and the install price!!!!!! |
| 09:56.58 | mort_gib | The very reason I'm into Asterisk, the local telco also offers a PBX, that is overpriced and under featuerd! |
| 09:57.18 | BBHoss | what did you expect? :) |
| 09:57.24 | mort_gib | Then we can get Avaya.. |
| 09:57.50 | mort_gib | Nothing much, my boyscout days are over ;-) |
| 09:57.57 | BBHoss | BUT! You can pay only $500 for 10 voice mailboxes!!!!!! lolzorz |
| 09:58.39 | mort_gib | Exactly! and SIP -eh, uhm that's with the VOIP package, that will cost you some £450 extra... |
| 09:59.00 | BBHoss | shit its more than that |
| 09:59.15 | mort_gib | i don't know, I don't sell Avaya... |
| 09:59.18 | mort_gib | :-) |
| 09:59.18 | BBHoss | the local telco here wanted 20 grand for some nortel |
| 09:59.31 | BBHoss | just to connect two offices over IP |
| 09:59.57 | mort_gib | Yes, but these are all small installs, They don't even offer VOIP (the local telco) |
| 10:01.03 | BBHoss | well i think they were going to run it over point-to-point PRI lines then push SIP IP traffic over that, so they could get even more cash :) |
| 10:01.29 | *** join/#asterisk sergey (n=sergey@91.189.233.71) |
| 10:02.19 | mort_gib | no, just ISDN lines for incoming calls DID and outgoing local calls. -No VOIP provider can do that more cost efficient than the local telco (Monopoly) |
| 10:03.24 | mort_gib | My only fear is hardware though... I read the books, and they all warn me about hardware |
| 10:03.39 | BBHoss | which hardware, the telephony? |
| 10:03.47 | mort_gib | recommendations?? No pc/server |
| 10:04.00 | BBHoss | for one ISDN line? |
| 10:04.13 | BBHoss | 3 |
| 10:04.35 | BBHoss | i dont see why a basic p4 system couldnt run that |
| 10:04.39 | mort_gib | no 12-16 users ISDN lines, mix of SIP and analog handsets, 8 Sip's |
| 10:04.57 | mort_gib | well I have to quote under £6500... |
| 10:04.58 | BBHoss | ok so now you have 12-16 ISDN lines? |
| 10:05.19 | BBHoss | you'll have a hell of a time with that |
| 10:05.28 | mort_gib | Nope, they only need some 6 concurrent lines (ew did the study using their call ercords) |
| 10:05.37 | mort_gib | why?? |
| 10:05.44 | BBHoss | especially having that many IRQ interrupts |
| 10:05.55 | JT | so that's 3 BRIs |
| 10:05.58 | JT | not 12 to 16 |
| 10:06.12 | mort_gib | Nope, 12-16 I would use PRI |
| 10:06.16 | BBHoss | ok, i was about to say :) |
| 10:06.27 | mort_gib | :-) |
| 10:06.51 | mort_gib | So hardware recommendation?? |
| 10:06.59 | BBHoss | sangoma has got digium beat on the BRI cards |
| 10:07.14 | BBHoss | but for pc hardware |
| 10:07.16 | zeeesh | getting problem with voicemail.conf.. i hv configured voicemail . i can easily hear voice msgs by using sip client .. i want to receive these voicemail in my hotmail or yahoo account.. in voicemail.conf in general section where "attach=yes" but still fail to receive any msg in my hotmail or yahoo account? |
| 10:07.36 | BBHoss | if you want room for growth, i would get a good mobo and maybe a core2duo |
| 10:07.44 | mort_gib | I gather that form this list and the feedback from my vendor... |
| 10:07.47 | JT | what the hell |
| 10:07.51 | JT | get a proper server |
| 10:07.55 | JT | not this core2duo junk |
| 10:08.23 | mort_gib | Budget is some £800, but most servers comes with PCi Experss |
| 10:08.23 | BBHoss | ok spend 60% more for xeons |
| 10:08.29 | mosty | zeeesh, can you successfully send mail from the command line? |
| 10:09.02 | BBHoss | mort_gib: also, i would get the hardware DSP echo cancellation to keep load off of the cpu |
| 10:09.04 | Frogzoo | mosty: sure, hundreds of ways |
| 10:09.36 | mosty | Frogzoo, of course, but i want zeeesh to test their mta |
| 10:09.46 | zeeesh | <mosty>: i can hear my voicemail by using any xlite... but i m unable to send these wave files to any hotmail or yahoo account? |
| 10:09.58 | Frogzoo | mosty: oh, telnet host 25 |
| 10:10.01 | *** part/#asterisk scream_ (n=stas@80.81.57.44) |
| 10:10.05 | JT | people still use hotmail and yahoo? :o |
| 10:10.11 | BBHoss | heh |
| 10:10.28 | mort_gib | Yes, that was high up on my list. Really I'm just trying to make sure, 100% sure that the project is a success, I have many more clients in the pipeline!! |
| 10:10.29 | mosty | zeeesh, is your mail server configured? |
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| 10:11.05 | zeeesh | <mosty>: no |
| 10:11.08 | mort_gib | Zeeesh> Hotmail/Yahoo might not accept emails form Dynamic IP ranges and IP's that don't resolv... |
| 10:11.10 | JT | mort_gib: you want to be 100% sure it's a success by cheaping out? |
| 10:11.30 | tzafrir | For that reason I use Yahoo as my default search engine |
| 10:11.35 | mosty | zeeesh, then that's your problem. of course asterisk can't send email |
| 10:11.41 | zeeesh | <mosty>:although my sendmail service is running |
| 10:11.51 | JT | yawn, pointless exercise, using second rate search engines, tzafrir |
| 10:12.13 | mort_gib | JT, no quite the opposite! By spending what is requierd! |
| 10:12.17 | tzafrir | JT, do you actually have any idea how not to rely on google? |
| 10:12.18 | mosty | zeeesh, well what do sendmail's logs say? |
| 10:12.27 | JT | tzafrir: sure |
| 10:12.34 | BBHoss | actually yahoo was rated more accurate than google, but i still hate it because its not accurate to me |
| 10:12.52 | JT | rated by crackpots ;) |
| 10:13.00 | tzafrir | The most annoying thing about them is that they don't give real links in the search results |
| 10:13.15 | JT | what are you talking about tzafrir ? |
| 10:13.26 | tzafrir | search.yahoo.com |
| 10:13.42 | tzafrir | And for that reason I never bother quoting their search results |
| 10:14.13 | JT | mort_gib: you just said there were low budgets |
| 10:14.48 | mort_gib | -Eh, did I?? I said I have to quote a 14 user install less than £6500 |
| 10:14.55 | BBHoss | JT: what processor would you suggest if not a core2duo? They are inexpensive and are plenty fast enough for 3 BRIs |
| 10:15.07 | mort_gib | That SHOULD be possible -right?? |
| 10:15.22 | BBHoss | mort_gib: that including labor? |
| 10:15.28 | mort_gib | Is the processor so important, I would have said the buss?? |
| 10:15.32 | JT | BBHoss: speed is irrelevant |
| 10:15.40 | JT | reliability is important |
| 10:15.42 | mort_gib | -Yes that includes my work |
| 10:15.48 | mort_gib | JT> Yes!!! |
| 10:15.55 | BBHoss | JT: so you mean like a Xeon? |
| 10:15.56 | tzafrir | mort_gib, for that ammount of users? not really. But maybe you need to do some other things |
| 10:16.11 | JT | it's better to get a slower second hand server grade machine than a new core2duo |
| 10:16.16 | JT | yes a xeon |
| 10:16.48 | JT | server grade most importantly |
| 10:16.55 | BBHoss | well the new xeons are the same cores as core2duo |
| 10:17.05 | JT | so what |
| 10:17.10 | JT | i'm not talking about the cores |
| 10:17.11 | mort_gib | JT> the client wants rackmount anyway, so I don't see any point in trying to fit a Desktop into rails ;-) |
| 10:17.40 | JT | heh |
| 10:17.47 | mort_gib | I have a working system here, with their specs, it took me some 6 hours (first install) |
| 10:18.39 | mort_gib | But with only two analog lines, so how much time would you spend on a small system like that?? |
| 10:19.00 | zeeesh | <mosty>: i hv safed voicemails in /var/spool/asterisk/voicemail/zeeesh/inbox/ .. how shud it connect with sendmail ? |
| 10:19.28 | mosty | zeeesh, you put the email address in voicemail.conf and it should email them |
| 10:20.18 | JT | heh |
| 10:22.07 | zeeesh | <mosty>: i hv mention in voicemail.conf my email address mentioned as new aliases .. will it work or still i hv missing something .. ? |
| 10:22.47 | mosty | do you see asterisk's attempts to send mail in your sendmail logs? |
| 10:23.18 | JT | mort_gib: it should be doable |
| 10:23.30 | mort_gib | :-) I think so too! |
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| 10:24.00 | zeeesh | <mosty>: let me check .. |
| 10:24.05 | mort_gib | Remember that I work in GBP £ |
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| 10:54.45 | lumoon | hi, i'm a new asterisk user and I'm desperated.... I have to configure simply a voicemail but I received always this notice/warnig: |
| 10:55.11 | tzafrir | anybody here uses rx_fax and tx_fax? I'm trying to send a simple tiff file from txfax to rxfax directly |
| 10:55.19 | tzafrir | And the whole thing seems to take forever |
| 10:55.38 | mosty | i do all my faxing with hylafax |
| 10:55.42 | tzafrir | Can anybody provide me a simple smoke test to show that those apps really work? |
| 10:55.51 | lumoon | "messagecount: IMAP user not set" |
| 10:56.20 | lumoon | "Can not leave voicemail" |
| 10:57.02 | lumoon | in extension I have this line: exten => _X.,n,VoiceMail(u41@default) |
| 10:57.13 | mosty | lumoon, http://www.voip-info.org/wiki/view/Asterisk+Voicemail+IMAP+storage |
| 10:59.12 | lumoon | ---uhm... but I must have a IMAP server even if I'd like simply to register the message (no mail)? |
| 10:59.37 | mosty | no, you can store voicemail on the local fs i believe |
| 11:00.51 | mosty | can you paste your voicemail.conf on a paste site? |
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| 11:01.38 | lumoon | I leave the voicemail.conf as it is... simply add "41 => 41, Mailbox lu" |
| 11:02.37 | tzafrir | lumoon, you want to use imap storage of voicemail? |
| 11:02.39 | lumoon | I've just cheked the permissions in /var/spool/asterisk/voicemail/default/41 |
| 11:02.52 | tzafrir | There are other voicemail storages |
| 11:03.38 | lumoon | no, I simply want to register message and use VoiceMailMan to hear the message |
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| 11:04.09 | lumoon | ...maybe.. for this... I've to use imap storage? |
| 11:04.19 | mosty | lumoon, show us your voicemail.conf, paste it to a paste web site |
| 11:04.41 | xoqa | ahoy mates. anyone aware of a free voip service which allows for free pc-to-landline calls? |
| 11:04.58 | mosty | xoqa, no such service exists |
| 11:05.53 | xoqa | mosty, that's what i thought :| |
| 11:06.00 | lumoon | ok... now I try to paste it... just a moment.. thanks |
| 11:06.11 | xoqa | i've found some near free services, but nothing completely free |
| 11:06.53 | mosty | xoqa, there's a reason for that |
| 11:07.19 | xoqa | because these people have to pay for services like ourselves? :) |
| 11:07.59 | mosty | no telephone company in the world is going to let you make calls for free |
| 11:08.47 | mosty | so if you want to connect to the regular old telephone network, you are going to have to pay |
| 11:11.20 | lumoon | my voicemail.conf... http://paste.turbogears.org/paste/1780 |
| 11:13.45 | lumoon | and asterisk log: |
| 11:13.56 | lumoon | [Nov 26 12:13:07] WARNING[2673]: app_voicemail.c:2383 messagecount: IMAP user not set for mailbox 41 |
| 11:13.57 | lumoon | [Nov 26 12:13:07] NOTICE[2673]: app_voicemail.c:2957 leave_voicemail: Can not leave voicemail, unable to count messages |
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| 11:24.14 | lumoon | any hint? |
| 11:25.07 | mosty | can you set verbose and debug to 10, does it give any more info? |
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| 11:27.18 | flujan | hi all. |
| 11:27.31 | *** join/#asterisk J4zen (n=Jvan4zen@a213-84-139-244.adsl.xs4all.nl) |
| 11:27.33 | flujan | I upgraded my system to version 1.4.14 now I am having issues with the load of asterisk |
| 11:27.45 | flujan | asterisk is consuming about 200% of CPU. Showed in top. |
| 11:28.01 | flujan | It starts to consume it from nothing... I am seeing no errors at the full log. |
| 11:28.06 | flujan | Someone with the same problem? |
| 11:28.26 | lumoon | mosty... unfortunately no more info.. :-( |
| 11:28.30 | mosty | flujan, anything in the logs? |
| 11:28.32 | J4zen | by chance, is anyone currently using a windows 2003 DHCP server that leases IP-adresses to VOIP-phones? |
| 11:28.40 | J4zen | or any Windows server that sends out DHCP to VOIP phones? |
| 11:28.48 | flujan | mosty: nops... |
| 11:29.49 | mosty | J4zen, why? |
| 11:30.09 | mosty | flujan, try disabling modules, see if you can narrow it down to something specific |
| 11:30.30 | flujan | mosty: do you mean using menuselect? |
| 11:30.45 | mosty | no, i mean in modules.conf (no need to recompile) |
| 11:30.46 | flujan | how can I disable modules I am not using or better... see which modules I am using. |
| 11:30.54 | mosty | "show modules" |
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| 11:35.49 | lumoon | can I register message without imap server? |
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| 11:37.05 | mosty | lumoon, you should be able to, but i haven't tested it in asterisk 1.4 |
| 11:38.37 | lumoon | in your opinion... the best solution is installing an imap server? |
| 11:39.11 | J4zen | mosty: I'd like to know if i could automatically assign a specific range of IP adresses to SNOM phones requesting DHCP. This would be possible if Joker signs were allowed ( All SNOM320 phones start with mac 000412, i could filter on that ) |
| 11:39.20 | tzafrir | lumoon, the simplest solution is not to use the imap storage support, but rather, the built-in voicemail storage |
| 11:40.01 | tzafrir | Maybe not as featureful, but much closer to "Just Works" |
| 11:40.13 | mosty | J4zen, just get a barcode reader and scan in all the MAC addresses for your phones |
| 11:40.17 | lumoon | and the voicemail storage is setting by default or I have to set something? |
| 11:40.51 | J4zen | mosty; MAC adresses are stored in the barcode? |
| 11:41.28 | mosty | J4zen, i think they are- on the outside of the snom boxes. they have the mac address written directly underneath a barcode... |
| 11:41.39 | lumoon | tzafrir, I don't set imap server but I can't leave message!! |
| 11:42.08 | J4zen | mosty: Thanks, ill look into that |
| 11:42.18 | J4zen | Other than that, no clue how to filter on a software-level? |
| 11:42.19 | tzafrir | lumoon, you can build app_voicemail instead with the local storage support |
| 11:42.20 | lumoon | where can I tell asterisk to use built-in storage? |
| 11:43.08 | mosty | J4zen, i could probably write a script to do it in unix, but windows- no idea |
| 11:44.09 | mosty | tzafrir, er, you have to pick imap or old style voicemail at compile time? |
| 11:45.19 | tzafrir | mosty, IIRC |
| 11:45.19 | lumoon | ...do you know if deb packages are set with imap by default? |
| 11:46.01 | tzafrir | lumoon, in the debs they used a nice trick to build app_voicemail with the three possible backends in three different modules |
| 11:46.14 | tzafrir | You have to make sure only one of them is loaded |
| 11:48.33 | mosty | tzafrir, nasty... what was the trick? (so i can incorporate it into my etch backport debs) |
| 11:48.44 | lumoon | great!...but I have no idea where I can view which is loaded.... sorry... :-( |
| 11:49.59 | mosty | lumoon, which distribution are you using? |
| 11:50.22 | tzafrir | mosty, http://svn.debian.org/wsvn/pkg-voip/asterisk/trunk/debian/patches/ , look at hack-multiple-app-voicemail |
| 11:50.49 | tzafrir | mosty, etch backports are provided there |
| 11:51.18 | tzafrir | mosty, check debian/backports/etch |
| 11:51.32 | tzafrir | (and debian/backports/etch-xorcom , maybe) |
| 11:51.40 | lumoon | mixed... debian etch / lenny / unstable... asterisk 1.4 |
| 11:51.52 | tzafrir | but the latter will remove imap altogether |
| 11:52.19 | tzafrir | the backports hook intended to make the package build cleanly under said distro |
| 11:53.49 | mosty | i have it being built cleanly already, i just wasn't aware that selecting imap voicemail would prevent the old style voicemail from working |
| 11:53.51 | hi365 | hey tzafrir! where did you find the externpassnotify option (for voice mail)?? it a. doesnt seem to be working (and b. i havnt seen it in the code, although i suck at c) |
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| 11:58.25 | hi365 | hmm, it doesnt seem to be in 1.4.11 what version is r47814 included in? |
| 11:59.40 | lumoon | tzafrir, I have /usr/lib/asterisk/app_voicemail.so... this means something? |
| 12:00.59 | mosty | lumoon, in /etc/asterisk/modules.conf make sure you have noload => app_voicemail_imap.so and the same for app_voicemail_odbc.so, but load => app_voicemail.so |
| 12:03.33 | lumoon | I don't have nothing like this... if I add it... what can I do then? |
| 12:03.59 | lumoon | ...or what I have to do then... :-) |
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| 12:09.16 | mosty | apparently debian (sid) provides three different app_voicemail modules, and you can only load one |
| 12:09.34 | mosty | so load the basic one, if you don't want imap or odbc voicemail |
| 12:12.21 | lumoon | mosty, but if I have only app_voicemail.so in /usr/lib/modules/asterisk/ .... it means asterisk loaded only this module... I'm wrong? |
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| 12:13.19 | mosty | lumoon, well you are running a bastardized debian box, i have no experience with testing at the moment, maybe that's why |
| 12:18.04 | lumoon | I thought it was simple... all books explain voicemail configuration in some few rows... as it is automatic... I think there's a big and stupid error in my configuration because it's impossible otherwise... :-( |
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| 12:18.58 | mosty | lumoon, it appears that the version of asterisk that you're running was compiled with only support for imap voicemail |
| 12:20.05 | lumoon | ...so I have to set a imap server or re-build asterisk... |
| 12:21.19 | mosty | lumoon, the version of asterisk in sid has a workaround, you do not need to rebuild asterisk |
| 12:24.10 | lumoon | mosty... which workaround? |
| 12:24.43 | mosty | it creates three app_voicemailXXX.so modules, you have to load the desired one in modules.conf |
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| 12:28.16 | rati | hi , when i am calling from from one SIP to other SIP soft phone it not displaying no , i want dispaly no |
| 12:28.36 | fujin | DISPLAY NO?? |
| 12:28.57 | fujin | Do both devices have callerid configured in Asterisk? |
| 12:30.07 | rati | i am not geting can give detail idea |
| 12:30.26 | fujin | you can get die in a fire |
| 12:30.41 | rati | shall i show u my dial paln and sip.conf |
| 12:30.50 | fujin | no |
| 12:33.21 | McDouglas | i have a strange error with my SPA phones. If i press XFER button the caller hears MOH till i finish the attended transfer, but if i press the BXFER button, the caller still hears what i say in the handset until i input the extension number and press dial. |
| 12:34.30 | rati | when i am calling arati(8000) user to 2000 , it displyas arati only i want dispaly 8000 in sted of name |
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| 12:41.58 | rati | <fujin> http://pastebin.ca/797918 |
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| 12:50.57 | rati | when i am calling from arati(8000) user to 2000 , it displyas arati only i want dispaly 8000 insted of name please gone through http://pastebin.ca/797918 |
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| 12:59.20 | McDouglas | one of my user want me to disable incoming call notification while already speaking with someone. Is it possible to do with asterisk? (or i have to look for some options on the device?) |
| 13:00.33 | mosty | i believe that's done on the device |
| 13:00.52 | mosty | snom phones have an option for that |
| 13:01.33 | tzafrir | agx, any chance you can help me with my silly rxfax/txfax issue? |
| 13:01.49 | tzafrir | http://pastebin.ca/797932 - I just can't send a fax from txfax to rxfax |
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| 13:02.44 | tzafrir | I must be missing something pretty basic. But then again, some RTFM would be nice |
| 13:03.35 | McDouglas | mosty: any chance you know what is the name of this feature so i can look for it? |
| 13:04.51 | mosty | McDouglas, on snom phones it's called "call waiting indication" or CWI or something similar |
| 13:05.10 | McDouglas | too bad i have d-link atas and linksys spa-s :\ |
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| 13:09.41 | Mw3 | tzafrir: txfax to rxfax does not work for me either |
| 13:10.50 | Mw3 | tzafrir: i read somewhere it doesnt even supposed to work. (but i dont know the reason) |
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| 13:11.15 | tzafrir | agx said it works for him |
| 13:11.47 | Mw3 | but we had the same issue as you, channels up, but no rtp |
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| 13:12.58 | Mw3 | and receiving from pstn was not too good. about 40-50% success rate. (with iaxmodem+hylafax almos every fax is ok). but this is another case |
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| 13:15.32 | badcfe | in a dialplan, if more extentions match, will i guaranteed get the first one mentioned in that context that matches? |
| 13:17.18 | Mw3 | no, the most specific will match first as i know |
| 13:17.48 | badcfe | oh wow |
| 13:18.10 | badcfe | there is actually a "best match" aproach when ising _ exten expressions .. ? |
| 13:18.12 | tzafrir | Mw3, It doesn't send *at all* |
| 13:18.37 | tzafrir | I must be missing something pretty basic |
| 13:18.39 | mosty | badcfe, use include => to include contexts, asterisk will check in the order of the include's |
| 13:19.15 | badcfe | that is if i want to force order as mentioned as toplevel and then go best-matching ok thanks. |
| 13:19.39 | badcfe | well i actually wants overall best match but thanks |
| 13:20.38 | coppice | tzafrir: is the app even starting up? |
| 13:20.38 | mosty | badcfe, include in order of decreasing prefix length |
| 13:20.46 | tzafrir | coppice, it starts up, and both rx_fax and tx_fax are in the main loop |
| 13:20.55 | tzafrir | now adding more debug prints... |
| 13:20.56 | Mw3 | tzafrir: but the channel is up, and txfax/rxfax is running. running for infinity ... |
| 13:21.01 | Mw3 | tzafrir: until soft hangup |
| 13:21.15 | coppice | tzafrir: is this over IP or PSTN? |
| 13:21.16 | tzafrir | yeah, just that |
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| 13:21.29 | tzafrir | coppice, directly from txfax to rxfax |
| 13:21.36 | Mw3 | we tried over sip and over iax |
| 13:21.37 | tzafrir | just a basic smoke test |
| 13:21.49 | tzafrir | After that I want to test it over PSTN a bit |
| 13:22.03 | coppice | oh, that doesn't work with the apps for asterisk. they expect incoming data to drive their timing |
| 13:22.33 | tzafrir | ok, so on to next stage (over zaptel) |
| 13:23.01 | coppice | the apps in callweaver will switch to a fudgy generator approach when the PSTN doesn't drive the timing |
| 13:23.17 | McDouglas | http://forums.whirlpool.net.au/forum-replies-archive.cfm/688681.html -> one of the posts says its possible to disable call waiting for one extension. How can i do it with asterisk (not tb) |
| 13:28.47 | mosty | McDouglas, set call-limit for the sip peer |
| 13:31.05 | rati | how to create data base table in mysqual |
| 13:31.30 | rati | now only i install the asterisk-addons |
| 13:32.11 | fujin | create table `name` |
| 13:34.56 | rati | command is not found in asterisk server , what to do ?? |
| 13:35.15 | fujin | apt-get install brain && done |
| 13:35.22 | fujin | ensure brain is connected before operating keyboard |
| 13:35.37 | fujin | other dependancies, i.e.; linux knowledge |
| 13:35.42 | fujin | may also require satisfaction |
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| 13:36.14 | jbot | sends a legion of lawyers after rati's head, courtesy of royk |
| 13:36.35 | fujin | ^5 royk |
| 13:37.01 | robl^ | hahah |
| 13:37.37 | [TK]D-Fender | "There can be only one!" |
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| 13:42.52 | Lasse123 | Could anyone inform me if the follow is a bug or some kind of twisted feature (do not want developers time by bugreporting something that may be known behaviour): |
| 13:43.00 | Lasse123 | We have 3 players in the drama: |
| 13:43.04 | Lasse123 | 1) SIP Phone client that supports only iLBC |
| 13:43.09 | Lasse123 | 2) Asterisk configured to support alaw, ilbc in that order (sip.conf). |
| 13:43.13 | Lasse123 | 3) PSTN provider supports all kinds of codecs |
| 13:43.17 | Lasse123 | Call from PSTN to sip phone works fine (PSTN -> alaw -> Asterisk -> ilbc -> SIP client) |
| 13:43.22 | Lasse123 | BUT if the order of the codecs in sip.conf is ilbc, alaw asterisk will send ilbc to PSTN provider and alaw to SIP client (allthough it indicates it only support ilbc in SDP) resulting in a failed call. |
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| 13:47.12 | [TK]D-Fender | Lasse123: pastebin your sip.conf masking only passwords |
| 13:47.14 | [TK]D-Fender | ~pb |
| 13:47.15 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines 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 |
| 13:47.17 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^ |
| 13:49.16 | McDouglas | i have a strange error with my SPA phones. If i press XFER button the caller hears MOH till i finish the attended transfer, but if i press the BXFER button, the caller still hears what i say in the handset until i input the extension number and press dial. |
| 13:50.06 | Tourinho | good day guys.. is there a way to reduce PLAYBACK delay when playing 2 files? |
| 13:50.47 | robl^ | McDouglas: it's a problem with the Linksys phones, not Asterisk. |
| 13:52.47 | [TK]D-Fender | Tourinho: What delay? |
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| 13:56.00 | fujin | McDouglas: that's not a problem with the phones |
| 13:56.02 | fujin | that's how it works. |
| 13:56.13 | fujin | read the manual that came with your phones |
| 13:56.13 | McDouglas | nice |
| 13:56.20 | fujin | bxfer (dial number) press dial = blind transfer |
| 13:56.26 | fujin | axfer initiates a 'hold' |
| 13:56.36 | robl^ | quick question -- when provisioning an IAXy, do you assign the server as an IP or can you use a FQDN? |
| 13:58.10 | [TK]D-Fender | robl^: IP only |
| 13:58.20 | [TK]D-Fender | robl^: IAXYZ doesn't do DNS |
| 13:59.44 | fujin | McDouglas: what's the error? :) |
| 14:00.04 | robl^ | [TK]D-Fender: ahh!! ok. That may cause slight issues then. do you know of any ATA that uses DNS? |
| 14:00.18 | [TK]D-Fender | robl^: All the Sipura/Linksys ones. |
| 14:00.38 | coppice | are here any ATAs which won't use DNS? |
| 14:00.53 | *** join/#asterisk _ys (i=yuri@91.151.196.254) |
| 14:00.59 | russellb | they probably all do it but the iaxy, heh |
| 14:01.22 | McDouglas | fujin: no error, its just annoying for the caller to hear the buttons (kinda loud) |
| 14:01.58 | Tourinho | [TK]D-Fender: I need to play 2 files. I would like to play then one after other.. but it takes some time to play second file.. is there a way to reduce this time, or to send 2 files to PLAYBACK? |
| 14:02.39 | *** join/#asterisk tobias (n=tobias@fvr.lightlink.com) |
| 14:03.36 | robl^ | coppice: hehe. good question. I haven't used any ATAs yet. all have been IP phones or analog phones hanging of a digium FXS card. But I may well need to put up 2 ATAs in remote locations. Unfortunately the server could possibly change IP addresses and get to the ATAs after deployment will be very difficult. They would have to be mailed back to me |
| 14:03.47 | [TK]D-Fender | Tourinho: There is no pause between files being played back unlessw its part of their audio. |
| 14:04.13 | [TK]D-Fender | McDouglas: Oh well, you're just going to have to live with it. |
| 14:04.30 | McDouglas | [TK]D-Fender: i could, but if the boss doesnt like it... |
| 14:04.59 | [TK]D-Fender | McDouglas: then buy a phone that doesn't do that then. |
| 14:05.14 | McDouglas | my more pressing problem atm is the lack of abbility to make call back work with attended transfers... old pbx could do it, asterisk cant |
| 14:05.29 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:05.58 | robl^ | McDouglas: Asterisk can, Linksys phones can't |
| 14:06.04 | [TK]D-Fender | McDouglas: Didn't we cover this yesterday? |
| 14:06.16 | McDouglas | [TK]D-Fender: yesterday? nope, but yes we did talka bout it |
| 14:06.26 | fujin | McDouglas: so mute first? |
| 14:06.33 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:06.36 | McDouglas | fujin: good idea :P |
| 14:06.37 | [TK]D-Fender | Why do you need callback when its an ATTENDED transfer? You never let the call go.... |
| 14:06.58 | McDouglas | [TK]D-Fender: i told you, our users dont use attended transfer the way you supposed to |
| 14:07.18 | fujin | what're you trying to do? |
| 14:07.29 | *** join/#asterisk e` (n=e@38.102.196.202) |
| 14:07.31 | McDouglas | they initiate the attneded transfer and they hang up |
| 14:07.39 | [TK]D-Fender | McDouglas: If they don't follow the simple instructions then they get what they deserve. This is very deep in TFB territory. |
| 14:07.40 | McDouglas | making it basicly a blind transfer |
| 14:07.50 | fujin | then why not do a blind transfer? |
| 14:07.57 | fujin | I'm lazy, I always do blind transfers. |
| 14:08.10 | McDouglas | [TK]D-Fender: i was instructed to mimic the behavior of the old pbx |
| 14:08.37 | fujin | McDouglas: you're doing it wrong! |
| 14:08.38 | fujin | try harder |
| 14:08.40 | [TK]D-Fender | McDouglas: Just wait till they start asking you to jump off bridges because your other friends did it too... |
| 14:08.41 | fujin | back to the drawing bard |
| 14:08.43 | fujin | etc |
| 14:09.22 | fujin | so, why do they use attended transfer to do blind transfers |
| 14:10.23 | [TK]D-Fender | fujin : because they can't let go of the "order of buttons" they were used to and fail to follow simple instructions. |
| 14:10.24 | McDouglas | fujin: old analog phones only had one button to make transfers |
| 14:10.43 | McDouglas | and that one button initiated an attended transfer |
| 14:10.44 | fujin | so they can't understand the difference between 'xfer' and 'bxfer' on their devices? |
| 14:10.50 | McDouglas | they probably can |
| 14:11.04 | fujin | It's a shame that the bxfer button is virtual and you have to scroll to the right to see it on the SPA's |
| 14:11.06 | McDouglas | but you have to press right on the navigation button once to get the bxfer softkey |
| 14:11.08 | fujin | That's one of my major gripes with em. |
| 14:11.09 | fujin | Yeah. |
| 14:11.11 | [TK]D-Fender | "but I was told to do something unreasonable so I'm going to beat it to death to no avail" |
| 14:11.29 | coppice | virtual buttons are good - no wear and tear issues :-) |
| 14:11.33 | fujin | That's true |
| 14:11.45 | fujin | McDouglas: sell linksys, buy polycom? |
| 14:11.49 | fujin | && profit? |
| 14:11.51 | [TK]D-Fender | McDouglas: Pass this back : TFB. You can ask as often as you want, but unless you;re directing it to Linksys and throwing a lot of money their way I don't they'll care. |
| 14:12.41 | robl^ | Asterisk + Linksys phones is NOT your old PBX. Microsoft Word is NOT Word Perfect. You can customize them bot to have similarities, but it is not possible to make them behave identically. It will involve some retraining. Otherwise -- why did you switch PBXes to begin with? |
| 14:12.45 | [TK]D-Fender | fujin : Polycom won't save them... they are still trying to pass of an attended transfer and have it treated like a blind transfer. No phone will do that. |
| 14:12.53 | McDouglas | fujin: no distributor in hungary, and they are kinda expensive |
| 14:13.39 | fujin | o_0 |
| 14:13.43 | fujin | give up and die |
| 14:13.45 | fujin | go back to your old PBX |
| 14:14.23 | coppice | I can almost see polycom's office from here, and they are still bloody expensive |
| 14:14.35 | fujin | Blah |
| 14:14.38 | fujin | IP330's are very cheap |
| 14:14.42 | fujin | same price as spa942 here |
| 14:14.45 | fujin | 250~nzd |
| 14:15.03 | McDouglas | 921s were the top i could make them purchase :P |
| 14:15.19 | fujin | damn |
| 14:15.20 | fujin | that's shit |
| 14:15.24 | fujin | those are the one-line ones, right?? |
| 14:15.34 | McDouglas | yes, we have only one lines/desktop |
| 14:15.56 | robl^ | McDouglas: tell them you did the best possible given the budget constraints they imposed. its THEIR fault ;-) |
| 14:16.17 | McDouglas | well, i guess thats gonna happen |
| 14:16.27 | *** join/#asterisk riddlebox (n=james@75-128-170-26.static.stls.mo.charter.com) |
| 14:16.43 | robl^ | what PBX did you have before? |
| 14:17.43 | McDouglas | bosch integral 3 |
| 14:17.44 | [TK]D-Fender | McDouglas: Forget the phons. What you want to do will not work on ANY phone I've ever heard of. |
| 14:18.09 | McDouglas | [TK]D-Fender: it did with bosch+analog phones |
| 14:18.40 | mosty | McDouglas, you could use features.conf and do something like dial *1 for atxfer, *2 for blindxfer |
| 14:18.49 | [TK]D-Fender | McDouglas: So keep the SPA's, they're fine. Your users are messed up in the head. Follow the instructions of face the consequences. |
| 14:19.07 | *** join/#asterisk Strom_M (n=strom@ip70-170-60-8.lv.lv.cox.net) |
| 14:19.10 | [TK]D-Fender | mosty: "but..but .... that's not how the OLD system worked! *wah*" |
| 14:21.19 | robl^ | "Do you want your calls to go not get lost!??! Press the buttons in THIS order before you hangup." |
| 14:24.21 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 14:25.00 | *** join/#asterisk shafique (i=cbd7b0ba@gateway/web/cgi-irc/ircatwork.com/x-67e5e6c6e39acbd7) |
| 14:25.31 | shafique | hi |
| 14:25.36 | shafique | anybody there? |
| 14:25.41 | shafique | i need a bit help |
| 14:26.37 | shafique | i am running Asterisk 1.2.16 on my gentoo box |
| 14:27.16 | [TK]D-Fender | .... |
| 14:27.30 | shafique | last time i have worked on this box with my seniors.. |
| 14:27.31 | [TK]D-Fender | shafique: feel free to speed along to your actual question.... |
| 14:27.36 | *** join/#asterisk fskrotzki (n=fskrot@host198.textwise.com) |
| 14:28.04 | shafique | but this time i have installed it by myself... now i am coming to question |
| 14:28.45 | endre | ~book |
| 14:28.45 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 14:29.08 | shafique | recordings of all calls i made are saved in /var/spool/asterisk/monitor/current |
| 14:30.30 | shafique | in past they all automatically moved after 24 hours to specific folder like if there is call record of agent a it will be saved in /monitor/2007/11/26/a/. |
| 14:30.51 | shafique | but after this installation they are not moving automatically |
| 14:31.08 | shafique | can anybody help me how can i do them move automatically? |
| 14:31.59 | [TK]D-Fender | ^^^^^^^^^^^ |
| 14:32.20 | [TK]D-Fender | shafique: this has nothing to do with Asterisk, but rather some sort of script you are going to have to make yourself. |
| 14:32.38 | robl^ | shafique: I don't think asterisk does that "move". It was likely a custom script written by the "senior" and running via cron |
| 14:32.46 | [TK]D-Fender | shafique: Cron is pretty clearly a good place to start, and I'm betting this is a BASH job.... |
| 14:32.52 | shafique | hmmm ok |
| 14:33.14 | shafique | then i should better learn to make such script? |
| 14:33.19 | shafique | :) |
| 14:34.02 | [TK]D-Fender | shafique: Yes. |
| 14:34.17 | [TK]D-Fender | shafique: feel free to go visit #bash |
| 14:34.32 | shafique | hmmm |
| 14:34.35 | robl^ | shafique: the script could be about 3-4 lines.. very simple. just read up on BASH |
| 14:34.38 | shafique | okay |
| 14:35.13 | shafique | [TK]D-Fender and robl^ thank for your Help |
| 14:35.21 | robl^ | shafique: or... if you have remote access to the other boxes installed with your senior, look for the script and just copy it |
| 14:35.23 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 14:35.49 | shafique | hmmm |
| 14:36.30 | shafique | there are 2 or 3 directories with cron jobs... like cron.daily,,, cron.weekly,,, cron.hourly blah blah |
| 14:36.58 | shafique | it was a daily job bcoz it run after every 24 hours... |
| 14:36.58 | robl^ | shafique: right. you said it copied them like once an hour.. so likely inside of cron.hourly |
| 14:37.16 | robl^ | ahh.. yeah.. if its every 24 hours, cron.daily |
| 14:38.47 | shafique | ok |
| 14:38.54 | robl^ | any scripts placed in cron.daily, will run once every day. so just look in that directory for a script that copies things from the monitor directory. |
| 14:39.12 | shafique | i am logged in on the remote system in our second office...let me copy these cron jobs :P |
| 14:39.26 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 14:39.50 | *** join/#asterisk mackes (n=root@65-121-253-83.dia.static.qwest.net) |
| 14:40.03 | robl^ | don't copy ALL of them.. just the ones you need. others may mess things up.. it dpends on what customizations and versions of software they use. you only want the one script |
| 14:42.16 | *** join/#asterisk ai-a (n=jake2@megan.healthnet.co.uk) |
| 14:43.12 | JerJer | mmm cronjobs |
| 14:43.15 | shafique | yes i understand |
| 14:43.55 | robl^ | hey JerJer. good morning to ya |
| 14:45.16 | *** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-2450276ed395e944) |
| 14:46.19 | JerJer | howdie - its more like afternoon already for me, been up since like 3am :) |
| 14:47.39 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 14:47.55 | anonymouz666 | JerJer: too much TCP reuse problems? :) |
| 14:48.21 | JerJer | nope, that project is done and solved. |
| 14:48.24 | ai-a | Any sangama / wanpipe / e1 sacr programmers here ? |
| 14:48.32 | anonymouz666 | JerJer: glad to hear. |
| 14:49.01 | JerJer | so was the customer - they kept calling it the holy grail |
| 14:50.13 | anonymouz666 | heh |
| 14:50.27 | JerJer | all i care about is OpenSER+Asterisk once again replaced some old ass lucent SIP switch |
| 14:50.53 | anonymouz666 | it's a great combo |
| 14:51.24 | JerJer | the next fun project I might be working on is a new age TTY/TDD system for a large inbound call center |
| 14:52.01 | anonymouz666 | I don't know what TTY/TDD is. |
| 14:52.02 | JerJer | caller with at TDD machine calls into a queue and gets text based prompts telling them they are in a queue, etc |
| 14:52.21 | JerJer | teletype - its how those without hearing make phone calls |
| 14:52.34 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 14:52.59 | anonymouz666 | ah ok |
| 14:53.08 | anonymouz666 | lunch time |
| 14:53.10 | anonymouz666 | bbl. |
| 14:53.41 | JerJer | the 'fun' part about the project is they don't want their agents to require a TDD machine themselves - they would like the text to show up on their 'screen pop' application |
| 14:53.51 | *** join/#asterisk jiboneus (n=jibone@60.51.108.104) |
| 14:58.31 | *** join/#asterisk duckz (n=duckz@85.204.47.228) |
| 14:59.22 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net) |
| 15:06.21 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:06.22 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 15:10.59 | *** join/#asterisk Champi (i=Champi@rootshell.fr) |
| 15:13.51 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 15:13.51 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:14.54 | niekie | Anyone got experience with "VoipBuster Pro"? |
| 15:15.03 | De_Mon | sounds retarded |
| 15:15.21 | niekie | De_Mon: it's not about how the name sounds :P |
| 15:15.41 | niekie | Apparently you get "1 year of free calls for 10 euro's" |
| 15:15.47 | *** join/#asterisk fskrotzki (n=fskrot@host198.textwise.com) |
| 15:15.55 | niekie | Max 300/min a week. |
| 15:16.45 | *** join/#asterisk hfb (n=hfb@pool-71-106-219-180.lsanca.dsl-w.verizon.net) |
| 15:18.14 | *** join/#asterisk clandmeter (n=Carlo@81.175.82.2) |
| 15:18.29 | niekie | Apparently there's a call set-up fee though. |
| 15:21.59 | *** join/#asterisk muiro (n=muiro@cpe-76-189-84-108.neo.res.rr.com) |
| 15:22.57 | *** join/#asterisk axisys (i=iqbala@outbound.silenceisdefeat.org) |
| 15:26.40 | *** join/#asterisk fskrotzki (n=fskrot@host198.textwise.com) |
| 15:27.07 | *** join/#asterisk jameswf (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 15:27.25 | *** join/#asterisk csmiga (n=chsmiga@68-114-1-65.dhcp.gsvl.ga.charter.com) |
| 15:27.47 | *** join/#asterisk fskrotzki (n=fskrot@host198.textwise.com) |
| 15:36.32 | *** part/#asterisk lumoon (n=lu@81-174-11-207.static.ngi.it) |
| 15:36.38 | *** part/#asterisk shafique (i=cbd7b0ba@gateway/web/cgi-irc/ircatwork.com/x-67e5e6c6e39acbd7) |
| 15:40.06 | niekie | Meh, I doubt I'll get it. |
| 15:50.25 | *** join/#asterisk dioedu (n=dioedu@201.7.117.114) |
| 15:50.41 | dioedu | hello... i have this message in my console... Rotated Logs Per SIGXFSZ (Exceeded file size limit) |
| 15:51.07 | dioedu | and asterisk is creating many files of queue_log, event_log and messages... |
| 15:51.58 | dioedu | does someone know what can I do to stop this ? |
| 15:53.32 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 15:54.12 | *** join/#asterisk RoyK (n=roy@62.92.9.241) |
| 15:54.31 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:54.56 | *** join/#asterisk jmesquita (n=jmesquit@201.7.117.114) |
| 15:55.59 | *** join/#asterisk fedya (n=fedya__@rrcs-71-43-242-130.se.biz.rr.com) |
| 15:56.18 | *** join/#asterisk Cyon (n=cyon@216.179.31.170) |
| 15:57.34 | *** join/#asterisk pifiu (n=someone@216.5.79.1) |
| 15:59.24 | *** join/#asterisk ZenBSDi (n=zenbsdi@unaffiliated/ZenBSDi) |
| 16:02.28 | fedya | if i have 10 phones all on the same network, i'm getting dropped calls, i just saw somewhere that they should all be using different local sip ports, will this fix the problem? |
| 16:03.03 | *** join/#asterisk j0 (n=dan@S01060016b6b541d2.va.shawcable.net) |
| 16:04.13 | pifiu | does anyone know how I can make 2 phones in 2 locations ring at once from one number? |
| 16:04.57 | dioedu | ok. Problem resolved... Both files, Master.csv, was very bigger... 2.0 GB... |
| 16:05.05 | *** join/#asterisk saftsack (n=saftsack@pD9E05758.dip.t-dialin.net) |
| 16:05.08 | mvanbaak | pifiu: Dial(SIP/1&SIP/2) |
| 16:05.17 | pifiu | right now I have an exten=>incomingdidnumber,4,Dial(IAX2/context/phonenumber1,60,r) |
| 16:05.25 | pifiu | i want to add another number |
| 16:09.11 | DrAk0 | [Nov 26 17:07:22] NOTICE[2387]: chan_iax2.c:2848 __auto_congest: Auto-congesting call due to slow response |
| 16:09.19 | blitzrage | pifiu: do as mvanbaak just suggested |
| 16:09.27 | blitzrage | plus you do not need 'r' |
| 16:09.48 | blitzrage | if the phone is ringing, the signalling will tell asterisk to generate ringing instead of generating it all the time (even if the other end isn't ringing) |
| 16:10.14 | *** join/#asterisk yannj_fr (n=yannj_fr@APuteaux-152-1-30-204.w82-120.abo.wanadoo.fr) |
| 16:11.39 | saftsack | does someone know where i can get the opal version from 21may2007? |
| 16:11.46 | fedya | we're getting a lot of dropped calls, how do i go about troubleshooting this? |
| 16:17.06 | *** join/#asterisk ussrback (n=MAX@80.92.183.84) |
| 16:17.26 | *** join/#asterisk russellb (i=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 16:17.27 | *** mode/#asterisk [+o russellb] by ChanServ |
| 16:18.38 | *** join/#asterisk magic_hat (n=geoffdou@h-74-2-87-16.chcgilgm.covad.net) |
| 16:19.47 | ussrback | Hi all. I have installed asterisk-addons. I want to use h323. addon automatically installed chan_ooh323.so in my modules directory. when i ran "modules load chan_ooh323", it gave me error about missing config file. i havent any ooh323.conf file. where can i get this file? |
| 16:20.17 | *** join/#asterisk c0rnflake (n=c0rnflak@38.112.4.210) |
| 16:20.44 | *** join/#asterisk ghento (n=ghento@d216-232-170-111.bchsia.telus.net) |
| 16:21.07 | magic_hat | hey everyone. I'm having problems w/ my * box... the setup itself works fine, but 45-60 minutes after boot * stops answering calls and the box begins rejecting ssh connections. ideas? |
| 16:21.45 | *** join/#asterisk ManxPower (n=manxpowe@225.sub-70-220-92.myvzw.com) |
| 16:22.13 | *** join/#asterisk csmiga (n=chsmiga@68-114-1-65.dhcp.gsvl.ga.charter.com) |
| 16:22.37 | ussrback | magic_hat: in your /var/log s |
| 16:23.05 | ussrback | if the problem is in ssh it might be OS prob first |
| 16:23.57 | tzafrir | magic_hat, does it answer pings? |
| 16:23.58 | magic_hat | which log am I looking at in /var/log? |
| 16:24.05 | magic_hat | tzafrir: yes |
| 16:24.17 | tzafrir | are you connected to it? |
| 16:24.50 | magic_hat | yes, it accepts registrations, etc. But when the greeter plays on inbound calls, it basically just locks up |
| 16:25.26 | tzafrir | magic_hat, disk issues? |
| 16:25.47 | magic_hat | I was getting a soft lock error, but that's not the case any more. |
| 16:25.52 | tzafrir | magic_hat, connect with ssh -v . Where does it hang? |
| 16:26.17 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:26.37 | magic_hat | tzafrir: I'll try that when the problem starts cropping up again. right now it's fine cuz I just rebooted. |
| 16:28.06 | *** join/#asterisk felix_da_catz (n=felix@PixNationalGroup.phonoscope.com) |
| 16:29.55 | [TK]D-Fender | ussrback: Go look in your addons source folder. If you spent more than 2 minutes searching, you'd have found it .... |
| 16:31.06 | ussrback | [TK]D-Fender: i was looking for it :) but may be i am a bad google :P |
| 16:31.27 | *** join/#asterisk pifiu-laptop (n=someone@216.5.79.1) |
| 16:31.41 | [TK]D-Fender | ussrback: I said in the SOURCE folder, not GOOGLE. |
| 16:31.55 | [TK]D-Fender | ussrback: You clearly aren't trying... |
| 16:31.57 | ussrback | i just get this config from some forum |
| 16:31.57 | ussrback | yes i know |
| 16:32.27 | coppice | [TK]D-Fender: most of the people here can be very trying |
| 16:32.46 | [TK]D-Fender | coppice: Funny thing is their only target seems to be our patience :) |
| 16:35.41 | *** join/#asterisk RoyK (n=roy@gprs-ggsn5-nat.mobil.telenor.no) |
| 16:36.17 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@216-110-94-252.static.twtelecom.net) |
| 16:37.04 | *** join/#asterisk csmiga (n=chsmiga@68-114-1-65.dhcp.gsvl.ga.charter.com) |
| 16:37.15 | ussrback | [TK]D-Fender: there is onli h323.conf.sample in asterisk-ooh323c directory |
| 16:37.28 | [TK]D-Fender | ussrback: \o/ |
| 16:37.29 | ussrback | should i cp this file as ooh323.conf ? |
| 16:37.40 | [TK]D-Fender | ~cluebat ussrback |
| 16:38.25 | [TK]D-Fender | ussrback: .... |
| 16:38.48 | ussrback | ??? |
| 16:38.55 | ussrback | http://pastebin.ca/798128 |
| 16:40.52 | *** join/#asterisk tripps (n=sean@72.20.150.196) |
| 16:42.08 | [TK]D-Fender | ussrback: what do YOU think you're supposed to do with "h323.conf.sample" ?! |
| 16:42.09 | *** join/#asterisk jmesquita (n=jmesquit@201.7.117.114) |
| 16:42.45 | ussrback | copy as ooh323.conf in /etc/asterisk directory |
| 16:42.50 | ussrback | and configure this file |
| 16:43.19 | [TK]D-Fender | ussrback: where to you get this brilliant idea to COMPLETELY rename the file instead of dropping the ".sample" ADD-ON ONLY? |
| 16:44.47 | ussrback | ??????? hm so can u please explain me, what i have to do in this case. cause there is not any ooh323.conf in my sources? |
| 16:45.55 | russellb | maybe ... it's ... just supposed to be called h323.conf ? |
| 16:46.23 | russellb | it would be pretty silly to have an h323.conf.sample for something that is expected to be named ooh323.conf ... |
| 16:46.39 | robl^ | russellb: it couldn't be THAT simple. ;-) |
| 16:46.42 | coppice | has ooh323 been abandoned? I keep hoping a really good pure C H.323 will appear, but it seems to be getting less and less likely |
| 16:46.50 | ussrback | whet i make load module chan_ooh323 is looking for ooh323.conf not h323.conf |
| 16:47.29 | russellb | coppice: there is nobody currently maintaining it |
| 16:47.45 | *** join/#asterisk yannj_fr (n=yannj_fr@APuteaux-152-1-34-103.w82-120.abo.wanadoo.fr) |
| 16:47.46 | russellb | but it's still there ... |
| 16:47.49 | ussrback | when i compiled addons it automaticaly put chan_ooh323.so module in my modules directory |
| 16:47.53 | coppice | so that implies abandoned? |
| 16:48.01 | ussrback | yes h323.conf is still ther |
| 16:48.07 | ussrback | this config was there |
| 16:48.13 | russellb | coppice: pretty much |
| 16:48.20 | ussrback | before addon compilation |
| 16:48.21 | coppice | kinda sad |
| 16:48.50 | russellb | the ooh323 folks that wrote it aren't interested in touching it unless they get paid lots of money |
| 16:49.19 | russellb | we may pay them again to get them to continue supporting it, but it hasn't been decided |
| 16:50.08 | ussrback | so what u suggest for h323? |
| 16:50.16 | coppice | those objective systems guys are a bit of a pain. |
| 16:50.31 | robl^ | besides some video conference systems and microsoft, does anyone still use h323? |
| 16:50.31 | russellb | ussrback: try chan_h323 or ooh323 ... use whatever ends up working for you |
| 16:50.49 | russellb | robl^: lots of legacy equipment, unfortunately |
| 16:51.05 | coppice | robl^ H.323 calls outnumber SIP calls 100:1 |
| 16:51.09 | fedya | does anyone use polycoms? do they require that a tftp server be present to configure from? or is it possible to manually configure them? |
| 16:51.18 | *** join/#asterisk lirakis (n=etamme@69.24.142.1) |
| 16:51.31 | russellb | fedya: they have a web interface |
| 16:51.52 | coppice | MGCP is also far more widely used than SIP |
| 16:52.10 | anonymouz666 | what is MGCP? :) |
| 16:52.12 | robl^ | coppice: wow! I didn't know that. I was thinking that SIP was the primary VoIP protocol these days. I rarely see support for it anywmore |
| 16:52.33 | russellb | robl^: most actively developed ... not necessarily the most widely deployed yet |
| 16:52.37 | robl^ | rarely see support for H323 or MGCP even |
| 16:52.44 | coppice | as my mother would say "empty kettles make the most noise" |
| 16:53.06 | [TK]D-Fender | fedya: You can use the phone interface itself, the web interface, or provisioning (more than just TFTP). I highly recommend the latter. |
| 16:54.00 | robl^ | fedya: polycoms can use tftp, ftp, or http. I suggest ftp |
| 16:55.34 | tzafrir | coppice, what specific versions of libtiff won't crash rx_fax (or is it tx_fax)? |
| 16:56.14 | coppice | most distros use 3.8.2 now. that's OK - it has two problems, but not killers |
| 16:56.27 | *** join/#asterisk Nukemizer (n=Nuke@67.137.28.165) |
| 16:58.38 | tzafrir | libtiff4-3.8.2-7 here (Debian Etch) . And either rx_fax or tx_fax crashes |
| 16:58.54 | tzafrir | post a backtrace? |
| 16:58.58 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 16:59.07 | coppice | it shouldn't be the fault of that version of libtiff |
| 16:59.23 | coppice | OK |
| 17:00.13 | *** join/#asterisk irule (n=irule@200.53.61.4) |
| 17:00.55 | *** join/#asterisk |omni| (n=rob@c-67-185-144-206.hsd1.wa.comcast.net) |
| 17:01.59 | tzafrir | coppice, http://pastebin.ca/798174 |
| 17:02.58 | agx | coppice: hopefully not my fault this time :) |
| 17:03.53 | agx | tzafrir: i've 3.8.2 |
| 17:04.43 | agx | tzafrir: can you share the tiff with me? i wanna test it too |
| 17:05.24 | tzafrir | agx, libtiff4 from Debian Etch |
| 17:06.29 | coppice | that looks weird. it has a kinda scrambled look like "built with version X, but running with version Y" kind of problems |
| 17:07.22 | tzafrir | it is built on a different system, but both run from the same repo, and they seem to have basically the same versions of everything |
| 17:07.33 | tzafrir | Any specific package you have in mind? |
| 17:07.57 | coppice | are you sure they don't have more than one version in different directories? this is a common issue |
| 17:08.55 | agx | tzafrir: i'm using same package: i386 debian etch libtiff4 3.8.2-7; perhaps you're mixing spandsp headers? |
| 17:09.36 | tzafrir | I'm using a spandsp deb I've just built of 0.0.4pre15 |
| 17:09.57 | tzafrir | The fact that spandsp still has an soname of 0 is an issue of its own |
| 17:10.03 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-18a44cfe9f3f40af) |
| 17:11.28 | tzafrir | yum. Just compared every single library in ldd, and they are the same |
| 17:12.15 | agx | tzafrir, if you can mailme the .tiff i can test it myself |
| 17:14.35 | fedya | i configured the polycom through the web interface, but it just reboots and says that no boot server was found and just keeps rebooting bceause it cant load MACADDRESS.cfg |
| 17:15.17 | alrs | tzafrir: Did Sangoma get in touch with you? |
| 17:16.15 | tzafrir | alrs, eventually: yes |
| 17:16.21 | tzafrir | working on a deb right now |
| 17:17.07 | tzafrir | agx, http://rapid.tzafrir.org.il/~tzafrir/fax_test.tiff |
| 17:18.13 | tzafrir | they should have a version out eventually that does not need to patch Zaptel. |
| 17:18.30 | tzafrir | With that version it would finally be possible to prepare a deb package |
| 17:18.43 | coppice | that's not a fax file |
| 17:18.53 | alrs | tzafrir: aces, thanks for the work |
| 17:18.54 | [TK]D-Fender | fedya: I'd advise flushing the phone's local configs and provisioning them from scratch.... |
| 17:19.21 | tzafrir | coppice, is there a sample fax file anywhere to test? |
| 17:19.49 | coppice | spandsp comes with a bunch of them :-) |
| 17:21.11 | coppice | interesting. that TIFF file makes bad things happen. I will have to investigate |
| 17:21.28 | coppice | it should just reject unsuitable tiffs |
| 17:22.22 | coppice | when I feed it to my test programs I get the same result you did |
| 17:23.44 | *** join/#asterisk dlynes (n=chatzill@d154-20-45-103.bchsia.telus.net) |
| 17:24.19 | coppice | tzafrir: thanks.my software behaves sanely with other wrong TIFFs I throw at it. I didn't know about that problem |
| 17:25.13 | coppice | tzafrir: if you go to the itutests/fax directory and run make, it should create a standard set of ITU test images as TIFFs |
| 17:25.14 | fedya | there's no way to reset factory settings until the phone actually boots up |
| 17:27.28 | [TK]D-Fender | fedya: setup a boot server for it with a different SIP/BR image than its running and it should cause a reformat. |
| 17:29.09 | *** join/#asterisk mog (n=mog@c-68-62-172-83.hsd1.al.comcast.net) |
| 17:29.09 | *** mode/#asterisk [+o mog] by ChanServ |
| 17:29.38 | tzafrir | coppice, right. Got some stuff to add to the spandsp-doc package, then |
| 17:32.00 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 17:33.07 | *** join/#asterisk Schumie (i=SteveWri@cpc1-rdng2-0-0-cust441.winn.cable.ntl.com) |
| 17:33.42 | coppice | ah, a test seems to have gone missing from the current version of spandsp. |
| 17:36.28 | coppice | OK. Fixed that one. |
| 17:43.41 | *** join/#asterisk jameswf (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 17:43.56 | *** join/#asterisk moprilo (n=nada@190.10.0.84) |
| 17:44.34 | moprilo | if i have 9003,1,Dial(SIP/9003) if the channel is unavailable hwo can i check that? to do a playback for channel un available.. |
| 17:44.57 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 17:45.40 | [TK]D-Fender | moprilo: check ${DIALSTATUS} after your dial. |
| 17:45.59 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 17:47.22 | rpm | http://www.microsoft.com/canada/uc/voipasyouare/default.mspx |
| 17:47.25 | rpm | thats funny. |
| 17:49.14 | *** join/#asterisk fholmes (n=felix@PixNationalGroup.phonoscope.com) |
| 17:50.50 | [TK]D-Fender | rpm.....nah, rip & replace :) |
| 17:51.08 | rpm | they did a good job on the movies/flash videos |
| 17:51.42 | *** join/#asterisk UserReg_CL (n=dede@164.77.196.217) |
| 17:51.51 | UserReg_CL | ~book |
| 17:51.51 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 17:52.23 | robl^ | Microsoft should hire Allison to do the voices |
| 17:52.45 | rpm | they should get homer simpson to do the voices |
| 17:53.31 | jameswf | if your screen has gond blue press DOH! system crashed |
| 17:53.31 | coppice | they need to get Dr Who. they need to get back to the 1990s before this stuff was old and failed |
| 17:54.04 | jameswf | s/gond/gone/ |
| 17:54.08 | robl^ | Microsoft TARDIS SP 2 |
| 17:54.23 | jameswf | ~~~~ |
| 17:54.23 | jbot | ARGH!!! STOP IT jameswf!!! |
| 17:54.28 | coppice | the TARDIS was the perfect colour for MS |
| 17:57.03 | [TK]D-Fender | Analog FTW! |
| 17:57.05 | robl^ | jameswf: at least it wasn't an IMB PCjr with the infrared keyboard ;-) |
| 17:57.23 | robl^ | s/IMB/IBM/ |
| 17:57.46 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 17:57.46 | jameswf | ! ! !! ! ! ! !! ! |
| 17:57.55 | jameswf | s/!/POO/g |
| 17:58.05 | Qwell | nice |
| 17:58.16 | jameswf | ;) |
| 17:58.20 | Qwell | ................................................. |
| 17:58.26 | Qwell | s/./!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!/ |
| 17:58.27 | *** join/#asterisk umay (n=chris@71-208-162-10.hlrn.qwest.net) |
| 17:58.32 | Qwell | d'oh, fail |
| 17:58.35 | Qwell | s/./!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!/g |
| 17:58.41 | Qwell | ................................................. |
| 17:58.42 | Qwell | s/./!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!/g |
| 17:58.50 | Qwell | didn't work |
| 17:58.54 | mog | no flooding |
| 17:58.55 | coppice | coppice |
| 17:59.02 | Qwell | mog: :D |
| 17:59.03 | coppice | s/cop/poc/g |
| 17:59.04 | jameswf | jameswf: |
| 17:59.25 | jameswf | s/jameswf/god/g |
| 17:59.36 | *** join/#asterisk A500mg (n=x@ACaen-151-1-58-166.w86-215.abo.wanadoo.fr) |
| 17:59.40 | A500mg | hello |
| 17:59.45 | jameswf | hi |
| 17:59.48 | jameswf | :D |
| 17:59.52 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 17:59.57 | coppice | high low |
| 18:00.06 | A500mg | i've a little question |
| 18:00.10 | jameswf | who's high |
| 18:00.18 | Qwell | jameswf: the bot |
| 18:00.19 | coppice | viagra can help with that |
| 18:00.20 | jameswf | we only answer big questions |
| 18:00.29 | jameswf | google handles little ones |
| 18:00.31 | A500mg | what is the "default" player for MOH in asterisk 1.4 ? |
| 18:00.33 | A500mg | :) |
| 18:00.39 | jameswf | ~high |
| 18:00.40 | jbot | howghdy |
| 18:00.45 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 18:00.46 | Qwell | A500mg: use native moh, and you'll be fine |
| 18:00.55 | *** join/#asterisk UnixDog (n=unixdog@adsl-69-234-205-102.dsl.irvnca.pacbell.net) |
| 18:00.57 | jameswf | ~pimp |
| 18:01.06 | A500mg | i use it and it's not fine :) |
| 18:01.21 | A500mg | good in local network |
| 18:01.22 | [TK]D-Fender | A500mg: DETAILS <- |
| 18:01.33 | A500mg | but not good with ZAP |
| 18:01.55 | A500mg | MOH says "tada..tada..CHHHH CHHHH .. tada... tada... CHCHHHH...3 |
| 18:02.05 | jameswf | ~pimp |
| 18:02.05 | jbot | it has been said that pimp is see jameswf |
| 18:02.29 | jameswf | woohooo |
| 18:02.35 | *** join/#asterisk PaulAviles (n=salinas9@dsl-7-36.cofs.net) |
| 18:03.11 | A500mg | musiconhold.conf is not changed |
| 18:03.26 | A500mg | mode=files and directory=/var/lib/asterisk/moh |
| 18:04.01 | PaulAviles | hey guys, i am getting 401 unauthorized at times, how do you go about finding the issue? |
| 18:04.09 | A500mg | i've a tdm01b with no problem, except MOH with very bad quality :( |
| 18:04.12 | PaulAviles | this is to the sip provider, teliax |
| 18:04.58 | *** join/#asterisk fskrotzki (n=fskrot@host.textwise.com) |
| 18:08.04 | [TK]D-Fender | PaulAviles: pastebin a complete failed transaction's SIP debug output. |
| 18:08.29 | UnixDog | ok whats not going on here |
| 18:08.31 | [TK]D-Fender | A500mg: "rmmod ztdummy" + restart * |
| 18:09.10 | pifiu-laptop | how can I make 2 phones ring at the same time? one a sip extension and the other a cellphone? |
| 18:09.32 | A500mg | i don't use ztdummy |
| 18:09.54 | A500mg | i've restart *, i've restart the server |
| 18:11.33 | *** join/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 18:11.37 | pifiu-laptop | i have exten => incomingdidnumber,4,Dial(SIP/phonenumber,60,r) |
| 18:11.40 | PaulAviles | hey guys, i am getting 401 unauthorized at times, how do you go about finding the issue, this is to teliax the sip provider |
| 18:11.44 | pifiu-laptop | how can i make it so it dials 2 numbers? |
| 18:12.00 | pifiu-laptop | i want it to ring my deskphone and the cellphone just in case im on either one |
| 18:12.04 | *** join/#asterisk key2 (n=Ritual@193.33.36.20) |
| 18:13.07 | *** join/#asterisk angom (n=angom@200.38.31.239.dsl.dyn.telnor.net) |
| 18:13.12 | A500mg | (i've try "modprobe ztdummy" and reload zap and *, same problem) |
| 18:14.03 | *** join/#asterisk bl4q (n=Bl@dslb-088-066-235-002.pools.arcor-ip.net) |
| 18:16.37 | *** join/#asterisk csmiga (n=chsmiga@68-114-1-65.dhcp.gsvl.ga.charter.com) |
| 18:17.16 | *** join/#asterisk ELBunce (n=erik@kde/developer/bunce) |
| 18:18.26 | [TK]D-Fender | A500mg: I just told you to REMOVE ztdummy. |
| 18:18.36 | [TK]D-Fender | PaulAviles: and I jsut told you what to DO. |
| 18:18.47 | *** join/#asterisk fujin (n=aj@unaffiliated/fujin) |
| 18:19.25 | *** join/#asterisk Igbothom_3rd (n=Hilton@office.quarkgroup.com.au) |
| 18:19.48 | *** join/#asterisk JayTee52 (n=jforde05@207-67-84-185.static.twtelecom.net) |
| 18:19.49 | A500mg | ztdummy was not used |
| 18:20.06 | A500mg | no need ztdummy with a digium card |
| 18:20.15 | PaulAviles | D-Fender, what? |
| 18:20.30 | [TK]D-Fender | [13:08]<[TK]D-Fender>PaulAviles: pastebin a complete failed transaction's SIP debug output. |
| 18:20.42 | *** join/#asterisk marc7 (n=marc@64.46.14.64.novuscom.net) |
| 18:21.01 | coppice | tzafrir: a new spandsp snapshot is available, which won't crash on those colour images |
| 18:21.28 | tzafrir | Nice to see my stupidity has helped :-) |
| 18:21.38 | marc7 | VoiceMailMain is crashing and hanging up my call when the msg####.gsm in /var/spool/asterisk/voicemail/..../ has a file size of 0, I'm running 1.4.13 but I don't see a fix for that in the 1.4.14 changelog |
| 18:22.09 | ManxPower | pifiu-laptop: How would you be calling the cellphone? VoIP provider, analog port, etc? |
| 18:22.17 | JayTee52 | I want to test videoconferencing on two X-Lite softphones. I've added videosupport=yes and h263p and h263 codecs to my sip.conf file. On one message board I was reading one entry that said I needed to add the codecs to extensions.conf? |
| 18:22.53 | ManxPower | marc7: it is well known voicemail blows up if the audio file is corrupted (like any audio file with 0 byte length is) remove the file. |
| 18:22.57 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkgroup.com.au) |
| 18:23.36 | marc7 | ManxPower: could this have anything to do with my changing the minimum length of a voicemail message in voicemail.conf (minmessage=2) recently? |
| 18:23.40 | ManxPower | most of the time issues with 0 byte length audio files is because you ran out of disk space. |
| 18:23.51 | ManxPower | marc7: no, |
| 18:24.07 | marc7 | that's odd. I couldn't tell you why we're starting to see that now. |
| 18:24.23 | ManxPower | at least it should not unless there is some major bug in that feature (many, many, many people use that feature) |
| 18:24.47 | ManxPower | marc7: how much disk space is available for that partition? |
| 18:24.57 | marc7 | 4GB |
| 18:25.40 | ManxPower | marc7: remove the file and any associated .txt files, rename them files so they are sequentially numbered, starting at 0. that should fix the issue. |
| 18:26.33 | marc7 | ManxPower: I can easily do that, I'm more concerned with why it's happening... I'm upgrading to 1.4.14 later this week, I'll let you guys know if it continues to be a problem, and if I can reproduce it. |
| 18:27.56 | ManxPower | JayTee52: any message board that says you need to "add codecs in extensions.conf" for video support is a message board that you should never consult again. |
| 18:28.42 | *** join/#asterisk PepOSX (n=pepOSX@190.79.246.105) |
| 18:30.50 | tzafrir | hmmm... fax sending is still stuck after I have replaced test page. |
| 18:31.00 | tzafrir | Too bad coppice has quit... |
| 18:34.28 | *** join/#asterisk mrempire (n=trefpunt@mrempire.demon.nl) |
| 18:34.29 | blitzrage | marc7: I think that was fixed, but I can't seem to find the bug on the tracker now to verify that -- might want to test that on your lab system before upgrading |
| 18:34.37 | JayTee52 | ManxPower, thank you. I suspected as much |
| 18:35.17 | marc7 | blitzrage: suppose you're right, we're not usually in the habit of testing new asterisk releases before they hit production ;-) |
| 18:35.32 | blitzrage | marc7: not so smart |
| 18:36.40 | marc7 | blitzrage: however we have a couple of servers still running 1.2.x because I don't feel like re-writing a dialplan that I know uses deprecated functions, so I'm sure when it actually counts we will test-in-advance |
| 18:37.02 | marc7 | and those servers have ~200K minutes a month of usage |
| 18:37.10 | blitzrage | marc7: all my production servers matter, so I tend to do some testing, but hey, what do I know? |
| 18:37.12 | blitzrage | heh |
| 18:37.26 | blitzrage | phone systems aren't usually critical anyways |
| 18:37.30 | marc7 | no ;-) |
| 18:37.34 | Qwell | who uses phones anymore? |
| 18:37.40 | blitzrage | I try my best not to |
| 18:37.46 | marc7 | fax is back in vogue |
| 18:38.01 | blitzrage | fax is the new email |
| 18:38.12 | ManxPower | the results of that asterisk-users survey were pretty interesting. |
| 18:38.19 | Qwell | ManxPower: which? |
| 18:38.41 | ManxPower | Qwell: I'll find the message |
| 18:39.31 | blitzrage | does anyone else in here have a MacBook Pro who can tell me if the cdrom on these machines are supposed to sound so sickly when injecting/ejecting a CD? |
| 18:39.51 | Qwell | blitzrage: buy me one, and I'll take one for the team, and let you know |
| 18:40.04 | blitzrage | Qwell: sure -- send me your billing address |
| 18:40.14 | UnixDog | ok permit and deny are not working |
| 18:40.22 | Qwell | blitzrage: P.O. Box 304 |
| 18:40.24 | *** join/#asterisk linxroute (n=linxrout@125.214.1.18) |
| 18:40.25 | blitzrage | :) |
| 18:40.30 | marc7 | blitzrage: it should sound unpleasant, but not deathly ill. if you have applecare, take it in, and if it's fixable they'll take care of it. |
| 18:40.38 | linxroute | ;) hi all |
| 18:40.54 | blitzrage | marc7: ok -- ya... it sounds unpleasant, but it works -- but it sounds like it's kinda crushing the disk a bit :) |
| 18:41.14 | blitzrage | and my spacebar is sometimes squeeky :) |
| 18:41.22 | marc7 | form over function. slot loading drives in small spaces are sometimes susceptible to that. |
| 18:41.41 | blitzrage | gotcha -- just wanted to make sure it wasn't supposed to be silent or something |
| 18:41.51 | marc7 | if you actually have to "pull" the disc out and hear grinding noises, then you win! |
| 18:42.11 | blitzrage | nope -- it "works" fine -- it's just.... loud :) |
| 18:42.57 | hi365 | dmtf seems pretty messed up on my zap channel - "most" buttions work, but like the number 2 just isnt accepted! |
| 18:43.03 | blitzrage | darn... guess I didn't enable remote configuration on my pfSense router |
| 18:43.20 | blitzrage | time to head to #pfsense! :) |
| 18:44.17 | hi365 | how can i toubleshoot? i tried using relaxdmtf=true - didnt help |
| 18:44.23 | *** join/#asterisk Navion (n=billp@75-105-41-123.cust.wildblue.net) |
| 18:44.41 | ManxPower | relaxdtmf almost never helps |
| 18:45.06 | hi365 | so what can i try? (things where working just fine till not to long ago...) |
| 18:45.06 | UnixDog | I would like to see pfsense and askozia merge |
| 18:45.11 | hi365 | UnixDog: same here |
| 18:45.21 | ManxPower | hi365: a common fix is to put in rxgain=somenumber for JUST the problem channels where somenumber is between -8 and 8 |
| 18:45.23 | blitzrage | that'd be nice |
| 18:45.26 | UnixDog | hey hi365 |
| 18:45.28 | ManxPower | Usually +2 or -2 helps |
| 18:45.31 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 18:45.35 | UnixDog | new daemonswitch pkg out |
| 18:45.50 | UnixDog | I did a new install script |
| 18:46.00 | UnixDog | and I updated the install page |
| 18:46.41 | Navion | Anyone have an idea about how to set up a door phone? Just needs to connect to an extension number as soon as it goes off hook. Be nice if it was dial restricted. |
| 18:47.22 | [TK]D-Fender | Navion: depends on the phone. this is doable with Digium FXS channels, and many ATA's including Linksys |
| 18:47.30 | UnixDog | then you should get a grandstream 200 |
| 18:47.55 | UnixDog | and set it for offhook dial to a ringgroup |
| 18:48.09 | UnixDog | so it rings all the phones in the hous |
| 18:48.10 | UnixDog | e |
| 18:48.22 | UnixDog | or a ata |
| 18:48.32 | UnixDog | and get a dump phone |
| 18:48.37 | Navion | So, it's an ATA option with some of the ATA's? |
| 18:48.42 | UnixDog | its a phone with out a dtmf |
| 18:49.01 | UnixDog | pad |
| 18:50.37 | UnixDog | just set the off hook mode |
| 18:50.47 | Navion | Sounds like an invitation for someone to bring their own tone pad. :) |
| 18:50.53 | UnixDog | to autodial a ring group |
| 18:51.38 | hi365 | ManxPower: i tried both +2/-2, folowed my a relaod chan_zap. didnt help. seems like disa is ignoring number 1-4 as the first digit |
| 18:51.46 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 18:53.00 | Navion | Oh cool, I found it... Grandstream GXW-4008 has an autodial in the port profile. |
| 18:53.41 | UnixDog | your using a gxw-8port |
| 18:53.46 | UnixDog | how do you like it |
| 18:53.55 | UnixDog | I heard they had audio issues |
| 18:55.37 | *** join/#asterisk BlaclSlik (n=jamesmar@77.246.64.134) |
| 18:55.47 | BlaclSlik | wew |
| 18:55.53 | BlaclSlik | back |
| 18:56.02 | BlaclSlik | Any bodyy here |
| 18:58.23 | Navion | UnixDog: I had some real greef with the GXW-4104 FXO but the FXS seems to be working well. The problems I had with the 4104 was extream side tone (AKA no echo return loss) which made the echo cancellers go nuts. |
| 18:58.40 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 18:59.24 | BlaclSlik | please i need a configured asterisk to make outbond calls with a wav message like 10000 number in one hour, how do i go about that |
| 18:59.35 | Navion | UnixDog: I haven't tried using them with a FAX machine. I guess that would be the ultimate test. |
| 19:00.34 | blitzrage | BlaclSlik: probably hiring a consultant -- and even then... an hour? depends what you need it to do I suppose. Easiest is probably to use a script that connects to the AMI and then generates the calls from there |
| 19:00.55 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 19:00.59 | ManxPower | or just a script that generates .call files. |
| 19:01.26 | BlaclSlik | umh.............means i need help or hire some one |
| 19:01.30 | *** join/#asterisk BZBW (n=wlwzhang@ip67-153-142-110.z142-153-67.customer.algx.net) |
| 19:01.53 | blitzrage | BlaclSlik: you can ask specific questions in here, but it's unlikely someone is going to hold your hand to create that |
| 19:02.15 | blitzrage | ManxPower: true, but depending on the number of calls, the AMI might be a better bet (depending on call density) |
| 19:02.25 | BlaclSlik | anyone willing to this message me |
| 19:03.24 | *** join/#asterisk iamthelostboy (n=nathan@125-236-212-46.adsl.xtra.co.nz) |
| 19:03.46 | BlaclSlik | Okay the situation i am in is i have a linux sever which i have install asterisk on |
| 19:05.10 | BlaclSlik | Okay the situation i am in is i have a linux sever which i have install asterisk on and i have a sip account with no sip device which i want it to be vitrual , then the to make the out bond call, is this possible |
| 19:05.43 | blitzrage | yes |
| 19:07.26 | BlaclSlik | so i have to configure the linux sever to work with the sip account and then code a script to generate the calls |
| 19:07.29 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 19:08.10 | iamthelostboy | hi.. i have users setup with qualify=yes, but when I do a 'sip show peers' the users have a status of unmonitored |
| 19:08.38 | iamthelostboy | and nokia and linksys phones seem to lose their way |
| 19:09.57 | blitzrage | BlaclSlik: yep. SIP accounts live in sip.conf |
| 19:11.32 | *** join/#asterisk Rhinoo_ah (n=ahonea@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 19:11.41 | BlaclSlik | Then how i get the script to generate calls and make the outbound call |
| 19:12.33 | niekie | BlaclSlik: do you want to call to PSTN numbers? Or VoIP systems? |
| 19:13.16 | niekie | BlaclSlik: there are multiple ways to make outbound calls.. Probably the easiest way would be to use the management API. |
| 19:13.19 | BlaclSlik | pstn |
| 19:13.39 | niekie | BlaclSlik: do you have a provider that connects you to the PSTN? |
| 19:13.54 | blitzrage | BlaclSlik: the script generates a "callfile" which is then moved into the appropriate directory /var/spool/asterisk/outgoing which then triggers the call, and when the call answers, then runs the defined dialplan logic in extensions.conf |
| 19:14.10 | [TK]D-Fender | 10000 cals / h = 166 calls / sec = z0mg system crash |
| 19:14.22 | blitzrage | indeed |
| 19:14.22 | [TK]D-Fender | ~wglwat |
| 19:14.22 | jbot | i guess wglwat is well, good luck with all that |
| 19:14.22 | niekie | [TK]D-Fender: :-o |
| 19:14.40 | blitzrage | I read it as 1,000, not 10,000 |
| 19:14.53 | blitzrage | need multiple boxes to handle that kind of load |
| 19:15.00 | [TK]D-Fender | BlaclSlik: PSTN connectivity will cost you, and you'r going to need a small cluster of servers and a boatload of bandwidth to survive this. |
| 19:15.03 | blitzrage | and a service provider who will handle that too :) |
| 19:15.11 | BlaclSlik | 100 pstn number same time |
| 19:15.13 | iamthelostboy | if i change it to qualify=200 it pretty much immediately gives sip_poke_noanswer peer is now unreachable, and it seems to stay that way, even though i can call from the phone |
| 19:15.15 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 19:15.25 | marc7 | hey guys, if I have three desks, three phones, three "extensions", but each person wants to be able to answer each 'extension'... does that mean I need to have 9 peers in sip.conf (three for each person) and setup each 'extension' as a queue? |
| 19:15.32 | variable_office | I am having a problem where the caller(behind nat) can hear and the dest cant hear. but it isnt consistant, and sometimes a reboot of the ata works, sometimes it doesnt |
| 19:15.39 | blitzrage | marc7: no -- use pickupgroups |
| 19:15.45 | marc7 | blitzrage: okay! |
| 19:15.49 | marc7 | *reads* |
| 19:15.53 | [TK]D-Fender | marc7: Go lookup "pickup groups" on the WIKI |
| 19:16.08 | variable_office | the problem seems to be random |
| 19:16.20 | variable_office | but that is from the users perspective |
| 19:16.21 | [TK]D-Fender | variable_office: "nat=yes" , canreinvite=no" , "qualify=yes" <- make sure these are set for that phone's entry |
| 19:16.55 | BlaclSlik | Blitzrage i get your point , what language i would use in scripting such script pretty good on php |
| 19:17.28 | blitzrage | BlaclSlik: any one you are familiar with -- but based on your call volume, you won't get this done in an hour :) |
| 19:17.34 | blitzrage | BlaclSlik: I like PHP, but that's just me |
| 19:18.22 | BlaclSlik | Blitzrage can i hire u to code this script for me i really need this even if not covered in an hr |
| 19:18.29 | variable_office | [TK]D-Fender, done already |
| 19:18.35 | blitzrage | yes, I can be hired through Digium |
| 19:19.12 | [TK]D-Fender | variable_office: make sure they aren't port forwarding or accounting for NAT in their own way and you should be fine |
| 19:19.28 | Navion | Has anyone played with overhead paging using the audio port in the PC? |
| 19:19.45 | variable_office | [TK]D-Fender, they dont know enough to do that, all they did was plug the router in i am sure |
| 19:20.00 | BlaclSlik | Blitrage what would be your charge |
| 19:20.31 | blitzrage | BlaclSlik: depends what the sales department charges -- you'll have to ask them about custom development |
| 19:20.50 | Navion | ~paging |
| 19:21.05 | Navion | ~audio |
| 19:21.05 | jbot | hmm... audio is usually a codec issue. start with trying to set 'disallow=all' and 'allow=alaw' in sip.conf or the channel's config file if not using sip |
| 19:21.22 | Navion | ~pc |
| 19:21.22 | jbot | A PC is a Personal Computer. A Macintosh is a PC. PC doesn't automatically mean x86, damn it. Politically Correct |
| 19:21.44 | variable_office | jbot, was the codec towards me? |
| 19:21.46 | robl^ | actually.. Macs are x86 now. ;-) |
| 19:22.10 | BlaclSlik | HEllo fellows can i get anybody whom is like freelanced to write this script for me because not to kin with contacting companies |
| 19:24.14 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 19:27.59 | iamthelostboy | qualify=yes doesnt seem to work in sip.conf... i have changed the users to qualify=200 and now i am getting a valid status |
| 19:28.21 | *** join/#asterisk ccesario_ (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 19:28.38 | [TK]D-Fender | Navion: "chan_oss.so" - "Dial(OSS/dsp)" |
| 19:28.44 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@216-110-94-252.static.twtelecom.net) |
| 19:29.10 | [TK]D-Fender | iamthelostboy: Could be taht your remote side is closing up UDP really fast. |
| 19:30.13 | *** join/#asterisk Shaun2222 (n=Shaun222@ip68-4-127-67.oc.oc.cox.net) |
| 19:30.26 | iamthelostboy | isn't qualify=yes the same as qualify=200 ? |
| 19:30.37 | [TK]D-Fender | iamthelostboy: No, yes=2000 |
| 19:30.41 | Shaun2222 | did VP just finally implement CallerID Name? |
| 19:30.48 | iamthelostboy | ahh... |
| 19:32.48 | BlaclSlik | HEllo fellows can i get anybody whom is like freelanced to write this script for me because not to kin with contacting companies |
| 19:33.12 | *** join/#asterisk Trionnis (n=blah@209.201.67.250) |
| 19:33.12 | [TK]D-Fender | BlaclSlik: Please don't spam your requests over and over. |
| 19:33.57 | BlaclSlik | no one did answer thats why , sorry about that |
| 19:36.06 | *** join/#asterisk saftsack (n=saftsack@pD9E05758.dip.t-dialin.net) |
| 19:37.28 | twisted | anyone have a sample tftp config file for a snom 370? |
| 19:41.52 | *** join/#asterisk blackhole (i=Mishu@unaffiliated/blackhole) |
| 19:42.08 | BlaclSlik | brb |
| 19:43.00 | blackhole | We are using a custom dialplan We are currently using 3 SIP lines from vonage, and have one zap line, our main number(ending in 1834) should function the same as the one zap line(we have the zap line until we can transfer the number to vonage). our other two numbers(ending in 1835 and 1836) are both direct lines. |
| 19:43.11 | blackhole | Now i have a problem |
| 19:43.12 | blackhole | people are sometimes sent to extension 18 who should be sent to the extension 15/16 ring group or put in queue |
| 19:43.17 | Qwell | problem #1: Vonage |
| 19:43.18 | blackhole | Can someone help? |
| 19:43.36 | blackhole | Qwell, means |
| 19:43.37 | blackhole | ? |
| 19:44.47 | robl^ | Vonage doesn't play nicely with Asterisk or other VoIP PBXes. They want you to only use their service and provided equipment / softphone |
| 19:45.02 | blackhole | Hmm.. |
| 19:45.22 | blackhole | robl^, Can it cause problems with extension ? |
| 19:45.54 | blackhole | robl^, Sometime people are sent to extension 18 who should be sent to the extension 15/16 ring group or put in queue |
| 19:46.07 | blackhole | robl^, Is there some way you can suggest me to debug and correct this..? |
| 19:46.30 | [TK]D-Fender | blackhole: this is YOUR dialplan, go look at where you send calls from each fo your vonage accounts. |
| 19:47.08 | [TK]D-Fender | blackhole: there really isn't much of an excuse for not seeing where your calls are falling or looking to see why they end up in a place you didn't expect them to. Thats what 'sip debug" is for. |
| 19:47.19 | *** join/#asterisk guillote_GNU (n=bancaria@host29.201-253-3.telecom.net.ar) |
| 19:47.32 | blackhole | [TK]D-Fender, I am a bit of new bie... ! |
| 19:47.39 | niekie | Ahem. |
| 19:47.42 | [TK]D-Fender | blackhole: Get searching... |
| 19:47.44 | niekie | sip set debug |
| 19:48.01 | robl^ | blackhole: and a liberal sprinkling of NoOp()s in the dialplan |
| 19:48.02 | [TK]D-Fender | blackhole: And don't ask us to guess. You need to show us specific backup for us to advise you |
| 19:48.09 | jameswf | ~newb |
| 19:48.09 | jbot | Don't bother telling us you're a "newb" or a "n00b". We can tell. |
| 19:48.27 | Qwell | jameswf: nice |
| 19:48.38 | robl^ | ~agents |
| 19:48.38 | jbot | from memory, agents is secret undercover employees that work for a branch of the federal government, such as FBI, NSA, IRS, AT&T or Secret Service. |
| 19:48.56 | tzanger | I haven't moved to Verbose() yet |
| 19:48.59 | tzanger | it looks nice though |
| 19:48.59 | jameswf | at&t ? lol |
| 19:49.01 | blitzrage | newb :) |
| 19:49.13 | tzanger | heh |
| 19:49.28 | jameswf | ~at&t |
| 19:49.28 | jbot | i heard at&t is the devil. |
| 19:49.35 | robl^ | blitzrage: I still have some legacy stuff that predates 1.0.... lots of NoOps there ;-) |
| 19:49.42 | jameswf | ~devil |
| 19:49.42 | jbot | <Kathie Lee Gifford> I've come for jameswf's sole |
| 19:49.43 | blitzrage | :) |
| 19:49.58 | blitzrage | the sole of your shoe? |
| 19:50.02 | nestAr | lol |
| 19:50.03 | blitzrage | left or right? |
| 19:50.19 | robl^ | Kathie Lee has a shoe fetish |
| 19:50.19 | km- | vonage. hehe. |
| 19:50.31 | jameswf | ~vonage |
| 19:50.32 | jbot | it has been said that vonage is a bunch of monkeys |
| 19:50.37 | blitzrage | hrmmmm wish I was better at gfx since I gotta make a new website here... :) |
| 19:50.44 | jameswf | ~monkeys |
| 19:50.44 | jbot | This problem, like many others in the computer industry, can be solved by the application of monkeys. |
| 19:51.17 | km- | I need to swap my sip service from vonage |
| 19:51.24 | km- | I hear they're going down the shitter soon |
| 19:51.26 | robl^ | blitzrage: you can use gimp to make stick figures. I do. |
| 19:51.36 | niekie | ~allison |
| 19:51.36 | jbot | somebody said allison was The IVR Voice, http://theivrvoice.com/ and http://thevoice.digium.com/ |
| 19:51.38 | blitzrage | robl^: ya, I just got it installed on my Mac ) |
| 19:51.40 | blitzrage | :) even |
| 19:51.50 | niekie | Stick figures rock. |
| 19:51.51 | niekie | Hehe. |
| 19:51.57 | *** part/#asterisk PaulAviles (n=salinas9@dsl-7-36.cofs.net) |
| 19:52.01 | blitzrage | just gonna make something REAL simple so I at least don't have the default CentOS page for my website anymore :) |
| 19:52.11 | blitzrage | at least until I get my old HD up and running and pull the data off of it |
| 19:52.45 | jameswf | took a week and a call to each member of the executive team and 1 call to the BBB to cancel my vonage service without a fee |
| 19:53.00 | robl^ | blitzrage: how about a giant arrow pointing to the center of the page with the text "Insert Website Here!" |
| 19:53.24 | km- | jameswf: that sucks |
| 19:53.35 | km- | jameswf: I have friends on the inside, they just have things handled for me |
| 19:53.50 | jameswf | ~mob |
| 19:53.57 | jameswf | ~motd |
| 19:53.57 | jbot | In Life, ibot is always logged in as root |
| 19:53.59 | *** join/#asterisk truz_`24 (n=truz_`24@74-128-233-23.dhcp.insightbb.com) |
| 19:54.27 | km- | nah, my current hack is trying to find a use for freenum |
| 19:54.40 | niekie | Though if I post it here I'll get a lot of calls.. :-\ |
| 19:54.42 | km- | like ipv6, the only thing that stops people from using the service is worth. |
| 19:54.47 | km- | what's it worth to someone? |
| 19:54.54 | niekie | km-: freenum? |
| 19:55.07 | km- | yeah, it's like enum. |
| 19:55.21 | km- | it's a technology that resolves phone numbers to voip endpoints |
| 19:55.29 | niekie | Ah. |
| 19:55.34 | niekie | Like e164.arpa? |
| 19:55.35 | km- | www.freenum.org |
| 19:55.38 | niekie | And e164.org? |
| 19:55.39 | km- | yes, exactly like that. |
| 19:55.41 | blackhole | I am looking to have someone solve some problems i m facing with asterisk, i will pay if he/she can solve it..? |
| 19:55.45 | km- | e164 is enum. |
| 19:55.52 | niekie | km-: not e164.org though. |
| 19:56.03 | jameswf | ~e164 |
| 19:56.03 | jbot | rumour has it, e164 is the numbering scheme for many telephone numbers, e164.org maps them into dns |
| 19:56.15 | km- | ok, my bad, e164.arpa is enum. |
| 19:56.38 | km- | :) |
| 19:56.54 | niekie | Haha. |
| 19:57.30 | km- | I've thought about starting a company doing asterisk consulting many times, but it just doesnt seem like the right direction to go. |
| 19:57.40 | km- | maybe because I hate consulting. |
| 19:57.59 | TJNII | km-: There's good money to be made prolonging problems |
| 19:58.14 | blackhole | km-, can u help me |
| 19:58.15 | UnixDog | I need to look at enum |
| 19:58.15 | *** join/#asterisk truz_`24 (n=truz_`24@74-128-233-23.dhcp.insightbb.com) |
| 19:58.15 | blackhole | ? |
| 19:58.16 | km- | yeah, that's my problem, I'm too honest :) |
| 19:58.27 | km- | brb, getting paged |
| 19:58.30 | km- | (oncall 4tl) |
| 19:58.52 | jameswf | I was thinking about going to work for fonality, they have got exploiting asterisk down to a science |
| 19:59.24 | *** join/#asterisk truz_`24 (n=truz_`24@74-128-233-23.dhcp.insightbb.com) |
| 20:00.21 | blackhole | I am looking to have someone solve some problems i m facing with asterisk, i will pay if he/she can solve it..? |
| 20:00.27 | blackhole | Please help me.. |
| 20:00.30 | blackhole | Its urgent..! |
| 20:00.34 | niekie | blackhole: what's the problem? |
| 20:00.56 | jameswf | I take ca$h up front |
| 20:01.04 | niekie | jameswf: haha. |
| 20:01.06 | blackhole | niekie, Honestly i don't know much about asterisk...-- people are sometimes sent to extension 18 who should be sent to the extension 15/16 ring group or put in queue |
| 20:01.31 | blackhole | niekie, We are using a custom dialplan We are currently using 3 SIP lines from vonage, and have one zap line, our main number(ending in 1834) should function the same as the one zap line(we have the zap line until we can transfer the number to vonage). our other two numbers(ending in 1835 and 1836) are both direct lines. |
| 20:01.41 | niekie | blackhole: without having a look at your dialplan, that would be quite hard to see. |
| 20:01.57 | blackhole | I will paste my dailplan.. |
| 20:02.21 | *** join/#asterisk clive- (n=pirch@dsl-242-189-88.telkomadsl.co.za) |
| 20:02.21 | km- | blackhole: do you use sip at all with asterisk currently? |
| 20:02.26 | niekie | blackhole: I'm not really experienced with ZAP stuff and hardware stuff though. I only have SIP devices. |
| 20:02.38 | blackhole | niekie, Okay n[ |
| 20:02.39 | blackhole | np |
| 20:02.46 | km- | blackhole: if so, please paste your zaptel.conf, sip.conf, extensions.conf to a pastebin and give me the links |
| 20:02.49 | blackhole | niekie, I will paste dailplan and then you can have look...' |
| 20:02.54 | blackhole | Sure.. |
| 20:02.57 | niekie | I don't really use ring groups too.. :\ |
| 20:02.58 | clive- | has anyone heard of a symbian iax phone yet ? |
| 20:03.17 | niekie | Hrmm.. I thought there was SIP for Symbian. |
| 20:03.19 | km- | clive-: can't say I have. but since iaxclient is open source, maybe someone will think about writing one. |
| 20:03.23 | niekie | But not sure about IAX. |
| 20:03.27 | jameswf | isbt synbian an adult xxx toy |
| 20:03.32 | Trionnis | haha |
| 20:03.33 | km- | no, that's a sybian |
| 20:03.38 | km- | symbian is a crappy os for smartphones |
| 20:03.39 | jameswf | s/isbnt/isnt/ |
| 20:03.40 | km- | ;) |
| 20:03.47 | clive- | lol..ok, I better nt make spelling mistakes again:) |
| 20:03.51 | clive- | not |
| 20:03.52 | niekie | km-: is it really that bad? |
| 20:04.05 | km- | I've never used it so I can't say for sure |
| 20:04.08 | km- | but I've heard people lambast it |
| 20:04.13 | km- | I'm a blackberry addict myself |
| 20:04.35 | jameswf | I has a blackberry, still looking for a good ebook reader for it |
| 20:04.44 | km- | mobipocket just released a new version |
| 20:04.59 | *** join/#asterisk GreggB (n=GreggB@66.206.86.107) |
| 20:05.07 | km- | if you haven't already, might I suggest getting the Viigo rss reader |
| 20:05.14 | km- | I dont know how I ever lived without it |
| 20:05.25 | niekie | I had a good laugh at the voice files I've found that come with Asterisk but aren't really for professional use :-) |
| 20:06.33 | *** join/#asterisk bmg505 (n=leon@196.209.179.149) |
| 20:06.57 | km- | yeah, some people enjoy making allison say silly shit |
| 20:07.11 | km- | we had all the prompts allison did re-recorded at our office |
| 20:07.19 | km- | people kept saying they thought she sounded like a pornstar. |
| 20:07.28 | Trionnis | she does |
| 20:07.35 | blackhole | km-, http://rafb.net/p/DbpVTr35.html this is extensions.conf |
| 20:07.36 | Trionnis | not that I mind... she's kinda hot... |
| 20:07.42 | blackhole | niekie, Please check it.. |
| 20:07.43 | Trionnis | *ahem* |
| 20:07.49 | blackhole | Sorry..! |
| 20:07.51 | km- | yeah, allison is hot. |
| 20:08.01 | jameswf | is that marks girlfriend? where did she come from? |
| 20:08.10 | Trionnis | haha |
| 20:08.14 | km- | holy no linefeeds. |
| 20:08.16 | blackhole | http://rafb.net/p/Br3hDH81.html |
| 20:08.19 | Trionnis | she's old enough to be Mark's mom dude |
| 20:08.19 | blackhole | Check this km- |
| 20:08.22 | Trionnis | :) |
| 20:08.33 | km- | blackhole, what's up with the no linefeeds? heh |
| 20:08.33 | blackhole | niekie, Please check http://rafb.net/p/Br3hDH81.html |
| 20:08.48 | km- | that's better |
| 20:09.00 | blackhole | km-, yes some word wrap text earlier, sorry..! |
| 20:09.03 | km- | OK, you said that calls coming in on 1834 are supposed to go to extension 18 but sometimes go to 34/35? |
| 20:09.18 | blackhole | yes |
| 20:09.37 | km- | wow, who wrote your extensions logic? |
| 20:09.49 | blackhole | some guy i have earlier.. |
| 20:09.54 | *** join/#asterisk tripps (n=sean@72.20.150.196) |
| 20:10.00 | km- | can you post your zaptel config, /etc/asterisk/zapwhatever |
| 20:10.06 | km- | zapata? zaptel? |
| 20:10.11 | blackhole | people are sometimes sent to extension 18 who should be sent to the extension 15/16 ring group or put in queue |
| 20:10.30 | blackhole | See they are supposed to got to 15/16 ring group but they go to extension 18 |
| 20:10.38 | blackhole | yes sure, i will just a min |
| 20:10.43 | km- | ok, calls that should be hitting 15/16 sometimes go to 18 |
| 20:10.48 | km- | but calls for 18 don't wind up in 15/16 |
| 20:11.09 | blackhole | yes |
| 20:11.14 | *** join/#asterisk lemanal (n=lemanal@ip68-14-106-198.no.no.cox.net) |
| 20:11.38 | blackhole | They should be sent to 15/16 ring group or put in queue but goes to extension 18 |
| 20:11.39 | zerocod3r | may I ask something, which voip phone works asterisk just fine? is there any voip phone with digium brand ;) |
| 20:12.12 | km- | zerocod3r: cisco 7940/7960's work amazingly well with asterisk when loaded with cisco's sip firmware. |
| 20:12.25 | km- | zerocod3r: if you were to ask what phone I'd confidently put on every desk in my enterprise, it'd be 7940's |
| 20:12.28 | Qwell | ~phones |
| 20:12.28 | jbot | phones is, like, http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. places ... |
| 20:13.14 | zerocod3r | ok.I will order cisco 7940 thanks |
| 20:13.28 | blackhole | km-, So any idea? |
| 20:13.41 | Qwell | zerocod3r: I would recommend not buying Cisco |
| 20:13.52 | km- | blackhole: I'm lookin. It seems unnecessarily complex |
| 20:13.56 | blackhole | Hmm |
| 20:13.58 | km- | Qwell: what's your beef with the 7940? |
| 20:14.02 | Qwell | It's Cisco |
| 20:14.05 | km- | or rather, cisco. |
| 20:14.13 | km- | ok, what's your beef with cisco? |
| 20:14.14 | blackhole | km-, Also mostly numbers dialed from office phones are not recognized on other phone systems |
| 20:14.14 | Qwell | They do sip very poorly |
| 20:14.35 | km- | blackhole: can you rephrase that? I have no idea what you mean. |
| 20:14.44 | robl^ | Cisco's licensing issues are a pain in the arse, esp few a a new user experimenting with Asterisk. |
| 20:14.49 | km- | Qwell: I've got 75 stations of it here and it works fine. |
| 20:15.07 | Qwell | km-: name one thing good about the Ciscos |
| 20:15.31 | *** join/#asterisk bantu (n=Miranda@p54A32D7D.dip0.t-ipconnect.de) |
| 20:15.39 | km- | extremely easy to manage |
| 20:15.44 | robl^ | Qwell: good thing about Cisco? they look nice. ;-) |
| 20:15.51 | km- | yes, desk appearance is fantastic |
| 20:15.54 | Qwell | km-: then you've never used a polycom |
| 20:16.09 | Qwell | I was actually hoping you would say sound quality |
| 20:16.12 | [TK]D-Fender | ~book |
| 20:16.12 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 20:16.16 | km- | no, I haven't, but I can't really say anything bad about ciscos. |
| 20:16.24 | km- | Qwell: why would I? They're polycom hardware for audio are they not? |
| 20:16.34 | Qwell | yes, that was going to be my point :p |
| 20:16.41 | Qwell | if you can't say something bad about Cisco, you've never used anything else. |
| 20:16.50 | nestAr | lol |
| 20:17.04 | km- | cisco > grandstream at least. |
| 20:17.12 | km- | can you agree to that? |
| 20:17.15 | blackhole | km-, You got any idea? |
| 20:17.16 | nestAr | sure |
| 20:17.18 | [TK]D-Fender | km-: Don't support presence. Firmware costs or you've got no warranty. Inferior call handling. Almost always more expensive that a Polycom equivalent. |
| 20:17.22 | km- | blackhole: where's the zapata config? |
| 20:17.33 | km- | and the polycoms do support call presence? |
| 20:17.56 | km- | I thought it was Asterisk's limitation that call presence wasn't reported in SIP. |
| 20:17.59 | blackhole | wait i will paste |
| 20:18.01 | [TK]D-Fender | km-: I was listing DIFFERENCES, so clearly, yes |
| 20:18.23 | [TK]D-Fender | km-: Lack of presence support is a Cisco SIP limitation |
| 20:18.25 | km- | [TK]D-Fender: thanks for the clarification on that. I must have been having a blonde moment. |
| 20:19.00 | km- | What do you have to do in asterisk to get presence indication working via sip on polycom phones, or is it just inherently supported? |
| 20:19.03 | blackhole | km-, http://rafb.net/p/faqXlB63.html -- zapata.conf |
| 20:19.25 | [TK]D-Fender | Cisco is a OK phone that is a bit of a pain to get started and lacks several features users want at a higher price. So really... why bother. |
| 20:19.39 | UnixDog | how do I lock access to a atseriskbox by a certian ip for inbound |
| 20:19.57 | jameswf | well thats crap mobireader no workie on linux |
| 20:19.58 | UnixDog | permit and deny does not work in general settings |
| 20:20.33 | [TK]D-Fender | UnixDog: PER ENTRY <-- |
| 20:20.48 | UnixDog | but there are no user entryies |
| 20:20.49 | [TK]D-Fender | UnixDog: if you want to limit globally, change your firewall |
| 20:21.09 | UnixDog | grrrr |
| 20:22.18 | *** join/#asterisk tobias (n=tobias@datadesk.lightlink.com) |
| 20:22.27 | km- | hmm. |
| 20:24.01 | km- | blackhole: your config is odd. |
| 20:24.26 | blackhole | km-, Hmm, Can you correct it..? |
| 20:24.51 | km- | blackhole: are VONAGE1/2/3 the phone numbers of your inbound lines? |
| 20:24.59 | km- | or are they some people's cellphones? |
| 20:25.20 | blackhole | they are inbound lines |
| 20:28.04 | *** join/#asterisk kaldemar (n=kalde@vipunen.hut.fi) |
| 20:28.10 | *** join/#asterisk pepo-- (n=pepOSX@190.79.246.105) |
| 20:28.23 | blackhole | km-, Why its odd? |
| 20:29.23 | *** join/#asterisk BBHoss (n=hoss@146.229.189.111) |
| 20:29.45 | *** part/#asterisk BBHoss (n=hoss@146.229.189.111) |
| 20:30.48 | km- | do you have any sip phones there? |
| 20:31.16 | km- | the reason why it's confusing is you guys are almost using as a switching platform |
| 20:31.27 | blackhole | km-, Hmm |
| 20:31.27 | km- | it doesn't look like you guys are using any regular extensions |
| 20:32.09 | blackhole | km-, i dnt understand that i need to learn more.. |
| 20:32.24 | blackhole | km-, can u tell me some book from which i can study these things? |
| 20:32.29 | hi365 | is it possible to to filter for a specific even with the manager? |
| 20:32.55 | km- | there's an oreilly asterisk book available I think. |
| 20:32.58 | km- | well, I know. |
| 20:33.04 | hi365 | ~book |
| 20:33.04 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 20:33.43 | blackhole | km-, Would that book clear these basics too me?? |
| 20:33.53 | jameswf | yes |
| 20:35.50 | km- | you bet |
| 20:37.00 | km- | what happened to your previous consultant? |
| 20:37.09 | km- | you paid him to make it work and it's not working |
| 20:37.35 | blackhole | km-, yes he told me it was working but it messing up in between. He left for Japan |
| 20:39.07 | blackhole | Anyways km- Thanks for your help..! |
| 20:39.24 | blackhole | I need to learn... thats the only alternative :( |
| 20:40.00 | Trionnis | ~osmosis |
| 20:40.00 | jbot | osmosis is, like, 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 ... |
| 20:40.08 | Trionnis | :> |
| 20:40.35 | blackhole | lol |
| 20:41.05 | Trionnis | I have many lumps and bruises from THE BOOK |
| 20:41.06 | km- | blackhole: yeah, what you really need is to get someone to sit down with you, figure out what you really need to have done, and then just rewrite the darn config |
| 20:41.17 | blackhole | yes... seems so..! |
| 20:41.21 | km- | blackhole: the way it looks like your thing is setup, you utilize one phone line to call the other and that just seems very weird |
| 20:41.32 | km- | instead you could have sip extensions on the backend and not utilize two phone lines |
| 20:41.42 | jameswf | ~@ |
| 20:41.42 | jbot | somebody said @ was where it's at |
| 20:41.44 | blackhole | hmm |
| 20:42.45 | blackhole | km-, Thanks for your help! |
| 20:42.54 | blackhole | km-, where are you from? |
| 20:43.31 | km- | philadelphia area |
| 20:43.49 | blackhole | Ahh, Okay |
| 20:44.04 | km- | yourself? |
| 20:44.58 | blackhole | Tennessee |
| 20:46.06 | blackhole | Clarksville....! |
| 20:46.21 | jameswf | jbot: diss newb |
| 20:46.26 | blackhole | Anyways km- Thanks for your support and help.. |
| 20:46.32 | jameswf | ~beer |
| 20:46.32 | jbot | ACTION has disconnected (Read error: 99 (Connection reset by beer)) |
| 20:46.43 | blackhole | I will leave for now... |
| 20:46.44 | km- | no problem. |
| 20:47.09 | km- | _ShrikE: you know, a lot of people in the city are saying AJ is the man, even though he threw that interception |
| 20:47.16 | km- | _ShrikE: did better than donovan would have |
| 20:47.21 | hi365 | is it possible to to filter for a specific even with the manager? |
| 20:48.13 | *** join/#asterisk BlaclSlik (n=jamesmar@77.246.64.134) |
| 20:48.54 | bkruse | hi365: wow have you just struke a huge topic of debate |
| 20:49.10 | bkruse | there are php libraries (the php-ami library, it is FANTASTIC!) and you can do that, but with manager alone, no. |
| 20:49.16 | km- | heh. |
| 20:49.25 | hi365 | bkruse: i.e. not from telnet? |
| 20:49.30 | bkruse | hi365: correct |
| 20:49.40 | bkruse | but you can, however, get action-ids for your requests (for use in javascript/ajam) |
| 20:49.48 | hi365 | basicly you can only do client-side filtering? |
| 20:50.01 | bkruse | hi365: this is correct, in which, you will have to get all the messages. |
| 20:50.14 | bkruse | of course, you can run the php lib on the system and connect locally (low traffic) and connect to your php server |
| 20:50.24 | twisted | hahaha |
| 20:50.26 | twisted | clarksville, tn. |
| 20:50.28 | bkruse | i know its a long way around, but for now, its the best way if you want to keep performance the highest |
| 20:50.33 | twisted | lol |
| 20:50.48 | km- | twisted: funny place? |
| 20:50.53 | hi365 | bkruse: thanks |
| 20:50.57 | km- | I've been through tennessee but not clarksville |
| 20:51.02 | twisted | i dated a girl from there once, so spent a lot of time there |
| 20:51.14 | twisted | it's like the little wanna-be city |
| 20:51.15 | km- | was she hot? |
| 20:51.29 | twisted | APSU is like the only thing that keeps the town populated... that and ft. campbell |
| 20:51.37 | twisted | km-: uh, of course |
| 20:51.43 | km- | I love cornfed midwestern girls |
| 20:51.45 | bkruse | hi365: np np |
| 20:51.58 | bkruse | twisted: when you coming down to go bowling with me again? |
| 20:52.16 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 20:52.29 | twisted | coming down? |
| 20:52.32 | twisted | i live in hsv |
| 20:53.22 | muiro | here's a small pastebin of what happens when I reload a dialplan. Can someone tell me what's going wrong? I'm not using ael. http://pastebin.com/d48af6d2 |
| 20:54.09 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@216-110-94-252.static.twtelecom.net) |
| 20:56.34 | ViKing78 | muiro: Do you have anything inside your extension.ael file? |
| 20:56.35 | *** join/#asterisk saik0 (n=saik0@adsl-9-249-228.mia.bellsouth.net) |
| 20:57.00 | saik0 | When i use follow me with an external extension (mobile phone) the calls get connected with no audio on either end. How can i find out what the problem is and what cani do to fix it? |
| 20:57.36 | *** join/#asterisk MrTelephone (n=MrTeleph@h697179-171.picriverisp.net) |
| 20:57.47 | MrTelephone | how can you get asterisk to forward to voicemail on busy? |
| 20:58.22 | MrTelephone | I mean.. when a user is in ringing state |
| 20:58.43 | BlaclSlik | any ivm user in here |
| 20:58.46 | MrTelephone | what happens is that voicemail works if the called party is in conversation, but it doesn't when the user has the phone offhook |
| 21:00.00 | [TK]D-Fender | MrTelephone: ... huh? |
| 21:00.04 | fetcher | MrTelephone: look at the "standard extension" example in extensions.conf, with conditional handling of s-BUSY, etc |
| 21:00.13 | MrTelephone | ok |
| 21:00.22 | ViKing78 | MrTelephone: You can uether use n+101 priority or use the ${DIALSTATUS} variable |
| 21:00.57 | ViKing78 | MrTelephone: http://www.voip-info.org/wiki-Asterisk+cmd+Dial - look at the section "Return codes" |
| 21:01.05 | [TK]D-Fender | MrTelephone: just call voicemail on the next priority after your dial. |
| 21:01.12 | muiro | ViKing78: I actually attempted to remove the ael file, but it didn't seem to help. It has some stuff in it but only what the installation put there |
| 21:01.16 | jameswf | ~vm |
| 21:01.16 | jbot | Emacs-based mail reader. URL: http://www.wonderworks.com/vm/ |
| 21:01.22 | jameswf | bah |
| 21:01.23 | *** join/#asterisk d-k-t (n=dt@60.176.197.184) |
| 21:02.16 | ViKing78 | muiro: don't remove it. Just comment out the contexts it's complaining about. |
| 21:03.01 | *** join/#asterisk mirco (n=mirco@p54B2409F.dip.t-dialin.net) |
| 21:06.29 | MrTelephone | exten => s,1,Dial(${ARG2},${ARG3}) |
| 21:06.29 | muiro | ViKing78: I commented out those contexts but it's still giving me the same warnings. |
| 21:06.30 | MrTelephone | exten => s,2,Voicemail(b${ARG1}) |
| 21:06.40 | MrTelephone | that should work |
| 21:08.22 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 21:08.30 | *** join/#asterisk Darthclue (n=e054502@fw149.nisd.net) |
| 21:09.23 | [TK]D-Fender | MrTelephone: yup |
| 21:09.51 | MrTelephone | the default config worked fine but it wouldn't work when someone was ringing another party |
| 21:09.58 | MrTelephone | but thats the same as s-BUSY |
| 21:10.39 | [TK]D-Fender | MrTelephone: Hree we don't care WHY they don't answer. In what case would you not want an unaswered call to hit VM? |
| 21:11.22 | muiro | ViKing78: I had to comment out a good portion of it and "ael reload" but the warnings are gone. I hope I don't need that to make calls. I guess we'll see when I get the zap card. |
| 21:11.47 | MrTelephone | tkfender, we'll see if your solution works there.. i'll let you know :P |
| 21:11.50 | ViKing78 | muiro: nope, they're just there for example ael layout |
| 21:12.08 | MrTelephone | it barely happens when someone phones another person during a time when they are ringing another person.. |
| 21:12.20 | muiro | ViKing78: ok, good. Strange, it's done this since I installed. I'll probably be playing with ael soon so we'll see |
| 21:12.26 | [TK]D-Fender | MrTelephone: Either they answer, or they don't. |
| 21:12.38 | [TK]D-Fender | MrTelephone: For whatever reason they don't = VM |
| 21:12.46 | MrTelephone | ok |
| 21:16.55 | *** join/#asterisk smithj (n=smithj@foresight/developer/smithj) |
| 21:19.18 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 21:19.18 | *** topic/#asterisk is Asterisk: The Open Source Telephony Application Platform (asterisk.org) -=- Asterisk 1.4.14 (2007/11/16), *-Addons 1.4.4 (2007/10/16), Zaptel 1.4.6 (2007/10/18), Libpri 1.4.2 (2007/10/16) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #asterisk-commits to monitor svn -=- #freepbx (freepbx.org) or #trixbox for trixbox (trixbox.org) support |
| 21:21.41 | *** join/#asterisk mirco (n=mirco@p54B2409F.dip.t-dialin.net) |
| 21:24.41 | *** join/#asterisk mirco (n=mirco@p54B2409F.dip.t-dialin.net) |
| 21:24.47 | *** join/#asterisk Defraz (n=t0tal@71-209-11-186.bois.qwest.net) |
| 21:25.45 | MrTelephone | some guy emailed me.. can I have the password to your registrar so I can change the pri and sec dns servers.. is the guy on crack? |
| 21:25.58 | MrTelephone | he can do a registrar transfer if he wants |
| 21:25.58 | fujin | Probably. |
| 21:26.45 | MrTelephone | same guy who told everyone he hacked the bank when he was little |
| 21:27.16 | fujin | ;\ |
| 21:27.20 | fujin | o_0 |
| 21:27.24 | fujin | I hacked all the banks. |
| 21:29.52 | *** join/#asterisk Defraz (n=t0tal@71-209-11-186.bois.qwest.net) |
| 21:40.17 | saik0 | When i use follow me with an external extension (mobile phone) the calls get connected with no audio on either end. How can i find out what the problem is and what can i do to fix it? |
| 21:40.28 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177583123.dsl.bell.ca) |
| 21:40.47 | *** join/#asterisk alephcom (n=chatzill@h66-112-187-16.mcsnet.ca) |
| 21:41.12 | *** join/#asterisk loko (n=rbrown@c-71-199-96-200.hsd1.pa.comcast.net) |
| 21:42.01 | *** join/#asterisk iratik (n=itariki@12-226-116-3.client.mchsi.com) |
| 21:42.07 | loko | file ? |
| 21:42.10 | iratik | How do I test to see if an sip register string works? |
| 21:43.35 | iratik | i mean... i'm telneting to port 5060 on the sip server my asterisk installation is supposed to point to ... and i'm getting connection refused |
| 21:43.35 | Nugget | telnet is eeeeeeevil! |
| 21:43.37 | *** join/#asterisk mopri (n=jjohn@201.198.78.23) |
| 21:43.44 | iratik | does that mean my sip server is down? |
| 21:43.54 | mopri | can check if a SIP channel is available?, so i can send it to playback? |
| 21:45.14 | iamthelostboy | is it possible to have one sip user logged into asterisk from 2 different phones, and have both ring when a call comes in? |
| 21:45.54 | iratik | anyone? |
| 21:46.02 | *** join/#asterisk `Sean (i=Sean@CPE002211569301-CM0011e6be76d9.cpe.net.cable.rogers.com) |
| 21:46.10 | `Sean | IS THERE ANYWHERE I CAN GET TOLLFREE TERMINATION? |
| 21:46.23 | `Sean | fucking bkw cut off tollfree termination from asterlink and ever since then its been problemville |
| 21:47.45 | alephcom | 'Sean, certainly there is. |
| 21:48.11 | alephcom | most voip providers will terminate toll free. They probably will no do it for free though, unless you have a channel agreement of some sort with them. |
| 21:48.11 | Nivex | I'm gonna go out on a limb here and suggest that walking into a channel, screaming, and blaming one of the head developers isn't the best way to get help. |
| 21:48.25 | *** join/#asterisk BlackSlik (n=jamesmar@77.246.64.134) |
| 21:48.39 | j0 | alephcom: but not all will port toll free numbers for you. |
| 21:48.58 | *** join/#asterisk Odie_flocon (n=odie@S0106000f3d5eae81.cg.shawcable.net) |
| 21:49.05 | alephcom | right, that's not what he asked for. |
| 21:49.17 | j0 | true :) |
| 21:49.36 | Nivex | HOW DO I GET WORLD PEACE?!?! |
| 21:49.37 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:49.43 | alephcom | lol |
| 21:49.45 | MrTelephone | fender thanks for the input earlier.. gotta jet tho |
| 21:49.47 | MrTelephone | talk to ou guys later |
| 21:49.57 | alephcom | Nivex: let me know when you figure it out |
| 21:50.07 | `Sean | Nivex bkw is a clown :P |
| 21:50.16 | alephcom | I guess you could like the romans did at carthage. "They made a desert and called it peace" |
| 21:50.38 | Nivex | alephcom: That seems to be the U.S. strategy at the moment :( |
| 21:50.38 | alephcom | umm, on second thought I really don't like that idea. :-) |
| 21:50.41 | BlackSlik | where can i get a usa tollfree number with sip/iax and i pay with my debit card |
| 21:50.44 | Odie_flocon | hey has anybody here used the * appliance? |
| 21:50.59 | russellb | ;) |
| 21:51.02 | file | loko: eh? |
| 21:51.24 | Odie_flocon | russellb you have? how do you like them? |
| 21:51.32 | iratik | i mean... i'm telneting to port 5060 on the sip server my asterisk installation is supposed to point to ... and i'm getting connection refused ... does that mean the sip server is down? |
| 21:52.03 | Nivex | iratik: no. SIP is UDP, so trying to make a TCP connection is pointless |
| 21:52.16 | russellb | Odie_flocon: I work for Digium ... I am very biased |
| 21:52.22 | loko | file talk in private |
| 21:52.27 | iratik | so there is no way to test an sip register string ? |
| 21:52.46 | russellb | Odie_flocon: i think it's awesome! ;) |
| 21:52.59 | *** part/#asterisk loko (n=rbrown@c-71-199-96-200.hsd1.pa.comcast.net) |
| 21:53.47 | Odie_flocon | Good I want a biased opinion. |
| 21:54.09 | *** join/#asterisk techie (n=techie@adsl-76-214-25-54.dsl.lsan03.sbcglobal.net) |
| 21:54.12 | `Sean | does someone have a toll free number i can dail to check if the callerid works? |
| 21:54.13 | Odie_flocon | ok I have a couple of questions for you then |
| 21:54.38 | j0 | `Sean: u in canada? |
| 21:54.51 | BadHorsie | what softphone software do you guys recommend? |
| 21:54.53 | `Sean | j0 yes but the gateway im using is in USA |
| 21:54.58 | `Sean | so a usa tollfree would work aswell :) |
| 21:55.00 | nestAr | can anyone tell me why my parked calls time out back to the Zap channel they are on.. IE: asterisk tries to dial ther same zap channel the call is parked on? |
| 21:57.23 | Odie_flocon | russellb ok how many trunks can you have on the * appliance? |
| 21:57.39 | russellb | 8 FXO trunks, and numerous SIP or IAX2 trunks |
| 21:57.56 | `Sean | russellb who do you use for tollfree termination? |
| 21:58.10 | russellb | `Sean: i don't use any providers right now |
| 21:58.28 | russellb | i rarely _use_ asterisk. i just make it. |
| 21:58.41 | Trionnis | that should tell you something right there ;) |
| 21:58.41 | Odie_flocon | russellb If I were to get the 4 fxo / 4 fxs model I could have numerous iax, or sip trunks? |
| 21:59.14 | russellb | Odie_flocon: yes |
| 21:59.51 | russellb | Trionnis: exaggeration, really :) |
| 22:00.05 | Odie_flocon | russellb: ok cool. now, is the dialplan as versatile as *'s is. |
| 22:00.11 | russellb | i use it constantly in test environments, and actively work with tons of different people using it in all shapes and sizes of production environments |
| 22:00.44 | russellb | Odie_flocon: it only lets you do a subset of what is possible in the dialplan from the GUI (as does any GUI). But, with this GUI, you are allowed to change the configuration files manually all you want |
| 22:01.01 | russellb | it even gives you a config file editor interface in the gui itself |
| 22:01.13 | Odie_flocon | ok. |
| 22:01.57 | Odie_flocon | russellb: so I could do a dailplan that allows me to use iax trunks once my fxo trunks are full? |
| 22:02.09 | russellb | Odie_flocon: absolutely |
| 22:03.09 | nestAr | == Timeout for Zap/2-1 parked on 701. Returning to park-dial,Zap/2,1 |
| 22:03.39 | Odie_flocon | russellb: Great, now how about voicemail. can I route my voicemail to another box??? |
| 22:04.15 | russellb | Odie_flocon: sure, you just point the voicemail extension to call some SIP or IAX2 number .. |
| 22:06.27 | Odie_flocon | russellb: Thanks for your help gotta run to Dr's appt. ttyl. |
| 22:07.08 | *** join/#asterisk CVirus (n=GoD@62.135.96.14) |
| 22:07.48 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 22:09.13 | *** join/#asterisk BBHoss (i=root@www.2gac.net) |
| 22:10.17 | rob0 | Is this anything I should worry about? That IP address is one of my Sipuras. [Nov 26 14:45:43] WARNING[23576]: chan_sip.c:12537 handle_response: Remote host can't match request BYE to call '0d76330a2a694d20601fea056646c173@192.168.7.96'. Giving up. |
| 22:10.42 | BBHoss | probably harmless |
| 22:11.02 | CVirus | BBHoss: yes ? |
| 22:11.48 | rob0 | It looks like the Sipura did answer the call (bridged from Zap/1.) |
| 22:12.02 | rob0 | and this might have been after hangup |
| 22:13.05 | mopri | i have callcentric, but when callcentric calls, and my peer sip where i'm fwd'n, is not online, i'll get a voicemail from callcentric, how can i catch this, so that my customers don't get thecallcentric vm? |
| 22:14.00 | De_Mon | nestAr I think it's a bug, I've just started trying to identify that myself. |
| 22:14.30 | De_Mon | nestAr when the call is parked you should see something on the CLI that says where its supposed to timeout to, and then when it does timeout, it goes somewhere else instead -- am I right? |
| 22:17.29 | BlackSlik | Hello people , Please i wanna know ,how i could add my sip account details on sip_custom.conf |
| 22:17.42 | fujin | what's sip_custom.conf? |
| 22:18.24 | BlackSlik | sip.conf is i may say fuji |
| 22:19.03 | [TK]D-Fender | BlackSlik,,,,, |
| 22:19.07 | [TK]D-Fender | ~book |
| 22:19.08 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 22:19.12 | Trionnis | ~osmosis |
| 22:19.13 | jbot | methinks 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 ... |
| 22:19.17 | Trionnis | :> |
| 22:19.41 | [TK]D-Fender | :D |
| 22:19.45 | BlackSlik | thanks |
| 22:20.35 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 22:21.48 | [TK]D-Fender | BlackSlik, Shorter answer... vi, vim, emacs, mn, nano, pico, gedit, kwrite, OOo Writer, amongst many others. |
| 22:23.28 | mopri | how can I catch this with extensions.conf (cause 3 - No route to destination)? |
| 22:24.05 | [TK]D-Fender | mopri : you need to debug the channel that caused it |
| 22:24.38 | *** join/#asterisk felix_da_catz (n=felix@PixNationalGroup.phonoscope.com) |
| 22:24.53 | BlackSlik | TK what i am looking for is where i can add my sip account details on sip.conf |
| 22:25.19 | mopri | [TK]D-Fender: i know it is because the SIP/peer i'm calling is not online, but I wanted to send a playback if that happened |
| 22:25.59 | [TK]D-Fender | mopri : then check the ${DIALSTATUS} after your dial and play it. |
| 22:26.26 | mopri | but for some reason, my provider only works if i put ######,1,.. |
| 22:26.42 | mopri | and not with s,1.. so the ${DIALSTATUS} is not working. |
| 22:27.19 | [TK]D-Fender | BlackSlik, "register =>"'s go after [general] and before any other section. your peers/users/friends all come after your "register =>"'s |
| 22:27.41 | [TK]D-Fender | mopri :that makes no sense. You put your check in AFTER your dial. |
| 22:27.54 | [TK]D-Fender | mopri : this has nothing to do with the EXTEN you're in. |
| 22:28.28 | mopri | [TK]D-Fender:ok.. i tried that already, but i guess i did something wrong, ..try'n again |
| 22:31.17 | *** join/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) |
| 22:32.03 | *** join/#asterisk sandorp (n=sandor@firewall2.wsi.net) |
| 22:32.09 | *** join/#asterisk csmiga (n=chsmiga@68-114-1-65.dhcp.gsvl.ga.charter.com) |
| 22:34.05 | sandorp | do I need to do something special for attended or blind transfers to work? from the sparse documentation I've read, it appears it should "just work" by pressing *2 or #1 which is what I think my features.conf says to do |
| 22:34.20 | sandorp | I'm running 1.4 |
| 22:34.56 | [TK]D-Fender | sandorp, pastebin your cal & failed attempt at verbose 10 |
| 22:37.42 | nestAr | De_Mon: yes, it should be timing out back to Sip/10XX |
| 22:37.51 | [TK]D-Fender | sandorp, and what phone are you using? |
| 22:38.20 | sandorp | http://www.pastebin.ca/798642 |
| 22:38.39 | sandorp | I am using x-lite as a SIP client .... I am attempting to transfer using x-lite |
| 22:39.23 | [TK]D-Fender | sandorp, -- Executing [s@macro-stdexten:1] Dial("Zap/1-1", "SIP/6010&IAX2/6010|20") in new stack <--- see this |
| 22:39.34 | [TK]D-Fender | sandorp, you are not ALLOWING DTMF transfer in your dial statement. |
| 22:39.38 | sandorp | that's my initial call going to my SIP client |
| 22:39.52 | nestAr | De_Mon: it actually doesn't seem to set the park-dial extention until it gets close to timing out. |
| 22:40.33 | nestAr | hrmm |
| 22:40.37 | nestAr | maybe that's not true either.. |
| 22:40.54 | wwalker | I just got some recordings from voice talent. they sound "buzzy". Should we get them as 48,44.1, or 8 K wavs? |
| 22:41.03 | wwalker | for conversion to gsm or sln? |
| 22:41.16 | *** join/#asterisk _Sam-- (n=sam@fresco.kneedraggers.com) |
| 22:42.09 | _Sam-- | [TK]D-Fender : you around? |
| 22:42.25 | nestAr | don't see him! |
| 22:42.30 | [TK]D-Fender | Today I'm SQUARE! |
| 22:42.44 | _Sam-- | i can make zaptel 1.4.5.1 fine from source, but cant make 1.4.6 |
| 22:42.59 | _Sam-- | would you fix it for a fee? im on asterisk 1.4.14, and zap 1.4.5.1 |
| 22:43.11 | _Sam-- | my asterisk cores about every 3 days, getting old. |
| 22:44.14 | _Sam-- | the zaptel 1.4.6 makes all the ko files, and i can get them to load if i put them in the right place, but something isnt right. |
| 22:44.16 | *** join/#asterisk Defraz (n=t0tal@71-38-91-152.bois.qwest.net) |
| 22:44.45 | [TK]D-Fender | _Sam--, pm me details. |
| 22:45.19 | Trionnis | don't worry Sam, he won't bill you for 2 months... you have a while to save up the cash ;) |
| 22:45.33 | _Sam-- | i pay before im billed, get better service that way :) |
| 22:50.35 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 22:51.10 | sandorp | [TK]D-Fender: I've added flags "two" to my Dial statement and asterisk still does not see the buttons pressed on the SIP client; you can see the log at http://www.pastebin.ca/798665 |
| 22:53.00 | *** part/#asterisk zerohalo (n=zeroHalo@pool-71-162-97-18.bstnma.east.verizon.net) |
| 22:54.53 | [TK]D-Fender | sandorp, you STILL aren't allowing your phones to transfer in that dial statement. read the INSTRUCTIONS. "core show application dial" |
| 22:55.38 | _Sam-- | sandorp : what are the clients? |
| 22:55.50 | sandorp | x-lite |
| 22:56.45 | sandorp | [TK]D-Fender: I thought adding the "t" flag enabled the use of the digits defined in features.conf ... did I not understand the docs right? |
| 22:57.42 | [TK]D-Fender | sandorp, go read them AGAIN. |
| 23:02.38 | nestAr | De_Mon: it's like it's backwards. it says it's parking the IAX phone side, but it's actually parking the Zap side. |
| 23:02.47 | nestAr | <PROTECTED> |
| 23:06.47 | sandorp | [TK]D-Fender: at the risk of my sounding dumb, the docs say "t = Allow the called party to transfer the calling party by sending the DTMF sequence defined in features.conf"; I added the "t" flag to the Dial statement you pointed out as being incorrect earlier; was I supposed to add the flag elsewhere? the x-lite client is answering the call so x-lite is the "called party", right? |
| 23:07.21 | CrazyTux[m] | I can access sound files like so: Playback(sub_dir_a/sub_dir_b/sub_dir_c/some_file) correct? |
| 23:07.37 | CrazyTux[m] | and that'd start from base path relative to /var/lib/asterisk/sounds. |
| 23:08.00 | [TK]D-Fender | sandorp, Also please note the DYNAMIC-FEATURES variable you should be setting |
| 23:08.13 | [TK]D-Fender | CrazyTux[m], Yes |
| 23:08.28 | *** join/#asterisk RoyK (n=roy@ip-29-2-149-91.dialup.ice.no) |
| 23:10.22 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 23:15.55 | *** join/#asterisk kavelot (n=lem0n@189-19-234-120.dsl.telesp.net.br) |
| 23:16.32 | kavelot | I can't recall which config must be done so Asterisk will query its peers ie each 2 seconds to check if them are alive... any hints? |
| 23:17.50 | *** join/#asterisk beighto (n=chatzill@12.176.156.130) |
| 23:17.51 | *** join/#asterisk Freman (n=freman@brdr-gw-01.benon.com) |
| 23:19.22 | _ShrikE | qualify=yes |
| 23:19.49 | kavelot | great, thanks |
| 23:20.05 | _ShrikE | qualify=bad |
| 23:20.14 | Freman | ummm... someone's just come to me and asked me a question I've never had to answer before. How can I configure asterisk to let people pick up other peoples incomming calls (via dialing an extension number, not hitting a line button)? |
| 23:20.35 | De_Mon | heh |
| 23:20.36 | russellb | Freman: app_pickupexten |
| 23:20.44 | Freman | thanks |
| 23:20.48 | Freman | I'll go read |
| 23:20.53 | russellb | maybe that's not it ... |
| 23:20.59 | *** join/#asterisk coppice (n=chatzill@102.204.17.210.dyn.pacific.net.hk) |
| 23:21.07 | De_Mon | no, that doesn't exist... |
| 23:21.10 | Freman | no, google says no |
| 23:21.27 | Freman | that's a res_features |
| 23:21.30 | russellb | app_directed_pickupo |
| 23:21.40 | russellb | the application is "Pickup" |
| 23:22.03 | russellb | alternatively, look at pickup groups |
| 23:22.26 | Freman | thanks |
| 23:22.31 | Freman | I know have direction (c: |
| 23:22.44 | beighto | I am trying to install wanpipe for some Sangoma a200 cards onto a copy of CentOS 4.5 64 bit. At the end of the setup it says "Wanpipe header files were not installed properly in /lib/modules/2.6.9-55.0.12.ELsmp/build/include directory". The printout fron uname -r is 2.6.9-55.0.12.ELsmp and under /usr/src/kernels I have 2.6.9-55.0.12.EL-smp-x86_64, any thoughts? |
| 23:24.24 | *** join/#asterisk tobias (n=tobias@nat1.ppckernel.org) |
| 23:26.20 | *** part/#asterisk mopri (n=jjohn@201.198.78.23) |
| 23:27.43 | JT | the error is for wanpipe headers... |
| 23:28.24 | beighto | I see... is there a tag I need to put on the end of setup -install for 64 bit os's? |
| 23:28.29 | De_Mon | ah haaa |
| 23:41.29 | *** join/#asterisk remmo (n=junk@203.32.47.250) |
| 23:43.08 | *** join/#asterisk Linx (n=linx@219-89-195-91.adsl.xtra.co.nz) |
| 23:43.16 | *** join/#asterisk grimsy (n=chatzill@203-206-220-214.perm.iinet.net.au) |
| 23:43.18 | sandorp | [TK]D-Fender: can you look at http://www.pastebin.ca/798723 ? I've set the DYNAMIC_FEATURES variable, but still can't transfer; some of the docs imply this is for enabling features in the applicationmap; does that mean I need to add atxfer, blindxfer, etc. to the applicationmap somehow? |
| 23:43.48 | Linx | having a problem with asterisk not playing wav files submitted to it from a vxml browser |
| 23:44.10 | *** join/#asterisk irule (n=irule@200.53.61.4) |
| 23:44.11 | Linx | although it was working earlier until it was restarted |
| 23:44.23 | Linx | WARNING: format_wav.c:170 check_header: Can only handle 16bits per sample: 1 |
| 23:44.39 | Linx | hmm probably shouldn't paste the log here |
| 23:44.49 | *** join/#asterisk BZBW (n=wlwzhang@ip67-153-142-110.z142-153-67.customer.algx.net) |
| 23:45.07 | sandorp | Linx: perhaps the encoding of the wav file is not compatible? |
| 23:45.13 | beighto | if anybody is still interested, I found out what my problem was with the wanpipe. I didn't have my dependencies installed, ncurses, flex, and libtermcap-devel. |
| 23:45.37 | Linx | sandorp: I have tried many formats... nothing seem sot be working, even GSM |
| 23:45.58 | Linx | asterisk is version 1.4.14, on debian etch |
| 23:46.46 | sandorp | Linx: you can always paste your log on www.pastebin.ca |
| 23:46.57 | Linx | yea was jsut doing that |
| 23:48.00 | _Sam-- | Qwell : you here? |
| 23:48.11 | _Sam-- | as an answer to this bug report, the answer is, its not fixed: http://bugs.digium.com/view.php?id=10156 |
| 23:48.33 | _Sam-- | if you want to post a workaround, the workaround is, if you have that problem, and dont use udev on your system, rm /etc/udev |
| 23:48.34 | Linx | sandorp: http://www.pastebin.ca/798732 |
| 23:48.49 | Qwell | sounds like a debian bug IMO |
| 23:48.55 | sandorp | anyone else want to take a stab at figuring out why I still can't transfer calls: http://www.pastebin.ca/798723 |
| 23:49.04 | _Sam-- | anythings possible -- im on debian sarge, and had that problem. |
| 23:49.08 | _Sam-- | and tk-fender couldnt fix. |
| 23:49.55 | sandorp | Linx: are you sure these files exist? |
| 23:50.17 | Linx | sandorp: the voicexml browser downloads them to /tmp |
| 23:50.25 | Linx | then instructs asterisk to play the file |
| 23:50.45 | Linx | yes... I have done a ls with the name of the file |
| 23:50.46 | sandorp | Linx: >> /tmp/X2MqoQclAdCpYOVq22FZDy== (format ulaw): No such file or directory |
| 23:50.58 | *** join/#asterisk mirco (n=mirco@p54B2409F.dip.t-dialin.net) |
| 23:51.03 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 23:51.03 | *** mode/#asterisk [+o anthm] by ChanServ |
| 23:51.04 | Linx | it exists with the extention .wav |
| 23:51.26 | Linx | in /tmp |
| 23:51.49 | Linx | when it was working... in the vxml log... it had the same thing listed |
| 23:51.54 | Linx | without extention |
| 23:52.27 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 23:52.35 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 23:52.52 | sandorp | Linx: do you have a copy of a log when it worked to compare to? |
| 23:54.03 | sandorp | the "src=filename==.wav" looks sort of strange, but I'm just guessing really |
| 23:54.50 | Linx | nope |
| 23:54.57 | Linx | unfortunatlly |
| 23:55.17 | Linx | but I do recall it still had ..... |
| 23:55.25 | Linx | [Nov 27 11:51:13] WARNING[7125]: format_wav.c:170 check_header: Can only handle 16bits per sample: 1 |
| 23:55.31 | Linx | but not the other 2 lines |
| 23:56.00 | sandorp | and your permissions on the files in /tmp are ok relative to the user running asterisk? |
| 23:56.05 | *** join/#asterisk rpm (n=russell@d154-5-130-82.bchsia.telus.net) |
| 23:56.26 | Linx | yes, openvxi and asterisk both run as same user |
| 23:58.11 | Linx | files in /tmp are owned by the user that both openvxi and asterisk run as |
| 23:59.15 | sandorp | wonder why it says: "Play Stream: . " ? almost looks like it is trying to play the current directory "." |
| 23:59.44 | sandorp | is the ==.wav actually part of the file name? |