| 00:00.03 | bitboy | hello |
| 00:00.06 | Dr-Linux | carl0s-: :P |
| 00:00.19 | Dr-Linux | dlynes_laptop: now ask carl0s- what it does :) |
| 00:00.42 | dlynes_laptop | Dr-Linux, it didn't do anything |
| 00:00.57 | Dr-Linux | dlynes_laptop: you are not using mIRC |
| 00:01.03 | dlynes_laptop | of course |
| 00:01.04 | dlynes_laptop | not |
| 00:01.11 | dlynes_laptop | i'm using x-chat like normal people |
| 00:01.18 | carl0s- | it worked for me. I'm connected via mIRC. (ew.) I was up until recently running FC5 on my machine but I got so absolutely sick of Evolution crashing every other time I tried to check my Exchange mailbox, and the flash-plugin not working properly. |
| 00:01.20 | *** join/#asterisk Kumba_ (n=kumba@office.crashsys.com) |
| 00:01.21 | bitboy | Anyone know if following possible-->dial an extension which dials a number. As soon as that call hangs up, another number is dialed, without closing the channel after the firsts call |
| 00:01.21 | Dr-Linux | yes i know |
| 00:01.23 | Dr-Linux | [dlynes_laptop VERSION reply]: xchat 2.6.2 Windows XP [Intel /1.60GHz] |
| 00:01.31 | dlynes_laptop | it lies! |
| 00:01.35 | bitboy | so like an autodial |
| 00:01.39 | dlynes_laptop | i'm running Intel 1.66GHz |
| 00:02.03 | Dr-Linux | aww |
| 00:02.20 | *** part/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 00:02.21 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 00:02.47 | carl0s- | hmm xchat on windows. I tried X-Chat on my girlfriends Intel iMac and it wasn't great. I think the OSX X-server doesn't integrate 100% with the desktop there. |
| 00:03.53 | Dr-Linux | bbl |
| 00:04.06 | jbroome | carl0s-: i think there's an osx native xchat build |
| 00:04.18 | *** join/#asterisk rowter (n=ING@dsl-200-78-93-62.prod-infinitum.com.mx) |
| 00:04.31 | jbroome | xchat-aqua actually |
| 00:04.36 | dlynes_laptop | jbroome, yeah..i seem to recall seeing one when I was looking for one for windows |
| 00:04.42 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.18.32.Dial1.SanJose1.Level3.net) |
| 00:04.56 | rowter | a TE205 card should go green even tho there is no channel configuration right? |
| 00:05.00 | carl0s- | ah. I'll have a look at that when I'm next over there. She's finally got a wireless network again so I can use my thinkpad from now on when I'm there anyway. |
| 00:05.04 | *** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.18.32.Dial1.SanJose1.Level3.net) |
| 00:05.07 | dlynes_laptop | rowter, no |
| 00:05.27 | Kumba_ | It should quit scrolling if the driver is loaded correctly... right? |
| 00:05.30 | dlynes_laptop | rowter, it'll only go green if you have your zaptel.conf set up, and you've run ztcfg -vvvvvvvvvv |
| 00:05.42 | rowter | dlynes_laptop, it will go red ? if there is no channel config? oohh |
| 00:05.50 | rowter | dlynes_laptop, let me set it up then.. |
| 00:05.53 | dlynes_laptop | Kumba_, to what do you refer? |
| 00:06.00 | Kumba_ | TE205p... |
| 00:06.17 | dlynes_laptop | Kumba_, what should quit scrolling? |
| 00:06.24 | Kumba_ | the lights will quit scrolling once a driver has loaded and correctly found the card... |
| 00:06.29 | wunderkin | i think he means the night rider thing |
| 00:06.34 | dlynes_laptop | ah |
| 00:06.36 | Kumba_ | Yeah, nightrider... |
| 00:06.38 | dlynes_laptop | yeah, correct |
| 00:06.43 | Kumba_ | sorry... forgot the technical terms :) |
| 00:06.48 | Strom_C | damn you, it was Knight Rider :) |
| 00:06.53 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 00:06.53 | *** mode/#asterisk [+o anthm] by ChanServ |
| 00:06.58 | *** part/#asterisk carrar (i=tim@osburn.com) |
| 00:07.02 | wunderkin | ok well im not familiar with it |
| 00:07.09 | Strom_C | KITT |
| 00:07.17 | Strom_C | am I the only one who actually watched the show? :) |
| 00:07.29 | Kumba_ | I watched it... but I was 10... |
| 00:07.33 | Strom_C | and I was 7 |
| 00:07.38 | droops | i never watched it |
| 00:07.40 | Strom_C | so you have no excuse :) |
| 00:07.49 | droops | and im older than strom |
| 00:07.52 | Kumba_ | I blame the molten hops and bong residues? |
| 00:08.18 | dlynes_laptop | Strom_C, no...i used to watch it every week |
| 00:08.29 | Strom_C | awesome |
| 00:08.37 | dlynes_laptop | Strom_C, then david hasselhoff defected and started babewatch |
| 00:08.39 | Kumba_ | The only thing I remember from that show was the car and Goliath... |
| 00:08.56 | *** join/#asterisk carl0s- (n=carl@compsup.demon.co.uk) |
| 00:09.11 | dlynes_laptop | Kumba_, how about the limey? |
| 00:09.13 | Strom_C | dlynes_laptop: they filmed 'baywatch' in my neighborhood growing up |
| 00:09.20 | Kumba_ | Dont remember the limey... |
| 00:09.42 | Strom_C | dlynes_laptop: i can clearly recall driving down PCH in the morning in february and seeing them running around in bathing suits in the cold |
| 00:09.43 | dlynes_laptop | Strom_C, well, the blond grew up in my neighborhood |
| 00:09.59 | dlynes_laptop | Strom_C, the one that starred in the pr0n with the motley crue guy |
| 00:10.09 | Kumba_ | I'm sure i'm about to get a beat-down for asking... but should I even bother with Trixbox when I plan to bring a T1 into one side of my TE205p and put a channel bank on the other? Or should I just start from source? |
| 00:10.13 | dlynes_laptop | pch? |
| 00:10.14 | Strom_C | jennifer titsalot? |
| 00:10.17 | Kumba_ | dlynes: Pamela Anderson? |
| 00:10.24 | dlynes_laptop | Kumba_, correct |
| 00:10.30 | dlynes_laptop | Kumba_, she's from down the road |
| 00:10.33 | Strom_C | Kumba_: dont bother with trixbox |
| 00:10.38 | dlynes_laptop | Kumba_, white rock, bc |
| 00:10.48 | rowter | dlynes_laptop, does asterisk needs to be off ? so ztcfg works correctly? |
| 00:11.02 | Kumba_ | Strom: Kewl... |
| 00:11.13 | dlynes_laptop | rowter, do you have chan_zap.so loaded? |
| 00:11.33 | Kumba_ | So if I go from Source... is CentOS still the linux variant of choice for *? |
| 00:11.37 | rowter | dlynes_laptop, yeah, I loop the card and it goes green.. |
| 00:12.01 | Strom_C | Kumba_: ew no |
| 00:12.02 | dlynes_laptop | loop the card? you mean put a loopback mod end in the port? |
| 00:12.14 | Strom_C | Kumba_: you can use whatever distro you like best |
| 00:12.18 | wunderkin | dlynes_laptop, you can do a remote loopback now |
| 00:12.19 | Kumba_ | I used to use slack before I converted to OBSD... |
| 00:12.20 | Strom_C | Kumba_: I happen to like debian |
| 00:12.32 | dlynes_laptop | wunderkin, remote loopback? |
| 00:12.51 | Kumba_ | Any preference as far as 2.4/2.6 kernel? |
| 00:12.55 | dlynes_laptop | Kumba_, come back into the fray |
| 00:12.57 | Strom_C | 2.6 definitely |
| 00:13.03 | *** join/#asterisk NoRemorse (n=bah@210-84-10-183.dyn.iinet.net.au) |
| 00:13.08 | NoRemorse | hi all |
| 00:13.14 | Kumba_ | is debian a somewhat minimalist install? |
| 00:13.17 | dlynes_laptop | Kumba_, yeah...slack 10.2 upgraded to a 2.6 kernel is the total shizzit |
| 00:13.17 | rowter | dlynes_laptop, it was a fisic loop.. set the cables together and it goes green.. but with the E1 it goes blinking.. |
| 00:13.22 | wunderkin | dlynes_laptop, well maybe im not using the right words but you can remotely loop the csu now |
| 00:13.30 | Kumba_ | I go into epilleptic seizures everytime I watch CentOS boot... |
| 00:13.35 | Strom_C | Kumba_: you can do a very minimal debian install |
| 00:13.50 | dlynes_laptop | whatever fisic means |
| 00:15.01 | dlynes_laptop | rowter, anyways...what I would do is shutdown asterisk, make sure your driver is loaded, and then do a ztcfg -vvvvvvvvvv |
| 00:15.14 | dlynes_laptop | rowter, then do a dmesg to make sure you didn't get any errors |
| 00:15.19 | NoRemorse | I am using firefly/IAX as a softphone, and when I call the client, iax2 debug shows the call being rejected by the client due to no compatable codecs, yet the calls etup shows it is trying to use: CODEC_PREFS : (g729|gsm|ulaw|alaw) (all codes are ticked in firefly config) any clues please? |
| 00:15.31 | dlynes_laptop | rowter, then make sure your zapata.conf channel configuration matches your zaptel.conf channel configuration |
| 00:15.41 | dlynes_laptop | rowter, then load safe_asterisk |
| 00:15.48 | Strom_C | NoRemorse: what does your iax.conf show? |
| 00:16.02 | NoRemorse | it allows all those codecs |
| 00:16.34 | NoRemorse | I also have an iaxy behind the same nat firewall it works fine |
| 00:17.10 | *** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-42-234.losaca.adelphia.net) |
| 00:17.35 | Kumba_ | dlynes: I need both slack CD's to install? or is disk 2 the kernel 2.6 version? |
| 00:19.17 | carl0s- | What's the deal with the Molex power connector on the TDM400P? Is it only needed if you have FXS modules, or if you have more than one module, or what. I just though, but I'm sure I didn't connect it up on mine, and it's working (apart from unbearable echo - I can hear myself through my earpeice). |
| 00:19.25 | *** part/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net) |
| 00:19.33 | hads | carl0s-: It's for FXS modules. |
| 00:19.40 | Kumba_ | carl0s: My understanding is FXS mod's... |
| 00:19.40 | Strom_C | carl0s-: connect it anyway |
| 00:19.48 | carl0s- | hads, thanks |
| 00:19.53 | carl0s- | Strom_C, OK. I will do. |
| 00:19.55 | *** join/#asterisk awe6 (n=lba@user-12lml5g.cable.mindspring.com) |
| 00:20.20 | hads | Yeah, like Strom_C said though, connect it as I have sometimes has weird issues when loading modules without it connected. |
| 00:20.46 | hads | s/has/had/ |
| 00:21.37 | NoRemorse | on a different topic I am getting this alot from a client and it then becomes unreachable and re-registers Unknown SIP command 'SI16384P/2.0' from '202.161.34.45 |
| 00:21.48 | carl0s- | Right. I'll check it in a min. just trying to find an acceptable font for X-Chat/win32. |
| 00:21.56 | NoRemorse | always the same trash |
| 00:23.23 | jbroome | carl0s-: http://www.silverex.org/download/ |
| 00:23.25 | Strom_C | sounds like the client has gone bonkers |
| 00:23.57 | carl0s- | jbroome, that's the version I'm using. Thanks though. The other one was the official xchat.org non-free version. |
| 00:24.01 | NoRemorse | it works apart fromt hat lol |
| 00:24.10 | *** join/#asterisk jbsolutios (n=jbenson@193.93.153.1) |
| 00:24.19 | andrejkw | Umm for some reason my PAP2 Line 1 always comes up as "Busy Here". Can someone help? |
| 00:25.18 | carl0s- | jbroome, that nice default font looks all blurry at size 10 on my system. size 9 is too small. I'm using courier normal now but it's a bit thin and badly spaced |
| 00:25.56 | Strom_C | carl0s-: Lucida Console |
| 00:26.45 | Kumba_ | Mmmm... torrent downloads... *watches his T1 croak* |
| 00:27.00 | andrejkw | Ok, I did *60 by accident, how do I undo? |
| 00:27.03 | carl0s- | Strom_C, that's not too bad. thanks :D |
| 00:27.21 | Kumba_ | Anyone got any experience with Polycom 301's? |
| 00:27.36 | NoRemorse | can anyone recomend another iax2 softphone please? |
| 00:27.40 | jbsolutios | hi all - I see that www.dundi.com is pointing to thevoice.digium.com? |
| 00:28.12 | jbsolutios | is that intentional? |
| 00:29.22 | *** join/#asterisk Dibbler_ (n=Dibbler@dsl-217-155-254-174.zen.co.uk) |
| 00:31.30 | Kumba_ | the span command is span = <card>,<Span>,<Timing>,<coding>,<framing> right? |
| 00:34.08 | wunderkin | the syntax is in the file.. |
| 00:35.13 | wunderkin | it is in file..... omg... watch out |
| 00:36.03 | Kumba_ | I dont see it in there... but i'll google around... |
| 00:36.36 | wunderkin | # span=<span num>,<timing>,<line build out (LBO)>,<framing>,<coding>[,yellow] |
| 00:37.34 | wunderkin | there are also examples in there |
| 00:38.17 | Kumba_ | well i've got a trixbox file... doesn't have anything... i'm just trying to get the lights to go green on my card while i'm waiting for slack to download so that I dont have to figure the settings out later :) |
| 00:38.21 | Kumba_ | I do appreciate the paste tho |
| 00:38.39 | Strom_C | Kumba_: like we said: eewwww trixbox |
| 00:38.45 | wunderkin | trix are for kids! |
| 00:38.55 | Strom_C | also the lights wont go green unless there's a circuit or a loopback connected |
| 00:38.56 | Kumba_ | I know... but ztcfg is still the same... kinda :) |
| 00:39.09 | Kumba_ | I have an RBS T1... |
| 00:39.16 | Strom_C | RBS? |
| 00:39.22 | Kumba_ | Robbed Bit Signalling... |
| 00:39.27 | Strom_C | ewwwwwwwwwwwwwwwwww |
| 00:39.32 | Kumba_ | Full-Channel T1... ESF/B8ZS... |
| 00:39.48 | Kumba_ | It's an existing circuit... I didn't feel like going through the hassle of having a PRI dropped in... |
| 00:39.48 | Strom_C | PRI for the win :) |
| 00:40.14 | wunderkin | span=1,1,0,esf,b8zs like that i think |
| 00:40.15 | Kumba_ | 20/20 Hindsight... |
| 00:40.31 | Kumba_ | Yeah... that's what i've come up with after your paste :) |
| 00:40.36 | Kumba_ | signalling = fxsls |
| 00:40.45 | Kumba_ | err that's zapata |
| 00:40.52 | Kumba_ | fxsls = 1-24 |
| 00:41.17 | *** part/#asterisk kcortez (n=kcortez@208.49.103.100) |
| 00:48.40 | *** join/#asterisk eDIsonxl (n=xian-lia@mail.kinyo.com.tw) |
| 00:52.38 | carl0s- | hmm. interesting. "The first thing to check if you experience echo cancellation with analogue (eg TDM400) cards is that the PSTN loadzone is set correctly. For instance if running in (default) FCC mode, and you are connected to a UK PSTN line, then you *will* observe harsh echo. You will need to change to using UK mode (see TDM400P) in this instance." |
| 00:53.06 | carl0s- | (I'm in the UK. when wctdm loads, it does say it's using FCC something-or-other). I have bad echo on calls out of the TDM400. |
| 00:53.41 | wunderkin | i think it means loadzone line in /etc/zaptel.conf |
| 00:54.05 | carl0s- | when I ran genzaptelconf, I did specify "-c uk". but i'll check now. |
| 00:54.24 | *** join/#asterisk NoRemorse (n=bah@210-84-10-183.dyn.iinet.net.au) |
| 00:54.37 | NoRemorse | hi all, has the voicemail.conf file format changed between 1.0 and 1.2? |
| 00:54.42 | carl0s- | yup. loadzone = uk. bugger. |
| 00:54.54 | *** join/#asterisk Druken (i=Druken@CPE00121716da99-CM00159a090acc.cpe.net.cable.rogers.com) |
| 00:54.59 | carl0s- | fxsks=4 ? |
| 00:56.07 | Druken | if i set nat=yes shouldn't it show a Y in the sip show peers? |
| 00:56.08 | *** join/#asterisk Splat (n=Splat@220-253-134-28.TAS.netspace.net.au) |
| 00:57.02 | Kumba_ | yay... green lights :) |
| 00:58.43 | NoRemorse | hi all, has the voicemail.conf file format changed between 1.0 and 1.2? logs are saying voicemail box blah not found in config file and it is there... |
| 00:59.19 | *** join/#asterisk Azrael (n=Azrael@orion.negativeblue.com) |
| 00:59.24 | Strom_C | NoRemorse: man, 1.0 was sooo long ago |
| 00:59.58 | NoRemorse | lol |
| 01:00.08 | NoRemorse | but voicemail.conf didnt change did it? |
| 01:00.12 | wunderkin | well you can check the sample file in /usr/src/asterisk/doc/configs or something like that |
| 01:00.41 | NoRemorse | looks the same :( |
| 01:00.53 | NoRemorse | i'll diff it on the 1.0.11 sample hehe |
| 01:00.55 | carl0s- | "While the TDM400P is certified for many countries, simply specifying the loadzone will not set your card up to suit local line imedences etc. This can lead to echo or bad sound quality even with agressive echo cancelation. " |
| 01:01.23 | hads | carl0s-: It's probably talking about the OPERMODE parameter when loading the module |
| 01:01.24 | Strom_C | NoRemorse: are you sending it to the right voicemail context? |
| 01:01.25 | carl0s- | sems I need to edit wctdm.c . Can I not just specify a parameter to the module? |
| 01:02.04 | carl0s- | hads, that'd be better. The document I'm reading says I need to edit wctdm.c and recompile. I guess they've made it easier now then? |
| 01:02.16 | Kumba_ | two green lights... this is the most fun i've had with this thing all night... (LOL) |
| 01:02.22 | Kumba_ | Time for dairy queen |
| 01:02.26 | NoRemorse | ahh there are voicemail contexts?! |
| 01:02.53 | hads | carl0s-: Yes, you can specify the OPERMODE parameter in /etc/modprobe.d/zaptel or /etc/modprobe.conf or whatever it is on $DISTRO |
| 01:03.05 | carl0s- | hads, oh, sorry. I should have carried on reading. You're quite right. The document was telling me to *look* at the source to see the available OPERMODE options |
| 01:03.06 | NoRemorse | is it 400|default or 405,default ? |
| 01:03.09 | Druken | has anyone gotten a wip300 connected and working properly on asterisk ? |
| 01:03.24 | *** part/#asterisk rene- (n=rene-@dsl-200-67-175-250.prod-empresarial.com.mx) |
| 01:03.36 | Strom_C | 400@default |
| 01:03.43 | NoRemorse | ty |
| 01:04.27 | hads | carl0s-: This is mine: install wctdm /sbin/modprobe --ignore-install wctdm opermode=NEWZEALAND boostringer=1 fastringer=1 && sleep 1 && /sbin/ztcfg |
| 01:04.51 | *** join/#asterisk rene- (n=rene-@dsl-200-67-175-250.prod-empresarial.com.mx) |
| 01:05.15 | carl0s- | hads, thanks. I'm just reading about boostringer now. |
| 01:05.27 | rene- | if an agent is logged of when in pause, when it logs back in will it be in paused state? |
| 01:06.04 | NoRemorse | so it is still Voicemail(u400) syntax? |
| 01:06.22 | wunderkin | NoRemorse, that changed recently but you want to specify the vm context there too |
| 01:06.30 | Strom_C | Voicemail(400@default,u) |
| 01:06.34 | NoRemorse | ahhh ok ty |
| 01:07.20 | NoRemorse | damn still not finding it |
| 01:07.47 | Strom_C | NoRemorse: pastebin your voicemail.conf and the exact error |
| 01:08.17 | NoRemorse | ah ok fixed it had wqrong context, NOW.. it is not using my old voice mail reocrdings ? |
| 01:08.55 | *** join/#asterisk kodok (n=makoata@bb219-74-196-86.singnet.com.sg) |
| 01:09.44 | NoRemorse | hmm new vm dir lol |
| 01:11.02 | NoRemorse | thanks guys |
| 01:11.19 | *** part/#asterisk jbsolutios (n=jbenson@193.93.153.1) |
| 01:12.35 | carl0s- | YAY. Module 3: Installed -- AUTO FXO (UK mode) |
| 01:13.13 | carl0s- | see how it sounds tomorrow. 2am is too late to be ringing people |
| 01:13.18 | *** join/#asterisk slayer192 (n=slayer19@adsl-71-146-227-206.dsl.okcyok.sbcglobal.net) |
| 01:16.45 | rene- | Qwell you are the acd resident expert, do you know weather an agent that was paused and then logged off from a queue will be in either paused or unpaused state whenever it logs back in?? |
| 01:21.09 | carl0s- | has anyone ever seen a miniPCI -> PCI adapter? It's the reverse of the normal PCI->miniPCI adapters. e.g. I want to try a TDM400P in a mini-pci slot |
| 01:22.20 | rene- | carl0s i havent, i remember looking for them in the past but didnt find nothing, i wanted to do the same that you |
| 01:22.20 | [andromeda] | I guess you can't use the free gizmo PSTN number with asterisk, every time i call in, i see the call on asterisks, but then i get redirected immediately to Gizmo's callwave voicemail |
| 01:23.49 | carl0s- | rene-, oh well. These guys apparently have some interesting miniPCI stuff. miniPCI u160 scsi? :) http://axiomtek.industrialpartner.com/industrial-pc/ |
| 01:25.06 | rene- | there are some oscure devices like miniPCI video cards and such, |
| 01:25.18 | rene- | minipci voice cards would surely be cool |
| 01:26.03 | carl0s- | rene-, Junghanns has a miniPCI quadBRI ISDN card which will be available for purchase in less than a month |
| 01:26.21 | rene- | sweet |
| 01:26.27 | rene- | those junghanns guys are cool |
| 01:26.35 | carl0s- | yeh, but no analog stuff :) |
| 01:26.37 | rene- | the gsm stuff is also really cool |
| 01:26.40 | carl0s- | yeah |
| 01:26.48 | *** join/#asterisk [Airwolf] (n=airwolf@dsl51B67B23.pool.t-online.hu) |
| 01:27.22 | rene- | they should do analog, i mean not everybody has fancy isdn bri at home or at their pbx heh |
| 01:27.38 | *** join/#asterisk gandhijee (n=gandhije@c-24-147-213-182.hsd1.ma.comcast.net) |
| 01:27.49 | carl0s- | OH YEAH. http://www.globalamericaninc.com/new_spec/spec2.php?id=748 |
| 01:29.32 | rene- | dude |
| 01:29.35 | rene- | those are awesome |
| 01:29.50 | *** join/#asterisk W9SH (n=W9SH@adsl-068-209-117-205.sip.asm.bellsouth.net) |
| 01:30.25 | rene- | are core duo/solo mini itx motherboards available? |
| 01:30.36 | carl0s- | I don't know if the miniPCI slot on a WRAP board is PCI 2.2, but if I can get one of those cheap enough I'll be trying my TDM400P on a WRAP embedded board. |
| 01:30.48 | rene- | most centrino motherboards have minipci instead of pci |
| 01:31.12 | rene- | cool |
| 01:31.41 | carl0s- | rene-, I'm not sure. I'm trying to not use PC hardware hence the WRAP board, but it's only for playing. I think LEX might have had a centrino mini-itx board. |
| 01:31.47 | dlynes_laptop | carl0s-, yeah, it's PCI 2.2; it's identical to regular PCI, but it only comes in 3.3V |
| 01:32.08 | dlynes_laptop | carl0s-, there's even miniPCI->PCI and PCI->miniPCI converters |
| 01:32.32 | rene- | but wrap does have full size pci doesnt it? or that was pc engines? |
| 01:32.45 | hads | One of the Soekris boards (4801?) has PCI |
| 01:32.52 | carl0s- | dlynes_laptop, cool. well, that miniPCI -> PCI convertor says it gives 2x PCI. 1 @ 3.3v and 1@5v. I don't know if that means the host miniPCI slot needs to be 5v though. |
| 01:33.13 | carl0s- | rene-, I think there might be a WRAP board with a PCI connector on it's edge, but not the one I've ordered. |
| 01:33.15 | dlynes_laptop | carl0s-, there's no such thing as 5V miniPCI |
| 01:33.43 | rene- | i saw the astlinux guy demo one of those at astricon |
| 01:33.47 | dlynes_laptop | carl0s-, so it probably uses a transformer to boost up the voltage to 5V |
| 01:34.05 | carl0s- | that's good then. I already knew of the PCI -> miniPCI adapters, but hadn't seen any the other way round for using full size PCI cards from a minPCI slot. |
| 01:34.19 | Druken | w00t! |
| 01:34.36 | Druken | finally got my WIP300 to register from behind the nat |
| 01:34.36 | *** join/#asterisk tlow (n=tlowe@bgp.terrorist.net) |
| 01:34.52 | dlynes_laptop | Druken, congrats |
| 01:35.09 | Druken | thanks :) |
| 01:37.38 | *** join/#asterisk pigpen2 (n=mark@fw.seamans.cc) |
| 01:37.39 | *** part/#asterisk tlow (n=tlowe@bgp.terrorist.net) |
| 01:37.48 | *** join/#asterisk digime (n=digime@75.8.126.175) |
| 01:37.59 | eDIsonxl | Hi |
| 01:38.16 | eDIsonxl | I am testing on my asterisk box |
| 01:38.49 | digime | anyone recommend a good incoming local DID provider? |
| 01:38.57 | eDIsonxl | but when I called , returning 'Forcing Marker bit, because SSRC has changed' |
| 01:39.25 | eDIsonxl | Then I had a one-way audio seesion |
| 01:39.37 | dlynes_laptop | digime, usually helps if you define 'local' |
| 01:39.49 | dlynes_laptop | digime, not everybody on here lives in the same city as you |
| 01:39.53 | eDIsonxl | How could I solve this problem |
| 01:40.19 | dlynes_laptop | [TK]D-Fender, welcome bakdc |
| 01:40.22 | [TK]D-Fender | y0 |
| 01:40.24 | *** join/#asterisk pdtmobile (n=ptinsley@c-68-53-40-50.hsd1.tn.comcast.net) |
| 01:40.48 | digime | dlynes_laptop: yes, let me give details: san diego 619 or 858 |
| 01:41.13 | *** join/#asterisk ivanfm (n=ivanfm@201.52.129.236) |
| 01:48.41 | eDIsonxl | Have anybody met a problem like I described? |
| 01:49.26 | rene- | dlynes_laptop: how expensive is that device, the minipci-pci adapter |
| 01:49.48 | file | moo |
| 01:49.54 | Qwell | moo? |
| 01:50.13 | file | or oom, but I'm not out of memory |
| 01:50.28 | dlynes_laptop | rene-, no idea...never bought one...i don't have any systems with miniPCI slots |
| 01:50.53 | rene- | they are not very commercial |
| 01:52.25 | [andromeda] | hmm, this is the same problem I am experiencing: http://forum.sipphone.com/viewtopic.php?t=1855&start=0 |
| 01:54.13 | *** join/#asterisk bitboy (n=amit@adsl-065-012-197-229.sip.bct.bellsouth.net) |
| 01:55.05 | *** part/#asterisk SkramX (n=MarkS@admins.sentiensystems.net) |
| 01:55.50 | bitboy | anyone tried predictive dialing in asterisk? Any suggestions |
| 01:56.37 | rene- | write your own, b) try the opensource ones like gnuidialer and vicidiarler |
| 01:56.50 | rene- | or stuff like sine dialer |
| 01:56.56 | rene- | commercial |
| 01:57.17 | bitboy | vicidialer doesnt seem to have it implemented yet. I would like to write one...but cant think of algorithm for autodialing |
| 01:57.29 | rene- | well people have been developing those for years |
| 01:58.04 | rene- | but basically it amounts to poll the pbx for connected users, available users and average call times |
| 01:58.33 | bitboy | within asterisk though....the idea is to dial a number after the current call ends...but how to do this without closing the channel after first call? |
| 01:58.53 | rene- | well if you only have one channel you cant do predictive |
| 01:59.16 | rene- | predictive dials ahead so you can have ready calls before or at the same time your current call ends |
| 01:59.39 | rene- | if you dial after the current call ends then you are doing what is called progressive |
| 01:59.46 | rene- | or automatic dialer |
| 01:59.56 | rene- | predictive dialer screens for busy signals |
| 02:00.04 | bitboy | ah. Ok then how to do progressive dialing :) |
| 02:00.04 | rene- | fax signals |
| 02:00.23 | rene- | the easiest way would be to use the call files |
| 02:00.32 | rene- | they wont get triggered |
| 02:00.43 | rene- | unless there are available channels |
| 02:02.33 | bitboy | but again...lets say I dial a "1" on my phone...this opens an extension which picks a number from a queue and dials it. How do I not have the channel close at the end of the call before picking off next number? |
| 02:03.43 | rene- | you want to trigger dialing using AGI |
| 02:04.26 | bitboy | hmm...I guess I better read up on AGI and call files....I have only set up a basic phone network with * |
| 02:04.42 | rene- | real man use manager interface |
| 02:04.50 | rene- | just kidding |
| 02:05.04 | rene- | i think call files are the way to go |
| 02:05.17 | bitboy | Oh cool...well thanks for the ideas |
| 02:05.17 | rene- | for progressive |
| 02:05.27 | rene- | your welcome |
| 02:08.50 | *** join/#asterisk trbldwine (n=trbldwin@c-71-194-161-170.hsd1.il.comcast.net) |
| 02:12.40 | Hmmhesays | anyone here good with as5300's? |
| 02:16.38 | [TK]D-Fender | Hmmhesays : I can throw them pretty far..... |
| 02:16.52 | *** join/#asterisk ivanfm (n=ivanfm@201.52.129.236) |
| 02:17.27 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 02:17.34 | shmaltz | tzafrir ping |
| 02:17.47 | shmaltz | tzafrir_laptop ping |
| 02:18.48 | Hmmhesays | [TK]D-Fender yeah |
| 02:18.51 | Hmmhesays | that doesn't help me |
| 02:19.08 | *** join/#asterisk NoRemorse (n=bah@210-84-10-183.dyn.iinet.net.au) |
| 02:19.18 | NoRemorse | hi all, can anyone recomend a decent iax softphone please? |
| 02:19.49 | wunderkin | softphones es teh suk |
| 02:20.09 | NoRemorse | yeah, but i still need 1 |
| 02:20.10 | Kumba_ | softphones... sounds like something from a Skinimax movie... |
| 02:20.35 | rob0 | haha, I've got a Linux console-based softphone which does SIP and IAX and call routing. |
| 02:20.55 | rob0 | It's called ... asterisk |
| 02:21.01 | dlynes_laptop | rob0, heh |
| 02:21.09 | NoRemorse | ffs |
| 02:21.34 | Kumba_ | I got green lights... that's all I care about... |
| 02:21.35 | *** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-42-234.losaca.adelphia.net) |
| 02:21.36 | rob0 | No kidding, I just set that up on my laptop for that very purpose. |
| 02:21.43 | Kumba_ | atleast until this ISO finishes downloading... |
| 02:21.59 | Kumba_ | use your sound card as an extension? |
| 02:22.13 | NoRemorse | stupid firefly keeps ringing when I click answer call |
| 02:22.28 | dlynes_laptop | NoRemorse, try ekiga? |
| 02:22.54 | *** join/#asterisk s0lid (n=s0lid@210.213.199.63) |
| 02:22.59 | NoRemorse | thansk will have a look |
| 02:23.11 | dlynes_laptop | NoRemorse, it only runs on linux, just so you know |
| 02:23.24 | NoRemorse | canceling link now... |
| 02:23.31 | Kumba_ | So green lights on my TE205p means that I have a good circuit? (IE, all that's left is to make asterisk work) |
| 02:23.32 | *** join/#asterisk fnordus (n=dnall@s142-179-111-243.bc.hsia.telus.net) |
| 02:23.44 | dlynes_laptop | Kumba_, correct |
| 02:23.50 | Kumba_ | Or can I get green lights and still have a messed up circuit? |
| 02:24.03 | dlynes_laptop | NoRemorse, heh...well that's why it works better...it doesn't run on windoze :) |
| 02:24.15 | dlynes_laptop | Kumba_, nah...green means all circuits are go |
| 02:24.22 | Kumba_ | sweet... |
| 02:24.34 | wunderkin | well things can still go wrong |
| 02:24.34 | Kumba_ | now if I can get this whole asterisk thing to work... i'll be set... |
| 02:24.39 | dlynes_laptop | so if you get asterisk fired up and it doesn't work |
| 02:24.52 | dlynes_laptop | then that means your zapata.conf is screwed up, usually |
| 02:25.06 | dlynes_laptop | but you could have hardware issues, too...and green light will mask that |
| 02:25.36 | Kumba_ | I guess I crimped my T1 crossover cable right too :) |
| 02:25.39 | dlynes_laptop | zaptel.conf and zapata.conf are not hte same |
| 02:25.43 | Kumba_ | good thing... only got 1 connector left |
| 02:26.03 | Kumba_ | yeah... but if my zaptel.conf looks good, I can just copy/paste it back in after I format and get rid of trixbox... |
| 02:29.16 | Kumba_ | Would I want to set channel 1-24 coming in from my T1 as fxsls in zaptel.conf, and channel 25-48 to fxols that is going out of my TE205p to the channel bank? |
| 02:29.16 | dlynes_laptop | ah |
| 02:29.23 | Kumba_ | or do I have that backwards? |
| 02:29.23 | dlynes_laptop | yeah....lose that piece of kruft |
| 02:29.52 | dlynes_laptop | Kumba_, it depends on what kinda signalling you're using |
| 02:30.06 | dlynes_laptop | Kumba_, do you not have disconnect supervision? |
| 02:30.40 | Kumba_ | Mmmmmm... disconnect supervision? (not sure exactly what you mean) |
| 02:30.50 | Strom_C | ...on a T1? |
| 02:31.06 | Strom_C | t1 just has to set the supervision bit on again |
| 02:31.27 | dlynes_laptop | Strom_C, he said he's got 24 channels coming in...i figured it was an analog t1 |
| 02:31.57 | Strom_C | "analog t1" |
| 02:32.01 | Kumba_ | It's esf/b8zs with loopstart... |
| 02:32.20 | Strom_C | thats like saying digital LP |
| 02:32.25 | Strom_C | there's no such thing |
| 02:32.34 | dlynes_laptop | LP? you mean 33-1/3? |
| 02:32.44 | Strom_C | theres CAS T1 and theres PRI, but there's not "analog T1" |
| 02:33.00 | Kumba_ | This isn't a PRI... |
| 02:33.02 | Kumba_ | unfortunately |
| 02:37.04 | bkw_ | Strom_C, well if you run FXO or FXS over a T1 what is that called? |
| 02:37.09 | bkw_ | like to a channel bank? |
| 02:37.27 | Strom_C | PCM :) |
| 02:37.32 | bkw_ | true |
| 02:37.53 | Strom_C | the T1 is digital. You may have analog inputs and/or outputs, but the T1 itself is digital. |
| 02:38.35 | Kumba_ | So... the 23 channels that are coming into the T1 I would want to setup as FXSLS in zaptel.conf? |
| 02:38.38 | Kumba_ | err 24 |
| 02:38.46 | Strom_C | yes |
| 02:39.20 | Kumba_ | and the next set of 24 channels that i'm coming out of asterisk and going to the channel bank would be FXOLS? |
| 02:39.48 | Kumba_ | or do I set that as FXSLS because I want to send FXSLS out of the channel bank? |
| 02:39.49 | Strom_C | I /think/ you can set your FXS ports on the channel bank to use FXO_KS, but dont quote me on that since I don't remember what I used for the only channel bank job I've ever done |
| 02:40.07 | Strom_C | mmm, waffle house |
| 02:40.13 | Kumba_ | dood... |
| 02:40.16 | Kumba_ | waffle house... |
| 02:40.26 | Strom_C | scattered covered diced |
| 02:40.30 | Strom_C | with tabasco |
| 02:40.34 | Strom_C | and coffee |
| 02:40.35 | Kumba_ | Scattered covered chunked |
| 02:40.37 | Strom_C | drooooooooool |
| 02:41.03 | Kumba_ | vi wafflehouse.conf |
| 02:41.33 | Strom_C | i think 75% of the reason i like waffle house so much is that I'm nowhere near one |
| 02:42.29 | *** join/#asterisk }btorch{ (n=kvirc@c-66-176-87-59.hsd1.fl.comcast.net) |
| 02:42.50 | }btorch{ | hey guys , what's up wioth asterisk and IAX2 ? |
| 02:43.24 | Strom_C | }btorch{: I don't know. What /is/ up with asterisk and IAX2? |
| 02:43.41 | }btorch{ | I got a p4 system which I set that uses iax2 and zap ... I have bought USB headsets and the iax2 to iax2 link is still kind of bad |
| 02:43.53 | Strom_C | }btorch{: which version of asterisk |
| 02:44.38 | }btorch{ | the linke from iax2 to zap is not ok either ... some times the connection is ok iax-to-iax and iax-to-zap but other times during the call the other end sounds like a robot talking |
| 02:44.49 | Strom_C | }btorch{: which version of asterisk |
| 02:44.52 | }btorch{ | 1.2.7 |
| 02:44.57 | }btorch{ | 1.2.7.1 |
| 02:45.05 | Strom_C | pastebin your iax.conf files |
| 02:45.10 | Strom_C | and consider upgrading to 1.2.10 ;) |
| 02:45.27 | }btorch{ | I don't think that is it though |
| 02:45.40 | Strom_C | pastebin your iax.conf files |
| 02:45.45 | }btorch{ | what about using sip ? is iax more cpu intensive ? |
| 02:45.50 | }btorch{ | ok |
| 02:45.50 | Strom_C | pastebin your iax.conf files |
| 02:46.17 | }btorch{ | btwe is www.asteriskgur.com offline ? |
| 02:46.18 | file | Strom_C: keep it gay! |
| 02:46.23 | *** join/#asterisk JunK-Y (n=junky@modemcable205.175-81-70.mc.videotron.ca) |
| 02:46.43 | }btorch{ | hehe |
| 02:46.52 | rob0 | "Up" is ^^ that way, altho it varies with global position. (My "up" might be down for an Asian.) |
| 02:47.37 | *** join/#asterisk tempest1 (n=asf@adsl-153-33-178.chs.bellsouth.net) |
| 02:49.35 | Strom_C | longest pastebin time in the history of pastebin |
| 02:50.01 | }btorch{ | on totaly unrelated question is it possible for a user to be registered like this ? 1506/lwaski 60.18.224.22 (D) 255.255.255.255 34674 |
| 02:50.15 | Strom_C | }btorch{: do you want me to help solve your problem or not? |
| 02:50.25 | }btorch{ | that's what iax2 show peers display for this user |
| 02:50.38 | }btorch{ | Strom_C: I'm pasting the file |
| 02:50.44 | Strom_C | use pastebin.ca |
| 02:50.46 | Strom_C | it's faster |
| 02:50.51 | Strom_C | eh |
| 02:51.14 | tempest1 | rafb.net is good too |
| 02:51.28 | file | whip it into shape! |
| 02:51.35 | Strom_C | shape it up! |
| 02:51.47 | file | actually I'm in a Rent mood... |
| 02:51.52 | file | La Vie Boheme! |
| 02:55.58 | JunK-Y | yay |
| 02:56.04 | Strom_C | I bet I could play every single one of my Cars albums before }btorch{ pastebins his iax.conf files |
| 02:57.42 | Strom_C | i dont mind you coming here / and tying up my line / cause when you're dialing oh so near / i hear the tones just fine |
| 02:58.01 | Strom_C | it's not the buttset that you wear / your thousand feet of twisted pair / i don't mind you coming here / and tying up my line |
| 02:58.11 | Strom_C | [instrumental break] |
| 02:58.37 | pdthome | scattered covered smothered and chunked the occasional diced |
| 02:59.13 | Strom_C | ewww, fake cheese |
| 02:59.32 | pdthome | lol |
| 02:59.58 | pdthome | it's waffle house for the love of god, it's not like the rest of it is fine dining |
| 03:00.13 | }btorch{ | here http://pastebin.ca/91997 |
| 03:00.13 | Strom_C | well, true |
| 03:00.16 | }btorch{ | sorry |
| 03:00.21 | Strom_C | but american cheese is not actual cheese |
| 03:00.38 | file | mmm cheeburger |
| 03:00.45 | Strom_C | }btorch{: that's it/ |
| 03:00.46 | Strom_C | ? |
| 03:00.50 | Strom_C | where's the rest of it? |
| 03:00.52 | pdthome | anybody going to cluecon? |
| 03:01.00 | }btorch{ | yes the rest of it are extensions |
| 03:01.07 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-49-155.cybersurf.com) |
| 03:01.10 | }btorch{ | let me add one for you |
| 03:02.35 | }btorch{ | http://pastebin.ca/92000 |
| 03:04.42 | Strom_C | }btorch{: how far away are the machines from each other? |
| 03:04.51 | *** join/#asterisk jzpian (n=jzpian@adsl-067-035-089-049.sip.mia.bellsouth.net) |
| 03:04.52 | Kumba_ | Should I do anything special with partitioning the drive? or just swap and the rest as / ? |
| 03:04.54 | }btorch{ | well , here is the deal |
| 03:04.57 | Kumba_ | for an asterisk server |
| 03:05.37 | rob0 | Weebles wobble but they don't fall down. |
| 03:05.59 | }btorch{ | when someone call from iax to iax within the office is ok , but now I got a use in japan that talks to me on iax-toiax and sometimes from his iax-to-my home phone |
| 03:06.28 | }btorch{ | when he calls me on either sometimes he sounds jitterty |
| 03:06.35 | Strom_C | }btorch{: so its only on calls to japan? |
| 03:06.35 | }btorch{ | I hear my echo |
| 03:07.01 | }btorch{ | no the problem is that not a lot of people have been using the system so I can't really tell |
| 03:07.11 | Strom_C | or, specifically, on calls to and from his line? |
| 03:07.28 | }btorch{ | but when I first setup the system I used zmonitor to arrive at the correct rx tx rate |
| 03:07.45 | Strom_C | }btorch{: ztmonitor doesnt have a thing to do with iax |
| 03:08.01 | pdthome | apples and bananas |
| 03:08.06 | }btorch{ | I think his end the call quality is ok according to him , on my end is where the problem seems to occur |
| 03:08.09 | Strom_C | }btorch{: if iax-iax calls are fine but its just problems with a single user, i would blame the user's internet connection |
| 03:08.39 | }btorch{ | I thought so but when I call from my cell to a user inside the office it s the same |
| 03:08.46 | }btorch{ | jitterty |
| 03:09.04 | Strom_C | ok, so when you call from your mobile phone into the pbx, how are you coming in? |
| 03:09.09 | Strom_C | via iax? |
| 03:09.29 | }btorch{ | if I speak slow that its ok but as soon as I start talking fast my voice starts to break up ton their end |
| 03:09.44 | Strom_C | answer my questions please |
| 03:09.53 | }btorch{ | my asterisk is connected to the PSTN |
| 03:10.08 | }btorch{ | goes from my mobile to PSTN asterisk |
| 03:10.14 | Strom_C | via IAX, or via a CAS T1, or via PRI, or via POTS? |
| 03:10.27 | }btorch{ | phone T1 |
| 03:10.31 | }btorch{ | PRI |
| 03:10.50 | }btorch{ | I have a digium on my asterisk |
| 03:11.03 | Strom_C | where is the user in relation to the PBX? |
| 03:11.05 | }btorch{ | that is plugged into my PSTN PRI line |
| 03:11.46 | }btorch{ | It's a iax user and the user is located in the LAN where the asterisk server is located |
| 03:12.00 | }btorch{ | same physiucal bldg |
| 03:12.04 | Strom_C | what kind of station equipment? |
| 03:12.16 | file | Strom_C: you are far too patient |
| 03:12.18 | }btorch{ | P4 2.8Ghz 2Gb |
| 03:12.31 | Strom_C | }btorch{: they're using a softphone? |
| 03:12.32 | }btorch{ | hyper-threading |
| 03:12.36 | }btorch{ | idefisk |
| 03:12.40 | *** join/#asterisk i-ball (n=i-ball@nat.hackerhalfwayhouse.org) |
| 03:12.40 | Strom_C | blech |
| 03:12.48 | i-ball | hola |
| 03:12.48 | Strom_C | buy your users real phones :) |
| 03:12.56 | }btorch{ | hehe tell my boss |
| 03:13.16 | Strom_C | }btorch{: have you run point-to-point pings? |
| 03:13.26 | Strom_C | }btorch{: what kind of variance in latency do you have |
| 03:13.47 | }btorch{ | they still have their old crappy siemens PBX with a bunch of siemens phones which gave me a hard time to configure asterisk with that box |
| 03:13.59 | }btorch{ | no have not checked on that |
| 03:14.08 | Strom_C | that would be a very good thing to test |
| 03:14.14 | }btorch{ | true |
| 03:14.26 | *** join/#asterisk pdtmobile (n=ptinsley@c-68-53-40-50.hsd1.tn.comcast.net) |
| 03:14.29 | Strom_C | go test it for 300 pings and tell me what happens |
| 03:14.36 | }btorch{ | I'll check on that meanwhile I'm gona go over my iax.conf file and enable a few things |
| 03:14.44 | Strom_C | no |
| 03:14.45 | }btorch{ | can't do that now |
| 03:14.46 | Strom_C | hang on man |
| 03:14.51 | Strom_C | lets work on ONE THING AT A TIME |
| 03:14.57 | Strom_C | what do you mean "cant do that now" |
| 03:15.01 | Strom_C | its a PING test |
| 03:15.10 | *** join/#asterisk tempest1 (n=asf@adsl-153-33-178.chs.bellsouth.net) |
| 03:15.15 | }btorch{ | you want me to ping a box at my office when I'm there in the LAN tright ? |
| 03:15.29 | }btorch{ | I guess I can vpn |
| 03:15.29 | Strom_C | ssh into the asterisk box and run the ping from there |
| 03:15.34 | Dovid | Do i need ztdummy for 2.6 kernel ? |
| 03:15.36 | }btorch{ | hold on |
| 03:15.41 | }btorch{ | doing that now |
| 03:15.53 | Strom_C | Dovid: if you have no tdm or t1 card and need timing, then yes |
| 03:16.19 | Dovid | even for 2.6 kernel or only 2.4 ?\ |
| 03:16.23 | }btorch{ | any special flags for ping besides -c 300 ? |
| 03:16.56 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 03:17.09 | Strom_C | }btorch{: no |
| 03:17.16 | Strom_C | Dovid: for both, yes |
| 03:22.00 | }btorch{ | ok it's done |
| 03:22.08 | Strom_C | ok...results? |
| 03:22.27 | }btorch{ | http://pastebin.ca/92008 |
| 03:22.28 | Qwell | "it said..stuff" |
| 03:22.53 | Strom_C | hmmm |
| 03:23.31 | Strom_C | }btorch{: at this point I would probably blame the station equipment. Try it with an IAXy or something. |
| 03:23.54 | }btorch{ | IAXy ? |
| 03:24.11 | Strom_C | Digium S101i IAXy analog terminal adapter |
| 03:24.18 | }btorch{ | oh ok |
| 03:24.38 | Strom_C | what codec are you using for your on-network calls? |
| 03:24.45 | Dovid | Strom_C: do i need to uncoment ztdummy or just make linux26 ? |
| 03:25.00 | Strom_C | Dovid: just make linux26 |
| 03:25.06 | Dovid | ok |
| 03:25.09 | Dovid | so make clean |
| 03:25.14 | Dovid | make |
| 03:25.15 | Strom_C | make clean; make install |
| 03:25.21 | Dovid | then make make linnux26 ? |
| 03:25.24 | Dovid | whats the ordeR? |
| 03:25.30 | Strom_C | no no no no no no |
| 03:25.36 | Strom_C | make clean; make install |
| 03:25.49 | }btorch{ | ulaw, alwa and gsm |
| 03:26.04 | Strom_C | }btorch{: gsm?! why? |
| 03:26.04 | Dovid | k |
| 03:26.09 | Dovid | and i still need to oply the cent os bug fix ? |
| 03:26.20 | Strom_C | Dovid: YES |
| 03:26.27 | }btorch{ | what do you use ? |
| 03:26.27 | Dovid | ok |
| 03:26.28 | Dovid | thansk |
| 03:26.35 | Dovid | ~centosbug |
| 03:26.37 | jbot | i guess centosbug is a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h" |
| 03:26.38 | Strom_C | }btorch{: only ulaw for on-net calls |
| 03:27.04 | }btorch{ | so on your iax.conf you disable all and only have ulaw ? |
| 03:27.15 | Strom_C | yes |
| 03:27.18 | *** join/#asterisk Telamon (i=telamon@blk-222-22-126.eastlink.ca) |
| 03:27.26 | }btorch{ | what if the call originates from a cell or a regular phone |
| 03:27.34 | }btorch{ | doesn't matter ? |
| 03:27.39 | Strom_C | }btorch{: that comes in over your PRI, right? |
| 03:28.41 | }btorch{ | yes outside call , but I also have regular digital phone connected to a siemens PBX which is also plugged to my * box over a zap |
| 03:28.58 | Dovid | how do i finr out my kernel ? uname - ? |
| 03:29.20 | Strom_C | }btorch{: so if its not coming in via iax, what difference does it make? |
| 03:29.30 | }btorch{ | when user dial 7 + extension the PBX routes the calls to my * box and that decides what to do according to my dialplan |
| 03:29.38 | }btorch{ | true |
| 03:29.45 | Strom_C | }btorch{: the zaptel cards pass everything to your asterisk box as slin anyway |
| 03:29.48 | Telamon | Does anyone know if IAX2 trunking still has jitter issues? I want to use it to connect a remote server to my main one (over the Internet, not LAN) where the PRI resides, but I don't want to introduce an echo problem. The phones will be using SIP to talk to the remote server via the LAN, if that makes a difference. |
| 03:30.02 | Telamon | Dovid: uname -a |
| 03:34.16 | i-ball | okay, so I'm reading this |
| 03:34.19 | i-ball | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ICES |
| 03:34.40 | i-ball | and it shows the following: |
| 03:34.58 | i-ball | eh.. shows the following as being in musiconhold.conf |
| 03:35.00 | [andromeda] | Just in case if anyone is wondering, this actually works: http://www.voip-info.org/wiki/view/IPKall |
| 03:35.13 | i-ball | random => quietmp3:/var/lib/asterisk/mohmp3,-z |
| 03:35.23 | i-ball | wth is "random =>"? |
| 03:37.33 | *** join/#asterisk cHr1Zt1An (n=cHr1Zt1A@200.121.195.218) |
| 03:38.07 | }btorch{ | Strom_C: check this http://pastebin.ca/92022 |
| 03:38.41 | i-ball | is the author of the tutorial implying that we can put in some random extension in place of "random =>"? |
| 03:38.44 | Strom_C | }btorch{: so? |
| 03:38.56 | Dovid | ~centosbug |
| 03:38.57 | jbot | hmm... centosbug is a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h" |
| 03:39.02 | }btorch{ | it chooses gsm automatically ? |
| 03:39.12 | Strom_C | i-ball: no, thats for musiconhold.conf |
| 03:39.19 | i-ball | yeah |
| 03:39.20 | Strom_C | }btorch{: dude, you have to disallow it |
| 03:39.27 | i-ball | but what does "random =>" mean? |
| 03:39.50 | hads | i-ball: It's a MOH class |
| 03:40.13 | i-ball | is this covered in the Asterisk: TFOT book? |
| 03:40.22 | hads | erm.. sorry, not quite a class. |
| 03:40.24 | *** join/#asterisk santiago (i=santiago@debian/developer/santiago) |
| 03:40.46 | hads | i-ball: See the sample musiconhold.conf for a description. |
| 03:42.31 | i-ball | looking in there now |
| 03:42.51 | *** join/#asterisk fnordus (n=dnall@s142-179-111-243.bc.hsia.telus.net) |
| 03:43.58 | i-ball | uhm.... |
| 03:44.12 | i-ball | maybe I'm looking in the wrong place but I don't see it in there |
| 03:44.35 | i-ball | does asterisk put the samples into /etc/asterisk or some other location by default? |
| 03:44.51 | Strom_C | i-ball: did you do 'make samples' when you installed asterisk? |
| 03:44.53 | i-ball | oh |
| 03:44.56 | i-ball | I found it |
| 03:45.32 | i-ball | ah, yeah, it's the one I was looking at already |
| 03:46.00 | i-ball | the only mention of "random" in there is: |
| 03:46.08 | i-ball | random=yes |
| 03:46.30 | Dovid | Strom_C: can u look at this ? |
| 03:46.31 | Dovid | http://pastebin.ca/92027 |
| 03:47.12 | Strom_C | what did you type? |
| 03:47.17 | Dovid | make clean |
| 03:47.25 | Strom_C | from zaptel-1.2.7? |
| 03:47.32 | Telamon | Dovid: You have the zaptel source installed somewhere other than /usr/src/zaptel-1.2.7 |
| 03:47.47 | Dovid | nope |
| 03:47.51 | Dovid | clean machine |
| 03:48.14 | Telamon | Dovid: Do a "ls /usr/src/zaptel-1.2.7" and make sure there is a Makefile in there. |
| 03:48.45 | Dovid | rebooted server |
| 03:48.46 | Dovid | one sec |
| 03:49.13 | *** join/#asterisk TommyTheKid (n=tommythe@mpk-edge.cto.sunit.net) |
| 03:50.44 | i-ball | http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf |
| 03:50.45 | TommyTheKid | So, while I am working the "Sangoma" angle at work (we currently have a few digium cards)... I was wondering if its worth my time to setup a 64 bit CentOS network boot and re-install my opteron server to take advantage of 64 bit? is it a major gain in performance? |
| 03:51.05 | Dovid | Strom_C: yup there is a mak file |
| 03:51.37 | Dovid | make* |
| 03:51.46 | Dovid | and when i do make I get |
| 03:51.55 | Strom_C | Dovid: what distro are you running? |
| 03:51.59 | mog_home | TommyTheKid: whats wrong with your digium stuff if you dont mind me asking? |
| 03:52.06 | Dovid | You do not appear to have the sources for the 2.6.9-34.0.2.ELsmp kernel installed. |
| 03:52.06 | Dovid | make: *** [linux26] Error 1 |
| 03:52.12 | Dovid | centOS 3.4 |
| 03:52.13 | TommyTheKid | mog_home: not dense enough? |
| 03:52.16 | Telamon | Dovid: Try "make -C /usr/src/zaptel-1.2.7 SUBDIRS=/usr/src/zaptel-1.2.7" |
| 03:52.19 | mog_home | ? |
| 03:52.19 | Dovid | i did the spinlcok.h fix |
| 03:52.28 | TommyTheKid | I am looking at the 8 port card and I want 3+ cards in one server |
| 03:52.31 | TommyTheKid | ... I think :) |
| 03:52.39 | mog_home | your gonna hit same problems with asterisk |
| 03:52.48 | mog_home | i have heard of yate scaling more cards |
| 03:53.00 | Dovid | Telamon: then i get this |
| 03:53.02 | Telamon | TommyTheKid: You might want to look into external channel banks if you are going to have that many PRIs. |
| 03:53.02 | mog_home | but benchmarks are you usually about the same |
| 03:53.16 | Dovid | http://pastebin.ca/92034 |
| 03:53.18 | mog_home | i see 2-3 cards in our machines here |
| 03:53.21 | TommyTheKid | how does an external channel bank help? |
| 03:53.23 | mog_home | i have heard of 5 |
| 03:53.27 | mog_home | but never seen it myself |
| 03:53.35 | mog_home | digium ones at least |
| 03:53.35 | Dovid | but i do have kernel sources |
| 03:53.40 | Telamon | TommyTheKid: You can load balance between multiple Asterisk servers. |
| 03:53.45 | Dovid | actually did yum install kernel-devel |
| 03:53.49 | TommyTheKid | oh, I plan to Telamon |
| 03:54.03 | mog_home | so why do you need 3+ in one machine ??? |
| 03:54.17 | TommyTheKid | I want to build 2 Sun X4200's each with as many lines as I can cram into them |
| 03:54.30 | TommyTheKid | conferencing servers |
| 03:54.35 | mog_home | <PROTECTED> |
| 03:54.47 | TommyTheKid | understood mog :) |
| 03:54.48 | mog_home | well doing that i would imagine you get same count |
| 03:54.56 | *** join/#asterisk shidan (i=shidan@CPE0013107d30c4-CM001371871af0.cpe.net.cable.rogers.com) |
| 03:55.07 | mog_home | esp with our new echo can board ^_^ |
| 03:55.15 | Telamon | mog_home: Hey, as a Diguim guy... Do you know if the jitterbuffer problems with IAX2 trunking have been fixed? |
| 03:55.19 | mog_home | which does work in 64 bit machines why the sangoma card doesnt ^_^ |
| 03:55.33 | mog_home | dtmf still an issue i believe Telamon |
| 03:55.34 | TommyTheKid | I have read about problems with the digium cards and intel e1000gX ? |
| 03:55.37 | mog_home | other wise it works |
| 03:56.00 | russellb | Telamon: if you're talking about a timestamps issue, that was fixed last year sometime |
| 03:56.02 | TommyTheKid | not sure if our specific MB would have it, but that would kill the operation |
| 03:56.45 | TommyTheKid | heh, I am also trying to use Solaris, which is always more fun when it comes to drivers :) |
| 03:57.09 | Telamon | mog_home: Hmm, even if DTMF is sent out of band? IE, via SIP-INFO? |
| 03:57.10 | *** join/#asterisk FuriousGeorge (n=FuriousG@ool-43536ea8.dyn.optonline.net) |
| 03:57.13 | *** part/#asterisk FuriousGeorge (n=FuriousG@ool-43536ea8.dyn.optonline.net) |
| 03:57.14 | *** join/#asterisk FuriousGeorge (n=FuriousG@ool-43536ea8.dyn.optonline.net) |
| 03:57.15 | mog_home | why hurt yourself |
| 03:57.23 | mog_home | dtmf is always out of band in iax2 |
| 03:57.23 | TommyTheKid | Sun on Sun :) |
| 03:57.28 | russellb | Telamon: that makes no sense ... how is DTMF sent with SIP INFO with IAX trunking? |
| 03:57.40 | mog_home | issue is to do with way frames are parsed |
| 03:58.10 | Telamon | russellb: IAX2 version of SIP-INFO. Basically, in the control headers rather than the RTP packets. |
| 03:58.12 | TommyTheKid | "to fly our own airplanes" .. something about dogfood? I dunno, take your pick, we'd rather run it in Solaris |
| 03:58.23 | russellb | Telamon: there is no RTP in IAX2 ........... |
| 03:58.48 | mog_home | well TommyTheKid you are gonna have a fun time getting stuff working in solaris no matter which way you go |
| 03:58.54 | mog_home | i heard someone is porting zaptel |
| 03:58.59 | TommyTheKid | "fun" |
| 03:59.03 | mog_home | but only have tdm cards done |
| 03:59.05 | TommyTheKid | yea, saw that |
| 03:59.06 | mog_home | fun as in pain in suffering |
| 03:59.12 | TommyTheKid | i know |
| 03:59.17 | wunderkin | yeah, mog_home is kinky like that |
| 03:59.18 | Strom_C | pain and suffering doesnt begin to describe it :) |
| 03:59.19 | russellb | mog_home: fun as in not possible |
| 03:59.26 | mog_home | it could be done |
| 03:59.28 | mog_home | given time |
| 03:59.32 | mog_home | but how legal it would be |
| 03:59.36 | mog_home | there we get into some fun |
| 03:59.50 | TommyTheKid | I have compiled ast several times, there are "issues" still, but that Joe guy has nice little packages, if a bit out of date :) |
| 03:59.51 | mog_home | a gpl dirivitve driver in a non gpl compat. kernel |
| 04:00.06 | mog_home | man i cant spell |
| 04:00.12 | Strom_C | derivative |
| 04:00.24 | *** join/#asterisk teknoprep (n=chris@unaffiliated/teknoprep) |
| 04:00.28 | TommyTheKid | they have to be re-written from specs I think.. I am no driver devl tho :) |
| 04:00.32 | shidan | Who understands T38 really well here |
| 04:00.42 | mog_home | there are no specs |
| 04:00.48 | mog_home | they are writing it from code |
| 04:00.54 | mog_home | is no other way to do it right now |
| 04:01.02 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 04:01.03 | TommyTheKid | hmm, well like I said, I have no clue, plus we use the te412p's right now |
| 04:01.14 | TommyTheKid | that was strange.. |
| 04:01.21 | mog_home | mmmm octasic echo canceller |
| 04:02.05 | TommyTheKid | we are hoping that by using the cards with onboard DSP we will be able to get more simultaneous calls |
| 04:02.12 | Strom_C | for some reason, it always looks like "Ocasek echo canceller" to me |
| 04:02.14 | mog_home | yeah a few |
| 04:02.34 | TommyTheKid | I am hoping I can put at least 8 T1s worth into a dual/dual opteron |
| 04:02.35 | mog_home | i mean 30mhz or so goes to echo cancelling |
| 04:02.41 | TommyTheKid | .. using conferencing |
| 04:02.48 | mog_home | yeah |
| 04:02.51 | Kumba_ | So should I go straight for Kernel 2.6.17.6? Or is asterisk not stable with it? |
| 04:03.10 | mog_home | asterisk has very little to do with linux kernel Kumba_ |
| 04:03.15 | mog_home | use whats stable |
| 04:03.34 | TommyTheKid | was hoping for 12-16 but got advice against it for IRQ issues? (which I am not sure how there could be with 4 CPUs) :) |
| 04:03.46 | Strom_C | 16 cards?! |
| 04:03.51 | TommyTheKid | PRIs |
| 04:04.10 | TommyTheKid | 4 cards (or 2 x 8 porters if I went non-digium) |
| 04:04.10 | iPBX | where can i find the wav or gsm for the ring sound? |
| 04:04.28 | iPBX | /var/lib/asterisk/sounds/???? |
| 04:04.29 | TommyTheKid | isnt the ring generated? |
| 04:04.37 | Strom_C | iPBX: it's generated; it's not a sound |
| 04:04.40 | mog_home | its generated |
| 04:04.43 | iPBX | crap |
| 04:04.46 | mog_home | you can do a monitor of it |
| 04:04.56 | mog_home | and sounds.txt in /var/lib/asterisk/sounds |
| 04:05.01 | mog_home | says what all the files are |
| 04:05.20 | TommyTheKid | as with every question I see here... what are you trying to accomplish, if you want to change it, I think thats quite possible |
| 04:05.36 | iPBX | any way to turn the ring volume down/ |
| 04:05.38 | iPBX | ? |
| 04:05.44 | Strom_C | iPBX: why? |
| 04:05.53 | iPBX | i noticed that the ring volume on the system is about twice as load as the ring sound i get from my telco |
| 04:06.06 | Strom_C | iPBX: whats your connection to the telco like |
| 04:06.19 | *** join/#asterisk daysmen3 (n=primus@host86-137-170-127.range86-137.btcentralplus.com) |
| 04:06.26 | *** join/#asterisk SkramX (n=MarkS@admins.sentiensystems.net) |
| 04:06.30 | SkramX | *anyone* making it to HOPE? |
| 04:06.36 | TommyTheKid | is there something like a PRI over T3? |
| 04:06.39 | TommyTheKid | CT3? |
| 04:06.48 | Strom_C | TommyTheKid: PRI is not inherently T1 |
| 04:06.54 | mog_home | yeah |
| 04:06.57 | Strom_C | TommyTheKid: you can do a 671B+D |
| 04:06.58 | mog_home | i mean you can get a t3 |
| 04:07.13 | mog_home | and an adtran multiplexer to break out to 28 t1s if you want |
| 04:07.19 | TommyTheKid | essentially I am using a back-to-back connection (T1-cross) from the corp PBX |
| 04:07.26 | iPBX | i have IAX service. When I call my #, I hear 1 ring by the telco, then my PBX picks up, the rings i hear from the PBX are twice as loud as the Telco Ring |
| 04:07.50 | iPBX | (quiet) ring... (loud) ring... (loud) ring... (loud) ring... |
| 04:07.58 | TommyTheKid | hehe |
| 04:08.16 | Strom_C | iPBX: are you supervising before the pbx rings? |
| 04:08.25 | TommyTheKid | have it answer and say "I am trying that extension now... (jeapordy music) ....." :) |
| 04:08.41 | iPBX | NV_FaxDetect |
| 04:08.44 | mog_home | well i have to go to work tommorrow |
| 04:08.44 | mog_home | gnite |
| 04:09.04 | *** join/#asterisk NoRemorse (n=bah@210-84-10-183.dyn.iinet.net.au) |
| 04:09.05 | NoRemorse | hi all |
| 04:09.06 | Kumba_ | i'm still at work :( |
| 04:09.06 | iPBX | 4 second delay before autoattendant answers |
| 04:09.16 | NoRemorse | I am using the DISA function to allow indials to 'call the extension of the party if they know the extension number" etc, and it all works fine, but I get 2 CDR records with the same uniqueid, which is playing havoc with my rating software. can anyone suggest how to cancel the first CDR from the dialplan? |
| 04:09.16 | Strom_C | iPBX: are you supervising |
| 04:09.25 | iPBX | i guess that's a yes |
| 04:09.47 | iPBX | yes, the call is answered by the PBX immediately when recieved |
| 04:09.47 | Strom_C | iPBX: or are you passing a "ringing" message |
| 04:09.51 | Strom_C | ok |
| 04:10.04 | Strom_C | so there should be no telco-side ringing at all then |
| 04:10.10 | i-ball | wow |
| 04:10.17 | iPBX | there's 1 ring before the call get's thru to me |
| 04:10.28 | iPBX | sometimes it's half a ring |
| 04:10.29 | Strom_C | iPBX: what bonkers iax provider are you using |
| 04:10.31 | i-ball | the sample file for musiconhold.conf is completely outdated |
| 04:10.32 | iPBX | voicepulse |
| 04:11.04 | Strom_C | thats extremely odd |
| 04:11.19 | Strom_C | i havent had voicepulse do that on my number |
| 04:11.29 | iPBX | that's even more disturbing, a ring that's 1/3rd quiet and 2/3's loud |
| 04:11.46 | iPBX | and the ring is 1.5 times normal length |
| 04:12.05 | iPBX | there's some delay getting the call from the telco to voicepulse |
| 04:12.06 | Strom_C | iPBX: whats the number |
| 04:12.09 | shidan | Ive had that happen to me too but only when using freepbx |
| 04:12.13 | Strom_C | iPBX: let me call it |
| 04:12.19 | Strom_C | iPBX: are you using freepbx? |
| 04:12.44 | iPBX | yep, nm |
| 04:12.52 | Strom_C | oh god |
| 04:12.54 | Strom_C | why |
| 04:13.00 | Strom_C | why can no one ever read the damned topic |
| 04:13.05 | Strom_C | why why why why why |
| 04:13.11 | iPBX | i know the topic |
| 04:13.12 | TommyTheKid | settle :) |
| 04:13.26 | shidan | haha |
| 04:13.40 | iPBX | i asked the same question there... |
| 04:14.01 | iPBX | <iPBX> where can i find the wav or gsm for the ring sound? |
| 04:14.02 | iPBX | <iPBX> /var/lib/asterisk/sounds/???? |
| 04:14.02 | iPBX | <N3GLV> should be in /var/lib/asterisk/sounds or something like that |
| 04:14.07 | iPBX | that was the sum of the response |
| 04:14.10 | iPBX | and it was even wrong |
| 04:14.11 | shidan | well why dont u paste bin the console during a call |
| 04:14.32 | iPBX | number is 207 321 5063 if you still wanna hear it |
| 04:14.37 | shidan | anyone here worked with t38 before? |
| 04:14.44 | Strom_C | my guess is that all that bonkers stuff freepbx has to do before it even gets to the answer() is delaying and causing the ring |
| 04:15.00 | iPBX | Strom_C that very well could be... |
| 04:15.20 | Strom_C | iPBX: seriously, dude....get rid of freepbx |
| 04:15.26 | Strom_C | learn asterisk :) |
| 04:16.32 | Dovid | ~centosbug |
| 04:16.36 | jbot | [centosbug] a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h" |
| 04:16.54 | iPBX | i know asterisk pretty well actually. I use it because i do alot of installs... every different client wants things to work differently. freepbx lets me do everything i want in about an hour's setup... |
| 04:17.09 | iPBX | and generally everything works very well |
| 04:17.10 | Strom_C | iPBX: ewwww |
| 04:17.32 | iPBX | plus i'm a coder and some of my code is in freepbx :-D |
| 04:18.07 | shidan | freepbx is much better now than it used to be |
| 04:18.19 | iPBX | i've done plain old asterisk installs before... sometimes would take me days to get it all working right |
| 04:18.32 | iPBX | i don't think i've done a setup since that's taken me more than 2 hours |
| 04:18.46 | Strom_C | iPBX: usually doesnt take me very long at all to get plain old asterisk working exactly as the client wants |
| 04:19.13 | TommyTheKid | So, I am having a problem that I am not sure how to "solve" .. When I add a "user" (an extension) I endup having to add an IAX entry (for IAX soft phones), a SIP entry for SIP soft phones, and SIP entries for every hard phone that may be associated with them (2 for my boss). People have already had problems with having more than one soft phone running.. one at work, one on laptop for example. The way I am handling it currently (adding a new |
| 04:19.15 | Strom_C | iPBX: and I find that freepbx severely limits flexibility |
| 04:19.48 | Dovid | how do i do the oposite of modprobe zaptel ? |
| 04:19.54 | Strom_C | rmmod zaptel |
| 04:19.57 | iPBX | zaptel modprobe |
| 04:20.02 | TommyTheKid | lol |
| 04:20.04 | Dovid | hehe |
| 04:20.06 | Dovid | really ? |
| 04:20.11 | Strom_C | rmmod zaptel |
| 04:20.12 | Dovid | k |
| 04:20.13 | TommyTheKid | lsmod rmmod modprobe |
| 04:20.18 | *** join/#asterisk godsmoke (n=godsmoke@cpe-66-108-202-75.nyc.res.rr.com) |
| 04:20.19 | iPBX | rm /* -rf |
| 04:20.38 | iPBX | sorry... couldn't resit |
| 04:20.42 | iPBX | don't try that command |
| 04:21.22 | TommyTheKid | I freaked out my PBX when I installed a new zaptel earlier today, I didn't realize it was going to try to replace the driver live |
| 04:21.57 | iPBX | i assume you mean it wouldn't take you long to setup the a POAS (plain old asterisk system??) because you aren't talking about writing out all the conf's by hand... you use some kind of generic template to start with right? |
| 04:22.14 | Strom_C | iPBX: I write it all out by hand |
| 04:22.38 | TommyTheKid | I started from the "make samples" actually... its not that bad, once you get a "system" going |
| 04:22.58 | shidan | strom sounds like pretty simple systems if they are that easy |
| 04:23.09 | Strom_C | there's beauty in simplicity |
| 04:23.14 | TommyTheKid | I actually really make use of include's .. you can use wildcards so something like users/*.conf is handy ;) |
| 04:23.42 | TommyTheKid | just copy a file and change the name/ext/etc and you have a new user |
| 04:23.55 | TommyTheKid | (well XXX reload obviously |
| 04:25.21 | bkw_ | that doesn't scale at all |
| 04:25.28 | bkw_ | once you reach a point it will take too long to reload |
| 04:25.35 | bkw_ | and the system will go nutz |
| 04:25.42 | iPBX | that's fine if they only need a simple system... i get people that want every feature under the sun... |
| 04:25.58 | shidan | exactly |
| 04:26.10 | Strom_C | iPBX: so its your job as a consultant to figure out what they really need and what they will never use |
| 04:26.38 | TommyTheKid | heh |
| 04:26.40 | iPBX | i have trouble selling them if they can't do everything a traditional standard off the shelf PBX can do |
| 04:27.08 | TommyTheKid | yeah, so that goes back to my indication that I was having trouble routing the call from one extension to several different possibilities of the user being logged in |
| 04:27.39 | TommyTheKid | I need some way for them to just have one SIP and one IAX account and if they are registered with 3 devices, it rings all three |
| 04:27.51 | Dovid | Strom_C: can u look at this ? |
| 04:27.52 | Dovid | http://pastebin.ca/92044 |
| 04:28.15 | godsmoke | udev not set up properly? |
| 04:28.18 | hads | http://www.google.co.nz/search?hl=en&ie=UTF-8&oe=UTF-8&q=exchange+rate+feed&btnG=Search&meta=cr%3DcountryNZ. |
| 04:28.24 | godsmoke | (to Dovid) |
| 04:28.33 | Dovid | udev is ? |
| 04:28.38 | TommyTheKid | i can do the sip part by frontending chan_sip with something like ser, but that adds yet another layer of complexity |
| 04:29.00 | godsmoke | Dovid: too complicated to explain here -- wikipedia it if you like -- but there's a README.udev in the source dir for zaptel |
| 04:29.02 | iPBX | nice chat Strom_C, thx for the insight |
| 04:29.21 | Kumba_ | I just want a dial tone :) |
| 04:29.51 | iPBX | I just want a fat white woman, but i think you'll have more luck getting what you want here Kumba_ |
| 04:30.00 | godsmoke | hahah |
| 04:30.10 | Kumba_ | Possibly... |
| 04:30.17 | Kumba_ | although, i'm sure there are fat white women on IRC... |
| 04:30.21 | iPBX | any single fat white women? |
| 04:30.30 | iPBX | in Maine? |
| 04:30.41 | TommyTheKid | google? |
| 04:30.51 | iPBX | Is there an asterisk PhoneSex mod? |
| 04:31.12 | *** join/#asterisk Dico_ (n=niko@60.51.217.61) |
| 04:31.20 | Strom_C | res_ohohohyes.so |
| 04:31.59 | iPBX | lol #include "handcuffs.h" |
| 04:32.13 | russellb | i have app_ohbabyohbaby.c |
| 04:32.16 | Kumba_ | hardprobe.h |
| 04:32.23 | russellb | didn't think it was appropriate to submit back to the community |
| 04:32.53 | *** join/#asterisk CoderCR (n=creyna@ip68-6-237-193.sd.sd.cox.net) |
| 04:32.57 | iPBX | ./etc/asterisk/condom.conf |
| 04:33.00 | Kumba_ | I wonder when someone's gonna write some games you can play on the phone with the softkeys... |
| 04:33.24 | *** part/#asterisk CoderCR (n=creyna@ip68-6-237-193.sd.sd.cox.net) |
| 04:33.27 | Kumba_ | I could look busy as hell playing a mean game of SipSolitare... |
| 04:33.36 | TommyTheKid | i wish someone would write app_iax for the polycom phones so I could turn off SIP :) |
| 04:33.47 | *** part/#asterisk SkramX (n=MarkS@admins.sentiensystems.net) |
| 04:33.53 | russellb | Kumba_: there are some already ... |
| 04:34.15 | russellb | i wrote one over a year ago, don't know where it is now |
| 04:34.24 | russellb | and there is a blackjack app to demo the speech recognition stuff |
| 04:34.33 | Kumba_ | Port zelda to asterisk... lol :) |
| 04:34.33 | iPBX | call 1800 555 TELL and say BlackJack at the main menu for IVR BlackJack w/ Voice Recognition |
| 04:34.55 | Strom_C | iPBX: their network engineer is my good friend :) |
| 04:35.50 | iPBX | cool... i've known that number for about 4 years now I think? I'm not on the road much anymore, but when I was travelling all over the states, i used to use it for driving directions religiously |
| 04:35.59 | iPBX | i mostly use it to listen to the AP news updates now |
| 04:36.27 | iPBX | i did actually play blackjack for a bit today :-p |
| 04:36.39 | iPBX | god i'm such a loser |
| 04:37.04 | Strom_C | i first used 800-555-TELL in...2000? |
| 04:37.20 | iPBX | i think it was about 2002 for me, but i'm not sure |
| 04:37.56 | iPBX | around that time anyways. i was working for a interconnect at the time... they had me drving everywhere |
| 04:38.54 | TommyTheKid | blackjack is cool |
| 04:39.02 | iPBX | at one time i could punch down a whole 25 pair cable in 1 min 45 seconds |
| 04:39.16 | Strom_C | iPBX: recite the color code |
| 04:39.19 | Strom_C | ready... |
| 04:39.19 | iPBX | on to a 66 block |
| 04:39.20 | Strom_C | go |
| 04:39.24 | iPBX | BOGBS |
| 04:39.38 | iPBX | and the inside |
| 04:39.42 | Strom_C | ....and the others? |
| 04:39.55 | TommyTheKid | as in the voice recognitian and sound quality was good |
| 04:39.56 | iPBX | WRBYP |
| 04:39.58 | Dovid | godsmoke: i did what the file told me and still having issues |
| 04:40.04 | iPBX | took me a second, i had to think |
| 04:40.09 | godsmoke | Dovid: the same issues, or different ones? |
| 04:40.12 | Dovid | same |
| 04:40.15 | godsmoke | hmm |
| 04:40.19 | godsmoke | then ask someone else |
| 04:40.19 | Strom_C | iPBX: who the hell told you it was "purple"? |
| 04:40.21 | Dovid | but if i do it again it goes thru |
| 04:40.22 | godsmoke | sorry |
| 04:40.29 | iPBX | violet w/e |
| 04:40.31 | Strom_C | iPBX: violet :) |
| 04:40.47 | iPBX | i remember purple because of the acronym while running backwards you puke |
| 04:40.47 | Dovid | meaning i do modprobe ztdummy and get error |
| 04:40.53 | Dovid | and i do it again and it goes |
| 04:41.07 | Strom_C | iPBX: i heard it as "Why run backwards? You'll vomit!" |
| 04:41.15 | iPBX | yea, puke and vomit |
| 04:41.23 | iPBX | lol |
| 04:41.25 | Strom_C | ah, telco humor :) |
| 04:41.38 | docelmo | OI! |
| 04:42.23 | *** join/#asterisk surfdue (n=surfdue@unaffiliated/surfdue) |
| 04:42.25 | surfdue | hey |
| 04:42.35 | iPBX | 1:45 sec on a 66 block, i could do a 110 or bix block in 1:10 |
| 04:42.36 | surfdue | how do I check what time it is if its after 10pm est for example go to message machine |
| 04:42.47 | Strom_C | surfdue: gotoiftime() |
| 04:43.12 | iPBX | of course that was just if i was challenging another tech... generally on site, I took my time to make sure it was perfect |
| 04:43.13 | surfdue | thank you |
| 04:44.02 | iPBX | nothing more annoying when you terminate the cabling with a few flipped pairs, then you go to hook up the phone system a few weeks later and something doesn't work, and it's like wtf |
| 04:44.30 | iPBX | more annoying when some other tech does that, and doesn't leave the slack to fix it |
| 04:44.38 | surfdue | Strom_C, the internet broke |
| 04:44.39 | surfdue | :? |
| 04:44.41 | TommyTheKid | nite |
| 04:44.44 | *** part/#asterisk TommyTheKid (n=tommythe@mpk-edge.cto.sunit.net) |
| 04:44.52 | Strom_C | surfdue: what? |
| 04:44.59 | surfdue | Strom_C, the internet is down |
| 04:45.01 | surfdue | :/ |
| 04:45.08 | surfdue | good says they are merging databases |
| 04:45.13 | surfdue | if google is down the world is down |
| 04:45.14 | surfdue | :( |
| 04:45.21 | iPBX | it's ok, I just finally finished reading the internet today |
| 04:45.23 | iPBX | nothing more to see |
| 04:45.30 | hads | I hate it when the Internet is down. |
| 04:45.33 | godsmoke | right, like when cogent and l3 had their bitchslapping cotest |
| 04:45.33 | surfdue | lucky |
| 04:45.35 | godsmoke | contest* |
| 04:45.44 | surfdue | can you explain Strom_C what the function is please, im not kidding i cant get to anywhere :/ |
| 04:46.04 | *** join/#asterisk konfuzed (n=Konf@H211.C18.B96.tor.eicat.ca) |
| 04:46.09 | iPBX | it must be Al Gore's fault... |
| 04:46.17 | godsmoke | it was |
| 04:46.38 | godsmoke | viruses, worms, spyware, and identity theft are also his doing |
| 04:46.41 | surfdue | stupid people |
| 04:46.42 | konfuzed | hey can these skype phones be configured to use a login from an asterisk server ? |
| 04:46.43 | surfdue | not you |
| 04:46.47 | godsmoke | I see no reason why we shouldn't lock him up |
| 04:46.49 | surfdue | the internet is a series of tubes |
| 04:47.06 | surfdue | if someone comes and puts alot into the tubes it blocks it for the rest of us |
| 04:47.06 | iPBX | that would be weird if he won the next election for president... i mean we'd have the inventor of the internet, as president... how cool would that be? |
| 04:47.06 | surfdue | lol |
| 04:47.19 | iPBX | tax breaks for the more time you spend on the net |
| 04:47.24 | surfdue | al gore invented the net? |
| 04:47.25 | surfdue | NO. |
| 04:47.31 | godsmoke | ... |
| 04:47.32 | godsmoke | yes |
| 04:47.34 | konfuzed | uh |
| 04:47.36 | godsmoke | don't you know anything? |
| 04:47.36 | surfdue | no.. |
| 04:47.38 | surfdue | nooo!! |
| 04:47.41 | surfdue | no really? |
| 04:47.45 | surfdue | your fucking kidding me :/ |
| 04:47.47 | iPBX | al gore invented the internet |
| 04:47.51 | iPBX | you didn't know that? |
| 04:47.51 | godsmoke | yes |
| 04:47.54 | surfdue | no one invented the internet |
| 04:47.59 | surfdue | it was started as a college project |
| 04:48.00 | surfdue | .. |
| 04:48.03 | konfuzed | no no no no no really Al Gore had nothing to so with the invention of the internet |
| 04:48.09 | iPBX | al gore did |
| 04:48.12 | surfdue | no lies |
| 04:48.12 | godsmoke | http://www.google.com/search?q=al+gore+invented+the+internet |
| 04:48.15 | surfdue | no.. |
| 04:48.17 | surfdue | i dont belive you |
| 04:48.19 | godsmoke | test it out for yourself |
| 04:48.24 | surfdue | im smarter then that i was in the college |
| 04:48.25 | surfdue | when they did it. |
| 04:48.39 | surfdue | i had one of the firt 26k modems |
| 04:48.42 | surfdue | its not al gore |
| 04:48.59 | godsmoke | direct quote: |
| 04:49.01 | konfuzed | the internet is merely TCP/IP with routing tables |
| 04:49.03 | godsmoke | Al Gore: "I took the initiative in creating the Internet." |
| 04:49.09 | konfuzed | times as many computers running it |
| 04:49.20 | konfuzed | or dedicated computers called routers |
| 04:49.28 | surfdue | lies.. |
| 04:49.39 | surfdue | well just explain to me the gotoiftime function please |
| 04:49.43 | surfdue | cuase al gores internet is dead |
| 04:49.43 | surfdue | lol |
| 04:50.12 | surfdue | oh i got to voip info |
| 04:50.13 | surfdue | nvm |
| 04:50.37 | konfuzed | more importantly , is there any hacks out there for using a skype phone to login to asterisk |
| 04:51.09 | docelmo | No cause skype isnt SIP |
| 04:51.57 | docelmo | err skype doesnt support sip they are proprietary |
| 04:52.49 | surfdue | im so confused |
| 04:52.52 | konfuzed | hmhmhmmhmmmm |
| 04:53.45 | surfdue | 10-8|*|*|* |
| 04:53.47 | surfdue | is that right/ |
| 04:53.51 | surfdue | 10am - 8pm |
| 04:53.55 | konfuzed | so not to be presumptious, does Asterisk support the protocols or technology used by Skype?? |
| 04:54.05 | hads | konfuzed: No |
| 04:54.25 | konfuzed | ok but take the usb skype phone and flash it to run a something like pc-phoneline code |
| 04:54.37 | konfuzed | or just embedded linux or something |
| 04:54.56 | surfdue | exten => 18006170141,1,GotoIfTime(10:00-20:00|mon-frii|*|*?open,s,1) |
| 04:54.56 | iPBX | anyone know the name Leonard Kleinrock ? |
| 04:54.59 | konfuzed | its just a phone set with a usb cable and a processor isnt it |
| 04:55.00 | surfdue | whats ?open mean? |
| 04:55.07 | konfuzed | ;^) |
| 04:55.08 | iPBX | anyone ever heard of him? |
| 04:55.23 | iPBX | i went to UCLA and studied under him |
| 04:55.29 | iPBX | he's a professor there, look him up |
| 04:55.35 | surfdue | anyone? |
| 04:55.45 | surfdue | exten => 18006170141,1,GotoIfTime(10:00-20:00|mon-frii|*|*?open,s,1) << whats ?open mean ? |
| 04:56.10 | iPBX | first person to look up Leonard Kleinrock on google and tell me why he's important gets a cookie |
| 04:56.10 | russellb | surfdue: open is the context |
| 04:56.18 | russellb | it's the same as any other goto in asterisk .... |
| 04:56.20 | surfdue | can i jsut take it off |
| 04:56.22 | i-ball | yeah, what's up with the |? |
| 04:56.31 | i-ball | Do those just replace the comas? |
| 04:56.55 | hads | i-ball: Yes you can use | or , |
| 04:56.57 | russellb | i-ball: yes, asterisk actually replaces commas with pipes '|' before processing |
| 04:57.02 | russellb | though | is really the standard |
| 04:58.51 | konfuzed | bang ! |
| 04:59.04 | Dovid | seen |
| 04:59.05 | i-ball | yeah, they really need to update the book |
| 04:59.06 | Dovid | ~seen |
| 04:59.14 | iPBX | i guess since no one cares who Professor Kleinrock is, I'll tell... He did almost all of the early development for ARPANET... the predecessor to USENET... he's credited as having the FIRST node on the internet |
| 04:59.15 | Dovid | ~seen Dovid |
| 04:59.17 | jbot | dovid is currently on #asterisk. Has said a total of 49 messages. Is idling for 2s, last said: '~seen Dovid'. |
| 04:59.22 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 04:59.30 | docelmo | ~seen my_butt |
| 04:59.33 | jbot | docelmo: i haven't seen 'my_butt' |
| 04:59.37 | iPBX | ~seen my_penis |
| 04:59.38 | jbot | iPBX: i haven't seen 'my_penis' |
| 04:59.38 | *** part/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 04:59.47 | Dovid | lol |
| 05:00.01 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:00.32 | Dovid | ~seen sean2222 |
| 05:00.34 | jbot | Dovid: i haven't seen 'sean2222' |
| 05:00.36 | Dovid | ~seen shaun2222 |
| 05:00.38 | jbot | shaun2222 <n=ndci@ip68-5-63-223.oc.oc.cox.net> was last seen on IRC in channel #asterisk, 29d 9h 42m 22s ago, saying: 'does it require vontage or can i use it to connect via sip to my asterisk server'. |
| 05:00.53 | docelmo | ~seen docE |
| 05:00.54 | jbot | doce <n=docelmo@66.237.242.41.ptr.us.xo.net> was last seen on IRC in channel #asterisk, 4d 11h 56m 54s ago, saying: 'whadup'. |
| 05:01.10 | }btorch{ | is there a way that on a dialplan context to have say exten => 2,1,.... and also a exten => _1XXX,1,... and not allow it to go two 2 when a user calls press 1244 |
| 05:01.21 | iPBX | from now on| I think I'll replace all my commas with pipes| i mean| i don't think it'll be that confusing| atleast now in this channel| right? |
| 05:01.38 | surfdue | yay it works |
| 05:01.42 | iPBX | omg i'm tired.... l8r folks |
| 05:01.46 | }btorch{ | I'm calling my system and I go through a menu and on that menu context I have an extension for 2 and also _1XXX |
| 05:01.47 | surfdue | now how do i set my voicemailbox 200 to a new message |
| 05:01.52 | Dovid | can some one help me with this again ? |
| 05:01.53 | Dovid | http://pastebin.ca/92064 |
| 05:01.54 | surfdue | instead of the person at extension bla.. |
| 05:02.10 | _Vile | hmm I'm in a while on an ast_waitfor(chan, -1) with an fr = ast_read(chan) and an if below that on if(fr->frametype == AST_FRAME_CONTROL) and it's just looping, never detecting that frametype -- any thought on what could cause that? |
| 05:02.12 | Dovid | sirfdude: log in to ur boz |
| 05:02.15 | Dovid | sirfdude: log in to ur vm box |
| 05:02.16 | surfdue | k |
| 05:02.17 | surfdue | im in |
| 05:02.19 | surfdue | how? |
| 05:02.24 | surfdue | no |
| 05:02.26 | surfdue | i have a gsm |
| 05:02.26 | surfdue | :p |
| 05:02.28 | }btorch{ | some times when I press 12XX its like it doesn't see the 1 but just the 2XX |
| 05:02.30 | *** join/#asterisk fnordus (n=dnall@s142-179-111-243.bc.hsia.telus.net) |
| 05:02.30 | surfdue | i mean mp3. |
| 05:02.33 | surfdue | i have an mp3 |
| 05:02.35 | Dovid | exten => 888,1,VoicemailMain |
| 05:02.44 | Dovid | or |
| 05:02.50 | surfdue | internal |
| 05:02.51 | Dovid | exten => 888,1,VoicemailMain(@context) |
| 05:03.09 | Dovid | can anyone help with http://pastebin.ca/92064 ??? |
| 05:03.55 | surfdue | can I use an mp3? |
| 05:04.04 | Dovid | hmm |
| 05:04.09 | Dovid | yes |
| 05:04.34 | Dovid | but u have to convert it to gsm file and rename the file that is there now with ur gsm file (that was an mo3) |
| 05:04.42 | Dovid | mp3* |
| 05:04.46 | surfdue | where is it |
| 05:04.53 | surfdue | the file taht says where yor recording is |
| 05:04.54 | Strom_C | you dont have to use gsm |
| 05:04.58 | surfdue | ok |
| 05:04.59 | Strom_C | you can use wav of slin |
| 05:05.07 | Strom_C | s/of/or/ |
| 05:05.29 | surfdue | Strom_C, i did have a question aswell i think you may know anyone else feel free to help, I have 20mb/s toll free => asterisk => pap2 |
| 05:05.33 | surfdue | i get alot of in and out |
| 05:05.38 | surfdue | like its trying to cancel echo |
| 05:05.43 | surfdue | how can I disable this? |
| 05:06.05 | Strom_C | surfdue: i'll be able to answer your question once you actually phrase it such that it makes sense |
| 05:06.17 | Dovid | StromC: can u look at my paste bin ? |
| 05:06.28 | surfdue | Strom_C, im sorry sir. Can you call my number and list what it sounds like? |
| 05:06.34 | surfdue | 1-800-617-0141 |
| 05:06.39 | surfdue | listen* |
| 05:06.54 | surfdue | wait it wont work cause I change the extensions |
| 05:07.01 | surfdue | Strom_C, basically its going in and out for some reason |
| 05:07.11 | Strom_C | surfdue: what do you mean "going in and out"? |
| 05:07.22 | surfdue | like works then doesnt everyother second |
| 05:07.34 | Strom_C | surfdue: put it back the way it was and let me dial it |
| 05:07.42 | surfdue | k |
| 05:07.43 | _Vile | sounds like cpu or jitter |
| 05:07.54 | _Vile | latency issues maybe |
| 05:08.13 | Strom_C | Dovid: what version of linux |
| 05:08.21 | Strom_C | Dovid: what distro, rather |
| 05:08.22 | i-ball | The free one. |
| 05:08.25 | surfdue | ok call |
| 05:08.30 | Dovid | Strom_C: Cent OS 4.3 |
| 05:08.41 | surfdue | _Vile, anything can help im on 1 gb proc and duel athlon |
| 05:09.29 | surfdue | err |
| 05:10.00 | surfdue | try now Strom_C sorry |
| 05:10.09 | surfdue | it just dosnt wanna work |
| 05:10.10 | surfdue | :) |
| 05:10.12 | surfdue | :(* |
| 05:11.10 | Strom_C | surfdue: jitter |
| 05:11.15 | surfdue | ok how do i fix that? |
| 05:11.21 | Strom_C | surfdue: you fix your network |
| 05:11.27 | surfdue | im sorry sir |
| 05:11.51 | surfdue | i dont know what you want me to do |
| 05:11.58 | Strom_C | surfdue: what kind of connection is the asterisk box on |
| 05:12.03 | surfdue | Strom_C, 20mb/s |
| 05:12.13 | Strom_C | ok, who is providing your number |
| 05:12.18 | surfdue | asterlink |
| 05:12.28 | surfdue | we are outputting a mier 0.38 KB/s of that 20mb |
| 05:12.29 | surfdue | atm |
| 05:12.53 | Strom_C | what happens when you try a different provider? |
| 05:13.53 | Dovid | Stom_C: any idea ? |
| 05:15.00 | Strom_C | Dovid: i dont know. centos is garbage. use something else. |
| 05:15.06 | Dovid | cant |
| 05:15.13 | Dovid | machine is in dedicated center |
| 05:15.18 | Dovid | never had this issue b4 |
| 05:15.39 | Strom_C | Dovid: well reinstall the OS or replace the machine or learn how the hell to use linux |
| 05:15.53 | *** join/#asterisk __flag__ (n=__flag__@59.163.66.98) |
| 05:16.21 | Dovid | ok |
| 05:16.26 | Kumba_ | zaptel is installed... again... and I got green lights... go me... |
| 05:17.07 | godsmoke | if I used SetMusicOnHold correctly, and then use "m" in a Dial command -- is there any explanation for why the music would work fine while on hold, but not during the ringing time? |
| 05:21.12 | Kumba_ | Think a 2.4ghz Machine with 1gb Ram can handle 18 lines and 8 phones? |
| 05:21.22 | Strom_C | Kumba_: easily |
| 05:21.25 | Kumba_ | sweet |
| 05:22.16 | *** join/#asterisk brut- (n=brut@66.173.4.254) |
| 05:22.44 | brut- | question: can asterisk play a straight mp3 or does it still have to be converted as of release 1.2.10? |
| 05:22.59 | Kumba_ | for future reference, if I want to add a module to asterisk, does that normally require recompiling? or does asterisk just link to it as an external module? |
| 05:23.19 | Strom_C | Kumba_: what do you mean "module"? |
| 05:23.36 | Kumba_ | Like if I want to add a web-based receptionist console... |
| 05:23.44 | Dovid | ~centosbig |
| 05:23.48 | Dovid | ~centosbug |
| 05:23.51 | jbot | i guess centosbug is a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h" |
| 05:23.54 | Strom_C | you mean FOP? |
| 05:23.57 | Kumba_ | like flash panel or something |
| 05:23.58 | Kumba_ | yeah... |
| 05:24.02 | Strom_C | you dont need to recompile for that |
| 05:24.07 | Kumba_ | ok... good... |
| 05:24.09 | Strom_C | that uses the manager interface |
| 05:24.16 | *** join/#asterisk pigpen2 (n=mark@fw.seamans.cc) |
| 05:25.07 | Kumba_ | Will I need to recompile if I plan on using SQL instead of hard files for the dialplan/etc? |
| 05:25.19 | Strom_C | no, dont think so |
| 05:25.23 | Kumba_ | Or is that loaded at runtime? |
| 05:25.23 | Kumba_ | kewl |
| 05:25.40 | *** join/#asterisk sponix (i=family@host-64-72-46-149.classicnet.net) |
| 05:26.21 | *** join/#asterisk CoderCR (n=creyna@ip68-6-237-193.sd.sd.cox.net) |
| 05:26.28 | *** part/#asterisk CoderCR (n=creyna@ip68-6-237-193.sd.sd.cox.net) |
| 05:26.43 | Kumba_ | And would you guys recommend MySQL or Postgre? (or does it not matter?) |
| 05:27.06 | surfdue | Strom_C, never tried |
| 05:27.07 | hads|home | text files are good :) |
| 05:27.17 | Strom_C | surfdue: try it |
| 05:27.22 | surfdue | Strom_C, like who? |
| 05:27.48 | Kumba_ | hehe... installed the P4 thermal notification on this kernel compile... and it keeps beeping that the CPU is too hot... guess i'll hafta fix that :) |
| 05:27.49 | Strom_C | surfdue: i dont know, voicepulse connect? |
| 05:28.22 | shidan | Kumba you wouldnt use a module for a web based console |
| 05:28.42 | surfdue | free? |
| 05:28.49 | Kumba_ | shidan: well I dont know what i'd use for a web based console... it's all new... :) |
| 05:28.57 | Strom_C | surfdue: no, you have to pay |
| 05:29.00 | shidan | youd use the manager api |
| 05:29.02 | surfdue | oh :( |
| 05:29.08 | surfdue | Strom_C, who can i test for free? |
| 05:29.13 | Strom_C | surfdue: for god's sake, it's FIVE DOLLARS |
| 05:29.20 | shidan | its a simple telnet interface |
| 05:29.20 | Nugget | telnet is eeeeeeevil! |
| 05:29.26 | hads|home | Grrr. |
| 05:29.27 | brut- | oh, aha, its in the readme |
| 05:30.58 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 05:30.59 | Strom_C | surfdue: or set up an account that I can dial into |
| 05:31.42 | brut- | so with format_mp3 compiled in, you can use mp3's running in 8kz mode in the moh application? |
| 05:32.03 | *** join/#asterisk brettnem (n=brettnem@72.29.102.158) |
| 05:34.13 | }btorch{ | anyone here uses meetmet ? |
| 05:34.24 | Strom_C | }btorch{: I use meetme |
| 05:35.12 | }btorch{ | for some wierd reason when I dial a conf number and join it musicon hold starts play like 2 seconds and then stops |
| 05:35.38 | }btorch{ | my music on hold is all default stuff |
| 05:36.11 | *** join/#asterisk SkramX (n=MarkS@admins.sentiensystems.net) |
| 05:36.31 | Strom_C | SkramX: no, I will not make a FWD number for my conference bridge :) |
| 05:37.27 | SkramX | Strom_C: right on |
| 05:37.31 | SkramX | pfft, IAX would be better |
| 05:37.33 | SkramX | im there now |
| 05:37.40 | Strom_C | you're on? |
| 05:37.41 | SkramX | im trying to get a wraspy voice so I can say "Hey big boys" |
| 05:37.43 | SkramX | Yes'si |
| 05:37.44 | SkramX | r |
| 05:37.48 | SkramX | 712 |
| 05:37.49 | Strom_C | oh just talk |
| 05:37.56 | SkramX | i shall |
| 05:39.57 | }btorch{ | Strom_C: what you use to play the mp3 files ? |
| 05:40.03 | Strom_C | mpg123 |
| 05:40.22 | }btorch{ | you are using 12.x ? |
| 05:40.31 | Strom_C | TWELVE?! |
| 05:40.37 | }btorch{ | 1.2.x |
| 05:40.44 | Strom_C | sorry, forgot it was 2018 |
| 05:40.56 | Strom_C | yes i'm using 1.2.x |
| 05:41.07 | Kumba_ | In zapata.conf, if I set signalling = fxsls, all associated groups (as you go line-by-line down the file) will be set-up using that signalling, till I define another type, and then all groups below that second type will inherit the second signalling type... right? |
| 05:41.20 | Strom_C | Kumba_: yes |
| 05:41.35 | Kumba_ | ok... :) |
| 05:41.57 | *** join/#asterisk timscott (n=a@d66-222-195-190.abhsia.telus.net) |
| 05:42.02 | timscott | hai i m a nub plz halp |
| 05:42.35 | Strom_C | timscott: i think you want #freepbx |
| 05:44.55 | Kumba_ | YAY! I got dial tone... |
| 05:46.28 | Strom_C | Kumba_: awesome |
| 05:46.59 | Kumba_ | Now I just need to get asterisk to hook the channel bank up to the outside T1... |
| 05:47.14 | Kumba_ | temporarily... till I can get these polycom's working right, and dialplans done, and etc... |
| 05:48.17 | Kumba_ | what would you suggest is the easiest way to have anything from Channel 25 sent to channel 1, and vica-versa? |
| 05:49.31 | Strom_C | so wait, you just want to have each channel ring an individual station? |
| 05:50.39 | Kumba_ | well... since it's 2am, i'm tired, and I think i've accomplished enough learning/brain meltdown... I just want to make asterisk be a pass-through... |
| 05:50.58 | Kumba_ | so that channel 1 is passed through to channel 25... and vica-versa... |
| 05:51.11 | *** part/#asterisk timscott (n=a@d66-222-195-190.abhsia.telus.net) |
| 05:51.14 | SkramX | sa... |
| 05:51.18 | SkramX | woops |
| 05:51.20 | Kumba_ | Since I dont feel like learning IVR/extensions/etc... |
| 05:51.58 | Kumba_ | but I did verify that I get tone from my channel bank now... and that I can call asterisk and it picks up... so that's good... |
| 05:52.06 | Strom_C | Kumba_: don't you have DNIS? |
| 05:52.12 | Kumba_ | DNIS? |
| 05:52.20 | Strom_C | dialied number identification service |
| 05:52.34 | *** join/#asterisk JohnJacob (n=dhorner@pool-71-127-102-43.aubnin.fios.verizon.net) |
| 05:52.38 | Kumba_ | CallerID? |
| 05:52.42 | Strom_C | no |
| 05:52.44 | Strom_C | dialied number identification service |
| 05:52.46 | Strom_C | er |
| 05:52.47 | Strom_C | dialed |
| 05:53.10 | _Vile | similar concept, the switch you are talking to wil tell you the number *dialed* |
| 05:53.16 | hads|home | The number someone dialed to get through to you. |
| 05:54.23 | _Vile | requires cas/e&m wink or a pri |
| 05:54.36 | _Vile | there's a few other standards that use it too' |
| 05:54.43 | _Vile | but those are the most common |
| 05:55.05 | Kumba_ | Dont have none of that... |
| 05:55.24 | Strom_C | Kumba_: barbaric |
| 05:55.26 | Kumba_ | I know... |
| 05:55.30 | _Vile | kumba, what's your setup? |
| 05:55.50 | _Vile | i saw channel bank |
| 05:55.55 | Kumba_ | TE205p... one span has a full T1 coming into it... the other span outputs to a channel bank.. |
| 05:55.56 | _Vile | and loopstart above i think |
| 05:56.15 | Kumba_ | I have verified that I get dial tone from the channel bank... and I can call in and here the asterisk example set-up |
| 05:56.41 | _Vile | so you have #s assigned to each channel on the bank |
| 05:56.50 | Kumba_ | now since I dont have dial plans/etc ready to go yet... |
| 05:56.55 | _Vile | err each channel on the t1 |
| 05:56.58 | Kumba_ | Yes |
| 05:57.06 | Kumba_ | the T1 originally just plugged into the channel bank |
| 05:57.28 | Kumba_ | I believe it used RBS... which is pretty basic... |
| 05:57.31 | Kumba_ | or... barbaric... |
| 05:58.07 | Kumba_ | I guess i'm just trying to shortcut it too much :) |
| 05:58.21 | Strom_C | Kumba_: each channel has a separate number associated with it? |
| 05:58.47 | Kumba_ | Each incoming channel from my provider has a phone number... (well 18 do, the other 6 aren't connected to anything on their end) |
| 05:58.56 | Kumba_ | but I have a full 24 channels coming in... |
| 05:59.04 | Kumba_ | Jul 19 01:54:21 WARNING[12400]: chan_zap.c:3922 zt_handle_event: Ring/Off-hook in strange state 6 on channel 9 |
| 05:59.06 | Strom_C | dear god, it's 1978 again |
| 05:59.24 | Kumba_ | What do you think that error is from? |
| 05:59.33 | Strom_C | beats me |
| 05:59.34 | benjk | can somebody please tell me their zaptel rules in /etc/udev/rules.d |
| 06:00.13 | benjk | for some reason the rules disappeared |
| 06:00.40 | benjk | seems to be volatile |
| 06:00.41 | _Vile | sec |
| 06:00.50 | benjk | second time this happened after a reboot |
| 06:01.18 | Kumba_ | I guess to do it right, I need to assign each channel on the Channel bank an Extension... and then set the routing of that extension to go to a channel on the T1... and then set the incoming routing for each channel on a T1 to go to a specific extension on the channel bank... |
| 06:01.23 | Kumba_ | but DAMN that sounds like work :( |
| 06:02.44 | benjk | can somebody please do grep -r zap /etc/udev/rules.d/* and tell me what it returns |
| 06:02.53 | Strom_C | Kumba_: please do yourself a favor and at least convert it into a hunt group with DNIS |
| 06:03.14 | Strom_C | benjk: it returns dead hookers |
| 06:03.15 | _Vile | haha |
| 06:03.19 | _Vile | he can't w/ ls |
| 06:03.21 | Snake-Eyes | is there a better way to grab real time cdr's other than using Asterisk Manager? eg some AGI script |
| 06:03.25 | _Vile | kumba |
| 06:03.27 | _Vile | easy |
| 06:03.38 | _Vile | for each channel |
| 06:03.43 | _Vile | in zapata.conf |
| 06:03.57 | _Vile | you need a context |
| 06:03.58 | _Vile | say |
| 06:04.04 | benjk | if you are running a 2.6 kernel and you have a zaptel device, you will have to have rules in /etc/udev |
| 06:04.17 | _Vile | context=t1_1 |
| 06:04.19 | Kumba_ | benjk: I sent it to you in message |
| 06:04.31 | benjk | yes saw it thanks a lot |
| 06:04.34 | _Vile | context=>t1_1 |
| 06:04.35 | Kumba_ | yup |
| 06:04.43 | _Vile | channel=>1 |
| 06:04.50 | _Vile | context=>t1_2 |
| 06:04.55 | _Vile | channel=>2 |
| 06:04.56 | _Vile | etc |
| 06:04.59 | _Vile | then |
| 06:05.06 | _Vile | build in your extensions.conf |
| 06:05.09 | _Vile | those contexts |
| 06:05.44 | Kumba_ | hmmmm |
| 06:05.52 | _Vile | use [t1_1] |
| 06:05.56 | _Vile | exten =>s,1,Dial(Zap/25) |
| 06:06.03 | _Vile | exten =>s,1,Hangup |
| 06:06.07 | _Vile | err |
| 06:06.13 | _Vile | s,2,Hangup |
| 06:06.14 | _Vile | done |
| 06:06.18 | _Vile | copy/paste a bunch of times |
| 06:06.33 | _Vile | Zap/26 for t1_2 |
| 06:06.35 | _Vile | get it? |
| 06:07.47 | Kumba_ | mmmm... so all my incoming channels I set the context as t1_<chan>... like t1_1, t1_2, t1_3... |
| 06:07.51 | *** join/#asterisk af_ (n=af@ip-170-209.sn1.eutelia.it) |
| 06:07.53 | _Vile | yep |
| 06:08.05 | Kumba_ | what exactly does setting the context like that accomplish? (just so I can understand) |
| 06:08.10 | _Vile | and build contexts for those t1_1, t1_2, etc |
| 06:08.19 | _Vile | it'll route that first channel to t1_1 |
| 06:08.22 | _Vile | in [t1_1] |
| 06:08.27 | _Vile | you will then dial zap/25 |
| 06:08.37 | _Vile | which is the first channel on the t1 going to your channel bank |
| 06:09.00 | Kumba_ | so in extensions I will have [t1_1] with the dial plan to go to chan 25? |
| 06:09.09 | _Vile | yep |
| 06:09.20 | Kumba_ | and then I would need to do the same thing for the second T1... but in reverse... |
| 06:09.31 | Kumba_ | so when channel 25 picks up, it send it to channel 1... |
| 06:09.35 | Strom_C | Kumba_: seriously dude....DNIS |
| 06:09.38 | _Vile | hehehe |
| 06:09.42 | _Vile | yeah dnis is easier |
| 06:09.55 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 06:09.57 | Kumba_ | well... I could wait a few weeks and get a PRI too... |
| 06:10.04 | Kumba_ | then I could set CID as well... |
| 06:10.08 | _Vile | kumba, yes you will need to play with that |
| 06:10.35 | _Vile | outbound, you could do a macro.. or something to save code |
| 06:11.08 | Kumba_ | ... |
| 06:11.09 | _Vile | if you don't care what channel it comes out on for outbound |
| 06:11.13 | Kumba_ | or...! |
| 06:11.20 | _Vile | then using callgroups would make life easier for you |
| 06:11.22 | Kumba_ | I could just hook the channel bank back to the T1 |
| 06:11.29 | _Vile | that's easiest |
| 06:11.41 | Kumba_ | go home... finish my 6-pack... and play dead :D |
| 06:11.46 | _Vile | bingo |
| 06:11.51 | Kumba_ | damn straight... |
| 06:13.07 | _Vile | I'm in a wait on an ast_waitfor(chan, -1) with an fr = ast_read(chan) and an if below that on if(fr->frametype == AST_FRAME_CONTROL) and it's just looping, never detecting that frametype -- any thought on what could cause that? |
| 06:14.30 | Kumba_ | b33r? |
| 06:14.32 | Kumba_ | :( |
| 06:17.10 | _Vile | ugh |
| 06:17.46 | Kumba_ | I'd love to help but i'm still mastering the complexities of channels :) |
| 06:18.06 | _Vile | haha |
| 06:18.46 | *** join/#asterisk fnordus (n=dnall@s142-179-111-243.bc.hsia.telus.net) |
| 06:24.11 | Kumba_ | hey vile... to use that context method you used... would I want to set group 1 (t1_1), and group 2 (t2_2) as call/pickup groups? |
| 06:24.56 | SkramX | Strom_C: people still on your conferencia? |
| 06:25.03 | Strom_C | yep |
| 06:27.41 | Kumba_ | ok |
| 06:30.13 | _Vile | kumba, you could, just not sure how it'd help you.. |
| 06:32.25 | Kumba_ | ok... each zap channel has a context... |
| 06:32.28 | Kumba_ | now for the extensions... |
| 06:34.41 | SkramX | Strom_C: hrmm, right on |
| 06:34.49 | SkramX | im chilling with the telephreak kids |
| 06:36.19 | russellb | telephreak? sounds l33t |
| 06:36.31 | Strom_C | bah, you just dont like my conference |
| 06:36.58 | russellb | who, me? i didn't know you had one |
| 06:37.02 | *** join/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg) |
| 06:37.05 | Strom_C | no, SkramX |
| 06:37.12 | russellb | k |
| 06:37.19 | littleball | hello, i am looking for voice whole sale provider in Germany. who can help or recommend? |
| 06:37.23 | Strom_C | russellb: |
| 06:37.30 | russellb | Strom_C: |
| 06:37.34 | Strom_C | 712-432-5282 |
| 06:37.46 | russellb | but but but ... that costs money |
| 06:37.56 | russellb | and i need to sleep, anyway |
| 06:38.00 | Strom_C | hehe ok |
| 06:43.21 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 06:43.58 | _Vile | i'll call the conf if someone there can help me w/ this: |
| 06:44.04 | _Vile | I'm in a wait on an ast_waitfor(chan, -1) with an fr = ast_read(chan) and an if below that on if(fr->frametype == AST_FRAME_CONTROL) and it's just looping, never detecting that frametype -- any thought on what could cause that? |
| 06:44.22 | _Vile | s/wait/while |
| 06:44.52 | Kumba_ | vile: I did that set-up you suggested... and it works for calls coming in from the T1... but if I try to originate a call from the channel bank... it doesn't connect... |
| 06:45.02 | _Vile | kumba yes |
| 06:45.03 | _Vile | now |
| 06:45.24 | SkramX | BAH! On my 12SP+ the caller can hear me, but I cannot hear them... |
| 06:45.26 | _Vile | in zapata.conf |
| 06:45.40 | _Vile | context=>outbound_25 |
| 06:45.45 | _Vile | channel=>25 |
| 06:45.47 | *** join/#asterisk Gunnar (n=gunnar@62.97.242.6) |
| 06:45.54 | _Vile | you need in extensions.conf |
| 06:46.02 | _Vile | [outbound_25] |
| 06:46.09 | _Vile | mmm sec |
| 06:46.27 | *** join/#asterisk s0lid (n=s0lid@gr-153-4.eglobalreach.net) |
| 06:46.43 | Kumba_ | context=T2-1 |
| 06:46.43 | Kumba_ | channel=>25 |
| 06:46.49 | _Vile | sure |
| 06:46.54 | _Vile | [T2-1] |
| 06:47.06 | Kumba_ | that's what I did for all the channels on the second T1... gave them contexts from T2-1 to T2-24... |
| 06:47.22 | Kumba_ | then in extensions.conf, I did the same set-up as I did for the first T1... |
| 06:47.28 | Kumba_ | was extensions.conf where I messed up? |
| 06:47.32 | _Vile | exten =>_.Dial(Zap/1/${EXTEN:${TRUNKMSD}}) |
| 06:47.39 | _Vile | err |
| 06:47.47 | _Vile | exten =>_.,1,Dial(Zap/1/${EXTEN:${TRUNKMSD}}) |
| 06:47.49 | i-ball | how do I stream a conversation from asterisk to the net? |
| 06:47.54 | _Vile | exten =>_.,2, |
| 06:48.07 | _Vile | err exten =>_.,2,Hangup |
| 06:48.12 | *** join/#asterisk vlt (n=dm@p54B31491.dip0.t-ipconnect.de) |
| 06:48.25 | _Vile | and T2-2 would be |
| 06:48.31 | _Vile | [T2-2] |
| 06:48.36 | _Vile | exten =>_.,1,Dial(Zap/2/${EXTEN:${TRUNKMSD}}) |
| 06:48.46 | _Vile | exten =>_.,2,Hangup |
| 06:48.51 | _Vile | etc |
| 06:49.31 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 06:50.03 | Kumba_ | So what exactly is going on here? |
| 06:50.09 | _Vile | well |
| 06:50.11 | Kumba_ | I understood the call incoming part ok... |
| 06:50.18 | _Vile | you have your inbound contexts |
| 06:50.20 | Kumba_ | this isn't as intuitive for me... |
| 06:50.25 | Kumba_ | right... |
| 06:50.25 | _Vile | now you need to set outbound contexts |
| 06:50.33 | _Vile | whihc would be on t2 |
| 06:51.04 | _Vile | pri = easier |
| 06:51.09 | Kumba_ | heh... |
| 06:51.14 | Kumba_ | mental note: next time get a PRI |
| 06:51.22 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 06:51.45 | _Vile | all you are doing is specifying a context for t2 |
| 06:51.49 | _Vile | above say channel 25' |
| 06:51.53 | Kumba_ | but is _., all part of telling it what to do on outgoing (originated) calls? |
| 06:51.57 | _Vile | saying use this context to dial out |
| 06:52.06 | _Vile | that's saying match anything |
| 06:52.12 | _Vile | and dial it |
| 06:52.28 | _Vile | it's a match |
| 06:52.29 | Kumba_ | But what if I want T2-2 to go to Zap channel 2? |
| 06:52.44 | Kumba_ | would I replace ${Exten with 2? |
| 06:52.45 | _Vile | then you are saying that by saying |
| 06:52.54 | _Vile | Dial(Zap/2/...... |
| 06:53.16 | _Vile | the $EXTEN crap |
| 06:53.20 | _Vile | is the # they dialed |
| 06:53.29 | Kumba_ | ohh... gotcha... |
| 06:53.32 | _Vile | the _. says to match anything |
| 06:53.44 | _Vile | and execute your dial command |
| 06:53.51 | _Vile | which dials the number they dialed |
| 06:53.54 | _Vile | over zap/2 |
| 06:53.55 | *** join/#asterisk fnordus (n=dnall@s142-179-111-243.bc.hsia.telus.net) |
| 06:54.09 | Kumba_ | and the MSD strips the digit off the front? |
| 06:54.12 | _Vile | yeah |
| 06:54.13 | _Vile | but |
| 06:54.15 | rob0 | i-ball: is it your wife talking to her bf? :) |
| 06:54.17 | _Vile | in your case |
| 06:54.24 | _Vile | you may want to set trunkmsd to 0 |
| 06:54.28 | _Vile | instead of 1 |
| 06:54.35 | _Vile | so |
| 06:54.39 | _Vile | ignore trunkmsd |
| 06:54.52 | _Vile | Dial(Zap/2/${EXTEN}) |
| 06:54.57 | Kumba_ | ahhh... ok... |
| 06:55.05 | Kumba_ | since I have 1978 technology |
| 06:55.15 | _Vile | ;) |
| 06:55.24 | _Vile | actually, channel banks are still used a lot |
| 06:55.32 | _Vile | and you can do a lot with your current setup |
| 06:55.34 | i-ball | hhahaa |
| 06:55.35 | i-ball | no |
| 06:55.44 | Kumba_ | I just didn't wanna buy ATA's... |
| 06:55.56 | Kumba_ | and this channel bank is provided under my contract... |
| 06:56.01 | _Vile | just gotta learn how to use zap |
| 06:56.04 | Kumba_ | figure it's the way to go :) |
| 06:56.11 | _Vile | and the extensions |
| 06:56.20 | _Vile | you'll have an auto attendant before no time |
| 06:56.45 | *** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com) |
| 06:57.09 | _Vile | no real big need to change unless you have a bunch of did's that do a bunch of different things |
| 06:57.24 | *** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135) |
| 06:57.35 | Kumba_ | well... I have to get all our voice prompts recorded for the Attendant... |
| 06:57.46 | Kumba_ | and do a dial plan... |
| 06:57.52 | Kumba_ | queue's... |
| 06:57.55 | _Vile | hardest part :) |
| 06:58.17 | _Vile | do you read voip-info.org? |
| 06:58.22 | _Vile | it will help you |
| 06:58.24 | Kumba_ | and then add timeslots to it... so from 9-5 people can bug the shit out of you... but the rest of the time you can die in voicemail hell... |
| 06:58.31 | Kumba_ | Yeah... i've been over it a lot... |
| 06:58.33 | _Vile | easy |
| 06:58.37 | _Vile | sec |
| 06:58.38 | Kumba_ | But I determined my problem was Trixbox... |
| 06:59.06 | Kumba_ | so I formatted, put slack 10.2 on it, upgraded to kernel 2.6... and an hour or two later i'm almost done... for today :) |
| 06:59.34 | Kumba_ | spent more time trying to make trixbox give me green lights on my digium card then I have doing a from-source install |
| 06:59.45 | Kumba_ | including downloading the iso's at 178K/sec |
| 06:59.49 | _Vile | include => aa-announce-oper-offhours|00:00-23:59|sat-sun|*|* |
| 06:59.49 | _Vile | include => aa-announce-oper-offhours|17:01-7:59|mon-fri|*|* |
| 06:59.49 | _Vile | include => aa-announce-oper-onhours|8:00-17:00|mon-fri|*|* |
| 07:00.25 | _Vile | hehehe |
| 07:00.40 | _Vile | i only use asterisk |
| 07:00.53 | _Vile | though bkw made freeswitch which im looking at too |
| 07:01.40 | DarKnesS_WolF | _Vile: hehe cool setup ;-) |
| 07:08.57 | Kumba_ | dear god... |
| 07:08.59 | Kumba_ | it... works...? |
| 07:09.19 | *** join/#asterisk Greek-Boy (n=grb@193.220.93.162) |
| 07:09.20 | Kumba_ | it... just... works... |
| 07:09.47 | Strom_C | Kumba_: congrats :) |
| 07:10.09 | _Vile | ;) |
| 07:10.14 | _Vile | now |
| 07:10.16 | _Vile | kumba |
| 07:10.21 | _Vile | hmm I'm in a while on an ast_waitfor(chan, -1) with an fr = ast_read(chan) and an if below that on if(fr->frametype == AST_FRAME_CONTROL) and it's just looping, never detecting that frametype -- any thought on what could cause that? |
| 07:10.34 | Kumba_ | Did you check the flux capacitor? |
| 07:10.38 | _Vile | haha |
| 07:11.31 | Kumba_ | wonder how faxes will work through this... |
| 07:12.05 | Kumba_ | I keep getting this in my logs... |
| 07:12.07 | Kumba_ | Jul 19 03:11:37 WARNING[12560]: chan_zap.c:3922 zt_handle_event: Ring/Off-hook in strange state 6 on channel 8 |
| 07:12.13 | Kumba_ | Think it's anything to worry about? |
| 07:12.55 | Kumba_ | it's always the channel answering the line that's giving that message... |
| 07:13.34 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 07:14.09 | _Vile | take this: "Ring/Off-hook in strange state" and -- www.google.com |
| 07:14.14 | _Vile | will turn a bunch of results |
| 07:14.46 | _Vile | use |
| 07:14.53 | _Vile | more results from lists.digium.com |
| 07:15.35 | *** join/#asterisk qdk (n=qdk@213.237.44.34) |
| 07:15.44 | E-bola | morning |
| 07:15.59 | Kumba_ | haha... guess the fax will work fine... just send a fax, from the CB, through *, through the T1, Trought the PSTN, back through the T1, back through *, and back out the CB... no problems... |
| 07:16.11 | Kumba_ | I dont think I can add anymore latency then that... |
| 07:16.25 | _Vile | fax will work fine |
| 07:16.46 | _Vile | check out the strange state tho |
| 07:17.00 | Kumba_ | Hmm... here's a new one... unable to create channel of type zap... |
| 07:17.25 | _Vile | more info? |
| 07:17.55 | Kumba_ | The T1 just went yellow... |
| 07:18.06 | Kumba_ | or atleast, the CLI said all 24 channels had an alarm... |
| 07:18.21 | _Vile | could explain it |
| 07:18.30 | Kumba_ | Jul 19 03:17:33 WARNING[12535]: chan_zap.c:6337 handle_init_event: Detected alarm on channel 24: Red Alarm |
| 07:18.35 | _Vile | do a zap show channels from the cli |
| 07:19.00 | _Vile | don't paste here |
| 07:19.22 | Kumba_ | it's showing them all with the right context's... |
| 07:20.28 | _Vile | well channel 24 is not up right? |
| 07:20.39 | _Vile | you said 19-24 was not live? |
| 07:21.12 | _Vile | you only have 1-18 active circuits on your telco provided t1 right? |
| 07:21.16 | Kumba_ | Right... only 18-24 are assigned numbers and accessible to the PSTN... but all channels had an alarm... |
| 07:21.22 | Kumba_ | err 1-18... |
| 07:21.30 | Kumba_ | 19-24 = unassigned, but are there... |
| 07:21.44 | _Vile | check the back of the pbx and see if the t1 is red/yellow |
| 07:21.48 | Kumba_ | Jul 19 03:17:33 WARNING[12535]: chan_zap.c:6337 handle_init_event: Detected alarm on channel 1: Red Alarm |
| 07:21.54 | _Vile | hm |
| 07:21.56 | Kumba_ | both lights are green |
| 07:22.18 | Kumba_ | shitty T1 line? |
| 07:23.10 | *** join/#asterisk creativx (n=creadure@196.82-134-19.bkkb.no) |
| 07:24.11 | _Vile | did it just happen, all 24 channels at the same time? or? |
| 07:24.27 | Kumba_ | Yeah... all done... 2-seconds later... all up... |
| 07:24.35 | Kumba_ | done = down |
| 07:24.46 | _Vile | probably a bounce, they work now? if so, don't worry about it |
| 07:24.52 | Kumba_ | yeah... |
| 07:25.39 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
| 07:25.56 | _Vile | if it bounces once a week |
| 07:25.59 | _Vile | then call the telco |
| 07:26.14 | _Vile | once every few months, don't :) |
| 07:26.24 | Kumba_ | I do keep getting a lot of notices about funny hooks and unable to create channel... |
| 07:26.37 | _Vile | could be a bouncing T |
| 07:26.46 | _Vile | i'd look into it |
| 07:26.58 | _Vile | mmm |
| 07:27.02 | Kumba_ | well... since I have the ability to log it now... I can see how many times it bounces... |
| 07:27.04 | _Vile | you need zttool |
| 07:27.08 | _Vile | yes |
| 07:27.14 | Kumba_ | didn't it come with the zaptel package? |
| 07:27.19 | _Vile | zttool can tell you how many violations you get |
| 07:27.20 | _Vile | yes |
| 07:27.34 | _Vile | if you get bpv's/bi-polar violations |
| 07:27.36 | _Vile | consistently |
| 07:27.40 | _Vile | *let the telco know* |
| 07:27.54 | _Vile | could be a timing thing too |
| 07:28.09 | Kumba_ | I need to put FOP on here too... cheaper then buying a receptionist phone for when the queue's get busy :) |
| 07:28.10 | _Vile | are you pulling timing from them in zaptel.conf? |
| 07:28.15 | *** join/#asterisk SwK (n=Silik0nJ@12-218-74-89.client.mchsi.com) |
| 07:28.17 | Kumba_ | yeah... i'm slave... |
| 07:28.20 | _Vile | good |
| 07:28.23 | _Vile | ok |
| 07:28.26 | _Vile | watch logs |
| 07:28.30 | _Vile | use zttool |
| 07:28.32 | _Vile | etc |
| 07:29.21 | Kumba_ | it didn't compile zttool... *read the makefile* |
| 07:29.26 | _Vile | :) |
| 07:29.27 | benjk | Is there any way to set the TON part of the called number in the Q.931 call setup message when dialing out? |
| 07:29.37 | _Vile | benj sec |
| 07:29.46 | benjk | I have got this ... |
| 07:29.47 | benjk | > Called Number (len= 6) [ Ext: 1 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '116' ] |
| 07:30.04 | benjk | but 116 is a special number of the phone company |
| 07:30.18 | benjk | thus TON must not be set to Subscriber Number |
| 07:30.20 | _Vile | hmmmm |
| 07:30.29 | benjk | it has to be 0 |
| 07:30.44 | _Vile | you got that msg back from * |
| 07:30.46 | _Vile | ? |
| 07:31.34 | benjk | bri intense debug will show you the complete Q.931 message transcript |
| 07:31.43 | benjk | or likewise pri intense debug |
| 07:31.46 | _Vile | yep |
| 07:31.58 | benjk | and when I switch that on and dial 116 |
| 07:32.04 | benjk | then I see this |
| 07:32.05 | benjk | > Called Number (len= 6) [ Ext: 1 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '116' ] |
| 07:32.10 | benjk | amongst other stuff |
| 07:32.35 | benjk | the phone company says that for calling 1xx numbers, TON must be set to 0 |
| 07:32.39 | benjk | but its 4 |
| 07:32.44 | *** join/#asterisk Juggie (n=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) |
| 07:33.01 | benjk | which makes perfect sense because 116 is the phone company operator, therefore not a subscriber number |
| 07:33.06 | Kumba_ | zttool wont compile... joy... |
| 07:33.10 | Kumba_ | *digs more* |
| 07:33.20 | benjk | yeah zttool is broken |
| 07:33.30 | SkramX | Strom_C, my buddy.... |
| 07:33.32 | SkramX | app_cepstral.c:26:33: ../include/asterisk.h: No such file or directory |
| 07:33.32 | SkramX | make: *** [app_cepstral.so] Error 1 |
| 07:33.32 | SkramX | sentien-support apps # pwd && file ../include/asterisk.h |
| 07:33.32 | SkramX | /usr/src/asterisk-1.2.4/apps |
| 07:33.33 | Kumba_ | ahhh... in new zaptel? |
| 07:33.34 | SkramX | ../include/asterisk.h: ASCII C program text |
| 07:33.37 | SkramX | eeeeeeeeeks |
| 07:34.12 | tzafrir_laptop | Kumba_, you need newt/libnewt? |
| 07:34.24 | benjk | so anybody any idea how to set the TON field on an outgoing call |
| 07:34.28 | Kumba_ | for the new ZTTool to compile? |
| 07:35.08 | _Vile | channels/misdn/isdn_lib.h: NUMPLAN_SUBSCRIBER=0x4, |
| 07:35.22 | benjk | I am using Zaphfc |
| 07:35.23 | _Vile | hm |
| 07:36.16 | _Vile | you could set it there |
| 07:36.18 | benjk | so you're saying this is something that can't be configured |
| 07:36.33 | benjk | has to be changed in the source code |
| 07:36.35 | _Vile | not as far as i know |
| 07:36.52 | _Vile | it could be coded |
| 07:36.53 | benjk | ok, not a problem then |
| 07:36.55 | _Vile | and patched in |
| 07:37.00 | _Vile | for future release |
| 07:37.01 | benjk | yeah, I will patch it |
| 07:37.06 | benjk | nah |
| 07:37.10 | *** join/#asterisk }btorch{ (n=kvirc@c-66-176-87-59.hsd1.fl.comcast.net) |
| 07:37.21 | *** join/#asterisk Johnnie (n=john@pdpc/supporter/active/Johnnie) |
| 07:37.27 | benjk | Digium isn't going to accept the patch anyway |
| 07:37.32 | }btorch{ | Strom_C: hey do you know where I can increase the registration timeout for iax ? |
| 07:37.50 | _Vile | probably not :) |
| 07:37.59 | }btorch{ | I got a that user now in japan who is trying but the registration times out in 30 seconds |
| 07:38.01 | benjk | and it will send some business my way |
| 07:38.34 | benjk | that's likely to be a firewall issue though btorch |
| 07:39.04 | benjk | tell them to enable qualify=yes |
| 07:39.10 | benjk | in their IAX peer |
| 07:39.27 | benjk | this should keep the entry in their NAT router's conntrack table alive |
| 07:40.08 | *** join/#asterisk MatsK (n=mats@141.221.181.62.in-addr.dgcsystems.net) |
| 07:40.21 | benjk | otherwise you could write a script that pings your server once every 29 seconds and let them run it in the background |
| 07:40.59 | benjk | reduce the interval in 1 sec steps until you find the exact value where its still working |
| 07:41.18 | creativx | well have ya seen.. mpg123 hogging 99% cpu |
| 07:41.23 | creativx | why am I still using that crap |
| 07:41.33 | benjk | use madplayer instead |
| 07:41.40 | }btorch{ | benjk: you mean turn qualify on on the user extension on eax.conf? |
| 07:42.01 | *** join/#asterisk implicit (n=implicit@ip68-4-84-39.oc.oc.cox.net) |
| 07:42.05 | benjk | no, your user will need to turn qualify on in their iax.conf |
| 07:42.10 | benjk | in the entry for your server |
| 07:42.51 | Kumba_ | Who was saying that zttool is broken? |
| 07:43.08 | }btorch{ | well I can turn on qualify to an individual per too |
| 07:43.10 | Kumba_ | is there not a working version somewhere? |
| 07:43.26 | _Vile | zttool works for me currently |
| 07:43.26 | benjk | what probably happens there is that when they register, their NAT router keeps track of the connection to your server so if your server sends something back, the NAT router still remembers where it has to be sent to |
| 07:44.00 | benjk | but at some point those conntrack entries expire |
| 07:44.00 | _Vile | want me to check my ver? |
| 07:44.01 | Kumba_ | please... |
| 07:44.02 | Kumba_ | 1.2.7 zttool wont compile for me |
| 07:44.13 | Juggie | #1, upgrade to 1.2.10 |
| 07:44.17 | Juggie | #2, whats the error? |
| 07:44.27 | benjk | if that time is shorter than your client re-registering, there will be no valid return path because the NAT router has forgotten you |
| 07:44.33 | Kumba_ | they have a zaptel 1.2.10? |
| 07:44.41 | Juggie | er, my bad. |
| 07:44.52 | Juggie | i wish they kept the version numbers in sync :) |
| 07:44.55 | Juggie | whats the error |
| 07:44.58 | Juggie | from the compile. |
| 07:44.59 | Kumba_ | I was just wanting to use zttool to tell me how many bounces i'm getting on my T1 |
| 07:45.07 | Kumba_ | since now I have means to log it... |
| 07:45.07 | benjk | btorch, the point is that the connection has to be reestablished from their end, not yours |
| 07:45.08 | Juggie | uhhuh, whats the compile error. |
| 07:45.11 | }btorch{ | there is no way to increase the registration timeout too ? |
| 07:45.23 | Juggie | reg timeout is in the sip.conf |
| 07:45.42 | benjk | it won't help you to increase the time out |
| 07:45.49 | Juggie | Kumba_, you going to tell me the error or not. |
| 07:46.02 | Juggie | you can set the registration time in there? what are you talking about. |
| 07:46.08 | Kumba_ | the compils error is a page and a half long... lots of implicit declarations of functions... and undeclared's... |
| 07:46.11 | Kumba_ | err compile |
| 07:46.14 | benjk | because what really matters is the entry in the user's NAT router's conntrack table |
| 07:46.20 | Juggie | Kumba_, www.pastebin.ca |
| 07:46.24 | Juggie | paste it there, and send me a link. |
| 07:46.41 | _Vile | later kumba, bed for me, good luck |
| 07:46.56 | Juggie | benjk, registration wont help nat. |
| 07:47.03 | Juggie | you need to use sip qualify |
| 07:47.06 | Kumba_ | http://pastebin.ca/92136 |
| 07:47.08 | benjk | if they can reconfigure their NAT router to increase the expiry period for conntrack entries, that would be ok, but I doubt a) the router allows that and b) they will know how to do it even if the router allowed it |
| 07:47.18 | Kumba_ | Thanks a million Vile... |
| 07:47.30 | benjk | Juggie that's for IAX not SIP |
| 07:47.39 | Juggie | sip has qualify |
| 07:47.41 | benjk | at least he'd asked about IAX |
| 07:48.01 | Juggie | Kumba_, what linux are you running? |
| 07:48.15 | Kumba_ | Slack 10.2 with kernel 2.6.17.6 |
| 07:48.27 | benjk | but its a moot point because the issue is the conntrack table entry in the NAT router of the user, in particular the expiry period of those entries |
| 07:48.33 | Kumba_ | The rest of the zaptel package works like a charm |
| 07:48.34 | *** join/#asterisk creativx (n=creadure@196.82-134-19.bkkb.no) |
| 07:48.55 | Juggie | Kumba_, does slack have a package manager? i'm not famalir with it. |
| 07:48.58 | Juggie | yum, apt? |
| 07:49.08 | Kumba_ | pkgtool |
| 07:49.12 | Kumba_ | what am I looking for? |
| 07:49.18 | Juggie | well, newt.h |
| 07:49.26 | Juggie | obviously means you need i'm guessing newt-dev |
| 07:50.02 | Juggie | on centos its newt-devel |
| 07:50.06 | Kumba_ | well... lemme look... |
| 07:50.11 | Juggie | which would also install plain ol newt as a dependancy. |
| 07:50.37 | Juggie | install that, and you will be good to go. |
| 07:50.58 | Juggie | allways look @ the first couple of lines of the error. |
| 07:51.01 | Juggie | the rest are junk |
| 07:51.14 | Juggie | because they all exist because newt.h failed to include. |
| 07:51.17 | rob0 | I don't think Slackware has newt. At least, it didn't in 10.0 and earlier. |
| 07:51.34 | rob0 | I built it from source once. |
| 07:51.40 | *** join/#asterisk Tordah (n=Ross@213-152-55-237.dsl.eclipse.net.uk) |
| 07:52.10 | *** join/#asterisk stephane_ (n=stephane@merlin.cabale.net) |
| 07:52.20 | stephane_ | jour/hi/hello |
| 07:52.47 | Kumba_ | 10.2 doesn't have a newt package... i'll be building from source... |
| 07:53.21 | rob0 | 11 is coming soon. You should consider upgrading to -current. |
| 07:53.50 | Tordah | Hey, I was wondering if anyone could help me with this error message: "Jul 19 08:49:26 NOTICE[10706]: app_dial.c:1040 dial_exec_full: Unable to create channel of type '(IAX2' (cause 66 - Channel not implemented) |
| 07:53.50 | Tordah | <PROTECTED> |
| 07:53.51 | Juggie | i cant even find the newt source |
| 07:55.18 | many | sounds more like chan_iax.so is not loaded |
| 07:55.18 | Kumba_ | I found it... |
| 07:55.22 | rob0 | I don't remember how I found it. But I do remember now that I didn't consider zttool worth the bother. :) |
| 07:55.43 | Tordah | well iax is working for incoming calls |
| 07:56.01 | rob0 | I've moved my zaptel over to a Slamd64 box. |
| 07:56.23 | Juggie | i use centos x86_64 |
| 07:56.26 | Juggie | with success. |
| 07:56.30 | many | in that case.. :)= |
| 07:56.35 | Kumba_ | all done |
| 07:56.39 | Kumba_ | and... it's bed time... |
| 07:56.48 | *** join/#asterisk bofh42 (n=bofh42@p5482A187.dip0.t-ipconnect.de) |
| 08:00.39 | *** join/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg) |
| 08:00.41 | littleball | hello, i am looking for voice whole sale provider in Germany. who can help or recommend? |
| 08:01.05 | *** join/#asterisk i-ball (n=i-ball@nat.hackerhalfwayhouse.org) |
| 08:01.07 | i-ball | hey |
| 08:01.53 | i-ball | when using MeetMe and using the "d" flag does the teleconference that is created get a random number assigned to it? |
| 08:02.03 | i-ball | or will it always use the number specified? |
| 08:06.22 | Strom_C | specified number |
| 08:06.46 | }btorch{ | I hate windows |
| 08:06.50 | i-ball | excellent, thanks |
| 08:06.56 | Strom_C | i-ball: think about it |
| 08:07.16 | Strom_C | i-ball: how can you get people to meet in a conference if you're throwing them in random conferences? |
| 08:07.34 | i-ball | I'm starting to understand that asterisk-ice tutorial |
| 08:07.38 | i-ball | I'm not |
| 08:07.52 | i-ball | I'm just making sure that it's going to use the assigned number. |
| 08:07.58 | i-ball | I don't want to assume these things. |
| 08:08.05 | }btorch{ | benjk: As I thought it's not the qualify is just that stupid windows keeps trying to talk to my server external IP when copnnected over the vpn and not the internal ip |
| 08:08.30 | }btorch{ | windows does that all the time |
| 08:08.34 | }btorch{ | it sucks |
| 08:09.09 | benjk | you should have told me about there being alternative routes (VPN and all that) ;) |
| 08:09.55 | benjk | and to be clear, it wouldnt have been anything to do with qualify even if it had been the NAT router's forgetfulness |
| 08:10.08 | Strom_C | benjk: he quit |
| 08:10.17 | benjk | oh |
| 08:10.22 | benjk | oh well |
| 08:10.26 | benjk | :) |
| 08:10.43 | Tordah | erm any idea with my problem? |
| 08:13.55 | littleball | hello, i am looking for voice whole sale provider in Germany. who can help or recommend? |
| 08:22.39 | *** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81) |
| 08:24.01 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
| 08:26.10 | *** join/#asterisk Kernel_core (i=Kernel_C@217.218.80.198) |
| 08:29.49 | Tordah | Hey, I was wondering if anyone could help me with this error message: "Jul 19 08:49:26 NOTICE[10706]: app_dial.c:1040 dial_exec_full: Unable to create channel of type '(IAX2' (cause 66 - Channel not implemented) |
| 08:33.14 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 08:33.23 | moon06 | hi all :) |
| 08:34.45 | moon06 | I'm having a big problem with my Asterisk installation under Gentoo ... I do not have any sound passing thru with SIP hard/soft phones |
| 08:35.22 | moon06 | if the incoming call is managed by chan_capi, it works great |
| 08:35.49 | *** join/#asterisk kristalino (n=kristali@84-50-84-146-dsl.trt.estpak.ee) |
| 08:35.50 | moon06 | but between 2 internal sip phones, there's absolutely no sound :( |
| 08:36.23 | moon06 | and I got the same Asterisk version with the same Asterisk config working on a redhat computer :( |
| 08:37.48 | moon06 | anyone that could help me ? :) |
| 08:39.41 | MrChimpy | tcpdump the sip ports on UDP to look for traffic |
| 08:39.49 | MrChimpy | check logs |
| 08:39.50 | MrChimpy | the usual |
| 08:41.32 | *** join/#asterisk jm|home (n=jamiem@dsl-217-155-242-137.zen.co.uk) |
| 08:41.39 | moon06 | MrChimpy, nothing in the logs |
| 08:42.59 | moon06 | gonna check tcpdump |
| 08:44.21 | *** join/#asterisk kapsel (i=kapsel@irc.thinkgeek.dk) |
| 08:44.40 | *** join/#asterisk [Airwolf] (n=airwolf@dsl51B6AE8A.pool.t-online.hu) |
| 08:45.42 | *** join/#asterisk nicox (n=nicox@83-64-42-210.prater.xdsl-line.inode.at) |
| 08:46.31 | moon06 | MrChimpy, my SIP phone has it's "UDO port 5004 unreachable" ... |
| 08:48.24 | moon06 | WTF it simply was my Grandstream SIP phone not working that time :( |
| 08:50.00 | i-ball | you said TWO internal SIP phones |
| 08:50.25 | i-ball | Your Grandstram SIP phone is one. |
| 08:50.29 | i-ball | What's the other? |
| 08:53.59 | nicox | Hello, did anybody know something about the chan_ss7 channel dirver? |
| 08:55.23 | nicox | anybody who tested it? |
| 08:55.48 | *** join/#asterisk tparcina (n=tparcina@lns02-1300.dsl.iskon.hr) |
| 08:55.53 | tparcina | hi channel |
| 08:56.02 | nicox | hi tparcina |
| 08:56.08 | tparcina | hi nicox |
| 08:56.50 | Tordah | lo |
| 08:58.55 | CMike | Anyone know I how hide a callerid on a outgoing zapchannel ? |
| 08:59.29 | moon06 | i-ball, X-lite |
| 09:00.17 | moon06 | when the capi incoming call goes to the Xlite SIP softphone, the sound works great |
| 09:00.23 | nicox | setvallerpres(prohib) |
| 09:00.28 | nicox | setcallerpres(prohib) |
| 09:00.45 | CMike | Hm.. |
| 09:01.02 | CMike | What does the "restricicid" field in SIP.conf do? |
| 09:01.15 | CMike | I thought that was for hiding the user callerid ? |
| 09:01.39 | CMike | <-- need an easy way to control the callerid for different sip-users |
| 09:01.55 | nicox | i don't know if asterisk is using this field also for outgoing zap channels |
| 09:02.06 | tparcina | CMike: analog or PRI? |
| 09:02.10 | CMike | E1 |
| 09:02.30 | CMike | so .. pri :) |
| 09:02.45 | nicox | talk with your telco |
| 09:02.58 | nicox | it must work with setcallerpres |
| 09:03.03 | tparcina | CMike: then you need to contact your provider to tell you how to send call to them so that CID is hiden |
| 09:03.16 | CMike | well the CID should only be hidden for some users |
| 09:03.26 | CMike | I have about 50000 DID numbers on that pri |
| 09:03.42 | tparcina | CMike: well, you will configure your dialpan for only those users |
| 09:04.47 | nicox | Hello, did anybody know something about the chan_ss7 channel dirver? |
| 09:04.52 | *** join/#asterisk jbsolutios (n=jbenson@193.93.153.1) |
| 09:06.22 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 09:07.58 | CMike | THe dialplan is the same for all users.. I used to use the callerid field for setting "Anonymous" but that was when I was using a SIP gw to PSTN.. Now I'm using zaptel |
| 09:08.08 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 09:08.22 | CMike | so I have to figure out a way to hide the callerid with a database entry |
| 09:08.27 | CMike | so to speak.. |
| 09:10.34 | tparcina | CMike: do you know how to hide callerid on one outgoing call? |
| 09:11.05 | tparcina | CMike: if not - contact you provider, it yes then do it like this |
| 09:12.12 | CMike | the q931 status says "number not screened" when I dial out |
| 09:12.19 | qdk | CMike: you use SS7? |
| 09:12.21 | tparcina | CMike: check who is calling, if he needs to hide callerid, let him dias thrue 1st dialplan, if he doesn't need then let him dials thrue 2nd dial plan. |
| 09:12.23 | CMike | nope |
| 09:12.50 | CMike | I'll have to try a few different dialplans I guess |
| 09:13.20 | qdk | CMike: i have the same issue when the call goes through a IAX channel |
| 09:14.23 | qdk | CMike: the SIP.conf setting gets lost and i cant trigger the setcallerpres on the (in my case) SS7 machine. |
| 09:14.38 | CMike | ah .. |
| 09:14.54 | CMike | I have to learn how to use SS7 later on :) |
| 09:15.03 | implicit | lol |
| 09:15.14 | CMike | I guess I'll have to put the "hidden users" in a different context.. |
| 09:16.01 | qdk | CMike: i fixed the issue with a dedicated (screened ID) IAX channel, and switch to it by context in sip instead of the sip options which is lost in IAX. |
| 09:16.09 | qdk | CMike: correct |
| 09:16.17 | CMike | ok.. thnx,. |
| 09:16.34 | CMike | another question .. |
| 09:16.42 | *** join/#asterisk Modcuts (n=bob@lan.proporta.com) |
| 09:17.22 | CMike | if I have a PRI that shows incoming CID, even if prohib. how to I screen the CID before dialing a SIP client. |
| 09:17.39 | CMike | is there a way to see on the zapchannel that this call is screening prohib ? |
| 09:18.34 | *** join/#asterisk s0lid (n=s0lid@gr-153-4.eglobalreach.net) |
| 09:19.38 | *** join/#asterisk jalsot (n=tamas@abacus.eworldcom.hu) |
| 09:19.44 | qdk | CMike: you need it on the ZAP for log and billing issues, but you can screen it on the channel you forward the call to. |
| 09:20.07 | CMike | sort of |
| 09:26.00 | moon06 | it's incredible, I jute tried with my 2nd Grandstream GXP-2000 phone, and still no sound ! |
| 09:26.38 | Modcuts | <moon06> :No errors? just no sound? |
| 09:26.40 | *** join/#asterisk _omer (n=_omer@202.38.51.2) |
| 09:26.47 | _omer | hi |
| 09:26.53 | moon06 | Modcuts, yes, just no sound passing thru |
| 09:26.53 | _omer | anyhelp ...... http://pastebin.ca/92188 |
| 09:26.59 | moon06 | but it works with Xlite |
| 09:27.11 | moon06 | between an outbound (capi) line and xlite |
| 09:28.20 | Modcuts | moon06: you got the vocoders in the grandstream set right? |
| 09:28.34 | moon06 | "vocoders" ? |
| 09:28.41 | moon06 | Codecs ? |
| 09:29.57 | Modcuts | yep, they are codecs but grandstream calls them vocoders on the account page |
| 09:30.23 | Modcuts | make sure you are using the same ones as set in asterisk or in the sip.conf? |
| 09:30.23 | moon06 | Modcuts, oh yes |
| 09:30.31 | MrChimpy | vocoders? who do they think they are? stevie wonder? cher? |
| 09:31.04 | Modcuts | MrChimpy: Don't ask me ask grandstream....... |
| 09:31.11 | MrChimpy | sparky the magic piano? |
| 09:31.16 | Modcuts | lol |
| 09:31.21 | *** join/#asterisk djulius (n=danj@bzq-88-155-205-172.red.bezeqint.net) |
| 09:32.08 | Modcuts | moon06: any luck? |
| 09:32.22 | moon06 | tried to change smth |
| 09:32.33 | moon06 | trying to make a call |
| 09:33.20 | moon06 | Modcuts, still nothing |
| 09:33.41 | moon06 | and if by example, I internally call the Voicemail, I don't hear anything |
| 09:33.58 | djulius | Hi everyone - newbie question here: I have a SIP account on an asterisk server in the US. I would now like to configure my Asterisk system at home to connect to the remote server, and allow me to make calls between the two. Can somebody help? |
| 09:34.13 | Tordah | you said voicemail.conf correctly? |
| 09:34.30 | _omer | anyhelp ...... http://pastebin.ca/92188 |
| 09:35.05 | moon06 | Tordah, yep |
| 09:35.30 | Tordah | what error message comes up? |
| 09:35.41 | Tordah | or |
| 09:35.47 | Tordah | there's just no sound? |
| 09:36.00 | Tordah | you got the sound files set up? :p does it say playing etc. file on asterisk console? |
| 09:36.38 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 09:37.52 | moon06 | Tordah, no error messages, and the console shows "playing ..." |
| 09:38.08 | moon06 | the Voicemail works great from an outbound ISDN line |
| 09:38.14 | Tordah | -- Playing 'vm-login' (language 'en') |
| 09:38.17 | Tordah | ? |
| 09:38.22 | Tordah | ah |
| 09:38.24 | moon06 | yes |
| 09:38.35 | Tordah | that's strange |
| 09:39.05 | Tordah | i haven't set up external voicemail. just works fine internally.. shouldnt be any problem with it really. |
| 09:40.06 | Tordah | what do you have in extensions.conf? |
| 09:40.19 | moon06 | Tordah, actually, there's no sound at all with this SIP hardphone at all |
| 09:40.30 | Tordah | oh. |
| 09:40.34 | Tordah | what phone is it? |
| 09:40.39 | moon06 | GXP-2000 |
| 09:40.53 | moon06 | but it still works with an Asterisk RedHat installation |
| 09:41.03 | Modcuts | i swear it's the codec stuff sounds like it, what codec are written in the sip.conf |
| 09:41.14 | moon06 | disallow=all |
| 09:41.14 | Tordah | yeah it could be |
| 09:41.17 | moon06 | allow=ulaw |
| 09:41.34 | moon06 | but smth strange in tcpdump on the Asterisk machine |
| 09:41.43 | moon06 | let me go look for a sec |
| 09:42.09 | Tordah | if its registering with server and dialing out, i had that problem it would dial with no sound, so it could be an issue with ports. But then it's internal there should be no problem |
| 09:42.30 | Tordah | use tethereal to see whats being sent between the two? |
| 09:43.36 | moon06 | my Asterisk machine says : ICMP 192.168.1.23 udp port 5008 unreachable, length 36 |
| 09:43.55 | Tordah | that's another thing on your server do you have a firewall enabled? |
| 09:44.16 | moon06 | nope |
| 09:44.27 | moon06 | actually I do have one on my gateway |
| 09:44.35 | moon06 | but this one isn't the gateway |
| 09:44.37 | Tordah | but it's internal. so that wouldn't affect it |
| 09:45.03 | *** join/#asterisk Gr1ncheux_ (n=devine@AStDenis-105-1-35-246.w80-8.abo.wanadoo.fr) |
| 09:45.10 | Tordah | make sure phone settings are correct to be honest. |
| 09:45.34 | Modcuts | The voicemail calls should be working i use the exact same phone and i have no problems with sound, the vocodders are all ulaw yes? and the message type is sip? |
| 09:45.54 | moon06 | Modcuts, yes and yes |
| 09:46.04 | Modcuts | that is very strange |
| 09:46.10 | moon06 | Tordah, gonna try with the settings that work for my other Asterisk installation |
| 09:46.40 | Tordah | good idea |
| 09:48.01 | Tordah | when I set it up it was as simple as "exten => 500,1,VoiceMailMain() ;" |
| 09:48.01 | moon06 | that's what I did |
| 09:48.26 | Tordah | is this just one phone it isn't working on |
| 09:48.27 | Tordah | or both? |
| 09:48.36 | moon06 | both |
| 09:48.44 | moon06 | but even DTMF don't work :( |
| 09:48.56 | moon06 | still not workin |
| 09:49.05 | Tordah | ; |
| 09:49.07 | Tordah | :< |
| 09:49.47 | Modcuts | Use a syslog server and set type to debug, and see what message are being sent, and you have two gxp2000s your trying this on or one? |
| 09:50.03 | moon06 | Modcuts, 2 |
| 09:50.09 | _omer | any body ...... http://pastebin.ca/92188 |
| 09:50.20 | Tordah | moon, can you hear anything at all on anything else? |
| 09:50.27 | Tordah | excluding voicemail |
| 09:50.37 | moon06 | nope |
| 09:50.39 | Tordah | if you dial from one to the other |
| 09:50.42 | Tordah | ah |
| 09:50.53 | Tordah | something isn't set up on phone properly then:p |
| 09:51.22 | moon06 | that's so strange cause it was working few mins ago with another Asterisk gateway |
| 09:51.50 | Tordah | ip address changed? |
| 09:52.05 | moon06 | Tordah, no ... |
| 09:52.47 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 09:53.10 | Tordah | alright, try dialing from one phone to the other directly on phone using ip addresses of each phone for example 192*168*23*59. |
| 09:53.11 | *** join/#asterisk Synyn (n=Synyn@cpe-72-181-72-81.houston.res.rr.com) |
| 09:53.21 | Synyn | hola |
| 09:53.30 | Tordah | actually, if you said it was working before probably dont need to check that:p |
| 09:53.45 | moon06 | ok, I try this |
| 09:54.24 | Tordah | something definitely is being blocked if they can dial and then not hear anything.. |
| 09:56.21 | *** join/#asterisk uwe (n=uwe@dogbert.palnet.com) |
| 09:59.04 | Synyn | anyone used the OEM X100P from DigitNetworks? |
| 09:59.24 | *** join/#asterisk Splat (n=Splat@220-253-101-189.TAS.netspace.net.au) |
| 09:59.38 | moon06 | Tordah, direct call doesn't work either |
| 09:59.54 | _4d4m_ | _omer: you need to install gnu c++ libraries |
| 09:59.57 | Tordah | lol |
| 10:00.11 | Tordah | sounds to me like an issue totally with the phones then |
| 10:00.50 | moon06 | when I call 192*168*1*23 it tells me 404 error |
| 10:00.50 | _omer | _4d4m_ : let me check plz..... |
| 10:01.11 | Tordah | maybe yours doesn't support that |
| 10:01.13 | Tordah | :P |
| 10:01.54 | Tordah | i don't know what to suggest really. can't even pinpoint the problem.. |
| 10:02.12 | _omer | _4d4m_ : yum install gnu (Cannot find a package matching gnu) |
| 10:05.19 | _4d4m_ | _omer: try yum update libstdc++ |
| 10:05.29 | _omer | ok |
| 10:06.04 | _omer | """"libstdc++ is installed and the latest version.""" |
| 10:06.59 | moon06 | Tordah, let's hard reset all :p |
| 10:07.34 | Tordah | yeah. start from scratch xD |
| 10:07.58 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 10:08.59 | _4d4m_ | _omer: has it installed anything, or is it telling you theres nothing to install? |
| 10:09.29 | _omer | no it didnt install anything .. |
| 10:09.30 | _omer | Finding updated packages |
| 10:09.31 | _omer | Downloading needed headers |
| 10:09.31 | _omer | libstdc++ is installed and the latest version. |
| 10:09.31 | _omer | No actions to take |
| 10:10.29 | jm|home | so I'm guessing mpg123 not terming is a known bug :S |
| 10:10.29 | _4d4m_ | _omer: what distro/version are you running? |
| 10:11.00 | moon06 | Tordah, still not working :( |
| 10:11.16 | Tordah | what are your settings, perhaps we should go through them |
| 10:11.16 | moon06 | for sure the problem comes from the machine |
| 10:11.31 | moon06 | which settings ? |
| 10:11.40 | Tordah | well, if you dial out to external can you hear through it? |
| 10:11.50 | moon06 | nope |
| 10:11.50 | _omer | _4d4m_ : RH9 |
| 10:12.18 | Tordah | settings from the phone, i.e. sip settings and network settings |
| 10:13.45 | moon06 | Tordah, actually, they're all the same as my working asterisk installation :( |
| 10:13.57 | _4d4m_ | _omer: I'm guessing your system has the wrong version of libstdc++ libraries. I'm afraid I'm not familiar with festival so cant really help there either |
| 10:14.25 | Tordah | alright then, copy the confs from the working one then |
| 10:14.26 | Tordah | simple:p |
| 10:15.20 | _omer | _4d4m_ : okey....but thanks for your efforts.. |
| 10:15.21 | _omer | :) |
| 10:15.56 | moon06 | Tordah, that's what I did :p |
| 10:16.26 | moon06 | but are these "unable to reach UDP port 5004" normal ? |
| 10:17.56 | moon06 | okay, I go eating |
| 10:18.02 | moon06 | see ya later |
| 10:18.29 | jm|home | hmm |
| 10:18.33 | Synyn | _omer: did you try yum install compat-libstdc* ? |
| 10:18.39 | jm|home | anyone else experiencing multiple instances of mpg123? |
| 10:20.05 | kay2 | someone here uses musiconhold with something else than mpg123 ? |
| 10:20.24 | kay2 | jm|home: mpg123 is not a very good idea |
| 10:20.32 | kay2 | jm|home: better use something else |
| 10:20.37 | *** join/#asterisk ChrisDE4 (n=ChrisDE@88.128.23.181) |
| 10:20.49 | _omer | Synyn : no ..but let me try |
| 10:22.08 | _omer | Cannot find a package matching compat-libstdc++-7.3-2.96.118.i386.rpm |
| 10:22.08 | _omer | No actions to take |
| 10:22.14 | Synyn | what system do yuou use? |
| 10:22.47 | Synyn | _omer: do you know the lib version you are missing? |
| 10:23.48 | _omer | Synyn : RH9 ...and you can check my paste at |
| 10:23.56 | _omer | http://pastebin.ca/92188 to get the version |
| 10:25.03 | Tordah | moon06. it shouldn't be blocked I swear thats a communication port |
| 10:25.11 | Synyn | hmm, try yum install libstdc++.so.6 |
| 10:25.41 | _omer | ok |
| 10:25.48 | jm|home | hmm |
| 10:26.07 | _omer | Cannot find a package matching libstdc++.so.6 |
| 10:26.07 | _omer | No actions to take |
| 10:26.08 | _omer | :( |
| 10:26.11 | *** join/#asterisk pnlarsson (n=niklas@c83-248-2-120.bredband.comhem.se) |
| 10:26.24 | Synyn | _omer: tough crowd there |
| 10:26.33 | *** join/#asterisk qdk (n=qdk@213.237.44.34) |
| 10:27.12 | _omer | should I forget about installing festival ? |
| 10:27.46 | Synyn | _omer: /shrug, I'm new to asterisk :D |
| 10:28.36 | _omer | I am very old in linux but still like a newbie |
| 10:28.41 | Synyn | _omer: you just need to get a copy of libstdc++.so.6 on your system and it will probably work |
| 10:28.42 | FlatFoot | moon86: UDP Port 5004 - RTP Windows Media Services |
| 10:28.52 | FlatFoot | sorry moon06 |
| 10:29.20 | *** join/#asterisk gr0mit (n=w10277@dhcp4.zuk40.mot-tools.co.uk) |
| 10:29.47 | _omer | Synyn : yeah ...but the prob is still there..where do I get it from and where do I copy this file in my system to get festival working ? |
| 10:29.54 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 10:31.01 | Synyn | _omer: where to put it is easy, any $LD_LIBRARY_PATH |
| 10:31.50 | _omer | means ? |
| 10:32.04 | Synyn | _omer: like /usr/lib |
| 10:32.12 | Synyn | _omer: what version of gcc do you have? |
| 10:32.51 | _omer | how to check ? |
| 10:33.34 | backblue | gcc -V? |
| 10:36.01 | *** join/#asterisk ericsmythe (i=eric@82.201.6.100) |
| 10:40.27 | ChrisDE4 | anyone experienced with call-limits now? |
| 10:40.55 | *** join/#asterisk moodperson (n=moodpers@ss13.lb4.ltk.com.ua) |
| 10:41.01 | moodperson | hi asterbots =) |
| 10:49.36 | *** join/#asterisk TeePOG (n=arno@dsl-145-155-145.telkomadsl.co.za) |
| 10:52.17 | TeePOG | hi hi |
| 10:56.12 | kay2 | someone here uses musiconhold with something else than mpg123 ? |
| 10:59.58 | *** join/#asterisk Sonderblade (n=mah@static-213.131.147.169.addr.tdcsong.se) |
| 11:02.43 | Sonderblade | how do you implement dnd in asterisk? Do you do it by adding logic to the extensions.conf file? |
| 11:03.35 | drray | I played with changing contexts |
| 11:03.49 | moon06 | Tordah, what's amazing is that with exactly the same conf in sip.conf, Xlite works great for incoming/outgoing calls |
| 11:04.04 | Tordah | hahaha |
| 11:04.05 | drray | dialing 363 from the phone changed teh context |
| 11:04.27 | Tordah | that's bizzare |
| 11:04.41 | DarKnesS_WolF | the CF syntax is changed on 1.2.x asterisk ?? |
| 11:05.29 | *** join/#asterisk saftsack (n=saftsack@p54A7E68B.dip.t-dialin.net) |
| 11:07.14 | *** join/#asterisk jbsolutios (n=jbenson@193.93.153.1) |
| 11:10.21 | jbsolutios | Hi everyone - any Manager experts around please? |
| 11:10.48 | kay2 | jbsolutios: ?? |
| 11:10.52 | kay2 | jbsolutios: ask ur question |
| 11:11.07 | Sonderblade | jbsolutios: yes |
| 11:11.16 | jbsolutios | in 1.2.9.1 it seems that there is a problem with the manager interface and SIP channels |
| 11:11.27 | kay2 | jbsolutios: what problem |
| 11:11.32 | *** join/#asterisk s0lid (n=s0lid@gr-153-4.eglobalreach.net) |
| 11:14.31 | jbsolutios | it seems that the callerID being passed over is the DID being called not the DID of the caller |
| 11:15.07 | jbsolutios | works fine with IAX |
| 11:16.25 | moon06 | wow, my network cards mac adress is 00:00[...]:00 |
| 11:18.06 | *** join/#asterisk qdk (n=qdk@213.237.44.34) |
| 11:18.21 | jbsolutios | it seems to be with the NewCallerid event I think |
| 11:19.40 | DarKnesS_WolF | greeeeeeee the CFIM is not working ! |
| 11:22.18 | jbsolutios | has anyone else seen this please? |
| 11:23.06 | backblue | jbsolutios: check the line in the code, that sends the callerid in the event |
| 11:23.14 | backblue | and check out what it's append |
| 11:24.24 | Synyn | has anyone used the OEM X100P from DIgitNetworks? good/bad/ugly? |
| 11:24.59 | jbsolutios | backblue - working my through it now |
| 11:26.26 | *** join/#asterisk Greek-Boy (n=grb@193.220.93.162) |
| 11:26.28 | *** join/#asterisk s0lid (n=s0lid@gr-153-4.eglobalreach.net) |
| 11:29.47 | *** part/#asterisk ChrisDE4 (n=ChrisDE@88.128.23.181) |
| 11:32.44 | moon06 | Tordah, you wont believe it |
| 11:32.51 | Tordah | what is it?:d |
| 11:33.10 | moon06 | it *so simply* was the network cards mac address on the asterisk gateway ! |
| 11:33.23 | Tordah | haha |
| 11:33.26 | Tordah | lol:D |
| 11:33.33 | jbsolutios | :D |
| 11:33.41 | Tordah | it always is something so simple that gets you for hours |
| 11:33.43 | moon06 | I simply plugged into the computer another card, this one has a nice mac address, and it works great ! |
| 11:33.44 | Tordah | =/ |
| 11:33.53 | moon06 | yep |
| 11:34.03 | moon06 | now let's begin the configuration of Asterisk :D |
| 11:34.08 | cy3o3 | Hmm |
| 11:34.11 | cy3o3 | Sup guys |
| 11:34.12 | Tordah | i got my thing working too;d iax incoming calls ;p |
| 11:34.27 | moon06 | ;-) Tordah |
| 11:34.27 | Tordah | sip incoming was so much easier=/ |
| 11:34.47 | Tordah | well, smooth sailing from now on =o |
| 11:35.28 | Tordah | well, i havent really helped much! moral support i guess |
| 11:35.28 | Tordah | xD |
| 11:35.34 | moon06 | ^^ |
| 11:36.03 | Tordah | what you gotta setup now? |
| 11:36.18 | Tordah | ive almost finished with my confs =o only gotta work out how to use groups now |
| 11:36.37 | moon06 | one chance here is being alone using my PBX :D |
| 11:36.53 | Tordah | lol |
| 11:36.54 | moon06 | so it'll be much easier than using groups and stuff |
| 11:36.56 | *** join/#asterisk chapeaurouge (n=chapeaur@80.92.83.35) |
| 11:37.00 | Tordah | yeah |
| 11:37.10 | Tordah | im doing it for my dads work |
| 11:37.28 | Tordah | keeps me entertained;) |
| 11:37.57 | moon06 | but I'll have to configure transfer on busy, transfer on unanswered calls -> my cellphone, and maybe moh (not that complicated, already works :p) |
| 11:38.08 | moon06 | and incoming faxes |
| 11:38.18 | saftsack | are there any telephon books based on mysql databases available for *? |
| 11:38.33 | Tordah | ah, that isnt that hard actually though |
| 11:38.43 | saftsack | or it would be possible to do on every number that calls an inverssearch for the name |
| 11:38.50 | Tordah | once you get used to it |
| 11:38.51 | Sonderblade | how does dnd work when you have call groups? |
| 11:38.52 | moon06 | the thing I'm happy with is having my Asterisk working on Gentoo, and doing my own config (I used to have my conf made by AMP) |
| 11:39.08 | Tordah | ah |
| 11:39.35 | Tordah | I've been doing this for just under 2 weeks. I'm a newb:< |
| 11:39.51 | moon06 | u mean Asterisk ? |
| 11:39.53 | Tordah | yeah |
| 11:39.56 | moon06 | oh |
| 11:40.28 | moon06 | I started using it about 1 year ago ... I still remember nights along trying to configure Asterisk :p |
| 11:40.35 | Tordah | haha |
| 11:40.44 | Tordah | it seems addictive! |
| 11:43.21 | moon06 | Tordah, more than that :p |
| 11:44.55 | Tordah | haha |
| 11:45.50 | Tordah | you started off with amp though, i just started with the proper thing=) |
| 11:46.15 | backblue | Synyn: x100p from digitnetworks? show me the url please. |
| 11:46.31 | backblue | i have bought 2 from x100p.com, and none of them work. |
| 11:46.43 | backblue | they dont even are recognized in my pci slots. |
| 11:47.43 | Tordah | well they are under warranty, send em back;p |
| 11:48.21 | backblue | no tks, they are far way from my country, would be shipper by other 2. |
| 11:48.47 | gr0mit | backblue, if you do lspci do you see them in the list? |
| 11:49.00 | gr0mit | they probably show up as tigerjet cards |
| 11:50.06 | backblue | gr0mit: offcourse not, as i said, they dont even are detected in my pci slots. |
| 11:50.10 | backblue | ofcourse |
| 11:50.28 | Synyn | are they 3.3 or 5v? |
| 11:50.54 | Synyn | can anyone do a sanity check on this... |
| 11:51.30 | Synyn | I wanna build a asterisk with fxo card to my PSTN, which is a vonage line, think I'll have any issues with that? |
| 11:52.06 | *** join/#asterisk mmmmmToop (n=mmmmToop@196.26.230.200) |
| 11:52.13 | gr0mit | eeew Synyn |
| 11:52.20 | gr0mit | looks nasty |
| 11:52.37 | Synyn | what? its smooth ) |
| 11:52.48 | gr0mit | lots of 2-4 wires |
| 11:52.49 | backblue | Synyn: that it's what i have to check. |
| 11:52.55 | gr0mit | = lots of ech |
| 11:52.58 | gr0mit | echo |
| 11:53.25 | mmmmmToop | Anyone fighting with bug: 0006626: Queues freeze if AgentCallbackLogin is used ...? |
| 11:53.26 | backblue | i dont know if they are for 3.3 or 5 volts |
| 11:53.31 | gr0mit | better to find a proper ip-telco |
| 11:53.43 | Synyn | gr0mit: what would cause all the echo in that scenario? |
| 11:53.59 | backblue | i dont see any echo in the fxo i have used. |
| 11:54.09 | gr0mit | each time you have a 2-4 wire hybrid you _will_ get echo |
| 11:54.18 | Synyn | I probably will, but I wanna try to use my vonage line as the PTSN so I can dial out while travelling |
| 11:54.37 | Synyn | can't the echo cancellation compensate? |
| 11:54.57 | gr0mit | well the prb with echo can is it adds distortion |
| 11:55.24 | drray | you can turn off echo cancel as well, as that can cause echo.. I've had luck turning the sound down to reduce the annoyance of echo |
| 11:55.37 | Synyn | well, its more of a test / learning thing, so as long as I can get it to make the call, I'm happy ) |
| 11:55.38 | gr0mit | so you will get artifacts from multiple e/c |
| 11:55.58 | gr0mit | if you are just playing around it should be ok |
| 11:56.01 | backblue | i have here a couple of x100p clones, they suck very hard, they dont suport callerid. |
| 11:56.08 | gr0mit | but not for production |
| 11:56.32 | gr0mit | backblue, they should support caller id. which country are you in? |
| 11:56.44 | Synyn | yeah, I read a lot on cards, the clones seem to be bunk, not the real firmware on them, the OEMs are supposed to be ok |
| 11:57.03 | backblue | gr0mit: portugal. they dont suport. |
| 11:57.18 | backblue | i never get them working, and they crash the machines. |
| 11:57.25 | gr0mit | what type of caller id do you use in .pt ? |
| 11:57.30 | drray | ADSI? |
| 11:57.36 | backblue | the chip it's ambient / md3200 |
| 11:57.43 | backblue | gr0mit: clip. |
| 11:57.48 | gr0mit | yes yes |
| 11:58.14 | gr0mit | but do they send caller id after first ring, or after polarity change? |
| 11:58.30 | backblue | how can i know, if the x100p.com card i bought its for 5V or 3.3V? |
| 11:58.39 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 11:58.49 | backblue | gr0mit: i dont know that, i dont have experience in analog lines |
| 11:59.06 | Synyn | backblue: its based on what your MB supports, you should be able to lookup the card and finds its pci voltage |
| 11:59.18 | backblue | i just pickup this clones, and they give me so much mess, i never had time to play with them, but i know they never worked for me, because they dont send callerd. |
| 11:59.21 | backblue | callerid |
| 11:59.36 | Synyn | backblue: if you can see it in devices, it most likely is working at your MB's voltage |
| 11:59.48 | backblue | Synyn: i know it depends on my mboard, but i this x100p does not say anything. |
| 12:00.04 | backblue | Synyn: has i sayed, i not see it in my devices. |
| 12:02.35 | Sonderblade | how do you do to call a ring group with more than one extension in it and one of the extensions has DND set? |
| 12:05.53 | backblue | what it's a DND set? |
| 12:07.22 | Sonderblade | backblue: DND set = extension has Do Not Disturb turned on |
| 12:08.37 | backblue | hoo.. :D |
| 12:08.50 | backblue | so you want to call a extension that have DND set... |
| 12:09.00 | backblue | why you should do that? |
| 12:09.32 | benjk | backblue, its not for 3.3V systems, a friend of mine burned his G5 PCI slot because he tried the card |
| 12:10.16 | backblue | so that's the problem, i was suspecting that. |
| 12:10.25 | Sonderblade | backblue: no.. I have a ring group that i want to call but when Asterisk Dial():s the ring group the extension with dnd set is also called which is wrong |
| 12:10.27 | benjk | besides, that card is so old, it couldnt possibly be 3.3V |
| 12:11.33 | benjk | look at the chip, its an Ambient PCI softmodem, then think about how long ago Ambient disappeared |
| 12:12.38 | [TK]D-Fender | Sonderblade : No it isn't. DND is not something the phone tells the server is "ON". It tells the phone to reject the incoming call as it occurs. Thats how SIP works. |
| 12:13.21 | backblue | benjk: i have ambient cards clones, and they are 3.3, but they dont have callerid suport in my country. |
| 12:13.55 | backblue | benjk: where do i find 3.3V x100p good cards? |
| 12:13.55 | benjk | well, you only have to blame yourself if your mobo burns off |
| 12:13.56 | *** join/#asterisk geoffl (n=geoff@gjctech.plus.com) |
| 12:14.12 | Sonderblade | [TK]D-Fender: i have DND implemented as a bool setting in AstDB so in my case it is something that is handled by asterisk |
| 12:14.20 | benjk | there is no such thing as a good Intel/Ambient softmodem card |
| 12:14.34 | benjk | the chipset is no longer manufactured |
| 12:14.55 | benjk | the Chinese manufacturers are now using refurbished chips to make "new" cards |
| 12:15.33 | benjk | some even use left over chips from earlier production runs which didn't make it through quality control |
| 12:15.55 | [TK]D-Fender | Sonderblade : Congratulations, thats DIALPLAN based DND, but it won't stop the Dial application you call. |
| 12:16.00 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 12:17.14 | Sonderblade | [TK]D-Fender: well obviously :) but I bet someone has solved it and I wanna know how |
| 12:18.00 | [TK]D-Fender | Sonderblade : Solved what? If you put that device into the dial string, its going to dial it, period. |
| 12:19.06 | [TK]D-Fender | Sonderblade : YOU have to check for the AstDB value and CHOOSE not to include the affected device from the dial-string. |
| 12:20.59 | Sonderblade | [TK]D-Fender: so the idea is to take a dail string like: SIP/100&SIP/101&SIP/102&SIP/103 and then filter out those extension in that list that has DND set before you call them? |
| 12:21.13 | *** join/#asterisk IMG-SD (n=IMG-SD@as2.imperialgroup.ca) |
| 12:22.29 | [TK]D-Fender | Sonderblade : or the reverse would probably be better |
| 12:22.37 | Sonderblade | [TK]D-Fender: it is not trivial to write that kind of code in asterisk's extensions.conf file |
| 12:23.06 | lunk | does anyone know how i can set flags for dialing with a .call file? |
| 12:23.23 | lunk | i need to set the tr flag, so the called party can make menu selections |
| 12:23.41 | *** join/#asterisk jcims (n=jcims@cpe-24-210-60-100.columbus.res.rr.com) |
| 12:24.28 | *** join/#asterisk |oranjia| (n=anban@dsl-146-24-225.telkomadsl.co.za) |
| 12:25.55 | [TK]D-Fender | Sonderblade : Fairly easy 20 line macro TOPS. |
| 12:27.23 | Sonderblade | [TK]D-Fender: can i see how you have done it then? :) |
| 12:28.02 | [TK]D-Fender | Sonderblade : I haven't done it on a multiple dial scenario before, just single, but I've already thought up how I'd do it. |
| 12:29.43 | Sonderblade | [TK]D-Fender: we have DND implemented for single dial, it is multiple dial that is the tricky part |
| 12:30.38 | Sonderblade | you need to do a string split, create a list, loop through list and then concatenate the good extensions into a new dial string |
| 12:31.01 | Sonderblade | but afaik, asterisk neither has a string split function or a for loop or an array type or string concatenation :) |
| 12:31.31 | tzafrir_laptop | string split as in Cut? |
| 12:31.54 | tzafrir_laptop | or splitting a string into an array? |
| 12:32.18 | tzafrir_laptop | string concatination in ${VAR1}${VAR2} |
| 12:33.01 | [TK]D-Fender | Sonderblade : easy way out = call a macro with multiple parms, and loop through the parms and concatenate THEM. |
| 12:33.48 | *** part/#asterisk jcims (n=jcims@cpe-24-210-60-100.columbus.res.rr.com) |
| 12:36.34 | Sonderblade | tzafrir_laptop, [TK]D-Fender: Thanks, now I know im on the right track |
| 12:38.10 | *** part/#asterisk moodperson (n=moodpers@ss13.lb4.ltk.com.ua) |
| 12:41.11 | *** join/#asterisk |oranjia| (n=anban@dsl-146-24-225.telkomadsl.co.za) |
| 12:48.57 | *** join/#asterisk murf (n=steve_mu@216.166.159.235) |
| 12:55.34 | *** join/#asterisk beyond (n=beyond@200.192.160.100) |
| 12:56.36 | E-bola | hey guys |
| 12:56.45 | E-bola | just setup my new linksys spa922 |
| 12:56.56 | E-bola | and its allmsot 5 seconds slower dialing out than my softphone is, do anybody know why? |
| 12:57.40 | *** part/#asterisk IMG-SD (n=IMG-SD@as2.imperialgroup.ca) |
| 12:58.05 | E-bola | hmm ignore my question |
| 12:59.51 | [TK]D-Fender | E-bola : ... sorry, did you say something? ;) |
| 13:02.58 | *** join/#asterisk Vec (n=Vector@dsl-146-119-118.telkomadsl.co.za) |
| 13:07.39 | *** join/#asterisk RoyK (n=roy@gprs-ggsn5-nat.mobil.telenor.no) |
| 13:07.42 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 13:08.54 | *** join/#asterisk gandhijee (n=gandhije@mail.win-ent.com) |
| 13:10.17 | RoyK | hi |
| 13:12.10 | *** join/#asterisk boch (n=root@201.216.241.97) |
| 13:12.20 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 13:12.20 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:12.41 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:13.58 | *** join/#asterisk loko (n=rbrown@c-71-199-123-66.hsd1.pa.comcast.net) |
| 13:15.13 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 13:15.13 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:15.25 | boch | is possible to know the maximum channels used in simultaneous in a pri ? |
| 13:15.52 | E-bola | hehe woudl be so cool if u could get the loinksys 922 to use its led as a niterider pulse |
| 13:15.52 | E-bola | hehe |
| 13:15.56 | E-bola | like the car kid |
| 13:16.15 | Synyn | np, know asm? ) |
| 13:16.25 | E-bola | very very little hehe |
| 13:16.36 | Synyn | jmp 14 -> |
| 13:17.39 | Synyn | my claim to faim is I did a cd crack for my windoze game back in the day, I was so proud of myself |
| 13:18.15 | MrChimpy | you did a crack for your own game? |
| 13:18.18 | MrChimpy | clever |
| 13:18.44 | Synyn | friend was trying to teach me how to use assembly, and I hated putting my cd in for rogue spear ) |
| 13:20.08 | MrChimpy | i like knightrider mode on the TE411P |
| 13:21.34 | Tordah | david hasslehoff pwnage |
| 13:22.05 | Synyn | heh, was surprised to see him recently in a movie |
| 13:22.20 | Tordah | what movie was he in? |
| 13:22.32 | Synyn | click |
| 13:22.34 | [TK]D-Fender | Dodgeball :) |
| 13:22.50 | Tordah | ah |
| 13:23.00 | Tordah | when did click come out? |
| 13:23.11 | Synyn | few weeks ago |
| 13:23.15 | E-bola | i need a click to call app that works with internet explore |
| 13:23.17 | Tordah | seen dodgeball that isnt recent! besides he didnt have that big a part, and it was just funny as hell;D |
| 13:23.28 | Tordah | ah kwel, maybe ill see it. whats it about?:p |
| 13:23.32 | Synyn | activex |
| 13:23.37 | Tordah | LIES! |
| 13:23.43 | lunk | chuck norris > david hasslehoff |
| 13:23.46 | Tordah | naw naw |
| 13:23.46 | lunk | who also was in Dodgeball |
| 13:23.55 | Tordah | lance armstrong! |
| 13:24.05 | lunk | 2 balls > 1 |
| 13:24.15 | Synyn | its about adam sandler getting a remote that stops/fastwards time |
| 13:24.24 | Tordah | ROFL |
| 13:24.26 | Tordah | lol |
| 13:24.30 | Tordah | sounds fun. |
| 13:24.30 | |oranjia| | has anyone used "answeronpolarityswitch" in zapata.conf. I am not getting any ringtone if i set it to "yes". When I set it to "no" the call duration is incorrect :( |
| 13:24.49 | Synyn | its funny, and actually kinda tragic too |
| 13:24.55 | Synyn | but a good ending |
| 13:25.23 | Tordah | has anyone seen that movie knightrider? |
| 13:25.25 | Synyn | sandler has come a long way from SNL |
| 13:25.30 | Synyn | but not that far actually |
| 13:25.37 | Tordah | with the crazy pwn car, and kits like zomg! |
| 13:26.01 | Synyn | lmao, how long you been up Tordah? |
| 13:26.26 | Tordah | i think its more of the variable how many hours sleep i had |
| 13:26.27 | Tordah | ;) |
| 13:26.38 | Synyn | knightrider2k is what you mean! |
| 13:26.43 | Tordah | yes! that's it |
| 13:27.17 | Synyn | have to have the reverse missle launchers pop out and blast you far far away |
| 13:27.20 | Tordah | and everyones like, omg the new one rox! and then david's like, no. no. no. |
| 13:27.36 | Tordah | old one had lasers >.< |
| 13:27.41 | Synyn | hehe |
| 13:27.51 | Synyn | and a friendlier eye |
| 13:27.57 | Tordah | anyway more on the right context you know anything about grouping? :p |
| 13:28.08 | Tordah | yeah, and a camper voice |
| 13:28.36 | Synyn | man, I wish my cards would get here already |
| 13:29.09 | Synyn | I ordered them like 7 hours ago... /tapfoot |
| 13:29.17 | Tordah | strong with the force this one is |
| 13:29.53 | Tordah | but patience he must learn |
| 13:31.42 | Synyn | truedat |
| 13:31.54 | [TK]D-Fender | Patience.... yeah yeah... how long is that gonna take?!?! |
| 13:32.01 | Tordah | rofl:D |
| 13:32.29 | Tordah | depends how long it takes for the cards to arrive;) |
| 13:33.29 | nicox | anybody there who tested chan_ss7 |
| 13:33.34 | *** join/#asterisk jm|home (n=jamiem@dsl-217-155-242-137.zen.co.uk) |
| 13:33.39 | Synyn | lol |
| 13:34.09 | Qwell | [TK]D-Fender: boot to the head |
| 13:34.12 | Tordah | anyone help me with grouping? cant find it in the Orly? Book |
| 13:34.23 | jm|home | hello |
| 13:34.36 | Synyn | buenas dias |
| 13:34.53 | jm|home | someone test my URI inbound for me? |
| 13:35.01 | Synyn | sure |
| 13:35.22 | jbalcomb | i can test it with my nessus server. |
| 13:35.31 | jm|home | Synyn: ok to pm? |
| 13:35.44 | E-bola | anybody uses click to call |
| 13:35.46 | E-bola | in windows? |
| 13:37.21 | E-bola | snap crashes :( with some font option |
| 13:37.58 | jm|home | hm |
| 13:38.01 | jm|home | I broke something |
| 13:38.10 | Tordah | snap |
| 13:38.31 | *** join/#asterisk Mike (n=mike@201.112.50.158) |
| 13:40.21 | Sonderblade | how do you use a macro that takes a variable number of arguments? |
| 13:41.00 | [TK]D-Fender | Qwell : *shooomp*! |
| 13:41.47 | Tordah | Bush: *fwap*! |
| 13:42.19 | *** part/#asterisk loko (n=rbrown@c-71-199-123-66.hsd1.pa.comcast.net) |
| 13:44.37 | *** join/#asterisk delmar (n=delmar@ip-58-28-158-154.ubs-dsl.xnet.co.nz) |
| 13:44.58 | *** join/#asterisk s0lid (n=s0lid@202.73.164.125) |
| 13:46.37 | [TK]D-Fender | Sonderblade : ALL macro's take a variable # of args. |
| 13:47.03 | [TK]D-Fender | Sonderblade : loop through them till you hit a blank. |
| 13:47.54 | Vec | How can I output a variable to the CLI for debugging, like an echo $var ? |
| 13:48.12 | docelmo | NoOP( |
| 13:48.14 | delmar | Hi everyone. I am having trouble with a TDM400P card I just received. lspci shows it as 0000:02:04.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface. cat /proc/pci also shows it there, and it has IRQ10 and nothing else is on IRQ10. when i load wctdm I see it find the card, module 0 and 1 not installed, 2 and 3 installed Auto FXO etc. ztcfg -vv outputs two channels 01 and 02 both FXS Kewlstart etc. |
| 13:48.14 | delmar | 2 channels configured... |
| 13:48.17 | docelmo | err NoOp() |
| 13:48.23 | Vec | k thanks |
| 13:48.27 | delmar | but.. asterisk will not work at all |
| 13:49.19 | docelmo | Start asterisk with -vvvvvvvc and see what the problem is |
| 13:49.30 | delmar | also, when i try to use fxotune ie fxotune -i 2 to at least tune the FXO's and create the file /etc/fxotune.conf it doesnt work |
| 13:49.39 | delmar | something is in fact wrong with the Zap channels |
| 13:49.43 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 13:49.48 | PakiPenguin | noon |
| 13:50.24 | *** join/#asterisk vgster (n=vgster@217.78.147.194) |
| 13:50.56 | delmar | when i do ./fxotune -i 2 i get output like.. /dev/zap/1 absent: No such device .. same for /dev/zap2 but for 3 onwards its /dev/zap/3 absent: No such device or address |
| 13:51.49 | docelmo | haha |
| 13:51.50 | delmar | So, as far as loading the modules and ztcfg is concerned.. the FXO modules are there... when I try to use fxotune, or run Asterisk.. it fails |
| 13:51.53 | docelmo | your using CentOS right? |
| 13:51.54 | Tordah | how can I set up grouping? |
| 13:52.28 | delmar | docelmo, who are u talkin to? |
| 13:52.33 | docelmo | you |
| 13:52.40 | *** join/#asterisk RoyK (n=roy@gprs-ggsn6-nat.mobil.telenor.no) |
| 13:52.47 | delmar | docelmo, no. Im running debian sid, kernel 2.6.10 |
| 13:52.51 | _MDC_ | ok, so now I've figured out why I get the "(Not enough bandwidth)" when trying to make a call to a H323 gk via oh323. The problem is that its not registrated correctly, asterisk/oh323 says it is, but only when using the gk's global h323 not my ordinary user account. OK, so short question; is the only thing I have to do in oh323.conf is to set alias to my h323 username and set gatekeeper to the gk ip? Obiovisly not, so |
| 13:52.51 | _MDC_ | what have I missed? |
| 13:53.17 | delmar | docelmo, but what was your idea anyway... |
| 13:53.24 | docelmo | delmar well shit.. that killed my idea.. I would check /dev/zap and make sure it exists |
| 13:53.38 | docelmo | if not there is a known centos bug that prevents them from being created |
| 13:53.40 | delmar | docelmo, the device is there in /dev |
| 13:53.42 | docelmo | I had the same problem |
| 13:53.47 | docelmo | hmmm |
| 13:53.54 | *** join/#asterisk Hmmhesays (n=Neg@24-117-135-28.cpe.cableone.net) |
| 13:54.07 | docelmo | I would say call Digium directly as you get free tech support from the masters for the cards |
| 13:54.09 | Sonderblade | [TK]D-Fender: how do i loop through them? |
| 13:54.11 | delmar | docelmo, for example.. under /dev/zap there is.. crw-r--r-- 1 root root 196, 1 Jul 19 20:00 1 |
| 13:54.22 | *** join/#asterisk sevard (n=sev@adsl-71-129-115-244.dsl.irvnca.pacbell.net) |
| 13:54.23 | [TK]D-Fender | Sonderblade : GotoIf |
| 13:54.34 | docelmo | if you go into /dev/zap does 1,2,3,etc exist? |
| 13:54.38 | sevard | How would one set ' |
| 13:54.46 | sevard | How would one set 'tos' without root privs? |
| 13:54.46 | *** join/#asterisk saftsack (n=saftsack@p54A7DA3A.dip.t-dialin.net) |
| 13:55.16 | Hmmhesays | Bah |
| 13:55.30 | delmar | docelmo, yeah .. for example.. in /dev/zap there is crw-r--r-- 1 root root 196, 1 Jul 19 20:00 1 |
| 13:55.30 | delmar | <PROTECTED> |
| 13:55.33 | Sonderblade | [TK]D-Fender: no i meant how to refer to the n:th argument? |
| 13:55.35 | *** join/#asterisk pdtmobile (n=ptinsley@c-68-53-40-50.hsd1.tn.comcast.net) |
| 13:56.08 | [TK]D-Fender | Sonderblade : ${ARG${count}} |
| 13:56.43 | Sonderblade | [TK]D-Fender: thanks i didn't know that was possible |
| 13:57.04 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 13:57.54 | *** join/#asterisk bkw_ (n=bkw_@asterisk/friend-and-developer/bkw) |
| 14:00.40 | *** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-42-234.losaca.adelphia.net) |
| 14:03.24 | Hmmhesays | bah, I hate cisco's documentation |
| 14:04.23 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 14:05.52 | Hmmhesays | I might be willing to pay someone if they can help me with this |
| 14:06.18 | sevard | Hmmhesays: see a doctor, use a comb and the ointment. |
| 14:06.30 | sevard | pay up. |
| 14:06.30 | Hmmhesays | sevard, funny |
| 14:06.40 | Synyn | lmao |
| 14:06.45 | sevard | I need 30.00 via paypal |
| 14:06.50 | Hmmhesays | haha |
| 14:06.54 | sevard | give it back. |
| 14:06.56 | Hmmhesays | fark you |
| 14:07.02 | Synyn | For his crack and hooker |
| 14:07.06 | Hmmhesays | it was worth it |
| 14:07.08 | sevard | bieach you still haven't got me a sixer |
| 14:07.09 | jm|home | er .... |
| 14:07.14 | Hmmhesays | I gave you some kickass dp shiat for free |
| 14:07.19 | jm|home | I can use an FXO card to pickup calls from the PSTN, right? |
| 14:07.28 | sevard | fuck that nigs |
| 14:07.36 | sevard | actually that hang up when called shit was tizzite |
| 14:07.37 | Synyn | correct |
| 14:07.43 | Hmmhesays | jm|home: I'll tell you if you configure my as5300 |
| 14:07.48 | sevard | what was a pretty neat trick |
| 14:07.57 | sevard | Hmmhesays: how can you set tos without root privs? |
| 14:08.16 | Hmmhesays | what a fantastically vague question |
| 14:08.26 | sevard | in asterisk poopyface. |
| 14:08.42 | sevard | if you run * as a nonroot user and try to tos your rtp streams tis not allowed. |
| 14:09.03 | sevard | in sip.conf |
| 14:09.04 | Hmmhesays | why not? |
| 14:09.28 | sevard | Jul 19 08:52:06 WARNING[21832]: rtp.c:1017 ast_rtp_settos: Unable to set TOS to 184 |
| 14:09.28 | Hmmhesays | what does it tell you? |
| 14:09.40 | sevard | I really don't want to go messing with iptables. |
| 14:10.02 | sevard | totally not comfortable doing on that on anything but a linksys running openwrt |
| 14:10.02 | Hmmhesays | what is that calling that needs root privileges |
| 14:10.06 | *** join/#asterisk FlyboySR22 (n=rsears@gateway.americanis.net) |
| 14:10.55 | Hmmhesays | iptables on a linksys is the same as iptables on a linux box |
| 14:11.51 | Sonderblade | how do you remove the last character of a string? |
| 14:12.22 | sevard | as far as i know you can't set tos in usermode |
| 14:12.30 | sevard | you'll need to give privs to something. |
| 14:12.32 | sevard | and i can't figure out what |
| 14:12.59 | Hmmhesays | look at rtp.c |
| 14:13.32 | *** join/#asterisk Precion (n=crhodes@adsl-75-7-75-29.dsl.milwwi.sbcglobal.net) |
| 14:14.08 | Hmmhesays | figure it out |
| 14:14.29 | *** join/#asterisk af_ (n=af@ip-170-209.sn1.eutelia.it) |
| 14:14.58 | *** join/#asterisk ambriento (n=melcon@200.192.160.100) |
| 14:15.08 | Synyn | Hmmhesays: could you just setup a sudo or root sticky bit? |
| 14:15.50 | *** join/#asterisk tuxd00d (n=tuxinato@netblock-68-183-136-97.dslextreme.com) |
| 14:16.59 | sevard | my girlfriend's dad was telling me about his old office job in a school doing books and you could smoke in the building so he'd smoke and drink tar all day, the secretary would do the same and all the girls in the next office smoked and drank coffee all day |
| 14:17.03 | sevard | what an awesome office. |
| 14:17.26 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
| 14:18.34 | jbalcomb | Who is using call Queues in Asterisk? Do you play an announcement before you send the call to the queue? How do you handle calls to the queue after hours? How do you handle calls to the queue during business hours when the queue has no members? If a member is available do they get the call directly without an announcement or hold music? |
| 14:18.52 | sevard | so many questions |
| 14:18.54 | *** join/#asterisk ariel_ (n=ariel_@74.8.35.2) |
| 14:19.16 | sevard | jbalcomb: lemme show you my company_x plan |
| 14:19.30 | *** join/#asterisk trbldwine (n=trbldwin@adam.ur.northwestern.edu) |
| 14:19.48 | Hmmhesays | got a wip-300 coming in the mail |
| 14:20.11 | file | Hmmhesays: tell me how it works... |
| 14:20.27 | *** join/#asterisk Ludo_ (n=Ludo@obelix.zoxx.net) |
| 14:20.30 | Ludo_ | hi |
| 14:20.42 | Ludo_ | some people already try to install web-meetme ? |
| 14:20.47 | Hmmhesays | file: prelim tests from my partner in crime say it works quite well |
| 14:21.14 | file | Hmmhesays: neat, I think it's an interesting phone considering it runs Linux and the source is available for some of it :) enough that you can build your own image |
| 14:21.20 | jbalcomb | Hmmhesays: your dog uses a phone? |
| 14:21.36 | *** join/#asterisk smackus (n=ckwall@63.149.122.93) |
| 14:21.38 | Hmmhesays | does asterlink offer did's othere than 800 numbers? |
| 14:21.44 | Hmmhesays | I was poking around the website last night |
| 14:21.49 | file | dunno |
| 14:22.04 | file | poke MikeJ[Laptop] or bkw |
| 14:22.05 | Hmmhesays | jbalcomb: huh? |
| 14:22.10 | sevard | jbalcomb: show answer most of your questions, highly commented, http://pastebin.ca/92354 |
| 14:22.16 | Hmmhesays | you still work for them? |
| 14:22.23 | file | no I do not |
| 14:22.32 | Hmmhesays | I see says the blind matty |
| 14:22.37 | smackus | I am trying to figure out how to make an autoattendant extension repeat once then hang up. The only thing I can figure out is the last priority sends to the first priority, but I do not want to send an extension into a loop. Is there a better way? |
| 14:22.52 | file | smackus: have a variable that increments and check it |
| 14:23.05 | *** join/#asterisk zamolxes (n=zamolxes@83.166.220.142) |
| 14:23.10 | smackus | are you willing to teach me how to do that? |
| 14:23.20 | file | you have everything you need |
| 14:23.36 | Hmmhesays | file: ever configure an as5300? |
| 14:23.42 | file | Hmmhesays: no |
| 14:23.48 | file | Hmmhesays: and even if I had, I wouldn't admit it |
| 14:24.01 | Hmmhesays | I got one sitting in somalia right now being a paperweight |
| 14:24.13 | smackus | here is my current extension that I need to repeat: http://pastebin.ca/92356 |
| 14:24.31 | jbalcomb | Hmmhesays: have you asked in #cisco? super helpful peeps over there. |
| 14:24.51 | jbalcomb | sevard: looks good. thanks. i'll go through it now. |
| 14:24.51 | file | smackus: all the tools you need to do this are in Asterisk, you just need to THINK about what it requires, and look and learn |
| 14:25.02 | Hmmhesays | yeah a few times |
| 14:25.09 | *** join/#asterisk tamp4x (n=tampon@www.vonworldwide.com) |
| 14:25.13 | smackus | understood... just don't even know where to begin to look |
| 14:25.19 | file | well think |
| 14:25.30 | tamp4x | is there something special needed to recognize e.164 numbers in asterisk |
| 14:25.33 | Hmmhesays | although I'm a cisco n00b so my questions are probably pretty stupid sounding |
| 14:25.40 | file | you need a variable that starts out at 0, you need to increment it when the timeout happens, you need a check on the value |
| 14:25.53 | jbalcomb | make variable, write variable, check variable increment variable, check variable, break? |
| 14:25.56 | tamp4x | keeps on looking for exten 's' |
| 14:26.14 | jbalcomb | Hmmhesays: what seems to be the trouble? |
| 14:26.53 | jbalcomb | woot!! all that and i'm the biggest hack coder i've ever worked with. ;) |
| 14:26.56 | Hmmhesays | jbalcomb: i have an as5300 with isdn e1's that I need to configure to terminate traffic from IP to Pots |
| 14:27.30 | Hmmhesays | I got the e1 up, I think, did the h323 gateway stuff |
| 14:27.31 | jbalcomb | Hmmhesays: hrmm.. that sounds like no small task. |
| 14:27.58 | Hmmhesays | configured a dial-peer, but the cisco documentation is confusing |
| 14:28.57 | tamp4x | what variable other than ${EXTEN} can be used to find the extensions ( the TO: number) |
| 14:29.32 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 14:30.23 | zamolxes | hi. My setup should allow users to prepay $4 or something , and then talk for $4. The system should hang up when the credit is consumed but play a sound file 1 minute before that. Everything interface-wise I wrote in perl with a postgres backend. I don't really know how to tackle the hanging up part . Some pointers on what should I be reading next? |
| 14:31.04 | *** join/#asterisk CoderCR (n=creyna@ip68-6-237-193.sd.sd.cox.net) |
| 14:31.09 | *** part/#asterisk CoderCR (n=creyna@ip68-6-237-193.sd.sd.cox.net) |
| 14:34.11 | Vec | Can you dial a name, for an extention rather than a number, or is the correct way to dial a number, I am using a soft phone ? |
| 14:34.26 | zamolxes | uhm, should I rephrase? :) |
| 14:34.28 | jbalcomb | sevard: i thought the timeout option in the queue function was deprecated in 1.2.4? |
| 14:35.06 | Synyn | zam: h = hangup |
| 14:36.09 | zamolxes | Synyn: ok, and how can I watch the call constantly and hangup when her credit is exhausted? My problem is I don't really know where to begin :) |
| 14:36.43 | Hmmhesays | zamolxes: set the absolute timeout before the call is connected |
| 14:37.18 | Synyn | zam: ah, you are trying to catch a event or poll for credits and then launch an event? |
| 14:37.37 | tamp4x | http://pastebin.ca/92372 <- anyone know why this is happening, i have +12037743993 as an exten in the dialplan, wtf is it coming up in the domain? |
| 14:38.55 | zamolxes | Synyn: basically when the user dials, I want to get the phone number, lookup the prefix+credits in postgres, decide how many minutes does the user has depending on that, and tell asterisk to hangup after N minutes and play after N-1 minutes |
| 14:41.24 | *** join/#asterisk nortex (n=breeves@snapper.titanspecialties.com) |
| 14:41.36 | angler | zamolxes, absolutehangup |
| 14:41.40 | zamolxes | ok |
| 14:42.38 | jbalcomb | sevard: exten = s,5,Goto(company_x-closed,s,1) means that if they don't get into the queue because its full, nonexistent, or there are no members they go to company closed? |
| 14:43.10 | angler | zamolxes, absolutetimeout actually, which looks like its not in trunk anymore |
| 14:43.27 | file | Set(TIMEOUT(absolute)=120) |
| 14:43.39 | angler | file, thanks |
| 14:43.41 | *** part/#asterisk Ludo_ (n=Ludo@obelix.zoxx.net) |
| 14:44.37 | zamolxes | thanks, i'll go hit the docs about all these |
| 14:44.38 | angler | zamolxes, also the S flag on Dial |
| 14:45.15 | angler | :) |
| 14:50.24 | jbalcomb | ~seen [TK]D-Fender |
| 14:50.31 | jbot | [tk]d-fender is currently on #asterisk (2h 51m 52s). Has said a total of 16 messages. Is idling for 54m 23s, last said: 'Sonderblade : ${ARG${count}}'. |
| 14:51.05 | X-Gen | ~seen jbot |
| 14:51.07 | jbot | jbot is currently on #asterisk-doc (15h 4m 59s) #ubuntu-utah (15h 4m 59s) ##t42 (15h 4m 59s) #how (15h 4m 59s) #ol (15h 4m 59s) #flyspray (15h 4m 59s) #asterisk (15h 4m 59s) #byumug (15h 4m 59s) #va (15h 4m 59s) #orkut (15h 4m 59s) #nslu2-linux (15h 4m 59s) ##ducleague (15h 4m 59s) #storm ... |
| 14:52.33 | jbalcomb | What does this 'i' do? exten = i,1,Goto |
| 14:52.49 | smackus | file: ok, so here is what I have been trying prior to asking the channel: http://pastebin.ca/92390 But I do not think that the syntax is right, because it still hangs up on me at the end of the auto attendant recording. I cannot find any asterisk specific documentation on how to do this. I have pieced this together from other boards I have read. |
| 14:52.49 | drray | invalid |
| 14:53.07 | jbalcomb | Is that just when they press a key that doesn't match anything we account for? |
| 14:53.13 | drray | jes |
| 14:53.26 | jbalcomb | ah, ok. danke |
| 14:53.46 | *** join/#asterisk Cresl1n (n=matt@gateway.digium.com) |
| 14:53.46 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 14:53.52 | nicox | anybody there who tested chan_ss7 |
| 14:54.23 | angler | smackus, use WaitExten after Background |
| 14:54.31 | [TK]D-Fender | jbalcomb : here |
| 14:55.04 | [TK]D-Fender | jbalcomb : link me to what you're looking at |
| 14:55.09 | angler | smackus, or set autofallthrough=no in general section of extensions.conf |
| 14:56.12 | smackus | angler: what does autofallthrough=no do? |
| 14:56.49 | angler | old behavior of asterisk where it waits for an extension to be entered if there is not another priority |
| 14:56.54 | jbalcomb | [TK]D-Fender: the uppers want to change the way our queues function. |
| 14:57.33 | gr0mit | nicox : yes, we are using chan_ss7 |
| 14:57.57 | *** join/#asterisk [Airwolf] (n=airwolf@dsl51B6AEE5.pool.t-online.hu) |
| 14:59.48 | jbalcomb | [TK]D-Fender http://pastebin.ca/92354 |
| 15:00.10 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 15:00.36 | Toerkeium | hello guys, anyone know a good SIP provider? |
| 15:02.03 | *** join/#asterisk stadanko (n=bill@mail.southerncarehospice.com) |
| 15:03.51 | nicox | good sip provider is www.platinplus.com |
| 15:04.01 | Toerkeium | thanks nicox |
| 15:05.21 | pdtmobile | I see new sip providers just about every time somebody asks that question, at some point I am going to see the same one twice... i just know it |
| 15:05.50 | pdtmobile | law of averages and all |
| 15:06.55 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
| 15:07.00 | [TK]D-Fender | jbalcomb : ok, so whats your question on this PB? |
| 15:07.16 | *** join/#asterisk Dr-Linux|work (n=Linux@202.125.139.198) |
| 15:09.52 | jbalcomb | [TK]D-Fender How can I make my queues work the way I'm being asked to do? |
| 15:11.37 | [TK]D-Fender | jbalcomb : ... I lost track of your requirements... |
| 15:11.56 | jbalcomb | [TK]D-Fender: I think I'll have to find out why we aren't using GotoIfTime and why our queues are memberless during business hours. |
| 15:12.34 | jbalcomb | [TK]D-Fender: they want me to do something with the caller when they can't join the queue. |
| 15:12.39 | [TK]D-Fender | jbalcomb : memberless during business hours is important. working hours is also important, but I keep manual on/off hour overrides in my setups |
| 15:12.44 | *** join/#asterisk dhill (i=dhill@fog.mindcry.org) |
| 15:12.55 | *** join/#asterisk RoyK (n=roy@gprs-ggsn6-nat.mobil.telenor.no) |
| 15:12.59 | dhill | does libpri allow me to use _any_ t1 card? |
| 15:13.02 | dhill | or just zaptel? |
| 15:13.09 | [TK]D-Fender | jbalcomb : They should ALWAYS be able to join the queue if it is considered "open for buisiness" |
| 15:13.39 | *** join/#asterisk marv[work] (n=timr@64.89.118.139) |
| 15:13.42 | [TK]D-Fender | dhill : just say what model you are looking at... |
| 15:14.01 | jbalcomb | [TK]D-Fender: I would think so as well. So if I turn off nojoin and leaveempty, what do i do with a call on hold for three minutes while someones on break? |
| 15:14.38 | dhill | I was looking at Sangoma and accoom (www.accoom.net) |
| 15:14.42 | [TK]D-Fender | jbalcomb : let them sit around, or have a queue timeout. if you let them sit, ALLOW them to quit to VM at their discretion. |
| 15:15.04 | [TK]D-Fender | dhill : Can't speak for the latter, but Sangoma uses ZAptel/libpri. |
| 15:15.15 | dhill | i will be using OpenBSD |
| 15:15.28 | *** join/#asterisk umay (n=chris@71-208-175-55.hlrn.qwest.net) |
| 15:15.43 | [TK]D-Fender | dhill : I wouldn't touch that other card with a 10ft pole... |
| 15:16.05 | dhill | openbsd has the san driver |
| 15:16.10 | jbalcomb | [TK]D-Fender: That's what I think. I wish I knew why these guys think theres a difference between someone sitting on hold because all members are unavailable and sitting on hold because there are no members. |
| 15:17.40 | [TK]D-Fender | jbalcomb : there IS. "Leaveempty" would kick them out if there are no agents LOGGED. If there are logged (but busy) agents then they would sit and wait |
| 15:17.40 | *** join/#asterisk pbx1 (n=pbx1@58.69.92.3) |
| 15:17.41 | *** part/#asterisk sevard (n=sev@adsl-71-129-115-244.dsl.irvnca.pacbell.net) |
| 15:18.38 | jbalcomb | [TK]D-Fender: i mean why they are willing to let someone sit for unavail but not unmanned. |
| 15:18.52 | jbalcomb | [TK]D-Fender from a customer services perspective |
| 15:19.17 | [TK]D-Fender | jbalcomb : unmanned has no future in sight, unavail can end their call. |
| 15:19.56 | dhill | [TK]D-Fender: I am just trying to figure out if I can get Asterisk to work on OpenBSD using the sangoma driver and libpri. I do not want to use the wanpipe driver. |
| 15:20.00 | [TK]D-Fender | jbalcomb : Unmanned shouldn't happen during bunsiness hours and staffing+policy should enforce that. with taht in mind you should always leave you caller in queu until THEY decide to leave. |
| 15:20.13 | [TK]D-Fender | dhill : You require Wanpipe. |
| 15:20.31 | jbalcomb | [TK]D-Fender: ... yes, that is where I was leaning. |
| 15:20.58 | dhill | how so? |
| 15:21.11 | *** join/#asterisk eKo1 (n=eKo1@190.4.7.90) |
| 15:21.41 | [TK]D-Fender | dhill : Sangoma runs off their driver which interfaces with Zaptel. ZAptel is second-banana in their setup |
| 15:22.04 | [TK]D-Fender | dhill : Whats the problem with it? |
| 15:22.31 | dhill | Is it an open or closed source driver? |
| 15:22.44 | jbalcomb | [TK]D-Fender: the way i see it there are four situations i need to account for in my queue structure... |
| 15:22.57 | [TK]D-Fender | dhill : Open. So again, whats the problem? |
| 15:23.22 | [TK]D-Fender | jbalcomb : 1/2 of them no doubt require reprogramming app_queue |
| 15:23.40 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 15:23.54 | RoyK | tag |
| 15:24.16 | jbalcomb | [TK]D-Fender: (1) 1+ members available; (2) 0 members avaialble; (3) no members, full, or nonexistant; (4) after hours |
| 15:24.57 | dhill | I am wondering why wanpipe is not in OpenBSD then. |
| 15:25.05 | dhill | perhaps it GPL'd |
| 15:25.13 | jbalcomb | because openBSD is awesome? |
| 15:25.41 | [TK]D-Fender | jbalcomb : after hours = dial plan. No members = joinempty/leaveempty. 1+ Avail (as in just ring, no MoH which I suspect is what you're thinking) = No-go. |
| 15:25.56 | [TK]D-Fender | dhill : What do you mean "not in OpenBSD"? |
| 15:26.08 | dhill | as the driver for sangoma cards |
| 15:26.17 | *** join/#asterisk mog_home (n=mogorman@68.62.237.103) |
| 15:26.17 | *** mode/#asterisk [+o mog_home] by ChanServ |
| 15:26.18 | [TK]D-Fender | dhill : Clarify... |
| 15:26.38 | nicox | do anybody know if is it possible to get 2 asterisk boxes working with only 1 SS7 link under chan_ss7 |
| 15:26.56 | [TK]D-Fender | jbalcomb : the intermittant queue message just plays on interval regardless of anything and won't trigger instantly if they are all busy. |
| 15:27.05 | dhill | alex from sangoma told me the san driver is old.. and that I should be using wanpipe |
| 15:27.09 | jbalcomb | [TK]D-Fender: ok, so after hours is handled by GotoIfTime and/or our DND check |
| 15:27.11 | dhill | http://www.openbsd.org/cgi-bin/man.cgi?query=san&apropos=0&sektion=4&manpath=OpenBSD+Current&arch=i386&format=html |
| 15:27.49 | dhill | so i am wondering what wanpipe has that the OpenBSD driver does not |
| 15:27.54 | jbalcomb | [TK]D-Fender: maybe we can put calls in the queue even if its unmanned... |
| 15:28.01 | [TK]D-Fender | jbalcomb : Don't trust phone-DND, use an AstDB to force it open/closed/normal. |
| 15:28.09 | *** join/#asterisk Kernel_core (i=Kernel_C@217.218.80.242) |
| 15:28.18 | [TK]D-Fender | dhill : Check their wiki, and when you're done with that, PHONE them. |
| 15:28.25 | jbalcomb | [TK]D-Fender: yeah, i'm using AstDB for that |
| 15:28.27 | [TK]D-Fender | jbalcomb : You certainly CAN. |
| 15:28.56 | [TK]D-Fender | jbalcomb : I use STD hours + manual overrides on all my setups. |
| 15:29.10 | eKo1 | nicox: What do you mean? |
| 15:29.22 | mut | stds are nasty |
| 15:29.43 | jbroome | it burns when i call! |
| 15:29.47 | [TK]D-Fender | mut : LIFE is a sexually transimtted disease which is in all cases FATAL. |
| 15:29.49 | *** join/#asterisk mtaht4 (n=m@reserve-64-79-114-30.wiline.com) |
| 15:29.52 | eKo1 | nicox: you should have one * box as the ss7 gateway and whatever number of * boxen sending calls to it. |
| 15:30.01 | mut | ah yes |
| 15:30.04 | mut | BUT |
| 15:30.05 | *** join/#asterisk unixgeek (n=unixgeek@216-220-234-197.exploremaine.com) |
| 15:30.10 | *** join/#asterisk hi365 (n=hi365@bzq-167-158.dsl.bezeqint.net) |
| 15:30.16 | mut | i'm immortal |
| 15:30.23 | jbalcomb | [TK]D-Fender: so now, if [0 members avaialble == no members] then i need a welcome msg, bail msg, and hold music ... |
| 15:30.49 | [TK]D-Fender | jbalcomb : 0 avail != 0 members. |
| 15:31.12 | jbalcomb | [TK]D-Fender: if i'm putting them both in the queue why wouldn't they be? |
| 15:31.19 | [TK]D-Fender | jbalcomb : and you can't really test that easily before entering the queue. |
| 15:31.26 | Synyn | Anyone used the DigitNetworks x100P cards? http://www.digitnetworks.com/store/product_info.php?cPath=22&products_id=28&osCsid=76eac566b0c64adb51d4e36ac91b140e |
| 15:31.51 | [TK]D-Fender | jbalcomb : they are different statuses. only diff is if you kick when no members logged. |
| 15:32.16 | jbalcomb | [TK]D-Fender if i was gonna kick why would i let them join? |
| 15:32.39 | [TK]D-Fender | jbalcomb : You might want to kick when the agents leave once they MADE it in. |
| 15:33.30 | jbalcomb | [TK]D-Fender: ok, so if leavewhenempty kicks in or the TIMEOUT runs out, doesn't it just drop to the next priority? |
| 15:33.40 | *** join/#asterisk n9urk (n=leonard@user-0ce2dhc.cable.mindspring.com) |
| 15:33.49 | [TK]D-Fender | jbalcomb : timeout is a seperate event unto itself. |
| 15:33.58 | *** join/#asterisk trbldwine (n=trbldwin@adam.ur.northwestern.edu) |
| 15:33.58 | *** part/#asterisk santiago (i=santiago@debian/developer/santiago) |
| 15:34.08 | [TK]D-Fender | jbalcomb : and all 3 return different queuexit values |
| 15:35.17 | jbalcomb | [TK]D-Fender: hrmm.. i though 0 and -1 were the only two values |
| 15:35.17 | [TK]D-Fender | jbalcomb : there is a channel var set on queue exit... |
| 15:35.47 | n9urk | hi all, * 1.2.5 is not recording the cdr data. I have not changed any settings to make it stop logging but nevertheless it does not record. Is there some setting in 1.2.5 that turns off cdr by default? |
| 15:35.50 | jbalcomb | [TK]D-Fender ok, so if they have different values, i can check the values in a GotoIf? |
| 15:36.19 | [TK]D-Fender | jbalcomb : naturally. |
| 15:36.22 | jbalcomb | n9urk i upgraded from 1.2.1 to 1.2.5 last week and nothing broke |
| 15:36.33 | Hmmhesays | SEvard |
| 15:36.34 | n9urk | jbalcomb: Thanks. |
| 15:36.37 | Hmmhesays | you one a windoze box? |
| 15:37.02 | jbalcomb | s/one/SAY WHAT!!/ |
| 15:37.11 | n9urk | jbalcomb: is there something obvious that I may have unintentionally did that killed the cdr logging? |
| 15:38.07 | *** join/#asterisk Koshatul (n=evangeli@ip157-65-132.cust.bit.net.au) |
| 15:38.26 | eKo1 | n9urk: What module(s) are you using for cdr logging? |
| 15:38.51 | n9urk | cdr-csv |
| 15:39.10 | jbalcomb | n9urk i'm not familiar with it enough to say. sorry. i would certain check the wiki for cdr logging though. |
| 15:39.49 | n9urk | jbalcomb: thanks anyawy! |
| 15:39.57 | eKo1 | n9urk: is the cdr_csv module loaded? |
| 15:40.35 | n9urk | eKo1: good question. What is the best way to know for sure? It seems that my cdr data ends approx. the time I upgraded to 1.2.5 |
| 15:40.58 | RoyK | perhaps 1.2.10 |
| 15:41.00 | eKo1 | In the Asterisk CLI, enter: load cdr_csv.so |
| 15:41.33 | n9urk | eKo1: it says it already exists |
| 15:41.42 | jbalcomb | n9urk you might do show modules to see it |
| 15:41.43 | eKo1 | Good, then it is loaded. |
| 15:41.56 | jbalcomb | n9urk and you might try unloading it and then loading it |
| 15:42.03 | eKo1 | I don't like show modules because it floods your screen. |
| 15:42.08 | jbalcomb | n9urk i have to do that to my voicemail module sometimes |
| 15:42.19 | hi365 | deos anyone know how to uninstall the Sangoma wanrouter drivers? they are detecting non existent cards and drining my system nuts! |
| 15:42.19 | hi365 | http://pastebin.ca/92446 |
| 15:42.30 | *** join/#asterisk saftsack (n=saftsack@p54A7FB9A.dip.t-dialin.net) |
| 15:42.31 | jbalcomb | eKo1: `asterisk -rx "show modules" | grep <module>` |
| 15:42.33 | *** join/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com) |
| 15:42.37 | n9urk | jbalcomb: do I need to restart * afterwards? |
| 15:42.42 | jbalcomb | n9urk nope |
| 15:42.45 | wunderkin | show modules like? |
| 15:43.04 | jbalcomb | yes, modules like show |
| 15:43.11 | eKo1 | jbalcomb: yeah, but you have to exit the cli for that |
| 15:43.32 | pdtmobile | hi365: you could just remove your wanpipe1.conf or not run wanrouter |
| 15:43.48 | jbalcomb | eKo1: i don't spend much time in the CLI realy |
| 15:44.22 | eKo1 | n9urk: check your cdr.conf to see if anything has changed. |
| 15:44.29 | eKo1 | n9urk: check your log files too |
| 15:44.32 | n9urk | jbalcomb: after reloading the mod it gave me an error message on permissions |
| 15:44.37 | jbalcomb | eKo1 the CLI seems about as usefull as tail -f /var/log/asterisk/messages |
| 15:44.40 | *** join/#asterisk Juggie (n=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) |
| 15:44.40 | n9urk | so I will attact that |
| 15:44.45 | n9urk | Thanks for the help thus far |
| 15:44.59 | eKo1 | jbalcomb: well, that depends on how you have logging set up. |
| 15:44.59 | jbalcomb | n9urk you have been checking your logs i hope/ |
| 15:45.00 | hi365 | pdtmobile: but i will need tor un it eventualy. It is needed to run the card. wanpipe1.conf doesnt mention anything about extera modules (FWI: im using 1-4, no way near the 9-10 and 16-17 its detecting) |
| 15:45.48 | pdtmobile | well then why are you asking how to uninstall it ;) |
| 15:46.01 | smackus | ok, I am looking for a bidder. My asterisk is not stable. it works, but over the course of a couple of days it starts throwing deadlock errors and eventually crashes. It is a production box, so it can have no downtime during the day. So I have been pulling 16 hour days for a couple of weeks trying to stabilize this bitch. Any takers? PM me. |
| 15:46.27 | *** join/#asterisk inv_arp[work] (i=junya@c-67-191-62-53.hsd1.fl.comcast.net) |
| 15:46.32 | n9urk | jbalcomb: I have just been monitoring the logs at teliax and that has served all the purposes I have needed, until now |
| 15:47.09 | *** join/#asterisk dwmw2_gone (n=dwmw2@baythorne.infradead.org) |
| 15:47.29 | hi365 | pdtmobile: good point. even with the card physicaly out of the computer it still "sees" the modules. So i need someway to flsu its "memory" and then ill reinstall it |
| 15:47.47 | pdtmobile | hi365: do you have zaptel.con configured for zap interfaces 8 9 16 17? |
| 15:47.54 | n9urk | jbalcomb: eKo1: Ok I got the problem fixed due to your guy's help. Thanks |
| 15:48.12 | *** join/#asterisk argos73 (n=mike@w010.z208036240.chi-il.dsl.cnc.net) |
| 15:48.44 | eKo1 | smackus: I suggest rebooting sometime late at night. |
| 15:48.50 | eKo1 | Every day. |
| 15:48.52 | pdtmobile | hi365: the A200 takes up 24 channels even if you only have two, just like a T1 card does. So you have to start numbering your other zap interfaces 24 + |
| 15:48.57 | hi365 | pdtmobile: at the moment its configured for nothing! #fxsks = 1-2 |
| 15:48.57 | hi365 | #fxsks = 9-10 |
| 15:50.05 | hi365 | hu? I dont understanbd what your trying to say, but surely regardless of where we start counting it should be in numerical order |
| 15:51.01 | pdtmobile | so your saying it's just picking those randomly? |
| 15:51.06 | hi365 | yup |
| 15:51.07 | *** join/#asterisk salviadud (n=ralfalfa@201.153.40.45) |
| 15:51.12 | pdtmobile | nice |
| 15:51.31 | *** join/#asterisk JohnJacob1 (n=JohnJaco@pool-71-127-102-43.aubnin.fios.verizon.net) |
| 15:51.45 | hi365 | Sangoma tech guy David claimed that means that the cards are defective, but thats only resonable IF the cards are randomly numbered when their in the system |
| 15:52.06 | hi365 | If their all out of the system, their obviously not to blame |
| 15:55.21 | *** join/#asterisk riddlebox (n=blah@24-207-167-238.dhcp.stls.mo.charter.com) |
| 15:55.36 | *** join/#asterisk Nobbie (n=no@fwb003.fw.is.co.za) |
| 15:55.38 | Nobbie | hi, |
| 15:55.46 | *** join/#asterisk asteriskwannabe (n=fholmes@rrcs-24-227-237-197.sw.biz.rr.com) |
| 15:56.19 | Nobbie | i have a problem with calls being dropped when they're blind transferred to a busy extension. how can i make them ring back to the transferee instead of dropping the call ? |
| 15:57.11 | asteriskwannabe | So I am looking at putting together a system that will handle two phone lines and one phone. Is the 4 port FXO/FXS card what I need? What kind of modules do I need? FXO or FXS? |
| 15:57.42 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.205.133.251) |
| 15:57.46 | angler | asteriskwannabe, TDM12B.... 1 FXS, 2FXO |
| 15:58.38 | [TK]D-Fender | asteriskwannabe : only 1 phone? |
| 16:00.08 | mut | do money market accounts usually have deposit and withdrawl limits, as in # of times? |
| 16:01.01 | *** join/#asterisk Mercestes (n=Merceste@216.54.143.2) |
| 16:01.08 | asteriskwannabe | Yes. It is a portable phone with a second base station. |
| 16:01.17 | asteriskwannabe | (Small office) |
| 16:01.42 | asteriskwannabe | The second base station does not have to connect to the phone line. It is more of a charging station and second phone. |
| 16:01.52 | salviadud | this time i called a rehab clinic in new york, like if i were british, i gave them a stanaphone number (also in new york) so they could contact me later. i love asterisk... it's sooooo evil |
| 16:01.55 | riddlebox | could me only allowing a couple of codecs affect not recieving caller id info on cingular cellphone? |
| 16:02.13 | *** join/#asterisk stubert (i=stu@techtools.actusa.net) |
| 16:02.41 | *** part/#asterisk tamp4x (n=tampon@www.vonworldwide.com) |
| 16:03.26 | hi365 | pdtmobile: my mistake |
| 16:03.34 | *** join/#asterisk JohnJacob (n=JohnJaco@pool-71-127-102-43.aubnin.fios.verizon.net) |
| 16:03.48 | hi365 | apperently it was in the messages scince yesterday |
| 16:04.20 | stubert | Is it possible for asterisk to manage sip and rtp traffic when the client attempts to connect via a url? |
| 16:04.57 | *** join/#asterisk gandhijee (n=gandhije@mail.win-ent.com) |
| 16:06.57 | asteriskwannabe | Thanks angler |
| 16:07.23 | gandhijee | anyone here use mdev in conjunction with zaptel? |
| 16:08.44 | *** join/#asterisk viperdude (n=jon@195.74.96.114) |
| 16:09.54 | *** join/#asterisk snoog (n=djc@68-188-220-62.dhcp.aldl.mi.charter.com) |
| 16:10.54 | snoog | This is probably a stupid question.. But if its in TFM I cant find it, or perhaps I'm looking in the wrong FM. I have been running asterisk as root, and I want to stop doing that. But when I try to run it as non-root, it is unable to create asterisk.pid in /var/run, becuase that is owned by root and is not world-writable. How do I eithe |
| 16:11.05 | snoog | This is probably a stupid question.. But if its in TFM I cant find it, or perhaps I'm looking in the wrong FM. I have been running asterisk as root, and I want to stop doing that. But when I try to run it as non-root, it is unable to create asterisk.pid in /var/run, becuase that is owned by root and is not world-writable. How do I tell asterisk it cant do that, and to stop trying |
| 16:11.15 | snoog | oop.. sorry for the dp |
| 16:11.54 | gandhijee | you can A) make the space use writeable, B) change the location of the pid |
| 16:12.01 | *** join/#asterisk ChkDigit (n=mike@static24-72-137-23.regina.accesscomm.ca) |
| 16:12.18 | gandhijee | or you can have it switch some how, like apache does |
| 16:12.19 | *** part/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com) |
| 16:12.23 | snoog | B would be preferrable.. how would I do that (and/or which FM should I look in to see how) |
| 16:12.39 | gandhijee | its probably something you modify in the sources |
| 16:12.43 | gandhijee | or makefile |
| 16:13.04 | gandhijee | and i am pretty sure there is info out there on how to run * as non-root |
| 16:13.08 | snoog | I would think that designin something that requires it to write to a location which is traditionally only writable by root would be poor security |
| 16:13.17 | gandhijee | i remember seeing some guides myself |
| 16:13.23 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.72.7) |
| 16:13.25 | snoog | one would hope |
| 16:13.33 | snoog | but i havent been able to find up to now |
| 16:13.44 | gandhijee | http://www.voip-info.org/wiki-Asterisk+non-root |
| 16:13.50 | gandhijee | google is your butt-buddy |
| 16:13.55 | snoog | tks |
| 16:14.08 | stubert | chown <astuser>: <fullpathandfilename> |
| 16:14.16 | snoog | silly me, I was googling for the Unable to open pid file '/var/run/asterisk.pid': Permission denied message |
| 16:14.27 | gandhijee | asterisk non-root |
| 16:14.31 | snoog | thinking that any such howto would say 'if you get this message, try this' |
| 16:14.40 | gandhijee | since thats what you are actually trying to do. |
| 16:15.03 | snoog | nod |
| 16:15.46 | snoog | If everyone suggests running asterisk not as root, you'd think that they wouldnt leave it set by default to have to run as root.. becuase im sure that encourages lots of people to do just that, instead of taking the time to manually adjust it to not |
| 16:16.05 | snoog | 'they |
| 16:16.06 | snoog | <PROTECTED> |
| 16:16.09 | gandhijee | u wouldn't happen to know how to use mdev would you? |
| 16:16.11 | snoog | 'they' = the * devs |
| 16:16.21 | snoog | i doubt it, since im not sure what it is |
| 16:16.27 | gandhijee | fair enough |
| 16:16.32 | snoog | what is it? |
| 16:16.53 | gandhijee | a replacement for udev for embedded systems |
| 16:17.01 | snoog | ah |
| 16:17.56 | gandhijee | yea |
| 16:18.22 | snoog | well I suppose if im gonna be recompiling I might as well go get the newest version |
| 16:18.36 | gandhijee | probably not a bad idea |
| 16:19.01 | snoog | i wish the freebsd port of it worked right |
| 16:19.02 | snoog | :P |
| 16:19.11 | snoog | actually I suppose that wouldnt help much |
| 16:19.11 | gandhijee | whats wrong with the BSD port? |
| 16:19.27 | gandhijee | you can setup jails/chroots in linux too IIRC |
| 16:19.29 | snoog | i dont remember.. i think it depends on zap, and zap doesnt work on freebsd.. or something |
| 16:19.37 | snoog | i remember i never could get it to compile |
| 16:19.42 | snoog | so I gave up and did it manually |
| 16:19.49 | snoog | and I dont even need zap |
| 16:19.49 | gandhijee | it should, zap was originally written on BSD |
| 16:20.00 | snoog | well lemme go see what make in the port does |
| 16:20.31 | *** join/#asterisk jero (n=jero@savoirfairelinux.net) |
| 16:20.51 | snoog | ===> Verifying install for /usr/local/include/zaptel.h in /usr/ports/misc/zaptel |
| 16:20.51 | snoog | ===> zaptel-0.11 "does not build on FreeBSD \< 5.x". |
| 16:21.14 | snoog | Yes, I need to upgrade, but the server is remote colo, and Im just not ready yet |
| 16:21.23 | gandhijee | O |
| 16:21.57 | snoog | In fact when I am ready, it will pretty much be leaving the exiting one running and building a new box to transition to |
| 16:22.20 | snoog | upgrading from 4 to newer is rather complex, i hear |
| 16:22.31 | snoog | as in far easier to just build new |
| 16:23.34 | snoog | Hrm.. -DWITHOUT_ZAPTEL |
| 16:23.49 | snoog | Of course I'll need to figure out how to patch rxfax and txfax into the port... |
| 16:24.03 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 16:24.11 | snoog | not to mention redo this after a cvsup, since its grabbing not the newest version of * |
| 16:24.41 | snoog | set out to do a simple thing like stop running * as root, and end up cvsuping my enteir system |
| 16:24.41 | snoog | sigh |
| 16:25.17 | wunderkin | dont use the port? |
| 16:25.33 | snoog | If I knew where to suggest that the * devs make * run as non-root properly by default, I' do so.. But I can't imagine where I do that that anyone would listen to |
| 16:25.35 | snoog | hrm |
| 16:25.40 | snoog | Yeah.. for now I think |
| 16:25.55 | *** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane) |
| 16:26.03 | snoog | stick with local src |
| 16:27.40 | *** join/#asterisk santiago (i=santiago@debian/developer/santiago) |
| 16:28.17 | *** join/#asterisk qdk (n=qdk@0x535eae17.boanxx9.adsl-dhcp.tele.dk) |
| 16:29.47 | *** join/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com) |
| 16:31.34 | nortex | What would be the best way to trigger an event in Asterisk, like run a call file, when an IAX trunk is unavaliable? |
| 16:32.02 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 16:32.02 | *** mode/#asterisk [+o Qwell[]] by ChanServ |
| 16:34.43 | *** join/#asterisk [Airwolf] (n=airwolf@dsl51B67C1E.pool.t-online.hu) |
| 16:34.54 | *** join/#asterisk pdtmobile (n=ptinsley@c-68-52-165-56.hsd1.tn.comcast.net) |
| 16:37.56 | jbalcomb | nortex: maybe the AMI |
| 16:41.12 | *** join/#asterisk RoyK (n=roy@gprs-ggsn5-nat.mobil.telenor.no) |
| 16:44.44 | Sonderblade | some hardphones have a "feature" so that when you press the "button" on the phone the old call is not hangup but you get a new dial tone |
| 16:45.08 | Sonderblade | i think it is supposed to be used for 3-way dialling, anyway is there a way to disable that really annoying feature? |
| 16:45.13 | russellb | asterisk sounds ... english, spanish, french ... ulaw, alaw, wav, gsm, g729 ... http://ftp.digium.com/pub/telephony/sounds/ |
| 16:45.16 | *** join/#asterisk smurf (n=smurf@debian/developer/smurf) |
| 16:45.33 | russellb | enjoy :) |
| 16:46.10 | Qwell[] | don't forget the fpm :D |
| 16:46.30 | russellb | oh yes, moh files in the same formats :) |
| 16:46.34 | russellb | no more mp3 for you! |
| 16:47.34 | *** join/#asterisk Mercestes (n=Merceste@216.54.143.2) |
| 16:49.30 | *** join/#asterisk fnordus (n=dnall@s142-179-111-243.bc.hsia.telus.net) |
| 16:50.24 | jbalcomb | If anyone wants the Polycom SoundPoint IP 501 SIP 1.6.6 firmware it is here: http://www.sendspace.com/file/spdxjc |
| 16:50.28 | nortex | jbalcomb, I was thinking more within the dialplan, Like if peer unreachable then do a system command to move a copy of a predefined call file to the spool |
| 16:51.17 | jbalcomb | nortex: use the AGI but just initiate a call based on values from the DB so you don't have to make a system call. |
| 16:52.32 | *** join/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net) |
| 16:52.53 | *** join/#asterisk Bullseye_Network (n=info@216.143.192.69) |
| 16:53.17 | TripleFFFF | when one trasnfers a call .. A call B .. B transfers to external line../...... why is B charged on the cdr ? |
| 16:53.20 | TripleFFFF | or how to prevent |
| 16:54.24 | nortex | But is there away to "trigger" this when a peer becomes unreachable or only when a call attempts to go accross that trunk? |
| 16:55.07 | jm|work | :| |
| 16:55.11 | TripleFFFF | triugger what .. just came |
| 16:55.22 | jm|work | I always have to wait 4 rings before * picks up from PSTN ?! |
| 16:55.34 | TripleFFFF | hmm |
| 16:55.38 | TripleFFFF | an option somewhere |
| 16:55.42 | TripleFFFF | whats the card |
| 16:55.47 | jm|work | despite immediate=yes |
| 16:56.03 | [TK]D-Fender | jm|work : Immediate=yes is NOT for LINES, its for PHONES. |
| 16:56.07 | jm|work | oh |
| 16:56.08 | TripleFFFF | dialing plan is blah ,s,1,answer ? |
| 16:56.12 | TripleFFFF | no wait there ? |
| 16:56.22 | jm|work | TripleFFFF: me? |
| 16:56.26 | jm|work | I have a Wait(1) |
| 16:56.30 | [TK]D-Fender | jm|work : Your issue is echo-training, callerID, and fax detection. |
| 16:56.37 | TripleFFFF | well pstn .. hmm |
| 16:56.44 | jm|work | [TK]D-Fender: cool. |
| 16:56.49 | TripleFFFF | remove the wait to see |
| 16:56.56 | jm|work | TripleFFFF: did that |
| 16:57.04 | [TK]D-Fender | jm|work : the 1 second wait isn't the issue. |
| 16:57.06 | TripleFFFF | wats the card |
| 16:57.12 | jm|work | some cloney thing |
| 16:57.13 | TripleFFFF | clone x ? |
| 16:57.17 | TripleFFFF | yeah |
| 16:57.24 | TripleFFFF | clones have problems getting callerid also |
| 16:57.31 | TripleFFFF | and caller id is between 2nd and 3rd ring |
| 16:57.45 | TripleFFFF | so maybe your version has a switch that says ..need caller id ;) |
| 16:57.46 | jm|work | hmm |
| 16:58.00 | jm|work | it does say that in zapata |
| 16:58.01 | TripleFFFF | hence the 4-5 rings.. 2-3- for callerid + 1 for wait |
| 16:58.08 | [TK]D-Fender | jm|work : Adding echotraining ADDS delay, and CID occurs between the 1st and 2nd ring. |
| 16:58.08 | TripleFFFF | here you go |
| 16:58.10 | TripleFFFF | nect ! |
| 16:58.23 | TripleFFFF | wheres BKW.. lol |
| 16:58.24 | jm|work | oh :( |
| 16:58.27 | [TK]D-Fender | and 1 ring +/- 4 sec |
| 16:58.42 | [TK]D-Fender | so wait(1) doesn't factor in. |
| 16:58.45 | [TK]D-Fender | (much) |
| 16:58.46 | TripleFFFF | I always have to wait 4 rings before * picks up from PSTN ?! |
| 16:58.47 | TripleFFFF | he said |
| 16:58.50 | TripleFFFF | not seconds.. but rings |
| 16:59.10 | [TK]D-Fender | TripleFFFF : Correct. But I never validated his analysis of the CAUSE. |
| 16:59.21 | [TK]D-Fender | TripleFFFF : Nor yours :) |
| 16:59.26 | TripleFFFF | lol |
| 16:59.34 | TripleFFFF | k |
| 16:59.35 | jm|work | 'Unknown' called me |
| 16:59.44 | [TK]D-Fender | TripleFFFF : You're batting 0 for 2 so far ;) |
| 16:59.59 | jm|work | [TK]D-Fender: so I removed a load of stuff from the zapata.conf |
| 17:00.01 | [TK]D-Fender | jm|work : Could be lack of caller ID, bad card, bad settings.... |
| 17:00.19 | jm|work | [TK]D-Fender: we have calledID on that line and I'm ringing from my mobile which sends |
| 17:00.21 | jm|work | anywho |
| 17:00.59 | Dr-Linux|work | my sound card is detected on my linux system, but i can't hear anything :( |
| 17:01.04 | jm|work | <PROTECTED> |
| 17:01.32 | [TK]D-Fender | Dr-Linux|work : do you think it just magically creates sound? |
| 17:03.12 | Dr-Linux|work | [TK]D-Fender, a song is playing ... |
| 17:03.57 | Sonderblade | how do you control which language asterisk chooses when it gets an incoming call? |
| 17:04.00 | [TK]D-Fender | Dr-Linux|work : What app? What environment? Checked your mixer? Checked your speakers? MRI? |
| 17:04.10 | Qwell[] | MRI? |
| 17:04.32 | jm|work | bah |
| 17:04.38 | jm|work | I'll just have to live with four ringhs |
| 17:04.39 | jm|work | -h |
| 17:04.51 | Dr-Linux|work | [TK]D-Fender, i checked with different players, currently it's playing with RealPlayer |
| 17:04.59 | jm|work | gives us chance to pick up the serial phone, I guess: it we're in the house |
| 17:05.11 | [TK]D-Fender | Qwell : Magnetic Resonnance Imaging.... I'm not sure he's "all there" ;) |
| 17:05.32 | Qwell[] | I'm just hoping that was sarcasm :P |
| 17:05.36 | [TK]D-Fender | :D |
| 17:05.43 | [TK]D-Fender | MOI?!?!! NEVER |
| 17:06.01 | *** join/#asterisk ph|ber (n=phiber@slackwaresupport.com) |
| 17:06.12 | ph|ber | is there any difference in the free * and the business one?? |
| 17:06.22 | ph|ber | and does the business come with the 729 codecs? |
| 17:06.26 | [TK]D-Fender | ph|ber : Yeah, Business one is supported and costs. |
| 17:06.33 | ph|ber | yea. i saw that. |
| 17:06.38 | [TK]D-Fender | ph|ber : Nope, still have to pay seperately like everything else. |
| 17:06.45 | ph|ber | damn |
| 17:06.53 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 17:06.58 | ph|ber | so other than cost and support theres no difference? |
| 17:07.04 | shmaltz | what type of hangup detection is used in the US? |
| 17:07.10 | [TK]D-Fender | ph|ber : Business one also allows integration in proprietary solutions without GPL getting in the way |
| 17:07.27 | ph|ber | can you buy just the manual? |
| 17:07.33 | ph|ber | the tech manual> |
| 17:07.34 | *** join/#asterisk System010 (n=jgargano@hide247.cybergnostic.com) |
| 17:07.34 | [TK]D-Fender | shmaltz : Polarity reversal usually (where even offered) |
| 17:07.42 | [TK]D-Fender | ph|ber : .... |
| 17:07.43 | [TK]D-Fender | ~book |
| 17:07.44 | jbot | book is probably a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 17:07.48 | [TK]D-Fender | THERE! |
| 17:08.07 | shmaltz | [TK]D-Fender, in my experience it's always offered, at least under Verizon Teritory |
| 17:08.07 | ph|ber | :P |
| 17:08.13 | Dr-Linux|work | [TK]D-Fender, thanks , it's fixed |
| 17:08.29 | *** join/#asterisk eKo1 (n=eKo1@190.4.7.90) |
| 17:08.34 | System010 | has anyone worked with an inter-tel axxess system before? |
| 17:08.36 | shmaltz | [TK]D-Fender, so you telling me what verizon gives is Polarity reversal |
| 17:08.39 | shmaltz | ? |
| 17:09.05 | ph|ber | anyone use the astgui? |
| 17:09.22 | *** join/#asterisk dmesg (n=dmesg@netbsd/user/dmesg) |
| 17:09.27 | dmesg | hi |
| 17:09.59 | [TK]D-Fender | shmaltz : Should be. |
| 17:10.11 | dmesg | asteris is good for phreking? :P |
| 17:10.14 | shmaltz | [TK]D-Fender, thanks |
| 17:10.33 | [TK]D-Fender | ph|ber : No GUI for yoooouuuuu!!!! |
| 17:11.02 | *** join/#asterisk bofh42 (n=bofh42@p54828F2A.dip0.t-ipconnect.de) |
| 17:11.03 | dmesg | where can i download the iso to install asterisk ? |
| 17:11.23 | System010 | dmesg: you mean the install? |
| 17:11.26 | [TK]D-Fender | LOL |
| 17:11.33 | dmesg | System010 yes |
| 17:11.46 | [TK]D-Fender | dmesg : * is an APP. there is no "ISO" :) |
| 17:11.56 | System010 | source files are on asterisk.org |
| 17:11.58 | dmesg | [TK]D-Fender ahhh |
| 17:12.03 | System010 | under the download section |
| 17:12.09 | dmesg | i see |
| 17:12.16 | [TK]D-Fender | dmesg : Install a Linux distro, download source, compile, enjoy. |
| 17:12.19 | snoog | FUCK |
| 17:12.25 | dmesg | so it can work on netbsd rigth? |
| 17:12.25 | Dr-Linux|work | [TK]D-Fender, now thinking how can i import .pst file from outlook to linux evolution mail client |
| 17:12.29 | snoog | I actually went ahead with trying the port |
| 17:12.33 | snoog | which wanted an upgraded spandsp |
| 17:12.37 | snoog | which was installed from port |
| 17:12.40 | [TK]D-Fender | dmesg : Dunno, but wouldn't bet on it. |
| 17:12.47 | snoog | but the new one keeps giving me some crap about std=c99 not being valid |
| 17:12.53 | dmesg | [TK]D-Fender ok |
| 17:13.01 | snoog | and now, the existing asterisk wont start becuase spandsp is missing, |
| 17:13.17 | snoog | and i dont think its possible to revert the port for spandsp back to its old one |
| 17:14.17 | [TK]D-Fender | snoog : "port"? |
| 17:15.02 | snoog | freebsd /usr/ports |
| 17:15.37 | [TK]D-Fender | snoog : Ah... more masochists trying to run * non-Linux.... |
| 17:15.56 | snoog | well is was running fine until I decided I wanted to stop running it as root |
| 17:16.04 | snoog | then I decided i might as well bump the version while I was at it |
| 17:16.17 | snoog | then I decided Id try the version in ports insead of local src |
| 17:16.33 | snoog | and linux is great for workstations, crap for servers |
| 17:16.34 | snoog | :P |
| 17:16.40 | snoog | *kidding* |
| 17:16.44 | snoog | but I do prefer freebsd on a server |
| 17:17.16 | snoog | unfortunately, im using a rather old dog.. RELENG_4 is getting crusty, but im not ready to upgrade yet |
| 17:17.22 | [TK]D-Fender | I prefer "functional" on a server :) |
| 17:18.20 | snoog | well everything else is functionall.. this box does quite a lot.. im just caught trying to use the 'official' install methods for a rather old version of 'official', and the world has passed me by |
| 17:18.39 | snoog | one of these days I need to buy some new iron and install a fresd freebsd6 on it |
| 17:18.45 | snoog | fresh |
| 17:18.54 | jm|work | is that why my callerid isn't working? |
| 17:19.46 | snoog | i doubt it |
| 17:20.42 | snoog | maybe if I install a newer gcc |
| 17:20.45 | snoog | sigh |
| 17:21.08 | *** join/#asterisk mog (i=ejabberd@68.62.237.103) |
| 17:21.08 | *** mode/#asterisk [+o mog] by ChanServ |
| 17:21.47 | kay2 | Is it possible to do videoconferencing with SS7 ? |
| 17:21.57 | *** join/#asterisk tempest1 (n=Brett@adsl-153-33-178.chs.bellsouth.net) |
| 17:23.13 | RoyK | kay2: ss7 is a signalling protocol, so it should be able to control any sort of communication |
| 17:23.19 | *** part/#asterisk dmesg (n=dmesg@netbsd/user/dmesg) |
| 17:24.04 | *** join/#asterisk docE (n=docelmo@66.237.242.41.ptr.us.xo.net) |
| 17:24.11 | RoyK | rotfl. http://en.wikipedia.org/wiki/World_Jump_Day |
| 17:24.51 | kay2 | RoyK: but basically, if someone calls a pstn number with a 3G phone in 3G mode, what happens ? |
| 17:25.41 | docE | That is stupid |
| 17:26.15 | RoyK | i would guess the call is rejected by pstn, unless it's setup to allow extra b channels for the video |
| 17:26.47 | RoyK | call the telco and ask if you can do that |
| 17:29.47 | *** join/#asterisk daysmen3 (n=primus@host86-137-170-127.range86-137.btcentralplus.com) |
| 17:29.55 | *** join/#asterisk mmmmmToop (n=mmmmToop@firewall.datapro.co.za) |
| 17:30.12 | mmmmmToop | anyone know how to stop a Snom from transfering with the Transfer button? |
| 17:30.49 | mmmmmToop | Wierd request I know...but this is the story: we need to transfer a call out a queue |
| 17:31.03 | kay2 | RoyK: and If I have a E1 |
| 17:31.04 | mmmmmToop | & it works fine using # to transfer, but if one uses the feature transfer it looses the call... |
| 17:31.20 | kay2 | and I call from my softphone using 3G mode, would it still be rejected ? |
| 17:31.23 | *** part/#asterisk smackus (n=ckwall@63.149.122.93) |
| 17:31.48 | kay2 | or it would send the data directly to the channel ? |
| 17:32.01 | *** join/#asterisk smackus (n=ckwall@63.149.122.93) |
| 17:32.30 | RoyK | kay2: 3g call -> some switch. that switch will most prrobably send the call to another switch which will reject the call since it's got incompatible media |
| 17:34.21 | System010 | so has anyone had luck connecting and Inter-Tel Axxess system to asterisk? |
| 17:36.09 | TripleFFFF | http://en.wikipedia.org/wiki/World_Jump_Day |
| 17:36.10 | TripleFFFF | ??????? |
| 17:36.34 | TripleFFFF | people got compiled --WITHOUT-BRAIN |
| 17:36.53 | RoyK | :) |
| 17:37.02 | TripleFFFF | lans to have 600 million people from the western hemisphere jump simultaneouslyThey claim this will move the Earth out of its current orbit |
| 17:37.20 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
| 17:37.53 | TripleFFFF | now why the hell would one want that.. even why would 600 mill want to suicide like that.. gravity is a precise thing.. me dodenst think getting sun and moon out the equatin is any good.. |
| 17:38.01 | kay2 | RoyK: but when using a E1, it sends 32 channel of 64kb each right ? |
| 17:38.19 | RoyK | 30 |
| 17:38.35 | RoyK | chan 0 is sync and (usually) chan 16 is dchan |
| 17:40.59 | kay2 | well even chan 0 and 16 is 64kb channel |
| 17:41.21 | RoyK | kay2: but the telco _will_ need to support this. any idea of how much bandwidth the video is? |
| 17:41.33 | kay2 | 64kb |
| 17:41.41 | RoyK | kay2: E1 is 2048kbps, yes, so each timeslot is 64kbps |
| 17:41.43 | *** join/#asterisk trbldwine (n=trbldwin@adam.ur.northwestern.edu) |
| 17:42.23 | gandhijee | i am trying to run some TMDoE, i belive i have my setup correct, but asterisks reports my TDMoE span as down |
| 17:42.38 | gandhijee | do i need an asterisk on the side that is exporting the TDMoE or something? |
| 17:44.29 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.238.118.Dial1.SanJose1.Level3.net) |
| 17:46.37 | *** join/#asterisk g__ (n=g@itd01fw-fibre.itdepartment.com) |
| 17:47.27 | g__ | I give up. Is there was I can test if an extension exists before jumping to it? |
| 17:48.28 | Qwell[] | g__: According to show application goto, if the exten doesn't exist, it'll continue on in the dialplan |
| 17:48.39 | kay2 | RoyK: but I can use one of the 64kb channel for data right ? |
| 17:48.42 | g__ | really? Cool.. I'll try it. |
| 17:48.46 | g__ | Thanks Qwell |
| 17:48.50 | kay2 | RoyK: or no ? |
| 17:48.53 | Strom_C | also, if you're designing your dialplan correctly, this shouldnt be an issue |
| 17:49.07 | Qwell[] | Strom_C: BAD |
| 17:49.10 | Qwell[] | BAD, BAD, BAD |
| 17:49.16 | Qwell[] | nevermind :P |
| 17:49.36 | Qwell[] | I was gonna dock you a point for forgetting a ' in shouldn't |
| 17:49.38 | i-ball | uh..there's an outhouse, you know |
| 17:50.06 | g__ | Strom_C: I guess it's a matter of opinion.. |
| 17:50.31 | g__ | And context (in original def' of context, not the asterisk one.) |
| 17:51.27 | RoyK | kay2: isdn-based videophones use one channel (or more) for video, yes |
| 17:51.42 | *** join/#asterisk okdo (n=goldenol@65.171.196.18) |
| 17:51.46 | okdo | hi |
| 17:52.27 | *** join/#asterisk russellb_ (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 17:52.27 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 17:52.31 | g__ | Qwell: actually, I suspect control is passed to the 'i' context rather than going to the next line. |
| 17:52.39 | okdo | I am using the fax extension --- which works well, I answer the channel and it drops to the fax extension when it detects the ring tone but it always rings the SIP phone (1) ring prior to passing the call to the fax extension --- I've inserted up to 3 seconds of Wait() and it doesn't help --- any thoughts? |
| 17:53.29 | RoyK | okdo: do you wait() after picking up the call? |
| 17:53.33 | Qwell[] | g__: Not with goto..not according to the docs |
| 17:53.51 | kay2 | RoyK: but is it possible to call from a 3G phone to a ISDN phone ? |
| 17:54.45 | okdo | yeah i do a Answer() and then Wait(2) and then Dial(SIP/.... |
| 17:55.18 | RoyK | okdo: perhaps playtones(ring) will help the users dialing in not to be disturbed..... |
| 17:56.04 | okdo | hmmm! good thought, let me test |
| 17:56.21 | RoyK | also, try wait(5) or so |
| 17:56.35 | RoyK | perhas 2 is a little too low |
| 17:57.23 | g__ | Qwell: well I tried.. I guess the documentation is wrong. |
| 17:58.02 | *** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.238.118.Dial1.SanJose1.Level3.net) |
| 17:58.20 | kay2 | RoyK: when I call from one ISDN to an other ISDN, it uses directly the line as a data channel |
| 17:58.21 | kay2 | right ? |
| 17:59.05 | tessier_ | What's the start of the art in VOIP faxing? Is it still a mess? |
| 17:59.15 | tessier_ | I've been out of the asterisk biz for the last year. |
| 17:59.30 | RoyK | kay2: it would be using one or more bchans for video in addition to the bchan for audio, yes |
| 17:59.54 | kay2 | RoyK: but when someone has a video phone and he places a call to an other number |
| 17:59.57 | kay2 | if it's just one call |
| 18:00.02 | kay2 | there is just one bchannel used |
| 18:00.20 | kay2 | am I right ? |
| 18:02.28 | *** join/#asterisk linlin (n=linlin@c-67-184-159-30.hsd1.il.comcast.net) |
| 18:02.43 | RoyK | iirc isdn-isdn video is done with one bchan per media or with several bchans for the video |
| 18:03.17 | kay2 | RoyK: with a digium quad E1 card |
| 18:03.25 | kay2 | RoyK: can I get the data directly from one channel ? |
| 18:03.30 | kay2 | RoyK: in raw ? |
| 18:03.57 | *** join/#asterisk ^Tr4sh^ (n=drttrtr@81-208-62-98.ip.fastwebnet.it) |
| 18:05.18 | RoyK | kay2: i don't think that would be the problem. i would rather think either the call would be rejected by the telco's switch or the video would be discarded on the way |
| 18:05.49 | i-ball | hey |
| 18:06.02 | i-ball | can anybody explain to me local channel creation in Asterisk? |
| 18:06.08 | RoyK | ~local |
| 18:06.11 | jbot | it has been said that local is like, is your system local to you? Can you physically touch it from where you're sitting, or maybe by going to another room? As opposed, say, to being 1500km from you, accessible only via air and sea (combined) travel, and installed in a restricted-access facility? This matters if we, say, try to restart your system and it doesn't. |
| 18:06.14 | RoyK | ~chan_local |
| 18:06.23 | RoyK | ~docs |
| 18:06.25 | jbot | somebody said docs was probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 18:06.25 | RoyK | ~book |
| 18:06.27 | jbot | it has been said that book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 18:06.43 | RoyK | kay2: do you |
| 18:07.09 | RoyK | kay2: do you really get a call on the PRI when dialling from the 3g phone? |
| 18:07.51 | *** join/#asterisk razu (n=razu@87-119-182-133.tll.elisa.ee) |
| 18:07.56 | *** join/#asterisk SplasPood (n=jwb@64.90.191.180.nyinternet.net) |
| 18:08.01 | RoyK | kay2: take a look with a full pri debug |
| 18:08.08 | RoyK | pri intense debug pri 1 |
| 18:08.11 | i-ball | uh... |
| 18:09.39 | RoyK | i-ball: it's a kind of advanced goto |
| 18:09.48 | RoyK | or perhaps not so advanced |
| 18:09.59 | *** join/#asterisk benjamin7062 (n=benjamin@mailserver.photodex.com) |
| 18:10.25 | kay2 | RoyK: well there is a local number when I call it with 3G option, I get video |
| 18:10.39 | i-ball | When I say that I need to know how to create local channels I mean: |
| 18:10.43 | i-ball | I saw an example that said: |
| 18:10.55 | i-ball | "Local/101@stream" |
| 18:11.09 | RoyK | kay2: erm. what do you mean? from 3g to 3g? |
| 18:11.13 | benjamin7062 | I have a channel that seems to be locked; when I do a show channels I get this. Can anyone explain this line: |
| 18:11.15 | benjamin7062 | Local/8090@phones-73 s@phones:1 Down (None) |
| 18:11.22 | benjamin7062 | It's just 'stuck' there? |
| 18:11.33 | kay2 | RoyK: from 3G to a ISDN |
| 18:11.34 | kay2 | E1 |
| 18:12.10 | RoyK | well. i really don't understand what you're asking about. sorry... |
| 18:12.11 | benjamin7062 | It shows the correct status of 'down' but won't go away and the phone is no longer in use... or making any sort of calls.. etc |
| 18:12.31 | *** join/#asterisk Bobcat_1966 (n=chatzill@cpe-069-132-139-254.carolina.res.rr.com) |
| 18:12.50 | RoyK | benjamin7062: does a 'soft hangup' work? or a restart? does asterisk want to restart at all? |
| 18:12.52 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:13.29 | benjamin7062 | RoyK, well, I was hoping to understand the issue before I clear the state. Since I'm not sure I can reproduce it reliably |
| 18:13.31 | RoyK | kay2: what sort of isdn phone do you use with video? how is this connected? i do not understand |
| 18:14.03 | RoyK | benjamin7062: just a sec |
| 18:14.13 | benjamin7062 | RoyK, I'm sure a restart will fix it, but I don't want to have to restart with these.. so I was wondering if anyone knew what condition would cause a 'Local' channel, and why they would sit there. It's showing the state of 'Down' correctly, just not going away |
| 18:14.20 | RoyK | local show channels? |
| 18:14.35 | benjamin7062 | yeah, it shows up there. |
| 18:15.02 | RoyK | benjamin7062: also, gdb -p asterisk-pid, thread info and 'thread apply all bt full' and file it in bugs.digium.com |
| 18:15.42 | benjamin7062 | bummer.. so it's probably a bug |
| 18:15.53 | benjamin7062 | okie dokie |
| 18:18.59 | snoog | alrighty.. so I installed gcc33, told the spandsp port to use it instead, and now its not complaining about c99 anymore.. |
| 18:19.08 | snoog | but not, its griping about 'INT16_MAX' undefined |
| 18:19.09 | snoog | sigh |
| 18:20.09 | snoog | in any case, i did manage to reinstall the old spandsp from a package.. so at least my existing setup is still working |
| 18:20.25 | *** join/#asterisk supjigatr (n=syslod@152.53.16.10) |
| 18:21.16 | *** join/#asterisk pdthome (n=pdthome@c-68-53-40-50.hsd1.tn.comcast.net) |
| 18:21.23 | supjigatr | Anyone seen occasional DTMF tones not recognized on inbound PRI calls to a menu? |
| 18:21.36 | snoog | Ok, I'm going to pretend I'm shopping for a new box.. Anyone have any recomendations for mainboards, cpu, etc? |
| 18:21.40 | i-ball | ~channels |
| 18:21.45 | RoyK | supjigatr: i beleive so |
| 18:21.52 | snoog | its been so long since I built any new machines |
| 18:21.55 | i-ball | ~local |
| 18:21.57 | jbot | i heard local is like, is your system local to you? Can you physically touch it from where you're sitting, or maybe by going to another room? As opposed, say, to being 1500km from you, accessible only via air and sea (combined) travel, and installed in a restricted-access facility? This matters if we, say, try to restart your system and it doesn't. |
| 18:22.07 | i-ball | ~/ |
| 18:22.09 | jbot | ~/ is your home dir silly!, or root, of all Unix |
| 18:22.12 | i-ball | eh.. |
| 18:22.18 | snoog | ~~ |
| 18:22.20 | jbot | Every moment in which im called upon is torture |
| 18:22.20 | RoyK | ~lart i-ball |
| 18:22.24 | trelane_ | ~botsnack |
| 18:22.25 | jbot | aw, gee, trelane_ |
| 18:22.27 | snoog | ~windows |
| 18:22.29 | jbot | hmm... windows is a 32 bit hack on a 16 bit operating system, originally designed for an 8 bit CPU, with a 4 bit system bus, made by a 2 bit company that can't stand 1 bit of competition... or the World of Warcraft bootloader, or the most important collection of bugs |
| 18:22.32 | supjigatr | RoyK: Any fix? |
| 18:22.32 | i-ball | shit, can anybody point me to a tutorial on local channel creation? |
| 18:22.32 | snoog | heh |
| 18:22.40 | benjamin7062 | ~fart |
| 18:22.41 | jbot | ACTION farts, releasing large quantities of methane and sulfur dioxide. "Evacuate the channel! GO! *gag* SAVE YOURSELVES *cough* MOVE *choke* MOVE!" |
| 18:22.41 | i-ball | what conf file do I need to edit? |
| 18:22.47 | *** join/#asterisk russellb_ (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 18:22.47 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 18:22.48 | snoog | ~billgates |
| 18:22.49 | jbot | somebody said billgates was http://www.geocities.com/TimesSquare/Dungeon/2170/fearthepenguin.jpg |
| 18:22.49 | Qwell[] | wow bootloader...haha |
| 18:22.58 | RoyK | supjigatr: not yet. it was some time ago. think it works now |
| 18:23.01 | snoog | ~slshdot |
| 18:23.03 | snoog | ~slashdot |
| 18:23.17 | supjigatr | RoyK: HEAD or asterisk current? |
| 18:23.25 | *** join/#asterisk russellb_ (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 18:23.25 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 18:23.34 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.72.7) |
| 18:24.14 | Qwell[] | russellb: having problems? |
| 18:24.43 | russellb | Qwell[]: yes |
| 18:28.05 | Bullseye_Network | when asterisk dumps where would it write the log? |
| 18:28.19 | *** join/#asterisk Cubano (n=Cubano@unaffiliated/cubano) |
| 18:28.23 | TripleFFFF | ./tmp |
| 18:28.28 | TripleFFFF | or depends on core |
| 18:28.42 | TripleFFFF | ./proc/sys/kernel/core_pattern |
| 18:28.45 | TripleFFFF | could look there |
| 18:28.54 | TripleFFFF | or sysctl -a |grep core |
| 18:29.21 | TripleFFFF | default is where app got aluched |
| 18:29.22 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.72.7) |
| 18:29.32 | TripleFFFF | so if you lauched from /root/ its in /root/pid.core |
| 18:29.36 | TripleFFFF | etc |
| 18:29.40 | TripleFFFF | sorry my enter key broken |
| 18:30.12 | Bullseye_Network | Launch with safe_asterisk on boot |
| 18:30.31 | gandhijee | i am trying to run some TMDoE, i belive i have my setup correct, but asterisks reports my TDMoE span as down |
| 18:30.35 | gandhijee | do i need an asterisk on the side that is exporting the TDMoE or something? |
| 18:30.54 | i-ball | for SIP channels I need to edit sip.conf |
| 18:30.54 | i-ball | for iax channels I need to edit iax.conf |
| 18:30.54 | i-ball | for local channels I need to edit --? |
| 18:30.54 | i-ball | or does nothing need to be edited? |
| 18:30.57 | i-ball | and they're just dynamically created? |
| 18:31.00 | Bullseye_Network | it should be in tmp but there was nada |
| 18:31.17 | i-ball | ? |
| 18:31.38 | [TK]D-Fender | i-ball : Local channel is a wait to dial INTO the dialplan from withing the dialplan. Like a GOTO, but more. |
| 18:32.20 | i-ball | ah, okay |
| 18:32.32 | i-ball | so can you explain to me the parts of the following line: |
| 18:32.40 | *** join/#asterisk lindy_R (i=HydraIRC@24.196.26.177) |
| 18:32.41 | i-ball | Local/101@stream |
| 18:33.08 | i-ball | is @stream the context? |
| 18:33.14 | i-ball | and 101 the extension? |
| 18:33.14 | [TK]D-Fender | i-ball : Correct |
| 18:33.25 | [TK]D-Fender | i-ball : yes |
| 18:33.29 | i-ball | PERFECT! |
| 18:33.32 | i-ball | Thanks a lot! |
| 18:33.40 | benjamin7062 | How do I hang up a 'local' channel if it's hung? |
| 18:33.50 | benjamin7062 | That is, if restart isn't an option |
| 18:34.27 | benjamin7062 | And soft hangup no workie |
| 18:34.41 | eKo1 | restarting is the only way |
| 18:34.42 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 18:34.48 | benjamin7062 | ~spit |
| 18:39.31 | jm|work | so has anyone got UK CallerID working on a FXO? |
| 18:39.51 | jm|work | the howtos I keep finding are somewhat legacy |
| 18:40.10 | i-ball | that's an understatement |
| 18:42.56 | *** join/#asterisk topping (n=topping@207.47.6.136.static.nextweb.net) |
| 18:45.37 | *** join/#asterisk topping (n=topping@207.47.6.136.static.nextweb.net) |
| 18:47.24 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 18:49.19 | jm|work | \: |
| 18:50.59 | marv[work] | are there any docs on asterisk's locking scheme? |
| 18:51.20 | *** join/#asterisk redondos (n=redondos@190.48.2.204) |
| 18:51.32 | redondos | AFAIK, I'm not using SQLite, but I'm getting this error: |
| 18:51.32 | redondos | Jul 19 12:50:19 ERROR[10145]: cdr_sqlite.c:157 sqlite_log: cdr_sqlite: attempt to write a readonly database |
| 18:52.17 | redondos | And `grep -i sqlite /etc/asterisk/*' finds nothing. |
| 18:52.45 | eKo1 | in your cli, type unload cdr_sqlite.so |
| 18:52.58 | *** join/#asterisk RoyK (n=roy@gprs-ggsn5-nat.mobil.telenor.no) |
| 18:53.20 | redondos | Good. Is it gone for good, or will it be brought back when I Restart? |
| 18:53.44 | redondos | It's back. :( |
| 18:54.01 | mmmmmToop | wouldnt even try...;) |
| 18:54.01 | redondos | What database does it want to write in? |
| 18:54.05 | redondos | There's nothing in /var/log/asterisk |
| 18:54.17 | redondos | mmmmmToop: What do you mean? |
| 18:54.17 | eKo1 | You need to add a noload => cdr_sqlite.so in modules.conf |
| 18:54.26 | redondos | Great, thanks. |
| 18:54.30 | mmmmmToop | sorry...wrong chat ;) |
| 18:54.35 | redondos | But what is it complaining about, where is the database supposed to be located at? |
| 18:56.00 | jm|work | bugger :| |
| 18:56.11 | jm|work | I patched the #DEFAULT_CIDRINGS 2 |
| 18:56.14 | jm|work | but still no joy |
| 18:56.34 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
| 18:56.53 | eKo1 | redondos: no clue |
| 18:56.59 | redondos | eKo1: Ok, thanks. |
| 19:02.35 | vader-- | ok power over ethernet experts |
| 19:03.04 | vader-- | can i create a box that has 3 outlets that is fed by one line that has poe and 3 data cables? |
| 19:03.10 | vader-- | to power 2 phones and an access point? |
| 19:04.52 | benjamin7062 | vader - no |
| 19:05.00 | eKo1 | probably not but feel free to experiment |
| 19:05.07 | benjamin7062 | vader -- you can feed 2 net connections in one line but that eats up the pairs for power |
| 19:06.52 | benjamin7062 | vader--, Never tried but you 'might' be able to split the power three ways... but to get 3 data AND poe devices on the same line won't work... since the data requires 4 wires each... no matter what |
| 19:07.08 | *** join/#asterisk s0lid (n=s0lid@124.6.176.99) |
| 19:07.40 | benjamin7062 | vader--, all of that was assuming the 'one line' feeding the box is standard cat-5 |
| 19:07.40 | vader-- | no i would bus the pairs for the poe and run 3 lines for data |
| 19:08.18 | vader-- | in 568B wireing schema the brown and blue pairs are for poe? |
| 19:08.28 | benjamin7062 | yes |
| 19:08.31 | vader-- | and the orange and green are for data |
| 19:08.35 | benjamin7062 | yes sir |
| 19:09.17 | *** join/#asterisk nortex (n=breeves@snapper.titanspecialties.com) |
| 19:10.15 | benjamin7062 | I'm assuming you want to pull 1 cat 5 cable to a location that has PoE -- then split that out 3 ways. That is too many -- even if you did some trickery on the other end of the line. There aren't enough wires in a standard cat-5 cable. |
| 19:10.46 | vader-- | na |
| 19:10.53 | vader-- | i want to pull 3 wires |
| 19:10.58 | vader-- | but only one wire has POE |
| 19:11.06 | vader-- | i want to split just the poe over the 3 plugs |
| 19:11.14 | benjamin7062 | Ahhh |
| 19:11.24 | [TK]D-Fender | vader-- : Buy another PoE switch |
| 19:12.47 | benjamin7062 | vader--, I suppose in theory -- that'd work but I'm not sure if you'd get enough power outta the pairs. |
| 19:13.45 | benjamin7062 | vader--, I'd have to agree with [TK]D-Fender -- just put another PoE switch in the back room.. =) If you are pulling 3 cables then it makes sense. But, I suppose we don't know what you are trying to accomplish |
| 19:13.53 | godsmoke | yeah, the wires are going to have safety limits |
| 19:14.00 | godsmoke | about how much power can go through them |
| 19:14.15 | i-ball | ~asterisk-ices.xml |
| 19:14.24 | *** join/#asterisk sevard (n=sev@adsl-71-129-115-244.dsl.irvnca.pacbell.net) |
| 19:14.26 | benjamin7062 | Is it 24v dc? |
| 19:14.33 | *** join/#asterisk mog_home (n=mogorman@68.62.237.103) |
| 19:14.33 | *** mode/#asterisk [+o mog_home] by ChanServ |
| 19:15.39 | benjamin7062 | I would think it would work so long as long as 3x power for devices !> 24v dc (or whatever the magic number is) |
| 19:15.48 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 19:15.48 | *** mode/#asterisk [+o russellb] by ChanServ |
| 19:16.05 | *** join/#asterisk Defraz (n=t0tal@fw.centrisys.com) |
| 19:16.08 | benjamin7062 | I'd spend $50.00 personally. ;-) |
| 19:16.30 | godsmoke | well the voltage doesn't go down by splitting the power |
| 19:16.34 | godsmoke | just the current |
| 19:18.39 | benjamin7062 | <-- not a power expert... was speaking in theory... like, having a 20amp circuit in a house.. and splitting that circuit off to several outlets... So long as the outlets combined don't go above 20amp.. the breaker never trips. That kinda thing, however it works in the real world |
| 19:19.23 | TripleFFFF | current does go down neither |
| 19:19.52 | TripleFFFF | all dpeends on which leg pulls the most.. thing to consider is using wires htat can take avg 15 amp so usually 20% more. .hence 20amp gage |
| 19:19.54 | *** join/#asterisk froguz (n=xxxxx@pc-95-155-104-200.cm.vtr.net) |
| 19:20.51 | *** join/#asterisk russellb_ (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 19:20.51 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 19:21.38 | froguz | Hi |
| 19:22.11 | TripleFFFF | hi there |
| 19:22.20 | froguz | wath is the simplest way to know wich codec is being used in a call? |
| 19:22.25 | froguz | any CLI command? |
| 19:22.52 | sevard | sip debug. |
| 19:22.56 | RoyK | sip show channel ... |
| 19:23.05 | RoyK | sevard: that's overkill and only works with call setup |
| 19:23.13 | RoyK | froguz: sip show channels |
| 19:23.15 | RoyK | if it's sip |
| 19:23.52 | froguz | thanks |
| 19:24.52 | *** part/#asterisk g__ (n=g@itd01fw-fibre.itdepartment.com) |
| 19:25.55 | *** join/#asterisk rpm (n=russell@S01060002b3d10d24.cg.shawcable.net) |
| 19:26.53 | Damin | Anyone using a Plantronics CS50 headset w/ a Polycom 501? |
| 19:27.15 | *** join/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com) |
| 19:27.48 | *** part/#asterisk russellb_ (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 19:29.21 | *** join/#asterisk Ironhand (i=arjen@meek.xs4all.nl) |
| 19:30.00 | [TK]D-Fender | Damin : I've used a simial unit, what of it? |
| 19:30.27 | gandhijee | anyone know where i can find the newt libs so i can compile zttool? |
| 19:30.49 | [TK]D-Fender | Damin : Yup, it uses the same optional handset lefter as mine did. |
| 19:31.52 | Damin | [TK]D-Fender: Do you need the lifter to have it answer from the headset Call Control button? |
| 19:32.32 | [TK]D-Fender | Damin : Yup... very annoying. |
| 19:32.43 | [TK]D-Fender | Damin : this is not an "intelligent" solution |
| 19:32.46 | Damin | [TK]D-Fender: Yes.. extremely annoying.. |
| 19:33.06 | [TK]D-Fender | Damin : I ran my call center on IP 600's with lifters. means you don't see the caller ID, etc. |
| 19:33.18 | *** join/#asterisk nighty_ (n=nighty@66-163-28-100.ip.tor.radiant.net) |
| 19:33.18 | [TK]D-Fender | Damin : if you're away from your desk. |
| 19:33.23 | nighty_ | hi :) |
| 19:33.27 | froguz | is there a way to know if asterisk is making transcoding? |
| 19:33.46 | [TK]D-Fender | Damin : more function to often just use an ATA with a cordless phone w/ BT |
| 19:34.03 | [TK]D-Fender | froguz : use "show channels" and look at both sides of the call. |
| 19:34.04 | nighty_ | anyone succeeded in compiling chan-sccp under freebsd 6.1 ? |
| 19:34.07 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) |
| 19:34.42 | sevard | I have a question -- in sip.conf you can set the astdb= field to set fields in the database, but it doesn't look like that's working, is there something you haev to enable? |
| 19:35.37 | [TK]D-Fender | sevard : pastebin what you're doing to set it, and what you're doing to test it. |
| 19:43.37 | *** join/#asterisk jgoo (n=foo@ppp200-161.adsl.forthnet.gr) |
| 19:43.57 | jgoo | quick way to test is asterisk is running? (something like -vvv??) thanks |
| 19:44.19 | [TK]D-Fender | ps -A|grep asterisk |
| 19:44.41 | jgoo | aah yes :) |
| 19:45.31 | jgoo | :( no not running, I just setup trixbox and it died when I did install-ZAPHFC |
| 19:45.48 | jgoo | hrm, I will keep that part in #freepbx ;)' |
| 19:45.57 | *** join/#asterisk russellb_ (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 19:45.57 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 19:47.20 | sevard | [TK]D-Fender: in sip.conf astdb = CLIENTS/SIP-1338=pimpninjas |
| 19:47.53 | sevard | [TK]D-Fender: extensions.conf: exten => s,4,Noop(${DB_EXISTS(CLIENTS/SIP-1338)}) ; should be "pimpninjas" |
| 19:48.26 | sevard | should be "1", not "pimpninjas" |
| 19:48.33 | sevard | but it returns 0 |
| 19:48.54 | [TK]D-Fender | sevard : that a raw paste? |
| 19:49.22 | sevard | if you're asking if they've been censored in any way, no |
| 19:49.35 | *** join/#asterisk linlin (n=linlin@c-67-184-159-30.hsd1.il.comcast.net) |
| 19:49.36 | [TK]D-Fender | sevard : REMOVE TH WHITESPACE |
| 19:49.50 | sevard | between astdb and = and CLIENTS? |
| 19:49.51 | [TK]D-Fender | "astdb=CLIENTS/SIP-1338=pimpninjas |
| 19:49.55 | sevard | kk |
| 19:50.41 | smackus | question... when upgrading the polycom provisioning files, ie phone1.cfg and sip.ld and such... what files do I have to mess with. Can I just add the new sip.ld file and call it good? |
| 19:51.00 | [TK]D-Fender | smackus : Should have sip.ver as well. |
| 19:51.01 | sevard | sip reload, database show CLIENTS, nothing |
| 19:51.04 | smackus | ok |
| 19:51.07 | smackus | thanks |
| 19:51.15 | [TK]D-Fender | sev, do another exten text |
| 19:51.43 | sevard | still returns 0 |
| 19:51.55 | [TK]D-Fender | sevard : more complete pastebing of sip/extensions/CLI |
| 19:52.26 | *** join/#asterisk TommyTheKid (n=tommythe@mpk-edge.cto.sunit.net) |
| 19:52.49 | sevard | hang on. |
| 19:53.06 | *** part/#asterisk ariel_ (n=ariel_@74.8.35.2) |
| 19:54.16 | TommyTheKid | Having a slight problem with IAX.. I have a trunk between two hosts. One of the two hosts has multiple IP addresses. It appears like IAX is sourcing the connection from the primary system IP address and not the address it receive the connection on. |
| 19:54.42 | TommyTheKid | the first (old) server says that the new one is unreachable because of this (I think) |
| 19:56.10 | vader-- | ok i made the box to do the poe |
| 19:56.15 | vader-- | it powers two phones |
| 19:56.17 | vader-- | no problem |
| 19:56.26 | vader-- | when i plug the access point in it no power up |
| 19:56.28 | vader-- | but |
| 19:56.42 | vader-- | if i plug the access point in before plugging the phones in and then plug the phones in it works |
| 19:56.43 | vader-- | weird |
| 19:57.39 | [TK]D-Fender | vader-- : high initial current draw overload |
| 19:59.16 | *** join/#asterisk TrixVox (n=Spy007@c-69-248-121-104.hsd1.nj.comcast.net) |
| 19:59.39 | vader-- | ya |
| 19:59.46 | vader-- | but all 3 are working |
| 19:59.47 | vader-- | :) |
| 19:59.48 | vader-- | yay |
| 19:59.52 | vader-- | i smert |
| 20:00.21 | smackus | ok, so I have changed those files and I am still getting a config error. 0x2040 |
| 20:01.02 | smackus | what I read is that it is looking for a bootrom.ld. isnt that the same as sip.ld? |
| 20:01.19 | sevard | pastebin fails: http://nopaste.snit.ch:8001/7711 |
| 20:03.36 | *** join/#asterisk Egonis (n=chultay@207.245.14.10) |
| 20:04.09 | Egonis | I just got a Sangoma A200 w/ Hardware Echo Cancel, but when I enable it, I get no audio -- the /dev/wp1ec device node exists, should I change it's ownership? |
| 20:04.12 | *** join/#asterisk lirakis (n=lirakis@ool-45775a5e.dyn.optonline.net) |
| 20:04.14 | lirakis | hello hello |
| 20:04.56 | lirakis | ... ive got moh setup.. and in my console.. it shows that it trys to start moh.. then it gives me some output about trying to schedule some thing in the past.. then it immediatly terminates moh |
| 20:05.00 | lirakis | NOTICE[10430]: res_musiconhold.c:511 monmp3thread: Request to schedule in the past?!?! |
| 20:05.05 | lirakis | that is the exact line i get |
| 20:06.29 | TripleFFFF | mpg123 |
| 20:06.30 | TripleFFFF | does that |
| 20:06.37 | TripleFFFF | use madplay |
| 20:06.42 | TripleFFFF | and uninstall mpg123 |
| 20:06.46 | TripleFFFF | next ! |
| 20:10.40 | [TK]D-Fender | sevard : those only get reset if you place a CALL. |
| 20:10.54 | russellb | lirakis: install ztdummy |
| 20:11.11 | sevard | [TK]D-Fender: to or from the SIP line in question? |
| 20:11.15 | *** join/#asterisk Johnnie (n=john@pdpc/supporter/active/Johnnie) |
| 20:12.08 | [TK]D-Fender | sevard : from IIRC |
| 20:12.26 | sevard | well it's not getting set when i dial that extension from 1338 |
| 20:12.28 | sevard | sooo... |
| 20:12.29 | [TK]D-Fender | lirakis : SCrew both of those and use native MoH. |
| 20:12.38 | [TK]D-Fender | sevard : SHOW |
| 20:13.04 | smackus | [TK]D-Fender: bootrom.ld and sip.ld are the same? |
| 20:13.07 | sevard | show what? |
| 20:13.15 | [TK]D-Fender | smackus : NO. |
| 20:13.22 | [TK]D-Fender | sevard : the call being placed. |
| 20:13.29 | sevard | oh, alright |
| 20:13.38 | smackus | what i thought... but bootrom was not in the zip file |
| 20:14.54 | sevard | http://nopaste.snit.ch:8001/7712 |
| 20:15.33 | [TK]D-Fender | smackus : Correct. SIP application & BootROM are 2 entirely seperate packages |
| 20:16.15 | sevard | unless they're only set after Answer(), i don't see the problem |
| 20:19.48 | [TK]D-Fender | sevard : do you see ti in "database show"? |
| 20:19.51 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 20:20.00 | a | dude, a is a tight nick. |
| 20:20.05 | Dr-Linux | a |
| 20:20.14 | Strom_C | a |
| 20:20.28 | a | during or after the call? |
| 20:20.32 | rob0 | ~nick a |
| 20:20.41 | Strom_C | but now your terminal will ding every time someone says a word |
| 20:20.42 | [TK]D-Fender | a : after |
| 20:20.46 | [TK]D-Fender | a : as in NOW. |
| 20:20.57 | a | negative |
| 20:20.59 | Dr-Linux | a |
| 20:21.10 | Dr-Linux | ~a |
| 20:21.12 | jbot | extra, extra, read all about it, a is not b |
| 20:21.13 | [TK]D-Fender | Strom_C : He clearly isn't as self-absorbed as you ;) |
| 20:21.33 | Cresl1n | a = b |
| 20:21.38 | Cresl1n | not it is :-D |
| 20:21.39 | rob0 | a lot of things will be falsely flagged as being addressed to "a" |
| 20:21.44 | Strom_C | [TK]D-Fender: ? |
| 20:21.45 | Cresl1n | *now |
| 20:21.46 | Cresl1n | :-) |
| 20:21.48 | sevard | stfu. |
| 20:22.01 | Dr-Linux | lol |
| 20:22.47 | sevard | pwned |
| 20:22.57 | [TK]D-Fender | Strom_C : For having his name all highlighted and alarms, bells, whistles, and fireworks at the mere mention of your name :) |
| 20:23.08 | Strom_C | heh |
| 20:23.20 | Dr-Linux | wow today i have 36 contact online in MSN |
| 20:23.43 | sevard | Dr-Linux: fuck you, child of bitch -- what do you think about?! |
| 20:24.21 | smackus | sevard: easy |
| 20:25.22 | Dr-Linux | sevard: your sister is also onilne with me, she is nude on cam.. she has sexy cunt .. woww so please DND |
| 20:25.40 | sevard | that'd work if i had a sister. |
| 20:25.44 | sevard | you must have my grandmother |
| 20:25.45 | sevard | enjoy. |
| 20:25.53 | Strom_C | sevard: will you please cool it |
| 20:26.06 | sevard | Strom_C: chillax dude. |
| 20:26.18 | Dr-Linux | sevard: what the fuck with you?? huh |
| 20:26.19 | sevard | [TK]D-Fender: what do you think |
| 20:26.40 | file | both of you, Dr-Linux & sevard, cool it |
| 20:26.47 | Juggie | yah really. |
| 20:26.50 | Juggie | wtf do you think this is. |
| 20:26.53 | Dr-Linux | file: why he is abused me first???? |
| 20:26.56 | Cresl1n | for shizzle, that's what I say |
| 20:27.00 | sevard | nobody is angry at anyone, we're fucking around - relax :) |
| 20:27.05 | Qwell[] | Cresl1n: fo :) |
| 20:27.06 | Dr-Linux | i didn't come here for that, |
| 20:27.10 | Juggie | this isnt 'the fucking aronud' channel |
| 20:27.10 | file | there's other channels to do that in |
| 20:27.12 | Juggie | this is about asterisk |
| 20:27.14 | Cresl1n | yeah |
| 20:27.15 | smackus | some of us dont appreciate it. |
| 20:27.18 | Cresl1n | #drama for one |
| 20:27.21 | sevard | anyyyway |
| 20:27.23 | sevard | [TK]D-Fender: what do you think |
| 20:27.30 | Dr-Linux | if he is drunk or what then he should go and fuck her month, but he shouldn't abuse someone :@ |
| 20:27.34 | [TK]D-Fender | sevard : not sure, but IG@G for now. back considerably later... |
| 20:27.39 | file | Dr-Linux: stop |
| 20:28.14 | [TK]D-Fender | file : Gimme Ops ;) |
| 20:29.05 | file | [TK]D-Fender: :D |
| 20:29.11 | sevard | hahahaha |
| 20:29.20 | [TK]D-Fender | (actually asking for it seems wierd.... and questionably appropriate from a business perspective though not a moral one) |
| 20:29.58 | [TK]D-Fender | ORLY? |
| 20:30.03 | sevard | YARLY |
| 20:30.12 | Strom_C | http://orlyguide.ytmnd.com/ |
| 20:30.33 | [TK]D-Fender | ok, but seriously I'm off.. later all! |
| 20:30.55 | file | Strom_C: Strommmy Boy |
| 20:31.08 | JohnJacob | freeswitch |
| 20:31.21 | Strom_C | JohnJacob: dont start |
| 20:31.25 | Strom_C | file: file file |
| 20:33.15 | Dr-Linux | ~sphinx |
| 20:33.16 | jbot | Sphinx is a speaker-independent large vocabulary continuous speech recognizer under Berkeley's style license. http://sourceforge.net/projects/cmusphinx/ http://cmusphinx.sourceforge.net/html/cmusphinx.php http://www.speech.cs.cmu.edu/sphinx/Sphinx.html |
| 20:37.15 | TripleFFFF | whats advatages of using ael ? |
| 20:37.22 | *** join/#asterisk n9urk (n=leonard@user-0ce2dhc.cable.mindspring.com) |
| 20:37.36 | MikeJ[Laptop] | TripleFFFF, it's easier to look at |
| 20:37.51 | TripleFFFF | ah |
| 20:38.08 | TripleFFFF | well a pretty girl too and that dont make it into the cvs ;) |
| 20:38.28 | TripleFFFF | yeah got sphynx working here.. in fact ill recompile all that crap on my laptop |
| 20:39.00 | *** join/#asterisk js_78743 (n=me@63.172.175.147) |
| 20:39.03 | TripleFFFF | seems some clients cant tell what to press even if theres only 2 options... so ill accept help ,help me , and help me m#$%!@# ucker |
| 20:39.37 | n9urk | can someone help me with getting cdr logging my mysql set up? I have installed unixodbc and have downloaded mysqlodbc and cannot figure out where odbcinst.ini is. Can anyone help? I have been following the instructions on the voip-info wiki |
| 20:39.56 | TripleFFFF | hmmm locate is your friend |
| 20:40.02 | TripleFFFF | normally /etc/odbcinst.ini |
| 20:40.17 | TripleFFFF | why not use mysql module instead of odbc ? |
| 20:40.17 | Dr-Linux | TripleFFFF: sphynx? |
| 20:40.32 | TripleFFFF | i had prolblems in the past on lots of coonenect |
| 20:40.40 | TripleFFFF | Dr-Linux yeah |
| 20:40.59 | TripleFFFF | well Dr-Linuxsphinx |
| 20:41.02 | Dr-Linux | TripleFFFF: what's sphynx? |
| 20:41.04 | Dr-Linux | aww |
| 20:41.11 | TripleFFFF | its same app.. with a typo |
| 20:41.12 | TripleFFFF | ;) |
| 20:41.16 | Dr-Linux | that's what i'm looking for since last month |
| 20:41.41 | n9urk | TripleFFFF: The wiki isn't the best linked around. can you send me a link to using the MYSQL mod? I would much rather use that mod but couldn't figure it out |
| 20:41.45 | Dr-Linux | TripleFFFF: can i PM you? |
| 20:41.52 | TripleFFFF | hehe works nice.. basically you make a server listen to packets.. client sends packets to it ( audio ) it interprets and send a string of what recognized back to client |
| 20:41.57 | TripleFFFF | not really |
| 20:42.07 | TripleFFFF | Dr-Linux its hell to compile etc |
| 20:42.16 | TripleFFFF | but since im redoing ,.,. ;) |
| 20:42.40 | Dr-Linux | TripleFFFF: i can't find any document for it on net |
| 20:42.48 | Dr-Linux | TripleFFFF: can you point me any? |
| 20:43.08 | Dr-Linux | TripleFFFF: i compiled new version 3.x |
| 20:43.19 | Dr-Linux | but still not understanding it |
| 20:43.34 | TripleFFFF | hmm .. mybraindump --all-data --table=sphinx-location >> braindump.tql <-- .tql extensions is new. .its the triplefff query language |
| 20:43.43 | TripleFFFF | i use 2 |
| 20:43.50 | TripleFFFF | 3+ is .. hmmm too slow |
| 20:43.53 | TripleFFFF | but nice recog paterns |
| 20:43.57 | TripleFFFF | just too slow for voip |
| 20:44.05 | sevard | so what does everyone think |
| 20:44.12 | TripleFFFF | v2 has like 1 sec delay . 3+ 3-4-5-6 sec |
| 20:44.13 | sevard | i can wait an underdermined amount of time for a 42 inch plasma |
| 20:44.16 | sevard | and i've been waiting since may |
| 20:44.23 | sevard | or i can get a 37 inch hdtv LCD right now |
| 20:44.24 | TripleFFFF | get one now |
| 20:44.28 | TripleFFFF | plasma |
| 20:44.32 | TripleFFFF | depends on room lights |
| 20:44.35 | sevard | keep waiting for the huge crystal clear plasma? or settle for the 5 inch smaller LCD righ tnow |
| 20:44.37 | TripleFFFF | where is it gonna be ? |
| 20:44.38 | sevard | both HDTV both widescreen |
| 20:44.42 | sevard | in my living room |
| 20:44.48 | TripleFFFF | light from windows on it ? |
| 20:44.50 | froguz | my FXO gateway has allways activated the VAD for G.729. is there a way i can make asterisk support this? |
| 20:44.51 | TripleFFFF | if so plasma |
| 20:44.51 | sevard | moderatly light some glare sometimes |
| 20:45.01 | TripleFFFF | if no lights at all.. (sun/etc) lcd if you can afford |
| 20:45.06 | sevard | i have a 19inch crt curtis mathis |
| 20:45.14 | sevard | it's free |
| 20:45.16 | sevard | i won it |
| 20:45.16 | TripleFFFF | g;are+lcd = bad |
| 20:45.22 | TripleFFFF | won from where |
| 20:45.23 | n9urk | can anyone send me a link or more info on loading cdr_mysql? |
| 20:45.27 | TripleFFFF | i want to win one too |
| 20:45.30 | sevard | i'm talking about tv clarity, i've seen LCDs in the stores and they blow |
| 20:45.33 | TripleFFFF | voipinfo type cdr mysql |
| 20:45.45 | TripleFFFF | sevard but on glares youll see nothing |
| 20:45.55 | sevard | the plasma is like worth 2400 and the LCD is ~1200 |
| 20:45.58 | Dr-Linux | n9urk: what you need about cdr_mysql? |
| 20:46.04 | TripleFFFF | where you get it from ? |
| 20:46.08 | sevard | everyfreegift.com |
| 20:46.20 | Dr-Linux | n9urk: that's in asterisk addons |
| 20:46.21 | sevard | i've been waiting since may for them to 'get it in stock' |
| 20:46.23 | TripleFFFF | you need to give bJ's for it ? |
| 20:46.27 | sevard | but seriously they're at *#$&^@ing walmart. |
| 20:46.35 | n9urk | Dr-Linux: what do I need to do to get it to load? |
| 20:46.38 | sevard | TripleFFFF: nah, i'm already on the master list. |
| 20:46.52 | n9urk | Dr-Linux: the wiki doesn't really tell how to load it. |
| 20:47.06 | TripleFFFF | load => res_mysql.so |
| 20:47.19 | Nugget | http://www.diyturbo.net/ot/walmart.jpg |
| 20:47.29 | Dr-Linux | n9urk: just compile asterisk addons and copy the mysql conf file to /etc/asterisk/ dir |
| 20:47.42 | TripleFFFF | oh everyfreegift.com is a spam me i love it thing ok |
| 20:47.45 | TripleFFFF | never mind |
| 20:48.07 | sevard | yeah, it's all a scam unless you know how to play the game |
| 20:48.18 | sevard | but i played and i just want my damn tv. |
| 20:48.37 | sevard | so would you keep waiting for a plasma or get the 5 inch smaller lcd right now |
| 20:49.06 | n9urk | Dr-Linux: thanks. Where do I get it? |
| 20:49.16 | *** join/#asterisk jgoo (n=foo@ppp200-161.adsl.forthnet.gr) |
| 20:49.26 | jgoo | okily doke, all working, w00t |
| 20:49.39 | Dr-Linux | n9urk: do you have asterisk-addons source? |
| 20:49.43 | sevard | Nugget: that's priceless. |
| 20:49.43 | TripleFFFF | sevard tell us man; ) |
| 20:49.45 | jgoo | now, specifics of SIP sertup for this Xlite softphone, I get a 408 / 404... |
| 20:49.49 | TripleFFFF | just signup then cancel ? |
| 20:49.55 | TripleFFFF | signup call MC card stolen ? |
| 20:49.57 | TripleFFFF | what what |
| 20:50.00 | *** join/#asterisk IvyUK (n=mark@194.201.148.132) |
| 20:50.03 | Dr-Linux | n9urk: if not, then get one from asterisk.org |
| 20:50.23 | TripleFFFF | jgoo make sure your xten is passing username in displayname |
| 20:50.25 | Dr-Linux | and compile it |
| 20:50.33 | jgoo | ok it worked, I chenged domain to proxy |
| 20:50.34 | sevard | TripleFFFF: a friend of mine gave me his 'debit' card, which was a giveway from boing something.com i have the card at home and can tell you where he got it, anyway, it's a gift card that's in the form of a debit card not tied to anyone |
| 20:50.43 | jgoo | erm, waht is default dial out code? |
| 20:50.56 | jgoo | also - where do you set the country codes? |
| 20:51.00 | TripleFFFF | and |
| 20:51.02 | jgoo | (for which dial tones) |
| 20:51.16 | TripleFFFF | you subed on all things on that 300$ filled card.. then they will rebill and get niet |
| 20:51.23 | TripleFFFF | so get your prize while they dont know it |
| 20:51.24 | sevard | it had 25 bucks on it and all the offers cost 23.75 to sign up |
| 20:51.36 | TripleFFFF | sevard pm me 10-20 on card |
| 20:51.45 | sevard | i can't refill the card |
| 20:51.54 | n9urk | Dr-Linux: Aha. I am dling it now. Thanks |
| 20:51.57 | sevard | oh, 10-20, heh |
| 20:52.03 | sevard | i'll do that when i get home |
| 20:52.03 | TripleFFFF | location ; |
| 20:52.05 | TripleFFFF | k |
| 20:52.09 | *** join/#asterisk carl0s- (n=carl0s@compsup.demon.co.uk) |
| 20:52.25 | TripleFFFF | www.boingboing.net/ |
| 20:53.03 | jm|work | `Kevin: |
| 20:53.10 | jm|work | are you cursor.biz man? |
| 20:54.01 | n9urk | How do I get * add ons for 1.2.5 ? When I downloaded asterisk-addons-1.2-current.tar.gz and tar -xzf'd it it came out saying 1.2.3.? |
| 20:54.10 | js_78743 | <PROTECTED> |
| 20:54.35 | TripleFFFF | sip and sip ? |
| 20:55.18 | TripleFFFF | just make sip context on both to incoming .. then in extensions .conf ad d[incoming] and in there add _X.,1,dial(SIP/${EXTEN}@OTHERPROVIDERSIPNAME) |
| 20:55.30 | Dr-Linux | n9urk: do not care about that |
| 20:55.32 | TripleFFFF | even s,1,dial |
| 20:55.35 | rob0 | n9urk: http://www.asterisk.org/ ... see "Asterisk Downloads" |
| 20:55.41 | js_78743 | yes, both SIP. I'm trying to prove to the provider that their SIP isn't working (or at least isn't SIP). It works fine with my 941 |
| 20:56.08 | TripleFFFF | who might provider be |
| 20:56.15 | jgoo | OK so it seems my outbound lines are not setup, hrm |
| 20:56.18 | n9urk | Dr-Linux: ok addons for 1.2.3 == addons for 1.2.5 ? |
| 20:56.33 | jgoo | also, anyone use linphone applet?? any good? I want to reskin if possible |
| 20:56.39 | js_78743 | Zingotel ... |
| 20:56.44 | TripleFFFF | never hear |
| 20:56.45 | Dr-Linux | n9urk: the same as your asterisk version |
| 20:57.03 | Dr-Linux | n9urk: but don't care, just have one |
| 20:57.19 | *** join/#asterisk hohum (n=dcorbe@12.195.58.235) |
| 20:57.31 | sevard | i'm taking the LCD |
| 20:57.35 | js_78743 | I've been using the default context across the board and am just trying to Playback(hello-world) on in-bound |
| 20:57.42 | websae | go #Freeswitch :) |
| 20:58.32 | n9urk | Dr-Linux: Ok thanks. am I missing something there is not a configure file in the tar |
| 20:58.36 | n9urk | ? |
| 20:58.51 | sevard | done. |
| 20:58.55 | sevard | mplayer? |
| 20:58.58 | sevard | bah |
| 20:59.17 | jgoo | oh ok so I just plugged my line from S0 on the ISDN box to my HFC card, and I got a shyteload of 'empty HDLC frame to bad CRC received' messages come up on my console |
| 20:59.40 | *** join/#asterisk nicchap (n=nicchap@216.209.85.2) |
| 20:59.41 | *** join/#asterisk sponix (i=family@host-64-72-46-149.classicnet.net) |
| 20:59.57 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 21:00.14 | Dr-Linux | n9urk: didn't you compile it? |
| 21:00.37 | Dr-Linux | tar -zxvf ..asterisk-addons.....tar.gz |
| 21:00.41 | Dr-Linux | make |
| 21:00.42 | n9urk | Dr-Linux: How do I compile it. there is no configure file |
| 21:00.46 | Dr-Linux | make install |
| 21:01.09 | n9urk | Dr-Linux: does it not need to configure? |
| 21:01.49 | Dr-Linux | n9urk: when compilation done, you will get a config file |
| 21:02.03 | Dr-Linux | then just copy and paste it in the /etc/asterisk/ dir |
| 21:02.11 | Dr-Linux | then configure it |
| 21:04.13 | n9urk | Dr-Linux: Thanks for the help I think I have it compiled now |
| 21:04.37 | Dr-Linux | n9urk: cool |
| 21:06.28 | *** join/#asterisk foo (n=foo@unaffiliated/foo) |
| 21:06.37 | n9urk | Dr-Linux: then do I need to load a module in the * console? |
| 21:06.56 | *** part/#asterisk foo (n=foo@unaffiliated/foo) |
| 21:07.06 | Dr-Linux | n9urk: yes do it |
| 21:07.16 | n9urk | Dr-Linux: how do I do it? |
| 21:07.31 | Dr-Linux | n9urk: you need to put mysql info in the mysql .conf file to make it wor |
| 21:07.34 | Dr-Linux | work |
| 21:08.26 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 21:08.27 | n9urk | Dr-Linux: you mean the cdr_mysql.conf? I did that. from the console do i need to do this "load cdr_mysql.so"? |
| 21:08.55 | Dr-Linux | yes |
| 21:09.25 | Dr-Linux | n9urk: you may have somehting like cdr_addons_mysql |
| 21:12.41 | n9urk | Dr-Linux: got it. I got this error: "Jul 19 17:12:16 ERROR[5255]: cdr_addon_mysql.c:437 my_load_module: Failed to connect to mysql database asterisk on localhost." |
| 21:13.12 | lirakis | i totally fuxored my pbx messing with moh.. he he |
| 21:13.28 | Dr-Linux | n9urk: your asterisk is not connecting to mysql DB |
| 21:14.09 | lirakis | n9urk: what is your cdr_mysql conf like? |
| 21:14.27 | n9urk | Dr-Linux: I figured that much. I have the db up in phpmyadmin. I have the username and pw right |
| 21:14.44 | n9urk | hostname |
| 21:14.46 | n9urk | dbname |
| 21:14.47 | lirakis | n9urk: make sure you have the right host specified.. i e 127.0.0.1 . also the domain must be right for your mysql user.. |
| 21:14.54 | n9urk | table |
| 21:14.55 | n9urk | username |
| 21:14.57 | n9urk | password |
| 21:14.58 | n9urk | port |
| 21:14.59 | n9urk | sock |
| 21:15.02 | n9urk | userfield |
| 21:15.17 | n9urk | I have hostname=localhost |
| 21:15.19 | lirakis | n9urk: nurk wtf are you talking about... stop posting for no reason |
| 21:15.27 | lirakis | n9urk: try 127.0.0.1 |
| 21:15.34 | TripleFFFF | ~tell enter |
| 21:15.40 | Dr-Linux | n9urk: comment the port line then check |
| 21:15.47 | n9urk | lirakis: ummmmmmmm wtf are you talking about? |
| 21:15.48 | TripleFFFF | ~tell about enter key |
| 21:16.44 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 21:16.48 | lirakis | n9urk: if you dont want to play nice then just go read this on your own.. http://www.voip-info.org/wiki-Asterisk+cdr+mysql |
| 21:17.08 | lirakis | n9urk: i just set mine up today without any trouble by following those directions |
| 21:17.41 | n9urk | lirakis: thanks for the link |
| 21:18.18 | n9urk | Dr-Linux: I commented out the port line and that didn't work |
| 21:18.29 | Dr-Linux | n9urk: just paste your cdr_mysql.conf file in my pvt |
| 21:19.12 | *** join/#asterisk trbldwine (n=trbldwin@adam.ur.northwestern.edu) |
| 21:19.41 | lirakis | humm... i guess i didnt fnark up my pbx.. a reboot did the trick |
| 21:20.07 | Dr-Linux | n9urk: hhm.. i saw your config |
| 21:20.15 | Dr-Linux | n9urk: just put that >> sock=/var/lib/mysql/mysql.sock |
| 21:20.24 | Dr-Linux | and uncomment last line as well |
| 21:21.02 | Dr-Linux | n9urk: then "reload" and check the cdr status on cli |
| 21:21.19 | n9urk | still not connected Dr-Linux |
| 21:21.41 | Dr-Linux | hhm.. |
| 21:21.45 | n9urk | Dr-Linux: I have connected to mysql via the cli with the username and pw in the conf file |
| 21:22.09 | Dr-Linux | n9urk: change hostname=localhost |
| 21:22.32 | n9urk | Dr-Linux: that didn't work |
| 21:23.01 | Dr-Linux | n9urk: did you reload the module? |
| 21:23.19 | n9urk | Dr-Linux: yes I did |
| 21:23.41 | Dr-Linux | n9urk: what distor you are using, and what's mysql version? |
| 21:23.51 | TripleFFFF | sphinx guy u there ? |
| 21:23.51 | Dr-Linux | it looks like problem with your DB setting |
| 21:23.56 | n9urk | Dr-Linux: Gentoo |
| 21:24.15 | n9urk | Dr-Linux: mysql 4.1.14 |
| 21:24.23 | Dr-Linux | TripleFFFF: i'm also looking for a sphinx guy , but never find one |
| 21:24.29 | *** join/#asterisk s0lid (n=s0lid@124.6.176.99) |
| 21:24.30 | lirakis | hrmm.. i got moh to work |
| 21:24.33 | lirakis | but it sounds like buthole |
| 21:24.35 | lirakis | ha ha |
| 21:24.47 | Dr-Linux | n9urk: hhm.. try update your mysql users password |
| 21:25.29 | n9urk | Dr-Linux: what do you mean? I am able to log into mysql from the cli with the username and pass in my cdr_mysql.conf file |
| 21:26.50 | Dr-Linux | n9urk: please make sure about mysql.sock path |
| 21:27.05 | Dr-Linux | i don't know about gentoooo |
| 21:27.53 | lirakis | Dr-Linux: are you referring to my install? |
| 21:28.38 | Dr-Linux | lirakis: i'm talking to n9urk |
| 21:29.01 | lirakis | Dr-Linux: im still using mpg123 .. im emerging madplay right now.. i am making the assumtion it can play a stream the same way.. (im streaming di.fm for moh) .. although i have tried the default moh setup too and it sounds terrible over the phone also. |
| 21:29.03 | n9urk | Dr-Linux: I think that is going to fix it |
| 21:29.05 | TripleFFFF | ahahahah sphinx up and running |
| 21:29.07 | lirakis | .. oh i see.. i am running gentoo as well |
| 21:29.11 | TripleFFFF | total time to get up 4 minutes |
| 21:29.21 | TripleFFFF | # ./client.pl yes.gsm |
| 21:29.21 | TripleFFFF | FTYPE: gsm |
| 21:29.21 | TripleFFFF | Result: YES |
| 21:29.33 | TripleFFFF | remote server : SERVER RESULT: YES |
| 21:29.39 | n9urk | Dr-Linux: hmmmm, that didn't do it |
| 21:30.11 | TripleFFFF | now making asterisk to work with that .. so i can call home and say. ( is anyone in my home.. ) will check microphone activity and answer ;) yes or no |
| 21:30.15 | *** join/#asterisk TeePOG (n=arno@dsl-145-155-145.telkomadsl.co.za) |
| 21:31.36 | Dr-Linux | n9urk: hhm... i have done that alot of time even mysql server on remote end , never had a problem |
| 21:32.19 | Dr-Linux | n9urk: can i paste you my mysql .conf file in your pvt? |
| 21:32.27 | n9urk | Dr-Linux: please do |
| 21:32.31 | TripleFFFF | linux stil need phinx help ? |
| 21:32.56 | Dr-Linux | TripleFFFF: yes |
| 21:33.06 | TripleFFFF | hehe got my second server up |
| 21:33.20 | TripleFFFF | only 267 words recognize |
| 21:33.24 | TripleFFFF | but still |
| 21:33.29 | TripleFFFF | recongnizxed laugh and coughs too |
| 21:33.56 | n9urk | Dr-Linux: is it 'username' or "user" |
| 21:33.57 | n9urk | ? |
| 21:34.08 | TripleFFFF | both |
| 21:34.25 | Dr-Linux | n9urk: user |
| 21:34.29 | n9urk | changing username to user seems to have fixed it |
| 21:34.37 | n9urk | thanks for your help |
| 21:34.45 | Dr-Linux | n9urk: yeah please try |
| 21:34.56 | n9urk | Dr-Linux: looks like it connected |
| 21:35.05 | n9urk | Dr-Linux: I will make a call and see |
| 21:35.36 | Dr-Linux | n9urk: ok good to know |
| 21:35.48 | Dr-Linux | save those setting with you for the future |
| 21:36.02 | *** join/#asterisk tdonahue-laptop (n=tdonahue@vonmail.vonworldwide.com) |
| 21:37.21 | n9urk | Dr-Linux: it is working now. Thanks for the help |
| 21:37.44 | Dr-Linux | n9urk: cool |
| 21:41.46 | lirakis | is it still required that you have "[class]" verbatim .. some where in your music on hold config? my sample doesnt have it .. i replace 'class' with my class name .. that is correct right? |
| 21:42.45 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 21:44.07 | file | ~centosbug |
| 21:44.10 | jbot | extra, extra, read all about it, centosbug is a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h" |
| 21:44.34 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 21:44.40 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 21:45.53 | *** join/#asterisk Pazzo (n=thomas@host130-250-static.72-81-b.business.telecomitalia.it) |
| 21:46.31 | *** join/#asterisk h3x0r (i=hex@ip70-189-236-254.lv.lv.cox.net) |
| 21:48.56 | *** part/#asterisk smackus (n=ckwall@63.149.122.93) |
| 21:50.51 | Dr-Linux | file: the command given for redhatbug doesn't work |
| 21:51.12 | lirakis | is there a way to enable moh for all extensions?? |
| 21:51.42 | TripleFFFF | what thats that bug ? |
| 21:51.45 | *** part/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 21:51.47 | Dr-Linux | lirakis: why are you so so AFTER moh? |
| 21:52.09 | lirakis | Dr-Linux: i am just playing around with a pbx in my basement to learn about asterisk |
| 21:52.14 | lirakis | .. right now im just playing with moh |
| 21:52.16 | TripleFFFF | spinlock is what ? |
| 21:52.24 | TripleFFFF | ~centosbug |
| 21:52.26 | jbot | rumour has it, centosbug is a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h" |
| 21:53.12 | Dr-Linux | lirakis: then remove mp123 player and use asterisk native player, anthm or someone developed |
| 21:53.18 | TripleFFFF | oh for zaptel |
| 21:53.30 | TripleFFFF | likaris or madplay |
| 21:53.44 | lirakis | i have emerged madplay |
| 21:53.52 | lirakis | but i should totally get rid of mpg123?? |
| 21:54.09 | TripleFFFF | yes |
| 21:54.09 | Dr-Linux | yes |
| 21:54.12 | TripleFFFF | uninstall |
| 21:54.16 | TripleFFFF | some libs make that |
| 21:54.53 | TripleFFFF | why no accoutn number sound ? |
| 21:55.07 | TripleFFFF | so we can you .. please-enter-your.gsm then account-number.gsm |
| 21:55.10 | TripleFFFF | but no accountnumber |
| 21:55.58 | *** join/#asterisk MatsK (i=MatsK@83.233.97.229) |
| 21:56.13 | TripleFFFF | got it |
| 21:56.31 | TripleFFFF | press-enter.gsm your-account.gsm number.gsm |
| 21:56.34 | TripleFFFF | darn |
| 21:56.47 | TripleFFFF | oh no its press .. darn anyone knowo what i gen bu ? |
| 21:57.24 | lirakis | okay.. im unmerging mpg123 .. thanks for the advice |
| 22:00.55 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 22:07.08 | Zodiacal | what should i know before buying a polycom soundpoint 601. it says it comes with sip. will i be able to upgrade that firmware? will polycom give me access to the firmware? |
| 22:07.21 | Zodiacal | do i need a service contract with polycom like cisco? |
| 22:07.33 | Dr-Linux | no |
| 22:07.46 | Zodiacal | the firmware upgrades are public? |
| 22:08.05 | Zodiacal | dr-linux no to all those questions? |
| 22:08.07 | Zodiacal | :) |
| 22:08.31 | hads | Zodiacal: You will need to get the firmware from an authorised reseller. |
| 22:08.44 | *** join/#asterisk enmaca (n=enmaca@200.53.44.19) |
| 22:08.54 | hads | but you don't need a contract. |
| 22:09.18 | Zodiacal | does polycom list their authorized reselers? |
| 22:09.52 | Qwell[] | The person who sold you the phone would be an authorized reseller... |
| 22:10.06 | hads | Unsure, just ask the reseller you are thinking about buying it off if they can supply you the firmware. If not, buy it somewhere else. |
| 22:10.43 | Zodiacal | i was just going to buy one off amazon to try it out... anyone know of a better place? |
| 22:10.49 | Zodiacal | amazon = $250 |
| 22:11.18 | Zodiacal | even "like new" |
| 22:12.30 | Zodiacal | does mgcp offer a lot more features than sip? |
| 22:13.05 | jarrod | no |
| 22:13.07 | *** join/#asterisk Agrajag- (n=filip@c211-30-4-5.artrmn1.nsw.optusnet.com.au) |
| 22:13.08 | jarrod | it definitely does not |
| 22:13.13 | Zodiacal | okie |
| 22:13.14 | jarrod | and it is much older |
| 22:13.18 | jarrod | and the asterisk mgcp code is crap |
| 22:13.20 | Zodiacal | thanks guys i'll give it a try! |
| 22:13.25 | Zodiacal | brb |
| 22:16.20 | Synyn | afternoon guys |
| 22:16.42 | Dr-Linux | good morning |
| 22:16.56 | drray | the polycoms are ugly |
| 22:16.58 | drray | imo |
| 22:16.59 | *** join/#asterisk tempest1 (n=Brett@adsl-144-61-127.chs.bellsouth.net) |
| 22:17.02 | drray | pardon me |
| 22:17.12 | Synyn | anyone using the OEM X100P cards? |
| 22:18.14 | Synyn | just got me 2 for 30 bux, damn cheap, I hope they are not really that cheap ) |
| 22:19.38 | Dr-Linux | digium x100p comes with $25? |
| 22:19.58 | *** join/#asterisk nortex (n=breeves@snapper.titanspecialties.com) |
| 22:20.04 | nortex | Got a polycom question, does anyone know of a way in the configuration files to have the phones keep the ringer volume level through a reboot? I have the other volumes taken care of, but I cannot find how to set this in the admin guide. |
| 22:20.49 | *** join/#asterisk Egonis (n=chultay@207.245.14.10) |
| 22:21.18 | Zodiacal | drray what do u prefer? |
| 22:21.22 | Egonis | I have a Sangoma A200 w/ HW Echo Cancel but when I enable it, there is no audio -- any ideas? I am using the latest Wanpipe |
| 22:22.04 | drray | Zodiacal - I like cisco, it's the ultimate pretty girlfriend. Treats you like garbage, but looks better than someone that would treat you better |
| 22:22.18 | Zodiacal | haha |
| 22:22.26 | Zodiacal | ciscos are getting on my nervs.. |
| 22:22.30 | Dr-Linux | Zodiacal: really i have Cisco and polycom on same desk, but cisco one is nice |
| 22:23.23 | drray | I don't like cisco one lick |
| 22:23.28 | drray | but the phone just works |
| 22:23.35 | drray | once you jump through all the hoops |
| 22:23.55 | Zodiacal | sip doesn't have hints |
| 22:24.06 | Dr-Linux | drray: you don't like Cisco, you don't like Polycom, what you like then?? |
| 22:24.11 | Egonis | exit |
| 22:24.18 | *** part/#asterisk tempest1 (n=Brett@adsl-144-61-127.chs.bellsouth.net) |
| 22:24.22 | Dr-Linux | xstream? |
| 22:24.54 | Dr-Linux | s/xstream/grandstream |
| 22:24.57 | jm|work | 0800 dialup?! |
| 22:24.59 | jm|work | that's going back a bit ;) |
| 22:26.02 | drray | I use cisco |
| 22:28.37 | enmaca | I there |
| 22:29.40 | TripleFFFF | .install Asterisk::AGI |
| 22:29.44 | TripleFFFF | how one installs this ? |
| 22:29.52 | TripleFFFF | what cpan name i mean lol |
| 22:30.04 | Strom_C | TripleFFFF: asterisk.gnuinter.net |
| 22:30.11 | Strom_C | download the tarball and follow the directions |
| 22:30.18 | TripleFFFF | tks |
| 22:30.19 | enmaca | its posible to asterisk generate sip calls on a specific interface via configuration option on sip.conf? |
| 22:31.36 | *** join/#asterisk Az_au (i=[imqgHiY@216.127.73.119) |
| 22:34.10 | *** join/#asterisk enjay- (n=enjay@71.216.165.97) |
| 22:34.32 | *** join/#asterisk tomlobato (n=tomlobat@201-68-70-211.dsl.telesp.net.br) |
| 22:35.22 | enjay- | Anyone know a site that could help me troubleshoot jitter issues.. |
| 22:36.34 | nortex | Anybody here use the penalty option in a queue? |
| 22:40.27 | *** join/#asterisk fnordus (n=dnall@s142-179-111-243.bc.hsia.telus.net) |
| 22:47.19 | *** join/#asterisk tomlobato (n=tomlobat@201-68-70-211.dsl.telesp.net.br) |
| 22:48.44 | *** join/#asterisk tomlobato (n=tomlobat@201-68-70-211.dsl.telesp.net.br) |
| 22:50.08 | *** join/#asterisk key2 (n=key2@gob75-2-81-56-64-17.fbx.proxad.net) |
| 22:52.02 | TripleFFFF | hj,,, |
| 22:52.04 | TripleFFFF | hmm |
| 22:52.34 | Dovid | any way to set a certaine ext. on the polycom to ring silently ? |
| 22:56.19 | Dovid | any way to set a certaine ext. on the polycom to ring silently ? |
| 22:57.28 | *** join/#asterisk rue_mohr (n=not@bdr2.fieldrd.scrd.ca) |
| 22:57.47 | rue_mohr | hello, I'd like to do something, I'm not sure quite how |
| 22:57.47 | TripleFFFF | ok i know why sphinxs doesnt work |
| 22:58.21 | rue_mohr | I have extensions 6402 and 6430, which I need to ring togethor, nomatter which one is called |
| 22:58.36 | rue_mohr | do I do that in extensions.conf? |
| 22:58.48 | TripleFFFF | what king of idiot would write this |
| 22:58.48 | TripleFFFF | while (defined(my $b = read $fh, my($buf), 4096)) { |
| 22:59.14 | Dovid | rue_mohr |
| 22:59.19 | rue_mohr | yes? |
| 22:59.21 | Dovid | let me understand |
| 22:59.31 | Dovid | u have 2 phones and when one ie clled u want the other to ring ? |
| 22:59.38 | rue_mohr | yes |
| 22:59.44 | Dovid | can u have one ext. that rings at both ? |
| 22:59.52 | enjay- | ring group |
| 22:59.53 | rue_mohr | hold wait |
| 23:00.07 | rue_mohr | dialing either number to ring both phones |
| 23:00.13 | Dovid | ok |
| 23:00.20 | *** join/#asterisk tomlobato (n=tomlobat@201-68-70-211.dsl.telesp.net.br) |
| 23:00.24 | Dovid | so create 2 new extensions that call them |
| 23:00.35 | Dovid | gona do it in paste bin |
| 23:00.37 | Dovid | and post here |
| 23:00.40 | *** join/#asterisk tomlobato (n=tomlobat@201-68-70-211.dsl.telesp.net.br) |
| 23:00.49 | rue_mohr | no, I need the old extentsions to ring both phones |
| 23:01.16 | *** join/#asterisk tomlobato (n=tomlobat@201-68-70-211.dsl.telesp.net.br) |
| 23:01.24 | rue_mohr | if I make two entries in extens... no that would recurse... ick |
| 23:01.25 | rue_mohr | hmmm |
| 23:01.53 | rue_mohr | maybe I should see if I can do this on the norstar sets |
| 23:02.07 | Bullseye_Network | exten => 444,1,Dial(SIP/phone1&SIP/phone2) |
| 23:02.09 | rue_mohr | I dont think it does it though |
| 23:02.18 | rue_mohr | yes, but I cant do |
| 23:02.40 | Dovid | http://pastebin.ca/92816 |
| 23:03.04 | rue_mohr | exten => 6402,4Dial(zap/g2/6402&zap/g2/6430,15) |
| 23:03.10 | Dovid | rue_mohr: look at this http://pastebin.ca/92816 |
| 23:03.19 | rue_mohr | I'm pretty usre it wouold not like that |
| 23:03.38 | rue_mohr | but, I cant change the destinatins numbers |
| 23:03.52 | rue_mohr | it needs to be dialed as 6402 and 6430 |
| 23:03.58 | Dovid | ok |
| 23:04.02 | Dovid | so the phones that u have now |
| 23:04.07 | Dovid | change thier extensions |
| 23:04.16 | Dovid | and switch around the exten numbers |
| 23:04.21 | jm|work | what # do people use for voicemail? |
| 23:04.24 | TripleFFFF | man |
| 23:04.27 | rue_mohr | yea, I'm believing thats the only way |
| 23:04.28 | *** join/#asterisk pdthome (n=pdthome@c-68-53-40-50.hsd1.tn.comcast.net) |
| 23:05.05 | *** join/#asterisk linlin (n=linlin@c-67-184-230-25.hsd1.il.comcast.net) |
| 23:05.09 | rue_mohr | jm|work well, our local office uses 6100 numbers with 6100 being hte * box |
| 23:05.19 | jm|work | I see |
| 23:05.19 | TripleFFFF | look AGI records a gsm hmm and send to sphinx server.. but it wont work.. only finds coughs... now. if i go in and take the 14868.gsm it wrote and send direct to server .. i get/ => /installs/sphinx/work/client.pl ./14868.gsm |
| 23:05.23 | rue_mohr | so we use 6100 6300 and 6400 as per our 3 offices |
| 23:05.23 | TripleFFFF | FTYPE: gsm |
| 23:05.23 | TripleFFFF | Result: HELP |
| 23:05.32 | jm|work | I was thinking '123' |
| 23:05.36 | jm|work | but seems a bit ..... meh |
| 23:05.47 | TripleFFFF | so i guess. the bin streaming of a gsm soesnt work great |
| 23:05.50 | rue_mohr | jm|work depends on your dial plan really |
| 23:06.01 | jm|work | aye |
| 23:06.10 | jm|work | atm I have 1xx and 2xx extensions |
| 23:06.12 | jm|work | for two sites |
| 23:06.27 | rue_mohr | our phones are 6000 mv boxes are 5000 |
| 23:06.32 | rue_mohr | with sites int eh 100 |
| 23:06.34 | rue_mohr | 's |
| 23:06.45 | rue_mohr | site 1 is 6100 site 2 is 6200 |
| 23:06.57 | rue_mohr | extenal numbers are 6800 |
| 23:07.02 | jm|work | hm |
| 23:07.07 | *** join/#asterisk brockj49464_home (n=chatzill@63.87.56.153) |
| 23:07.09 | rue_mohr | 6900 is portable numbers |
| 23:07.22 | rue_mohr | 6700 are testing targest |
| 23:07.30 | jm|work | 6000 makes festival say "Mary had a little lamb" |
| 23:07.31 | *** part/#asterisk umay (n=chris@71-208-175-55.hlrn.qwest.net) |
| 23:07.42 | rue_mohr | no wait, thats 7000 is testing |
| 23:07.51 | rue_mohr | 7121 1mw source |
| 23:07.59 | rue_mohr | 7123 hold music |
| 23:08.01 | jm|work | I need to read up on Festive now :S |
| 23:08.16 | rue_mohr | (not that we use hold music as nobody agrees on what it should be) |
| 23:08.19 | *** part/#asterisk brockj49464_home (n=chatzill@63.87.56.153) |
| 23:08.20 | jm|work | yeah |
| 23:08.29 | jm|work | I need to set up a # for the operator |
| 23:08.33 | rue_mohr | 0 |
| 23:08.46 | jm|work | "To check your voicemail press 1 ...." |
| 23:08.46 | jm|work | etc. |
| 23:09.01 | rue_mohr | jm|work use the same pattern as the local cell phones |
| 23:09.14 | jm|work | that's 123 for voicemail :/ |
| 23:09.16 | jm|work | depends on carrier |
| 23:09.22 | rue_mohr | here its 1 for first message 5 to repeat, 7 to delete etc etc |
| 23:09.29 | jm|work | yeah |
| 23:09.31 | rue_mohr | yea, use what people are farmiliar with there |
| 23:10.01 | jm|work | "to repeat the message, press 1 .... to store the message, press 2 .... to delete the message, press 3" |
| 23:10.07 | jm|work | "Message deleted. Main menu ...." |
| 23:10.36 | rue_mohr | just remember '9 to hear a duck quack' |
| 23:10.40 | jm|work | the automagic emailling of messages is good |
| 23:11.03 | rue_mohr | I have to go read a think norstar book... |
| 23:11.07 | rue_mohr | thick |
| 23:11.50 | rue_mohr | see if I can make the phones right togethor at the ics level |
| 23:12.11 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-152-23-125.red.bezeqint.net) |
| 23:12.57 | *** join/#asterisk okdo (n=goldenol@65.171.196.18) |
| 23:13.34 | TommyTheKid | Has anyone use the sangoma A108d? |
| 23:19.24 | okdo | anyone have issues with spandsp and fax receive quality being bad? |
| 23:19.30 | okdo | a lot of the pages seem to get chopped up |
| 23:19.33 | TripleFFFF | im trying to push a binary file trough a socket to remote server listening .. can one tell me how in c ? |
| 23:19.43 | TripleFFFF | wrong chan.. sorry |
| 23:20.30 | rue_mohr | waddya bet two norstar phones wiht the same line assignment ring togethor... |
| 23:21.20 | *** join/#asterisk JaredBluestein (n=Jared@nwlnnhbas01-pool0-a223.nwlnnh.tds.net) |
| 23:24.19 | *** join/#asterisk Kumba_ (n=kumba@office.crashsys.com) |
| 23:24.34 | JT | morning |
| 23:26.42 | rue_mohr | "broadcast ring group ability is possible with multiple target line assignment across a group of sets |
| 23:26.44 | rue_mohr | " |
| 23:26.48 | rue_mohr | so I think thats "yes" |
| 23:27.32 | *** part/#asterisk mog (i=ejabberd@68.62.237.103) |
| 23:31.22 | *** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-42-234.losaca.adelphia.net) |
| 23:31.45 | Kumba_ | Are there any utilities for planning out extensions.conf? |
| 23:32.11 | Qwell[] | Kumba_: only tool required is your brain |
| 23:32.11 | Kumba_ | :( |
| 23:32.12 | Qwell[] | pen/pencil and paper might also be good |
| 23:32.40 | Strom_C | Kumba_: you could always hire a consultant |
| 23:32.46 | rue_mohr | whats complex about your extensions conf |
| 23:33.06 | Kumba_ | I've never done one before... :) that's about it... |
| 23:33.23 | rue_mohr | ok, well, what external numbers are you handling |
| 23:33.39 | Kumba_ | 12-lines... |
| 23:33.47 | rue_mohr | ok |
| 23:33.53 | Kumba_ | 8 inside extensions |
| 23:34.07 | rue_mohr | and how many external numbers are bound to your 12 lines |
| 23:34.15 | Strom_C | see, why would you trust someone who misspells "amateur"? :) |
| 23:34.22 | rue_mohr | hah! |
| 23:34.22 | Kumba_ | 12... |
| 23:34.33 | Qwell[] | or "experience" |
| 23:34.42 | hads | 12 lines and 8 extensions? |
| 23:34.47 | rue_mohr | ok, map out which phones each of the 12 numbers go to |
| 23:34.57 | rue_mohr | hads its plausable |
| 23:35.12 | Qwell[] | Strom_C: dcap allowed you to add "highly"? heh |
| 23:35.12 | rue_mohr | did each plus a depeartment number |
| 23:35.22 | Strom_C | Qwell[]: yes :) |
| 23:35.33 | hads | rue_mohr: That's numbers, not lines. |
| 23:35.34 | Strom_C | im going to milk this dcap thing for all it's worth |
| 23:35.42 | rue_mohr | so its 2 peoplple in each of 4 departments |
| 23:35.54 | rue_mohr | ok... |
| 23:36.13 | rue_mohr | so you have 12 numbers from the telco, and 8 phones |
| 23:36.18 | rue_mohr | however, |
| 23:36.21 | rue_mohr | map them all out |
| 23:36.30 | rue_mohr | say when |
| 23:37.45 | JT | 12 lines and 8 extensions would seem a little pointless, unless most call response work is done by IVR or other automated process |
| 23:37.54 | rue_mohr | Kumba_ done? |
| 23:38.04 | rue_mohr | JT not neccissarily |
| 23:38.17 | JT | why not |
| 23:38.18 | rue_mohr | lets say you have 8 emplyees, divided into 4 departments |
| 23:38.22 | Kumba_ | I need 4 queue's... one for sales, warranty, accounting, and customer service... |
| 23:38.28 | JT | i said lines not numbers |
| 23:38.32 | rue_mohr | so each employee would have a did |
| 23:38.40 | rue_mohr | and each department would have a did too |
| 23:38.47 | Strom_C | Kumba_: I seriously recommend you upgrade to a PRI with DIDs |
| 23:39.05 | rue_mohr | the cost point is 13 |
| 23:39.05 | lirakis | im trying to figure out how to make a menu type dial plan.... i want to dial 2000, then have it say enter the extention of the person you want to reach, then if they enter 1, or 2, it will dial 2001, or 2002 (wich are defined in my sip.conf) .. i cant figure out how to do that.. it asks the question.. but then it just hangs up. and if i dial 1 or 2 instead of 2000 .. it connects me to 2001 or 2002 respective.. |
| 23:39.09 | Toerkeium | guys, I am trying to install *. I need it for a pure VOIP install. Do I need zaptel and libpri packages? I want to use meetme and IAX |
| 23:39.19 | JT | Kumba_: so not actually 12 physical phone lines? |
| 23:39.32 | Kumba_ | It's 12 physical phone lines... |
| 23:39.41 | hads | Toerkeium: You will need zaptel but not libpri. |
| 23:39.42 | Strom_C | Kumba_: i thought it was a T1 |
| 23:39.47 | Toerkeium | thanks hads |
| 23:39.49 | rue_mohr | Kumba_ if it goes over 13, go to a T1 |
| 23:39.51 | Pazzo | Toerkeium: meetme needs zaptel (ztdummy) |
| 23:39.53 | lirakis | im finding a pastebin to post my extensions.conf |
| 23:39.53 | Kumba_ | It is... but it's not PRI... |
| 23:40.08 | Kumba_ | There are 12 lines in the round robin for my 800-number... |
| 23:40.11 | JT | ah, so you're in the US? so they would be analogue lines, which can't do DID :( |
| 23:40.21 | Strom_C | Kumba_: "12 physical phone lines" would indicate 12 POTS lines |
| 23:40.27 | Strom_C | JT: you can do DID with analog |
| 23:40.29 | Kumba_ | Sorry... |
| 23:40.33 | Strom_C | JT: its kludgy though |
| 23:40.37 | JT | Strom_C: distinctive ring? |
| 23:40.47 | rue_mohr | Kumba_ so you have 12 channels on a T1? |
| 23:40.52 | rue_mohr | dosn't matter anyhow |
| 23:40.54 | Strom_C | JT: no, polarity reversal and then DTMF |
| 23:40.57 | lirakis | here is my extensions.conf file that i have tried to setup a menu with |
| 23:40.59 | lirakis | http://pastebin.ca/92840 |
| 23:41.02 | JT | sounds fun |
| 23:41.22 | Kumba_ | I have a full T1... 24 channels... each channel represents one FXS line... |
| 23:41.35 | rue_mohr | but you have only activated 13 channels? |
| 23:41.37 | JT | half a dozen BRIs are so much nicer than a dozen analogue lines :) |
| 23:41.48 | Strom_C | Kumba_: like I said....save yourself the headache and change it to PRI |
| 23:41.54 | rue_mohr | you DO know that 11 lines is more than enough for an office of about 100+ people usually |
| 23:42.02 | JT | Kumba_: sounds like he already has one |
| 23:42.03 | JT | err |
| 23:42.05 | JT | Strom_C: |
| 23:42.13 | Kumba_ | We fill up all 12 lines daily |
| 23:42.14 | Strom_C | JT: no, I helped him yesterday |
| 23:42.20 | rue_mohr | ok |
| 23:42.21 | Strom_C | JT: it's a CAS T1 |
| 23:42.27 | rue_mohr | I suppose if its a call centre |
| 23:42.33 | JT | with a channel bank? |
| 23:42.44 | Strom_C | JT: going into a digium t1 card |
| 23:43.01 | JT | ok |
| 23:43.21 | Kumba_ | That part is all working... |
| 23:43.42 | lirakis | any help on this phone menu ?? .. the asterisk book i have is not very clear on this |
| 23:43.54 | Strom_C | lirakis: lemme have a look |
| 23:44.07 | Kumba_ | if I had DID on 6 lines... once those 6 lines are full i'm done correct? |
| 23:44.23 | lirakis | Strom_C: thanks for taking a second to look at it |
| 23:44.45 | Strom_C | lirakis: you want extension 2000 to goto extension s of a second context where you play the menu and have extensions 1 and 2 |
| 23:45.02 | Strom_C | lirakis: but this begs the question of why you're doing that when you can just dial 2001 and 2002 directly |
| 23:45.33 | lirakis | Strom_C: i mjust doin git for academic purposes... i have a asterisk box i set up in my basement.. i want to learn how to do stuff with it |
| 23:45.54 | Strom_C | lirakis: ok ;) |
| 23:45.56 | lirakis | .. certainly its not necessary.. and is faster to dial 2001 .. or 2002.. but i want to know how to set up a meny |
| 23:45.59 | lirakis | *menu |
| 23:46.10 | lirakis | .. i will see if i cant figure it out from what you have said |
| 23:47.56 | rue_mohr | Kumba_ sorry, I cant answer that |
| 23:48.24 | rue_mohr | Kumba_ I know you can have just 1 line with a did number on it, and you can fill all your channels with calls to that number |
| 23:48.45 | rue_mohr | line? |
| 23:48.58 | rue_mohr | I think this is just outside my scope |
| 23:49.13 | rue_mohr | numbers must apply to the T1 as a pool |
| 23:49.14 | Dovid | can i have a Queue have an extension as a member ? |
| 23:49.27 | Qwell[] | Dovid: sure, use chan_local |
| 23:49.37 | rue_mohr | hah |
| 23:49.37 | Dovid | i didnt see that on the wiki |
| 23:49.46 | Dovid | let me search for it |
| 23:49.52 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 23:50.07 | rue_mohr | people asking questions are like mice in a field, every so often a hawk swoops down and answers one |
| 23:50.29 | Qwell[] | most people are expected to have RTFM |
| 23:50.39 | Dovid | lol |
| 23:50.46 | *** join/#asterisk murf (n=steve_mu@216.166.159.177) |
| 23:51.04 | Dovid | Qwell so i do Member => Local/exten@context ? |
| 23:51.12 | Qwell[] | Dovid: pretty much |
| 23:51.15 | Dovid | hanks |
| 23:51.17 | Dovid | thanks |
| 23:51.24 | rue_mohr | anyone know norstar that well, I have assigned multiple line assignments to the same two phones, and they dont ring togethor... |
| 23:51.24 | lirakis | Strom_C: okay.. i have moved the menu to a seperate context called [menu] so my first line of [internal] is exten => 2000,1,Goto(menu) then i play the soun & etc. Now i cant dial 1 or 2 and have it connect to 2001 or 2002 directly.. which is good, but if i dial 2000 it plays the message then hangs up immediatly |
| 23:51.58 | lirakis | do i need a goto(menu,s,2) to keep it looping for input?? |
| 23:52.03 | Strom_C | what? |
| 23:52.08 | lirakis | ha ah |
| 23:52.22 | Strom_C | you need to goto(menu,s,1) |
| 23:52.26 | lirakis | .. i think i need to make it loop.. otherwise it doesnt wait for input.. it just hangs up |
| 23:52.40 | lirakis | Strom_C: right.. ha ha it just took me a second to realize that |
| 23:55.02 | *** join/#asterisk nentis (n=nentis@hotblack.opensourcery.com) |
| 23:55.28 | nentis | Is it possible to use an ATA for connecting a modem? If not, is there a way to use dialup over VoIP? |
| 23:55.55 | Strom_C | nentis: it'll work if all you're doing is connecting over a LAN to a PRI :) |
| 23:56.21 | nentis | mm. nope. VoicePulse is our PSTN provider using IAX. |
| 23:56.26 | Strom_C | nentis: but for longer-distance communications, modem over voip is tricky at best and just a miserable exercise in failure at worst |
| 23:58.06 | rue_mohr | OH answer DN's! |
| 23:58.08 | rue_mohr | ?? |
| 23:58.25 | Strom_C | rue_mohr: uh...what? |
| 23:58.54 | rue_mohr | I'm trying to figure out the norstar ics that we use to interface with teh norstar phones |
| 23:59.07 | rue_mohr | I need to get both phones to ring with each other |
| 23:59.20 | lirakis | Strom_C: is there a way to have a kind of "dummy" extension that i can use just to jump back to? right now i Goto(menu,s,2) (which is right after Answer()) and it plays the "enter the ext... " over and over again while it waits for input. Can I creat a dummy exten entry just after the Backgroud() .. so it doesnt play the sound over and over again? |
| 23:59.24 | rue_mohr | I think I need to add answer DN's |
| 23:59.28 | Strom_C | dude, this is #asterisk, not #nortel :) |
| 23:59.33 | rue_mohr | er? |
| 23:59.40 | rue_mohr | is there a #nortel? |
| 23:59.48 | Strom_C | lirakis: how about setting the response timeout |
| 23:59.55 | Strom_C | and making use of the t extension :) |
