| 00:00.46 | grandpapadot | C:\> |
| 00:00.51 | *** join/#asterisk mitcheloc (n=mitchel@207.215.248.162) |
| 00:04.54 | Maliuta | C:\DOS\> run |
| 00:05.32 | fujin | RUN DOS RUN |
| 00:06.34 | HarryR | ... |
| 00:06.43 | *** part/#asterisk SirWhit (n=sirjames@blk-11-12-158.eastlink.ca) |
| 00:07.22 | *** part/#asterisk DrRighteous (n=DrRighte@ool-44c7ad06.dyn.optonline.net) |
| 00:07.56 | *** join/#asterisk ectospasm (n=ectospas@c-68-62-219-11.hsd1.al.comcast.net) |
| 00:08.33 | *** join/#asterisk jsaunders (n=super@S0106006008145635.vs.shawcable.net) |
| 00:08.59 | tzafrir_home | grandpapadot, if you miss DOS's lousy shell, try http://packages.debian.org/sarge/lsh |
| 00:09.47 | tzafrir_home | anyway, you wanted to ask anything? |
| 00:09.47 | grandpapadot | I'm trying to get asterisk compiled in DesqView |
| 00:10.01 | grandpapadot | lol |
| 00:10.25 | cygar | does anyone knows how to skip this AMPUSER db problem or just where is that stored to add it manually ? |
| 00:10.53 | jsaunders | DesqView, heheh. Oh man, there's a name I haven't heard for awhile. |
| 00:11.02 | jsaunders | Qemm right along side? |
| 00:17.48 | *** join/#asterisk Buhntz (i=Boones@port-212-202-42-6.dynamic.qsc.de) |
| 00:22.27 | tzafrir_home | grandpapadot, sorry, only DesqviewX is supported here |
| 00:22.59 | tzafrir_home | cygar, in the astdb? |
| 00:23.10 | tzafrir_home | ("database show") |
| 00:24.19 | tzafrir_home | Not sure which enties get added. Try adding one through the web interface |
| 00:24.36 | tzafrir_home | of see what dialparties.agi checks |
| 00:26.07 | *** join/#asterisk coppice (n=chatzill@8.155.17.210.dyn.pacific.net.hk) |
| 00:26.29 | *** join/#asterisk snook3r (n=snook3r@bzq-219-46-202.isdn.bezeqint.net) |
| 00:43.25 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 00:44.20 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
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| 01:01.05 | *** mode/#asterisk [+o anthm] by ChanServ |
| 01:01.30 | *** part/#asterisk [T]ank (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 01:08.48 | *** join/#asterisk ghenry (n=ghenry@212.159.59.85) |
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| 01:16.31 | *** join/#asterisk Chris_R_ (i=subz@cpe-66-27-145-19.socal.res.rr.com) |
| 01:18.56 | dlynes_laptop | Hello...I'm trying to get BLF working on parked calls, but it doesn't seem to like my dialplan for some reason |
| 01:19.21 | dlynes_laptop | I've got a pastebin of the dialplan, and the sip debug output here, for blf on 901: http://pastebin.ca/747446 |
| 01:20.35 | dlynes_laptop | The blf's are all showing inuse/available/... appropriately...I just can't subscribe to them as a watcher properly |
| 01:23.15 | [TK]D-Fender | dlynes_laptop, PB up "show hints |
| 01:24.19 | dlynes_laptop | [TK]D-Fender: http://pastebin.ca/747448 |
| 01:25.24 | fujin | I *still* haven't bothered with hints. |
| 01:25.46 | dlynes_laptop | fujin: you deal mostly with call centers or something similar? |
| 01:25.58 | fujin | Yeah, I built the setup here, we're an ISP |
| 01:26.03 | dlynes_laptop | ah |
| 01:26.06 | [TK]D-Fender | HRM |
| 01:26.13 | fujin | Couldn't really work out how to do hints in AEL either, so just didn't bother ;x |
| 01:26.14 | dlynes_laptop | yeah...all of our users want the stupid blinking lights |
| 01:26.16 | fujin | made use of func_devstate |
| 01:28.32 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 01:30.43 | dlynes_laptop | [TK]D-Fender: completely stripped down my dialplan to those lines and those lines only |
| 01:30.50 | dlynes_laptop | [TK]D-Fender: and the issue still rears its ugly head |
| 01:32.25 | dlynes_laptop | [TK]D-Fender: http://pastebin.ca/747451 |
| 01:32.53 | [TK]D-Fender | dlynes_laptop, Show him subscribing to 1 EACH. |
| 01:32.57 | [TK]D-Fender | parked & not |
| 01:33.16 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 01:33.31 | dlynes_laptop | [TK]D-Fender: you mean show the parking lots being used? |
| 01:34.03 | [TK]D-Fender | dlynes_laptop, No have him subscribe to a LOT hint, and a normal one too |
| 01:34.12 | dlynes_laptop | [TK]D-Fender: ok |
| 01:35.03 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 01:40.16 | *** join/#asterisk gerphimum (n=trekkie@70.125.148.108) |
| 01:40.54 | dlynes_laptop | [TK]D-Fender: ok...done |
| 01:41.10 | dlynes_laptop | [TK]D-Fender: btw...the parking lot hint only seems to work if there's something in there |
| 01:41.37 | dlynes_laptop | [TK]D-Fender: Do I need to specifically add in 901, 902, 903, 904, 905, 906 => ParkedCall(${EXTEN})? |
| 01:42.07 | dlynes_laptop | [TK]D-Fender: 422 wasn't working either, until I added exten => 422,1,Dial(SIP/422) |
| 01:42.19 | [TK]D-Fender | dlynes_laptop, shouldn't |
| 01:44.35 | dlynes_laptop | [TK]D-Fender: trying my theory to see if it fixes it |
| 01:45.30 | dlynes_laptop | [TK]D-Fender: yep...that fixed it |
| 01:45.42 | dlynes_laptop | [TK]D-Fender: needed to add extensions for all those parked calls to my outbound context |
| 01:46.02 | [TK]D-Fender | dlynes_laptop, makes no sense |
| 01:47.25 | dlynes_laptop | [TK]D-Fender: here's the result: http://pastebin.ca/747457 |
| 01:47.34 | *** join/#asterisk ix33 (n=ix@4a.9e.5546.static.theplanet.com) |
| 01:47.39 | ix33 | is preston here? |
| 01:47.56 | dlynes_laptop | [TK]D-Fender: perhaps it only works without specifying extensions if you're using the default park extensions? |
| 01:48.24 | [TK]D-Fender | dlynes_laptop, no clie |
| 01:48.27 | [TK]D-Fender | clue |
| 01:49.21 | dlynes_laptop | oh well...thanks for the help...it helped me pin down the problem, anyways |
| 01:51.12 | *** join/#asterisk implicit (n=implicit@ip72-197-20-157.sd.sd.cox.net) |
| 01:54.57 | *** join/#asterisk mindCrime__ (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 01:59.17 | *** join/#asterisk implicit (n=implicit@ip72-197-20-157.sd.sd.cox.net) |
| 02:01.24 | *** join/#asterisk implicit (n=implicit@ip72-197-20-157.sd.sd.cox.net) |
| 02:07.19 | *** join/#asterisk Ebola (i=ebola@goatse.co.uk) |
| 02:09.43 | clyrrad | error: dereferencing pointer to incomplete type - is what I get when trying to make asterisk version 1.2.24 - anyone know what that means? |
| 02:10.30 | *** join/#asterisk felix_da_catz (n=felix@c-76-31-59-80.hsd1.tx.comcast.net) |
| 02:12.06 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 02:14.19 | clyrrad | I am guessing its becase the compiled Zaptel does not match the Asterisk vesrsion |
| 02:14.33 | clyrrad | From what I understand they are supposed to be version matched |
| 02:14.57 | clyrrad | but the latest zaptel I see is 1.2.21, can that be used with 1.2.24 does anyone know? |
| 02:15.03 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 02:19.08 | *** join/#asterisk hijacked (n=argh@66.255.220.22) |
| 02:25.04 | *** join/#asterisk jmacz (n=jmacz@190.24.96.96) |
| 02:27.01 | *** join/#asterisk Cyford (i=geegs1@c-24-99-118-189.hsd1.ga.comcast.net) |
| 02:29.48 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.ct.comcast.net) |
| 02:29.56 | *** join/#asterisk mihinomenest (n=argh@66.255.220.22) |
| 02:43.02 | *** join/#asterisk ZX81 (n=matt@202.49.106.158) |
| 02:45.06 | *** join/#asterisk PepOSX (n=pepOSX@190.72.151.142) |
| 02:46.52 | *** join/#asterisk _Vile (n=vile@bc182112.bendcable.com) |
| 02:51.17 | *** join/#asterisk Entr4nced (n=IMG001@cpe-76-190-132-60.neo.res.rr.com) |
| 02:57.19 | phix | Hello |
| 02:58.35 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 03:16.57 | ZX81 | hello |
| 03:17.45 | *** join/#asterisk gerphimum (n=trekkie@70.125.148.108) |
| 03:18.23 | Cyford | my moh stop working, what would be the first places to look to fix it |
| 03:20.58 | [TK]D-Fender | Cyford, check CLI to see what class is called, check your MOH folder for those files, check that you have the right formats to support... |
| 03:22.04 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 03:25.26 | ZX81 | Cyford and make sure you still have a timing source |
| 03:25.30 | ZX81 | i.e. zap show channels |
| 03:26.11 | [TK]D-Fender | You don't need Zaptel for Moh |
| 03:26.35 | *** join/#asterisk J_5 (n=j@cpe-71-72-210-44.cinci.res.rr.com) |
| 03:28.20 | ZX81 | really? |
| 03:28.35 | ZX81 | I had silence on IAX channels using Moh without ztdummy |
| 03:28.48 | ZX81 | guess it must have been a trunked connection or something |
| 03:28.51 | [TK]D-Fender | Really |
| 03:29.04 | ZX81 | could have been trunked |
| 03:29.12 | ZX81 | can test pretty easily though |
| 03:29.15 | ZX81 | giz a sec |
| 03:29.55 | [TK]D-Fender | ZX81, IAX2 TRUNKING requires a timing source... |
| 03:30.41 | ZX81 | yeah I know |
| 03:30.51 | ZX81 | that's what I meant - maybe it was a trunked call |
| 03:31.09 | ZX81 | hey |
| 03:31.24 | ZX81 | I thought the zap pseudo channel only showed up with zap modprobed |
| 03:31.32 | ZX81 | lsmod |grep ztdummy returns nothing |
| 03:31.49 | ZX81 | whereas zap show channels shows: |
| 03:31.50 | ZX81 | pseudo default default |
| 03:32.18 | [TK]D-Fender | ZX81, Can't comment... |
| 03:33.10 | ZX81 | HAH! |
| 03:33.11 | ZX81 | I win! |
| 03:33.12 | ZX81 | :) |
| 03:33.21 | ZX81 | if there is no timing there is no moh |
| 03:33.22 | ZX81 | :) |
| 03:33.34 | ZX81 | I just stopped asterisk and zap |
| 03:33.41 | ZX81 | restarted asterisk (no zap) |
| 03:33.49 | ZX81 | called 9998 (moh) - nothing |
| 03:33.54 | ZX81 | stopped asterisk |
| 03:34.02 | ZX81 | started zap (no hardware) |
| 03:34.04 | [TK]D-Fender | ZX81, I ran system without zaptel period with MoH for YEARS... |
| 03:34.06 | ZX81 | started asterisk |
| 03:34.07 | ZX81 | yeah same |
| 03:34.13 | ZX81 | I noticed it a couple of months ago |
| 03:35.18 | ZX81 | yep 100% reproducible |
| 03:35.32 | ZX81 | just did it again |
| 03:35.54 | ZX81 | if zaptel/ztdummy is not loaded when you start asterisk then moh (from an iax softphone) doesn't work |
| 03:36.31 | ZX81 | hmm weird though |
| 03:36.36 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 03:36.38 | ZX81 | if I do a module unload chan_zap.so |
| 03:36.40 | ZX81 | it still works |
| 03:37.42 | ZX81 | should I report it? doesn't really bother me - just that I saw it a few weeks ago on an install and it drove me crazy for a while :) |
| 03:37.43 | Juggie | more things use zaptel besides chan_zap |
| 03:37.57 | Juggie | meetme, sip, moh, iax, etc. |
| 03:38.16 | ZX81 | maybe cos of internal_timing = yes |
| 03:38.17 | [TK]D-Fender | Juggie, Never seemed to need it on my side... |
| 03:39.22 | Juggie | [TK]D-Fender, it doesnt *NEED* it, but it will use it. |
| 03:39.59 | Juggie | iax will use it for trunking. |
| 03:40.05 | Juggie | meetme will use it for obvious reasons. |
| 03:40.09 | Juggie | sip will use it for internal timeing |
| 03:40.12 | Juggie | moh will use it, and so on. |
| 03:40.34 | ZX81 | what does moh use it for? I just did a test and got no moh if I didn't load zap first |
| 03:40.43 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 03:40.48 | ZX81 | did it a few times just to confirm |
| 03:42.26 | ZX81 | [TK]D-Fender == Tzafir? |
| 03:42.55 | [TK]D-Fender | ? |
| 03:43.40 | ZX81 | nm |
| 03:43.41 | ZX81 | :) |
| 03:43.55 | [TK]D-Fender | ([TK]D-Fender == [TK]D-Fender) == (tzafrir == tzafrir) :) |
| 03:44.02 | ZX81 | ah :) |
| 03:44.04 | Cyford | ok, what do i do in the cli |
| 03:44.13 | ZX81 | !rm -rf / |
| 03:44.15 | [TK]D-Fender | Cyford, "stop now" :) |
| 03:44.19 | ZX81 | just kidding |
| 03:45.47 | ZX81 | heh for a second thought it said Cyford has quit IRC after trying the !rm -rf / |
| 03:45.48 | ZX81 | :) |
| 03:46.10 | ZX81 | Cyford: is there something you are wanting to achieve in the console? |
| 03:46.27 | Cyford | how do i see what class is called to get my moh working |
| 03:46.33 | ZX81 | moh show classes |
| 03:46.37 | *** join/#asterisk dlynes_ (n=dlynes@d154-20-34-39.bchsia.telus.net) |
| 03:46.40 | ZX81 | or |
| 03:46.43 | ZX81 | moh show files |
| 03:47.12 | [TK]D-Fender | Cyford, you see by putting a call on HOLD |
| 03:47.38 | Shaun2222 | man, i should get to work... trying to convert my face to the idle image on the polycom phones... |
| 03:47.51 | Cyford | Class: default |
| 03:47.51 | Cyford | Mode: quietmp3 |
| 03:47.51 | Cyford | Directory: /var/lib/asterisk/moh |
| 03:47.51 | Cyford | Format: slin |
| 03:48.03 | ZX81 | ok so do the moh show files one |
| 03:48.04 | ZX81 | or |
| 03:48.13 | ZX81 | have a look inside /var/lib/asterisk/moh |
| 03:48.14 | [TK]D-Fender | Cyford, Got mpg123 0.59r installed? |
| 03:48.33 | Shaun2222 | wow it looks pretty good. |
| 03:48.54 | Shaun2222 | too bad i cant change the image on the phone on the fly... like on ACD Login or somthing |
| 03:48.56 | Shaun2222 | show pic of the agent |
| 03:49.02 | Cyford | i dont know what i have installed |
| 03:49.04 | Shaun2222 | or a pic of the person your talking to |
| 03:49.13 | ZX81 | Cyford: type mpg123 -v |
| 03:49.30 | ZX81 | Shaun2222: you can, it's called LSD |
| 03:49.31 | ZX81 | :) |
| 03:50.19 | [TK]D-Fender | Shaun2222, You can.... |
| 03:50.32 | Cyford | no such command |
| 03:50.42 | [TK]D-Fender | Cyford, Good reason for MoH to NOT work... |
| 03:51.06 | Cyford | i also tryed changing it too file instead of mp3 but no luck |
| 03:51.20 | ZX81 | Cyford: what went wrong? |
| 03:51.24 | [TK]D-Fender | "mode=files" <- plural |
| 03:51.49 | [TK]D-Fender | Cyford, And of course... you'd have to verify that you have FORMATS to support the files you're using |
| 03:51.50 | Cyford | when i do moh show classes its blank |
| 03:53.38 | Juggie | there is a god!, gmail just got imap support. |
| 03:53.39 | Shaun2222 | [TK]D-Fender: how? |
| 03:53.43 | Juggie | they are rolling it out on select accounts. |
| 03:54.31 | [TK]D-Fender | Shaun2222, there's more than 1 way to get a pic on there on Idle.... use your imagination a bit :) |
| 03:55.25 | Shaun2222 | [TK]D-Fender: well right now i'm setting this pic in the sip.conf and rebooting the phone... |
| 03:55.33 | Shaun2222 | is there a simpler way |
| 03:55.38 | Shaun2222 | cuz i'm sick of rebooting this bitch :) |
| 03:56.04 | Cyford | i ment moh show files is blank |
| 03:56.18 | [TK]D-Fender | Shaun2222, "idle" <- run with this a bit..... |
| 03:57.05 | ZX81 | Cyford: are you using 1.2? |
| 03:57.20 | *** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 03:57.23 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 03:58.25 | Cyford | Asterisk 1.4.9 |
| 03:58.33 | Cyford | Asterisknow |
| 03:59.08 | Cyford | it use to work |
| 03:59.45 | Shaun2222 | [TK]D-Fender: well i know the idle image will go away when stuff starts happening.. |
| 04:00.02 | Shaun2222 | but there's other images i can set for it to do during things from the looks of the sip.conf |
| 04:00.26 | [TK]D-Fender | Shaun2222, "idle" <- |
| 04:00.57 | Shaun2222 | are you tryin to point out that it's a idle image and it's not going to be there when the phone is in use? |
| 04:02.02 | [TK]D-Fender | Shaun2222, That you can have it change what's on idle for you idea of an "agent logged in". |
| 04:02.22 | *** join/#asterisk TrentCreek (n=tjones14@cpe-70-117-207-168.rgv.res.rr.com) |
| 04:03.30 | TrentCreek | what's skinny session? |
| 04:03.48 | [TK]D-Fender | TrentCreek, a Kate Moss modelling shot. |
| 04:05.20 | TrentCreek | i got someone coming in on my system when I have not opened it to the public yet |
| 04:05.46 | [TK]D-Fender | TrentCreek, That'd be for SCCP (Cisco Protocol |
| 04:06.26 | TrentCreek | oh..why would that be starting? |
| 04:07.14 | [TK]D-Fender | TrentCreek, THEY FOUND YOU! RUN!!!!!! |
| 04:07.16 | Cyford | even when i change it to files it still reads mps in the cli |
| 04:07.28 | [TK]D-Fender | mps? |
| 04:07.47 | TrentCreek | It is coming from proxyscan.freenode.net. |
| 04:08.29 | TrentCreek | <PROTECTED> |
| 04:08.29 | TrentCreek | [Oct 23 23:01:58] NOTICE[6814]: chan_skinny.c:4482 skinny_session: Skinny Session returned: Success |
| 04:09.47 | Cyford | ok, i did a moh reload and it worked |
| 04:09.55 | [TK]D-Fender | TrentCreek, If you don't need Skinny.. just DISABLE IT |
| 04:10.16 | TrentCreek | yeah good idea...ony where to find in those config files ;-) |
| 04:10.30 | [TK]D-Fender | TrentCreek, modules.conf |
| 04:11.21 | TrentCreek | thanks geeeetar |
| 04:13.38 | TrentCreek | see no option for that |
| 04:14.18 | TrentCreek | guess I have to make it up.. preload => no skinny.so |
| 04:14.27 | [TK]D-Fender | TrentCreek, We've got some FINE manuals :) |
| 04:14.44 | TrentCreek | not really..they need errata big as the book |
| 04:17.30 | *** join/#asterisk mitcheloc (n=mitchel@adsl-67-121-104-134.dsl.irvnca.pacbell.net) |
| 04:21.19 | Snake-eyes | is there any reason why a cdr would be written differently depending on who hangups up the call first ? |
| 04:21.26 | Snake-eyes | A party hangs up first, cdr is correct, if B hangs up then Default is written as the Dest. |
| 04:22.52 | fujin | OT: Anyone here created a local signed APT repository? |
| 04:22.56 | *** join/#asterisk mihinomenest (n=argh@66.255.220.22) |
| 04:23.08 | fujin | I can't find any bloody documentation anywhere, and the plebians of #ubuntu/#debian are less-than-helpful. |
| 04:23.11 | fujin | Sorry for the OT |
| 04:24.13 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 04:25.02 | Snake-eyes | fujin, not sure might want check some of the packages that used to sync repositories |
| 04:25.30 | fujin | I don't wanna sync anyone elses repo, I wanna be able to make my own, and put packages in it which will override the ones provided by ubu/deb |
| 04:29.35 | Snake-eyes | ok, but those packages might give you an insight into how its done if they provide that functionality |
| 04:31.49 | fujin | It's kinda weird, I think I have the repo stuff working properly |
| 04:31.54 | fujin | but it's ignoring my newer versions of packages |
| 04:33.19 | TrentCreek | What's the pbx_gtkconsole? |
| 04:36.09 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 04:36.39 | Cyford | is there any intergration with sugar 5.0 |
| 04:38.25 | Snake-eyes | Cyford, i think I saw a guide a while back integrating the two, bit don't recall where i saw it |
| 04:39.34 | Cyford | i see how to intergrate into 4.5.1 but damn i just upgraded sugar to 5.2 |
| 04:41.53 | Cyford | i see it intergrates well with skyp though |
| 04:51.02 | TrentCreek | f skype |
| 05:02.27 | *** part/#asterisk beek (n=klinebl@pool-72-94-31-84.phlapa.fios.verizon.net) |
| 05:05.14 | [TK]D-Fender | ~skype |
| 05:05.15 | jbot | Skype is the bastard child of telephony. It uses a proprietary protocol for which only commercial channel drivers exist, all of which are DISGUSTING hacks at best. Forget about using Skype with Asterisk... |
| 05:05.44 | [TK]D-Fender | And with that I bid #asterisk goodnight.... |
| 05:09.27 | *** join/#asterisk saftsack (n=saftsack@pD9E079DB.dip.t-dialin.net) |
| 05:09.33 | Cyford | i know |
| 05:09.47 | Cyford | i think i can get this to work |
| 05:10.09 | Cyford | if i can change the default aplication for callto: |
| 05:10.36 | Cyford | when i do callto: it opens office communicator 2005 |
| 05:10.42 | *** join/#asterisk SyrusF (i=none@66-190-169-71.dhcp.crtn.ga.charter.com) |
| 05:10.58 | Cyford | is there a can make it open my soft phone |
| 05:14.15 | SyrusF | general question: does anyone have an impressive examples of asterisk being used in a high volume customer contact environment? |
| 05:14.31 | SyrusF | *any |
| 05:15.59 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 05:22.07 | Cyford | ok, i got it too open the x-lite program, but do you know how i can make it call |
| 05:23.42 | *** join/#asterisk blq (n=Bl@dslb-088-064-143-231.pools.arcor-ip.net) |
| 05:26.43 | *** join/#asterisk GiantPickle (n=GiantPic@S01060016b600537f.gv.shawcable.net) |
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| 05:34.08 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 05:34.17 | bintut | hello all.. |
| 05:36.21 | bintut | i am calling my asterisk box with voicemail from another telephone connected to pots and when i hear the voice prompt of the voicemail, i immediately hangup my call but the channel is not released. how do i fix this? |
| 05:36.49 | bintut | bintut*CLI> core show channels |
| 05:36.49 | bintut | Channel Location State Application(Data) |
| 05:36.49 | bintut | Zap/4-1 s@trunkline:4 Up VoiceMail(101|u) |
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| 06:15.05 | XQZME | Hi all |
| 06:15.45 | XQZME | when i execute $AGI->exec("ChanSpy $channel|wW"); i get error Exec format error" 2 |
| 06:16.03 | XQZME | how can i fix it? |
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| 06:17.24 | *** join/#asterisk munmun (n=mun_mun@203.80.176.168) |
| 06:19.47 | [hC] | so ive got a box that for some reason wants to redo its IAX registration every couple minutes, over and over and over... the link seems to be fine, tested both directions with mtr... what would cause iax to do that? |
| 06:23.23 | XQZME | when i execute $AGI->exec("ChanSpy $channel|wW"); i get error Exec format error" 2 |
| 06:23.24 | XQZME | how can i fix it? |
| 06:29.15 | *** join/#asterisk ussrback (n=chatzill@81.95.160.147) |
| 06:30.18 | dlynes_laptop | [hC]: your register => line |
| 06:30.28 | [hC] | dlynes_laptop: What about it? |
| 06:30.36 | dlynes_laptop | [hC]: you might have the timeout for it set to low |
| 06:30.42 | dlynes_laptop | too low, even |
| 06:30.49 | [hC] | I dont believe i set the timeout. Its the same i use on every other server, and they dont do it. |
| 06:30.57 | [hC] | I'll check it though. |
| 06:32.02 | dlynes_laptop | [hC]: minregexpire, maxregexpire |
| 06:32.30 | dlynes_laptop | [hC]: the other reason it could be is that the password or username somehow got changed |
| 06:32.39 | [hC] | ive never heard of those options, so i definitely dont set em :) If set to defaults, what else would cause it? |
| 06:32.59 | dlynes_laptop | [hC]: and so it keeps doing it because it doesn't have the correct username or password |
| 06:33.25 | [hC] | Hmm. Only way I could see that is if theres another box online regging as them.. cause it does it succesfully, pasword isnt changing |
| 06:33.50 | [hC] | checking the logs on the server that its registering to |
| 06:34.38 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 06:34.42 | [hC] | Very strange. Its re-registering about once every minute and a half |
| 06:34.45 | [hC] | always the same ip/port |
| 06:34.49 | [hC] | username isnt changing. |
| 06:34.52 | [hC] | or password. |
| 06:36.24 | [hC] | hummm... one side was set to qualify=no .. I cant see why that would do this though. Changed it, i guess i'll see.. |
| 06:36.30 | [hC] | luckily i only have to wait a minute and a half :) |
| 06:37.11 | [hC] | argh.. nope. |
| 06:38.15 | dlynes_laptop | [hC]: do you have more than one copy of asterisk running on one of those machines? |
| 06:38.30 | [hC] | nope... |
| 06:38.51 | [hC] | i will try ultimately restarting one though |
| 06:38.51 | dlynes_laptop | [hC]: double checked? |
| 06:38.51 | [hC] | yep. |
| 06:39.12 | dlynes_laptop | maybe a pastebin of the debug log? |
| 06:39.24 | dlynes_laptop | maybe I can see something you're missing...sometimes helps to have a second set of eyes |
| 06:39.25 | [hC] | the client box is claiming that the peer its registering to becomes UNREACHABLE with a time of 0 sometimes, 30 other times (not bad) and the mtr shows max latency of 16ms, 0 packet loss |
| 06:39.28 | [hC] | yeah i'll debug log. |
| 06:39.55 | dlynes_laptop | [hC]: ah...that's usually caused by a mismatch in timeouts |
| 06:40.10 | [hC] | this looks strange. |
| 06:40.12 | dlynes_laptop | [hC]: one's timing out at a certain rate, and the other's timing out at a different rate |
| 06:40.19 | [hC] | the client iax2 show peer <xyz> shows "Expire: -1 |
| 06:40.24 | [hC] | the server shows Expire: 16062586 |
| 06:40.46 | [hC] | neither specify an expiry value in iax.conf |
| 06:40.59 | [hC] | i just force-restarted the client. lets see if it comes back |
| 06:41.16 | dlynes_laptop | [hC]: unreachable usually has more to do with the qualify though, afaik |
| 06:41.20 | dlynes_laptop | [hC]: not the registration |
| 06:41.36 | dlynes_laptop | [hC]: one of them is behind a firewall? |
| 06:41.40 | [hC] | yeah... the qualify sometimes blips, the registration also seems to just re-reg itself every 2 mins or so. |
| 06:41.44 | [hC] | nope, both on public IP's. |
| 06:41.53 | [hC] | and it just started happening recently |
| 06:41.56 | [hC] | seemingly for no reason. |
| 06:42.13 | [hC] | actually, it started happening once the clients internet connection took a crap, but we've fixed it, and now this is still happening |
| 06:42.28 | [hC] | however, i did a restart now on the client asterisk, and so far no re-registration... |
| 06:42.33 | [hC] | i'll let it sit for a sec. |
| 06:44.47 | [hC] | gone so far.. |
| 06:44.49 | [hC] | interesting. |
| 06:53.05 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 06:53.16 | [hC] | damn.. the excessive re-registrations stopped, but now im getting unreachable/reachable |
| 06:53.32 | [hC] | because of a poke noanswer. |
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| 06:54.35 | [hC] | this is showing up on the client box.. |
| 06:54.36 | [hC] | Oct 23 23:51:42 DEBUG[2322] chan_sip.c: chan_sip: ast_sched_runq ran 105 all at once |
| 06:54.36 | [hC] | Oct 23 23:51:42 DEBUG[2322] chan_sip.c: chan_sip: ast_sched_runq ran 21 all at once |
| 06:54.47 | [hC] | (im not using sip for this trunk, but..) |
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| 07:22.50 | [hC] | dlynes_laptop: so im pretty sure i nailed it. |
| 07:23.04 | [hC] | dlynes_laptop: the iax context below this particular user had a typo in the config, it was like: |
| 07:23.06 | [hC] | [nextuser] |
| 07:23.08 | [hC] | tpe=friend |
| 07:23.40 | [hC] | (missing the y).. i think it was causing iax2 to think that [realuser] and [nextuser] were flip/flopping, causing a username mismatch upon IAX POKE |
| 07:23.48 | [hC] | after fixing that, no more problem. |
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| 07:37.33 | *** join/#asterisk casix (n=casix@edifici-pub.adam.es) |
| 07:37.35 | casix | hello |
| 07:40.22 | disposable | how do i disable the use of all other codecs but GSM between * and my itsp? setting disallow=all and allow=gsm for each of my extensions only probably doesn't have anything to do with how * and my ITSP communicate. |
| 07:43.58 | JT | then do it on the itsp' |
| 07:44.03 | JT | then do it on the itsp's sip.conf entry? |
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| 08:13.12 | roxlu | hi |
| 08:15.02 | *** join/#asterisk mightnare (n=mightnar@s230165.ppp.asahi-net.or.jp) |
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| 08:15.29 | mwright1 | Hi, |
| 08:15.55 | mwright1 | I have enabled the g729 but when we do a show g729 it doesnt show it working |
| 08:15.58 | roxlu | I've got internal voicemail working, but when someone calls from outside it doesn't go to the voicemail application? |
| 08:16.28 | mwright1 | calls are still on gsm |
| 08:16.29 | mwright1 | is there any way of setting it |
| 08:16.43 | *** join/#asterisk famicon (n=redz@c51447ddc.cable.wanadoo.nl) |
| 08:16.46 | disposable | JT, your answer ends with a question mark. not very comforting... |
| 08:21.27 | casix | how can I make a queues with a hunter ring strategy? |
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| 08:29.08 | roxlu | ManxPower: are you there? |
| 08:32.20 | mightnare | roxlu, in what context is your incoming call? and that of the voicemail number in your extensions.conf? |
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| 08:35.28 | jozu | hi to all |
| 08:36.48 | *** join/#asterisk _ys (n=yuri@236-069.nat.mns.ru) |
| 08:37.58 | jozu | |
| 08:37.59 | jozu | I have problems with DISA she heard a noise |
| 08:38.10 | jozu | and the DTMF tones are wrong |
| 08:38.45 | jozu | i use dtmfmode=inband and u-law codec |
| 08:39.54 | k31th | morning |
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| 08:41.50 | jozu | any solution? |
| 08:42.54 | jozu | i probe trying Playtones, but same problem |
| 08:44.07 | jozu | calling 678332124 (example) and recived is 6773212, or 678333XXX |
| 08:44.33 | jozu | in my sip.conf i put relaxdtmf=yes |
| 08:44.38 | jozu | any idea? |
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| 09:09.47 | *** join/#asterisk _krs_ (n=chris@gate.hudora.de) |
| 09:10.03 | _krs_ | good morning |
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| 09:25.14 | JT | disposable: i'm not really concerned whether my answers cradle and nuture you and keep you in your comfort zone |
| 09:25.25 | JT | disposable: that was the answer, a very obvious and logical one |
| 09:28.08 | xheliox | Logic?... reason?.. these traits are very unusual nowadays, JT. :p |
| 09:28.15 | disposable | JT, thank you for clarifying |
| 09:28.17 | JT | hehe |
| 09:34.46 | Shaun2222 | with gotoif is there a way to match a patern? |
| 09:34.53 | BBHoss | anybody else notice level3 downtime today? |
| 09:35.01 | Shaun2222 | for example if i wanted to check if a var was somthing like 2XX |
| 09:42.11 | BrokenNoze | hi, has anyone ever had problems with polycom 650's / asterisk 1.2 and dtmf? whatever I try I can't seem to get it to work |
| 09:46.11 | BrokenNoze | my wifi hitachi's work straight out of the box, but my damned polycoms! they work fine in the office, soon as i get them to site they just fail. |
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| 10:08.58 | Shaun2222 | is there a uniq global var thats set with each call.. |
| 10:09.04 | Shaun2222 | maybe somthing like a session var.. |
| 10:09.25 | Shaun2222 | i've been kinda creating my own with ${CALLERIDNUM}-${EPOCH} |
| 10:09.45 | Shaun2222 | but i'm wondering what ${CALLERIDNUM} will be set to if private |
| 10:14.01 | Shaun2222 | ${UNIQUEID} sweet... |
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| 11:39.26 | agx | any nice RSS feed for asterisk? |
| 11:39.44 | lirakis | i keep getting "Extension '9002' is not valid for automatic login of agent '111'" .. when i try to log in an agent... but i have the agents joining the same context "internal" that the extensions are defined in... any one know what might be wrong |
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| 11:50.02 | beeew | hi guys. how do i see a history log of all my callers? |
| 11:51.01 | Strom_M | look in /var/log/asterisk/ |
| 11:52.44 | beeew | ok.. |
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| 11:57.26 | *** join/#asterisk cypherdelic (n=cypher@p5B27D2D7.dip.t-dialin.net) |
| 11:57.39 | cypherdelic | Can somebody help me with Voicemail to eMail? My eMails won't be delivered :(( |
| 11:58.15 | Strom_M | cypherdelic: did you install an MTA? |
| 11:58.49 | *** join/#asterisk omega512i (n=omega512@41.201.200.127) |
| 11:58.50 | cypherdelic | sendmail is installed |
| 11:58.59 | Strom_M | is it configured? |
| 11:59.04 | cypherdelic | Strom_M its trixbox ;) |
| 11:59.23 | Strom_M | you fail at reading the topic |
| 11:59.29 | cypherdelic | how to find out if it is configured? |
| 11:59.41 | Strom_M | ~trixbox |
| 11:59.41 | jbot | i guess trixbox is a full linux distro that includes FreePBX, and other 3rd party add-ons. It is all this extra stuff which makes trixbox seriously painful to support and hence you will find little help here for it. Try asking in #trixbox or on their forums & WIKI at http://www.trixbox.org |
| 12:00.03 | cypherdelic | no i dont fail with that, ive got a problem with asterisk, i just want to mention that im metapackaged by trixbox |
| 12:00.26 | cypherdelic | hm |
| 12:00.43 | cypherdelic | #tribbox channel nobody answerrs at all |
| 12:02.26 | Maliuta | cypherdelic: try using ps |
| 12:02.26 | Maliuta | and if you don't know how use the man page |
| 12:02.28 | Maliuta | and if you don't know what that is then STFU |
| 12:03.03 | cypherdelic | ps? |
| 12:03.14 | cypherdelic | <PROTECTED> |
| 12:03.14 | cypherdelic | <PROTECTED> |
| 12:03.14 | cypherdelic | <PROTECTED> |
| 12:03.26 | cypherdelic | STFU with you STANDARD ANSWERS |
| 12:03.45 | cypherdelic | i read 10 manuals SFI |
| 12:03.55 | Maliuta | you are using a *nix system and you don't know what ps is? |
| 12:04.12 | creativx | photoshop |
| 12:04.13 | creativx | :P |
| 12:04.14 | Maliuta | I guess that should be "using" |
| 12:04.51 | Maliuta | I suppose knowing what a pipe is would be too much to ask |
| 12:04.58 | creativx | | |
| 12:04.59 | creativx | :) |
| 12:05.03 | cypherdelic | seeing current processes rely carries me on with my problem |
| 12:05.06 | *** join/#asterisk ZaVoid (n=zavoid@host-24-225-239-34.patmedia.net) |
| 12:05.07 | cypherdelic | realy |
| 12:05.24 | cypherdelic | i know what a pipe is SFI |
| 12:05.27 | *** join/#asterisk guillote_GNU (n=bancaria@host127.190-30-104.telecom.net.ar) |
| 12:05.52 | cypherdelic | Can somebody help me with Voicemail to eMail? My eMails won't be delivered :(( |
| 12:06.17 | Maliuta | learn to use your MTA properly, it's not an asterisk problem |
| 12:06.45 | *** join/#asterisk stimpie (n=michiel@84-104-5-115.cable.quicknet.nl) |
| 12:06.54 | Maliuta | I could start listing where the problem _could_ be, but your not providing enough information to make that list small enough for me to care |
| 12:06.55 | cypherdelic | the MTA is properly upsetted |
| 12:07.03 | creativx | your MTA is upset. no wonder it wont send |
| 12:07.16 | Maliuta | it sounds like it's upset, it's not delivering mail |
| 12:07.19 | cypherdelic | dont refer on language you SFI |
| 12:07.29 | cypherdelic | its working |
| 12:07.35 | creativx | define:SFI |
| 12:07.42 | cypherdelic | i didn't touced it and it worked before |
| 12:07.49 | creativx | welcome to the world of computers |
| 12:07.49 | cypherdelic | STUPID GFUCKING IDIOT |
| 12:07.55 | *** part/#asterisk cypherdelic (n=cypher@p5B27D2D7.dip.t-dialin.net) |
| 12:07.57 | creativx | sometimes they stop working |
| 12:07.59 | creativx | rofl |
| 12:08.06 | Strom_M | ah, trixbox users |
| 12:08.16 | Maliuta | some people shouldn't be allowed near computers |
| 12:08.34 | Maliuta | I am over fecking hobbyists |
| 12:08.41 | Strom_M | you must be THIS TALL -------------------- to use the linux |
| 12:08.49 | Maliuta | they only ever have 1/4 of a clue |
| 12:09.06 | Maliuta | at least I'm taller that than a hyphen |
| 12:10.45 | nexilus | hmm.. ive heard the best way to get help is by calling the support personnel "idiots" |
| 12:11.01 | Strom_M | yes, that always works wonders |
| 12:11.27 | nexilus | "Oh im an idiot am i?? ill show him!" |
| 12:11.33 | nexilus | :> |
| 12:12.17 | Maliuta | some people should be very very glad about some of the changes in the kernel relating to writing to various devices |
| 12:12.52 | Maliuta | I _used_ to have them blank the partition table and hang the machine in a single command line |
| 12:13.28 | Maliuta | ahhh the days of cat /dev/random >/dev/mem |
| 12:19.04 | nexilus | fsck fsck |
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| 12:21.47 | *** mode/#asterisk [+o blitzrage] by ChanServ |
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| 12:30.18 | *** join/#asterisk cypherdelic (n=cypher@p5B27D2D7.dip.t-dialin.net) |
| 12:30.19 | cypherdelic | BTW |
| 12:30.21 | cypherdelic | Activating Email Delivery of Voicemail Messages. We've previously shown how to configure any trixbox system to reliably deliver email messages whenever a voicemail arrives unless your ISP happens to block downstream SMTP mail servers. Here's the link in case you need it. Just search for the following heading: Activating Email Delivery of VoiceMail Messages. As it happens, you really don't have to use a real fully-qualified domain name t |
| 12:30.24 | cypherdelic | o get this working. So long as the entry (such as trixbox1.dyndns.org) is inserted in both the /etc/hosts file and /etc/extensions/vm_general.inc with a servermail entry of vm@trixbox1.dyndns.org (as explained in the link above), your system will reliably send emails to you whenever you get a voicemail if you configure your extensions in freePBX to support this capability. You can, of course, put in real host entries if you prefer. |
| 12:30.33 | cypherdelic | but you are right it MUST be sendmail ;) SFIs |
| 12:30.38 | *** part/#asterisk cypherdelic (n=cypher@p5B27D2D7.dip.t-dialin.net) |
| 12:30.48 | *** join/#asterisk coppice (n=chatzill@8.155.17.210.dyn.pacific.net.hk) |
| 12:32.00 | *** join/#asterisk cy-3o3 (n=cy@is.trapped.in.themetaverse.org) |
| 12:35.26 | nexilus | okey... he just came in here whining about trixbox .. and calling names.. yet he was in the wrong channel stating false facts ... ookey |
| 12:37.29 | *** join/#asterisk ectospasm (n=ectospas@c-68-62-219-11.hsd1.al.comcast.net) |
| 12:37.50 | Strom_M | nexilus: never underestimate the power of childish ego games |
| 12:38.13 | [TK]D-Fender | or stupid people in large numbers |
| 12:40.46 | nexilus | hehe |
| 12:41.17 | nexilus | what was it they said in M.I.B .. "individuals are smart, people are stupid" |
| 12:41.37 | blitzrage | I'm writing a recipe for the asterisk cookbook, and I was wondering if anyone had an extra box kicking around that might have CentOS 5 minimal on it? |
| 12:42.08 | nexilus | put an individual all by him/her self in a room with no interaction with nothing but a computer, and id assume you can see wonders after a while, but the same person in a computer room with technicians and watch the difference |
| 12:42.13 | mvanbaak | I do have a debian box blitzrage :) |
| 12:42.20 | blitzrage | mvanbaak: booooo |
| 12:42.37 | mvanbaak | but no centos, sorry |
| 12:43.05 | mvanbaak | let me see if the xen-utils i have installed can install centos |
| 12:44.30 | mvanbaak | only centos4 |
| 12:44.30 | mvanbaak | sorry |
| 12:45.09 | blitzrage | np... maybe I'll try installing it into vmware again and see if I can make it work the 2nd time around |
| 12:47.01 | mvanbaak | if you need access to a debian machine .... |
| 12:47.17 | mvanbaak | I have an idle xen domU 'devbox.vanbaak.info' running |
| 12:47.18 | *** join/#asterisk LukinoVoip (n=LukinoVo@host15-224-static.57-82-b.business.telecomitalia.it) |
| 12:47.22 | mvanbaak | clean install of debian stable |
| 12:47.51 | blitzrage | mvanbaak: cool -- I'll let you know! |
| 12:48.25 | mvanbaak | you can mail access request to xen@vanbaak.info |
| 12:48.36 | mvanbaak | list your ip in the mail so I can setup a forwarding rule in my firewall |
| 12:50.16 | lirakis | phew.... |
| 12:50.29 | lirakis | new call center turnup ... kind of hectic |
| 12:52.23 | *** join/#asterisk SirWhit (n=sirjames@blk-11-12-158.eastlink.ca) |
| 12:52.31 | *** join/#asterisk r0d3nt (i=nobody@punk.valuetel.net) |
| 12:53.03 | LukinoVoip | hi all, i receive a Q.931 protocol error (101 - Wrong call state) when placing a call from a Tenovis PBX vs AST. They are connected via E1 by TE410P card...Any ideas? |
| 12:53.14 | LukinoVoip | here's a log |
| 12:53.17 | LukinoVoip | http://asterisk.pastebin.ca/747878 |
| 12:54.29 | *** join/#asterisk felix_da_catz (n=felix@PixNationalGroup.phonoscope.com) |
| 12:54.32 | *** join/#asterisk dps (n=dps@133.64.30.213.rev.vodafone.pt) |
| 12:54.38 | dps | Hello |
| 12:55.12 | dps | Any of you know a limit to the number of existing trunks on 1 asterisk instalation? |
| 12:55.52 | [TK]D-Fender | dps: Huh? |
| 12:56.05 | blitzrage | dps: there is no limit other than what your computer can do |
| 12:56.21 | blitzrage | dps: there is no cap or limit programmed into the code |
| 12:56.34 | dps | thank you blitzrage |
| 12:57.57 | [TK]D-Fender | blitzrage: Now you're all set! |
| 12:58.06 | blitzrage | :D |
| 12:58.56 | *** join/#asterisk Runlvl (n=juan@190.2.40.57) |
| 12:59.11 | blitzrage | don't use it all in one place now! |
| 12:59.32 | Runlvl | Hi guys! For spanish help , visit http://www.asterisk-la.org :-) |
| 12:59.47 | *** part/#asterisk ming_zym (n=ming_zym@121.0.31.121) |
| 13:01.14 | [TK]D-Fender | Runlvl: Coolo... I'm a little rusty. Do you teach Russian as well? |
| 13:01.54 | Runlvl | Runlvl, Spanish support for asterisk... |
| 13:02.02 | *** join/#asterisk mihinomenest (n=argh@66.255.220.22) |
| 13:02.25 | [TK]D-Fender | Runlvl: Kill-joy :( |
| 13:02.50 | _x86_ | roffle |
| 13:03.07 | Runlvl | [TK]D-Fender, I can help you in english anyway ;-) |
| 13:03.19 | _x86_ | Runlvl: tango las gotas los pantelones |
| 13:03.38 | _x86_ | Runlvl: the only help TK needs is mental... ;-) |
| 13:03.43 | Runlvl | _x86_, hahahaha |
| 13:04.45 | _x86_ | ;)4 |
| 13:04.59 | _x86_ | haha |
| 13:07.40 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 13:08.38 | *** join/#asterisk BadPacket (n=John@unaffiliated/badpacket) |
| 13:09.05 | dps | Guys, what's the signal protocol that cisco uses? |
| 13:09.09 | dps | Ski....? |
| 13:09.13 | dps | skiming? |
| 13:09.26 | blitzrage | skinny |
| 13:09.27 | blitzrage | SCCP |
| 13:09.46 | dps | Do you by any chance know if Asterisk can interact with it? |
| 13:09.52 | [TK]D-Fender | skanky |
| 13:09.56 | dps | without the use of h323 or sip trunks? |
| 13:10.05 | [TK]D-Fender | dps: Yes. |
| 13:10.32 | blitzrage | dps: yes -- it is an entirely separate protocol -- it does not 'use' h.323 or sip.... which are separate protocols of themselves |
| 13:10.41 | mvanbaak | chan_skinny works great here :) |
| 13:10.49 | blitzrage | it'd be like asking if I can use IAX2 without SIP |
| 13:10.58 | dps | yes i understand |
| 13:11.10 | mvanbaak | you can run IAX2 without SIP ???? |
| 13:11.18 | blitzrage | mvanbaak: sometimes |
| 13:11.24 | mvanbaak | ;) |
| 13:11.27 | blitzrage | mvanbaak: IAX2 really just packages itself in SIP msgs |
| 13:11.34 | mvanbaak | aaaaaaaaaah |
| 13:11.39 | blitzrage | SIP is the underlying protocol to IAX2! |
| 13:11.40 | mvanbaak | now it makes sense |
| 13:11.41 | blitzrage | :D |
| 13:11.52 | file | you could.... |
| 13:13.19 | [TK]D-Fender | Man bites dog, news at 11! |
| 13:15.19 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:18.48 | *** join/#asterisk Qwell (i=north@pdpc/sponsor/digium/Qwell) |
| 13:18.48 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 13:19.10 | beeew | i've read in the TFOT book that IAX is ideal for running heavy concurrent loads.. |
| 13:19.26 | beeew | is it worth the effort to switch to IAX vs SIP? |
| 13:19.51 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 13:20.28 | JT | that's misleading |
| 13:20.34 | JT | it's not good for heavy loads |
| 13:20.51 | JT | it saves you a little bandwidth for a small amount of concurrent calls |
| 13:20.58 | JT | you get problems when you try and push it |
| 13:21.07 | *** part/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 13:21.30 | beeew | i'm new to all of this JT.. |
| 13:21.40 | beeew | SIP is cool then? |
| 13:22.02 | [TK]D-Fender | beeew: Advisable |
| 13:22.26 | guillote_GNU | hi people, can i use asterisk as an sms server? |
| 13:22.35 | beeew | u guys aware of a good DID number that can handle big load? |
| 13:23.10 | beeew | a good DID company |
| 13:23.22 | [TK]D-Fender | beeew: Level3 |
| 13:23.32 | dps | any of you had any experience regarding the connection between asterisk and skinny? |
| 13:23.39 | dps | Not to phones |
| 13:23.45 | dps | but from Asterisk to call managers |
| 13:23.51 | Qwell | dps: not possible |
| 13:23.55 | [TK]D-Fender | dps: * only does Skinnk TO phones, not direct to CCM |
| 13:24.21 | dps | So there's not a skinny trunk... |
| 13:24.23 | dps | ok |
| 13:24.38 | beeew | D-Fender, level3.com loos like a fiber optics co. they do DID number service too? |
| 13:24.56 | JT | a fiber optics company, lol! |
| 13:25.03 | JT | i think you mean "a telco" |
| 13:25.05 | jordanb | level3 is a colo |
| 13:25.13 | JT | ... |
| 13:25.18 | JT | no it's a telco |
| 13:25.30 | JT | telecommunications company |
| 13:25.47 | beeew | : T sorry u had to spell that all out |
| 13:26.01 | wwalker | level 3 is first and foremost a bureaucracy, secondarily they provide data, telephony, and colocation services... |
| 13:26.38 | JT | every man and their dog can provide co-location |
| 13:26.41 | beeew | anyone holding the stock? |
| 13:26.41 | JT | even i do ;) |
| 13:26.43 | beeew | :P |
| 13:27.35 | beeew | looks like it's on sale.. |
| 13:31.45 | *** join/#asterisk unas2 (n=unas@77-57-8-95.dclient.hispeed.ch) |
| 13:32.03 | *** join/#asterisk fskrotzki (n=fskrot@host198.textwise.com) |
| 13:35.20 | unas2 | help needen: if i forware a call to my mobilephone, the call rings (on my mobilefone) but i cant hear the caller and he can't hear me?! . any ideas... i forward the call with exten => 101,5,Dial(NUMBER,30,tT) |
| 13:35.48 | unas2 | is the problem the ,30,tT? |
| 13:36.18 | waKKu | firewall ? |
| 13:36.57 | unas2 | hmm... there is a firewall.. but outgoing calls are working... |
| 13:37.05 | unas2 | only forwarding to my mobilephone |
| 13:37.09 | [TK]D-Fender | unas2: Behind NAT? |
| 13:37.15 | unas2 | if i call my mobilephone... no problem |
| 13:37.20 | unas2 | jep.. behinde nat |
| 13:37.26 | [TK]D-Fender | ~sipnat |
| 13:37.26 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 13:37.28 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^ |
| 13:37.57 | unas2 | but outgoing calls are perfect... so could it be a NAT problem? |
| 13:38.20 | [TK]D-Fender | Yes |
| 13:38.32 | unas2 | thanks... |
| 13:41.35 | roxlu | hi there |
| 13:42.05 | *** join/#asterisk qdk (n=qdk@193.164.155.113) |
| 13:42.11 | unas2 | so nat=1 and qualify=yes are enought? |
| 13:44.55 | roxlu | i've got a strange thing over here... I've got a sip account at budgetphone.nl. I've added the entries in my sip.conf and incoming calls work. |
| 13:45.30 | roxlu | Though outgoing calls not :( but..... when I use my budgetphone account directly in X-lite, connect and close the application, it suddenly starts working with my asterisk account??? |
| 13:45.56 | *** join/#asterisk Dirk- (n=a@82-33-155-212.cable.ubr04.wiga.blueyonder.co.uk) |
| 13:46.37 | Dirk- | <PROTECTED> |
| 13:49.47 | mvanbaak | roxlu: gheh, now there's a nice problem to debug |
| 13:49.56 | roxlu | hi mvanbaak ! |
| 13:50.12 | roxlu | yesterday incoming/outgoing seemed to work both with budgetphone |
| 13:51.23 | nexilus | Hey anyone here happen to know how to use arp to get the ip adress of a certain hw adress? |
| 13:51.50 | cpm | nexilus, you can do it the braindead way |
| 13:52.22 | nexilus | and whats that? |
| 13:52.28 | *** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-f661dba0fdc10fea) |
| 13:53.11 | cpm | broadcast ping the subnet you suspect, and check your arp cache before it expires for the mac address |
| 13:53.21 | cpm | pretty brain dead, |
| 13:54.09 | cpm | or use nmap -sP |
| 13:54.24 | mvanbaak | roxlu: and today it's not working ? |
| 13:55.05 | unas2 | if i can call from outbound my asterisk server behind nat, forware the call outbound (mobile phone), than we can hear each other... but if i fix a forward in my extensions... we can't here each other |
| 13:55.15 | unas2 | can't be a nat problem! |
| 13:55.38 | roxlu | mvanbaak: yes.... this morning incoming was the only thing that worked.. but no anymore ... strange... |
| 13:55.52 | *** join/#asterisk munmun (n=mun_mun@203.80.176.168) |
| 13:55.54 | [TK]D-Fender | unas2: "fix a forward"? |
| 13:55.59 | *** join/#asterisk saftsack (n=saftsack@s0933.vpn.hrz.tu-darmstadt.de) |
| 13:56.05 | mvanbaak | that's why I gave up on budgetphone |
| 13:56.14 | roxlu | mvanbaak: but I'm copying a backup now... |
| 13:56.14 | *** join/#asterisk geminidomino (n=ciro@65.41.157.192) |
| 13:56.35 | roxlu | mvanbaak: when I got it working I'll give you my configs |
| 13:57.11 | roxlu | okay outgoing works again :D |
| 13:57.26 | [TK]D-Fender | unas2: pastebin your sip.conf masking only passwords. |
| 13:57.27 | mvanbaak | nice |
| 13:57.28 | [TK]D-Fender | ~pb |
| 13:57.28 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 13:57.29 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^ |
| 13:58.13 | roxlu | mvanbaak: yesterday someone helped me a lot with this |
| 13:58.25 | mvanbaak | ah |
| 13:58.34 | roxlu | mvanbaak: and we split the budgetphone into to contexts (not totally sure what we did :) ) |
| 13:58.46 | geminidomino | Are there any known issues with the zaptel drivers on ubuntu Feisty? |
| 13:59.36 | *** join/#asterisk ussrback (n=MAX@81.95.160.147) |
| 13:59.37 | mvanbaak | not that I know |
| 13:59.58 | mvanbaak | roxlu: did you create a seperate type=user and type=peer in sip.conf ? |
| 14:00.05 | mvanbaak | because that's what was next on my list |
| 14:00.06 | roxlu | yes |
| 14:00.07 | geminidomino | bugger |
| 14:00.11 | roxlu | :-) |
| 14:02.55 | [TK]D-Fender | ~sipnat |
| 14:02.56 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 14:03.08 | roxlu | mvanbaak: when I call using my mobile, I see it in the CLI log:http://paste-it.net/4143 but no phone rings |
| 14:03.26 | unas2 | exten => 101,20,Dial(SIP/0792929258@41445001670,60,tT) (what means tT ... ?) sorry for that question.. but couldn't found a good man. |
| 14:03.57 | ussrback | Hi all |
| 14:04.03 | ManxPower | <PROTECTED> |
| 14:04.13 | ussrback | anyone uses perl for AGI scripts? |
| 14:04.23 | ussrback | i need some help |
| 14:04.48 | mvanbaak | where's the rest of the debug ? |
| 14:04.50 | creativx | unas2: cli show application dial |
| 14:04.58 | roxlu | thats it |
| 14:05.32 | ManxPower | roxlu: How are you today? |
| 14:05.40 | roxlu | Hi ManxPower ! |
| 14:05.52 | roxlu | mvanbaak: ManxPower is the one who helped me a lot!! |
| 14:06.12 | [TK]D-Fender | unas2: please rad the guide. You have NOT configured your system properly for NAT FORGET about your Dial statement for now. |
| 14:06.15 | roxlu | ManxPower: well.. this morning nothing worked anymore :( ... I put back my backup but only outgoing is working now |
| 14:06.48 | mvanbaak | roxlu: ManxPower is known to be helpfull with sip trouble |
| 14:06.50 | ManxPower | roxlu: give me about 30 mins. I have no finished my coffee. |
| 14:06.50 | ussrback | i use this statemant in my perl $AGI->exec("ChanSpy $channel|wW"); , but it gives error "Exec format error" 2" |
| 14:07.00 | roxlu | of course :-) |
| 14:07.05 | mvanbaak | he also gave me some tips to get xs4all working |
| 14:07.07 | roxlu | mvanbaak: yes indeed |
| 14:07.24 | ManxPower | ussrback: learn perl. Try $AGI->exec("ChanSpy", "$channel,wW"); |
| 14:07.29 | roxlu | I think it's strange providers don't provide a howto or an example config |
| 14:07.41 | ManxPower | all providers suck |
| 14:07.59 | mvanbaak | lol ManxPower |
| 14:08.02 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net) |
| 14:08.10 | roxlu | well.. at least budgetphone.nl assured me that they will put my config on their website when it works :-)... at least that is something |
| 14:09.05 | ManxPower | roxlu: do a "sip debug" and then do a failed call. put the CLI output and your sip.conf (sans passwords) on pastebin.ca |
| 14:09.13 | mvanbaak | roxlu: the weird part with that debug is that I dont see any calls to a dailplan statement |
| 14:09.36 | ussrback | ManxPower: I've already tried this, but no success |
| 14:09.43 | roxlu | mvanbaak: indeed.. I think there is something wrong with my sip.conf |
| 14:10.36 | mvanbaak | pastebin it please :) |
| 14:10.55 | roxlu | yes working on it :-) |
| 14:11.24 | mvanbaak | hhmm, I never seen anyone using the template functions in sip.conf (cept for my own config) |
| 14:12.49 | roxlu | there you go: http://pastebin.ca/747968 |
| 14:13.33 | ManxPower | roxlu: what calls are working and what calls are not? |
| 14:13.50 | *** part/#asterisk munmun (n=mun_mun@203.80.176.168) |
| 14:14.36 | ManxPower | roxlu: look at line 167 |
| 14:14.47 | ManxPower | ..er.. 176 |
| 14:15.12 | mvanbaak | 404 |
| 14:15.13 | mvanbaak | yay |
| 14:15.20 | roxlu | ManxPower: incoming calls arn't working |
| 14:15.29 | ManxPower | Looking for s in default (domain 84.107.142.180) |
| 14:15.36 | ManxPower | that is the problem. |
| 14:15.42 | roxlu | okay |
| 14:16.00 | roxlu | but the 's' is in [incoming] right? |
| 14:16.36 | ManxPower | roxlu: what is happening is that the incoming call is not matching anything in sip.conf so asterisk uses in the info in [general] |
| 14:16.44 | mvanbaak | this looks weird to me: |
| 14:16.47 | mvanbaak | Found peer 'budgetphone' |
| 14:16.49 | ManxPower | I will keep looking at the pastebin. |
| 14:16.57 | roxlu | Okay |
| 14:17.12 | mvanbaak | in my opinion it should find 3171XXXXXx |
| 14:17.17 | ManxPower | mvanbaak: correct. |
| 14:17.43 | roxlu | ManxPower: the only thing I changed as far as I remember, are the context names for internal phones (10/20) |
| 14:17.44 | mvanbaak | why not make that a peer as well ? |
| 14:17.53 | *** join/#asterisk rantsh (n=rantsh@190.36.185.139) |
| 14:18.02 | roxlu | though internal calls aren't working either now |
| 14:18.03 | rantsh | hello everybody |
| 14:18.05 | file | due to your configuration the remote side is matching against the budgetphone peer, and requesting authentication (because insecure is not set to do IP based matching) |
| 14:18.17 | ManxPower | mvanbaak: because I'm a traditionalist and by tradition asterisk does not accept calls from type=peer. |
| 14:18.21 | *** part/#asterisk lirakis (n=eric@69.24.142.1) |
| 14:18.23 | file | therefore the remote side probably freaks out and doesn't try to send the INVITE again with authentication, because in it's world their clients don't do that normally |
| 14:18.47 | mvanbaak | ManxPower: I only have: type=peer |
| 14:18.59 | mvanbaak | I dont have any type=user entries in my sip.conf |
| 14:19.08 | ManxPower | 31711111111 is type=user |
| 14:19.16 | mvanbaak | make it type=peer |
| 14:19.40 | mvanbaak | both my incoming and outgoing entries are 'type=peer' |
| 14:19.49 | [TK]D-Fender | UGH |
| 14:19.51 | *** join/#asterisk callguy (n=callguy@pool-72-70-79-233.bstnma.east.verizon.net) |
| 14:19.52 | ManxPower | roxlu: it can't hurt setting it to type=peer. |
| 14:19.58 | ManxPower | it's not working now afterall. |
| 14:19.59 | file | roxlu: add insecure=very to the budgetphone peer entry and see what happens |
| 14:20.08 | [TK]D-Fender | roxlu: It'd be working fine if you just SET YOUR CONTEXT in [budgetphone] ! |
| 14:20.21 | file | as for the 404 Not Found that is from the OPTIONS packet, checking to see if the Asterisk box is alive |
| 14:20.23 | ManxPower | [TK]D-Fender: do you also hate the fact that the whole type=peer/friend/user is a total and complete chaos |
| 14:20.26 | [TK]D-Fender | roxlu: Looking for s in default (domain 84.107.142.180) |
| 14:20.41 | *** join/#asterisk grandpapadot (n=null@mail.heavylogic.com) |
| 14:20.48 | [TK]D-Fender | roxlu: You only set a cotext in [genera], and it doesn't even EXIST. |
| 14:20.56 | mvanbaak | lol |
| 14:21.06 | ManxPower | [TK]D-Fender: the context in [general] should not exist |
| 14:21.09 | mvanbaak | I would ditch the 3171..... one |
| 14:21.12 | ManxPower | as he is not accepting unauthenticated calls. |
| 14:21.14 | [TK]D-Fender | roxlu: Go put "context=incoming" under [budgetphone] |
| 14:21.37 | ManxPower | roxlu: I'll give you a merged version, just a min |
| 14:21.40 | [TK]D-Fender | ManxPower: Well "whatever" to that, but he only needed 1 line to END this. |
| 14:21.45 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 14:21.57 | mvanbaak | lol |
| 14:21.59 | file | 2 lines actually |
| 14:22.06 | roxlu | Okay I've put context=incoming in [budgetphone] |
| 14:22.16 | mvanbaak | roxlu: also do as file said |
| 14:22.24 | mvanbaak | roxlu: add: insecure=very |
| 14:22.41 | ManxPower | roxlu: try this: http://pastebin.ca/747979 |
| 14:22.45 | roxlu | okay |
| 14:23.18 | ManxPower | In asterisk peer and user are sort of, but not really the same thing. |
| 14:23.23 | mvanbaak | and remove the 3171XXXXX ones |
| 14:23.25 | ManxPower | at least in 1.2 and 1.4 |
| 14:23.49 | roxlu | mvanbaak: what do you mean? |
| 14:24.00 | roxlu | ManxPower: looking at it right now |
| 14:24.04 | mvanbaak | you have two users in sip.conf |
| 14:24.07 | rantsh | I'm having some problems with get_variable in agi |
| 14:24.09 | mvanbaak | [3171XXXXX] |
| 14:24.13 | mvanbaak | and one commented out |
| 14:24.18 | mvanbaak | remove them both :) |
| 14:24.19 | ManxPower | roxlu: he means comment out the [3171XXXXX] section |
| 14:24.30 | roxlu | ah |
| 14:24.34 | rantsh | paste-binned some info in case anyone can check what might be the problem |
| 14:24.36 | rantsh | http://pastebin.com/d65f2dc68 |
| 14:24.36 | roxlu | okay, |
| 14:24.58 | roxlu | is order in [budgetphone] important? |
| 14:25.07 | mvanbaak | no |
| 14:25.08 | ManxPower | rantsh: you are not use Asterisk::AGI |
| 14:25.13 | ManxPower | roxlu: should not be |
| 14:25.22 | mvanbaak | only when you are trying to force codecs |
| 14:25.35 | file | rantsh: and you are assigning the dialplan variable "foo" to the variable $q, not $foo |
| 14:26.25 | roxlu | okay testing... |
| 14:26.53 | *** join/#asterisk Dan0maN_Work (n=dschuh@64.149.174.136) |
| 14:26.58 | rantsh | manxpower: what do you mean? |
| 14:27.07 | roxlu | hmm still not |
| 14:27.16 | roxlu | I'm gonna put back exactly what I had when it was working |
| 14:27.40 | rantsh | file: sorry, changed the var names to pastebinned but foo is actually $q at my script, my apologies |
| 14:27.41 | roxlu | though, I've got my softphone on my laptop connected directly to my budgetphone account.. can that be a problem? |
| 14:28.03 | mvanbaak | ehm, yeah |
| 14:28.13 | ManxPower | rantsh: I'm not a Perl Guru, but all my AGIs start with: use Asterisk::AGI; |
| 14:28.24 | ManxPower | roxlu: Yes! |
| 14:28.36 | ManxPower | roxlu: you can't have 2 devices connect to the same account for many providers |
| 14:28.42 | roxlu | so maybe it didn't work yesterday, but it only looked like it was working... (though it was redirecting calls to my other softphone) |
| 14:29.04 | *** part/#asterisk LukinoVoip (n=LukinoVo@host15-224-static.57-82-b.business.telecomitalia.it) |
| 14:29.13 | rantsh | manxpower: I'll try that, but it's always been enough for me to put AGI :p |
| 14:29.59 | ManxPower | rantsh: also you are not processing the start of the AGI |
| 14:30.00 | ManxPower | $AGI = new Asterisk::AGI; |
| 14:30.00 | ManxPower | %input = $AGI->ReadParse(); |
| 14:31.03 | ManxPower | Asterisk is sending you a bunch of stuff and your app is not processing it. Also, of course, even if everything was working you have the logic issue file told you about. |
| 14:31.53 | wwalker | How do I get asterisk to put var/run under the PREFIX directory? I used ./configure --prefix=/opt/asterisk-4-1.4.11 --localstatedir=/opt/asterisk-4-1.4.11/var --sysconfdir=/opt/asterisk-4-1.4.11/etc |
| 14:31.54 | rantsh | yup, that last one was a mistake of me trying to put standar var names on the pastebin, screwed up the pastebin but I checked the script and it is fine |
| 14:31.57 | *** join/#asterisk mugawuki (n=mugawuki@extranet.lehighgas.com) |
| 14:32.09 | wwalker | and I still get Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory |
| 14:33.21 | rantsh | thanks manxpower %input = $AGI->ReadParse(); did the magic |
| 14:33.30 | ManxPower | wwalker: then asterisk is NOT running |
| 14:33.35 | ussrback | Anybody knows correct syntax for perl agi ? when i execute $AGI-> exec('chanspy',"$channel|wW"); i got Exec format error |
| 14:33.37 | ManxPower | wwalker: start asterisk as "asterisk -cvvv" |
| 14:33.47 | rantsh | how do I access those variables? $input{agicallerid} ??? |
| 14:34.06 | wwalker | ManxPower: read |
| 14:34.58 | wwalker | during configure defaults.h is being modified showing that var/run belongs under the prefix : defaults.h:#define AST_RUN_DIR "/opt/asterisk-4-1.4.11/var/run" |
| 14:35.03 | ManxPower | rantsh: example: http://www.fnords.org/~eric/fax2email.txt |
| 14:35.42 | ManxPower | wwalker: but that is not what you said. |
| 14:36.00 | ManxPower | wwalker: check with #asterisk-dev as that sounds like a bug. But before you do that, check /etc/asterisk/asterisk.conf |
| 14:36.18 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 14:36.20 | roxlu | ManxPower: okay internal calls and outgoing is working... only incoming not.. |
| 14:36.27 | roxlu | I'll paste my configs AND CLI |
| 14:36.54 | mvanbaak | yes please |
| 14:37.28 | *** join/#asterisk TrentCreek (n=tjones14@cpe-70-117-207-168.rgv.res.rr.com) |
| 14:37.42 | ManxPower | roxlu: are you SURE your softphone is no longer running? |
| 14:37.52 | roxlu | yes |
| 14:38.02 | roxlu | well not the softphone connected to budgetphone |
| 14:38.05 | ManxPower | roxlu: chances are the softphone registered with the providdr and the provider is sending the calls to the internal ip if the softphone |
| 14:38.21 | roxlu | I was thinking somthing like that :( |
| 14:38.41 | ManxPower | roxlu: do a unload chan_sip.so and a load chan_sip.so in the CLI |
| 14:38.49 | ManxPower | that should make asterisk reregister |
| 14:39.02 | ManxPower | roxlu: did you make any NAT or firewall changes? |
| 14:39.06 | roxlu | http://pastebin.ca/747999 |
| 14:39.10 | roxlu | now |
| 14:39.12 | roxlu | no |
| 14:41.03 | mugawuki | If a SIP call comes into an asterisk node that's using ARA to store sip peers, the node should check the sip_peers table to find the fullcontact of the destination user, even if the node the call came into wasn't the one that the sip peer actually registered with, right? |
| 14:41.42 | mugawuki | For example, in a replicated ARA database situation |
| 14:41.49 | mvanbaak | roxlu: try this as sip.conf: http://pastebin.ca/748003 |
| 14:41.54 | roxlu | ManxPower: can you see something strange in my conf....... ahh |
| 14:42.35 | ManxPower | roxlu: try mvanbaak's bastardized usage of peer. |
| 14:42.47 | roxlu | haha okay |
| 14:43.21 | mvanbaak | ManxPower ;) |
| 14:43.29 | mvanbaak | it's working great for xs4all here |
| 14:43.50 | *** join/#asterisk Tili (n=tili@0.Red-83-53-150.dynamicIP.rima-tde.net) |
| 14:44.07 | roxlu | okay trying now |
| 14:44.31 | mvanbaak | ManxPower: if you look at the sip debug you see: 'found peer budgetphone' |
| 14:44.34 | ManxPower | mvanbaak: I come from a time when you HAD to have type=peer and type=user for many gateways. |
| 14:44.51 | [TK]D-Fender | ....... |
| 14:44.53 | mvanbaak | ManxPower: 1.0 ? |
| 14:44.54 | [TK]D-Fender | :| |
| 14:45.09 | mvanbaak | pre 1.0 ? |
| 14:45.10 | roxlu | mvanbaak: yes that works |
| 14:45.27 | mvanbaak | [in his best elvis voice] thank you thank you |
| 14:45.34 | roxlu | hahaha !! |
| 14:45.40 | roxlu | mvanbaak++ |
| 14:45.45 | roxlu | ManxPower++ |
| 14:45.55 | mvanbaak | sip-- |
| 14:47.06 | roxlu | but.. why is this working suddenly? |
| 14:47.33 | ManxPower | *scream* apparently the rainstorm sunday night BURIED one of the BellSouth/AT&T pedestals under several feet of mud and water. |
| 14:47.42 | ManxPower | they are calling for a backhow |
| 14:47.46 | ManxPower | backhoe |
| 14:48.05 | ManxPower | mvanbaak: long before 0.65 |
| 14:49.38 | k31th | afternoon |
| 14:49.42 | *** join/#asterisk Dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 14:50.23 | roxlu | mvanbaak: could you explain me a bit why it works now? |
| 14:50.35 | mvanbaak | sure |
| 14:50.51 | mvanbaak | in current asterisk, if a sip call comes in it starts looking for a peer |
| 14:50.58 | mvanbaak | it will find the [budgetphone] one |
| 14:51.28 | mvanbaak | like file said, without the 'insecure=very' asterisk sends a 'proxy auth required' |
| 14:51.30 | roxlu | okay... and when [1002] had type=peer as well? |
| 14:51.37 | roxlu | okay |
| 14:51.47 | mvanbaak | because the ip there is not the same as the ip the invite comes from |
| 14:51.58 | mvanbaak | so you fixed that by setting 'insecure=very' |
| 14:52.10 | mvanbaak | the second issue was what [TK]D-Fender said |
| 14:52.16 | mvanbaak | you had no context defined |
| 14:52.28 | [TK]D-Fender | ... |
| 14:52.31 | mvanbaak | so the calls ended up in a non-existant context |
| 14:52.45 | mvanbaak | adding 'context=incoming' fixed that problem for you |
| 14:52.54 | roxlu | okay |
| 14:53.04 | mvanbaak | the [3171XXXX] user was never used |
| 14:53.12 | mvanbaak | so adding stuff to that was useless |
| 14:53.46 | roxlu | so in "Jip en janneke taal", I call my VOIP number, asterisk sees an incoming request, it looks for a type=peer, finds it at the [budgetphone] ----> incoming (in extensions.conf) etc.. |
| 14:53.58 | mvanbaak | precies |
| 14:54.01 | roxlu | okay nice |
| 14:54.14 | roxlu | well.. than you've fixed the budgetphone.nl asterisk problemn :-) |
| 14:54.27 | mvanbaak | lol |
| 14:54.45 | mvanbaak | shows how much I had to learn in the days when I tried to fix it on my box |
| 14:54.50 | ManxPower | roxlu: you understand that adding context=incoming under [general] means any system on the planet route calls thru your system, right. |
| 14:55.06 | mvanbaak | ManxPower: he put it in his [budgetphone] peer |
| 14:55.11 | mvanbaak | not in the [general] |
| 14:55.14 | *** join/#asterisk Blueneon (n=blue@dsl-146-29-195.telkomadsl.co.za) |
| 14:55.19 | *** join/#asterisk etfonhomey (n=chatzill@mail.advancmed.org) |
| 14:55.22 | ManxPower | mvanbaak: Just making SURE. |
| 14:55.26 | mvanbaak | ok |
| 14:55.28 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 14:55.30 | roxlu | Thanks a lot !! |
| 14:55.39 | roxlu | nice people here in #asterisk :D |
| 14:55.43 | mvanbaak | my [general] has: context=default |
| 14:55.54 | ManxPower | why don't they just REMOVE type=user and friend???? |
| 14:55.55 | mvanbaak | because I allow enum calls in |
| 14:56.05 | mvanbaak | ManxPower: I wonder as wel |
| 14:56.10 | ManxPower | mvanbaak: but you accept calls from untrusted sources. |
| 14:56.12 | Blueneon | anyone know why on earth my calls hear silence when I start a threewaycall? If i press (R) they hear nothing, but if I transfer the call (R) + Ext they get onhold music... im using TDM400 |
| 14:56.28 | mvanbaak | ManxPower: yup, you have to when you want to be reached using your enum stuff |
| 14:56.35 | ManxPower | Blueneon: 1.2 or 1.4? do you have a MOH class for those channels |
| 14:56.39 | Blueneon | 1.4 |
| 14:56.45 | *** join/#asterisk saftsack (n=saftsack@pD9E079DB.dip.t-dialin.net) |
| 14:56.46 | roxlu | how can I secure my sip.conf so I can only allow calls from certain IPs ? (outgoing calls than) |
| 14:56.48 | Blueneon | yes the MOH are loaded |
| 14:56.53 | ManxPower | mvanbaak: we don't want weird people calling us. |
| 14:56.56 | ussrback | does voicemail database automaticaly supported in * 1.4? cause by wiki its necessary to copy mysql-vm-routines.h to apps directory and install it again. but i cant see any mysql-vm-routines.h file in my addons |
| 14:57.23 | mvanbaak | roxlu: the [1000] and [1002] you mean ? |
| 14:57.27 | Blueneon | or at least i think so, as like i mentioned music can be heard when the caller is transferred etc |
| 14:57.36 | roxlu | yes or .. when someone hacks my machine or something.. |
| 14:57.47 | ManxPower | Blueneon: look in /etc/asterisk/zapata.conf for musiconhold= settings |
| 14:57.51 | [TK]D-Fender | Blueneon: So they only get MoH after you enter your first digit AFTER the "R"? |
| 14:57.52 | mvanbaak | when someone hacks your machine there's nothing you can do |
| 14:57.58 | mvanbaak | because they will have root access |
| 14:58.05 | mvanbaak | so they can simply alter the config |
| 14:58.05 | roxlu | ah :-) okay |
| 14:58.08 | roxlu | true |
| 14:58.13 | roxlu | so now is everything secure? |
| 14:58.20 | mvanbaak | yeah |
| 14:58.23 | roxlu | nice |
| 14:58.24 | ManxPower | [TK]D-Fender: I suspect his channels don't have an moh class, but when the call comes in he uses "m" in his dial |
| 14:58.32 | Blueneon | TK: pretty much yes, and that i would think is because my dial plan says m in it |
| 14:58.32 | roxlu | I'll add some firewall rules to make it safe :D |
| 14:58.44 | Blueneon | Manx thats exactly right |
| 14:58.56 | file | ussrback: are you following a guide of some sort? |
| 14:58.56 | ManxPower | Blueneon: do what I told you to do. |
| 14:59.12 | ussrback | yes http://www.voip-info.org/tiki-index.php?page=Asterisk+voicemail+database |
| 14:59.12 | Blueneon | but i'm loading the class up in zapata.conf |
| 14:59.16 | Blueneon | musiconhold=default |
| 14:59.19 | Blueneon | ^ its there mate |
| 14:59.22 | [TK]D-Fender | Blueneon: After following ManxPower's directions, if it still isn't working, pastebin your dialplan and zapata. |
| 14:59.28 | [TK]D-Fender | ~pb |
| 14:59.29 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 14:59.31 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^^6 |
| 14:59.33 | Blueneon | guess i'll pastbin then |
| 14:59.34 | Blueneon | hehe |
| 14:59.40 | file | ussrback: find a new guide, if it is mentioning those then it is for an old old Asterisk version |
| 14:59.44 | mvanbaak | if you want to make sure the [1000] and [1002] can only connect from a speficic ip use something like this: http://pastebin.ca/748021 |
| 14:59.48 | [TK]D-Fender | Blueneon: And CLI output of a failed attempt |
| 15:00.22 | ussrback | i did not found any new guide . its only one for voicemail database |
| 15:00.34 | ManxPower | ussrback: I thought I told you that much of the information on the Wiki is old, outdated, or just plain wrong. |
| 15:00.47 | ManxPower | ussrback: there is nothing in the doc/ directory of Asterisk? |
| 15:01.03 | enalert | I have a Polycom 501 which is continually rebooting looking for a TFTP server that no longer exists, what's the best way to get it to boot up to the menu so I can access it via the web interface? (I have all passwords and such for the phone, just looking for some documentation) |
| 15:01.23 | ussrback | no |
| 15:01.34 | file | I disagree |
| 15:02.22 | [TK]D-Fender | enalert: Only reason for a continuous reboot cycle is if the configs it last loaded were corrupted. |
| 15:02.24 | mvanbaak | disagree on what ? |
| 15:02.30 | enalert | [TK]D-Fender, how do I clear it? |
| 15:02.33 | file | doc/extconfig.txt details how to store voicemail user information in a database (including what a schema should look like) and if you are using postgresql then doc/voicemail_odbc_postgresql.txt details how to store voicemail message themselves in Postgresql using ODBC |
| 15:02.44 | [TK]D-Fender | enalert: Provide it GOOD configs clearly... |
| 15:02.57 | [TK]D-Fender | enalert: Go set up a new provisioning server. |
| 15:03.00 | enalert | [TK]D-Fender, we're no longer running a TFTP server |
| 15:03.01 | enalert | damn |
| 15:03.07 | enalert | I was hoping you wern't going to say that |
| 15:03.16 | [TK]D-Fender | enalert: they support FTP, HTTP, etc... |
| 15:03.21 | _x86_ | it's insanely easy to setup a TFTP server |
| 15:03.23 | [TK]D-Fender | enalert: TFTP sucks |
| 15:03.35 | ussrback | @file: ok i'll try. Thanks |
| 15:03.42 | _x86_ | [TK]D-Fender: polycom phones can provision off of HTTP? |
| 15:03.46 | Blueneon | http://pastebin.com/m324ca696 |
| 15:03.53 | [TK]D-Fender | _x86_: yes, HTTPS, FTPS as well |
| 15:03.54 | coppice | what's wrong with TFTP? its an excellent bootstrapping scheme |
| 15:04.10 | _x86_ | [TK]D-Fender: i thought they only did FTP/TFTP/FTPS |
| 15:04.16 | [TK]D-Fender | coppice: You just like term because it sounds like bondage :p |
| 15:04.17 | Blueneon | [TK]D-Fender and ManxPower: ^ thats my zapata, extensions and CLI output |
| 15:04.25 | _x86_ | would be cool if they did SFTP :) |
| 15:05.06 | [TK]D-Fender | Blueneon: I want to see the ENTIRE call from start to finish, vewrbose 10 <--- |
| 15:05.10 | ManxPower | Blueneon: callprogress=yes could easily be causing the problem |
| 15:05.39 | Blueneon | let me remove callprogress and show the entire call |
| 15:05.47 | [TK]D-Fender | Blueneon: AND I want a 2nd call with MoH working through other means as well as your MoH config |
| 15:07.08 | Blueneon | http://pastebin.com/m2b5130ad |
| 15:07.14 | roxlu | how can I see the current running version of asterisk? (is there a CLI command?) |
| 15:08.20 | [TK]D-Fender | roxlu: "show version" |
| 15:08.28 | Blueneon | http://pastebin.com/m5642a7b <- MOH working on transfer |
| 15:08.41 | roxlu | Thanks |
| 15:08.45 | *** join/#asterisk seanbright (n=elixer@65.207.74.18) |
| 15:09.01 | Blueneon | http://pastebin.com/m67a3742d <- MOH config |
| 15:10.58 | Blueneon | hello? |
| 15:11.35 | *** join/#asterisk Blueneon (n=blue@dsl-146-29-195.telkomadsl.co.za) |
| 15:11.44 | Blueneon | think i got disconnected there |
| 15:11.46 | Blueneon | <Blueneon> http://pastebin.com/m5642a7b <- MOH working on transfer |
| 15:11.56 | ManxPower | Blueneon: you did not get disconnected |
| 15:12.15 | ManxPower | you just did not get a response in 2 seconds like you expected |
| 15:12.57 | ManxPower | Blueneon: just remember callprogress=yes is an alias for randomlydisconnectmycalls=yes |
| 15:13.02 | [TK]D-Fender | Blueneon: pastbin a failed attempt please. |
| 15:13.03 | Blueneon | no... i've been getting disconnected all day |
| 15:13.09 | Blueneon | so i assumed it happend again |
| 15:13.26 | Blueneon | [TK]D-Fender: i already did paste a failed attempt |
| 15:13.34 | [TK]D-Fender | Blueneon: Which link? |
| 15:14.09 | *** join/#asterisk Juggie (n=Juggie@CPE001601df17fb-CM000a73a18a20.cpe.net.cable.rogers.com) |
| 15:14.22 | Blueneon | http://pastebin.com/m324ca696 |
| 15:14.36 | Blueneon | right at the bottom u can see that there is never any music on hold started |
| 15:14.39 | [TK]D-Fender | Blueneon: one with ALL OF THE CALL please.. |
| 15:14.45 | Blueneon | k sec |
| 15:15.02 | [TK]D-Fender | Blueneon: Never come in here with jsut little bits & pieces. |
| 15:15.14 | [TK]D-Fender | Blueneon: Drives people crazy... |
| 15:15.24 | ManxPower | Blueneon: Do you work for the Bush Administration? They also always give partial information too. |
| 15:15.36 | roxlu | is it VoiceMail or Voicemail what I need to use? |
| 15:15.42 | [TK]D-Fender | ManxPower: No, but I'd like to make THEM glow in the dark... |
| 15:15.45 | Blueneon | http://pastebin.com/m5e549b50 |
| 15:15.49 | ManxPower | roxlu: doesn't matter. |
| 15:16.01 | roxlu | oh I thaught they vere case sensitive |
| 15:16.14 | ManxPower | roxlu: some stuff is, application names are not |
| 15:16.19 | roxlu | ah |
| 15:16.39 | Blueneon | as u can see on line 26 i press (R), but no music was ever started again |
| 15:17.06 | [TK]D-Fender | Blueneon: You have "musiconhold=default" in your zapata.conf and SHOULD have " musicclass=default" <------- |
| 15:17.19 | [TK]D-Fender | Blueneon: Get your config parms right. |
| 15:17.52 | roxlu | mvanbaak: do you have some dutch sound files? (for example the voicemail menu?) |
| 15:18.24 | Blueneon | still doesn work |
| 15:18.31 | [TK]D-Fender | Blueneon: You'll have to restart * |
| 15:18.43 | Blueneon | i did |
| 15:18.47 | Blueneon | (ofc) :) |
| 15:19.10 | [TK]D-Fender | Blueneon: Do a NEW call, internal only. No more outside meddling around. |
| 15:19.31 | Blueneon | same result |
| 15:19.44 | [TK]D-Fender | .......... |
| 15:19.47 | Blueneon | want the paste? |
| 15:19.50 | [TK]D-Fender | DUH! |
| 15:19.55 | ManxPower | Blueneon: stop and start asterisk or reload? |
| 15:19.57 | [TK]D-Fender | AAAAAAAAARRRRRRRRRRRGGGGGGGGGGGHHHHHHHHH |
| 15:20.01 | Blueneon | http://pastebin.com/m5525d4c5 |
| 15:20.03 | Blueneon | hehe |
| 15:20.21 | *** join/#asterisk af_ (n=getsmart@81-174-44-189.dynamic.ngi.it) |
| 15:20.25 | Blueneon | i stopped it from within CLI and started it again, i also did: service asterisk restart |
| 15:21.12 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 15:21.12 | [TK]D-Fender | Blueneon: pastebin "zap show channel 3" and repeat for 4 |
| 15:21.52 | [TK]D-Fender | Blueneon: Acutallly... ALL of them |
| 15:22.08 | Blueneon | http://pastebin.com/m26b0175c |
| 15:22.16 | Blueneon | err... that 3/4 |
| 15:22.25 | Blueneon | will do 1/2 aswell |
| 15:22.58 | Blueneon | http://pastebin.com/m23ccbcda |
| 15:23.24 | Blueneon | 1 and 4 are lines coming in, 2 and 3 are the internals |
| 15:23.38 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 15:23.44 | [TK]D-Fender | Blueneon: New PB of zapata.conf please. |
| 15:24.08 | ManxPower | Wow! GMAIL is now starting to support IMAP over SSL |
| 15:24.28 | ManxPower | FINALLY a reason to consider switching |
| 15:24.36 | Nugget | nice, it's embarassing for them that they haven't so far. |
| 15:24.47 | Blueneon | http://pastebin.com/mab3fc73 |
| 15:24.49 | Nugget | they need to start supporting smtp/ssl too |
| 15:25.06 | Juggie | they do i though |
| 15:25.10 | Juggie | *thought |
| 15:25.23 | Nugget | that's not what I'd heard, but I'll admit I've never verified. |
| 15:25.42 | ManxPower | Nugget: the IMAP setup instructions for Thunderbird shows authenticated SMTP setup w/TLS |
| 15:26.31 | ManxPower | Nugget: See http://mail.google.com/support/bin/answer.py?answer=77662 |
| 15:26.43 | [TK]D-Fender | Blueneon: Ok, I'm out of ideas right now.... |
| 15:26.48 | [TK]D-Fender | Blueneon: Keep those around. |
| 15:27.07 | Blueneon | okies, thanks for trying to help though, you're a star :) |
| 15:28.32 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 15:29.18 | Nugget | the only gmail.com I see in my maillog is spam, so I dunno. :) |
| 15:31.08 | Nugget | yeah, the gmail smtp servers are not doing ssl |
| 15:31.28 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-372dfabd1115a6ec) |
| 15:32.06 | mvanbaak | roxlu: no |
| 15:32.20 | mvanbaak | roxlu: but I remember there is a soundpack on the internet somewhere |
| 15:32.36 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:32.37 | ManxPower | Nugget: I guess I should report the documentation error to google then |
| 15:33.24 | ManxPower | Nugget: you are of course using port 587, right? |
| 15:33.24 | Nugget | it might be supported for senders, but for server-to-server they're not doing tls |
| 15:33.36 | Nugget | I don't have a gmail account. |
| 15:33.39 | ManxPower | as it says in the link I gave you |
| 15:33.54 | Nugget | I was just talking about server-to-server |
| 15:33.58 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.178.16) |
| 15:34.20 | *** part/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 15:34.58 | *** join/#asterisk _ys (n=yuri@236-069.nat.mns.ru) |
| 15:35.46 | ManxPower | Nugget: *shrug* I could care less about server to server. |
| 15:36.14 | *** join/#asterisk Schumie (n=Steve@cmarfw01.marlow.spinvox.com) |
| 15:36.28 | *** join/#asterisk cervajs2 (n=cervajs@cervajs.fpf.slu.cz) |
| 15:37.13 | cervajs2 | hi, someone who is using chan_ss7? |
| 15:39.13 | Blueneon | [TK]D-Fender: just figured out something interesting |
| 15:39.42 | Blueneon | When i use my IAX phone to call internal then press the Hold button the IAX software, the music is started |
| 15:39.58 | Blueneon | Wierd |
| 15:40.03 | *** join/#asterisk FlatFoot (n=bigflatf@80.88.192.83) |
| 15:42.12 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 15:42.12 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:43.22 | Blueneon | http://pastebin.com/m720effc3 |
| 15:43.31 | Blueneon | there is a paste of it working from IAX |
| 15:43.33 | Blueneon | :/ |
| 15:46.56 | *** join/#asterisk blq (n=Bl@dslb-088-067-019-040.pools.arcor-ip.net) |
| 15:47.07 | outtolunc | shouldn't you be using m(class) |
| 15:48.04 | Blueneon | ? |
| 15:48.09 | ManxPower | outtolunc: his issue is the caller does not hear MoH when ON HOLD. |
| 15:48.12 | outtolunc | in your dial line |
| 15:48.20 | ManxPower | i.e. during dialing in a 3-way call. |
| 15:48.21 | Blueneon | the dial is fine |
| 15:48.22 | outtolunc | ah |
| 15:48.25 | Blueneon | its only onhold |
| 15:48.30 | Blueneon | but with iax it works |
| 15:48.46 | ManxPower | Blueneon: Oddly, IAX2 and Zapata are totally different. |
| 15:48.46 | Blueneon | just my normal phones (R) doesnt seem to send the caller into music mode |
| 15:48.55 | ManxPower | outtolunc: I saw a similar issue with ringing on transfer. |
| 15:49.10 | ManxPower | Blueneon: call it FLASH instead of (R) |
| 15:49.15 | Blueneon | ok |
| 15:50.04 | Blueneon | on the normal phones the flash works in some ways but not in others, ie. when pressing flash i get the dialtone and i can start making another call, but the caller left on hold is left with silence. |
| 15:50.58 | Blueneon | so like manx said, I assume its asterisk not knowing its music on hold class to use. Albeit that it is set in zapata |
| 15:51.03 | disa-help | oh fun times |
| 15:51.05 | disa-help | firewalling asterisk |
| 15:51.09 | outtolunc | what asterisk version? |
| 15:51.11 | disa-help | anyone got any good docs/pointers? |
| 15:51.11 | disa-help | heh |
| 15:51.17 | disa-help | http://www.voip-info.org/wiki/view/Asterisk+firewall+rules <-- already been there |
| 15:51.33 | disa-help | seems that agents can get calls from the queue, but can dial out, after i've applied my rules |
| 15:51.49 | Blueneon | does flash send any type of DTMF? |
| 15:52.28 | bkruse | Blueneon: I do not believe so |
| 15:52.43 | Blueneon | do u think that the rxflash timing might need to be changed? |
| 15:53.13 | bkruse | Blueneon: for what reason? |
| 15:53.30 | *** join/#asterisk phix (i=threat@123-243-44-131.tpgi.com.au) |
| 15:53.35 | phix | hey |
| 15:53.59 | Blueneon | well perhaps asterisk isnt detecting the flash correctly |
| 15:54.13 | phix | Any suggestions for SIP Phones? |
| 15:54.18 | phix | Brand? Model? etc..? |
| 15:54.20 | mvanbaak | ~phones |
| 15:54.21 | jbot | phones is, like, http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. places ... |
| 15:54.22 | Blueneon | i've been reading that diff phones have diff times for the flash and thus the setting in asterisk |
| 15:54.45 | phix | mvanbaak: thank yuo |
| 15:55.37 | phix | mvanbaak: also, are there any protocols used / supported by asterisk and SIP / IAX phones that can use a global address book? |
| 15:55.45 | phix | or don't these things exist? |
| 15:55.55 | ManxPower | Blueneon: if the flash timing had to be changed then three-way calls would not work |
| 15:56.10 | Blueneon | makes sense i guess |
| 15:56.25 | Blueneon | i just dont understand why on earth its not working |
| 15:56.29 | phix | Some of the high end phones I have seen support XML, which I guess would be used to setup / use a centralised address book |
| 15:56.54 | ManxPower | Blueneon: perhaps you have discovered a big. |
| 15:56.58 | ManxPower | and a bug too |
| 15:57.06 | phix | ok so I should stay clear of Grandstream? |
| 15:57.14 | ManxPower | ~gs |
| 15:57.14 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 15:57.15 | phix | any particular reason? |
| 15:57.31 | phix | Blueneon: ok |
| 15:57.45 | lmadsen | experience |
| 15:57.50 | coppice | I go near Grandstream quite often |
| 15:58.16 | coppice | they are in Shenzhen science park :-) |
| 15:58.26 | phix | ok, and I suppose I should stear clear from any product that has Skype stickers on it? |
| 15:58.47 | lmadsen | oh no... skype is the best |
| 15:58.54 | disa-help | ~firewall |
| 15:58.54 | jbot | well, firewall is This is a form of Internet security that stands between a private network and the Internet. It is like a wall in that it can prevent unwanted traffic from passing either way. Some firewalls have proxy functions built in. In fact, the distinction between a firewall and a proxy is often blurry. Add in the differences and similarities between a ... |
| 15:59.01 | coppice | the best what? |
| 15:59.03 | disa-help | ~firewall asterisk |
| 15:59.10 | lmadsen | coppice: the best of the best! |
| 15:59.12 | lmadsen | </sarcasm> |
| 15:59.29 | phix | lmadsen: :P |
| 16:00.19 | phix | jbot: not really, it can stand between any network, not just between a private network and the Internet |
| 16:00.35 | *** join/#asterisk hfb (n=hfb@pool-71-106-219-180.lsanca.dsl-w.verizon.net) |
| 16:01.17 | phix | jbot: a firewall filters traffic, a proxy forwards and handles request on behalf (in proxy) of a client |
| 16:01.31 | phix | someone update jbot :) |
| 16:03.13 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 16:04.22 | FlatFoot | ~snom |
| 16:04.22 | jbot | i heard snom is like all German products. High quality, but wacky engineering. :) |
| 16:04.41 | FlatFoot | does jbot answer anything at all ? |
| 16:05.08 | coppice | All German engineering tends towards a Tiger Tank |
| 16:05.21 | FlatFoot | ~sausage |
| 16:05.22 | jbot | somebody said sausage was ground up animal parts stuffed into an sphincter, grilled so that you don't gag |
| 16:05.50 | Qwell | yeah, that's pretty accurate |
| 16:06.00 | Qwell | if not a bit one-sided |
| 16:06.28 | phix | coppice: heh |
| 16:06.45 | [TK]D-Fender | phix: Given your location & probable budget I'd sujjest Linksys |
| 16:06.46 | phix | coppice: anything big and metal :) |
| 16:06.49 | *** join/#asterisk fskrotzki (n=fskrot@host198.textwise.com) |
| 16:06.55 | [TK]D-Fender | suggest* |
| 16:07.49 | coppice | look at german cars. they go to italy and get cars designed with lots of window area. then they do a couple of local spins until they end up with slits. then they go back to italy, where the glass is put back |
| 16:08.18 | FlatFoot | coppice: ? |
| 16:08.27 | *** join/#asterisk bantu (n=Miranda@p54A32BBA.dip0.t-ipconnect.de) |
| 16:08.50 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 16:09.29 | phix | [TK]D-Fender: I am looking at them, they look interesting however I really want something minimal (I do not need the following: PPPoE, 2 ethernet ports / switch, DHCP server, router / NAT, etc.) |
| 16:11.24 | phix | However I would like a resonable sized dispaly (4 or more line LCD display), support for centralised address book (whether it be XML, WAP, SQL, LDAP, etc..), SIP (of corse :)), g.721 at least, hold and maybe transfer (but I don't mind pressing *2 or #2). |
| 16:11.59 | phix | I guess for enterprise stuff I would be looking at a Cisco? |
| 16:12.08 | phix | well Linksys == cisco sort of |
| 16:12.40 | Blueneon | [TK]D-Fender: why would my iax hold work with moh, but the zap not? |
| 16:12.42 | cpm | not at all. That's marketing. |
| 16:12.48 | phix | at least 2 lines |
| 16:12.56 | cpm | linksys is owned by cisco, but they are separate shops |
| 16:13.00 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 16:13.14 | phix | Blueneon: ummm you need to used framed MOH for ZAP? |
| 16:13.20 | coppice | linksys being a kinda 5 and dime |
| 16:13.24 | Blueneon | phix? |
| 16:13.29 | phix | cpm: yeah, cisco being more highend than linksys |
| 16:13.34 | Qwell | coppice: I'd think Cisco is the 5 and dime |
| 16:13.34 | cpm | a bit, yeah |
| 16:13.37 | phix | Blueneon: *shrugs* |
| 16:13.45 | Qwell | because they nickel and dime you to death |
| 16:13.51 | cpm | yup |
| 16:13.56 | coppice | Qwell: i think they are a well matched pair |
| 16:13.57 | phix | Blueneon: Looking at some one elses config they had music on hold set to framed mode |
| 16:14.19 | Blueneon | hmm |
| 16:14.22 | phix | Qwell: hmmmm, American |
| 16:14.34 | Blueneon | in musiconhold.conf? |
| 16:14.39 | Blueneon | or zapata.conf |
| 16:14.40 | Qwell | phix: hmm? |
| 16:14.42 | coppice | American as sweet and sour |
| 16:17.31 | Blueneon | phix i cant seem to find any settings for frame |
| 16:18.23 | *** join/#asterisk rpm (n=russell@75.155.167.90) |
| 16:19.02 | JT | Qwell: that's a very american way of talking |
| 16:19.07 | JT | <@Qwell> because they nickel and dime you to death |
| 16:19.21 | phix | Blueneon: hmmm ok, ignore me then :) |
| 16:19.28 | phix | ~softphone |
| 16:19.28 | jbot | something that should be drug out into the street and shot |
| 16:19.37 | phix | hmmm |
| 16:19.51 | k31th | what does that mean, another way of saying a ripp off or they are expensive? |
| 16:19.52 | phix | I am after a list of SIP clients for computer |
| 16:20.12 | phix | k31th: nfi, I am not American :) |
| 16:20.30 | De_Mon | why do I see debug messages on the console with 'core set debug 0' |
| 16:20.32 | phix | I don't even know which coin is worth more than the other |
| 16:20.42 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 16:21.22 | phix | I do know that AUD to USD is pratically 1:1 |
| 16:21.24 | phix | yay for me :P |
| 16:21.48 | k31th | UKP here... |
| 16:21.55 | Qwell | JT: howso? |
| 16:22.26 | phix | k31th: exchange rate from AUD to UKP is a bitch :( in saying that I am still comming over next year :) |
| 16:22.57 | k31th | yeah |
| 16:23.03 | k31th | i can imagen your pain. |
| 16:23.24 | phix | I wouldn't mind getting a job over there so I have something to bring back with me :) |
| 16:23.34 | k31th | lol |
| 16:23.47 | coppice | visiting the UK is bad, but working there is true masochism |
| 16:24.03 | phix | Whether it be duty free grog or UKP |
| 16:24.43 | phix | coppice: heh, It was good when I visited last year :) then again I don't think I would survive a winter |
| 16:24.57 | k31th | lol |
| 16:26.17 | k31th | do all you guys use asterisk with no web gui and do a manual install each time? |
| 16:26.38 | Nugget | That's the only way to do it and stay sane. |
| 16:26.59 | phix | k31th: I have setup asterisk twice, on both occasions I did not use a GUI |
| 16:27.38 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 16:27.45 | phix | Nugget: I quite like learning about the dial plan and other stuff |
| 16:28.21 | k31th | yeah, well i have installed from source but never really ran it, i have tried both elastix / freepbx / trixbox... i checked open ports on trixbox seemed to have just about every service running including ircd |
| 16:28.52 | phix | I am also quite sane, I hope, does talking to ones self == insanity? or only if you talk back? |
| 16:28.59 | k31th | so now after trying all those I am doing a source install on centos5... and i have "the Asterisk book" open |
| 16:29.23 | phix | k31th: heh, I don't like the idea of an asterisk distro |
| 16:29.28 | grandpapadot | Anyone know of a faq to auto-provision (tftp) a LinkSys PAP2T (unlocked)? The admin guide is quite vague ... |
| 16:29.28 | phix | I like Debian |
| 16:29.47 | phix | k31th: use a Debian based system :) |
| 16:29.54 | k31th | yeah im a debian user... |
| 16:30.01 | grandpapadot | debian rocks. |
| 16:30.04 | phix | yay |
| 16:30.15 | phix | k31th: so why use centos then? |
| 16:30.22 | phix | that uses yum right? |
| 16:31.28 | phix | hmmm, sleep |
| 16:32.54 | *** join/#asterisk tru_`z24 (n=truz_`24@74-129-166-232.dhcp.insightbb.com) |
| 16:33.41 | k31th | phix: well yeah you have a point. |
| 16:33.55 | k31th | might as well roll with debian on it as I am used to that. |
| 16:39.19 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 16:42.18 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:42.45 | [hC] | It used to work for me, to set up a hint on a SIP device like: hint,SIP/123@somehost and I could retrieve status of 123 from somehost, but now that doesnt seem to work anymore. How do other people do this? |
| 16:43.09 | Blueneon | phix, ManxPower, [TK]D-Fender, just to let you know, I've upgraded to 1.4.10 and it seems to have fixed the issue with the onhold music, thanks for all your help :) |
| 16:46.19 | *** join/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 16:47.43 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 16:48.41 | [TK]D-Fender | [hC]: Shouldn't |
| 16:49.17 | [hC] | [TK]D-Fender: shouldnt what? that shouldnt work? |
| 16:49.52 | [hC] | [TK]D-Fender: I just had the sip guest account enabled on both sides, and could have sip hints monitored like that between two asterisk boxes. It definitely worked at one point. |
| 16:49.54 | [TK]D-Fender | [hC]: Correct. AFAIK * can only report on things it has directo control over. * doesn't subscribe to random outside resources. |
| 16:51.17 | *** join/#asterisk angom (n=angom@201.143.89.82.dsl.dyn.telnor.net) |
| 16:52.11 | Mw3 | ~h323 |
| 16:52.11 | jbot | from memory, h323 is An ITU-T standard for packet-based multimedia communications systems. This standard defines the different multimedia entities that make up a multimedia system - Endpoint, Gateway, Multipoint Conferencing Unit (MCU), and Gatekeeper - and their interaction. This standard is used for many voice-over-IP applications, and is heavily dependent on ... |
| 16:53.34 | roxlu | ManxPower: are you still here? |
| 16:54.23 | *** join/#asterisk grugnog (n=Grugnog@ip68-108-241-244.sb.sd.cox.net) |
| 16:54.34 | *** part/#asterisk Dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 16:55.37 | grugnog | Does anyone have any clues on why MEETME_EXIT_CONTEXT isn't working for me on 1.4? |
| 16:55.49 | grugnog | I am calling MeetMe with the 'X' parameter |
| 16:56.15 | grugnog | there is an extension in the destination context |
| 16:56.46 | grugnog | I have checked that DTMF is working - it's just that it's not doing anything... |
| 17:12.55 | roxlu | Does someone knows how I can record phone calls using a filename like [phonenumber/name-time] ? |
| 17:14.04 | *** join/#asterisk ussrback (n=MAX@80.92.183.84) |
| 17:15.15 | *** join/#asterisk hrmphh (i=patrick@notchill.com) |
| 17:15.23 | hrmphh | Oct 24 09:31:07.69: [ 1140]: RECV: reject TCF (zero run too short, min 1200) |
| 17:15.27 | hrmphh | any idea what the means from hylafax? |
| 17:16.17 | hrmphh | this url (http://techpubs.sgi.com/library/tpl/cgi-bin/getdoc.cgi?coll=fw&db=man&fname=/usr/freeware/catman/u_man/cat4/hylafax-log.Z) says that The received TCF was deemed unacceptable because there was too high a percentage of non-zero data in it. |
| 17:17.55 | *** join/#asterisk shtoom (n=shtoom@59.93.123.172) |
| 17:18.53 | shtoom | Hi I am not able to record call using Monitor application I am getting the following error , can some one please help me |
| 17:18.55 | shtoom | [Oct 24 22:45:57] WARNING[14869]: file.c:194 ast_writestream: Unable to translate to format wav, source format g729[Oct 24 22:45:57] WARNING[14869]: channel.c:2935 ast_write: Failed to write data to channel monitor write stream |
| 17:19.17 | mvanbaak | shtoom: error tells you all |
| 17:19.31 | mvanbaak | you dont have g729 codec so asterisk cannot trascode it to wav |
| 17:20.28 | shtoom | but I've checked with make menuselect and all format_*codectypes* are enabled |
| 17:20.32 | *** join/#asterisk mitcheloc (n=mitchel@ppp-67-126-240-11.dsl.irvnca.pacbell.net) |
| 17:20.34 | roxlu | mvanbaak: can you help me with recording calls? |
| 17:21.02 | mvanbaak | sure |
| 17:21.05 | shtoom | mvanbaak : Thanks for your response I'll check for g729 codec |
| 17:21.06 | mvanbaak | mixmonitor |
| 17:21.17 | roxlu | mvanbaak: not Record()? |
| 17:21.21 | mvanbaak | shtoom: it's a commercial codec you have to buy from digium |
| 17:21.34 | mvanbaak | roxlu: record is for recording soundfiles |
| 17:21.40 | shtoom | mvanbaak : But the call is happening between sip client and trunk thru asterisk |
| 17:21.49 | shtoom | with out any trouble |
| 17:21.54 | roxlu | mvanbaak: what do you mean? |
| 17:22.15 | mvanbaak | shtoom: asterisk can do passthru of codecs it cannot transcode. so both endpoints need to understand g729 |
| 17:22.39 | mvanbaak | roxlu: record is not meant to record calls between channels |
| 17:22.46 | roxlu | okay |
| 17:22.58 | mvanbaak | record is meant to record your ivr prompts using a phone |
| 17:23.09 | mvanbaak | it will act as an endpoint |
| 17:23.17 | roxlu | okay (don't know what it is).. |
| 17:23.18 | roxlu | ah |
| 17:23.21 | mvanbaak | mixmonitor will simply monitor the channel and write a file |
| 17:23.26 | shtoom | mvanbaak: I got you. Thanks for your time :) |
| 17:23.29 | mvanbaak | ivr == voice menu |
| 17:23.36 | *** join/#asterisk iPod-nano (n=iPod-nan@c-68-43-60-193.hsd1.mi.comcast.net) |
| 17:23.42 | iPod-nano | Got a Linksys PAP2. |
| 17:23.49 | roxlu | so like: exten => _XX,n,MixMonitor(${NUMBER}-{TIME}.gsm) ? |
| 17:24.17 | mvanbaak | roxlu: put that before the actual Dial |
| 17:24.24 | roxlu | okay |
| 17:24.44 | roxlu | what's a smarty naming rule for this? number and time?) |
| 17:25.16 | iPod-nano | If anybody can help me, my PAP2 works for a while, but then it won't ring if I call it. |
| 17:25.24 | mvanbaak | roxlu: whatever you want |
| 17:25.30 | roxlu | okay |
| 17:25.40 | *** join/#asterisk teknoprep (n=Chris@unaffiliated/teknoprep) |
| 17:25.41 | mvanbaak | it all depends on what you want to do with the recording |
| 17:25.42 | teknoprep | hey all |
| 17:25.46 | mvanbaak | we use a lot of variables in it |
| 17:25.54 | roxlu | can you show me some? |
| 17:25.57 | teknoprep | is it possible to use a Skype phone as a softphone or hardphone with Asterisk |
| 17:26.05 | teknoprep | possibly with IDEfisk ? |
| 17:26.06 | mvanbaak | so our webbased CRM application can link it to the right customer based on number etc |
| 17:27.33 | iPod-nano | The PAP2 was a steal on eBay. 23 bucks. |
| 17:27.35 | *** join/#asterisk hijacked (n=argh@66.255.220.22) |
| 17:27.38 | mvanbaak | ${customerid}_${queuename}_${timestamp} |
| 17:27.49 | mvanbaak | that's one of the schemas we use |
| 17:28.01 | roxlu | do you have an example which records the phonecall like [number]-[time].wav/gsm ? |
| 17:29.31 | *** join/#asterisk Keltus (i=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 17:29.40 | mvanbaak | ${CALLERID(num)}-${STRFTIME(${EPOCH},,%C%y%m%d%H%M)} |
| 17:29.46 | *** join/#asterisk absd (n=chatzill@124-168-3-1.dyn.iinet.net.au) |
| 17:30.05 | absd | Can anyone point me to a good beginners doc on setting up basic call routing in asterisk? I've got a basic conceptual understanding of extensions and contexts but need some simple worked examples preferably |
| 17:30.14 | roxlu | ah thanks! ... and one last thing :-) Can I dynamically set the caller-name based n the incoming phone number? |
| 17:30.16 | mvanbaak | ~book |
| 17:30.17 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 17:30.26 | mvanbaak | absd: look there |
| 17:30.36 | mvanbaak | roxlu: yeah |
| 17:30.48 | roxlu | using a PHP script even? |
| 17:32.05 | EnterSadman | there is 3rd day that im fighting against the exec command in AGI :) , but without any success. anyone can help me with that? |
| 17:32.26 | mvanbaak | roxlu: I use a script for that indee |
| 17:32.33 | roxlu | okay nice |
| 17:32.44 | mvanbaak | have a look at Agi |
| 17:32.49 | roxlu | ok |
| 17:32.53 | teknoprep | so anyone know of a way to use a skype phone with asterisk ? |
| 17:32.54 | [TK]D-Fender | roxlu: Stop now and go read THE BOOK. |
| 17:33.00 | [TK]D-Fender | ~skype |
| 17:33.01 | jbot | Skype is the bastard child of telephony. It uses a proprietary protocol for which only commercial channel drivers exist, all of which are DISGUSTING hacks at best. Forget about using Skype with Asterisk... |
| 17:33.03 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^ |
| 17:33.19 | teknoprep | WOW |
| 17:33.26 | mvanbaak | better: forget about Skype at all |
| 17:33.34 | teknoprep | i don't want to use skype tho |
| 17:33.39 | teknoprep | just one of there USB phones |
| 17:33.43 | teknoprep | well a 3rd party usb phone |
| 17:33.43 | absd | whoa, is that oreilly book free to download, or is that url somehow kinda dodgy and letting me download something I shouldn't see? |
| 17:34.03 | teknoprep | also is there software like IDEfisk that runs on PocketPC ? |
| 17:34.07 | [TK]D-Fender | teknoprep: Even worse |
| 17:34.12 | roxlu | [TK]D-Fender: i'm gonna buy the book friday indeed |
| 17:34.14 | mvanbaak | absd: it's the PDF of the book |
| 17:34.20 | [TK]D-Fender | teknoprep: No you won't be finding drivers to support it... |
| 17:34.24 | teknoprep | ok |
| 17:34.38 | absd | mvanbaak: sure, but can I read it legally, or do I need to purchase it somehow? |
| 17:34.41 | [TK]D-Fender | teknoprep: Get googling. |
| 17:34.46 | teknoprep | oh [TK]D-Fender were you talking about IDEfisk for pocketpc ? |
| 17:35.09 | mvanbaak | absd: it's released under a creative common license |
| 17:35.10 | [TK]D-Fender | teknoprep: First was about Skype. thats a dead issue. There are SIP phones for PPC |
| 17:35.31 | teknoprep | i was looking for something ... preferably IDEfist |
| 17:35.35 | teknoprep | or whaterver its called now |
| 17:35.38 | teknoprep | Zoiper |
| 17:35.45 | mvanbaak | schmoiper |
| 17:36.45 | teknoprep | last question.. what is idefisk home page |
| 17:36.56 | teknoprep | i seem to find 8000 places to download it but no homepage |
| 17:37.13 | teknoprep | wow... nvm... www.zoiper.com aye? |
| 17:38.00 | [TK]D-Fender | teknoprep: Yes |
| 17:38.13 | *** join/#asterisk guillote_GNU (n=guillote@190.7.27.17) |
| 17:39.12 | EnterSadman | my $tempchan1 = "$channel"."|"."wW"; |
| 17:39.13 | EnterSadman | <PROTECTED> |
| 17:40.56 | *** join/#asterisk syle2 (n=blag@unaffiliated/syle) |
| 17:41.39 | *** join/#asterisk gardo (n=gardo@121.97.178.73) |
| 17:42.08 | hrmphh | anyone have problems receiving faxes w/hylafax? looks like some remote modems/faxmachines are incompatible? |
| 17:43.44 | Alan_Hicks | Arg! This is crazy. I'm reading through all the documentation on the Polycom Soundpoint IP 320, but I can't figure out how to put a "." in an IP address stored inside the phone. |
| 17:44.27 | Alan_Hicks | Anyone know how this is done? I need to enter 172.16.200.1, but I can only enter 172162001. |
| 17:44.51 | *** join/#asterisk TestMaster (n=Dan@xplr-ts-w10-208-114-135-205.barrettxplore.com) |
| 17:44.54 | TestMaster | hello all |
| 17:45.01 | *** join/#asterisk chris_1 (n=chris@ng1.kurtkrenn.com) |
| 17:45.04 | TestMaster | anyone here good with php/billing increments |
| 17:45.04 | syle2 | testmaster! |
| 17:45.17 | [hC] | Alan_Hicks: i normally hit '*' for a dot on phones. |
| 17:46.13 | syle2 | hmm nope, thought that would run faster in c so did it there |
| 17:46.46 | TestMaster | syle2 would you be able to show me how you did it in C? |
| 17:47.29 | syle2 | wrote it over a year ago can;t remember, but was just some simple math formula i put together and some loops |
| 17:47.50 | TestMaster | see its something i need to figure out is the math formula |
| 17:50.23 | *** join/#asterisk Braxus (n=bhsieh@66.147.214.164) |
| 17:52.04 | syle2 | its just trial and error really |
| 17:52.11 | roxlu | mvanbaak: I'm testing my outgoing calls, can you maybe check if this is correct: http://paste-it.net/e0c072c |
| 17:52.15 | syle2 | start with something and keep building on it |
| 17:54.53 | mvanbaak | roxlu: put the mixmonitor before the dial |
| 17:54.57 | mvanbaak | now it will never be called |
| 17:55.16 | absd | mvanbaak: thanks for the link to that asterisk book -- freakin awesome... I've got this nailed now... cheers. |
| 17:55.23 | roxlu | mvanbaak: I tried that, but than my outgoing calls won't work |
| 17:55.23 | mvanbaak | hang on, going to make coffee |
| 17:55.29 | *** join/#asterisk mocker (n=user@198.247.173.227) |
| 17:55.33 | roxlu | mvanbaak: good idea :-) |
| 17:56.02 | mocker | Is Polycom BLF dependent on other Polycoms? Or can a Polycom phone see when a Cisco is on the line? |
| 17:56.14 | mocker | (I know that Cisco can't see when Polycom is) |
| 17:56.44 | [TK]D-Fender | mocker: thats because Cisco SIP doesn't support presence. |
| 17:56.48 | absd | Alan_Hicks: some devices that don't use dots accept notation of 12 digits (ie 172016200001 for the IP you gave) |
| 17:56.58 | [TK]D-Fender | mocker: * is what sends out presence info. |
| 17:57.21 | mocker | [TK]D-Fender: So in theory, one Polycom should be able to see when other Ciscos are on the phone? |
| 17:57.21 | [TK]D-Fender | mocker: so any phone that supports presence can support any kind of device * can report on. |
| 17:57.30 | mocker | Oh, except Ciscos suck. |
| 17:57.33 | [TK]D-Fender | mocker: Yes, not just in theory |
| 17:57.50 | mocker | Ahh, so even w/ Cisco's suck it will work? |
| 17:59.08 | [TK]D-Fender | mocker: I've already answered your question COMPLETELY. |
| 17:59.19 | [TK]D-Fender | mocker: Stop trying to reword it fruitlessly |
| 17:59.53 | mocker | [TK]D-Fender: With Cisco being so crappy, presence information work will it not? |
| 18:00.25 | [TK]D-Fender | mocker: Cisco doesn't inform ASTERISK <- ZIts the other way around. |
| 18:00.40 | mocker | [TK]D-Fender: Last one was a joke. :) |
| 18:01.26 | mvanbaak | in sccp it works ;) |
| 18:01.42 | mvanbaak | and also in chan_skinny |
| 18:02.37 | Alan_Hicks | absd: That's impossible. How does it know that 172162001 means 172.16.200.1 or 172.162.0.1? |
| 18:03.06 | Qwell | Alan_Hicks: note the 0's |
| 18:03.07 | absd | Alan_Hicks: notice I put extra 0's in such that it's 172 016 200 001 |
| 18:03.15 | Alan_Hicks | OH! |
| 18:03.21 | Alan_Hicks | Thanks. |
| 18:03.21 | roxlu | mvanbaak: when I put the MixMonitor as the first line, I can't maky any outgoing calls |
| 18:03.34 | absd | Alan_Hicks: no guarantees, just I've seen that work... :) worth a shot |
| 18:04.15 | Alan_Hicks | That'll play hell when ipv6 gets here. |
| 18:04.32 | mvanbaak | LOL Alan_Hicks |
| 18:04.34 | mvanbaak | yeah |
| 18:04.57 | mvanbaak | You are IPv6 2001:888:152c:0:21b:77ff:fe6e:1d3f |
| 18:04.59 | mvanbaak | lol |
| 18:05.01 | Alan_Hicks | They'll have to figure out some way to make a ":" character or I'll give up! :^) |
| 18:05.10 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 18:05.11 | Qwell | mvanbaak: 2001:888:152c::21b:77ff:fe6e:1d3f |
| 18:05.16 | Qwell | drop the 0s :D |
| 18:05.49 | mvanbaak | this is what php is giving me in $_SERVER["REMOTE_ADDR"] |
| 18:05.55 | Qwell | then it's broken |
| 18:06.00 | mvanbaak | duh |
| 18:06.03 | mvanbaak | it's by design |
| 18:06.18 | absd | it'd suck doing hex conversions to decimal so you could type it on a phone handset anyway |
| 18:06.29 | Alan_Hicks | Exactly. |
| 18:06.30 | mvanbaak | hahahahaha |
| 18:06.38 | mvanbaak | typing ipv6 on a phone handset |
| 18:06.57 | Qwell | it's only 32 chars (plus colons) |
| 18:06.59 | mvanbaak | da horror |
| 18:07.09 | Alan_Hicks | 32 hex characters you mean. |
| 18:07.19 | Qwell | a char is a char is a char |
| 18:07.24 | mvanbaak | I'm glad there's something called DNS |
| 18:07.31 | Qwell | mvanbaak: what's DNS? |
| 18:07.56 | mvanbaak | ;) |
| 18:08.00 | Alan_Hicks | Yeah, but when I have to hit "333" just to make an "e", that's a lot of characters. |
| 18:08.19 | absd | and how would it know you didn't mean "333" rather than "e" |
| 18:08.27 | mvanbaak | T9 |
| 18:08.36 | absd | ugly |
| 18:08.42 | Alan_Hicks | Exactly. You'd have to toggle back and forth between numerical and alphabetic input. |
| 18:09.57 | Qwell | 16 digit keypads |
| 18:10.04 | Qwell | we've already got ABCD, just add EF |
| 18:10.35 | Alan_Hicks | That'll work, but I can already see the problems with end-users. |
| 18:10.41 | roxlu | mvanbaak: do you know why my outgoing calls stop working when I add the MixMonitor as first? |
| 18:10.54 | Qwell | Alan_Hicks: why are users dialing by IP address in the first place? |
| 18:10.54 | Alan_Hicks | "I swear I dialed 1-888-deadbeef but it never rang!" |
| 18:11.03 | mvanbaak | gimme 15 minutes to drink coffee ok |
| 18:11.12 | Alan_Hicks | Qwell: I'm just learning here and exploring the workings of the phone right now. |
| 18:11.20 | roxlu | oh sorry |
| 18:11.31 | mvanbaak | no worries |
| 18:12.02 | TestMaster | anyone here know php/billing increments and can help me out this into a working math formula? |
| 18:12.02 | absd | I want SOME company to make a geek phone... with a chording keyboard, or even just a morse-code reader.... a one-buttoned phone... Morse isn't too hard to learn to 20 to 30wpm |
| 18:12.30 | roxlu | TestMaster: whats you question |
| 18:12.40 | TestMaster | I cant figure out how to do 30/6 |
| 18:12.53 | roxlu | divide 30 by 6? |
| 18:12.56 | Qwell | 5 |
| 18:13.42 | TestMaster | roxlu ok but the first 30 seconds are billed at 30 seconds no matter what right. what i cant figure out is how to put that in to a working php script |
| 18:14.18 | roxlu | so, only the first 30 seconds? or per 30 seconds? |
| 18:14.21 | Qwell | if time > 30, time -= 30, billinc += 5, billinc += time / 6 |
| 18:14.49 | Qwell | else billinc += 5 |
| 18:14.57 | Qwell | something |
| 18:15.07 | TestMaster | ok let me try that in php second |
| 18:15.43 | Qwell | you need to add a billing increment by 5 every time, regardless of whether time is more than 30 seconds. if it is more than 30 seconds, subtract 30 |
| 18:15.59 | grandpapadot | modula? |
| 18:16.17 | rpm | people that use qmail need to be shot. |
| 18:16.31 | Nugget | yes. |
| 18:16.34 | TestMaster | Qwell ok. |
| 18:16.42 | Nugget | qmail was the best thing going in 1997. it hasn't changed but the world has. |
| 18:16.49 | roxlu | rpm: why? (not tham using amail) |
| 18:17.10 | Nugget | now qmail is the source of endless frustration and pain for the rest of the internet |
| 18:17.15 | rpm | roxlu, i can't even tell if that is a question. |
| 18:17.31 | roxlu | rpm: hahaha what do you use? |
| 18:17.34 | Alan_Hicks | Friends don't let friends use qmail. |
| 18:17.43 | rpm | roxlu, postfix or sendmail. |
| 18:17.51 | roxlu | okay |
| 18:18.28 | *** join/#asterisk tsgbill (n=chatzill@h207.210.28.71.ip.alltel.net) |
| 18:18.46 | Alan_Hicks | I prefer postfix myself, and I stay as far away from anything using qmail as humanly possible. |
| 18:19.10 | mvanbaak | sendmail is a pain as well |
| 18:19.18 | mvanbaak | using m4 as config language |
| 18:19.33 | *** join/#asterisk dug (n=chatzill@c-76-102-23-25.hsd1.ca.comcast.net) |
| 18:19.52 | rpm | sendmail is great despite its history. |
| 18:20.03 | *** part/#asterisk ussrback (n=MAX@80.92.183.84) |
| 18:20.04 | *** join/#asterisk tru_`z24 (n=truz_`24@74-129-166-232.dhcp.insightbb.com) |
| 18:21.13 | dug | does asterisk set the timezone for the emails anywhere? http://www.pastebin.ca/748248 you can see my machine is set to pdt (-0700) but the Date on the email is +0000? I see in voicemail.conf there is a tz setting but is is ";" out |
| 18:21.23 | mvanbaak | roxlu: I'm back |
| 18:21.24 | mvanbaak | :) |
| 18:22.29 | roxlu | yes!!! |
| 18:22.40 | rpm | I'm front. |
| 18:23.01 | mvanbaak | lol |
| 18:23.02 | rpm | </troll> |
| 18:23.04 | roxlu | http://paste-it.net/e0c072c when I put the MixMonitor line as first (1) and than the dial as second (n), I can't call out anymore |
| 18:23.18 | mvanbaak | maybe you need the monitor application :) |
| 18:23.33 | roxlu | why? |
| 18:23.34 | mvanbaak | http://www.voip-info.org/wiki/view/Asterisk+cmd+Monitor |
| 18:24.01 | mvanbaak | I have no idea |
| 18:24.06 | mvanbaak | we only do inbound recording |
| 18:24.40 | mvanbaak | but I do know the way you put it in that past-it it will not work |
| 18:24.47 | mvanbaak | because first you do the call handling etc |
| 18:24.57 | mvanbaak | and when the call is done you go to mixmonitor |
| 18:25.12 | roxlu | yes I changed that.. so the first line has the mixmonitor |
| 18:25.22 | seanbright | dug: no, the timezone is not specified on e-mail generation |
| 18:25.28 | *** join/#asterisk rantsh (n=rantsh@190.36.185.139) |
| 18:25.37 | mvanbaak | roxlu: look at this: http://www.voip-info.org/wiki/index.php?page=Monitor+setup+sample |
| 18:26.26 | rantsh | hey people |
| 18:26.26 | roxlu | mvanbaak: the Monitor alone works |
| 18:26.42 | dug | seanbright: why is the timezone on the email different than the system tz/time? |
| 18:26.54 | rantsh | anyone knows where I can find the addons and GUI for download??? the link in asterisk.org is broken |
| 18:27.18 | seanbright | dug: i do not know. |
| 18:27.18 | Qwell | rantsh: what link? |
| 18:27.48 | seanbright | dug: what version of asterisk? 1.4? |
| 18:28.01 | dug | 1.4 |
| 18:28.20 | rantsh | the addons download link |
| 18:28.30 | seanbright | then it should be using your timezone into |
| 18:28.30 | Qwell | rantsh: where? |
| 18:28.31 | seanbright | err |
| 18:28.36 | rantsh | sends me to this page http://downloads.digium.com/pub/asterisk/ and it doesn't load |
| 18:28.46 | seanbright | dug: then it should be using your timezone info (i was wrong before) |
| 18:28.51 | Qwell | oh, right |
| 18:28.52 | rantsh | http://www.asterisk.org/downloads and click |
| 18:28.58 | rantsh | modules and addons |
| 18:29.05 | mvanbaak | downloads is on the svn box as well ? |
| 18:29.07 | Qwell | yeah, that's on a server that's currently down... lemme find the cname for the other box |
| 18:29.11 | Qwell | mvanbaak: it's one of the mirrors |
| 18:29.16 | rantsh | thanks |
| 18:29.17 | mvanbaak | ah |
| 18:29.51 | k31th | evening |
| 18:30.00 | *** join/#asterisk geminidomino (n=ciro@65.41.157.192) |
| 18:30.02 | seanbright | dug: run this at the CLI "voicemail show zones" |
| 18:30.25 | geminidomino | dumb question: anyone else having trouble hitting the digium svn and ftp servers? |
| 18:30.36 | Qwell | geminidomino: yes |
| 18:30.47 | mvanbaak | geminidomino: yeah, it's dead |
| 18:31.01 | k31th | what happend to the *-current.tar.gz |
| 18:31.02 | mvanbaak | they are working on it |
| 18:31.03 | geminidomino | ok, then it's not just me. Thank you... Anyone know of any mirrors offhand? |
| 18:31.16 | roxlu | mvanbaak: hmm it works :-) (even with mixmonitor) |
| 18:31.18 | dug | seanbright: http://www.pastebin.ca/748274 |
| 18:31.18 | mvanbaak | Qwell is digging up the cname |
| 18:31.23 | *** join/#asterisk dlynes_laptop (n=dlynes@216.251.149.66) |
| 18:31.27 | *** join/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 18:31.30 | mvanbaak | roxlu: yeah, mixmonitor should work as well |
| 18:31.33 | Qwell | I know the name of the box, but not what it should be called, heh |
| 18:31.38 | mvanbaak | lol |
| 18:32.07 | rantsh | Qwell: mirros http://downloads.digium.com/pub/ and http://ftp2.digium.com/pub/ are down too |
| 18:32.07 | seanbright | dug: and you are central? |
| 18:32.07 | Qwell | rantsh: same box |
| 18:32.07 | dug | pacific |
| 18:32.07 | Qwell | downloads should be a roundrobin |
| 18:32.10 | rantsh | Qwell: HAHA so much for a mirror hehehehe |
| 18:32.13 | roxlu | mvanbaak: I used a -> instead of => |
| 18:32.15 | seanbright | of course... :) you need to create a new zone line, and then set your tz= value to that |
| 18:32.16 | Qwell | somebody dig CNAME it, would ya? |
| 18:32.37 | geminidomino | ... |
| 18:32.43 | geminidomino | CNAME ftp.digium.com |
| 18:32.46 | mvanbaak | ;; ANSWER SECTION: |
| 18:32.46 | mvanbaak | downloads.digium.com. 3600 IN CNAME ftp.digium.com. |
| 18:32.46 | mvanbaak | ftp.digium.com. 3600 IN A 216.27.40.102 |
| 18:32.54 | rantsh | Qwell sorry for my ignorance but how do you that? |
| 18:33.09 | dug | seanbright: so set tz=pacific in voicemail.conf |
| 18:33.16 | deeperror | is this due to the outage @ bandwidth.com? |
| 18:33.31 | seanbright | dug: yes, but you have to create a 'pacific' line under [zonemessages] |
| 18:33.32 | mvanbaak | Qwell: I had one, till you guys moved to this php tool to track downloads |
| 18:33.33 | Qwell | deeperror: is there an outage at bandwidth? |
| 18:33.36 | mvanbaak | that's when I gave up |
| 18:33.45 | Qwell | mvanbaak: I mean two mirrors we ran |
| 18:33.49 | mvanbaak | ah |
| 18:33.56 | mvanbaak | remove the php stuff |
| 18:34.02 | mvanbaak | and you'll get a danish mirror for free |
| 18:34.10 | mvanbaak | or gimme rsync access :) |
| 18:34.15 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 18:34.16 | Qwell | deeperror: how out are we talking here? |
| 18:34.24 | deeperror | Qwell: i've been getting e-mails from them regarding some type of major outage |
| 18:34.37 | k31th | dos? |
| 18:34.53 | roxlu | I there support for remote addressbooks with voip/asterisk? |
| 18:35.19 | deeperror | it seems that was an issue with voip service last night for 2h 10m |
| 18:35.24 | mvanbaak | Qwell: what state is the mantis box located ? |
| 18:35.35 | deeperror | • The root cause of the service outage was attributed to successive primary and redundant device failures in California and Virginia, which are part of our underlying service provider’s network. The failures occurred in rapid succession. The initial cause of these failures is known and has been corrected. Continuing analysis by service provider focusing on improving time to repair is ongoing. |
| 18:35.36 | Qwell | mvanbaak: one of the carolinas |
| 18:35.51 | Qwell | deeperror: nice |
| 18:36.25 | mvanbaak | bandwidth.com has a map where you can check outages |
| 18:36.31 | mvanbaak | ;; ANSWER SECTION: |
| 18:36.31 | mvanbaak | downloads.digium.com. 3600 IN CNAME ftp.digium.com. |
| 18:36.31 | mvanbaak | ftp.digium.com. 3600 IN A 216.27.40.102 |
| 18:36.33 | mvanbaak | oops |
| 18:36.36 | mvanbaak | http://bandwidth.com/content/services?page=availability |
| 18:36.37 | dug | seanbright: under zonemessages the format is "eastern=America/New_York|'vm-received' Q 'digits/at' IMp" do I need to create a pacific=America/Los_Angles| .... or just a tz=pacific? |
| 18:37.13 | mvanbaak | roxlu: I created my own |
| 18:37.16 | seanbright | dug: pacific=Americ... |
| 18:37.17 | mvanbaak | very simple |
| 18:37.28 | mvanbaak | small mysql database |
| 18:37.38 | mvanbaak | a php script that is called using Agi() |
| 18:37.40 | mvanbaak | that's it |
| 18:38.02 | mvanbaak | at work we integrated it with our CRM |
| 18:38.06 | *** join/#asterisk jaike (n=jaike@203.177.199.188) |
| 18:38.11 | *** join/#asterisk callguy (n=callguy@pool-72-70-79-233.bstnma.east.verizon.net) |
| 18:38.11 | k31th | so am i right in thinking i cant download asterisk right now. |
| 18:38.18 | mvanbaak | k31th: indeed |
| 18:38.26 | *** join/#asterisk KuJaX (n=kuj@customtrading.dsl.xmission.com) |
| 18:38.28 | geminidomino | k31th: That's the long and short of it |
| 18:38.33 | deeperror | so if i'm trying to install zaptel it trys to download some extras is there any way to mirror that or point to a mirror? |
| 18:38.40 | k31th | yeah, well damit lol |
| 18:38.55 | mvanbaak | deeperror: Qwell is looking for a mirror |
| 18:38.58 | k31th | i had planned a install this evening, doh. |
| 18:39.05 | Qwell | mvanbaak: I don't think there is a second one anymore. |
| 18:39.11 | mvanbaak | ouch ! |
| 18:39.16 | Qwell | I can't find it anyhow |
| 18:39.24 | mvanbaak | what ? |
| 18:39.25 | deeperror | i've got current copies of the main file |
| 18:39.26 | mvanbaak | you lost it ? |
| 18:39.34 | Qwell | I thought I knew the box it was on, but... |
| 18:39.44 | mvanbaak | call the wife |
| 18:39.51 | mvanbaak | they are way better in finding stuff ;) |
| 18:39.53 | Qwell | heh |
| 18:41.08 | k31th | well i googled the file name i cant find it any where |
| 18:41.28 | deeperror | how long has downloads.digium been offline? |
| 18:41.57 | dug | seanbright: can I set a default tz or do I have to set it for each mailbox? |
| 18:42.11 | file | 4 or 5 hours? |
| 18:42.25 | k31th | shit |
| 18:42.27 | seanbright | dug: you can set the default |
| 18:42.31 | seanbright | dug: under [general] |
| 18:42.59 | *** topic/#asterisk by Qwell -> Asterisk: The Open Source Telephony Application Platform -=- Asterisk 1.4.13 (Oct. 10, 2007) Addons 1.2.8, 1.4.4 (Oct. 16, 2007) -=- Join #asterisknow or #asterisk-gui for AsteriskNOW and Asterisk-GUI info -=- Join #asterisk-commits to monitor svn changes -=- Join #freepbx for freepbx/#trixbox for trixbox support. -=- http://www.digium.com/en/company/switchvox-acquisition-faq.php -=- Mantis/Public SVN/Web SVN/download mirror is down. We know. |
| 18:43.28 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 18:44.21 | *** join/#asterisk tiav (n=tiav@ram94-3-82-225-11-215.fbx.proxad.net) |
| 18:49.19 | [hC] | Qwell: omgwtfbbq, did you know that svn is down? |
| 18:49.22 | [hC] | :) |
| 18:49.30 | k31th | lol |
| 18:49.32 | Qwell | [hC]: not internally it isn't :P |
| 18:49.46 | [hC] | ITS A CONSPIRACY!! |
| 18:50.00 | Qwell | sorry, you'll just have to buy BE |
| 18:50.10 | Qwell | err, did I say that out loud? |
| 18:50.14 | k31th | BE? |
| 18:50.22 | k31th | business edition ? |
| 18:50.25 | Qwell | k31th: Asterisk Business Edition |
| 18:51.22 | k31th | forced marketing ? |
| 18:51.32 | *** join/#asterisk rantsh (n=rantsh@190.36.185.139) |
| 18:52.23 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 18:53.01 | *** join/#asterisk juanjoc (n=juanjoc@200.69.219.113) |
| 18:54.05 | jaike | am having DTMF problems with sonus voip gateways, dtmfmode=rfc2833 wont work. anyone experienced this problem? |
| 18:54.33 | *** part/#asterisk geminidomino (n=ciro@65.41.157.192) |
| 18:55.51 | *** join/#asterisk zerohalo (n=zeroHalo@pool-72-70-79-233.bstnma.east.verizon.net) |
| 18:55.56 | *** join/#asterisk ectospasm (n=ectospas@c-68-62-219-11.hsd1.al.comcast.net) |
| 18:56.05 | *** join/#asterisk Siya (n=djerk@217-195-248-251.dsl.easynet.nl) |
| 18:56.12 | Siya | hello |
| 18:56.21 | ManxPower | jaike: So you have a GrandStream Phone <-> Asterisk <-> Sonus Gateway <-> PRI ? |
| 18:56.24 | k31th | olleh |
| 18:56.26 | *** join/#asterisk Schumie (i=SteveWri@cpc2-rdng2-0-0-cust382.winn.cable.ntl.com) |
| 18:56.35 | Siya | Anyone who can point me to a mirror of svn.digium.com |
| 18:56.40 | dug | seanbright: still no luck added tz=pacific in general and pacific=America/Los_Angeles|'vm-received' Q 'digits/at' IMp and none of my extensions have tz set... |
| 18:56.42 | k31th | lol... |
| 18:56.44 | Siya | hiya k31th |
| 18:56.53 | jaike | ManxPower: polycom |
| 18:57.04 | jaike | ManxPower: polycom 301s and 430s |
| 18:57.05 | dug | seanbright: and restarted asterisk ;) |
| 18:57.05 | seanbright | dug: have you reloaded? |
| 18:57.06 | ManxPower | jaike: other than that, it is correct? |
| 18:57.07 | mvanbaak | Siya: there are none |
| 18:57.09 | jaike | yes |
| 18:57.12 | seanbright | dug: ah |
| 18:57.19 | k31th | no one know knows of a mirror |
| 18:57.20 | Siya | mvanbaak: bummer |
| 18:57.21 | seanbright | dug: well sorry i couldn't help |
| 18:57.30 | Siya | mvanbaak: goeie avond |
| 18:57.31 | *** join/#asterisk Op3r (n=edwin@121.97.179.227) |
| 18:57.35 | *** join/#asterisk dlynes_laptop (n=dlynes@216.251.149.66) |
| 18:57.37 | ManxPower | jaike: so you have dtmfmode=rfc2833 for BOTH the Polycoms AND the Sonus entries in sip.conf? |
| 18:57.40 | mvanbaak | goeie avond |
| 18:57.43 | dug | I know I set the timezone in asterisk to -0700 somwhere |
| 18:57.46 | Siya | :) |
| 18:58.10 | jaike | ManxPower: yes, rfc2833 for all |
| 18:58.19 | ManxPower | jaike: I assume you set the Sonus for RFC2833 as well, right? |
| 18:58.20 | *** part/#asterisk Speedy2 (n=Javier_6@cpe-66-75-4-134.san.res.rr.com) |
| 18:58.36 | ManxPower | The default dtmfmode for the polycoms is rfc2833 unless you were stupid and changed it. |
| 18:58.41 | Siya | mvanbaak: would you happen to know how long it's been down? |
| 18:58.54 | Qwell | Siya: 4-5 hours |
| 18:59.00 | *** join/#asterisk trippss (n=ss@72-20-150-196.dhcp.cmts1.phonoscope.net) |
| 18:59.02 | jaike | ManxPower: our provider commpartners is migrating from telica to sonus |
| 18:59.28 | ManxPower | jaike: if the Sonus is set for INBAND or INFO and Asterisk is set for RFC2833 then don't expect DTMF to work |
| 19:00.00 | jaike | ManxPower: ok. will check with the provider |
| 19:00.03 | jaike | thanks |
| 19:00.09 | ManxPower | jaike: what codec between asterisk and the gateway? |
| 19:00.32 | *** join/#asterisk dlynes_laptop (n=dlynes@216.251.149.66) |
| 19:01.46 | jaike | ManxPower: g729. we need to use it to save on bandwidth, but wont work with inband |
| 19:02.44 | *** join/#asterisk Xenon3DN (n=Xenon@mail.3dnature.com) |
| 19:02.54 | *** join/#asterisk twilson (i=terry@nat/digium/x-4b9543587b681f3e) |
| 19:03.35 | roxlu | ManxPower: how can I grant access to my asterisk for an external user? (do I need to change the type=xxx ?) |
| 19:04.17 | deeperror | type=friend ? |
| 19:04.35 | roxlu | I don't know.. I have that, but the login fails |
| 19:04.43 | *** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 19:04.46 | generalhan | hey all |
| 19:04.51 | deeperror | can you see the connection from CLI? |
| 19:05.17 | Siya | Qwell: did you see the solution to the dual registries from same phone/ata? |
| 19:05.20 | ManxPower | roxlu: you set them up with an entry in sip.conf just like your other sip phones. |
| 19:05.30 | roxlu | an nothing else? |
| 19:05.44 | ManxPower | roxlu: you are not using NAT or your budgetphone calls would not work |
| 19:05.58 | generalhan | my boss wants to get some form of extensive logging with out phone system and im trying to find out if something like this has already been made. we need things like, how long a caller is in the queue before it is picked up. And how long a rep is on the phone with someone |
| 19:06.02 | ManxPower | roxlu: asterisk does not have the concept of "local" or "remote" users. They are JUST users. |
| 19:06.06 | *** join/#asterisk bigwilson (n=tim@ppp-70-251-246-162.dsl.rcsntx.swbell.net) |
| 19:06.08 | roxlu | ah okay |
| 19:06.14 | ManxPower | the only exception to that is if NAT is involved and it is not involved in your case. |
| 19:06.23 | generalhan | s/out/our/ |
| 19:06.36 | *** join/#asterisk Op3r (n=edwin@121.97.179.227) |
| 19:07.25 | roxlu | the other user gets an 408 error |
| 19:07.29 | roxlu | using x-lite |
| 19:07.42 | *** join/#asterisk krondorl (n=chatzill@tfi1meg.1meg.golden.net) |
| 19:08.07 | krondorl | Allo allo!! |
| 19:08.18 | ManxPower | roxlu: if the REMOTE side is behind NAT then you need nat=yes for their entry. |
| 19:08.27 | ManxPower | roxlu: 408 is Authentication Required? |
| 19:08.41 | roxlu | yes hee needs to authenticate |
| 19:09.04 | ManxPower | roxlu: ALL SIP calls start out unauthenticated, the server sends back a 408, the client connects again using the auth info |
| 19:09.33 | roxlu | okay |
| 19:09.41 | ManxPower | so simply seeing a 408 does not really mean anything other than the client did try to connect. |
| 19:09.53 | ManxPower | just remember that [theinfohere] in sip.conf must match the username the client is sending |
| 19:10.25 | roxlu | yes |
| 19:13.12 | *** join/#asterisk bantu (n=Miranda@p54A32BBA.dip0.t-ipconnect.de) |
| 19:15.12 | ManxPower | with sip debug off, you should still be seeing messages indicating the cause of the failed stuff (usually registration info messages) |
| 19:15.29 | ManxPower | sip no debug (or whatever the 1.4 version of that command) |
| 19:15.51 | dlynes_laptop | sip set debug off |
| 19:16.47 | k31th | jesus what did they do to that mirror kill it |
| 19:17.03 | grandpapadot | Anyone know how to make a phone to phone call with a LinkSys PAP2 (dual-port) and two analog phones using anything but g711a? |
| 19:17.22 | dlynes_laptop | k31th: they already know about the problem |
| 19:17.39 | dlynes_laptop | k31th: type /topic |
| 19:18.03 | k31th | i know, |
| 19:18.09 | k31th | they know i know |
| 19:18.20 | k31th | i was just saying |
| 19:18.25 | dlynes_laptop | your name should be kenny |
| 19:18.31 | ManxPower | k31th: what is so gosh darn time sensitive that you need access to Digium svn RIGHT NOW? |
| 19:18.37 | dlynes_laptop | Just like the south park dood |
| 19:18.52 | k31th | uhh wtf? |
| 19:19.01 | dlynes_laptop | cartman |
| 19:19.06 | k31th | i dont want to access the digium svn |
| 19:19.07 | mocker | Wow, gmail adds imap support. |
| 19:19.15 | k31th | ohhh i see |
| 19:19.25 | dlynes_laptop | mocker: they've had it for a while |
| 19:19.38 | ManxPower | dlynes_laptop: it's not on my gmail account. 8-( |
| 19:19.54 | dlynes_laptop | ManxPower: really? I've been using it for about 3 months now |
| 19:20.11 | ManxPower | dlynes_laptop: What e-mail client are you using to connect to google mail via IMAP? |
| 19:20.12 | Qwell | dlynes_laptop: pop or imap? |
| 19:20.19 | mocker | http://mail.google.com/mail/help/about_whatsnew.html |
| 19:20.20 | Qwell | imap was just added |
| 19:20.21 | dlynes_laptop | ManxPower: thunderbird |
| 19:20.25 | mocker | Says 'Just launched' |
| 19:20.28 | Qwell | and only on some accounts |
| 19:20.46 | ManxPower | dlynes_laptop: and you are not using the "GMAIL" account type, but the IMAP account type? |
| 19:21.00 | dlynes_laptop | ManxPower: oh...maybe it's the gmail account type then :) |
| 19:21.04 | Qwell | still no imap on mine |
| 19:21.06 | ManxPower | *nod* |
| 19:21.12 | dlynes_laptop | ManxPower: it just seemed to be exactly like imap :) |
| 19:21.19 | ManxPower | Qwell: me neither. IMAP w/SSL is what could make me switch to gmail |
| 19:21.25 | Qwell | tls |
| 19:21.34 | ManxPower | dlynes_laptop: until you use a client without special support for Gmail |
| 19:21.52 | ManxPower | Qwell: SSL, TLS, whatever it is. |
| 19:22.10 | dlynes_laptop | ManxPower: it's supported pop3 for a while now, too....but I had no use for pop3 |
| 19:22.17 | ManxPower | me neither |
| 19:22.23 | dlynes_laptop | ManxPower: i use the smtp once in a while, though |
| 19:22.51 | dlynes_laptop | ManxPower: it modifies your from address on all outbound mail though, to reflect your gmail address |
| 19:23.35 | ManxPower | dlynes_laptop: only if you don't tell it not to. |
| 19:23.37 | k31th | not if you hiost your domain with them it wont. |
| 19:23.42 | k31th | hosrt |
| 19:23.45 | k31th | host! |
| 19:23.48 | dlynes_laptop | ManxPower: oh...is that a new option? |
| 19:23.50 | k31th | bad lagg |
| 19:24.01 | ManxPower | My significant other uses gmail all the time and his From address is his @fnords.org address. |
| 19:24.07 | dlynes_laptop | ManxPower: i've had it set for that for probably two years now, and never checked new options on that since |
| 19:24.17 | generalhan | can anyone help me to figure out how i would get the amount of time a caller was in a queue before someone picked up ? |
| 19:24.49 | davevg-btwtech | generalhan, check queue_log |
| 19:24.53 | ManxPower | dlynes_laptop: Settings/Accounts/Send Mail As |
| 19:24.55 | [TK]D-Fender | generalhan: Its all very clear in the queue log as to hold long the hold time was when answered... |
| 19:25.01 | dlynes_laptop | ManxPower: thanks |
| 19:25.17 | generalhan | queue_log huh, thanks ill check into that |
| 19:25.18 | *** join/#asterisk destructure (n=kwatz@66.193.229.254) |
| 19:25.32 | *** join/#asterisk iPod-nano (n=iPod-nan@c-68-43-60-193.hsd1.mi.comcast.net) |
| 19:25.56 | iPod-nano | Can anybody give me the correct model numbers for an intel modem card that I can modify? |
| 19:26.13 | Qwell | modify? I'm sure you could modify any of them |
| 19:26.25 | Qwell | what do you want it to actually do? |
| 19:26.31 | *** join/#asterisk rtasterisk (n=mozveren@se167-1-82-242-148-65.fbx.proxad.net) |
| 19:26.32 | iPod-nano | Act as an FXO interface. |
| 19:26.34 | rtasterisk | hello all |
| 19:26.37 | Qwell | iPod-nano: save your time |
| 19:26.38 | ManxPower | Qwell: prolly magically turn it into an X101P |
| 19:26.40 | Qwell | buy a real card |
| 19:26.43 | Qwell | ~cheap |
| 19:26.44 | jbot | cheap is, like, a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 19:26.51 | outtolunc | he's probably referring to the diff between the HW and SW versions |
| 19:27.04 | rtasterisk | I tried asterisk 1.4 and dont understand why the priority scheduling changed |
| 19:27.13 | Qwell | rtasterisk: priority scheduling? |
| 19:27.16 | iPod-nano | I'm referring to this: http://www.voip-info.org/tiki-index.php?page=X100P+clone |
| 19:27.28 | Qwell | iPod-nano: yes, they are junk. Don't waste your time on them. |
| 19:27.29 | rtasterisk | an error is now must treated by a variable s-STATUS .. |
| 19:27.35 | [TK]D-Fender | iPod-nano: If you're going off the wiki and trying to find cheap junk.... |
| 19:27.38 | [TK]D-Fender | ~wglwat |
| 19:27.38 | jbot | extra, extra, read all about it, wglwat is well, good luck with all that |
| 19:27.43 | rtasterisk | before its was +101 to handle an error |
| 19:27.56 | Qwell | rtasterisk: priority jumping was silly |
| 19:28.00 | ManxPower | iPod-nano: it lists the numbers right there! |
| 19:28.04 | Qwell | there is a far better way to handle it |
| 19:28.11 | rtasterisk | why silly ? |
| 19:28.23 | ManxPower | however, those chips have not been made for several years |
| 19:28.56 | ManxPower | rtasterisk: no, the result of Dial() is in DIALSTATUS and HANGUPCAUSE |
| 19:29.13 | *** part/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 19:29.17 | rtasterisk | I think use a variable as a extension "s-BUSY" to handle a error, is maybe more silly |
| 19:29.23 | rtasterisk | than priority jumping |
| 19:29.32 | iPod-nano | Well I searched for one of the numbers on eBay and got a few results, but I don't know for sure if that's what I actually want. |
| 19:29.37 | ManxPower | rtasterisk: Um, you are reading std-exten wrong |
| 19:30.01 | ManxPower | rtasterisk: See the Goto(s-${DIALSTATUS},1) ? That jumps to s-WHATEVERDIALSTATUSEVALUATEDTO |
| 19:30.10 | ManxPower | iPod-nano: we cannot help you. |
| 19:30.14 | rtasterisk | Yes I now |
| 19:30.34 | rtasterisk | but you assume return value is now a extension ... |
| 19:30.37 | ManxPower | so if the dest was busy it would evaluate to Goto(s-BUSY,1) |
| 19:30.39 | rtasterisk | its not very clear |
| 19:30.43 | rtasterisk | I think |
| 19:30.46 | [TK]D-Fender | rtasterisk: no, we are ENSURING that it is. |
| 19:30.48 | Qwell | how is it less clear than +101? |
| 19:30.55 | rtasterisk | now :) |
| 19:30.57 | rtasterisk | no |
| 19:31.17 | ManxPower | not a single one of my current macros use a Goto(s-${DIALSTATUS},1) |
| 19:31.21 | rtasterisk | for my point of view, its the same level :) |
| 19:31.25 | [TK]D-Fender | rtasterisk: And who says +101 is an ERROR? What about other status'? Why would one be more special than the other? Now THAT would be silly. |
| 19:31.42 | ManxPower | rtasterisk: no, because with +101 you could only handle ONCE condition, now you can handle any number of conditions |
| 19:31.48 | Qwell | [TK]D-Fender: status'? nice |
| 19:32.02 | Qwell | [TK]D-Fender: I would've gone with statii myself |
| 19:32.09 | ManxPower | +101 was a bad idea to start out with and I'm glad it is finally gone in 1.4 |
| 19:32.29 | rtasterisk | I think EAL is much better than extensions.conf |
| 19:32.39 | rtasterisk | Why dont use EAL as standard ? |
| 19:32.44 | Qwell | rtasterisk: and how would you have done priority jumping with AEL? |
| 19:32.44 | *** join/#asterisk AsTeRiSk_1 (n=A@190.80.139.29) |
| 19:32.59 | [TK]D-Fender | Qwell: Yes, were it in the dictionary :) |
| 19:33.44 | rtasterisk | :) |
| 19:33.45 | [TK]D-Fender | rtasterisk: AEL (get it right, its only 3 friggen letters) gets parsed back to extensions.conf language anyways. It doesn't actually do MORE. |
| 19:34.19 | rtasterisk | I know but AEL seems to present a better logic ... |
| 19:34.51 | outtolunc | TK's point is that 'logic' boils down to standard dialplan code |
| 19:35.05 | [TK]D-Fender | outtolunc: Correct |
| 19:35.31 | [TK]D-Fender | rtasterisk: But you seem to thrive on illusions so go right ahead an believe "whatever" |
| 19:35.50 | ManxPower | For one thing AEL didn't even work very well until 1.4 |
| 19:36.21 | rtasterisk | its just a question :) |
| 19:36.36 | rtasterisk | Its plan to support SIP/TCP for 1.6 ? |
| 19:36.47 | rtasterisk | does anyone have any information about it ? |
| 19:36.48 | [TK]D-Fender | rtasterisk: Last I heard, yes. |
| 19:37.23 | AsTeRiSk_1 | Hello, some body can help me? tnks for your time and help i having problems incoming calls the DEBUG[13519] chan_zap.c: Sent deferred digit string:, but i can meke outbound calls |
| 19:37.49 | ManxPower | rtasterisk: there is no plans that I am aware of to support RTP over TCP (all "sip audio" is really RTP) |
| 19:38.26 | ManxPower | so the only thing you would get with SIP/TCP is....um...uh...what advantage IS there of SIP/TCP? |
| 19:38.32 | [TK]D-Fender | RTP != SIP, SIP= UDP/TCP |
| 19:38.47 | AsTeRiSk_1 | i am using a Normal T1 E&M W |
| 19:39.30 | [TK]D-Fender | ManxPower: Advantage of TCP is that the control channe can work better behind NAT since it can be persistant and wouldn't need to be forwarded when there are multiple clients behind the same router. |
| 19:39.51 | anthm | he probably means the signaling over TCP which is a requirement in the sip RFC to support both TCP and UDP transports |
| 19:39.57 | [TK]D-Fender | ManxPower: marginally more secure for being state based |
| 19:40.02 | [TK]D-Fender | (barely) |
| 19:40.18 | rtasterisk | the advantage can be to use asterisk and Microsoft LCSE both with openser ... |
| 19:40.39 | rtasterisk | because its seems LCSE only support SIP/TCP |
| 19:40.42 | ManxPower | openser does not support SIP/UDP? |
| 19:41.08 | ManxPower | openser can't translate from SIP/UDP to SIP/TCP? |
| 19:41.25 | rtasterisk | translate ? |
| 19:41.53 | ManxPower | rtasterisk: don't worry, once 1.6 comes out I'm sure you'll discover that LCSE does something that makes it not work with any decent SIP system. |
| 19:42.00 | ManxPower | rtasterisk: convert |
| 19:42.04 | anthm | according to the spec it's mandatory to support both and be able to do either transparently |
| 19:42.20 | anthm | the transport is not tied to the protocol |
| 19:42.27 | *** join/#asterisk Mrchicken (n=administ@200.71.58.39) |
| 19:42.33 | Mrchicken | hello |
| 19:42.43 | Mrchicken | Anybody knows if AEL2 supports arrays? |
| 19:42.58 | rtasterisk | TCP can be a problem on large networks |
| 19:43.48 | rtasterisk | and TCP stack of Linux seems to be not very crontrolable by applications |
| 19:43.59 | rtasterisk | and present timers problems ... |
| 19:44.27 | AsTeRiSk_1 | hello, some body can help ? incoming calls the DEBUG[13519] chan_zap.c: Sent deferred digit string: |
| 19:45.31 | anthm | despite any criticism of tcp its simply a requirement of a sip ua to be able to handle it |
| 19:46.38 | *** join/#asterisk Agnt_0rnge (n=chatzill@mail.enplan.com) |
| 19:47.00 | ManxPower | AsTeRiSk_1: I have never seen that error in the almost 5 years of using Asterisk |
| 19:47.17 | rtasterisk | does anyone knows if it exist a app_firewall ? |
| 19:47.25 | ManxPower | anthm: so really BOTH Asterisk AND the microsoft piece of crap are out of spec |
| 19:47.33 | rtasterisk | like voice firewall ? |
| 19:47.38 | ManxPower | rtasterisk: now you are just being stupid. |
| 19:47.39 | anthm | if they do one or the other yes |
| 19:47.54 | rpm | rtasterisk, its called layer7 iptables. |
| 19:48.04 | [TK]D-Fender | ManxPower: What makes you think he's only starting NOW ;) |
| 19:48.26 | ManxPower | [TK]D-Fender: I didn't mean to imply that he just started. |
| 19:48.36 | rtasterisk | for example, i want to deny international calls during the night |
| 19:48.45 | rtasterisk | its something possible with AGI |
| 19:48.49 | rtasterisk | i know |
| 19:48.52 | ManxPower | rtasterisk: incoming international calls or outgoing international calls? |
| 19:48.59 | rtasterisk | outgoing |
| 19:49.06 | ManxPower | THAT is a dialplan thing, you putz |
| 19:49.09 | [TK]D-Fender | rtasterisk: go read THE BOOK. |
| 19:49.12 | [TK]D-Fender | ~book |
| 19:49.13 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 19:49.13 | anthm | the actual reason it's mandatory is because of another rule in the udp transport that says you cannot put more than 90% of the mtu in a single udp packet when sending a sip message and if the total size of the message exceeds that size you are forced to change to tcp |
| 19:49.55 | rtasterisk | but exceed 1500 bytes for a SIP message seems to be very rare ?? |
| 19:50.02 | ManxPower | anthm: so really, the only time Asterisk's non-support of SIP/TCP (which you say is an RFC violation) is when connecting to some device that also violates the RFC. |
| 19:51.27 | [TK]D-Fender | ManxPower: RFC = no comment ;) |
| 19:51.35 | De_Mon | Yikes, I just started reading the debian pkg-voip-maintainers mailing list... |
| 19:51.36 | ManxPower | 8-) |
| 19:51.49 | *** join/#asterisk cypherdelic (n=cypher@p5B27EE2D.dip.t-dialin.net) |
| 19:51.56 | jordanb | ManxPower, Well, or in a situation where you must use TCP for some other reason. |
| 19:52.02 | ManxPower | De_Mon: are they as crazy as all the other package maintainers? |
| 19:52.04 | jordanb | Suppose you need to tunnel over ssh or something. |
| 19:52.10 | karleeto | anyone use HPEC? |
| 19:52.14 | Alan_Hicks | I'm having some trouble attempting to configure a TDM400P card for use with *. I'm following the book (First edition, 2005) and have done everything according to instructions in the initial configuration. |
| 19:52.15 | ManxPower | karleeto: yes |
| 19:52.22 | karleeto | i |
| 19:52.39 | anthm | well technically it's always an RFC violation but i doubt asterisk follows any of the other req either |
| 19:52.43 | Alan_Hicks | Basically, I'm attempting to call my wctdm card and run the Echo() application just to ensure everything is operating properly. |
| 19:52.44 | karleeto | i've been seeing a LOT of: |
| 19:52.47 | anthm | or anyone for that matter |
| 19:52.53 | anthm | cos it's all insane |
| 19:53.00 | karleeto | hpec_channel_alloc: No channels available |
| 19:53.11 | Alan_Hicks | However, * never answers. It just rings forever. |
| 19:53.15 | karleeto | in my dmesg |
| 19:53.16 | ManxPower | karleeto: It looks like you did not buy enough HPEC licenses |
| 19:53.37 | De_Mon | ManxPower the whole process of managint a project with as many dependancies on other packages as asterisk is a lot harder than I though |
| 19:53.37 | ManxPower | how many Zap channels do you have and how many HPEC licenses do you have? |
| 19:53.39 | anthm | the funny part is that the spec also requires you to support a max udp packet size of 65k |
| 19:53.50 | anthm | in case the tcp doesnt work |
| 19:53.54 | Alan_Hicks | When I run "ztcfg -vv" it says at the bottom "4 channels to configure." Am I right in assuming that something's wrong with zaptel and not *? |
| 19:54.07 | karleeto | ManxPower: yeah, i have one license and added a second module last week |
| 19:54.11 | [TK]D-Fender | Alan_Hicks: Doesn't mean your dialplan is right or ZAPATA is either... |
| 19:54.16 | [TK]D-Fender | Alan_Hicks: PASTEBIN is your friend... |
| 19:54.20 | ManxPower | karleeto: then you ran out of licenses |
| 19:54.20 | Alan_Hicks | IIUC, that should say "4 channels configured." |
| 19:54.22 | anthm | so you are required to support 65k packets but only supposed to use 1200 bytes of it |
| 19:54.39 | rtasterisk | on IP yes |
| 19:54.54 | [TK]D-Fender | Alan_Hicks: It should if you have 4 channels, but that doesn't mean everything will work like you want. |
| 19:54.54 | Alan_Hicks | [TK]D-Fender: Give me a moment and I'll see about pastebin. I've got no X, so it might be tricky with lynx. |
| 19:54.55 | karleeto | ManxPower: so i just need to call em and have em increase the number of modules for my license |
| 19:55.12 | *** join/#asterisk javb (n=javb@190.80.224.20) |
| 19:55.13 | ManxPower | karleeto: I have no idea, but calling them would be a good place to start. |
| 19:55.18 | AsTeRiSk_1 | hello, some body can help ? incoming calls the DEBUG[13519] chan_zap.c: Sent deferred digit string: |
| 19:55.23 | Alan_Hicks | My card has four FXO modules. |
| 19:55.32 | javb | Hi, i`m getting this error... any ideas: http://dpaste.com/23286/ |
| 19:56.01 | [TK]D-Fender | javb: Seem pretty blatant.. what more do you need? |
| 19:56.44 | ManxPower | javb: "logger reload" |
| 19:57.03 | k31th | wooo |
| 19:57.05 | k31th | its up:D |
| 19:57.36 | Alan_Hicks | http://pastebin.com/m3c1f6b25 |
| 19:57.44 | javb | Well, dont know why i have "read only file system" .. dont know how to take this off, dont know why happened.. can u help with your expirience? |
| 19:57.58 | ManxPower | [TK]D-Fender: Even I admit my newest macro-std-exten-v2 is a total mess. |
| 19:58.00 | Alan_Hicks | That's everything I've changed beyond a stock "make samples" |
| 19:58.11 | *** part/#asterisk krondorl (n=chatzill@tfi1meg.1meg.golden.net) |
| 19:58.20 | AsTeRiSk_1 | hello, some body can help ? incoming calls the DEBUG[13519] chan_zap.c: Sent deferred digit string: |
| 19:58.22 | Alan_Hicks | And that shows the output of ztcfg -vv. Any help will be appreciated. |
| 19:58.23 | ManxPower | javb: it LOOKS like the files were deleted out from under Asterisk and it is not happy about it. |
| 19:58.52 | tzafrir | Alan_Hicks, that message from ztcfg is normal |
| 19:58.53 | [TK]D-Fender | Alan_Hicks: Looks decent, but you might not have plugged into the right JACK |
| 19:59.03 | tzafrir | unless you got an error after it |
| 19:59.04 | ManxPower | javb: DO those files EXIST? |
| 19:59.17 | javb | ManxPower: I CANT even create a folder anywhere in that Linux box. |
| 19:59.23 | [TK]D-Fender | Alan_Hicks: pastebin "zap show channels" now as well. |
| 19:59.30 | Alan_Hicks | Let me double-check. I plugged into the 1 listed as "1" in the diagram in the book, and I believe I've tried it in every jack. |
| 19:59.34 | ManxPower | javb: Oh! Well then leave. Go to a Linux channel. |
| 19:59.35 | [TK]D-Fender | javb: Guess you're screwed |
| 19:59.57 | Alan_Hicks | [TK]D-Fender: I don't have a "zap" command. |
| 20:00.01 | [TK]D-Fender | Alan_Hicks: You COULD also have a dead module, but PB first... |
| 20:00.10 | [TK]D-Fender | Alan_Hicks: Guess you didn't compile * AFTER Zaptel... |
| 20:00.17 | [TK]D-Fender | Alan_Hicks: Which is what you have to do. |
| 20:00.30 | tzafrir | the message of ztcfg changed in zaptel 1.4.6 , because indeed it is issued before ztcfg actually attempts to configure the channels |
| 20:00.31 | [TK]D-Fender | Alan_Hicks: Try loading it manually : "module load chan_zap.so" |
| 20:00.41 | Alan_Hicks | Actually, I'm pretty sure I did, but it's possible I didn't actually install zaptel until after compiling *. |
| 20:01.06 | Alan_Hicks | [TK]D-Fender: In the * console you mean? |
| 20:01.07 | [TK]D-Fender | Alan_Hicks: Very believable that it was not done in the right order. |
| 20:01.10 | [TK]D-Fender | Alan_Hicks: Yes. |
| 20:01.20 | [TK]D-Fender | Alan_Hicks: Just to see if the module is there. |
| 20:01.32 | [TK]D-Fender | Alan_Hicks: If it fails, go recompile everything in the right order |
| 20:02.11 | Alan_Hicks | Module 'chan_zap.so' already exists. |
| 20:02.12 | AsTeRiSk_1 | some body can help ? |
| 20:02.34 | *** join/#asterisk beek (n=klinebl@pool-72-94-31-84.phlapa.fios.verizon.net) |
| 20:02.44 | Mrchicken | Anybody knows if AEL2 supports arrays? |
| 20:02.46 | tzafrir | Alan_Hicks, zap show channels |
| 20:03.10 | tzafrir | does it show anything at all? anything more than "pseudo"? |
| 20:03.16 | Alan_Hicks | I don't really have any way to pastebin that output. |
| 20:03.29 | Alan_Hicks | "1 incoming default" |
| 20:03.32 | AsTeRiSk_1 | i having problems incoming calls TE210P E&M |
| 20:03.48 | *** join/#asterisk fujin (n=aj@unaffiliated/fujin) |
| 20:03.54 | Alan_Hicks | [TK]D-Fender: Sorry about that "don't have a zap command" thing earlier. I didn't realize you meant inside the * console. |
| 20:04.45 | [TK]D-Fender | ........... |
| 20:05.03 | rtasterisk | what are today SIP level load balancing solution for asterisk ? |
| 20:05.37 | [TK]D-Fender | Alan_Hicks: go change "channel => 1" to "channel => 1-4" and restart *. This will help you see what jack is plugged in. |
| 20:05.47 | [TK]D-Fender | rtasterisk: SER |
| 20:05.48 | dlynes_laptop | rtasterisk: /join #openser |
| 20:05.52 | AsTeRiSk_1 | this is my error http://dpaste.com/23287/ |
| 20:06.02 | ManxPower | someone is trying to argue with me that echo on pure IP calls is impossible unless the endpoint causes it. |
| 20:06.23 | Alan_Hicks | http://pastebin.com/m21babe5c |
| 20:06.25 | ManxPower | (on -users mailing list) |
| 20:06.36 | *** join/#asterisk Jubalint (n=HoBob@adsl-072-148-059-225.sip.ard.bellsouth.net) |
| 20:06.44 | dlynes_laptop | ManxPower: not quite impossible....it can still be caused by the acoustic coupling in the handset/headset |
| 20:07.05 | ManxPower | Mrchicken: 1.4 has an ARRAY function. AEL just compiles into standard dialplan extensions.conf. You do the math |
| 20:07.15 | [TK]D-Fender | dlynes_laptop: Those would technically be "endpoints" ;) |
| 20:07.16 | dlynes_laptop | ManxPower: if they don't have decent acoustic coupling echo cancellation in their software |
| 20:07.17 | ManxPower | dlynes_laptop: I said "unless the endpoint causes" |
| 20:07.21 | AsTeRiSk_1 | this is my error http://dpaste.com/23287/ |
| 20:07.21 | *** join/#asterisk killfill (n=killfill@pc-164-134-45-190.cm.vtr.net) |
| 20:07.23 | killfill | hey.. |
| 20:07.32 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@70.240.164.157) |
| 20:07.34 | dlynes_laptop | ManxPower: yeah...wasn't paying attention :) |
| 20:07.42 | Mrchicken | ManxPower, my question is this: I trying to get some records from mysql |
| 20:07.49 | Alan_Hicks | [TK]D-Fender: Done. Output from "zap show channels" is the same as the link I just pasted, except that it lists channels 2, 3, and 4. They have to same values as 1. |
| 20:07.51 | ManxPower | feel free to post on the mailing list on the subject |
| 20:07.52 | [TK]D-Fender | AsTeRiSk_1: No-one here has an answer for your question. Go post it on the mailing lists. |
| 20:08.06 | [TK]D-Fender | Alan_Hicks: Good, now test each port. |
| 20:08.13 | Alan_Hicks | yes sir |
| 20:08.17 | killfill | i want to connect from a client app (.net or ruby) and ask if the phone i use has an active call. this is agi for right?.. |
| 20:08.18 | Mrchicken | and I'd like each row to go to $var[1] |
| 20:08.37 | ManxPower | Mrchicken: Sorry, I did not mean to imply that I know how to use the ARRAY() function nor did I mean to imply that I have ever used a database with Asterisk in any way. |
| 20:09.08 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 20:09.29 | killfill | or agi's are for 'server' things? |
| 20:09.37 | ManxPower | Mrchicken: When was the last time you did "show applications" and "show functions" in the Asterisk CLI. |
| 20:10.03 | seanbright | killfill: if you want a client app to connect to asterisk and ask it questions, you want to look into AMI |
| 20:10.05 | Mrchicken | ummm a long time ago :P |
| 20:10.21 | seanbright | killfill: you can find lots of info on that at http:///www.voip-info.org |
| 20:10.26 | seanbright | killfill: just search for AMI there |
| 20:10.27 | ManxPower | killfill: I can't imagine any way you might do what you want to do, other than write a full blown application to monitor the state of all calls. You would use AMI (Asterisk Manager Interface) for that. |
| 20:10.28 | killfill | aah |
| 20:10.31 | Alan_Hicks | Ok.... |
| 20:10.36 | Alan_Hicks | Ports 2, 3, and 4 work. |
| 20:10.39 | Alan_Hicks | Port 1 does not. |
| 20:10.48 | killfill | yup.. i wish to obtain info about the call im recieving... |
| 20:10.55 | killfill | (for a mini callcenter) |
| 20:11.21 | [TK]D-Fender | Alan_Hicks: now shut down and swap modules around to confirm if its 1 dead |
| 20:11.31 | Alan_Hicks | Yes sir. |
| 20:11.43 | seanbright | killfill: great. go to www.voip-info.org and search for AMI. |
| 20:11.44 | [TK]D-Fender | Alan_Hicks: We'v isolated it. the hard part is done. |
| 20:11.44 | killfill | i wan make this "event driven" right?.. i.e. connect and make AMI notify me on events? |
| 20:11.54 | killfill | wan/can |
| 20:12.01 | seanbright | killfill: yup. go ahead over to www.voip-info.org and search for 'AMI' |
| 20:12.07 | killfill | greate. thanks! |
| 20:12.17 | Alan_Hicks | [TK]D-Fender: Thanks. Bad hardware is something I *NEVER* look for until everything else has been exhausted. |
| 20:12.18 | seanbright | heh |
| 20:12.21 | Alan_Hicks | You saved me some time. |
| 20:12.21 | ManxPower | killfill: don't get frustrated my the limitations of AMI. |
| 20:12.32 | Alan_Hicks | But how exactly will moving it around help? |
| 20:12.44 | killfill | ManxPower, what exactly do you mean? |
| 20:12.55 | [TK]D-Fender | Alan_Hicks: Well your config was fine to test 1 port, but that assumed it was wired right, so we very quickly checked everything. Test couldn't have gone faster... |
| 20:13.13 | [TK]D-Fender | Alan_Hicks: it will confirm if its the MODULE, or the slot its in thats dead. |
| 20:13.15 | killfill | hm.. "rami" ruby for ami, is from 2005.. :P |
| 20:13.18 | ManxPower | killfill: go read the wiki |
| 20:13.29 | [TK]D-Fender | Alan_Hicks: Dead module = replace module. Dead base card = replace card. |
| 20:13.31 | Alan_Hicks | Thanks. Hope it ain't the slot.... I'll check back in later. |
| 20:13.39 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 20:13.53 | [TK]D-Fender | Alan_Hicks: This is all really bottom-up stuff. |
| 20:15.00 | killfill | Asterisk will not generate a CDR record, if i dont anwear the incomming call, will it? |
| 20:15.04 | *** topic/#asterisk by Qwell -> Asterisk: The Open Source Telephony Application Platform -=- Asterisk 1.4.13 (Oct. 10, 2007) Addons 1.2.8, 1.4.4 (Oct. 16, 2007) -=- Join #asterisknow or #asterisk-gui for AsteriskNOW and Asterisk-GUI info -=- Join #asterisk-commits to monitor svn changes -=- Join #freepbx for freepbx/#trixbox for trixbox support. -=- http://www.digium.com/en/company/switchvox-acquisition-faq.php |
| 20:15.25 | [TK]D-Fender | Qwell: \o/ |
| 20:16.20 | fujin | ooh |
| 20:17.00 | ManxPower | killfill: that is a very bad assumption |
| 20:17.10 | ManxPower | killfill: and easily verifiable by you |
| 20:17.16 | killfill | heh.. |
| 20:17.32 | killfill | well, i verify it. maybe i have a bad setup? |
| 20:17.47 | killfill | it sounds strange really.. thats why i asked |
| 20:17.56 | killfill | i should see the "ring" events, right |
| 20:18.33 | ManxPower | "ring event"? |
| 20:19.21 | ManxPower | In the modern world of voicemail, etc, almost all calls are answered, BTW. |
| 20:19.43 | ManxPower | [TK]D-Fender: I called the CLEC to ask for a credit for 2 days of downtime. |
| 20:19.50 | killfill | oh sure.. but i wish to execute something when the phone rings. i.e. before ppl answears.. |
| 20:19.59 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 20:20.12 | ManxPower | killfill: System, TrySystem, AGI, and I'm sure there are a few I forgot. |
| 20:20.19 | Katty | mew. |
| 20:20.33 | ManxPower | heck, I generally execute 100 lines of dialplan code before a phone actually rings. |
| 20:20.54 | ManxPower | (more like 50, actually) |
| 20:21.08 | De_Mon | and it could really be cleaned up and done in about 10 |
| 20:21.11 | killfill | i ment on a client app.. but never mind. i need to read more. |
| 20:21.58 | De_Mon | where would you on irc go to find people that use snmp? |
| 20:22.46 | seanbright | #snmp? |
| 20:22.47 | trippss | De_Mon: #snmp? |
| 20:22.48 | seanbright | :) |
| 20:22.54 | ManxPower | De_Mon: Feel free. http://www.fnords.org/~eric/macro-std-exten-v2.inc |
| 20:22.57 | *** part/#asterisk AsTeRiSk_1 (n=A@190.80.139.29) |
| 20:23.13 | De_Mon | oh crap |
| 20:23.15 | Dan0maN_Work | all, i would like to get some opinions on PBX managers (such as thirdlane, switchvox, etc.) |
| 20:23.17 | De_Mon | trippss I was lonely |
| 20:25.20 | De_Mon | ManxPower asterisk has this cool app called app_voicemail! |
| 20:25.55 | ManxPower | De_Mon: yup. I use it all the time. |
| 20:26.19 | ManxPower | killfill: the only thing you can "execute before dial" is dialplan stuff. |
| 20:31.03 | Dan0maN_Work | no opinions? ;) here's the deal... the pres of my company asked me to evaluate * as a replacement for our lucent definity. |
| 20:31.21 | Dan0maN_Work | i have been lurking here for about 2 months now |
| 20:31.27 | Dan0maN_Work | and attended astricon |
| 20:31.39 | ManxPower | Dan0maN_Work: then you know we never talk about those things, as we don't use them |
| 20:31.49 | Dan0maN_Work | that's what i'm looking for Manx |
| 20:31.50 | ManxPower | A PBX manager is a PERSON, not a PROGRAM. 8-) |
| 20:31.55 | Dan0maN_Work | looking for reasons why |
| 20:32.10 | ManxPower | Dan0maN_Work: the primary reason is that they FORCE you into doing things their way. |
| 20:32.23 | ManxPower | Their way almost never is the same as your needs. |
| 20:32.29 | Dan0maN_Work | my pres initially wanted me to look into it because he "heard" about it from a friend of his |
| 20:32.35 | Dan0maN_Work | after i got back from astricon |
| 20:32.42 | ManxPower | Asterisk is not a PBX. It is a toolkit to build your own PBX. |
| 20:32.44 | Dan0maN_Work | i told him that it would be a long road to get set up, and maintain |
| 20:32.53 | Dan0maN_Work | k. understood |
| 20:32.58 | Qwell | Dan0maN_Work: it's really not |
| 20:33.01 | ManxPower | Dan0maN_Work: good planning make maint pretty easy. |
| 20:33.04 | Qwell | not a long road, that is |
| 20:33.43 | Dan0maN_Work | Qwell, i'm new at this. completely. never used sip, never touched *. most i know about it is my linux background |
| 20:33.54 | Qwell | Dan0maN_Work: hey, that's more than most |
| 20:33.54 | ManxPower | Dan0maN_Work: do you know anything about telecom? |
| 20:34.11 | ManxPower | To be good at Asterisk you really need to know Telecom, Linux, Networking, AND SIP. |
| 20:34.11 | Qwell | Dan0maN_Work: go to a bootcamp course |
| 20:34.16 | Dan0maN_Work | Manx: enough to get me by so far. but nothing too deep |
| 20:34.27 | Dan0maN_Work | i'm the network admin here too |
| 20:34.43 | Qwell | alternatively, buy a boxed deal, like switchvox |
| 20:34.45 | ManxPower | Dan0maN_Work: that doesn't mean you know ANYTHING about networks. 8-) |
| 20:34.51 | Qwell | ManxPower: too true |
| 20:34.58 | Dan0maN_Work | he has since "talked" to his friend about what he's doing. he told him if i'm not looking into a gui for it, i should be |
| 20:35.06 | Dan0maN_Work | everyhting i've read in here from you guys says differently |
| 20:35.11 | Dan0maN_Work | which i can understand |
| 20:35.14 | Qwell | Dan0maN_Work: stay far away from trixbox. |
| 20:35.17 | ManxPower | Dan0maN_Work: we are very biased against GUIs. |
| 20:35.30 | Dan0maN_Work | if the features aren't in the gui, it prolly messes with the config files when you add it manually |
| 20:35.34 | Dan0maN_Work | ok |
| 20:35.34 | ManxPower | if you want to talk to people that like guis, then go to a GUI channel like #trixbox |
| 20:35.51 | Dan0maN_Work | Manx: i already know to stay away from trix ;_) |
| 20:36.07 | Dan0maN_Work | right now, i'm looking to build a case to NOT use a GUI |
| 20:36.07 | Qwell | Dan0maN_Work: then like I said - you already know a lot more than most do ;) |
| 20:36.16 | Dan0maN_Work | heh. thanks Qwell |
| 20:36.23 | ManxPower | The last person here with a GUI question was fed to the 'gators before he even finished asking it. |
| 20:36.26 | Dan0maN_Work | i've been lurking here for a bit |
| 20:36.53 | [hC] | Dan0maN_Work: the reason we dont talk about guis is because most of the people in here hack on asterisk on a daily basis and have for quite some time. It all depends if you want to get that into it, etc, or you just want to set up a phone system and then move on to something else. |
| 20:37.20 | Dan0maN_Work | ok. thanks hC |
| 20:37.23 | ManxPower | Dan0maN_Work: The difference is "I'm sorry, but our GUI can't do that" .vs. "I'll look into it and see if we can impliment that request" |
| 20:37.26 | [hC] | Dan0maN_Work: if you want to just set up a phone system and have generally all the stuff you'll need, trixbox or something more advanced like switchvox will do just fine. |
| 20:37.57 | [hC] | Dan0maN_Work: but if you want to get into massaging asterisk to do what you want, expect to not use a gui. compare it to being able to make linux do what you want, where as windows does what it decides you should be able to do. |
| 20:38.25 | Qwell | [hC]: or not do, in many cases |
| 20:38.27 | Dan0maN_Work | the problem stems from out of 3 IT people (including phone support), i'm the only person who knows linux, networking, and the most of the telephony |
| 20:38.47 | Qwell | Dan0maN_Work: sounds like you need a new IT dept |
| 20:38.53 | ManxPower | Our IT manager keeps asking me when we will have a GUI end user management interface. I always say "when someone gives me a list of features and requirements". That shuts him up for a while. |
| 20:39.11 | Dan0maN_Work | preaching to the choir Qwell ;) |
| 20:39.15 | *** join/#asterisk Assid (n=assid@unaffiliated/assid) |
| 20:39.21 | flujan | hi guys... |
| 20:39.24 | generalhan | has anyone in here used anything like QueueMetrics, with good results ?? The boss-man wants all kinds of information to be ready at the drop of a hat, to be able to smack some slackers around, and this looks to me like what he wants. |
| 20:39.29 | Assid | bah.. i think i need help fine tuning quality on this thing |
| 20:39.30 | flujan | I am having problems with the Pickup app. |
| 20:39.40 | flujan | I can pickup extension to extension calls |
| 20:39.42 | *** join/#asterisk TrentCreek (n=tjones14@cpe-70-117-207-168.rgv.res.rr.com) |
| 20:39.50 | flujan | but not incoming calls to a specific extension. |
| 20:39.55 | flujan | here is my extensions.conf |
| 20:39.58 | flujan | http://pastie.caboo.se/110553 |
| 20:40.04 | Assid | err.. whats the command to see the jitter buffer etc? |
| 20:40.25 | ManxPower | exten => _1XXXXX,1,Pickup(${EXTEN:1@incoming} |
| 20:40.31 | flujan | Incoming calls enters on the incoming context... |
| 20:40.34 | *** join/#asterisk punkgode (n=punkgde@rev-200-40-119-222.netgate.com.uy) |
| 20:40.37 | dlynes_laptop | Assid: iax2 jitter buffer? |
| 20:40.46 | Assid | dlynes_laptop: sip |
| 20:40.52 | flujan | ManxPower: I will Pickupcalls from both context? |
| 20:40.59 | Assid | trying to fine tune the settings for some better clarity |
| 20:41.47 | ManxPower | flujan: according to "show application pickup" you can only do one context per invocation |
| 20:42.26 | ManxPower | but you knew that already. |
| 20:42.42 | ManxPower | flujan: clever use of contexts quickly solves that issue. |
| 20:42.56 | ManxPower | specifically include => but be careful of security considerations. |
| 20:43.14 | *** join/#asterisk cypherdelic (n=cypher@p5B27EE2D.dip.t-dialin.net) |
| 20:43.20 | _x86_ | anyone ever hear of / use a company called TNCI |
| 20:43.33 | _x86_ | for point-to-point data T1's |
| 20:43.34 | flujan | ManxPower: you men include the incoming context on the default one? |
| 20:44.30 | ManxPower | flujan: assuming that does not cause any security considerations, yes. |
| 20:44.42 | ManxPower | but the pastebin you gave doesn't work anyway. |
| 20:44.56 | ManxPower | there is no way for incoming calls to get into the default context. |
| 20:45.19 | flujan | ManxPower: yeap I know it... it is handle by a agi... |
| 20:47.28 | ManxPower | flujan: pretty irresponsible of you to not give us a working, easy to understand config. |
| 20:47.28 | flujan | ManxPower: even using the @incoming is is not working... :( |
| 20:47.48 | flujan | ManxPower: sorry about that... :( |
| 20:47.56 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 20:47.56 | ManxPower | flujan: does your AGI do any Dialing? |
| 20:48.15 | flujan | ManxPower: yeap... it receives the call and dial to a extension |
| 20:48.18 | peanut- | how do I call an extension and make it pickup withour ringing? |
| 20:48.23 | ManxPower | not going to work with Pickup then |
| 20:48.35 | ManxPower | peanut-: you cannot. |
| 20:48.45 | k31th | have a bit of a problem, I am trying to register a new handset. I have created an entery in sip.conf for exten 1000 , the phones does not register and the CLI displays nothing, it is as if the phone is not contacting the server... however if i set the phone to auth using an extension that does not exsist i get "No matching peer found" |
| 20:48.51 | ManxPower | or do you mean have the DEVICE answer automatically like paging. |
| 20:48.55 | flujan | ManxPower: any solution that I can use? |
| 20:49.01 | De_Mon | O_o |
| 20:49.11 | ManxPower | flujan: none I can think of. |
| 20:49.22 | De_Mon | Dilbert does NOT have an erect tie you sick-o |
| 20:49.33 | peanut- | ManxPower: poo. |
| 20:49.34 | ManxPower | the calls go into the magical black box called "agi" then magically appear where they are supposed to. |
| 20:49.36 | flujan | if I use always the dialplan not the agi script do you think it will work? |
| 20:49.58 | ManxPower | flujan: We might have a chance if diagnosing the problem at least. |
| 20:50.48 | De_Mon | k31th turn on sip debugging and find out what it's doing |
| 20:50.55 | TrentCreek | I see i3nary has not been here in a few days. He home USED to be in an Diego |
| 20:51.18 | flujan | ok I will try to quit the agi and use the dialplan. :) |
| 20:51.38 | De_Mon | (sip debug) |
| 20:53.53 | peanut- | ManxPower: I guess I mean page |
| 20:54.15 | ManxPower | peanut-: then the answer is "it is totally and %100 dependent on the phone you are using" |
| 20:54.22 | k31th | De_Mon: sip set debug* |
| 20:54.56 | peanut- | ManxPower: thanks |
| 20:55.14 | [hC] | has anyone come up with a limit at which paging starts to fail (number of paged phones) - i get reports of people getting pages all the time and not hearing all the audio, of course cause there are about 60 phones joining the meetme at once... is there a more scalable solution yet? |
| 20:55.43 | k31th | I am behind a NAT however the phone normally registers fine? |
| 20:58.00 | peanut- | k31th: is that a question. |
| 20:58.25 | *** join/#asterisk |Rain| (i=rain@2001:440:eeee:fffb:42:0:0:2) |
| 20:58.38 | Alan_Hicks | Said I'd check back in, so here am I. |
| 20:58.54 | ManxPower | [hC]: yes. |
| 20:59.19 | Alan_Hicks | Turns out the problem was with the TDM400P card itself, not the module. The smaller three-prong plug at the top had broken loose from the card in slot TEL1. |
| 20:59.53 | [hC] | ManxPower: and that is? :) |
| 21:00.25 | |Rain| | with 1.4.13 and the configuration in http://themuffin.net/tmp/asterbork/, dialing the queue (200) and then hanging up doesn't make the call go away (it'll keep trying queue members), and if you don't hang up but try to mash buttons instead, the queue-exit context doesn't actually work. I'm looking for ideas, 'cause I'm all out |
| 21:00.35 | ManxPower | 256 max length for application options. would you like the patch I use for my system to bump it up to 8192 bytes? |
| 21:00.49 | k31th | peanut-: yes |
| 21:01.22 | flujan | ManxPower, why a agi script that do dialing ruin the Pickup app behavior? |
| 21:02.04 | peanut- | k31th: no it's not. |
| 21:02.10 | ManxPower | [hC]: The problem for YOU is that you have too many phones being paged. You are trying to initate 60 calls at the same time. |
| 21:02.29 | ManxPower | [hC]: our solution was to not put so many phones into a page group. |
| 21:02.51 | ManxPower | But we still ran up against line length issues because the paging destinations we use are very long. |
| 21:03.14 | ManxPower | flujan: you would have to pay me to debug your AGI script before I could know that and I have no interest in doing that. |
| 21:03.30 | [hC] | ManxPower: that would of course be a solution, but when i am trying to page an entire company with multiple floors/offices/etc it doesnt work so well to just eliminate people from the group.. |
| 21:03.57 | Alan_Hicks | Now if I can just figure out how to configure this SIP phone..... |
| 21:04.27 | ManxPower | [hC]: There are 2 problems. One problem is that you have a max of about 256 chars (at least in 1.2, I don't know if 1.4 has fixed that issue) for application data (the dests to page). The other is that your system cannot seem to handle all those people in a meetme conference. |
| 21:05.01 | ManxPower | My solution for problem 2 won't work for you. It sucks to be you. |
| 21:05.07 | [hC] | ManxPower: yeah, of course. I wonder if something like app_conference would work better. |
| 21:06.44 | k31th | http://pastebin.ca/748478 thats what sip debug gives me |
| 21:06.50 | k31th | any ideas guys? |
| 21:06.54 | dug | which do I modify to make the gain louder for a zaptel card, rxgain in my zapata.conf? |
| 21:07.31 | ManxPower | dug: rxgain or txgain. |
| 21:07.43 | tzafrir | rxgain is audio recieved into asterisk . txgain: transmitted from asterisk (to the card) |
| 21:07.51 | ManxPower | just remember those settings will apply to ALL channels after the option unless you override it. |
| 21:08.14 | dug | ManxPower: sorry ... I meant to say for what I hear... which is rxgain correct (on the fxo port) |
| 21:09.52 | ManxPower | dug: it TOTALLY depends on your perspective. |
| 21:10.09 | ManxPower | do you want to change the gain on an FXO port, an FXS port, or an IP phone? |
| 21:10.16 | Katty | my phone doesn't seem to want to dial two digit numbers. |
| 21:10.23 | Katty | i'm sitting here staring at the digitmap timeout... it says 3 |
| 21:10.29 | k31th | hows it going Katty |
| 21:10.33 | ManxPower | for an fxo port, received audio gain would be controled by rxgain |
| 21:10.36 | Katty | is that the one i change to make it think 2 numbers is okay to dial? |
| 21:11.03 | ManxPower | it actually says "3 seconds" |
| 21:11.18 | dug | ManxPower: all my phones are quiet (SIP and Zap) .... so I think the best place to change the gain is on the fxo port ( I only have one incoming/outgoing line) |
| 21:11.19 | k31th | 2 digit ext?? |
| 21:11.21 | k31th | ewww |
| 21:12.00 | Katty | ManxPower: hrmm.. |
| 21:12.10 | Katty | ManxPower: then it's actually the digit map itself that needs changing :< |
| 21:12.21 | ManxPower | Katty: bingo! |
| 21:12.25 | Katty | great. :< |
| 21:12.32 | JerJer | don't anyone have a heart attack: static realtime on a queue...do i just use queues.conf => blah,foo in extconfig? |
| 21:12.53 | Katty | also! |
| 21:12.53 | k31th | Katty: is this a trixbox ? |
| 21:12.55 | Katty | hi k31th (= |
| 21:13.01 | Katty | k31th: it doesn't matter. |
| 21:13.04 | k31th | hi |
| 21:13.06 | Katty | k31th: it's just a polycom issue |
| 21:13.11 | k31th | ahh |
| 21:13.12 | flujan | ManxPower: so it is caused by the agi script? |
| 21:13.17 | flujan | I am using a ruby agi script... |
| 21:13.28 | ManxPower | flujan: I DON'T KNOW. |
| 21:13.29 | JerJer | the wonderful wiki reads like it was wrote in Chinese and babelfished to engrish |
| 21:13.37 | flujan | ManxPower: ok... sorry... :D |
| 21:13.38 | Katty | k31th: but yes, this particular server does have trixbox |
| 21:13.51 | ManxPower | flujan: but it is impossible to diagnose your problems with an AGI script involved. |
| 21:13.51 | flujan | ManxPower: no need to get angry. ;) |
| 21:14.08 | ManxPower | flujan: it's like the 3rd time you asked and the 3rd time I answered. |
| 21:14.14 | Alan_Hicks | Shouldn't I see something in the console when a SIP phone authenticates to *? |
| 21:14.23 | Katty | ManxPower: http://pastebin.ca/748489 <- there's my digit map |
| 21:14.28 | ManxPower | Katty: but you are on Polycom phones, right? |
| 21:14.32 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:14.43 | Katty | ManxPower: yes'r, ip501s |
| 21:15.03 | Katty | ManxPower: i think i need to add some sort of [0-9]xx thingy |
| 21:15.04 | k31th | ahhh wats this phone doing |
| 21:15.10 | Katty | ManxPower: but i really don't get what i'm looking at here |
| 21:15.15 | Katty | ManxPower: think you can edjimicate me? |
| 21:15.29 | ManxPower | Katty: what are you trying to dial? |
| 21:15.34 | ManxPower | the actual digits. |
| 21:15.50 | Katty | 46 and 02 |
| 21:15.52 | k31th | http://pastebin.ca/748478 is thie "no nat" the issue ? |
| 21:16.00 | ManxPower | too bad you are not using "9" for an outside line or you could get rid of all those timeouts. |
| 21:16.42 | ManxPower | why not add |XXT| to the dialplan? |
| 21:16.51 | ManxPower | or |xxT| as the case may be. |
| 21:17.21 | Katty | what does xxT mean? |
| 21:17.27 | Katty | or, more specifically the T |
| 21:17.38 | [TK]D-Fender | Katty, 2 digits + wait |
| 21:17.44 | Katty | oh ah |
| 21:17.49 | ManxPower | xx = any two digits, T = wait for the digit timeout. |
| 21:18.00 | [TK]D-Fender | Katty, best dialplan : x.T|*.T|#.T |
| 21:18.03 | ManxPower | in your case 3 seconds |
| 21:18.14 | Katty | should it be | [0-9]xxT? |
| 21:18.19 | ManxPower | Katty:, worst dialplan: x.T|*.T|#.T |
| 21:18.32 | ManxPower | Katty: that specifies THREE digts, not TWO digits |
| 21:18.39 | Katty | oh! |
| 21:18.43 | Katty | k'then |
| 21:18.49 | ManxPower | if you have a T, then you have to wait for the timeout or press SEND. |
| 21:18.59 | Katty | yes, that's good |
| 21:19.02 | [TK]D-Fender | Katty, best dialplan : x.T|*.T|#.T ;) |
| 21:19.23 | ManxPower | some people apparently don't mind waiting 3 extra seconds before anything happens -- my users mind. |
| 21:19.36 | *** join/#asterisk |dennis| (n=dennis@200.32.236.18) |
| 21:19.36 | Katty | most of our people hit send |
| 21:19.38 | *** join/#asterisk HCEvan (i=edude@pool-72-77-226-91.tampfl.fios.verizon.net) |
| 21:19.47 | Katty | but a few of them sit there and wait |
| 21:20.15 | ManxPower | Katty: what happens if the timeout happens a moment before they press the SEND softbutton? i.e. what does that SEND softbutton change into? |
| 21:20.15 | [TK]D-Fender | ManxPower, Your users are genetic throwbacks and your dialplan harkens back to < 1.0 :) |
| 21:20.28 | ManxPower | [TK]D-Fender: my dialplan is almost all 1.2! |
| 21:20.51 | [TK]D-Fender | ManxPower, Procrastination : The art of keeping up with YESTERDAY. |
| 21:21.05 | ManxPower | [TK]D-Fender: it isn't not broken.... |
| 21:21.15 | [TK]D-Fender | ManxPower, You you have not yet full attained even! |
| 21:21.39 | [TK]D-Fender | s/You you/Yet you/ |
| 21:21.56 | k31th | anyone look at my pastebin ? i cant see any errors in the output from sip set debug? |
| 21:21.56 | Katty | [TK]D-Fender: what would *xx.T do? |
| 21:22.06 | ManxPower | My users are hateful bastards, most of which don't even known a computer. |
| 21:22.18 | Shaun2222 | [TK]D-Fender: you ever seen the polycom's not listen to sntp server offset? |
| 21:22.34 | Shaun2222 | running latest and greatest bootrom and sip |
| 21:22.40 | flujan | ManxPower: don't be mad at me but I removed the agi script... |
| 21:22.45 | Shaun2222 | my phones look to be stuck on GMT from the looks of it |
| 21:22.46 | flujan | and it is still not working. |
| 21:22.49 | flujan | http://pastie.caboo.se/110576 |
| 21:22.54 | flujan | this is the new dialplan. |
| 21:23.08 | Katty | i guess that would be any *two digit number. |
| 21:23.42 | [TK]D-Fender | Katty, *xx.T = * 2 (or more) digits + wait |
| 21:23.53 | ManxPower | Katty: . usually means "1 or more digits" |
| 21:23.59 | flujan | the SIP/40000 extension start ringing and then I try to pickup without success... |
| 21:24.02 | [TK]D-Fender | Shaun2222, pastebin what you filled in. You never answered the other day |
| 21:24.06 | Katty | hmm neat. |
| 21:24.12 | ManxPower | Polycoms might be . mean 0 or more digits, I don't know for sure |
| 21:24.18 | Katty | we have x.T and *xx.T at the FTP on our network |
| 21:24.19 | [TK]D-Fender | Shaun2222, and I AM relatively sure I know what your issue is.... |
| 21:24.50 | ManxPower | flujan: you need to understand what an extension is. A phone is not an extension. |
| 21:25.03 | ManxPower | a SIP account is not an extension. An exten in a line in extensions.conf starting with exten => |
| 21:25.38 | ManxPower | Katty: it is usually his fault. |
| 21:25.40 | flujan | ManxPower: ok. So pickup will only get extensions not devices right? |
| 21:25.58 | ManxPower | flujan: for like the 99th time, Pickup only works with extensions, not devices. |
| 21:26.00 | Katty | hrmm. |
| 21:26.04 | Katty | it's still doing it. sigh. |
| 21:26.05 | Shaun2222 | [TK]D-Fender: uhh, i dont think i brang this up the other day but here's my sip.cfg.. http://pastebin.ca/748507 |
| 21:26.10 | ManxPower | The answer is not going to change no matter how many times you ask. |
| 21:26.14 | Shaun2222 | well the sntp section |
| 21:26.18 | flujan | ManxPower: now I see ... :( |
| 21:26.19 | Shaun2222 | if you want the full thing let me know. |
| 21:26.26 | [TK]D-Fender | Shaun2222, tcpIpApp.sntp.gmtOffset="-8.0" <--- see this? |
| 21:26.30 | Shaun2222 | ya |
| 21:26.43 | ManxPower | maybe Asterisk needs to come up with a term other than "extension" to mean "extension". |
| 21:26.50 | [TK]D-Fender | Shaun2222, Now if you actually read the admin guide you'd know that this field is counted in >>>SECONDS<<< |
| 21:26.54 | ManxPower | I suggest "blork". |
| 21:26.54 | k31th | christ, this phone is killing me. |
| 21:27.10 | [TK]D-Fender | Shaun2222, So congratulations on your 8 second time zone change! |
| 21:27.11 | ManxPower | blork => 3556,1,Dial(SIP/3556) |
| 21:27.26 | Shaun2222 | [TK]D-Fender: after trusting the manual with the idle image i gave up on it ;) |
| 21:27.28 | k31th | hahahah |
| 21:27.42 | ManxPower | Hopefully when people say "that only works with blorks" they don't think it is a device or sip account. |
| 21:27.42 | [TK]D-Fender | Shaun2222, Merry Christmas. |
| 21:27.44 | outtolunc | extension = mystical number assignment akin to a post office box on a wall of post office boxes |
| 21:27.45 | Shaun2222 | haha |
| 21:28.07 | Shaun2222 | [TK]D-Fender: thanks, let me see when i set it right |
| 21:28.15 | [TK]D-Fender | ok, I'm out for a while...... |
| 21:28.16 | [TK]D-Fender | later |
| 21:28.21 | k31th | later |
| 21:31.00 | k31th | SIP/2.0 401 Unauthorized |
| 21:32.52 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 21:33.31 | k31th | the phone just says Not Registered |
| 21:33.40 | k31th | not failed... |
| 21:36.38 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 21:37.49 | Shaun2222 | [TK]D-Fender: thanks, setting it in seconds fixed the prob... |
| 21:42.25 | *** join/#asterisk UserReg_CL (n=dede@164.77.196.217) |
| 21:42.37 | UserReg_CL | hi.. helpme please.. |
| 21:42.50 | ManxPower | UserReg_CL: hi.. ask a question please.. |
| 21:42.56 | k31th | lol |
| 21:42.56 | UserReg_CL | need know password root for mysql (in trixbox) |
| 21:43.03 | ManxPower | UserReg_CL: we cannot help you |
| 21:43.05 | k31th | oh jesus |
| 21:43.07 | ManxPower | try #trixbox |
| 21:43.18 | UserReg_CL | void #trixbox :) |
| 21:43.30 | ManxPower | UserReg_CL: nobody here uses trixbox. |
| 21:43.35 | k31th | ManxPower: any idea on my problem? |
| 21:43.43 | punkgode | hello, I'm having problems getting the clock source from the PRI line (E1) using a TE110P, do I need to setup anything more? here is my zaptel.conf -> http://rafb.net/p/6yuuxJ12.html |
| 21:43.50 | ManxPower | k31th: I was not thinking about your problem |
| 21:44.16 | k31th | all help welcome :D |
| 21:44.59 | ManxPower | punkgode: regardless of what any of the utilities say, you have it set correct. |
| 21:46.22 | k31th | [1000] |
| 21:46.22 | k31th | type=friend |
| 21:46.22 | k31th | context=phones |
| 21:46.22 | k31th | host=dynamic |
| 21:46.23 | k31th | secret=1000 |
| 21:46.37 | ManxPower | k31th: flood and die. |
| 21:46.37 | punkgode | ManxPower, I do think so, this same setup is working perfectly with other Digium cards |
| 21:46.41 | *** join/#asterisk Uatec (n=uatecuk@adsl.ntsols.com) |
| 21:46.43 | Uatec | evening |
| 21:47.01 | k31th | <PROTECTED> |
| 21:47.03 | punkgode | ManxPower, it's just this one, that refuses to get the clock from the pstn |
| 21:47.05 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 21:47.32 | punkgode | ManxPower, I'm also getting lot's of IRQ misses |
| 21:47.38 | Uatec | is it possible to timestamp every line in the CLI? |
| 21:48.06 | Uatec | also, when i connect to the CLI using "asterisk -R" on one of my servers, i get everything in colour |
| 21:48.13 | ManxPower | punkgode: how do you know it is not getting sync from the PSTN? |
| 21:48.13 | Uatec | but on another one, it's all in black and white |
| 21:48.24 | Uatec | how can i change the blac and white one to be in colours? |
| 21:48.31 | *** part/#asterisk UserReg_CL (n=dede@164.77.196.217) |
| 21:49.20 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 21:49.35 | ManxPower | k31th: for one thing that is not a valid sip.conf |
| 21:49.41 | [hC] | Qwell: so i just got my AA50.. is this for real, the only thing i can get to is port 80? |
| 21:49.44 | punkgode | ManxPower, zttool reports it, I'm also getting voice problems, fax transmit problems, and some errors displayed on asterisk logs |
| 21:49.55 | Qwell | [hC]: that's all that's running by default |
| 21:50.05 | [hC] | Qwell: how do i get more? :) |
| 21:50.13 | ManxPower | punkgode: zttool frequently incorrectly reports sync source |
| 21:50.14 | Qwell | [hC]: enable ssh from the GUI |
| 21:50.22 | ManxPower | punkgode: your REAL problem is the IRQ misses. |
| 21:50.29 | Qwell | oh, and other stuff would be running too, I guess.. like SIP/IAX2 |
| 21:50.36 | k31th | ManxPower: it has [general] at the top. |
| 21:50.42 | [hC] | Qwell: yeah... i havent found ssh yet :) |
| 21:51.15 | ManxPower | punkgode: your IRQ misses will cause all of the problems you describe, including echo and dropped calls. |
| 21:52.06 | [hC] | doh |
| 21:52.10 | [hC] | Tab called networking. of course! |
| 21:52.15 | punkgode | ManxPower, yep, is there any way of knowing if the clock source is correct? besides the IRQ problem |
| 21:52.18 | ManxPower | Why is it I don't get ANY calls from my boss until 4:30, then he starts calling |
| 21:52.55 | ManxPower | punkgode: I was told by digium developers that if you set 1 in the 2nd field of the span= line then that is the sync source. |
| 21:53.15 | ManxPower | now if you continue to obsess over the sync source you will waste time. |
| 21:53.31 | punkgode | ManxPower, ok, I'll solve the other problem and pray xD |
| 21:53.37 | punkgode | thxs |
| 21:53.39 | ManxPower | outtolunc: it used to be 6:30pm. I broke him of that habit VERY fast. |
| 21:53.46 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 21:54.21 | k31th | i am now getting SIP/2.0 200 OK but it is not registering. |
| 21:56.43 | Alan_Hicks | What a nightmare today has been. :^) I can't get my Polycom Soundpoint IP 320 to register with Asterisk. I get the following error over and over again in the console. |
| 21:56.47 | Alan_Hicks | [Oct 24 14:55:55] NOTICE[3269]: chan_sip.c:14861 handle_request_register: Registration from '<sip:510@172.16.200.1>' failed for '172.16.200.31' - No matching peer found |
| 21:57.18 | Alan_Hicks | When I attempt to dial an extension (say, 611 which is just an Answer(); Echo() test), I get a busy signal on the phone. |
| 21:57.20 | ManxPower | Alan_Hicks: you do not have a [510] section of sip.conf |
| 21:57.32 | Alan_Hicks | ManxPower: Negative. |
| 21:57.50 | fujin | PASSTEBINNN |
| 21:57.58 | Alan_Hicks | Doing so already. |
| 21:58.02 | ManxPower | Alan_Hicks: then put the [501] section of sip.conf on pastebin.ca change ONLY the password |
| 21:58.12 | fujin | I'm going to go with "you're doing it wrong" |
| 21:58.15 | Alan_Hicks | http://pastebin.com/m7a39f345 |
| 21:58.26 | Xenon3DN | Hey, is anyone here an admin for the asterisk-users mailing list? I've subscribed successfully under two different e-mail addresses, but my message to it asking a question never posts to the list, and I can't figure out why. I'm on tons of other lists just fine... |
| 21:58.30 | fujin | 510 != alan |
| 21:58.31 | fujin | next |
| 21:58.33 | Alan_Hicks | ManxPower: I'm not worried about the password. This is just a test box. |
| 21:58.39 | ManxPower | Alan_Hicks: your phone is trying to register as user "501" It is simple as that. |
| 21:58.43 | Alan_Hicks | fujin: I realize this now. |
| 21:58.49 | fujin | Congratulations |
| 21:58.56 | ManxPower | you need a [501] section of sip.conf or you need to make your phone register as the user you are expecting. |
| 21:58.58 | Alan_Hicks | ManxPower: Ok, thanks. I didn't understand what it was trying to do. |
| 21:59.16 | Alan_Hicks | Got it. Thanks. |
| 21:59.25 | *** join/#asterisk Primer (n=vi@sh.nu) |
| 21:59.27 | ManxPower | Alan_Hicks: you are faster than most 8-) |
| 21:59.39 | Alan_Hicks | fujin: And yes, I was sure I was doing something wrong, just didn't know what. :^) |
| 21:59.41 | ManxPower | Most of the time the user argues with us for a while first. |
| 21:59.56 | Alan_Hicks | ManxPower: I've been around the block, just not the * block. |
| 22:00.14 | Alan_Hicks | You come in ##slackware, and I'm the guy the dumb newbs are arguing with. :^) |
| 22:00.38 | Assid | is there an issue with 1.4.12.1 and voicemail ? |
| 22:00.46 | Assid | apparently its cutting off the users |
| 22:00.55 | generalhan | hey guys, i have never used "Agents" before, so im trying to understand this before i start playing around with it. if i have an agent that is logged in and is heard hold music while they wait for a call to come in, how do they go about making a call? |
| 22:00.57 | *** join/#asterisk [T]ank (n=ckwall@206.71.78.158) |
| 22:01.34 | Primer | Is it possible to have a binary package for asterisk then separate binary packages for each type of voicemail? |
| 22:01.40 | Primer | or must the base asterisk be compiled with a specific type of voicemail? |
| 22:02.58 | trippss | when regsitering two * server to each other via IAX2, say we have IAX2 trunk site1 as [site1] context in iax.conf in site 1 and site2 iax trunk as [site2] context on site 2. on site1 is the registration string site1:pass@site2 or site2:pass@site2? |
| 22:02.59 | Assid | Primer: related to me ? |
| 22:03.18 | [hC] | Primer: are you asking if you can have a compiled asteirsk binary (eg /usr/bin/asterisk) that will work with interchangable app_voicemail.so's, one supporting IMAP and one not? |
| 22:03.33 | Alan_Hicks | "Peer 'alan' is now Reachable." Thanks. |
| 22:03.37 | [hC] | Primer: I'm pretty sure that even if you compile IMAP support in, you can disable it via config directives in voicemail.conf, but dont quote me. |
| 22:03.51 | Primer | Assid: Am I related to you? I have no idea! |
| 22:04.12 | Primer | [hC]: yes, that is exactly what I'm asking |
| 22:04.23 | Primer | [hC]: too late, I'm quoting you! |
| 22:04.39 | [hC] | :) |
| 22:04.56 | ManxPower | "Medtron is working, but the internet is not!" The customer apparently forgot they get to Medtron over the internet |
| 22:05.21 | Xenon3DN | Medtron sounds like a giant fighting robot. |
| 22:05.36 | ManxPower | Xenon3DN: it's a medical billing outsource company |
| 22:05.55 | Xenon3DN | \Of course. It just sounds like it should be the name of a giant fighting robot. ;) |
| 22:06.16 | ManxPower | Trust me, if you have ever called tech support you would think they are a giant fighting robot. |
| 22:06.27 | Xenon3DN | Biiig ;) |
| 22:07.34 | nestAr | anyone know why when i use the rc.debian.asterisk script that my zap channels don't work? |
| 22:08.19 | ManxPower | Xenon3DN: this customer apparently does not realize that 90% of the problems reported to their consultants end up being forwarded to me to fix. But, no, they can't just e-mail me direct, they have to have someone screw up their request first before I get it. |
| 22:08.37 | nestAr | lol |
| 22:09.16 | ManxPower | The only IT think I don't do for them is desktop support, printer support, and physical wiring stuff. |
| 22:09.48 | ManxPower | I wonder how exactly they think their cable monkey can fix a call routing problem. |
| 22:10.05 | Xenon3DN | Aw, VOIP and printers are a natural couple! |
| 22:10.27 | Alan_Hicks | ManxPower: I know the feeling. Today, I could have spent all morning working on this stuff and be hours ahead of where I am now, but no... some luser decided they had to call their third-party e-mail host and delete a bunch of old users and change everyone's password, and naturally they screwed it all up. |
| 22:10.33 | peanut- | I want a giant fighting robot.. |
| 22:10.54 | Alan_Hicks | If they had called me first, it would have taken me 15 minutes to make the changes and things would have been done right the first time. |
| 22:10.56 | mvanbaak | I want a giant bag of money |
| 22:11.04 | Qwell | peanut-: Qwell Communications will make you one - for the right price. |
| 22:11.28 | ManxPower | Qwell: Only if they don't get shutdown by the NSA |
| 22:11.29 | mvanbaak | peanut-: beware, it will be based on chan_skinny.c |
| 22:11.58 | *** join/#asterisk unstable (n=unstable@tor/regular/sid) |
| 22:12.02 | unstable | I'm looking for a good analog phone, one of those nice office speaker phones, and a cordless headset to go with it. Anyone know a good product? |
| 22:12.26 | Siya | when building * from svn are there prerequisits for the asterisk-addons? |
| 22:12.44 | Siya | like mp3 packages which need to be preinstalled etc? |
| 22:12.53 | ManxPower | Siya: that is the funniest question I have seen all week. |
| 22:13.30 | *** join/#asterisk geminidomino (n=vircuser@fl-207-30-169-168.sta.embarqhsd.net) |
| 22:13.58 | Siya | ManxPower: glad to have made your day :) |
| 22:14.15 | Assid | err |
| 22:14.23 | Assid | i think this version has a problem with voicemail |
| 22:14.51 | Assid | when its recording.. if ts for 3-4 seconds and nothing is said and the call is cut.. the file size is just 0KB |
| 22:14.57 | Primer | Assid: my question was not related to yours, btw. Sorry for the obscure answer |
| 22:14.59 | Xenon3DN | unstable: I think this is just what you need. ;) |
| 22:15.00 | Xenon3DN | http://xenon.arcticus.com/automatic-electric-company-model-40-telephone |
| 22:15.15 | unstable | heh |
| 22:15.17 | geminidomino | Ok... I need some serious n00b help here... I've tried two machines, Ubuntu and trixbox, with a known working T1 card, and the zaptel driver just won't see it. The system does (it shows up in lspci) though. I've rebooted, restarted the zaptel service, rebuilt the drivers... any ideas what obvious thing I'm missing? |
| 22:15.34 | codefreeze | Siya: Advice: read the source in the addons. They usually tell about what they need. Look around there for docs. |
| 22:15.49 | Siya | codefreeze: thanks |
| 22:15.53 | unstable | Xenon3DN: So there is no solution for me? |
| 22:16.14 | Xenon3DN | Sorry, I'm not the person to ask. I'm just giving you comedic relief. |
| 22:16.53 | unstable | What is that company.. aas something? |
| 22:16.58 | unstable | they make phones |
| 22:17.37 | *** join/#asterisk dlynes_ (n=dlynes@216.251.149.66) |
| 22:17.41 | tzafrir | generalhan, modprobed the relevant zaptel module? |
| 22:18.21 | tzafrir | generalhan, sorry, meant geminidomino |
| 22:18.31 | generalhan | lol, no worries ! |
| 22:18.34 | tzafrir | geminidomino, what version of zaptel? |
| 22:18.38 | geminidomino | tzafrir: Sure did. |
| 22:18.47 | geminidomino | Tried 1.4 SVN, 1.4.6, and 1.2.15 |
| 22:18.53 | geminidomino | (I was desperate and tried the ubuntu pkg) |
| 22:19.02 | tzafrir | what do you have now? |
| 22:19.06 | geminidomino | 1.4.6 |
| 22:19.19 | tzafrir | lsmod | grep ^zaptel |
| 22:19.31 | geminidomino | its loaded, as is tor2 |
| 22:19.56 | tzafrir | also, what T1 card is it, exactly? |
| 22:20.21 | tzafrir | ls /proc/zaptel |
| 22:20.36 | geminidomino | only thing in /proc/zaptel/1 is the ztdummy... |
| 22:21.16 | peanut- | can you make Monitor() just choose a random file? |
| 22:21.59 | tzafrir | rmmod tor2; modprobe tor2 |
| 22:22.20 | peanut- | or is there a variable you can set to a timestamp and specify that as the lonitor recording file? |
| 22:22.36 | tzafrir | peanut-, make up a name using ${RAND}? |
| 22:22.38 | geminidomino | tzafrir: No dice... |
| 22:22.45 | Qwell | geminidomino: what model card? |
| 22:22.52 | tzafrir | geminidomino, what distro is it? |
| 22:22.54 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 22:23.33 | geminidomino | tzafrir: occurs on both Ubuntu and trixbox 2.2.4 (been fighting with this for 8 hours) |
| 22:23.48 | tzafrir | right now |
| 22:24.00 | geminidomino | trixbox 2.2.4 |
| 22:24.26 | tzafrir | could you please pastebin: tail /var/log/messages |
| 22:24.37 | geminidomino | gimme a sec... lost connection. |
| 22:24.42 | peanut- | tzafrir: I didn't know there was a rand |
| 22:24.52 | Primer | Is it possible to have a binary package for asterisk then separate binary packages for each type of voicemail? Or must the base asterisk always be compiled with a specific voicemail app? |
| 22:25.01 | peanut- | tzafrir: you know how to set a var to be system time? |
| 22:25.05 | geminidomino | Qwell: Trying to remember the brand... It's a tormenta2 from Phoneq or something like that |
| 22:25.24 | JT | Qwell: delayed response, but, nickel and dime are american things. |
| 22:25.25 | Qwell | and you're sure they don't require patches or anything like that? |
| 22:25.26 | tzafrir | peanut-, show function RAND (or is it RANDOM?) |
| 22:25.29 | Qwell | JT: of course |
| 22:25.51 | tzafrir | geminidomino, they have their own module, which they call tor3 |
| 22:25.59 | Qwell | such as that ^ |
| 22:26.06 | *** join/#asterisk Magotari (n=karol@chello089076064182.chello.pl) |
| 22:26.09 | geminidomino | No, I didn't know that... could be the info I needed. :) |
| 22:26.10 | tzafrir | (Phoneiq) |
| 22:26.23 | nestAr | odd, if i comment out the AST_GROUP in the init script, it works fine.. |
| 22:27.04 | geminidomino | hrm... no website |
| 22:27.15 | peanut- | where are functions defined? |
| 22:27.27 | Assid | yeah there seems to be a problem |
| 22:27.49 | Assid | if a voicemail is less than a few seconds.. and theres nothing said.. then the voicemail gsm file size is 0KB |
| 22:28.06 | tzafrir | PhonicEQ, that is |
| 22:28.11 | Assid | also if the file is 0kb, then voicemailmain just hangs upo |
| 22:28.18 | nestAr | as a bonus, i get colors in the console now. |
| 22:28.39 | geminidomino | there they are. Thanks, tzafrir |
| 22:28.45 | peanut- | oh STRFTIME does what I need, neat |
| 22:31.16 | Assid | anyone |
| 22:31.37 | geminidomino | ok... I'll try the tor3 when the connection comes back up. Thanks for pointing it out. |
| 22:32.33 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 22:32.48 | trippss | what would cause a "registration refused" in an iax2 trunk? |
| 22:33.13 | trippss | assuming the password and username are correct/ |
| 22:34.54 | *** part/#asterisk Primer (n=vi@sh.nu) |
| 22:35.10 | geminidomino | Thanks Qwell and tzafrir |
| 22:36.06 | Magotari | Excuse me, I have a question. Is there any problem with running asterisk on User Mode Linux? |
| 22:36.20 | *** join/#asterisk Alowishus (n=jpenix@fwsdo.projectdesign.com) |
| 22:36.23 | Qwell | Magotari: UML is the one that runs in Windows, right? |
| 22:36.36 | Magotari | No, it runs only in Linux. |
| 22:36.40 | Magotari | Linux inside a Linux. |
| 22:36.53 | generalhan | anyone know if you can use global variables in agents.conf ? like in extensions.conf i specify a call recording file using ${DATE} and ${EXTEN}, can i do the same thing in the recording section of agents.conf ?? |
| 22:36.56 | Uatec | why run linux inside linux? |
| 22:37.02 | Uatec | why don't you just run linux? |
| 22:37.19 | Magotari | Because with UML you can have 20 machines in a network inside one physical computer. |
| 22:37.27 | Magotari | Which it what is required here. |
| 22:37.48 | JT | Magotari: uml is completely inappropriate for asterisk |
| 22:37.58 | Magotari | Aha. Any idea why? |
| 22:38.07 | JT | because it has terrible performance |
| 22:38.11 | JT | it proxies all IO calls |
| 22:38.28 | JT | try xen or another virtualisation scheme |
| 22:38.33 | Magotari | Hmm... Let me rephrase my question. |
| 22:38.36 | Magotari | Would it run at all? |
| 22:38.48 | JT | yes, audio would be stuffed |
| 22:39.20 | Magotari | Yes, I got the idea, I just wanted to know if it would run. Thanks for help everyone. Goodnight. |
| 22:39.29 | fakhir | can anyone clue me in to why my did context seems to not want to work for me |
| 22:39.30 | fakhir | http://pastebin.com/m29400ae9 |
| 22:39.30 | fakhir | i want calls from a particular number to go to a voice menu but all other calls to go to a ring group |
| 22:39.30 | fakhir | problem is even when i call from that number the call gets sent to the ring group |
| 22:42.11 | *** join/#asterisk dmangot (n=dmangot@pnapgw.terracottatech.com) |
| 22:42.34 | dmangot | Does anyone know when the 'send me my password' functionality on Asterisk.Org will get fixed? |
| 22:42.34 | generalhan | anyone have any insight on using things like ${DATE} inside agents.conf ?? |
| 22:42.55 | JT | what sort of insight? |
| 22:43.51 | karleeto | where is redhat or centos or trixbox supposed ot run /usr/sbin/zaphpec |
| 22:43.56 | karleeto | _enable |
| 22:43.59 | generalhan | JT: i use extensions.conf to specify a CallFileName to specify where to save recorded(monitor()) calls go. but i want to start playing around with agents and there is a section in there to specify a folder, i would like to use some of thsoe variables in that path |
| 22:44.28 | generalhan | wow, typing nightmare, sorry about that ! lol |
| 22:45.14 | karleeto | modprobe.conf> |
| 22:49.13 | generalhan | well, i just tried it, and the answer is, no ${DATE} does not work in there ! lol |
| 22:49.39 | peanut- | I have option 'n' set in my Monirot() but I still get an -in.wav and a -out.wav, why is that? |
| 22:50.37 | generalhan | that sux, this agent thing was looking really promising ... but there is no way i can just one folder for all the calls with 5,000 - 10,000 calls a day that would become impossible to manage. there has to be some way to specify a file name and path for the agent recording |
| 22:51.15 | JT | peanut-: MixMonitor |
| 22:52.35 | generalhan | JT: any ideas on how i might get that done ? |
| 22:56.23 | *** join/#asterisk craigk (n=ckowald@58.174.122.198) |
| 22:57.04 | *** part/#asterisk Xenon3DN (n=Xenon@mail.3dnature.com) |
| 23:03.34 | *** join/#asterisk Dovid (n=Dovid@bzq-88-155-170-112.red.bezeqint.net) |
| 23:03.36 | Dovid | hi |
| 23:03.42 | Dovid | is it possible to have under sip.conf |
| 23:03.51 | peanut- | no. |
| 23:04.33 | Dovid | host=212.212.212.0 (so that any host starting with 212.212.121.X will work ) ? |
| 23:05.15 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 23:05.18 | *** join/#asterisk agx (n=badpengu@81-174-46-216.dynamic.ngi.it) |
| 23:05.29 | *** part/#asterisk dmangot (n=dmangot@pnapgw.terracottatech.com) |
| 23:07.56 | TrentCreek | I see i3nary has not been here in a few days. His home USED to be in an Diego |
| 23:11.10 | Assid | err.. can someone clarify this for me ? [Oct 24 19:10:33] WARNING[4715]: app_voicemail.c:6960 vm_exec: Prefixing the mailbox with an option is deprecated ('u205@ila') |
| 23:11.34 | dlynes_laptop | Assid: Voicemail(205@ila,u) instead |
| 23:11.57 | Assid | crap.. i hope i have this as a macro |
| 23:11.59 | dlynes_laptop | Assid: The new voicemail application is more flexible than the old one |
| 23:12.10 | Assid | how would it be any different? |
| 23:12.21 | Assid | they access the smae thing |
| 23:12.22 | dlynes_laptop | Assid: now you're not limited to one character for options |
| 23:12.37 | dlynes_laptop | Assid: you can also have multiple parameters |
| 23:13.09 | dlynes_laptop | Assid: i.e. Voicemail(205@ila|optiona|optionb|optionc|optiond|...) |
| 23:13.17 | Assid | eh.. its gonna play the unvailable or busy depending what i send it.. i dont get this.. how can you be busy AND unavailable ? |
| 23:13.30 | Assid | so it plays 2 recording? |
| 23:13.36 | dlynes_laptop | Assid: no |
| 23:13.53 | dlynes_laptop | Assid: I believe they changed it to allow for future expansion...adding options that it doesn't currently support |
| 23:14.23 | Assid | okay thank god i have it in a macro.. lemme see if this will work |
| 23:14.31 | Assid | 205@ila,u right ? |
| 23:14.48 | Assid | or | as well |
| 23:15.35 | dlynes_laptop | Assid: correct |
| 23:15.59 | Assid | okay fixed that.. |
| 23:16.16 | Assid | but i still do think voicemail has a bug ifd you leave a message less than a few seconds , and dont say anything |
| 23:16.23 | Assid | i saw 0KB gsm files |
| 23:16.34 | Assid | and then voicemailmain just cuts you off |
| 23:16.46 | Assid | i..e when you go and try to access it |
| 23:17.01 | ManxPower | Assid: set your minmessagelength |
| 23:17.23 | ManxPower | Assid: the ONLY time I've seen 0byte voicemail files is when the filesystem was full. |
| 23:17.39 | ManxPower | even if there was no audio, it would still need a header to be a valid gsm file. |
| 23:17.48 | Assid | ManxPower: had to clean out 22 messages.. i know |
| 23:18.22 | Assid | ManxPower: cant explain why.. ive had a few people leave me 2-3 seconds voicemails on my other box.. (older asterisk) worked fine |
| 23:21.17 | Assid | yeah.. 1.2 didnt show me this bug |
| 23:21.48 | Assid | err is there a way to save to gsm + mp3 (for emailing) ? |
| 23:22.11 | ManxPower | Assid: you can save as mp3????? |
| 23:22.26 | Assid | i get my emails on my cell phone.. and i cant open gsm files |
| 23:22.27 | Alan_Hicks | Hey guys. Hoping you'll help me out once more. |
| 23:22.36 | Assid | ManxPower: nope.. was hoping someone would have an idea |
| 23:23.04 | Alan_Hicks | I setup a simple dialplan to call the second line on my SIP phone. "exten => 100,1,Dial(SIP/alan)" |
| 23:23.29 | Alan_Hicks | When I do this though, I pick up the phone and dial "100" and nothing happens. |
| 23:23.40 | ManxPower | Alan_Hicks: the brand (and sometimes model) are important anytime you say "SIP phone" |
| 23:23.53 | Alan_Hicks | If I change it to "exten => 511,1,Dial(SIP/alan)" and dial 511, everything works as expected. |
| 23:24.03 | Alan_Hicks | ManxPower: Polycom Soundpoint IP 320. |
| 23:24.18 | ManxPower | Alan_Hicks: in SIP the "dialplan" is controlled by the phone. The phone collects the digits, then sends them to asterisk en-mass when it thinks it should. |
| 23:24.19 | punkgode | Alan_Hicks, check your phone's dialplan |
| 23:24.40 | Alan_Hicks | Ok, that makes sense. |
| 23:24.45 | ManxPower | Alan_Hicks: press the Send softbutton when you are done dialing (you can fix this issue later) |
| 23:25.02 | Alan_Hicks | Any ideas on what to do to check the phone's dialplan? |
| 23:25.08 | Alan_Hicks | Just RTFM? |
| 23:25.27 | punkgode | Alan_Hicks, yep, depends on your phone |
| 23:25.28 | ManxPower | Alan_Hicks: pretty much. Polycoms have a steep learning curve, but they are some of the best phones out there. |
| 23:25.51 | ManxPower | Alan_Hicks: you can connect to the phone using a web browser for BASIC setup if you don't want to set up a provisioning server. |
| 23:26.11 | ManxPower | (the phone web server takes a while to start after the phone is done booting) |
| 23:26.33 | ManxPower | Alan_Hicks: you should get the Admin Guide for the Polycom. |
| 23:26.42 | Alan_Hicks | Yeah, I had to do that to setup the phone anyhow as it wouldn't let me enter the correct IP address to asterisk if I didn't. |
| 23:27.00 | Alan_Hicks | I didn't see anything about a dialplan in it, but I'll double check and read the docs. Thanks. |
| 23:28.24 | Alan_Hicks | ManxPower: Got it, just gotta read it. |
| 23:28.38 | ManxPower | in the dialplan "|" separates entries. "x" means any single digit. "." means 'one or more digits' or 'zero or more digits' (I can never remember which, in Asterisk's dialplan "." means 1 or more digits". "," means continue dialtone. "[2-5]" means any single digit 2-5. "T" means wait for the timeout (default to 3 seconds, I think) |
| 23:29.05 | ManxPower | Alan_Hicks: look at the current dialplan on the phone using the web interface |
| 23:29.27 | punkgode | Assid, think out of the box, setup a linux box with fetchmail, extract the attachment convert it to mp3 an resend :P. I'm joking... or maybe not... :) |
| 23:29.47 | Assid | too complicated |
| 23:29.53 | punkgode | Assid, booo |
| 23:29.55 | punkgode | xD |
| 23:30.33 | Alan_Hicks | ManxPower: Maybe I'm just dumb, but I'm not seeing it there. |
| 23:31.00 | ManxPower | Alan_Hicks: I've not configured a polycom phone via the web interface in years. But trust me, it's in there. |
| 23:31.07 | Alan_Hicks | ok |
| 23:31.36 | Alan_Hicks | Could it possibly be named something obscure and different? |
| 23:31.43 | *** join/#asterisk kev88 (n=kev8888@70.51.58.167) |
| 23:31.46 | ManxPower | Alan_Hicks: http://www.fnords.org/~eric/polycom-config-examples/ Save that URL for when you are ready to set up a provisioning server. |
| 23:32.09 | *** join/#asterisk jameswf (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 23:32.12 | Alan_Hicks | Thanks. |
| 23:32.14 | *** join/#asterisk Ritzerisk (n=Ritztech@24.120.190.66) |
| 23:32.25 | Ritzerisk | how do i add the most recent Gui .. |
| 23:32.38 | ManxPower | Ritzerisk: We don't do GUIs here. |
| 23:32.46 | ManxPower | go to the channel for your GUI (whatever it is) |
| 23:32.57 | *** join/#asterisk coppice (n=chatzill@8.155.17.210.dyn.pacific.net.hk) |
| 23:32.58 | *** join/#asterisk marl (n=matt@82-40-218-233.cable.ubr01.dunb.blueyonder.co.uk) |
| 23:33.06 | ManxPower | Alan_Hicks: what is on the top of the phone web page? |
| 23:33.27 | Alan_Hicks | ManxPower: Home, General, network, SIP, and Lines. |
| 23:33.35 | ManxPower | Alan_Hicks: try lines |
| 23:33.42 | ManxPower | or SIP |
| 23:33.43 | Alan_Hicks | It's not there. I looked. :^) |
| 23:33.51 | Assid | alrite im outta here |
| 23:33.58 | Assid | 5 am aint a good time for me to be awake |
| 23:33.59 | ManxPower | what are you using to log in. Admin as the username? |
| 23:34.05 | Alan_Hicks | Polycom/456 |
| 23:34.10 | marl | hi, can someone point me to a good howto on running asterisk for multiply setups on one box? eg. running 3 seperate company pbx's from one box, without using vserver style setup? |
| 23:34.11 | ManxPower | Ah, yes, that would be correct. |
| 23:34.15 | ManxPower | what version of the firmware? |
| 23:34.24 | *** join/#asterisk fskrotzki (n=fskrot@cpe-74-74-245-250.rochester.res.rr.com) |
| 23:34.38 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 23:34.47 | ManxPower | marl: no. However, ALL of that would be controlled by "contexts" |
| 23:35.13 | Alan_Hicks | Would that be the Bootrom version number? 3.2.3.0021. |
| 23:37.06 | ManxPower | no, but if you have bootrom 3.2, then you have a non-ancient SIP firmware |
| 23:37.11 | marl | thanks ManxPower, had looked at that, but was wandering if there was any other way of doing it |
| 23:38.24 | kev88 | Anyone running Music on hold with IAX2 on Asterisk? |
| 23:38.56 | marl | anyone know how to stop asterisk trying to access /dev/tty9 ? |
| 23:39.19 | Alan_Hicks | Don't run safe_asterisk, or edit the script? |
| 23:39.21 | ManxPower | marl: I believe that is controlled by the init script or safe_astersik |
| 23:39.42 | kev88 | Any idea why Music on hold doesn't work on IAX2? |
| 23:39.58 | *** join/#asterisk irule (n=irule@200.53.61.4) |
| 23:40.01 | ManxPower | kev88: doesn't work or is garbled/distorted? |
| 23:40.14 | kev88 | Well, it shows it's playing in the CLI, but it's totally silent |
| 23:40.26 | irule | hi there, where may I find a complete dialplan so I may just add my sip phones¡ |
| 23:40.36 | irule | ?????????????? |
| 23:40.40 | ManxPower | irule: No such thing exists. |
| 23:40.46 | irule | why? |
| 23:40.50 | ManxPower | It cannot exist because Asterisk is not a PBX. |
| 23:41.01 | ManxPower | It is a toolkit that allows you to build a PBX. |
| 23:41.33 | ManxPower | irule: if you want a plug-n-pray setup then buy a commercial product (asterisk based or not, we don't care) |
| 23:41.38 | irule | ManxPower well, isnt a nice dialplan missing for the average joe? |
| 23:41.43 | JT | no |
| 23:41.49 | JT | there is no average setup |
| 23:41.58 | ManxPower | irule: every single asterisk box out there has a different dialplan |
| 23:42.21 | ManxPower | irule: where are you located? |
| 23:42.35 | ManxPower | Even with the PSTN each country has it's own national dialplan |
| 23:42.37 | irule | well, I found out that I must react to a zillion situations even on the most basic of setups |
| 23:42.44 | irule | Mexico |
| 23:43.25 | ManxPower | irule: yes, asterisk setup is difficult, complex, error prone, and just downright nasty. If you don't want to deal with that then buy a commercial solution. |
| 23:43.33 | irule | well, wouldnt there be less basic questions if we had a dialplan wiki where joe may just copy/paste or something? |
| 23:43.42 | Alan_Hicks | Or you could end up like me and chasing down your phone's dialplan. |
| 23:43.45 | ManxPower | irule: define "dialplan" |
| 23:44.05 | Alan_Hicks | Given that each phone is different, the same dialplan wouldn't work for each setup, even if Asterisk was configured the exact same way. |
| 23:44.08 | Alan_Hicks | Right? |
| 23:44.08 | irule | ManxPower yes I get it, you work for some PBX company that has not hit NYSE yet :S |
| 23:44.24 | ManxPower | Alan_Hicks: correct. |
| 23:44.24 | marl | wat is the differance between safe_asterisk and asterisk? |
| 23:44.26 | *** join/#asterisk grimsy (n=chatzill@203-206-220-214.perm.iinet.net.au) |
| 23:44.37 | ManxPower | marl: safe_asterisk is a shell script. Go read it. |
| 23:44.50 | Alan_Hicks | marl: asterisk is a binary, safe_asterisk is a script that calls asterisk with options. |
| 23:45.06 | codefreeze | irule: yes, the permutations to cover all the different situations would be in the millions or billions. A generic dialplan is the holy grail of just about every commercial developer out there that bases their solutions on asterisk. |
| 23:45.13 | ManxPower | irule: I'm a tech consultant. |
| 23:46.54 | irule | codefreeze but wouldnt it be nice to hace at least a nice dialplan to build on? I mean, that takes all the core functions into consideration, documented and commented, that allow you to start a little further than, what to do when there is BUSY? |
| 23:47.00 | ManxPower | irule: here are some things that can impact your "dialplan", your connection to the PSTN (Zap, IAX, SIP) as well as the specific config options for each provider, the sip phones (each sip phone requires different configuration), your extensions (how many digits, what digits are allowed, not allowed) |
| 23:47.05 | Ritzerisk | i dont seem to see a trunk config file |
| 23:47.35 | ManxPower | Ritzerisk: that is because Asterisk does not support trunks except for IAX2 trunking. |
| 23:48.39 | Ritzerisk | haha . |
| 23:49.00 | marl | thanks ManxPower, Alan_Hicks, had not relised that safe_ast was a script, got it sorted :) |
| 23:49.12 | ManxPower | I'll bet he things us Asterisk people knows about whatever GUI he is using to control Asterisk. |
| 23:49.18 | Alan_Hicks | marl: Check the book. It'll really help you along. |
| 23:49.22 | Alan_Hicks | ~book |
| 23:49.23 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 23:49.26 | Ritzerisk | someone told me wrong though ... what config would i put my context-from-pstn and dtmf mode |
| 23:49.33 | codefreeze | irule: trixbox and others have 'generic' dialplans that cover all sorts of contingencies. It makes for really complicated and messy dialplans. |
| 23:49.41 | ManxPower | Ritzerisk: what protocol? |
| 23:49.52 | Ritzerisk | sip |
| 23:50.02 | ManxPower | all sip stuff is in sip.conf |
| 23:50.04 | Ritzerisk | would it be the sip conf |
| 23:50.07 | codefreeze | irule: so we basically just give the configs/example.extensions.conf & ael |
| 23:50.08 | Ritzerisk | k |
| 23:50.21 | Ritzerisk | even a sip trunk |
| 23:50.29 | ManxPower | THERE IS NO SUCH THING AS A SIP TRUNK!!!!!!! |
| 23:50.42 | irule | codefreeze indeed, I tested that, just download the code, took out the dialplan and built onthat, and learned a lot on the way, but it is not documented so some part are just way over my league |
| 23:51.00 | ManxPower | I don't care what some craptastic gui calls them, they are not "sip trunks" |
| 23:51.29 | Alowishus | do they achieve the same result? |
| 23:51.33 | ManxPower | sorry, I should have done /ignore rather then hold down the shift key. |
| 23:51.48 | ManxPower | Much better. |
| 23:51.48 | Ritzerisk | haha well via the mitel i have to purchase Sip Tunks so what are they called in asterisks case |
| 23:51.56 | irule | ManxPower I own a linksys SPA3102 that has 1 FXO and 1 FXS, that will be a SIP trunk! :P |
| 23:52.00 | Alowishus | Ritzerisk: you're probably thinking of a SIP gateway |
| 23:52.02 | JT | <PROTECTED> |
| 23:52.07 | codefreeze | irule: if you want to start some awesome, generic dialplan that's well commented, works 'out of the box', etc. etc, --no-one will stop you...! |
| 23:52.08 | Alowishus | I think it's just a terminology thing |
| 23:52.11 | JT | no such thing as a sip trunk |
| 23:52.14 | JT | ever |
| 23:52.25 | Alan_Hicks | ManxPower: Wow, I really don't understand any of those examples you linked me to. :^) |
| 23:52.34 | Ritzerisk | bandwidth.com uses them to send off to all the Voip switches |
| 23:52.48 | ManxPower | Alowishus: partially a terminology thing. |
| 23:52.54 | JT | what does bandwidth.com use Ritzerisk ? |
| 23:52.58 | codefreeze | irule: and after you've got it written, there's probably 150 people or so, that will send you further requirements! |
| 23:52.58 | Alowishus | Alan_Hicks: you having Polycom trouble? |
| 23:53.15 | ManxPower | Alan_Hicks: you will someday -- that's why I said save the URL. |
| 23:53.16 | Alan_Hicks | Alowishus: I'm having Polycom/Asterisk learning curve. |
| 23:53.24 | irule | so, who wants to join me in creating a cool open source generic dialplan to build on? |
| 23:53.25 | Alan_Hicks | ManxPower: And that's why I did. |
| 23:53.44 | ManxPower | Someone that has a polycom handy might want to help Alan_Hicks find the dialplan stuff via the web interface. |
| 23:54.04 | ManxPower | irule: good luck with that. |
| 23:54.06 | Alan_Hicks | Alowishus: I'm more than a little green. Right now I'm just getting a base of knowledge that I can build on. |
| 23:54.55 | irule | who is the #asteirsk maintaner, is there someone online right now? |
| 23:55.01 | Alowishus | Alan_Hicks: sec I can maybe help guide you |
| 23:55.11 | Alan_Hicks | Alowishus: Thank you. |
| 23:55.43 | ManxPower | irule: there isn't one. There are a few people with op status that get involved if someone is massively disruptive to the channel but I've only seen that happen a few times in the years I've been her. |
| 23:56.15 | Ritzerisk | haha. |
| 23:56.26 | irule | well, once I have a web address I will ask it to be included in the /topic :D |
| 23:56.34 | Alan_Hicks | Alowishus: It's a Polycom Soundpoint IP 320 if that helps you. |
| 23:56.56 | ManxPower | Alan_Hicks: for the most part all polycoms have the same basic firmware. |
| 23:57.20 | JT | the topic is full enough as is |
| 23:57.37 | irule | naaa lol |
| 23:57.44 | Qwell | irule: there is always somebody around |
| 23:59.00 | irule | cool thans for the repply Qwell, I want to start an open source dialplan poroject because it is necesary for many people to fully get into asterisk faster, no offense but, help in here would have better quality with this included, dontya think? |
| 23:59.14 | alrs | irule: adhearsion.com |
| 23:59.41 | Alowishus | Alan_Hicks: the basics are that the phone boots and picks up the <mac>.cfg file... which tells it which config files to pick up next... it reads those left to right and loads them in order, and values defined in earlier files override ones in later files... so basically you have your specific -phone config, all the way down to the Polycom provided sip.cfg which has defaults for everything else you didn't define |
| 23:59.47 | Qwell | open source dialplan? |