| 00:00.25 | jblack | workout time. have fun all |
| 00:00.43 | Katty | byebye |
| 00:06.00 | gilevy | i am getting this error: sent into invalid extension 'return' in context 'from-did-direct', but no invalid handler |
| 00:06.33 | *** join/#asterisk KavanS (n=KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 00:06.48 | Katty | gilevy: look at your goto |
| 00:06.50 | gilevy | but i cannot find the 'from-did-direct' context in extensions_additional.conf |
| 00:06.58 | *** join/#asterisk sahafeez (n=sahafeez@65-119-47-100.dia.static.qwest.net) |
| 00:06.58 | gilevy | i see the goto |
| 00:06.59 | Katty | oh. nevermind. |
| 00:07.13 | Katty | lookings like you're using trixbox or something. |
| 00:07.19 | gilevy | freepbx |
| 00:07.59 | gilevy | no good? |
| 00:08.09 | Katty | you might ask in #freepbx. |
| 00:08.24 | *** part/#asterisk dlewis (n=dlewis@about/security/staff/dlewis) |
| 00:09.23 | gilevy | fair enough thanks |
| 00:09.28 | *** part/#asterisk gilevy (n=gil@c-24-10-28-163.hsd1.ca.comcast.net) |
| 00:15.48 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 00:17.38 | *** join/#asterisk darkskiez_ (n=dz@62-50-230-103.client.stsn.net) |
| 00:20.58 | *** join/#asterisk _bugz_ (n=bugz@adsl-99-154-128-215.dsl.lsan03.sbcglobal.net) |
| 00:27.35 | *** join/#asterisk KavanS (n=KavanS@173.50.141.22) |
| 00:34.44 | jblack | that was fun |
| 00:42.40 | *** join/#asterisk VooDooNOFX (n=Joe@rrcs-24-43-123-93.west.biz.rr.com) |
| 00:46.02 | *** join/#asterisk gilevy (n=gil@c-24-10-28-163.hsd1.ca.comcast.net) |
| 00:46.16 | *** part/#asterisk gilevy (n=gil@c-24-10-28-163.hsd1.ca.comcast.net) |
| 00:49.24 | *** join/#asterisk KavanS (n=KavanS@173.12.184.89) |
| 00:50.02 | *** join/#asterisk sircolin (n=sircolin@my83-216-68-241.mynow.co.uk) |
| 00:56.28 | ChannelZ | fuck now people are using Flickr to spam!? |
| 01:20.05 | *** join/#asterisk KavanS (n=KavanS@71.117.242.28) |
| 01:48.59 | *** part/#asterisk sahafeez (n=sahafeez@65-119-47-100.dia.static.qwest.net) |
| 01:59.04 | drmessano | I hate google |
| 02:09.24 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 02:21.12 | p3nguin_ | Is this something to worry about? NOTICE[24233]: rtp.c:1285 ast_rtp_read: Unknown RTP codec 100 received from '24.171.73.149' |
| 02:36.41 | ChannelZ | no.. that's g729a I think (if its hex) |
| 02:37.25 | *** join/#asterisk KavanS (n=KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 02:39.38 | ChannelZ | or maybe not.. hmm |
| 02:45.36 | ChannelZ | I dunno what 100 is. Undefined |
| 02:46.36 | ChannelZ | http://www.asteriskguru.com/tutorials/unknown_codec_received.html #6 ? |
| 02:46.50 | drmessano | 6) Unknown RTP codec 100 received: |
| 02:46.50 | drmessano | You probably connected some fax to some ATA, and then try to send faxes over t.30 to asterisk, this is not good, this fails in almost all cases. |
| 02:46.50 | drmessano | The real solution is to use a t.38 enabled ATA and patch asterisk to use t.38. |
| 02:46.50 | drmessano | references: |
| 02:46.50 | drmessano | http://bugs.digium.com/view.php?id=5090 |
| 02:47.01 | drmessano | gah |
| 02:48.04 | ChannelZ | yes pasting is fun! |
| 02:48.48 | drmessano | I didn't intend to paste all of it, hence the "gah" |
| 02:48.56 | ChannelZ | :) |
| 02:48.58 | drmessano | Thought that would have been self explanatory |
| 02:49.06 | ChannelZ | it was |
| 02:49.34 | drmessano | Thats not what Ubuntu told me about you |
| 02:49.47 | drmessano | Sure, sometimes, Ubuntu lies |
| 02:51.05 | ChannelZ | unga bunga |
| 02:54.04 | Katty | mmm, this salad is divine. |
| 02:54.48 | Katty | tho the avacado isn't really ripe enough :/ |
| 03:02.25 | *** join/#asterisk Xetrov` (n=xetrov@unaffiliated/xetrov/x-827361) |
| 03:20.15 | p3nguin_ | I had a bread bowl from Domino's, so I'm good for a while. Bread and pasta... lots of carbs! |
| 03:21.05 | *** join/#asterisk psilikon (n=psilikon@221-116.96-97.tampabay.res.rr.com) |
| 03:21.47 | *** join/#asterisk btsteve (n=tstevens@24-196-234-39.dhcp.gwnt.ga.charter.com) |
| 03:24.09 | p3nguin_ | Is there a guide for allowing me to dial into my * system, enter a PIN, and then receive a dial tone which will allow me to then dial out? My brain isn't working well enough tonight to know where to start. |
| 03:24.20 | ChannelZ | Disa() |
| 03:24.44 | p3nguin_ | I don't want to set up billing and calling cards and crap like that. Just a simple gateway will do. |
| 03:25.05 | ChannelZ | Disa() |
| 03:25.23 | p3nguin_ | Yes, I'm checking core show application DISA right now. |
| 03:25.38 | p3nguin_ | This will likely get me going in the right direction. Just needed a kickstart. |
| 03:26.16 | ChannelZ | Or you can roll your own by making an extension that is your PIN, and then making it jump to a context with a pattern extension for dialing out |
| 03:26.45 | p3nguin_ | That's a good idea, too. |
| 03:31.23 | ChannelZ | hmm ok how do I get my * to do ringers and things as UK? |
| 03:31.52 | ChannelZ | oh wait it is working |
| 03:31.59 | ChannelZ | I guess Zoiper makes it's own tones |
| 03:34.58 | psilikon | I was getting an error trying 'make menuselect' with * 1.6.0.15 on CentOS 5.4. Keep getting a message stating that ncurses and/or newt are not installed, but they are. |
| 03:35.40 | ChannelZ | curses! |
| 03:36.28 | psilikon | works with 1.6.1.6 |
| 03:36.47 | ChannelZ | ARGH Skype pisses me off with all these fucking pop-ups and tips and suggestions |
| 03:36.57 | ChannelZ | YES I know my microphone is quiet, because I haven't fucking said anything yet |
| 03:46.08 | Katty | scooby doo |
| 03:46.10 | Katty | WHERE ARE OU |
| 03:47.49 | KavanS | does anyone do anything fancy with dual wan/voip providers? |
| 03:47.56 | KavanS | i.e. trunk fail detection, etc? |
| 04:03.04 | *** join/#asterisk dkirker (n=dkirker@99.2.134.30) |
| 04:13.26 | *** join/#asterisk JonathanB (n=Adium@cpe-24-242-201-118.tx.res.rr.com) |
| 04:21.40 | *** join/#asterisk voipmonk (n=voipmonk@dsl-67-204-18-84.acanac.net) |
| 04:31.41 | *** join/#asterisk voipmonk (n=voipmonk@dsl-67-204-18-84.acanac.net) |
| 04:49.14 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 04:58.13 | *** join/#asterisk voipmonk_ (n=voipmonk@CPE00222d5081e0-CM00222d5081dc.cpe.net.cable.rogers.com) |
| 05:09.08 | raden | Katty scooby doo bee dooo |
| 05:11.31 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 05:23.46 | *** join/#asterisk linagee6 (n=linagee@barghest-1-pt.tunnel.tserv3.fmt2.ipv6.he.net) |
| 05:24.17 | linagee6 | [TK]D-Fender: ack. wtf. I was googling myself and I remember one time you said you were a JOAT and somehow I knew that meant jack of all telephones. Why could I not have figured that out now? :( |
| 05:25.27 | [TK]D-Fender | linagee6: .... huh? |
| 05:25.31 | linagee6 | [TK]D-Fender: n/m |
| 05:25.42 | linagee6 | [TK]D-Fender: http://ibot.rikers.org/%23asterisk/20071121.html.gz |
| 05:26.45 | linagee6 | corydon, [TK]D-Fender, dad, linagee, etc. the cruft of IRC. :( |
| 05:26.53 | [TK]D-Fender | linagee6: Wow... doesn't quite sound like me... |
| 05:27.03 | linagee6 | [TK]D-Fender: nor me. geez how time has changed things |
| 05:27.35 | linagee6 | [TK]D-Fender: wow. 2 years ago. that's like forever in internet time |
| 05:29.00 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 05:29.00 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 05:29.22 | *** join/#asterisk JKac3BEq (n=JKac3BEq@173.20.68.31) |
| 05:36.19 | *** join/#asterisk JKac3BEq (n=JKac3BEq@173-20-68-31.client.mchsi.com) |
| 05:51.45 | raden | what word is in the orion constellation ? |
| 05:54.17 | leifmadsen | con? |
| 05:54.22 | leifmadsen | tell? |
| 05:54.32 | leifmadsen | or? |
| 05:54.34 | leifmadsen | ion? |
| 05:56.36 | jplank | stencil |
| 05:57.39 | jplank | I was never good at this game |
| 06:03.40 | [TK]D-Fender | is now known as ... oh wait... I was always just me :p |
| 06:11.42 | pwn3dnewb | :( |
| 06:13.13 | ChannelZ | is now known |
| 07:08.20 | phix | hey |
| 07:08.26 | phix | I don't know you ChannelZ |
| 07:18.01 | ChannelZ | Clearly you need to scroll back. |
| 07:32.04 | *** join/#asterisk ttl- (n=patrick@94-224-78-192.access.telenet.be) |
| 07:41.53 | *** join/#asterisk tzafrir__laptop (n=tzafrir@local.xorcom.com) |
| 07:47.15 | TJNII | Wow. I'm in that log, too. Cool? |
| 08:02.46 | *** join/#asterisk frk2 (n=faraz@zivios/member/fkhan) |
| 08:21.20 | TJNII | This is completely unrelated to *, but does anyone know why ioctl(Serial_fd, TIOCMIWAIT, &status) wouldn't block like it is supposed to? |
| 08:22.10 | TJNII | I will be so happy when I get PICs talking ethernet so I can drop this RS-232 crap. |
| 08:24.20 | *** join/#asterisk VooDooNOFX (n=joe@user-118brks.cable.mindspring.com) |
| 08:27.18 | VooDooNOFX | I'm having some NAT issues (I think it's NAT related) with one way audio. Is there a way to disable Packet2Packet bridging, and allow all traffic to flow over * to help fight those issues? |
| 08:28.58 | TJNII | Because the third argument isn't a pointer with TIOCMIWAIT..... |
| 08:29.18 | TJNII | Which means this code used to work on luck... |
| 08:29.28 | TJNII | VooDooNOFX: Yes |
| 08:29.36 | TJNII | 1 sec. |
| 08:30.05 | TJNII | canreinvite=no will keep * in the RTP path. |
| 08:33.33 | VooDooNOFX | i've already got it for the sip trunk, i'll add it for each phone also |
| 08:35.40 | TJNII | Or in general.. |
| 08:35.48 | VooDooNOFX | like magic. it works just fine now |
| 08:36.31 | VooDooNOFX | Is there a "monitor" application that can be installed to allow an phone user to visually see the status of other * users' telephones? |
| 08:36.48 | VooDooNOFX | or to see the queue? |
| 08:37.47 | TJNII | Quiet! Before you get [TK]D-Fender ranting about key systems! |
| 08:37.58 | TJNII | Look into dialplan hints. |
| 08:38.13 | TJNII | I don't remember what exactly it is called, BLF or something like that. |
| 08:39.33 | VooDooNOFX | yeah, I've seen that. But it looks like it just notify's a phone or web application that a call is waiting |
| 08:39.50 | VooDooNOFX | I was hoping to see something like the 3cx call queue viewer |
| 08:40.58 | TJNII | It notifies if the "extension" is in use. |
| 08:41.22 | TJNII | Or at least that is my understanding / experience. |
| 08:54.00 | *** join/#asterisk crazybyte (n=crzp@unaffiliated/crazypenguin/x-000001) |
| 09:06.19 | TJNII | breaks out the scope |
| 09:06.26 | TJNII | Let's get serious. |
| 09:08.57 | *** join/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 09:09.30 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 09:09.37 | VooDooNOFX | yes. Phones are very serious. |
| 09:09.44 | VooDooNOFX | puts on my serious face |
| 09:10.22 | *** part/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 09:20.03 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 09:23.09 | ChannelZ | I can't find my monocle |
| 09:41.17 | *** join/#asterisk wam (i=wam@unaffiliated/wam) |
| 10:10.41 | *** join/#asterisk Jabka (n=jabka@bzq-79-179-206-10.red.bezeqint.net) |
| 10:14.40 | *** join/#asterisk ming_zym (n=ming_zym@218.108.49.4) |
| 10:18.06 | *** join/#asterisk scardinal (n=supreme@90.184.100.119) |
| 10:35.14 | *** join/#asterisk meesterarend (n=frans@vc-41-192-68-83.umts.vodacom.co.za) |
| 10:42.48 | *** join/#asterisk TiToyz (n=TiToyz@aut75-5-82-239-181-57.fbx.proxad.net) |
| 10:57.27 | *** join/#asterisk puzzled (n=foobar@83.163.53.136) |
| 11:10.45 | *** join/#asterisk coppice (n=chatzill@host86-132-13-104.range86-132.btcentralplus.com) |
| 11:18.08 | *** join/#asterisk crazybyte (n=crzp@unaffiliated/crazypenguin/x-000001) |
| 11:28.35 | *** join/#asterisk TSM (n=the_soft@87-194-32-212.bethere.co.uk) |
| 11:31.04 | *** join/#asterisk bn-7bc (n=bjarne-i@mac.lan.noare-1.holmedal.net) |
| 11:38.02 | *** join/#asterisk scardinal (n=supreme@90.184.100.119) |
| 11:54.34 | *** join/#asterisk Bladerunner05 (n=feelme@81-174-56-54.static.ngi.it) |
| 11:58.53 | *** join/#asterisk scardinal (n=supreme@90.184.100.119) |
| 12:12.39 | Bladerunner05 | hello with digium tdm410p and asterisk 1.4.26.2 is better zaptel o dahdi ? |
| 12:18.55 | *** join/#asterisk Da-Geek (n=Da-Geek@nat/redhat/x-uxlcugeamkinyaez) |
| 12:33.29 | *** join/#asterisk DND (n=arabia@213.132.40.2) |
| 12:33.40 | DND | hi guys, how can i enable conference from x-lite? |
| 12:33.50 | DND | what should be the settings in asterisk? |
| 12:36.47 | *** join/#asterisk kazaa_lite (n=msaleem@cpc1-lamb4-0-0-cust590.bmly.cable.ntl.com) |
| 12:38.25 | *** join/#asterisk Buklov (n=buklov@213.138.71.254) |
| 12:48.56 | tzafrir | Bladerunner05, DAHDI. DAHDI is basically the latest version of Zaptel |
| 12:50.03 | coppice | tzafrir: have far have you got with your SIP driver for DAHDI? :-) |
| 12:50.34 | tzafrir | coppice, SIP? why SIP? |
| 12:51.01 | tzafrir | I can install Asterisk (or whatever) on the box and use... |
| 12:51.04 | coppice | tzafrir: your comments on the * mailing list :-) |
| 12:51.05 | tzafrir | H.323 |
| 12:52.40 | tzafrir | I wasn't really serious. It's technically possible, but as the Sipura boxes are so common, I would have expected someone would have already done that if it was easy |
| 12:53.30 | coppice | you made a comment about putting openwrt on them. most of the linksys VoIP boxes don't run linux |
| 12:57.26 | tzafrir | That still does not make it technically impossible. Just more porting work :-) |
| 13:06.55 | niekie | Wooo! |
| 13:07.04 | niekie | has a new VoIP device :D |
| 13:07.49 | *** join/#asterisk Chodorenko (n=chodoren@ext.one.by) |
| 13:08.09 | coppice | tzafrir: I don't think most of those devices have enough RAM to run Linux. Linux seems to be the standard think in new VoIP designs, though |
| 13:24.07 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 13:29.07 | Katty | bad dreams suck :< |
| 13:30.23 | coppice | isn't sucking a sign of a good dream? :-\ |
| 13:32.57 | Katty | ^_- |
| 13:38.53 | EmleyMoor | Is there a way to ring two Zap FXS ports at the same time with the same cadence, but only specify the cadence once? |
| 13:45.51 | *** join/#asterisk flohack (n=fhackenb@chello084115131198.3.graz.surfer.at) |
| 13:47.22 | flohack | Hi! I know it's possible, I just forgot how to do it: Which command will let asterisk wait until the channel is hung up? I use a Dial(..,,G(...)) command, and therefore need to wait for the callers to complete the call. |
| 13:49.18 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 13:51.26 | *** join/#asterisk gardo (n=gardo@110.55.232.121) |
| 14:03.02 | tzafrir | EmleyMoor, Dial(Zap/1r1&Zap/2r2) |
| 14:03.12 | tzafrir | Note that ring cadences are global |
| 14:07.15 | jks | Hi guys! - Using app_queue: is there a simple way to set callerid seperately for each member of a queue? (without having to create a outbound dialing context for each member9 |
| 14:14.22 | EmleyMoor | tzafrir: That isn't posssible without a rewrite of my dialplan |
| 14:15.10 | EmleyMoor | tzafrir: I was hoping to be able to use a single r2 (say) to apply to both - not an r2 each |
| 14:15.51 | tzafrir | EmleyMoor, I bet you can patch chan_dahdi to make that also a per-channel definition (the default ring cadence number) |
| 14:16.26 | EmleyMoor | How about common non-default? |
| 14:16.34 | tzafrir | and no: in that example one phone gets r1 and the other: r2 |
| 14:16.54 | EmleyMoor | Hmmm |
| 14:17.03 | tzafrir | so if you need to later on ring Zap/2 and Zap/3 it gets trickier, I guess |
| 14:17.17 | EmleyMoor | Not enough time to rewrite for it now |
| 14:17.44 | EmleyMoor | David will just have to make do with default on the bedroom phone while I'm away |
| 14:32.51 | *** join/#asterisk flohack (n=fhackenb@chello084115131198.3.graz.surfer.at) |
| 14:37.24 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:37.24 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:38.28 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 14:40.05 | *** join/#asterisk kazaa_lite (n=msaleem@cpc1-lamb4-0-0-cust590.bmly.cable.ntl.com) |
| 14:41.53 | Bladerunner05 | using TDM410p and DAHDI in italy when someone call asterisk and the caller hang up, asterisk leave the line opened.. |
| 14:43.56 | [TK]D-Fender | ~cds |
| 14:43.57 | infobot | [~cds] Call Discconect Supervision is a service placed on analog lines to be able to signal you that that the calling party has hung up. This is typically done either by a momentary battery cut, or by a polarity reversal on the line. |
| 14:44.03 | [TK]D-Fender | ^^ go ask your telco |
| 14:44.15 | [TK]D-Fender | BlaI believe I've been telling you this for over a year now. |
| 14:44.27 | [TK]D-Fender | Bladerunner05: I believe I've been telling you this for over a year now. |
| 14:44.49 | Bladerunner05 | TK]D-Fender: of course, but with dadhi on another box it happen again.. :) |
| 14:45.19 | *** join/#asterisk voipmonk (n=voipmonk@dsl-67-212-10-157.acanac.net) |
| 14:45.21 | [TK]D-Fender | Bladerunner05: If they have issues with this your only other hope is using "callprogress=yes" and having an indications zone to detect an audio pattern signalling a hangup |
| 14:45.45 | [TK]D-Fender | Bladerunner05: Well the answer is the same. It will pretty much always be the same |
| 14:46.57 | Bladerunner05 | TK]D-Fender: I do that |
| 14:47.26 | *** join/#asterisk TSM (n=the_soft@87-194-32-212.bethere.co.uk) |
| 14:50.08 | *** join/#asterisk jbw (n=jbw@87.232.105.162) |
| 14:53.18 | *** join/#asterisk sudhir492 (n=sudhir@adsl-210-52-132.mco.bellsouth.net) |
| 14:53.43 | sudhir492 | Is anyone using Cisco 7940 |
| 14:54.45 | leifmadsen | some people are, I'm sure |
| 14:55.00 | sudhir492 | I need some help with dialplan.xml |
| 14:55.07 | leifmadsen | ~ask |
| 14:55.08 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 14:55.13 | sudhir492 | thats why I asked if I can find someone here |
| 14:56.02 | sudhir492 | Thanks infobot for the tip. I will adhere to that |
| 14:56.52 | *** join/#asterisk voipmonk (n=voipmonk@67.212.10.157) |
| 14:57.05 | sudhir492 | How do I make the phone dial 10 digits automatically, instead of pressing "Dial" softkey |
| 14:58.08 | leifmadsen | sudhir492: what is your current dialplan? |
| 14:58.17 | sudhir492 | My dialplan.xml at present has only 3 lines: |
| 14:58.29 | sudhir492 | <DIALTEMPLATE> |
| 14:58.30 | sudhir492 | <PROTECTED> |
| 14:58.30 | sudhir492 | </DIALTEMPLATE> |
| 14:59.03 | sudhir492 | But it doesn't 10 digits automatically |
| 14:59.09 | leifmadsen | sudhir492: http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79xx |
| 14:59.25 | sudhir492 | there is a long wait, 15 sec or so unless someone presses dial softkey |
| 15:00.36 | sudhir492 | I have read that page many times, but still not have been able to make my phone work properly. |
| 15:01.23 | manxpower | sudhir492: have you tried asking on the mailing lists? |
| 15:03.06 | TSM | sudhir492: your dial plan on the phone is wrong then |
| 15:03.16 | sudhir492 | Actually, I did not specifically ask on the mailing list, but did search through the archive |
| 15:03.38 | manxpower | sudhir492: Searching the archive is almost as good as asking. |
| 15:03.53 | TSM | sudhir492: http://www.loligo.com/asterisk/Cisco/79xx/2003-04-27.examples/dialplan.xml |
| 15:04.07 | sudhir492 | Yes, the phone is not interpreting the dialplan correctly. Or maybe it is not reading that correctly :-( |
| 15:04.58 | manxpower | Cisco phones are not as popular with Asterisk users as they once were. Polycom phones are just as good but come with free firmware. Linksys are much cheaper and come with free firmware. |
| 15:05.03 | sudhir492 | TSM, just before I logged in to freenode, I was on that page |
| 15:05.46 | TSM | sudhir492: start from scratch then with a default dialplan, see if that sorts the problem, then start putting your rules back in |
| 15:05.55 | sudhir492 | manxpower, you are quite right about Polycom phones. I used them most of the time. But this new client of mine had about 25 Cisco phones, and I had to configure them |
| 15:06.35 | sudhir492 | Everything works fine now, except the dialpan |
| 15:07.33 | TSM | wipe the dialplan and start again |
| 15:08.23 | sudhir492 | Let me try that. |
| 15:08.26 | manxpower | sudhir492: Cisco phones also tend to be poorly documented. |
| 15:08.49 | sudhir492 | Who knows, I might have put some binary char in the file. |
| 15:10.09 | TSM | manxpower: i agree that polycoms are good, the documentation is quite fully featured, i still think that some bits could be tweeked to make it far better, mainly in the area of the efk's |
| 15:10.16 | *** join/#asterisk LemensTS (n=customgt@71.86.32.146) |
| 15:10.21 | sudhir492 | manxpower, not just that, they are such pain in the !@# that they dont give access to any information, unless you sign up for their support contract |
| 15:11.02 | manxpower | TSM: Oh, there are some things that could be much better documented on Polycoms (custom key mapping, etc) |
| 15:11.06 | sudhir492 | Which too is not easy. I called them to pay for support contract, but they told me to find some vendors etc |
| 15:12.58 | TSM | manxpower: i would like especialy to have the ability to have softkeys and efk to show up depending on what number calls, ie in a conference, or door phone etc..., hopfully one day |
| 15:13.17 | TSM | manxpower: also better support of some of their normal keys in asterisk |
| 15:13.36 | LemensTS | my sip phone will not make a call out thru the server out to the itsp...i see a problem at line 160, but im not sure what it means http://pastebin.com/m36aedf7d - my sip phone: 305...my ip 71.86.32.146....sip server ip 67.231.241.218....itsp ip 174.37.177.66.... |
| 15:15.13 | manxpower | not found means exactly that. number not found on the destination server. |
| 15:15.56 | manxpower | You are asking the destination server to dial 0015733780904 is that what you expect? |
| 15:16.13 | sudhir492 | TSM, I started with a new dialplan.xml |
| 15:16.22 | sudhir492 | still no luch |
| 15:16.24 | sudhir492 | luck |
| 15:17.03 | manxpower | sudhir492: have you confirmed the phone actually loads the config file off the server? |
| 15:17.07 | sudhir492 | TSM, do you have a default dialplan.xml for cisco 79XX |
| 15:18.17 | LemensTS | manxpower: yes i want to send to the ITSP 0015733780904 |
| 15:18.36 | LemensTS | so that means its unroutable on the ITSp server? |
| 15:19.06 | manxpower | LemensTS: It could mean it's not a valid number on the ITSP dialplan. It could also mean that your account does not have permission to dial that number. |
| 15:19.32 | LemensTS | manxpower: ok thanks, that was what i was thinking been fighting with this itsp just wanted to check |
| 15:19.39 | [TK]D-Fender | LemensTS: It means they don't like that number. Make sure its a legal FORMAT for them |
| 15:19.40 | *** join/#asterisk Tim_Toady (n=moi@adsl72-43.kln.forthnet.gr) |
| 15:19.44 | manxpower | LemensTS: where is your ITSP located, where is the number you are dialing located? |
| 15:20.55 | LemensTS | itsp is europe i think. number is usa |
| 15:21.05 | *** join/#asterisk Cyon (n=cyon@fxp1.dmz1.cro1.bestweb.net) |
| 15:21.06 | manxpower | you think? |
| 15:21.32 | LemensTS | Yea i didnt create the account, they just gave me an IP for sip.conf |
| 15:21.39 | [TK]D-Fender | LemensTS: "0015733780904" <- sure doesn't look like a USA number. Go check how they expect you to dial international calls. |
| 15:21.43 | manxpower | If it is in europe, then the international dial prefix might be 00, if the ITSP is in USA then you don't need the 00, you would just need the 1 before the area code. |
| 15:21.59 | LemensTS | yea ive tried it with 00 and without |
| 15:22.03 | manxpower | LemensTS: it's time to contact your ITSP. |
| 15:22.24 | [TK]D-Fender | LemensTS: and I doubt "00" = USA. I think you're missing a COUNTRY PREFIX there... |
| 15:22.28 | sudhir492 | manxpower, the phone doesnot seem to be reading dialplan.xml |
| 15:22.39 | manxpower | [TK]D-Fender: it would be 001 00 + country code of 1 |
| 15:23.20 | sudhir492 | in SIPDefault.cnf, I have specified : dial_template: "dialplan" |
| 15:23.38 | manxpower | sudhir492: do you actually see the phone download the files? |
| 15:23.47 | LemensTS | yea ive called from other servers in europe doing 001 |
| 15:23.49 | manxpower | (like in the FTP or TFTP logs) |
| 15:24.26 | [TK]D-Fender | LemensTS: then clearly 001 isn't for USA <- |
| 15:24.27 | sudhir492 | I see that it downloaded SIPDefault.cnf. (I had verified that earlier too) and other things in that files do take effect. |
| 15:24.38 | [TK]D-Fender | LemensTS: So go ask them for the proper format |
| 15:24.50 | sudhir492 | However, I did not see any request for dialplan* file coming to the tftpserver |
| 15:25.07 | sudhir492 | So, something may be wrong with my SIPDefault.cnf. |
| 15:26.22 | *** join/#asterisk afterbyte (n=tribal@190.8.40.59) |
| 15:28.49 | TSM | sudhir492: http://www.loligo.com/asterisk/Cisco/79xx/current/dialplan.xml |
| 15:29.43 | TSM | sudhir492: http://www.loligo.com/asterisk/Cisco/79xx/current/SIPDefault.cnf |
| 15:31.37 | Kobaz | how would i get text messages on a non-cell number |
| 15:31.56 | manxpower | Kobaz: are you in europe? |
| 15:32.32 | Kobaz | nope |
| 15:33.00 | manxpower | Kobaz: Asterisk's SMS support expects standard PSTN SMS, which is NOT available in the USA/Canada |
| 15:33.05 | Kobaz | ah |
| 15:33.08 | TSM | in the UK we have that, just send a txt to a landline number, a computer reads it out |
| 15:33.10 | Kobaz | those punks |
| 15:33.14 | Kobaz | oh hmm |
| 15:33.24 | Kobaz | TSM: i've had that before... gotten a text to speech on my land line |
| 15:33.27 | manxpower | TSM: *nod* That is not available in the USA. |
| 15:34.16 | manxpower | You can purchase a PSTN SMS compatible phone. But Asterisk would have to dial that phone direct to send the SMS. |
| 15:34.52 | Kobaz | yeah i'm just talking about recieving |
| 15:34.58 | Kobaz | text to speech is fine |
| 15:35.16 | TSM | TTS is proly the only realy way to do it then |
| 15:35.37 | manxpower | Kobaz: then your question becomes "How to I make an outbound call and have Asterisk speak text". |
| 15:35.57 | Kobaz | no |
| 15:35.57 | Kobaz | heh |
| 15:36.25 | Kobaz | it becomes, what combination of itsp provider and source sender do i need... to recieve text messages from the outside world |
| 15:36.39 | manxpower | Good luck with that. |
| 15:36.42 | Kobaz | hehe |
| 15:36.49 | Kobaz | does some experiments |
| 15:37.52 | *** part/#asterisk ming_zym (n=ming_zym@218.108.49.4) |
| 15:53.12 | sudhir492 | TSM, thanks for the link. I cannot understand, why it is not loading dialplan.xml |
| 15:53.54 | sudhir492 | I compared SIPDefault.cnf and dialplan.xml from John Todd's page. They are almost the same |
| 15:55.06 | *** join/#asterisk Da-Geek (n=Da-Geek@nat/redhat/x-lyribqznawtoivhv) |
| 15:56.29 | sudhir492 | Actually, I remember now, that I had already copied his SIPDefault.cnf to experiment |
| 15:58.48 | Kobaz | dialplan in xml... ack |
| 16:01.36 | [TK]D-Fender | Kobaz: Go try FreeSWITCH :) |
| 16:08.39 | Kobaz | heh |
| 16:08.56 | TSM | freeswitch has xml comfig i think |
| 16:09.17 | Kobaz | the problem is... if we did go with freeswitch, it would be a year+ of coding that would be dumped |
| 16:13.22 | *** join/#asterisk Athahira (n=chatzill@81.10.122.185) |
| 16:13.35 | Kobaz | i would like to check it out at some point |
| 16:21.21 | Bladerunner05 | TK]D-Fender: I put callprogress=yes in chan_dadi.conf in /etc/asterisk but the box when called hangs only for timeout |
| 16:21.45 | Bladerunner05 | any italian that have problem "hangup" with tdm410p + asterisk 1.4.22 + dahdi ? |
| 16:22.13 | Bladerunner05 | c'รจ qualche italiano qui ? |
| 16:22.15 | p3nguin_ | SIPDefault.cnf and dialplan.xml are Cisco phone files, not VoIP switch config files. |
| 16:24.40 | p3nguin_ | kobaz: It's not that bad. http://paste-bin.com/view/fc0e8caa |
| 16:45.47 | geneticx | Hello, I have a 7960 IP phone that everytime I start to dial 1 it dials automatically without waiting for me to finish the number.. how or where would I change this behavior? |
| 16:47.00 | jblack | ip phones usually have an internal dialplan. |
| 16:47.39 | p3nguin_ | geneticx: You might need to adjust your dialplan.xml to change the behavior. |
| 16:47.44 | geneticx | jblack: yeah some sort of digimap |
| 16:47.45 | jblack | find the phone's configuration, hit google if you need, and go from there. |
| 16:48.01 | jblack | morning p3nguin_ |
| 16:48.13 | p3nguin_ | Check the dial_template option in SIPDefault.cnf. |
| 16:48.16 | p3nguin_ | jblack: hi |
| 16:48.44 | p3nguin_ | If it says dialplan, then check the tftp directory for dialplan.xml. That's the most common name for it. |
| 16:49.28 | *** join/#asterisk cyberfab007 (n=cyberfab@CPE001b11cf4f69-CM0014f85c3ada.cpe.net.cable.rogers.com) |
| 16:49.40 | geneticx | p3nguin_: Oh! damn yeah I see it now. Thanks man |
| 16:50.23 | *** join/#asterisk Chodorenko (n=chodoren@ext.one.by) |
| 16:52.17 | jblack | Awesome. tor on android |
| 16:52.40 | jblack | Now I can look up my bomb making recipes on my cell phone without getting caught. |
| 16:53.30 | p3nguin_ | geneticx: Do you know of any services URL that is worth a crap for the Cisco phones? |
| 16:53.31 | drmessano | Tor... Great for the end user, horrible for Sysops |
| 16:53.56 | jblack | Yeah. Trying to keep tor inside a firewall is damn near impossible. |
| 16:54.11 | drmessano | No, I mean, for those running the exit nodes |
| 16:54.38 | jblack | It's definitely a headache to be taken with open eyes. |
| 16:55.04 | drmessano | Why the hell would I want to full out that sort of activity.. heh |
| 16:55.08 | drmessano | funnel |
| 16:55.10 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 16:55.16 | drmessano | word / thought disconnect |
| 16:55.31 | *** join/#asterisk _bugz_ (n=bugz@99.129.215.181) |
| 16:55.36 | p3nguin_ | I/O error |
| 16:56.07 | jblack | You might not. I have an exit node. |
| 16:56.35 | drmessano | What happens when the FBI comes knockin? |
| 16:56.43 | jblack | for running an exit node? |
| 16:56.45 | geneticx | p3nguin_: not really..what have you been using so far? |
| 16:57.07 | drmessano | When requests for kiddie porn are made from your internet connection |
| 16:57.10 | jblack | I tell 'em I run tor, and they go away. |
| 16:57.25 | drmessano | "Whats a Tor, perv?" |
| 16:57.37 | geneticx | lol |
| 16:57.50 | jblack | Can you show me a pile of people sitting in jail for running tor exit nodes? |
| 16:58.03 | jblack | For that matter, can you show me one? |
| 16:58.51 | p3nguin_ | geneticx: services_url: "http://phone-xml.berbee.com/menu.xml" ;but it sucks. |
| 16:58.55 | *** join/#asterisk sircolin (n=sircolin@my83-216-68-241.mynow.co.uk) |
| 16:59.02 | jblack | I'll take the risk of being the first person to be jailed for runing an exit node, because I don't think it's illegal to do so |
| 16:59.21 | drmessano | I'm not even sure it's been tested yet.. I think the lack of publicized cases isn't an indication that you can blow off the feds because "I just run this exit node, I am not the perv", but simply because it's not been an issue as of yet |
| 16:59.22 | jblack | And I'll run an exit node so that tor is where when I need it. =) |
| 17:00.02 | jblack | There's been similiar cases with plausible deniability and open wireless access points. |
| 17:00.22 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 17:00.24 | drmessano | Ok |
| 17:00.27 | jblack | If anything, it ironically makes you safer, becuase they can no longer prove that you're the only person using the account. |
| 17:00.45 | Katty | yawns a bit, stares blankly |
| 17:00.55 | jblack | mornin |
| 17:00.59 | Katty | morning |
| 17:01.00 | drmessano | If there's precedent about shared connections that you can viably use in court, I suppose that's different |
| 17:01.20 | jblack | Yeah, there's case law on that sort of thing. |
| 17:01.42 | jblack | btw, you know who invented tor? And I shit you not. |
| 17:01.44 | jblack | The CIA. |
| 17:02.03 | drmessano | I would need more than the EFF telling the court "NUH UH" and richard stallman showing up in tights to make me feel better about it |
| 17:02.14 | drmessano | I think I heard that somewhere |
| 17:02.20 | Katty | i uhh. |
| 17:02.22 | Katty | hmm. |
| 17:02.28 | Katty | maybe i don't want to know what's going on. |
| 17:02.39 | jblack | katty: Oh, nothin. |
| 17:03.34 | jblack | drmessano: I can understand that position |
| 17:03.58 | jblack | There's also cases that an open wap isn't a legal defense. |
| 17:04.14 | jblack | That case was in texas. |
| 17:04.48 | jblack | btw, http://www.torproject.org/eff/tor-legal-faq.html.en |
| 17:04.50 | drmessano | Ah |
| 17:05.09 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 17:05.09 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 17:05.44 | geneticx | p3nguin_: yeah I haven't seen many services url's other than this one. Although I have seen people using their own .pl scripts to acommodate their needs, but I'm not an expert on it |
| 17:06.09 | p3nguin_ | I'm not a programmer, or I probably would have written my own services. :/ |
| 17:06.18 | geneticx | yup |
| 17:06.49 | p3nguin_ | AND I would have shared the URL for other people who aren't programmers. |
| 17:08.33 | *** join/#asterisk grharry (n=root@ppp-94-67-126-205.home.otenet.gr) |
| 17:08.35 | geneticx | well if I come across one I will sure let everyone aware on this channel of it's existance. |
| 17:10.49 | Katty | holy snickerdoodles. in case anyone was curious, a single large sweet potato contains 692% of your days worth of Vitamin A |
| 17:10.49 | grharry | I am running asterisk 1.6.1.6 and trying to install codec_g72[39] from asterisk.hosting.lv however when I restart asterisk it complaints about format_au.so missing -- which I am not able to locate in the source tree make options ..... do I miss some lib ??? |
| 17:18.31 | *** join/#asterisk Athahira (n=chatzill@81.10.122.185) |
| 17:18.56 | Katty | http://www.recipezaar.com/Baked-Salmon-28199 <- lunch |
| 17:19.35 | ChannelZ | hmm I've never IM'd on Facebook until now.. everyone the person sent me a message, it just makes this annoying pop sound like unplugging your speakers. |
| 17:21.44 | sircolin | use pidgin's Facebook plugn then you don't even need to go there |
| 17:22.00 | Katty | trillian also has an addon for facebook chat. |
| 17:22.05 | Katty | as does diggsby |
| 17:22.17 | ChannelZ | hmm interesting |
| 17:22.17 | drmessano | Pidgin's Facebook plugin rocks |
| 17:22.24 | drmessano | The author is a cool dude |
| 17:22.26 | Katty | goes to start salmon |
| 17:23.32 | *** join/#asterisk ChUbB (n=IceChat7@62-31-213-230.cable.ubr12.aztw.blueyonder.co.uk) |
| 17:25.20 | *** join/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 17:26.07 | *** part/#asterisk grharry (n=root@ppp-94-67-126-205.home.otenet.gr) |
| 17:27.16 | geneticx | p3nguin_: I assume you have checked these others listed here right? http://www.voip-info.org/wiki/view/Asterisk+Cisco+79XX+XML+Services |
| 17:27.33 | *** part/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 17:27.44 | p3nguin_ | Yeah, most of them are broken or aren't really any good. |
| 17:28.28 | geneticx | uff, ok. |
| 17:32.42 | *** join/#asterisk manxpower (n=ewieling@24.42.221.26) |
| 17:33.15 | *** join/#asterisk nix8n82 (n=nate@63.162.27.14) |
| 17:34.24 | ChannelZ | drmessano: hmm neat, thanks for the pidgin/fb pointer, seems to work |
| 17:34.39 | ChannelZ | err sircolin rather, you mentioned it first sorry! |
| 17:35.03 | sircolin | np enjoy |
| 17:35.09 | drmessano | I mentioned it better |
| 17:35.13 | sircolin | lol |
| 17:35.50 | ChannelZ | so what happens if I'm on FB and someone yells at me? Does it pop up in both places? |
| 17:36.02 | sircolin | mines speaks irc too in a nice steven hawkins voice |
| 17:36.05 | sircolin | yup |
| 17:36.24 | ChannelZ | ok |
| 17:36.45 | ChannelZ | heh - I did my voicemail greet as stephen hawkings |
| 17:37.48 | *** join/#asterisk voipmonk (n=voipmonk@dsl-67-212-10-157.acanac.net) |
| 17:37.52 | sircolin | there's a pigin plugin that allows festival to speak what ever is in the pidgin window |
| 17:38.28 | sircolin | you should try it I lets you listern to many chats at the same time |
| 17:38.31 | p3nguin_ | Stephen Hawking is a funny guy. |
| 17:38.38 | Chodorenko | Howe to security asterisk rtp stream ? |
| 17:39.20 | manxpower | Chinorro: VPN |
| 17:40.30 | Chodorenko | voip phone -> asterisk, phone is not support vpn |
| 17:40.42 | Kobaz | put a vpn in beteen |
| 17:40.51 | Kobaz | and make sure you don't use compression... or you |
| 17:40.59 | Kobaz | 'll get long dropouts |
| 17:41.24 | manxpower | Chinorro: It would be almost impossible today to make the entire stream encrypted. |
| 17:41.25 | jblack | I don't think that's true. |
| 17:41.30 | russellb | i heard that some snom phones are including VPN support now |
| 17:41.44 | manxpower | There is SRTP, but it's not well supported, you would also have to make sure your ITSP supported SRTP |
| 17:41.52 | jblack | I mean, it could be true, if either end is cpu starved, but I've done voip on compressed vpn links with little problem. |
| 17:41.52 | manxpower | And your phone and Asterisk. |
| 17:41.53 | Kobaz | jblack: don't think what's true? |
| 17:42.13 | manxpower | russellb: What variety of "VPN" support does SNOM have? |
| 17:42.28 | russellb | i think it was openvpn |
| 17:42.30 | Kobaz | jblack: if you have *any* packet loss, when you drop one vpn packet, it will drop several voip packets |
| 17:42.39 | Chodorenko | my phone is suport srtp |
| 17:42.49 | manxpower | Chodorenko: does your ITSP? |
| 17:42.57 | russellb | Chodorenko: Asterisk doesn't (yet) |
| 17:43.00 | manxpower | russellb: does Asterisk support SRTP yet? |
| 17:43.15 | jblack | kobaz: Um... how well do you think a tunnel really manages to compress ulaw/gsm/etc ? |
| 17:43.20 | manxpower | Does any RTP proxy support SRTP? |
| 17:43.30 | Kobaz | jblack: depends on the type of compression |
| 17:43.40 | manxpower | Header compression is the most useful. |
| 17:43.50 | Chodorenko | @russellb: in tls asterisk support SRTP , why not support without ? |
| 17:43.57 | jaytee | I bet HappyClown does but it's still in beta I think |
| 17:44.03 | jblack | Yeah, the headers get crunched up well. But voice data doesn't compress well on tunnels, for obvious reasons. :) |
| 17:44.10 | russellb | Chodorenko: I'm not sure I understand that question. |
| 17:44.17 | Kobaz | but doing voip on a compressed vpn over the public internet, is just asking for a crappy connection |
| 17:44.25 | drmessano | ~HappyClownPBX |
| 17:44.26 | infobot | [HappyClownPBX] is currently in closed beta, approaching 12GB in size, uses Asterisk for it's core, it pwns, is also now compatible with the Diahatsumashiniriki Keyotason 200LP-A11 SIP phone |
| 17:44.31 | drmessano | Yep, beta |
| 17:44.34 | Katty | returns |
| 17:44.34 | Kobaz | over lan, or private lines, that's another story |
| 17:44.49 | jblack | Kobaz: Dude, you're full of it. Time to back off quietly. :) |
| 17:44.56 | Kobaz | jblack: try it someday |
| 17:45.19 | Kobaz | jblack: packet loss is painfully obvious when you have compression |
| 17:45.19 | Chodorenko | @russellb: by manual in TLS tunel SIP TCP asterisk encode RTP as SRTP , without TLS srtp not present. Why? |
| 17:45.37 | russellb | I still don't understand |
| 17:45.45 | russellb | Asterisk supports SIP TLS, but not SRTP |
| 17:45.53 | manxpower | Chodorenko: TLS and SRTP are NOT the same thing. |
| 17:45.55 | russellb | why, because it's not done yet |
| 17:46.05 | russellb | when will it be done? when it's done |
| 17:46.07 | russellb | :-) |
| 17:46.07 | jblack | kobz: I've done it for half a dozen clients. I know for a _fact_ you're full of shit. :) |
| 17:46.09 | Chodorenko | thx |
| 17:46.18 | Kobaz | jblack: you have excellent connectivity then |
| 17:46.25 | Chodorenko | @russellb: thx |
| 17:46.29 | russellb | you're welcome. |
| 17:46.34 | Katty | hmm |
| 17:46.38 | Kobaz | jblack: i've noticed the exact opposite... so... i know for a *fact* you are also full of shit |
| 17:46.42 | Katty | why don't we drop the topic and find something new to talk about |
| 17:46.46 | Chodorenko | @russellb: SRTP not planning in future ? |
| 17:46.48 | Kobaz | jblack: so, lets agree that we're both full of shit... :P |
| 17:46.55 | jblack | plonk |
| 17:46.58 | manxpower | jblack: People think packet loss is much higher in the internet than it really is. It also depends on the type of VPN. |
| 17:47.05 | Katty | because telling each other you're full of shit is just going to make your afternoon grumpy |
| 17:47.11 | Kobaz | heh |
| 17:47.23 | russellb | SRTP support is close, but nobody is actively working on it right now. |
| 17:47.33 | Kobaz | you would be surprised how many udp packets are dropped |
| 17:47.33 | jblack | manxpower: Yah. Just wish they'd clear the old-dialup paradigm out of their heads. |
| 17:47.33 | russellb | it'll get finished up when it makes it high enough on the priority list. |
| 17:48.03 | Chodorenko | i use Vpn in most my clients , compression and mppe not have problem in sound |
| 17:49.04 | jblack | katty: It's over. I ignored him way back at "plonk" |
| 17:49.22 | Kobaz | heh |
| 17:49.29 | Kobaz | like it said, it just depends(tm) |
| 17:49.40 | Kobaz | s/it/i |
| 17:49.41 | Katty | lol. |
| 17:49.48 | manxpower | jblack: I imagine most TCP based VPNs would suck for voice. |
| 17:49.48 | Katty | let's find something better to argue about. |
| 17:50.06 | Chodorenko | @russellb: how to raise the priority of this problem? and where you can see the priorities |
| 17:50.33 | jblack | manxpower: I would have thought so, but in my experience, not so much. It all comes down to that latency |
| 17:50.38 | manxpower | Chodorenko: you have 3 options. 1) Write it yourself 2) pay someone to write it 3) use a different PBX |
| 17:50.41 | russellb | well, it's there for anyone to work on (as an open source project) |
| 17:50.47 | russellb | When I speak of priorities, I mean Digium priorities. |
| 17:51.30 | russellb | By asking about it, that does affect it. |
| 17:51.33 | manxpower | jblack: *nod* I don't have to deal with that sort of thing very much. We have 50ms latency, no jitter over our VoIP network. |
| 17:51.38 | russellb | If I hear about it enough, it'll get bumped up. |
| 17:52.03 | russellb | Keep in mind, though, that we literally have 500 to 1000 identified items that we could be working on :-) |
| 17:52.04 | jblack | manxpower: Yeah. a tcp tunnel over _consumer_ grade dsl , and you're usually looking at 80-90 ms. |
| 17:52.38 | Kobaz | jblack: depends how far away the source and destination are... and how well managed the network is |
| 17:52.46 | jblack | SOHO grade is usually the same, sometimes slightly better. |
| 17:53.26 | drmessano | app_tor |
| 17:53.27 | Chodorenko | @russellb: digium likely to understand that that the problem of traffic safety is very important at the moment is not difficult to intercept the audio path from kasteriska to the client, so it's a real problem |
| 17:53.34 | manxpower | Come to think of it, I currently run my voice over a VPN! |
| 17:53.47 | Kobaz | jblack: i have a vpn from new york to california, the vpn only adds 2-5ms |
| 17:54.28 | drmessano | SNOM phones have OpenVPN built-in |
| 17:54.36 | drmessano | Just sayin |
| 17:54.41 | jblack | drmessano: Heh. That could be some day. :) |
| 17:55.38 | Chodorenko | standard wireshark have function decode rtp trafic in two direction |
| 17:55.44 | Kobaz | if all of your customer vpns are 80ms location agnostic, you have some crappy networks.. heh |
| 17:56.12 | Kobaz | ...or a terrible vpn |
| 17:56.32 | manxpower | Chodorenko: and a plain telephone to listen on normal PSTN calls. |
| 17:57.05 | manxpower | People seem to think the PSTN is "secure" for some reason. |
| 17:57.29 | Chodorenko | manxpower: you say not true path , PSTN is not secure |
| 17:58.53 | manxpower | In my experience nobody cares about securing the RTP stream. |
| 17:58.57 | *** join/#asterisk bluOxigen (n=asad@static-host119-73-66-158.link.net.pk) |
| 17:59.43 | manxpower | If you are doing something illegal or involved in political protests then security is very important. |
| 17:59.51 | Chodorenko | manxpower: wiretapping ordinary phones was not a problem even during the president nickson :) |
| 17:59.56 | manxpower | For everyone else I think it's a great idea, but not practical yet. |
| 17:59.58 | sircolin | I don't think anybody expects a secure call from pstn |
| 18:00.26 | manxpower | sircolin: then why do they care so much about securing their VoIP calls? |
| 18:00.47 | drmessano | manxpower: HACKERS |
| 18:00.48 | jaytee | President Nickson? Was he the one between Truman and Eisenhower or was he the one between Reagan and Ross Perot? |
| 18:01.06 | manxpower | jaytee: je' |
| 18:01.17 | drmessano | EVEL HAXORS lissenin to mah calls with teh linux |
| 18:01.25 | manxpower | he's the one that put the USA on it's current, totally unsustainable, policy on drugs. |
| 18:01.28 | drmessano | leenux laptops |
| 18:02.10 | Chodorenko | *in time president Nickson:) |
| 18:02.16 | jaytee | Nixon? or J. Edgar? |
| 18:02.26 | nix8n82 | that is funny we are losing the war on drugs |
| 18:02.35 | drmessano | It really is harder/easier than you think to listen to a VoIP call.. You don't just hold a bucket under the NID outside |
| 18:02.42 | manxpower | nix8n82: Exactly my point. |
| 18:02.55 | drmessano | So I dont completely get the paranoia |
| 18:04.33 | flohack | Hi! I know it's possible, I just forgot how to do it: Which command will let asterisk wait until the channel is hung up? I use a Dial(..,,G(...)) command, and therefore need to wait for the callers to complete the call. |
| 18:04.35 | Chodorenko | IMHO security voice information must be a high priority |
| 18:04.41 | p3nguin_ | jblack: How come I have 136ms latency from my phone to * with about 30 feet of cable and one switch in the mix, but only have 33ms to a phone on the other side of the internet, which also runs through that same switch? |
| 18:05.15 | manxpower | p3nguin_: would that be latency shown in Asterisk "sip show peers" or is that actual real packet latency as measured by something like ping. |
| 18:05.25 | Kobaz | p3nguin_: i haven't looked at the code for checking latency in asterisk, but from what I can tell... asterisk ip latency is a damn lie |
| 18:05.34 | Chodorenko | flohack: cli> soft hangup ? |
| 18:05.53 | p3nguin_ | manxpower: I was reading the qualify times. |
| 18:05.55 | manxpower | Kobaz: it's "latency in responding to a SIP OPTIONS packet", it's not ICMP (ping) latency. |
| 18:05.56 | Kobaz | flohack: what are you trying to do |
| 18:06.03 | nix8n82 | I still wonder if the war on drugs is just a way to make money by brutal means, and an effort to arm brutal people, in turn to kill less desirable people/or population control over areas with rich resources |
| 18:06.15 | Kobaz | manxpower: ah, good to knowe |
| 18:06.15 | manxpower | p3nguin_: qualify times are not network latency, they are more SIP latency |
| 18:06.41 | p3nguin_ | manxpower: Okay, how can I reduce that? |
| 18:06.49 | manxpower | nix8n82: I think people that profit from drugs use much of that money to ensure that drugs stay illegal |
| 18:07.01 | manxpower | p3nguin_: Use a different phone. |
| 18:07.27 | p3nguin_ | Maybe a firmware upgrade could change it? |
| 18:07.31 | manxpower | If most drugs were legalized then the money stream will try up. |
| 18:07.48 | manxpower | p3nguin_: it's the software/hardware of the phone latency. |
| 18:07.50 | Kobaz | p3nguin_: doubtful, but look at the changelog for the firmware updates |
| 18:08.06 | Kobaz | p3nguin_: i wouldn't be concerned with latency as reported in asterisk show peers |
| 18:10.35 | p3nguin_ | I have a 7912 that's got a 20ms qualify time, but a 7940 with over 130ms. Ping times are 66.5 ms and 0.940 ms, respectively. |
| 18:10.48 | flohack | Kobaz: I'm trying to write a 'click to dial' script. It should first call me, let me press one to be connected, two to reject. It should then call the user who pressend the button and play him a message if I rejected as soon as he picks up. If I accepted it should bridge the channels. |
| 18:11.07 | nix8n82 | I don't think the money would dry up, but lets say the put a $5 cap/metric ton of any time of drug it would leave vast amounts of money for..I don't know health care, scientific research,education and feeding those that really need it.. |
| 18:11.51 | nix8n82 | then they could have a war on assholes |
| 18:12.10 | Kobaz | flohack: that's backwards |
| 18:12.29 | Kobaz | flohack: it should call the user who requested the call first.. once then pick up... call the other person |
| 18:12.39 | Kobaz | flohack: which is just a simple originate command via AMI |
| 18:13.13 | flohack | Kobaz: True, that would make it easier |
| 18:13.45 | flohack | Kobaz: I'll try that and see if I run into any problems, thanks! |
| 18:14.08 | flohack | Chodorenko: Thank you too for your reply! |
| 18:14.44 | Chodorenko | flohack: :) no thx , i glade help you |
| 18:16.42 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:17.44 | *** join/#asterisk JonathanB (n=Adium@cpe-24-242-201-118.tx.res.rr.com) |
| 18:19.56 | Katty | wishes salmon would bake faster |
| 18:20.40 | *** join/#asterisk jicksta (n=jicksta@c-98-210-108-28.hsd1.ca.comcast.net) |
| 18:27.12 | Kobaz | my first asterisk module. yay |
| 18:44.47 | Katty | :> |
| 18:44.49 | Katty | has lunch! |
| 18:45.39 | Katty | dang. they weren't kiddin. |
| 18:45.41 | Katty | it IS good. |
| 18:46.00 | Katty | http://www.recipezaar.com/Baked-Salmon-28199 |
| 18:46.10 | *** part/#asterisk EmleyMoor (i=phil@topdeck.tinsleyviaduct.com) |
| 18:46.23 | Katty | i put the salmon on a salad containing spinach, celery, and broccoli |
| 18:46.37 | Katty | then used the leftover marinade that the fish was cooking in as a salad dressing |
| 18:46.42 | Katty | and put parmesan on top :> |
| 19:06.00 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 19:07.00 | *** join/#asterisk dlewis (n=dlewis@about/security/staff/dlewis) |
| 19:11.06 | *** join/#asterisk BadHAL (n=nn@24-179-237-8.dhcp.eucl.wi.charter.com) |
| 19:12.00 | Kobaz | mmm. asterisk-dev is dead |
| 19:12.11 | Kobaz | where's the doxygen for all the code |
| 19:12.30 | ChannelZ | bah who needs docs |
| 19:12.56 | Kobaz | heh |
| 19:13.34 | Kobaz | what's the max length of a channel variable? |
| 19:23.01 | *** join/#asterisk dkirker (n=dkirker@99.2.134.30) |
| 19:28.15 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 19:31.31 | *** join/#asterisk LemensTS (n=customgt@71.86.32.146) |
| 19:31.31 | LemensTS | http://pastebin.com/m8c1d19c i cannot get this gotoif statement to go to the right place...its asterisk 1.6 |
| 19:35.03 | ChannelZ | Only put n(less) on the first priority of that sequence |
| 19:35.54 | LemensTS | Yea just now seen that as you typed that thank you. guess i never tried doing more than 1 before. thanks! |
| 19:36.30 | ChannelZ | It's a label jump, not like a separate priority class. |
| 19:39.50 | Chodorenko | @russellb: realization SRTP https://reviewboard.asterisk.org/r/191 |
| 19:41.45 | drmessano | Chodorenko: I think he's well aware of the progress |
| 19:43.42 | Chodorenko | drmessano: strange that not said can be found here, why? |
| 19:47.13 | VooDooNOFX | anyone know of a free public domain music site? |
| 19:47.23 | Kobaz | freesounds |
| 19:47.31 | drmessano | It hasn't been touched in months, it's not a priority.. He told you that much |
| 19:47.52 | Kobaz | http://www.freesound.org/ |
| 19:47.56 | drmessano | He didnt say there wasn't ANY code written.. |
| 19:48.17 | drmessano | Like a lot of things not yet implemented, its in its infancy, and no resources are working on it |
| 19:49.12 | *** join/#asterisk twanny796 (n=twanny@85.232.220.146) |
| 19:54.28 | Kobaz | sizesql = (newsize / 512 + 1) * 512; |
| 19:54.39 | Kobaz | wow... that's a really strange way to double the size of a number |
| 19:55.46 | Kobaz | well newsize starts at 512... so the first run it doubles |
| 19:56.09 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 19:56.26 | Kobaz | oh i see.. mmm |
| 19:56.36 | Kobaz | anyways... is there a limit to the number of asterisk channels? |
| 19:56.57 | drmessano | no |
| 19:57.16 | Kobaz | k |
| 19:57.25 | VooDooNOFX | only hardware/bandwidth restrictions |
| 19:57.51 | Kobaz | yeah.. it's just bumping up newsize by 512 |
| 20:00.16 | *** join/#asterisk werdan7 (n=w7@freenode/staff/wikimedia.werdan7) |
| 20:01.46 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 20:01.46 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 20:07.11 | *** join/#asterisk manxpower (n=ewieling@24.42.221.26) |
| 20:08.40 | Kobaz | drmessano: is there a limit to the length of a channel variable |
| 20:12.11 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 20:12.11 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 20:17.38 | Kobaz | <PROTECTED> |
| 20:17.42 | Kobaz | [Oct 25 16:14:56] WARNING[22459]: pbx.c:8196 pbx_builtin_setvar_multiple: MSet: ignoring entry 'SIP/bob-b7947920' with no '=' (in 2799@trunkhandler_branch-in:3 |
| 20:17.58 | Kobaz | how do i not trigger that warning, when having ',' as a value for a variable |
| 20:22.58 | ChannelZ | do \, |
| 20:23.44 | Kobaz | hmm, if i put it in quotes.. it doesn't complain |
| 20:23.45 | Kobaz | well this is the result of an application call |
| 20:23.56 | Kobaz | ChanList(); Set(channels="${CHANLIST_RESULT}"); |
| 20:24.43 | *** join/#asterisk mchou (n=quassel@unaffiliated/mchou) |
| 20:26.33 | ChannelZ | well Set(barf=foo\,manchu) works |
| 20:28.43 | Kobaz | yeah |
| 20:28.54 | Kobaz | i know about escaping |
| 20:29.11 | Kobaz | mmm, this is so awesome... i <3 my new asterisk module |
| 20:31.09 | svenx | what does it do? |
| 20:31.13 | Kobaz | http://pastebin.ca/1643004 |
| 20:32.10 | Kobaz | right now it just returns all channels |
| 20:32.15 | Kobaz | i'm gonna put regex matching in next |
| 20:32.21 | Kobaz | and then submit the code |
| 20:34.44 | Kobaz | no more asterisk -rx 'core show channels concise', and crap |
| 20:35.06 | *** join/#asterisk bn-7bc (n=bjarne-i@mac.lan.noare-1.holmedal.net) |
| 20:43.30 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 20:44.44 | Katty | leifmadsen: so. |
| 20:44.48 | Katty | leifmadsen: about those photos... |
| 20:44.54 | Katty | leifmadsen: hinthinthint |
| 20:45.17 | ChannelZ | pitchas! dirrrrrty pitchas! |
| 20:45.18 | Katty | leifmadsen: you can't /move/ and not post photos of the new place! |
| 20:45.43 | Chodorenko | drmessano: http://svn.asterisk.org/svn/asterisk/team/group/srtp :) |
| 20:46.43 | TJNII | ChannelZ: That's our website! Nyaaaah! |
| 20:47.40 | leifmadsen | Katty: haha, well the fact it hasn't been built yet will make it difficult to take pictures |
| 20:47.57 | ChannelZ | dirtypitchas.com ? |
| 20:48.14 | *** join/#asterisk geneticx (n=geneticx@adsl-2-215-240.mia.bellsouth.net) |
| 20:49.26 | TJNII | You should see what tweetypie.org looks like. I thought I saw a putty ...... tat. |
| 20:50.23 | ChannelZ | heh |
| 20:51.49 | Katty | leifmadsen: well. now you have something to do. |
| 20:51.52 | Katty | leifmadsen: :P |
| 20:54.08 | leifmadsen | heh |
| 20:58.45 | Katty | well regardless, i'm very happy for you two |
| 20:58.47 | Katty | hugs leifmadsen |
| 20:58.57 | leifmadsen | Katty: thanks! |
| 20:58.58 | Katty | congratulations (= |
| 20:59.06 | leifmadsen | 1700+ sqft will be nice |
| 20:59.17 | Katty | any pets? |
| 20:59.25 | *** join/#asterisk bkruse (n=bkruse@76.73.154.120) |
| 20:59.25 | *** mode/#asterisk [+o bkruse] by ChanServ |
| 21:14.12 | *** join/#asterisk geneticx (n=geneticx@adsl-2-215-240.mia.bellsouth.net) |
| 21:16.45 | [TK]D-Fender | leifmadsen: upgrading? |
| 21:18.02 | *** join/#asterisk `Sauron (n=sauron@dsl001-130-033.aus1.dsl.speakeasy.net) |
| 21:20.59 | *** join/#asterisk icyValk77 (n=icyValk7@host86-146-251-237.range86-146.btcentralplus.com) |
| 21:21.01 | *** join/#asterisk coppice (n=chatzill@host86-132-13-104.range86-132.btcentralplus.com) |
| 21:22.10 | *** join/#asterisk retentiveboy (n=pdugas@74.95.28.37) |
| 21:23.47 | *** join/#asterisk TimToady_ (n=moi@adsl81-194.ath.forthnet.gr) |
| 21:24.49 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 21:29.34 | *** join/#asterisk russellb_ (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 21:29.34 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 21:32.02 | *** join/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
| 21:34.29 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-218-155-146.cablep.bezeqint.net) |
| 21:44.44 | *** part/#asterisk lifeforms (n=walter@clone.lfms.nl) |
| 21:50.18 | *** join/#asterisk KavanS (n=KavanS@173.12.184.89) |
| 21:59.33 | *** join/#asterisk Tim_Toady (n=moi@adsl81-194.ath.forthnet.gr) |
| 22:04.40 | *** join/#asterisk ZenBSDi (n=ZenBSDi@unaffiliated/ZenBSDi) |
| 22:10.15 | *** join/#asterisk niekie (i=quasselc@dreamworld.bergnetworks.com) |
| 22:14.52 | *** join/#asterisk sdaf4 (n=none@adsl-065-005-214-034.sip.jan.bellsouth.net) |
| 22:15.34 | sdaf4 | hello |
| 22:22.59 | *** join/#asterisk jmacz (n=jmacz@190.24.97.225) |
| 22:24.39 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:25.20 | *** join/#asterisk astassistant (n=support@67.101.7.254) |
| 22:30.05 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 22:30.05 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 22:35.28 | *** join/#asterisk kazaa_lite (n=msaleem@94-193-98-124.zone7.bethere.co.uk) |
| 22:36.23 | *** join/#asterisk kazaa_lite_ (n=msaleem@94-193-98-124.zone7.bethere.co.uk) |
| 22:57.27 | *** join/#asterisk Caplain (i=shayne@2001:470:5:fb:851d:bd38:83e:3582) |
| 23:01.51 | jblack | hrmm. my 8 gig micro sdhc card is only 7.4 gigs in size |
| 23:02.25 | ChannelZ | hurray for overhead |
| 23:07.21 | jblack | of course, 7.4gigs is still a lot on a cell phone. |
| 23:09.13 | *** join/#asterisk tzafrir__laptop (n=tzafrir@bzq-218-155-147.cablep.bezeqint.net) |
| 23:09.38 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 23:09.51 | *** join/#asterisk linuxstb (n=linuxstb@rockbox/developer/linuxstb) |
| 23:09.55 | jblack | 16 gig cards are only gonna have 14.9 gigs on 'em, and 32... 29 gigs. |
| 23:11.19 | drmessano | You need a better file system |
| 23:11.35 | jblack | it's not the filesystem. It's the numbering system |
| 23:12.45 | p3nguin_ | Perhaps a combination of those two factors? |
| 23:12.54 | drmessano | No true.. If I use a file system that automatically compresses seldom used files and retrieves them as needed, with an 82% chance they'll open properly, like NTFS, i'm good |
| 23:13.46 | jblack | 8*(1000^3)/(1024^3) = 7.4 |
| 23:14.01 | drmessano | p3nguin_: File system comes into play when you wonder why 6GB of data takes up more than 6GB |
| 23:14.36 | jblack | I'll just cut back on the number of movies I keep on my cellphone. |
| 23:14.36 | p3nguin_ | Combine that with GiB, and you lose lots of space. |
| 23:15.23 | jblack | I wonder if android phones can do ext2 |
| 23:18.29 | Katty | hugs jblack |
| 23:19.01 | jblack | Your bf is gonna beat me up, katty. :) |
| 23:19.46 | jblack | this would be cool, if it takes ext2. Anyone wanna guess? |
| 23:20.30 | jblack | hugs katty back |
| 23:20.34 | Chainsaw | jblack: It's a linux-based system, why wouldn't it take ext2? |
| 23:20.39 | Katty | jblack: no he's not. i hug everyone. |
| 23:20.56 | Chainsaw | jblack: We're not talking about some niche filesystem like Reiser4 here. |
| 23:21.16 | jblack | chainsaw: There's a number of linux devices out there that don't "work with linux". |
| 23:21.21 | jblack | Ironic, I know |
| 23:21.51 | Chainsaw | jblack: That tends to be about interfacing with an outside device, not something as basic as reading an SD card. |
| 23:22.03 | jblack | Looks like it doesn't. |
| 23:22.06 | Chainsaw | jblack: But you can sit here and argue about it or you can stick one in. |
| 23:22.16 | jblack | what do you think I'm trying right now? |
| 23:22.36 | Katty | Kale. |
| 23:22.37 | Chainsaw | I have no idea. I probably don't want to know. |
| 23:22.39 | Katty | with beets. |
| 23:22.49 | jblack | Android phones do not like sd cards formatted in ext2. |
| 23:22.51 | Katty | that actually sounds pretty disgusting. |
| 23:23.23 | jblack | Beets with anything, especially more beets, are disgusting |
| 23:23.40 | Katty | i agree. |
| 23:23.57 | Katty | corn chips, on the other hand, are delightful. |
| 23:24.10 | jblack | unless they have beets on them. |
| 23:24.15 | Katty | yes. |
| 23:24.19 | Katty | so let's not put beets on them. |
| 23:24.25 | Katty | let's put salsa on them instead. |
| 23:24.34 | Katty | salsa and sour cream MIXED together! |
| 23:24.35 | Katty | MmmMMMm! |
| 23:24.44 | eppigy | 8D |
| 23:24.58 | jblack | Hmmm. Beet salsa |
| 23:25.44 | Katty | Eww. |
| 23:25.46 | Katty | no. |
| 23:25.49 | Katty | HIDAVE |
| 23:25.53 | Katty | what did you have for dinner |
| 23:26.42 | *** join/#asterisk ManxPower (n=ewieling@24.42.221.26) |
| 23:26.43 | eppigy | Katty: taco bell |
| 23:26.45 | Katty | :< |
| 23:26.49 | eppigy | :< |
| 23:27.00 | Katty | i thought you were gonna start cooking |
| 23:27.03 | Katty | RIBS |
| 23:27.06 | Katty | and hamburgers |
| 23:27.13 | Katty | and steaks! |
| 23:27.32 | eppigy | well |
| 23:27.34 | jblack | Ohh, know what makes hamburgers better? |
| 23:27.45 | eppigy | i have just been kinda blah |
| 23:27.50 | jblack | You put ketchup on them. And mustard, and onions.. and especially.... |
| 23:27.53 | ManxPower | jblack: when someone cooks them for you |
| 23:27.54 | jblack | BEETS! |
| 23:27.58 | ManxPower | oh |
| 23:28.01 | Katty | NO BEETS |
| 23:28.30 | jblack | now has beets on the mind |
| 23:29.11 | Katty | :< |
| 23:29.23 | Katty | eppigy: busy with work and such? |
| 23:29.35 | jblack | unignore all |
| 23:29.48 | Katty | jblack: sweet potato fries would probably make a hamburger better. |
| 23:30.05 | Katty | jblack: with lots of crispy lettuce on the burger, and onion, and thick slices of tomato :> |
| 23:30.14 | jblack | hmmmmmm |
| 23:30.22 | Katty | and BACON |
| 23:30.26 | *** part/#asterisk linuxstb (n=linuxstb@rockbox/developer/linuxstb) |
| 23:30.32 | jblack | BACON BACON BACON! |
| 23:30.38 | Katty | yes. i do like me some bacon. |
| 23:30.55 | Katty | http://farm4.static.flickr.com/3502/4040490244_e4246c7c85.jpg <- it's BACON |
| 23:30.59 | jblack | Yes. Sometimes I make my bacon better by putting bacon on it |
| 23:31.05 | Katty | mmm. |
| 23:31.07 | ManxPower | I still think they taste better when someone else cooks them. 8-| |
| 23:31.10 | jblack | bacon is the anti-beet |
| 23:31.49 | Katty | i agree. |
| 23:32.15 | jblack | so, android not only doesn't like sdcards in ext2, but it gets _really_ confused when given a card formatted in such a way. Such a shame, really |
| 23:32.23 | Katty | :< |
| 23:32.28 | Katty | that's most unfortunate. |
| 23:32.52 | jblack | It kinda makes sense to me. I think their root filesystem is jffs, so they may not have ext-anything in the kernel at all |
| 23:33.19 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 23:33.37 | *** join/#asterisk Iamnacho (i=Iamnacho@98.186.180.143) |
| 23:47.42 | *** join/#asterisk shadebob (n=chatzill@41.92.10.186) |
| 23:48.19 | shadebob | Hi, I search online predictive dialer. Any idea ? |
| 23:48.35 | *** join/#asterisk moy (n=moy@CPE001cdfec4cee-CM00080dab8485.cpe.net.cable.rogers.com) |
| 23:58.42 | *** join/#asterisk mog (n=mog@c-68-62-169-247.hsd1.al.comcast.net) |
| 23:58.42 | *** mode/#asterisk [+o mog] by ChanServ |
