| 00:00.03 | dennisharrison | russellb, I personally think the best way is to just get some media time |
| 00:00.06 | blitzrage | ~roulette |
| 00:00.06 | jbot | ACTION watches blitzrage pull the trigger: BANG! |
| 00:00.10 | russellb | dennisharrison: gotcha ... well using vicidial is the best option, unless you want to code up a script to do it |
| 00:00.12 | dennisharrison | ... other then the crap that was on cnn today |
| 00:00.13 | blitzrage | DAMNIT! |
| 00:00.17 | fujin | actually, fuck it |
| 00:00.20 | fujin | I've already made $850 USD off it |
| 00:00.22 | fujin | You can have it. |
| 00:00.22 | fujin | http://homepages.maxnet.co.nz/~djfu/dial-threaded.pl.txt |
| 00:00.37 | fujin | make sure you install Parallel::ForkManager, if you don' thave it |
| 00:00.50 | dennisharrison | thanks fujin |
| 00:00.52 | fujin | and modify the relevant stuff |
| 00:00.57 | fujin | username/secret, Data: line |
| 00:01.04 | fujin | and the channel aswell to reflect your sip peer |
| 00:01.08 | fujin | (or IAX) |
| 00:02.20 | fujin | I'm sure anyone with a bit of perl-fu could modify that as necessary |
| 00:02.22 | nhuisman_work | anyone know if it's possible to ugprade the glibc in asterisknow? |
| 00:02.31 | fujin | nhuisman_work: that's probably a bad idea |
| 00:02.32 | nhuisman_work | my fonulator config for redfone won't work with 2.3.x |
| 00:02.36 | nhuisman_work | needs 2.4 |
| 00:03.08 | fujin | Don't use asterisknow? :P |
| 00:03.14 | nhuisman_work | yeah i guess not |
| 00:03.34 | fujin | I'd say you probably could upgrade it, but upgrading glibc generally means upgrading everythign that was built against it. |
| 00:03.43 | russellb | or don't use redfone :) |
| 00:03.49 | fujin | or that. |
| 00:03.57 | nhuisman_work | yeah I already have the hardware |
| 00:04.07 | fujin | ebay? :} |
| 00:04.12 | nhuisman_work | i wonder if I can just run 2.4 in parallel with 2.3.x |
| 00:04.13 | fujin | some other poor sod will have it |
| 00:04.24 | fujin | multiple glibc :\ |
| 00:04.32 | fujin | nhuisman_work: maybe in a chroot |
| 00:04.48 | nhuisman_work | this is only a test anyways, i'm going to use asterisk be but it's not arrived yet. |
| 00:04.48 | nny_1 | [TK]D-Fender: when your around hit me up, got some good news |
| 00:04.59 | nhuisman_work | i just wanted a quick way to get a linux box installed with asterisk on it |
| 00:05.05 | nhuisman_work | know of any other out of the box installs? |
| 00:05.15 | fujin | trixbox, freepbx |
| 00:05.16 | fujin | callweaver |
| 00:05.23 | fujin | they're all shit, though |
| 00:05.24 | fujin | ;> |
| 00:05.27 | nny_1 | anyone have any thoughts on high volume high availability systems? |
| 00:05.36 | fujin | high volume high availablity? |
| 00:05.38 | nny_1 | like 3000 phones |
| 00:05.39 | russellb | i have some thoughts. |
| 00:05.45 | fujin | doable. |
| 00:05.47 | russellb | but that's it ... just thoughts |
| 00:05.48 | fujin | not entirely with Asterisk, though |
| 00:05.55 | nny_1 | lol |
| 00:06.00 | fujin | probably a combination of SER, asterisk (as the backend), and $cluster_software |
| 00:06.05 | fujin | redhat-cluster-suite, perhaps |
| 00:06.07 | nny_1 | figured clustering software |
| 00:06.11 | nny_1 | SER? |
| 00:06.16 | fujin | heartbeat v2, maybe |
| 00:06.23 | fujin | yes, ser |
| 00:06.26 | nny_1 | sip router? |
| 00:06.32 | fujin | kind of. |
| 00:06.47 | fujin | ~ser |
| 00:06.48 | jbot | well, ser is Sip Express Router - see http://www.iptel.org/ser/, or an old secret method of obtaining a havoc of NAT problems, or at #ser |
| 00:06.51 | fujin | ~openser |
| 00:06.51 | jbot | i guess openser is an open source GPL project that aims to develop a robust and scalable SIP server. It is spawned from FhG FOKUS SIP Express Router (SER) and it promotes a development strategy open for contributors and contributions. From project's website http://www.voip-info.org/wiki/view/About+OpenSER |
| 00:07.54 | nny_1 | interesting |
| 00:09.10 | Yourname`` | vicidial sucks |
| 00:10.18 | nny_1 | i think everytime someone says sucks in here, confetti should shoot out of my monitor and sirens should go off :) |
| 00:10.26 | nny_1 | i feel that way about snoms |
| 00:10.28 | nny_1 | ~ snom |
| 00:10.29 | jbot | snom is, like, like all German products. High quality, but wacky engineering. :) |
| 00:10.34 | nny_1 | lol |
| 00:10.35 | nny_1 | nice |
| 00:12.48 | fujin | OH |
| 00:12.49 | fujin | MY |
| 00:12.49 | fujin | GOD |
| 00:12.50 | fujin | http://www.disappearing-car-door.com/ |
| 00:15.06 | nhuisman_work | yeah you want a sip proxy probably |
| 00:16.03 | nhuisman_work | the problem with asterisk is that it makes calls route through the server |
| 00:16.28 | nhuisman_work | i know sipexchange routes internal calls direct phone to phone. |
| 00:16.33 | nhuisman_work | correct if I'm wrong btw. |
| 00:17.21 | Qwell | you're wrong |
| 00:18.30 | russellb | Qwell: now correct him! |
| 00:18.55 | nhuisman_work | Qwell, so asterisk can be setup to use internal calls direct ? |
| 00:18.58 | nhuisman_work | and not through the server? |
| 00:20.47 | nhuisman_work | by the way by pass through the server I mean media streams |
| 00:22.02 | jnc | is it user= or username= ? |
| 00:22.12 | jnc | I knew it is secret= and not password= |
| 00:22.43 | jnc | <PROTECTED> |
| 00:22.51 | jnc | can't figure that one out |
| 00:22.56 | jnc | no username is being sent |
| 00:23.53 | nny_1 | fujin: always wonder what you would have to do if the battery died |
| 00:23.57 | nny_1 | accident etc. |
| 00:24.07 | fujin | manual override |
| 00:24.11 | fujin | learn2readdd |
| 00:24.57 | Qwell | russellb: what's the fun in that? :p |
| 00:25.01 | Qwell | nhuisman_work: yes |
| 00:25.07 | Qwell | and signaling too |
| 00:25.37 | nhuisman_work | yes it can be changed to allow direct point to point media streaming? |
| 00:25.46 | Qwell | yes, among others |
| 00:26.35 | nhuisman_work | well then I must have been reading some pro non-asterisk spam. Oh well, good to know though. |
| 00:28.01 | mikecx | pro non, what a good combo of words. most people would have just said anti, but you went the extra mile |
| 00:28.02 | JT | nhuisman_work: do you really need endpoints to talk direct to each other? |
| 00:28.28 | nhuisman_work | mikecx, yeah :) |
| 00:28.32 | mikecx | :-P |
| 00:28.50 | nhuisman_work | JT well if you have 30k phones in an organization most of the traffic is going to be between them |
| 00:29.02 | nhuisman_work | if it all has to pass through your asterisk servers i assume that is going to be quite a load |
| 00:29.06 | nhuisman_work | vs just point to point. |
| 00:29.15 | JT | nhuisman_work: there will be quite a load in either case |
| 00:29.47 | nhuisman_work | I would assume the bulk of the network load on the boxes would be the media streams |
| 00:30.14 | JT | network load perhaps, but not cpu load |
| 00:30.24 | fujin | providing the codecs are all the same, I wouldn't expect much cpu load |
| 00:30.31 | fujin | transcoding is usually the killer. |
| 00:31.03 | nhuisman_work | i guess if you just dump 10 gige and trunk them then maybe it wouldn't be a problem |
| 00:31.25 | nhuisman_work | you'd probably need a pretty big cluster to run 30-50k phones |
| 00:32.05 | *** join/#asterisk Maliuta (n=nikolai@203.201.152.211) |
| 00:32.27 | Qwell | JT: network load causes CPU load |
| 00:32.34 | JT | one asterisk box could not flood a 10gigE link |
| 00:32.43 | JT | Qwell: not like sip though |
| 00:32.48 | Qwell | rtp |
| 00:33.08 | JT | yes but it will take a lot more work to process sip than rtp |
| 00:33.13 | Qwell | nope |
| 00:33.25 | Qwell | network card interrupts will kill your box with enough calls |
| 00:33.46 | JT | sure but that's at a lower level to asterisk anyway |
| 00:34.01 | nhuisman_work | you would need to have a bunch of cpus then turn on the interupt balancing |
| 00:34.20 | fujin | WSX. |
| 00:34.21 | fujin | ESX |
| 00:34.22 | fujin | rather. |
| 00:34.35 | JT | nhuisman_work: meh, X86 still sucks for that sort of stuff |
| 00:34.49 | nhuisman_work | vs x64? |
| 00:34.50 | fujin | in a SIP/IAX only situation (with no Meetme/zap requirements), I'm sure a vmware esx farm could handle whatever you tossed at it. |
| 00:34.57 | JT | it will get flooded with interrupts |
| 00:35.00 | JT | yes it's still X86 |
| 00:35.12 | Qwell | fujin: nope, the host OS is still interrupting |
| 00:35.18 | fujin | OS(s) |
| 00:35.21 | JT | sure if you had lots of physical machines |
| 00:35.22 | nhuisman_work | yeah but you could still split it up a lot |
| 00:35.26 | mikecx | where is the sla code in the repository? |
| 00:35.30 | JT | or forget about X86s |
| 00:35.33 | Qwell | mikecx: app_meetme, mostly |
| 00:35.35 | JT | and go ASICs ;) |
| 00:35.42 | mikecx | Qwell: thanks |
| 00:36.15 | jnc | I've got a trunk defined, can I force this to register some how? |
| 00:36.19 | jnc | I'm debugging the settings |
| 00:36.20 | Qwell | ~trunk |
| 00:36.20 | jbot | hmm... trunk is is a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. |
| 00:36.43 | jnc | oh, crappy freepbx term |
| 00:36.49 | jnc | what is it, SIP peer? |
| 00:36.59 | fujin | indeed |
| 00:37.18 | jnc | okay so I have this SIP peer ;) can I poke it somehow to make it attempt registration? |
| 00:37.35 | Qwell | jnc: add a register line, see the sample sip.conf |
| 00:37.37 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177585332.dsl.bell.ca) |
| 00:38.04 | jnc | hm, must not be in my sip.conf |
| 00:38.10 | jnc | was this present for asterisk 1.2.x? |
| 00:38.13 | Qwell | yes |
| 00:39.16 | jnc | even though I'm going to a 8fxs/2fxo device on the lan here, I heard that I didn't need a register line |
| 00:39.20 | jnc | I needed peer and user auth |
| 00:39.26 | jnc | then I read that user auth is depreciated |
| 00:39.40 | Qwell | if the device is static, just set host= in the peer |
| 00:40.00 | jnc | okay and that will attempt to authenticate when it is used? or when asterisk reloads |
| 00:40.09 | Qwell | when it's used |
| 00:40.30 | jnc | can I force it to make an attempt (for debugging reasons only) from asterisk console? |
| 00:40.40 | mikecx | hrrm. I don't get why asterisk uses meetme instead of call parking for sla |
| 00:40.57 | Qwell | mikecx: it can't use parking |
| 00:41.02 | Qwell | if a call is parked, it isn't active |
| 00:41.16 | mikecx | Qwell: ahh, guess that makes sense |
| 00:42.29 | *** join/#asterisk mo3nga (n=crudi@rrcs-24-242-163-106.sw.biz.rr.com) |
| 00:42.36 | *** join/#asterisk dexpdx (n=dexpdx@66-162-134-242.static.twtelecom.net) |
| 00:42.41 | mo3nga | hey guys - is there any issues in having multiple SIP calls to AGI? |
| 00:43.05 | dexpdx | anyone got any suggestions for an affortable cordless voip speaker phone |
| 00:43.05 | mo3nga | i keep getting "ast_openstream_full" errors w/ multiple calls |
| 00:50.26 | *** join/#asterisk adjohn (n=adjohn@p5087-ipad408marunouchi.tokyo.ocn.ne.jp) |
| 00:50.51 | *** join/#asterisk GiantPickle (n=GiantPic@S01060016b600537f.gv.shawcable.net) |
| 00:59.26 | mikecx | is there a chance that Progress() could be used to detect a fax? |
| 01:00.46 | *** join/#asterisk Maliuta (n=nikolai@203.201.152.211) |
| 01:08.06 | *** join/#asterisk docelmo (n=vircuser@c-68-32-135-157.hsd1.de.comcast.net) |
| 01:11.24 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-0bcf4782d337097c) |
| 01:18.14 | `Sauron | Is there a web frontend to one of the SQL CDR modules? |
| 01:18.53 | Nugget | I wrote one and then Digium totally broke it. I blame file. |
| 01:19.14 | `Sauron | file? |
| 01:19.29 | Nugget | <file> |
| 01:20.02 | Nugget | /whois file |
| 01:20.06 | Nugget | that file. |
| 01:20.19 | `Sauron | n=file@asterisk/developer-and-muffin-lover/file |
| 01:20.28 | `Sauron | hum |
| 01:20.30 | `Sauron | Aha. |
| 01:20.50 | `Sauron | Bummer. |
| 01:23.57 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 01:23.57 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 01:29.15 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 01:36.44 | nny_1 | ahhh Sauron is in #asterisk... heh guess he's looking for a new ring.... |
| 01:36.46 | nny_1 | :) |
| 01:39.46 | jnc | One ringtone to rule them all |
| 01:39.59 | `Sauron | and getting dropped into the telemarketer jail |
| 01:40.13 | jnc | BritneySpears_HitMyBabyOneMoreTime.mP3 |
| 01:40.17 | `Sauron | nny_1: Save the jokes, they're old. |
| 01:40.22 | nny_1 | lol i bet |
| 01:40.29 | nny_1 | probably almost as old as you are |
| 01:41.13 | Mercestes | Wow, he's as touchy as the real Sauron. Maybe it really is him. |
| 01:41.19 | nny_1 | lol |
| 01:41.54 | nny_1 | he would be the ultimate BOFH |
| 01:41.57 | *** join/#asterisk thinko (i=jdoe6alp@smaug.rackdragon.com) |
| 01:42.19 | Mercestes | LMAO> "What's your username?" |
| 01:42.31 | nny_1 | lol i am having to install the snmpd package just get a tasty init script |
| 01:42.41 | nny_1 | and then remove it and use the source |
| 01:42.44 | nny_1 | mwararar |
| 01:43.01 | nny_1 | tried to use /etc/init.d/skeleton, but it made things explode |
| 01:43.08 | Mercestes | Nice. |
| 01:43.18 | `Sauron | Hehn. |
| 01:45.37 | nny_1 | Mercestes, you have any favorites for sites to put a howto on (voip-info vs cookbook) etc. |
| 01:45.45 | Maliuta | nny_1: no, I _am_ the ultimate BOFH ... Maliuta was the favourite torturer of Ivan IV (aka Ivan the Terrible), the name is synonymous with "absolute complete evil" in eastern slavic languages :) |
| 01:45.56 | nny_1 | Maliuta: woah.. niiiice |
| 01:46.05 | nny_1 | coolest name evar |
| 01:46.29 | Maliuta | an it's been my nick for nigh on 10 years now |
| 01:46.37 | Maliuta | registered with nickserv here for about 7 |
| 01:46.44 | nny_1 | nice |
| 01:46.51 | nny_1 | nny is a johnen vasquez character |
| 01:47.05 | `Sauron | nny_1: Actually, I've used this as my nick (earlier it was w/o the `) for > 10 years. Hell, almost 15 at this point |
| 01:47.06 | nny_1 | my original nick i am waiting on a password reset |
| 01:47.23 | `Sauron | Long before the movies made the name commonplace. |
| 01:47.40 | nny_1 | heh |
| 01:48.59 | jer | i'm getting a "_macro_exec: Context 'macro-dialext' for macro 'dialext' lacks 's' extension, priority 1" ... i just moved "macro-dialext" to a realtime extensions db, set up the macro-dialext context in extensions.conf to: switch => Realtime/@ ... and in my SQL, the entry exists as it did in the flat file (proper context, extension 's' for all 3 priorities, etc). anybody have any idea what's going wrong? |
| 01:49.39 | nny_1 | if anyone wnats to piss themselves |
| 01:49.41 | nny_1 | http://www.youtube.com/watch?v=4QAlt4Sfl7Q |
| 01:49.44 | nny_1 | good times |
| 01:51.32 | Mercestes | My nick is.........heh, original. :P |
| 01:55.10 | nny_1 | well damn that didn't work either |
| 01:55.19 | nny_1 | (init script) |
| 01:56.47 | nny_1 | ahh damn source doesnt put snmpd in /usr/sbin |
| 01:57.34 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 01:57.48 | *** join/#asterisk joat (n=joat@ip70-160-147-169.hr.hr.cox.net) |
| 01:58.15 | nny_1 | er wait no |
| 01:58.16 | nny_1 | lol |
| 01:58.19 | nny_1 | /usr/local/bin |
| 01:58.21 | nny_1 | :\ |
| 01:58.49 | jnc | hmm.. anyone familiar with using ekiga on a vpn connection? |
| 01:58.51 | jnc | does it work? |
| 01:59.05 | jnc | still trying to debug an asterisk setup here |
| 01:59.15 | nny_1 | yheah |
| 01:59.17 | nny_1 | yeah* |
| 01:59.19 | nny_1 | i have done it here |
| 01:59.45 | nny_1 | pretty straight forward |
| 02:00.46 | jnc | I'm getting a timeout |
| 02:00.50 | jnc | so it's not even connecting |
| 02:01.03 | jnc | gotta wonder, what the trick is in configuring ekiga to make this work |
| 02:01.10 | nny_1 | ok cool snmpd is up |
| 02:03.55 | jer | realtime extensions seem to be not working for me, getting a "macro 'dialext' lacks 's' extension, priority 1" when calling an extension that uses that macro; here's the pastebin of all relevant information: http://pastebin.ca/826221 |
| 02:06.22 | nny_1 | lol bad package manager |
| 02:06.29 | nny_1 | uninstalled my conf file too |
| 02:06.54 | joat | jer, i'm not so sure that SendDTMF means what you think it does... |
| 02:09.49 | jer | joat, works fine when it's defined in extensions.conf |
| 02:10.04 | joat | hmmm... |
| 02:10.09 | jer | exten => s,2,SendDTMF(${ARG1}) ... the problem with realtime extensions only exhibits itself when used with an 's' extension |
| 02:10.26 | jer | i define for example, local extensions in it, it works fine |
| 02:12.57 | blitzrage | jer: how are you calling the macro? |
| 02:13.01 | Mercestes | It's kinda like, using Mysql as an overglorified text editor....so not worth it. |
| 02:13.33 | blitzrage | jer: you shouldn't really use realtime like that -- create a static extensions.conf file and use func_odbc to store information in the DB |
| 02:13.55 | jer | blitzrage, like this: exten => otherexten,1,Dial(IAX2/otherprovider/1234|60|M(dialext^200)) |
| 02:14.48 | jer | blitzrage, well, my extensions.conf is a static file, i just want certain items to be pulled from the DB so my wife can make changes to the system (i don't want her editting config files, she's not very unix savvy) |
| 02:15.19 | blitzrage | it's easier for your wife to modify a SQL table than a flatfile? |
| 02:15.31 | jer | but one of them is the ivr, which i was haivng problems with; so i reverted back to statically defining it in extensions.conf, and picked a simple macro (this macro-dialext) to test with, same problem |
| 02:15.40 | jer | blitzrage, no, it's easier for her to point and click at a web page |
| 02:15.48 | jer | which is how this stuff is added to/removed from the db |
| 02:15.59 | blitzrage | interesting concept... :) |
| 02:16.18 | blitzrage | not sure why your Macro isn't working... sorry |
| 02:16.49 | jer | =/ thanks anyway |
| 02:16.49 | *** join/#asterisk teknoprep (n=tekno@unaffiliated/teknoprep) |
| 02:18.02 | jnc | nny_1: you had it working over a VPN? |
| 02:18.05 | teknoprep | well |
| 02:18.10 | teknoprep | i am going to have to learn asterisk now |
| 02:18.11 | nny_1 | jnc: yeah |
| 02:18.11 | teknoprep | lol |
| 02:18.22 | teknoprep | i am tired of relying on freepbx to configure asterisk |
| 02:18.42 | jnc | nny_1: did you configure ekiga to proxy through to the asterisk box? |
| 02:18.49 | teknoprep | what is the latest book on learning asterisk's configs |
| 02:18.51 | jnc | I don't get it, no connection gets through |
| 02:18.54 | teknoprep | is it still asterisk tfot |
| 02:19.09 | drmessano | ~book |
| 02:19.10 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 02:20.40 | teknoprep | ty |
| 02:21.11 | teknoprep | i'll first start by dropping my box at home and setting up everything manually |
| 02:21.41 | blitzrage | if you have something setup already... keep it, and use it for a reference |
| 02:22.14 | blitzrage | freepbx isn't going to necessarily write a great dialplan, but there could be some useful syntax in there for things that you want to try and replicate |
| 02:23.17 | *** join/#asterisk adjohn (n=adjohn@p5087-ipad408marunouchi.tokyo.ocn.ne.jp) |
| 02:26.08 | *** join/#asterisk mosty (n=mostyn@ppp191-34.static.internode.on.net) |
| 02:26.14 | *** join/#asterisk ar3dam (n=fl3pix@189.156.231.173) |
| 02:28.35 | jer | blitzrage, not sure if this is related to my problem or not (not likely considering the error message), but rt extensions don't seem to respect ${VAR} |
| 02:29.09 | blitzrage | them's the breaks when you use something non-standard |
| 02:30.04 | jer | indeed =] |
| 02:31.56 | *** join/#asterisk angom (n=Angel@201.170.49.106) |
| 02:32.29 | *** join/#asterisk infernix (n=nix@unaffiliated/infernix) |
| 02:38.50 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 02:38.50 | *** mode/#asterisk [+o anthm] by ChanServ |
| 02:41.51 | FuriousGeorge | hey all |
| 02:46.12 | *** join/#asterisk egypcio (n=egypcio@200.150.132.61) |
| 02:46.47 | *** join/#asterisk rhythmicmayhem (n=me@h242.225.39.162.ip.alltel.net) |
| 02:48.46 | teknoprep | blitzrage, yeah i was just going to copy the /etc/asterisk directory |
| 02:48.50 | teknoprep | to my laptop |
| 02:49.08 | teknoprep | blitzrage, i am going to use AsteriskNOW |
| 02:49.13 | teknoprep | blitzrage, since its a full install |
| 02:58.01 | teknoprep | what do you guys think of AsteriskNOW ? |
| 02:58.21 | *** part/#asterisk angom (n=Angel@201.170.49.106) |
| 03:03.09 | [TK]D-Fender | teknoprep, Good work... escape one GUI to become imprisoned by another. You've learned very little |
| 03:04.59 | teknoprep | lol |
| 03:05.07 | teknoprep | i was actually just going to use asterisk inside of it |
| 03:05.18 | teknoprep | i didn't want to compile everything from scratch is all |
| 03:05.39 | teknoprep | my home machine is a p3 800 |
| 03:05.43 | teknoprep | that would take some time |
| 03:06.01 | teknoprep | can i use asteriskNOW just as asterisk ? |
| 03:06.06 | teknoprep | without using the web gui |
| 03:08.04 | *** part/#asterisk egypcio (n=egypcio@200.150.132.61) |
| 03:08.51 | fujin | why would y ou? |
| 03:09.03 | jnc | the asteriskgui is a convenience front end |
| 03:09.06 | fujin | that's like, installing ubuntu desktop and using it as a server |
| 03:09.09 | jnc | I don't think you're required to use it |
| 03:10.02 | teknoprep | does asterisknow use a sql db for its configs? |
| 03:10.09 | Qwell | no |
| 03:10.26 | teknoprep | does it add stuff to the configs on a default install ? |
| 03:10.49 | [TK]D-Fender | teknoprep, How do I sum this up gently.... |
| 03:10.52 | mosty | teknoprep, asterisknow is a gui/config engine on top of asterisk |
| 03:10.59 | [TK]D-Fender | teknoprep, .... what a load of crap! |
| 03:11.07 | mosty | teknoprep, if you want asterisknow without the gui, just use asterisk |
| 03:11.16 | [TK]D-Fender | teknoprep, Compiling will take valuable 10 minutes out of your life. |
| 03:11.25 | teknoprep | astlinux a good choice? |
| 03:11.26 | fujin | if that |
| 03:11.56 | [TK]D-Fender | teknoprep, Keep running Forrest.... |
| 03:12.09 | teknoprep | [TK]D-Fender, heh |
| 03:12.17 | fujin | you're doing it wrong, as it were |
| 03:12.21 | fujin | http://doingitwrong.com |
| 03:13.14 | teknoprep | [TK]D-Fender, which linux distro do you prefer for your asterisk os ? |
| 03:13.26 | teknoprep | [TK]D-Fender, or asterisk systems |
| 03:14.37 | mosty | teknoprep, pick whichever distribution you're most familiar with |
| 03:15.07 | teknoprep | mosty, well i am pretty familiar with most of the major distributions BSD / debian / gentoo / redhat |
| 03:15.20 | teknoprep | BSD not being linux but whatever |
| 03:15.33 | Mercestes | I would suggest one of the latter 3. |
| 03:15.39 | mosty | teknoprep, i would recommend whichever you prefer out of centos/redhat/debian |
| 03:15.42 | Mercestes | my fav. being Gentoo but most people in here disagree iwth me. |
| 03:15.55 | Mercestes | like mosty. |
| 03:15.57 | RypPn | I wouldn't ;) |
| 03:16.15 | Mercestes | There is just something nice about "emerge asterisk" being my list of instructions, but...that's me. |
| 03:16.16 | teknoprep | gentoo is nice but i am definatly not compiling a system onto a p3 800 |
| 03:16.27 | Mercestes | Dude, that's what gentoo is *for* man. |
| 03:16.28 | *** join/#asterisk JT (n=j@unaffiliated/jt) |
| 03:16.32 | Mercestes | ricing out $20 pcs. |
| 03:17.46 | fujin | fuck that ;> |
| 03:20.48 | [TK]D-Fender | Mercestes, Did you pick up your complimentary fak Type-R sticker for your box? |
| 03:21.10 | Mercestes | Of course. |
| 03:21.25 | Mercestes | and a "got rice" sticker too. |
| 03:21.42 | teknoprep | i have alot to learn lol |
| 03:21.54 | teknoprep | i am reading through the from-internal setup in trixbox |
| 03:22.03 | teknoprep | in extensions.conf |
| 03:22.07 | Mercestes | ew, trixbox....nasty |
| 03:22.18 | Mercestes | trixbox is the devil. |
| 03:22.23 | teknoprep | i have been using elastix lately |
| 03:22.25 | teknoprep | i like it |
| 03:22.58 | teknoprep | but i want to setup a box that runs solely for small business that i can resell voip to ... where all they need is phones in there offices |
| 03:23.17 | teknoprep | and a small pfsense embeded router for QoS |
| 03:23.27 | Mercestes | YOu do that....and then I'll come in behind you after it breaks and charge them to wipe out trixbox and install regular asterisk. |
| 03:23.40 | teknoprep | no not using trixbox |
| 03:23.48 | Mercestes | It'd be like your my sales lead. :) |
| 03:23.49 | teknoprep | trixbox i have at my house |
| 03:23.54 | Mercestes | I'm sorry. |
| 03:24.12 | teknoprep | i have been using trixbox very successfully |
| 03:24.21 | teknoprep | but i am tired of not knowing what is actually going on |
| 03:24.33 | teknoprep | so i am going to setup this crap at my house without trixbox or elastix |
| 03:26.55 | mosty | is there a way to set the permissions to a specific value on files created by Monitor? i can do it with MixMonitor, but MixMonitor doesn't work with the g729 transcoder card |
| 03:29.51 | *** join/#asterisk mikecx (n=mikecx@cpe-76-181-117-188.columbus.res.rr.com) |
| 03:31.02 | Mercestes | mosty: Any way you could use system(chmod) ? |
| 03:31.54 | mosty | Mercestes, where would i put that in the call flow? in the h extension? |
| 03:32.10 | Mercestes | Yea. |
| 03:32.39 | mosty | i will try that |
| 03:37.29 | *** join/#asterisk watchy2 (i=watchy@70.247.77.22) |
| 03:37.45 | watchy2 | i got a freshly built * box and it seems to lag a little |
| 03:37.58 | watchy2 | anyone had issues with new core2duos? |
| 03:40.42 | *** join/#asterisk roeinstein (i=roeinste@c-71-193-30-237.hsd1.ca.comcast.net) |
| 03:42.15 | roeinstein | got a couple questions and I'm hoping maybe someone here can point me in the right direction :-) |
| 03:42.18 | mosty | do those cpu's support HT? have you tried disabling it? |
| 03:43.20 | flenders | hey, got something funny going on here. I updated asterisk to version 1.4.15 a few weeks ago, and now every 2 or 3 days, asterisk 'locks up'. everything seems normal, but we can't dial out. restarting asterisk seems to fix the issue. Sangoma A101 card, with wanpipe 3.2.1, zaptel 1.4.7, libpri 1.4. |
| 03:43.41 | flenders | hey, got something funny going on here. I updated asterisk to version 1.4.15 a few weeks ago, and now every 2 or 3 days, asterisk 'locks up'. everything seems normal, but we can't dial out. restarting asterisk seems to fix the issue. Sangoma A101 card, with wanpipe 3.2.1, zaptel 1.4.7, libpri 1.4.1, asterisk 1.4.15 |
| 03:43.52 | flenders | has anyone seem something simiilar? |
| 03:43.55 | mosty | flenders, try upgrading to asterisk 1.4.16.2 - there have been a few segfault/deadlock fixes |
| 03:44.23 | flenders | mosty, I thought about that, just wanted to see if someone else had similar issues |
| 03:44.54 | roeinstein | first off I have a couple different phone numbers right now and my sip provider passes that to me in a DNIS how can I make extensions unique for those numbers? right now I have dial plans setup for each number and they play different message but the extensions all work the same on all of them.. so basically if someone hit '1' I'd like different things to happen depending on the phone # called |
| 03:46.16 | flenders | roeinstein: use /enten_number on the register line on sip.conf |
| 03:46.26 | mosty | flenders, i read the changelog for 1.4.16.2 20 minutes ago, there are lots of segfault/deadlock fixes, i'm guessing it's an improvement |
| 03:46.42 | flenders | I was just reading that too |
| 03:47.56 | mosty | roeinstein, organise your dialplan in such a way to do that. set a channel variable for the called extension, and use that to jump to a context specifically for that extension |
| 03:48.36 | *** join/#asterisk coppice (n=chatzill@235.202.17.210.dyn.pacific.net.hk) |
| 03:48.51 | roeinstein | flenders, mostly, any example of this online you might be able to point me too? my knowledge of dial plans is pretty minimal, mine are pretty basic right now |
| 03:49.11 | [TK]D-Fender | roeinstein, the answer is to seperate your CONTEXTS. Again now is the time to STOP just throwing questions like that out here, sit down, and read THE BOOK |
| 03:49.15 | [TK]D-Fender | ~book |
| 03:49.15 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 03:49.16 | [TK]D-Fender | ^^^^^^^^^^^^^^^^ |
| 03:49.58 | roeinstein | I believe I have seperated them |
| 03:50.58 | roeinstein | haha, I don't need a detailed answer, maybe a page that goes into more detail about dial plans becuase the ones I have seen haven't helped me with what I am looking for |
| 03:51.11 | roeinstein | is that OK? |
| 03:51.18 | [TK]D-Fender | blitzrage, people miss PM's (Oh I didn't see the window) or when this channel isn't blocked to non-auth'd users are INCAPABLE of receiving them |
| 03:51.38 | blitzrage | that makes sense |
| 03:51.38 | [TK]D-Fender | roeinstein, There is a blatant chapter in the book. Go read. |
| 03:51.58 | blitzrage | but is speaking in CAPS on some WORDS really NECESSARY all the time? |
| 03:52.12 | [TK]D-Fender | blitzrage, No, only SOME of the time ;) |
| 03:52.33 | blitzrage | lowercase would have been just as effective |
| 03:52.48 | [TK]D-Fender | blitzrage, its all about the right emPHAsis on the right syLABle ;) |
| 03:52.50 | roeinstein | he OBVIOUSLY thinks it HELPS him make a POINT |
| 03:53.23 | [TK]D-Fender | roeinstein, Its proven effective, with 4 out of 5 psychologists in agreement. |
| 03:53.34 | coppice | maybe he's Nigerian |
| 03:57.54 | mosty | roeinstein, the book is the best place to start with this |
| 03:58.57 | Mercestes | actually, he really talks like that where he shouts random words in his sentence. |
| 03:59.54 | Mercestes | I think it's terets |
| 04:00.38 | Mercestes | :D |
| 04:00.51 | Mercestes | That's one way to do word association |
| 04:01.12 | [TK]D-Fender | Mercestes, a lot of osmosis will do you good. |
| 04:01.28 | Mercestes | Mmmmm...absorption through celluar membranes..... |
| 04:01.51 | Mercestes | cellular...... |
| 04:01.52 | Mercestes | even. |
| 04:01.56 | *** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 04:02.17 | Mercestes | hey, the pummeling works. |
| 04:02.31 | Mercestes | funny story...it took me forever to find "dictionairy.com" |
| 04:02.37 | [TK]D-Fender | Mercestes, yeah.. it's not "Nukular Science" ;) |
| 04:03.03 | [TK]D-Fender | </compoundcomicrelief> |
| 04:03.07 | Mercestes | lol |
| 04:03.30 | [TK]D-Fender | That's a bonus... I got to slam GWB in that one too... |
| 04:04.46 | roeinstein | yep reading, this is pretty helpful |
| 04:04.59 | Mercestes | I dunno, I gotta give the man props. He has every reporter on the planet hanging on every word he stutters in an ongoing attempt to catch, document, and advertise his every screwup on the 5 o'clock news, and he still has the brass to speak publicly. |
| 04:05.23 | Mercestes | I would have given up by now and hired a professional speaker. |
| 04:05.34 | jnc | N00bular Ircian program |
| 04:06.06 | Mercestes | but not GWB...he just keeps on screwing it up publicly without a care. That's the American way. |
| 04:06.48 | [TK]D-Fender | Mercestes, Or he's too dumb to acknowledge just how dumb he is. |
| 04:07.33 | Mercestes | it could be that. |
| 04:08.56 | Mercestes | I don't have high hopes of it getting better in the elections next year, though. |
| 04:10.02 | Mercestes | Got your pick between a muslim and the woman smart enough to marry an old horndog, and still publicly "supports him" after he banged an 18 year old intern. |
| 04:10.23 | Mercestes | I'd pick one of the other guys personally, but those seem to be the two that are winning. |
| 04:11.07 | [TK]D-Fender | Mercestes, Some corrections, Obama is CHRISTIAN, and Hillary has so much dirt on her that I don't think enough will wash clean for her to win. Her policy is ass. |
| 04:11.35 | [TK]D-Fender | Mercestes, Barack is too new, it sounds like a good start for him, but I don't see it. |
| 04:11.41 | Mercestes | Obama is "christian." atleast for the elections. |
| 04:11.44 | coppice | policy? what has that to do with getting elected? |
| 04:11.58 | jnc | Mercestes: not even a mention of Ron Paul? hope for america |
| 04:12.01 | [TK]D-Fender | Mercestes, personally I actually have 3 candidates that I would be THRILLED to see make it in. |
| 04:12.07 | Mercestes | his entire family clean on up to Moses is muslim, I don't buy that christian crap |
| 04:12.23 | jnc | [TK]D-Fender: steven colbert, ron paul, who's the third? |
| 04:12.24 | jnc | ;) |
| 04:12.33 | [TK]D-Fender | jnc, He's probably my #1, Kucinich a close 2nd, Gravel an ACTUAL "no hope in hell" third |
| 04:12.47 | jnc | poor mikey |
| 04:12.59 | [TK]D-Fender | jnc, Yup, he's WAY too cool to get anywhere. |
| 04:13.14 | [TK]D-Fender | jnc, He's a utopian. |
| 04:13.16 | coppice | america might have elected colin powell, but I can't think of any other black guy who stands a chance |
| 04:13.55 | mosty | wayne grady |
| 04:14.08 | [TK]D-Fender | coppice, He pronounces his name like a bottom part of an intestinal tract! If that doesn't scream "full of shit", I don't know what does! ;) |
| 04:14.20 | Mercestes | I was thinking that. |
| 04:14.26 | [TK]D-Fender | ZING! |
| 04:14.41 | Mercestes | The "powell" on the end of it doesn't sound too comfortable either. |
| 04:14.42 | [TK]D-Fender | I'm really on the ball tonight... |
| 04:15.03 | Mercestes | I'm not sure what a "powell" is but it sounds like an aggressive action verb to me. |
| 04:16.29 | jnc | onomotopeia (spelling?) |
| 04:16.39 | Hadi- | anyone here using Cisco IP phone |
| 04:16.40 | *** join/#asterisk pc500 (n=feaw@216-207-205-36.dia.static.qwest.net) |
| 04:16.40 | Mercestes | I think schwartzeneggar should be president. |
| 04:16.43 | Hadi- | with asterisk 1.2? |
| 04:16.48 | Hadi- | and codec g729? |
| 04:16.57 | Mercestes | Hadi-, once upon a time. Then I got a real phone |
| 04:17.13 | Hadi- | im having serious issues |
| 04:17.14 | Hadi- | with Silence Supression |
| 04:17.20 | Hadi- | and g729 codec |
| 04:17.22 | Hadi- | on asterisk |
| 04:17.23 | mosty | hadi: it should be disabled |
| 04:17.36 | Mercestes | Yea, I'm pretty certain asterisk 1.2 doesn't support silence supression. |
| 04:17.38 | jnc | Mercestes: not possible |
| 04:17.48 | Mercestes | jnc: I know....I'd vote for him anyway though. |
| 04:17.57 | Hadi- | mosty: it is |
| 04:17.59 | jnc | forbidden by the tiny sneeze left of the US constitution |
| 04:18.00 | Hadi- | im still losing audio |
| 04:18.16 | Mercestes | jnc: It's like...a friggen loophole. Schwartzeneggar is more american than muslim boy. |
| 04:19.16 | Mercestes | Yea, that pesky constitution.....I'm glad we got rid of all that crap like...freedom of speech and ....right to bear arms. I mean..what a dumb idea. Give everyone the right to offend one another, then the means to shoot whoever offended them. |
| 04:19.52 | Mercestes | suing offensive people into financial ruin for all eternity is a much better idea. |
| 04:20.02 | [TK]D-Fender | Mercestes, you still don't get that he is not Muslim.... |
| 04:20.03 | jnc | ...and no health insurance to pay for the gunshot wounds |
| 04:20.29 | Mercestes | [TK]D-Fender, I get that he's "not" Muslim. "not" in the sense that no one would vote for him if he was. |
| 04:20.33 | [TK]D-Fender | And Schwartzenegger?! EW! |
| 04:20.43 | jnc | Muslim? he's a douchebag, end of story |
| 04:21.00 | [TK]D-Fender | jnc, slightly yeah :) |
| 04:21.12 | jnc | btw that could be mistaken for hating on Muslims |
| 04:21.14 | Mercestes | I mean, please, mama's muslim and he's not? The only way that could be true is if he's agnostic/atheist. |
| 04:21.20 | [TK]D-Fender | jnc, not a serious douchebag. I'd say "mostly harmless" :) |
| 04:21.24 | jnc | I meant that, he's not a Muslim, because more importantly, he is a douchebag |
| 04:22.14 | jnc | it's like Al Gore flying around to warn people about global warming that happens from people flying around |
| 04:22.48 | jnc | you can't fly far enough away from Illinois state to get out of the hold of Richard Dailey |
| 04:23.11 | Mercestes | Global Warming is a hoax bred from uneducated fear. |
| 04:23.17 | jnc | no matter how many first class tickets upgraded from coach at the taxpayers expense you excuse yourself for |
| 04:23.39 | jnc | Obama has no plan |
| 04:23.50 | Mercestes | Obama has a nice suit tho. |
| 04:24.01 | jnc | none of the candidates I have seen do, with the exception of Colbert and Paul |
| 04:24.07 | jnc | lol |
| 04:24.08 | Mercestes | I don't know if he wears the same suit to every press conference or if he just has like 8 of the same one. |
| 04:24.11 | jnc | it is a nice suit |
| 04:24.24 | jnc | I know what you're talking about, I agree |
| 04:24.27 | jnc | it's striking |
| 04:24.47 | Mercestes | It is... |
| 04:24.53 | jnc | hizzunah |
| 04:24.55 | Nugget | I'll never forgive him for what he did to Meigs Field |
| 04:25.01 | jnc | lol |
| 04:25.01 | Nugget | what a fucker |
| 04:25.05 | drmessano | Isnt Richard Daley the guy that does the impression of the mayor of springfield on the simpsons? |
| 04:25.11 | jnc | I used to crew a boat there for some 10 years |
| 04:25.17 | jnc | one day, the fucking runway is gone |
| 04:26.02 | jnc | some kind of cloak and dagger douchebaggery took place at night |
| 04:26.05 | Nugget | I'm just glad I had a chance to land their before it was gone. |
| 04:26.11 | Nugget | er, "there" :) |
| 04:26.13 | jnc | ooh |
| 04:26.16 | jnc | pilot? |
| 04:26.19 | Nugget | yeah |
| 04:26.26 | *** join/#asterisk outtolunc (n=pchammer@c-67-174-216-60.hsd1.ca.comcast.net) |
| 04:26.28 | jnc | nice to hear that |
| 04:27.10 | jnc | we (speaking as a sailboater) enjoyed many a hippie-free year watching planes come in |
| 04:27.14 | jnc | thank you |
| 04:28.01 | coppice | douglas adams had presidents nailed perfectly, and nobody is a better illustration of what he said than bush |
| 04:28.37 | jnc | hitchhiker's guide is like a bible for today |
| 04:28.53 | jnc | I don't like the old testament, but damn I enjoy hitchhiker's guide |
| 04:28.58 | Mercestes | Well, the whole election thing is a hoax. "Ok, you get your choice of these three carefully selected candidates for president." |
| 04:29.03 | jnc | short stories with no point |
| 04:29.29 | Mercestes | jnc: Hithikers or the old testament? |
| 04:29.47 | Mercestes | *hitchhikers even |
| 04:29.51 | jnc | both, I thought |
| 04:30.00 | Mercestes | hehe |
| 04:30.03 | coppice | "Hey folks. Here's two guys you'd be a fool to buy a used car from, and you certainly wouldn't invite for dinner. Choose one to lead our country." |
| 04:30.04 | jnc | the stories run together but they're really not about anything |
| 04:30.32 | Mercestes | jnc: You really need to start specifying which book your referring to...I'm getting confused. |
| 04:31.10 | Mercestes | I think i should run for president |
| 04:31.15 | jnc | I'm saying that the "bible" and the "book" (hh2g) are roughly the same in literary excellence |
| 04:31.25 | roeinstein | ok hmm |
| 04:31.57 | roeinstein | I see that my phone numbers are being passed as an extension so I need to figure out how to do extensions for the extension? |
| 04:31.57 | roeinstein | lol |
| 04:32.06 | Mercestes | I propose a candadicy of tyranny, and I will forge a future of gonads and strife. |
| 04:32.10 | jnc | not like oh jesus dies there and there also |
| 04:32.16 | jnc | just that the style is very similar |
| 04:32.21 | jnc | short stories with no point |
| 04:33.28 | [TK]D-Fender | ~whee |
| 04:33.28 | jbot | [~whee] Weeeeeeeee! http://www.albinoblacksheep.com/flash/weeee |
| 04:33.46 | roeinstein | ok so this seems to be a problem, when my sip provider passes off the dnis it makes it an extension |
| 04:33.51 | jnc | after reading hitchhikers, I was nearly passed out from laughter, and then I had a profound sense of "explanation" for all things in life |
| 04:34.55 | jnc | :) |
| 04:35.04 | Mercestes | and I would approve all forms of torture |
| 04:35.11 | jnc | oh geeze |
| 04:35.17 | Mercestes | ....? |
| 04:35.28 | Mercestes | Can't be *that* bad, most of the girls I hang with want me to do stuff like that to them. |
| 04:35.40 | coppice | douglas adams didn't say anything really novel, but the way he said things really put them into perspective.... a bit like the total perspective vortex itself. |
| 04:36.14 | Mercestes | and I would invade New Zealand... |
| 04:36.22 | Mercestes | Just because.... |
| 04:36.37 | Mercestes | They could use the air time on the news. |
| 04:38.47 | coppice | Most guys would be happy just invading Jessica Alba's house |
| 04:38.59 | Mercestes | i'd be happy invading Jessica Alba. |
| 04:40.13 | Mercestes | Here, Jessie, have this roofie-colada. It's nummy. |
| 04:40.37 | Mercestes | Eh, screw it, we're talking about me being hypothetical president..I'd just have her brought in for...heh, questioning. |
| 04:43.54 | roeinstein | well I'm confused, my dialplans are broken up into different contexts |
| 04:44.37 | roeinstein | what I dont understand is how I tie the dnis my provider is passing me to the proper context |
| 04:44.58 | roeinstein | becuase right now it passes the the dnis AS the extension |
| 04:45.37 | Mercestes | roeinstein: I always did an exten => ${dnis},1,Goto(context,exten,priority) myself. |
| 04:45.44 | Mercestes | not literally, of course, but, something like that. |
| 04:47.36 | roeinstein | hmm nice :-) |
| 04:47.43 | roeinstein | thnx |
| 04:47.58 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 04:48.24 | Mercestes | So if one of your numbers is 1-800-277-4659 (a real number, btw) then you would do an exten => 8002774659,1,Goto(Context1,s,1) and do your magic there. |
| 04:54.48 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 05:04.30 | *** join/#asterisk piper69 (n=haiger@unaffiliated/piper69) |
| 05:04.41 | piper69 | !seen troy |
| 05:08.13 | *** join/#asterisk phix (i=threat@123-243-44-131.tpgi.com.au) |
| 05:08.17 | phix | hmmm |
| 05:08.33 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 05:08.33 | *** mode/#asterisk [+o russellb] by ChanServ |
| 05:08.43 | phix | I dial from my SIP phone, the other end picks up but my SIP phone still has the ringing tone |
| 05:08.46 | phix | wtf |
| 05:09.37 | roeinstein | Mercestes, thanks! |
| 05:09.44 | roeinstein | Mercestes, that was exactly what I was looking for |
| 05:09.57 | Mercestes | :) |
| 05:09.59 | Mercestes | np |
| 05:10.03 | Mercestes | I take paypals |
| 05:10.21 | Mercestes | phix: Does * see the other end pick up? |
| 05:10.24 | *** part/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 05:10.27 | *** join/#asterisk russellb_ (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 05:10.27 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 05:10.41 | phix | yes |
| 05:10.52 | coppice | there should be a paypal unnumbered swiss account edition |
| 05:11.01 | piper69 | Mercestes: can you help me with Cisco ATA 186 |
| 05:11.26 | roeinstein | sorry no pp or I definitely would :-) :-) |
| 05:11.29 | roeinstein | k I'm off :-) |
| 05:11.32 | *** part/#asterisk roeinstein (i=roeinste@c-71-193-30-237.hsd1.ca.comcast.net) |
| 05:11.40 | Mercestes | That's one way to get rid of 'em. |
| 05:11.47 | Mercestes | piper69, I'd rather not. |
| 05:11.59 | Mercestes | you could, hypothetically, start with what's broken... |
| 05:12.15 | piper69 | Mercestes: oh thank you anyways |
| 05:13.03 | phix | Mercestes: ? |
| 05:13.06 | phix | how does that help me? |
| 05:14.03 | *** join/#asterisk mog (n=mog@c-68-62-172-83.hsd1.al.comcast.net) |
| 05:14.03 | *** mode/#asterisk [+o mog] by ChanServ |
| 05:14.12 | Mercestes | phix: I've seena few phones "stick" and not really pick up even though you put the handset to your ear. Figured it'd be an easy solution. |
| 05:14.20 | Mercestes | Ciscos mainly. |
| 05:14.21 | *** join/#asterisk tengulre (n=tengulre@124.42.50.9) |
| 05:14.30 | phix | Mercestes: the phone I am using is a Nokia E65 |
| 05:14.39 | tengulre | hi,all |
| 05:14.39 | Mercestes | oh.... |
| 05:14.42 | Mercestes | that explains alot. |
| 05:14.50 | phix | it works when calling my voice mail on my asteriskl server |
| 05:14.57 | tengulre | how to test stun ? |
| 05:15.12 | phix | but it fails when calling a local SIP account or dialing via my VoIP provider |
| 05:15.44 | tengulre | I download the stun server & client on vovida.org ,but I don't know how to test it ? |
| 05:16.34 | phix | I cant call my phone either wtf |
| 05:16.39 | phix | it says it is registered |
| 05:17.02 | knn | I have an Dial command issue |
| 05:17.02 | knn | 1) Using Originate from php I land in <conf> context |
| 05:17.02 | knn | 2) then from there i issue Dial with G option two make the dialed channel to enter Meetme |
| 05:17.02 | knn | 3) The calling local channel goes to the <conf> context again and loops to issue Dial command again for the next conference particapant |
| 05:17.02 | knn | Everything works perfectly when * is run under a debugger, however without debugger when Dial just rings once for the third participant and then hangs up the third particapnt |
| 05:17.05 | knn | Any ideas |
| 05:17.10 | phix | SPAM |
| 05:22.39 | Mercestes | phix> I have yet to actually make the e65 really work. |
| 05:24.57 | FuriousGeorge | anyone familiar with peopleline? i know a guy in .ca who uses them, say they force him to use # as last 'digit' to complete a call |
| 05:25.08 | FuriousGeorge | wonder why that is |
| 05:27.08 | Mercestes | FuriousGeorge, The "#" key is often translated into a "dial" key on many phones. They are probably lazily circumventing a poorly designed dialplan. |
| 05:27.42 | FuriousGeorge | i thought that might be the case |
| 05:33.56 | *** join/#asterisk abdul7383 (n=abdul738@e179144048.adsl.alicedsl.de) |
| 05:34.42 | abdul7383 | hi |
| 05:38.06 | *** join/#asterisk LakeSolon (n=blake@64-83-209-86.dhcp.stcd.mn.charter.com) |
| 05:42.21 | ar3dam | hi there .. how i can use a modem like a fxo card? . |
| 05:43.30 | Mercestes | ar3dam, does your country use 110v or 220v? |
| 05:44.06 | ar3dam | :S .. what:S .. ? |
| 05:44.45 | Mercestes | Do your outlets put out 110v or 220v? |
| 05:44.53 | nny_1 | anyone have an opinion on WARNING[6385]: loader.c:363 load_dynamic_module: Error loading module 'res_snmp.so': /usr/lib/libnetsnmpagent.so.15: undefined symbol: boot_DynaLoader |
| 05:45.12 | nny_1 | i am determined to get this ox spewing snmp data tonight... -_- |
| 05:45.12 | ar3dam | ok, use both outlet .. |
| 05:45.35 | ar3dam | why mercestes? |
| 05:45.38 | Mercestes | If it's 220v you can put the leads pretty much anywhere on yoru body, if it's 110v, I would suggest a mucuous membrane, like your tongue or eyes. |
| 05:45.53 | Mercestes | ie: I would give up on the modem as an Fxo card deal. |
| 05:46.02 | Mercestes | if zaptel doesn't see it, it's impossible, and even if it does...it will sound like ass. |
| 05:46.22 | Mercestes | it's only a very specific chipset that even works and I kidna wish no one had mentioned it. |
| 05:46.53 | Mercestes | I'd just buy an FXO card. |
| 05:47.04 | nny_1 | lol |
| 05:47.15 | nny_1 | he just wouldn't* walk into that one easily |
| 05:47.22 | *** part/#asterisk dijungal (n=kdaniel@209.59.110.18) |
| 05:47.53 | Mercestes | nny_1, they never do. |
| 05:47.55 | Mercestes | ~cheap |
| 05:47.56 | jbot | extra, extra, read all about it, cheap is 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. |
| 05:48.16 | ar3dam | ja ja .. so funny man .. ok ok .. |
| 05:48.20 | dezenten | hehe |
| 05:49.13 | nny_1 | the boot_DynaLoader error is because libperl is missing (AFAICS net-snmp uses this). We should have autotools check for libperl and libnetsnmp stuff… |
| 05:49.27 | nny_1 | http://www.callweaver.org/ticket/51 |
| 05:53.31 | russellb | ?? |
| 05:55.41 | nny_1 | #*(&#@*($&!#(\ |
| 05:55.47 | nny_1 | I really really hate this shit |
| 05:55.53 | nny_1 | such garbage |
| 05:57.55 | dezenten | whats callweaver ? |
| 05:58.16 | nny_1 | dunno.. trying to get snmp working on asterisk here and I am having one problem after another |
| 05:59.03 | Mercestes | Callweaver could be referred to as "the rebel force" I suppose. It's a group of rogue programmers who thought asterisk sucked so they took the source code, viciously raped it, and then submitted the nasty, used, sweaty remains as a "new release." |
| 05:59.53 | dezenten | Mercestes: what you would call a fork |
| 06:00.01 | Mercestes | basically |
| 06:00.44 | Mercestes | It says it supports T.38 passthrough and runs in a virtualized environment, but, I've never even managed to make it work. |
| 06:01.00 | dezenten | virtualized enviroment ? |
| 06:01.03 | dezenten | like xen ? |
| 06:01.08 | dezenten | or chroot ? |
| 06:01.12 | Mercestes | xen, Vmware, etc. |
| 06:01.15 | dezenten | aah |
| 06:01.52 | dezenten | i use asterisk on xen sometimes |
| 06:01.53 | piper69 | where is troy? |
| 06:01.59 | dezenten | works really well |
| 06:02.06 | Mercestes | It's ok as long as you don't need a zap timer. |
| 06:02.14 | dezenten | yepp |
| 06:02.22 | Mercestes | I read somewhere that * was supposed to fix that soon somewhere. |
| 06:02.33 | dezenten | i use it mostly for testing our product |
| 06:03.30 | piper69 | guys i setp an ata for my dad and i sent it to him overseas, i tested before i send it and it was working fine, but now where my dad plugs it in it doesn;t work |
| 06:03.54 | piper69 | i am thinking it could be because of the IP?! |
| 06:03.55 | dezenten | that sux |
| 06:04.06 | dezenten | if he is behind nat |
| 06:04.15 | dezenten | that could be a problem |
| 06:04.35 | dezenten | not your dad behind nat... the ata that is |
| 06:04.36 | dezenten | =) |
| 06:05.10 | dezenten | i got all my voip-stuff on public ipadresses here |
| 06:05.11 | piper69 | i am trying to acccess his xp machine , but i don't know how to get the address of the ata |
| 06:05.35 | dezenten | check the dhcp-server |
| 06:05.50 | Mercestes | HIs country could be blocking VoIP traffic too. |
| 06:05.59 | dezenten | where does he live ? |
| 06:06.10 | piper69 | africa |
| 06:06.14 | dezenten | ah yes |
| 06:06.20 | piper69 | sudan in africa |
| 06:06.26 | dezenten | do you know how to configure vpn ? |
| 06:06.34 | piper69 | nope |
| 06:06.45 | Mercestes | google openvpn howto |
| 06:06.46 | dezenten | i was talking to an ISP from africa on VON summer this year |
| 06:06.52 | dezenten | they were blocking voip |
| 06:07.02 | dezenten | i dont know which country |
| 06:07.07 | dezenten | yeah |
| 06:07.23 | dezenten | openvpn is easy and goes thru almost anything |
| 06:07.35 | Mercestes | 3 bit encryption would be helpful too. Enough to mask VoIP traffic but not enough for them to go "Ahh! Encryption!" |
| 06:07.55 | dezenten | 3bit ? |
| 06:08.04 | Mercestes | Yea. |
| 06:08.04 | dezenten | how do you do that ? |
| 06:08.05 | piper69 | yarageel |
| 06:08.08 | dezenten | aah |
| 06:08.39 | dezenten | thats funny |
| 06:08.41 | Mercestes | You are, of course, risking an international event in which you could be sent to africa and made property of the province to pay of your debt of your obscene violation of their local laws. |
| 06:08.55 | coppice | http://www.xkcd.com/ seems to be on topic |
| 06:09.07 | dezenten | Mercestes: true |
| 06:10.12 | knn | Mercestes, do you have any idea on my Dial issue |
| 06:10.39 | dezenten | piper69: like Mercestes says.. you should check if its legal to use encryption and stuff there |
| 06:11.05 | Mercestes | knn: I suggest a consultant |
| 06:11.11 | Mercestes | knn: But otherwise, not really. |
| 06:11.32 | Mercestes | I never said he should check. I said he'd be brutually victimized by the african authorities. |
| 06:11.46 | Mercestes | was kind of hoping for it, actually. >.> |
| 06:11.57 | dezenten | Mercestes: yes, i told him to check because of your point there |
| 06:12.08 | Mercestes | spoil-sport. :P |
| 06:12.58 | knn | I have an Dial command issue, can anyone help... |
| 06:12.58 | knn | 1) Using Originate from php I land in <conf> context |
| 06:12.58 | knn | 2) then from there i issue Dial with G option two make the dialed channel to enter Meetme |
| 06:12.58 | knn | 3) The calling local channel goes to the <conf> context again and loops to issue Dial command again for the next conference particapant |
| 06:13.00 | knn | Everything works perfectly when * is run under a debugger, however without debugger when Dial just rings once for the third participant and then hangs up the third particapnt |
| 06:14.24 | dezenten | i havnt used context other than default in 2 years |
| 06:14.34 | *** join/#asterisk SwK (n=SwK@user-24-214-55-149.knology.net) |
| 06:18.30 | nny_1 | Mercestes: i am gonna have to put your callweaver quote in wikipedia |
| 06:18.58 | Mercestes | roflmao |
| 06:19.07 | piper69 | which tftp you guy recommand |
| 06:19.09 | Mercestes | I think that somehow..someone, somewhere, someone might take offense. |
| 06:19.14 | Mercestes | opentftp |
| 06:19.28 | piper69 | Mercestes: for windows |
| 06:19.45 | Mercestes | format c: /s /u |
| 06:19.57 | Mercestes | Then install linux. |
| 06:20.01 | Mercestes | then install opentftp |
| 06:20.02 | piper69 | ok |
| 06:20.30 | piper69 | Mercestes: i did format c: /s /u and it ask me to reboot |
| 06:20.41 | nny_1 | Mercestes: so honestly what do you think of the site? Never really done the cute approach before.. I kind of like it, but I am a weird one |
| 06:20.49 | Mercestes | I like it. |
| 06:21.43 | piper69 | Mercestes: should i say yes to reboot the system |
| 06:22.30 | Mercestes | piper69, no, that would damage the data on your harddrive. You should definately yank the power plug to cancel changes. Then plug it back in and tap the power button a few dozen times to clear the command outof memory. |
| 06:23.29 | piper69 | Mercestes: when i power it back on it says no operating system found |
| 06:23.29 | nny_1 | lol |
| 06:23.52 | piper69 | what does that me, it never did that to me before |
| 06:23.54 | Mercestes | Perfect! Now insert the linux cd you burned.... |
| 06:23.59 | Mercestes | ......oh wait...that was step one..damnit. |
| 06:24.58 | dezenten | :( |
| 06:25.04 | piper69 | Mercestes: how can i make it work again, i have to finish writting my sa paper for tomorrow's final |
| 06:25.27 | Mercestes | install linux and open office. |
| 06:25.36 | Mercestes | oh and, opentftp |
| 06:25.41 | piper69 | only one page to go and then i can come and chat again |
| 06:25.59 | piper69 | no i have windows |
| 06:26.07 | Mercestes | http://www.tftp-server.com/ btw |
| 06:26.35 | Mercestes | funny how google windows tftp returned a useful result. |
| 06:26.42 | piper69 | but i can't see the start button |
| 06:26.58 | piper69 | the screen is all black |
| 06:27.01 | dezenten | there is a tftpd in windows xp i think |
| 06:27.26 | piper69 | Mercestes: but it say operation system not found |
| 06:27.31 | piper69 | what does that mean |
| 06:27.39 | nny_1 | http://sourceforge.net/projects/filezilla |
| 06:27.41 | dezenten | it means you got windows |
| 06:28.30 | piper69 | but in the past i just press the power and it goes to where i can click start and use office |
| 06:28.44 | nny_1 | that feature has been upgraded |
| 06:28.50 | coppice | if you've got windows, does that mean you only have 6 months to live? |
| 06:28.52 | piper69 | now it just black screen |
| 06:29.22 | *** join/#asterisk jochieng (n=jochieng@217.194.147.193) |
| 06:29.28 | nny_1 | wonder why there isnt more defenstration jokes with windows |
| 06:33.01 | *** join/#asterisk gerphimum (n=trekkie@cpe-70-125-148-108.satx.res.rr.com) |
| 06:33.23 | *** join/#asterisk snazm (n=snazm@78.146.170.13) |
| 06:34.06 | *** part/#asterisk snazm (n=snazm@78.146.170.13) |
| 06:34.28 | drmessano | I got one |
| 06:34.44 | drmessano | Why is Windows more secure than Linux? |
| 06:34.46 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 06:35.12 | piper69 | Mercestes: btw, yo mama is sitting here on my lap and she was watching yo ass acting smart, she told me not to listen to you when you said format c: /s /u. she also said she will slap you in your ass when she comes back to Houston because you are gay |
| 06:35.20 | drmessano | Because Microsoft carefully HAHAHA LOLZ Sub7 HAXORED GREETZ LOL |
| 06:35.30 | drmessano | Best I could do being this tired |
| 06:36.07 | piper69 | lmao |
| 06:36.09 | Mercestes | drmessano, still, better tan piper's retort. |
| 06:37.55 | drmessano | I was trying to think of a good Vista joke... |
| 06:37.58 | Mercestes | but....A: my mom's never been to houston, B: as anyone in #asstricks will tell you, not only am I not gay, my viewpoints are actually quite offensive to the lispy types, and finally C:, he has a broken ATA and no TFTP server and no one to help him, so revenge is mine. |
| 06:38.12 | drmessano | But I couldnt find anyone using it that I could use as a reference |
| 06:38.14 | Mercestes | drmessano, try installing it. Always gives me a laugh. |
| 06:38.30 | Mercestes | I use it. You can use me as a reference. |
| 06:38.33 | drmessano | lol |
| 06:38.43 | Mercestes | Like I always said, my putty windows into linux have never looked cooler. |
| 06:38.50 | drmessano | ROFLLL |
| 06:38.58 | drmessano | I used Vista for 30 minutes once |
| 06:39.09 | Mercestes | OH, so you never even got to log in. |
| 06:39.17 | drmessano | AHHAHA!! FTW |
| 06:39.45 | drmessano | Actually.. I was trying to install Cisco VPN and VNC on it |
| 06:39.49 | drmessano | neither of them worked |
| 06:39.54 | drmessano | I got a headache from the aero glass |
| 06:40.18 | Mercestes | I kinda like Aero glass...but I also have a gforce gtx 8800 gfx card in my box. |
| 06:40.23 | drmessano | and I told him that even though it was his personal laptop, if he wanted company software on it, he better bring it back with XP installed |
| 06:40.36 | piper69 | Mercestes: revenge is mine, you remind me of Robin williams |
| 06:40.37 | Mercestes | My GPU is bigger than my other 3 computers...somehow..I couldn't bring myself to put linux on that. |
| 06:40.39 | piper69 | lol |
| 06:40.41 | awk | anyone had any issues with 1.4.16? |
| 06:40.45 | awk | buddies, queues, etc etc? |
| 06:40.51 | Mercestes | piper69, Nice, thanks. Robin Williams is actually kinda funny. |
| 06:41.03 | piper69 | ;) |
| 06:41.08 | Mercestes | drmessano, and my beryl cube spun *way* too fast. :( |
| 06:41.30 | Mercestes | *that* gave me a headache. It was like, blindness every time I used my wheel mouse on the wrong part of my screen. |
| 06:41.31 | drmessano | lol |
| 06:41.49 | *** join/#asterisk harpal (n=Harpal@124.125.255.24) |
| 06:41.49 | jochieng | Hello everyone -- how do i start asterisk,zaptel on boot? |
| 06:42.02 | drmessano | I got one.. Why does Linux have more security holes than Vista? |
| 06:42.25 | drmessano | Because you can actually get driver for your NIC in Linux |
| 06:42.30 | drmessano | drivers* |
| 06:42.38 | piper69 | Mercestes: my ATA is not broken ! it a matter of me knowing how to config it |
| 06:42.40 | piper69 | lol |
| 06:43.16 | Mercestes | drmessano, ....That is kinda true. =/ |
| 06:43.24 | *** join/#asterisk adjohn (n=adjohn@p5087-ipad408marunouchi.tokyo.ocn.ne.jp) |
| 06:43.25 | Mercestes | jopchieng: What OS are you on? |
| 06:43.35 | jochieng | debian etch |
| 06:43.47 | Mercestes | dont' supose you have rc-update =/ |
| 06:43.54 | Mercestes | you have to add it to your rc stuff. |
| 06:44.11 | Mercestes | in gentoo I would just rc-update add zaptel default but I dunno if debian etch has that. |
| 06:44.12 | jochieng | Mercestes:i installed using apt-get install zaptel \ asterisk |
| 06:44.55 | piper69 | they acutally say that if you place your Nikes close enought to your vista box .. it will acually install the driver "Found a new air jordan...do you want me to install it"? |
| 06:45.18 | drmessano | lol |
| 06:45.31 | drmessano | I loved the Dell ad |
| 06:45.38 | piper69 | which one |
| 06:45.44 | drmessano | The low spec PC "Great for booting the operating system" |
| 06:46.07 | drmessano | As in, all you could do is boot |
| 06:46.10 | drmessano | nothing more |
| 06:46.24 | piper69 | hahahahahh, i didn't saw that one |
| 06:46.50 | drmessano | It was a real Dell ad |
| 06:46.57 | drmessano | Side by side of 3 spec PCs |
| 06:47.12 | drmessano | and the middle was good for apps, the high end was good for gaming |
| 06:47.19 | drmessano | AKA "Minesweeper" |
| 06:47.37 | piper69 | drmessano: but dude they really have good service and equipment |
| 06:47.54 | piper69 | all my boxes are dell |
| 06:47.54 | drmessano | I dont blame Dell.. I blame the OS |
| 06:48.32 | piper69 | you name it from inspiron to latitude to diminssion all of it |
| 06:48.34 | piper69 | yes |
| 06:48.41 | piper69 | but the have ubuntu now |
| 06:48.45 | piper69 | no |
| 06:49.13 | drmessano | At least its not an HP |
| 06:49.35 | coppice | A big corporate Dell customer gets good service |
| 06:49.36 | coppice | A small Dell customer gets pissed on |
| 06:49.41 | *** join/#asterisk gerphimum (n=trekkie@cpe-70-125-148-108.satx.res.rr.com) |
| 06:49.41 | piper69 | omg , i hate HP i don't care what every body else say |
| 06:49.55 | drmessano | Im almost certain when they merged that HPs printers lived on, and Compaqs PCs became HPs |
| 06:50.06 | drmessano | Because "HP" PC's don't exist anymore |
| 06:50.23 | drmessano | At least not the decent crap I remember |
| 06:50.26 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 06:50.37 | piper69 | i like the new design laptop |
| 06:50.41 | drmessano | Ugh |
| 06:50.43 | piper69 | have you seen that |
| 06:50.51 | drmessano | I have an NX9420 from last year |
| 06:50.56 | drmessano | Its a piece of ELLLL CRAPPO |
| 06:51.30 | piper69 | "ELLLLL CRAPPO" |
| 06:51.43 | drmessano | Dual core 1.83 Pentium M I think.. which means it runs at 1.8GHZ 50% of the time, and 1.83GHZ the other 50% of the time |
| 06:52.47 | drmessano | It would be just like the "new HP" to disallow both cores to work at the same time, or during the same hour |
| 06:53.37 | *** join/#asterisk bkw_ (n=brian@adsl-64-149-47-251.dsl.tul2ok.sbcglobal.net) |
| 06:53.54 | piper69 | drmessano: i would love to sit and chat with you , but i really want to get this stupid ATA to work , i can't load the h323 on it |
| 06:55.13 | piper69 | *H.323 |
| 06:55.31 | Mercestes | drmessano, that's largely true. |
| 06:55.45 | drmessano | ok |
| 06:56.21 | drmessano | We've been an HP shop for 3 years now.. and I miss Dell more and more |
| 06:56.39 | drmessano | Everytime I get a new HP desktop I sob... just a little |
| 06:56.46 | Mercestes | Yea, The proliant series made it in, I think HP took over a few consumer Pcs, but a couple of the better compaq PCs stayed. |
| 06:56.50 | piper69 | if anyone would like to help me with my Cisco ATA 186 please drop me pm |
| 06:57.07 | Mercestes | Umm...no, you made fun of my mom. |
| 06:57.24 | piper69 | Mercestes: i will apologize |
| 06:57.26 | piper69 | ;) |
| 06:57.31 | Mercestes | Nah, it's ok. |
| 06:57.42 | drmessano | All the HP desktops have the Green plastic crap in them.. which was my first tipoff |
| 06:57.44 | piper69 | bastered |
| 06:57.45 | Mercestes | I think your doomed, honestly. |
| 06:57.52 | piper69 | why? |
| 06:57.53 | Mercestes | bastard, actually. |
| 06:57.58 | Mercestes | Well, 1, your using an ATA. |
| 06:58.01 | drmessano | basted? |
| 06:58.07 | piper69 | lol |
| 06:58.08 | Mercestes | mmmm..basted merc. |
| 06:58.23 | Mercestes | Me, personally, I would put an Asterisk box in Africa and IAX2 it over via a VPN. |
| 06:58.25 | Mercestes | but that's me. |
| 06:58.37 | Mercestes | 2: Your trying to do VoIP over seas and that generally tends to be a little painful at best. |
| 06:58.38 | piper69 | Mercestes: no no no |
| 06:58.41 | drmessano | HA |
| 06:58.46 | piper69 | wait |
| 06:58.55 | piper69 | this ATA is here in usa |
| 06:59.01 | drmessano | If youre going to put it in Africa, better install it on an OLPC and get someone to keep it cranked |
| 06:59.04 | Mercestes | I thought you said it was in Africa. |
| 06:59.10 | piper69 | the other ata is diffrent |
| 06:59.19 | piper69 | yes that was another one |
| 06:59.28 | Mercestes | so you have *2* atas that don't work? |
| 06:59.30 | drmessano | Who has Asterisk on an OLPC yet? |
| 06:59.42 | Mercestes | ~olpc |
| 06:59.43 | Mercestes | ? |
| 06:59.48 | drmessano | I had the idea of "One PBX Per Child" which is so damn american |
| 06:59.50 | Mercestes | save me a google? |
| 07:00.01 | drmessano | "One Laptop Per Child" PC's |
| 07:00.15 | Mercestes | Oh... |
| 07:00.25 | piper69 | no, the linksys ata works fine , i sent it to my dad to try and talk to him, but that one did work due to ip config |
| 07:00.29 | Mercestes | It can be done... |
| 07:00.43 | Mercestes | I have asterisk on my Linksys wrt54gl |
| 07:00.56 | Mercestes | It *might* even handle a phone call. |
| 07:01.01 | Mercestes | or two |
| 07:01.14 | piper69 | now i want to config this cisco ata 186 so i can use it here |
| 07:01.17 | *** join/#asterisk Maliuta_ (n=nikolai@119.11.99.9) |
| 07:01.34 | Mercestes | Might I suggest the admin guide? |
| 07:01.44 | drmessano | the world would be a better place if all those starving children in 3rd world countries had a working Asterisk PBX |
| 07:04.14 | Mercestes | Why would starving kids need phone calls that don't work half the time? |
| 07:04.16 | *** join/#asterisk tengulre11 (n=tengulre@124.42.50.9) |
| 07:05.05 | drmessano | So you're saying they need a decent Wireless infrastructure to go with it? |
| 07:05.18 | drmessano | Ok, but thats gonna cut into the food ration |
| 07:05.18 | Mercestes | s/wireless// |
| 07:06.26 | drmessano | Also need to cut a deal with Cisco for phones |
| 07:06.53 | drmessano | Do they make one thats Zebraproof? |
| 07:07.19 | *** join/#asterisk Abydos313 (n=abydos31@adsl-76-214-22-120.dsl.lsan03.sbcglobal.net) |
| 07:08.49 | Mercestes | s/cisco/polycom/ |
| 07:10.40 | piper69 | i keep getting upgrade failed |
| 07:12.30 | drmessano | lol |
| 07:12.52 | *** join/#asterisk sergee (n=serg@195.94.224.197) |
| 07:14.35 | piper69 | drmessano: you think its funny |
| 07:16.48 | drmessano | Well |
| 07:16.56 | drmessano | You said Cisco |
| 07:16.59 | drmessano | I LOL'ed |
| 07:17.41 | piper69 | omg |
| 07:18.30 | piper69 | looks like you guy feel better when someone else is strugling |
| 07:18.59 | drmessano | Not at all |
| 07:19.19 | drmessano | Have you reset it? |
| 07:19.25 | piper69 | ELLLLLLLLLLLL CRAPOOOO |
| 07:19.26 | drmessano | Checked the admin guide? |
| 07:19.44 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 07:23.07 | *** join/#asterisk docelm0 (n=vircuser@c-68-32-135-157.hsd1.de.comcast.net) |
| 07:31.20 | Mercestes | drmessano, they never read the admin guide |
| 07:39.34 | arctanx | guides++. I still had to make a comic about my asterisk setup though http://arctanx.id.au/comic/12.htm |
| 07:44.45 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53) |
| 07:45.59 | drmessano | LOL |
| 07:54.51 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 07:58.37 | arctanx | It's quite inexplicable really. I have asterisk 1.2 working fine with an SPA3000. Incoming calls have the CID prefixed with a letter, and it's set to ring on line 1 (the FXS port on the same device), but there's a selective forward to push all calls starting with that letter off to an asterisk extension, the result being that the caller doesn't get the phone picked up until asterisk does an Answer(). Found that on a forum. But come 1.4, an i |
| 07:59.20 | arctanx | And where in 1.2 I had outgoing calls going out through the FXO fine (cheers to this channel), I would get 503 circuit busy in 1.4. No idea at all why. So I went back to 1.2 :P |
| 07:59.38 | arctanx | The only reason I ever bothered with 1.4 was to get volgain in voicemail.conf |
| 07:59.40 | arctanx | </rant> |
| 08:05.26 | *** join/#asterisk adjohn (n=adjohn@p5087-ipad408marunouchi.tokyo.ocn.ne.jp) |
| 08:12.15 | drmessano | CentOS needs a "server CD" for 5.1 |
| 08:15.03 | *** join/#asterisk Schumie (n=Steve@cmarfw01.marlow.spinvox.com) |
| 08:15.59 | tzafrir | arctanx, what version of 1.4? |
| 08:16.16 | tzafrir | In such a case, please file a bug with a complete trace (sip debug) |
| 08:17.24 | tzafrir | arctanx, also, a script I use to test a different version of Asterisk with minimal disruptions to the installed version: |
| 08:18.32 | tzafrir | http://svn.digium.com/svn/asterisk/team/group/zapata_conf/contrib/scripts/live_ast |
| 08:18.47 | tzafrir | it should work with any asterisk >= 1.4 |
| 08:19.16 | jochieng | i am running v1.2 how do i add extension and agensts to start making basic calls |
| 08:19.26 | tzafrir | So it can help you testing asterisk 1.4 if you can afford a short downtime |
| 08:20.19 | tzafrir | jochieng, "add"? to what exactly? Is this a new installtion? |
| 08:20.26 | tzafrir | Or an existing one? |
| 08:20.34 | *** join/#asterisk joelsolanki (i=joelsola@202.160.161.94) |
| 08:20.37 | joelsolanki | hi all |
| 08:20.38 | jochieng | yes this is my very first time |
| 08:21.09 | joelsolanki | i have 4 incoming lines. when i make call i hear 3 ring then it goes to extension |
| 08:21.31 | joelsolanki | is there a way to make less rings or completely remove the ringing process |
| 08:21.38 | joelsolanki | and divert the call to extension. |
| 08:21.49 | joelsolanki | any hints ? |
| 08:22.45 | joelsolanki | it is rhino fxo card |
| 08:32.16 | *** join/#asterisk harpal (n=Harpal@124.125.255.24) |
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| 08:48.43 | *** join/#asterisk X-Filez (i=x-files@x-files.lv) |
| 08:50.07 | X-Filez | Ppls, Please help, I have SNOM 320 and Asterisk 1.4 , but SNOM random time and don't use phone or use, lost registration SIP, logs SNOM -> http://pastebin.com/m45869e32 , debug sip asterisk -> http://pastebin.com/m271a0a8a |
| 08:53.59 | *** join/#asterisk implicit (i=implicit@gateway/tor/x-6af4ff2c7f5d29a1) |
| 08:54.36 | FlatFoot | Xifilez: what firmware version on snom ? and that debug is showing a password mismatch |
| 08:54.46 | implicit | anyone here use SIPP? |
| 08:55.34 | FlatFoot | Xifilez: try using snom320-7.1.19-SIP-f.bin it's the only one i found that works correctly |
| 08:56.04 | X-Filez | FlatFoot: Version 7.1.30 |
| 08:56.47 | FlatFoot | X-filez: i tried all different versions and found that snom320-7.1.19-SIP-f.bin is the only one that operates the phone correctly |
| 08:57.41 | X-Filez | FlatFoot: hmm, i see in http://wiki.snom.com/Firmware/V7/Update , snom320 7.1.30 |
| 08:58.06 | FlatFoot | X-filez: http://snom.provu.co.uk/sw/snom320-7.1.19-SIP-f.bin |
| 08:58.09 | X-Filez | FlatFoot: this problem you have same ? and fixed in 7.1.19 ? |
| 08:58.29 | FlatFoot | X-filez: lots of diff probs all fixed with .19 |
| 08:58.31 | X-Filez | ok thanks |
| 08:58.57 | FlatFoot | X-filez: i have told snom and they are supposed to be making a new version that works :) |
| 08:59.18 | FlatFoot | X-filez: but that 401 error is a password mismatch |
| 08:59.49 | X-Filez | FlatFoot: :( mde |
| 08:59.58 | X-Filez | FlatFoot: you try write to dev snom ? |
| 09:00.31 | FlatFoot | X-filez: no i told my supplier who is the major UK retailer for snom and they are in contact with snom |
| 09:02.47 | X-Filez | FlatFoot: :) maybe next version fixed it... |
| 09:03.16 | FlatFoot | X-filez: maybe , but i would'nt hold your breath |
| 09:03.40 | coppice | contact with snom sounds rather unpleasant :-) |
| 09:04.08 | FlatFoot | coppice: i have never got through direct , only through my supplie |
| 09:04.09 | FlatFoot | r |
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| 09:18.42 | ice_croft | hi all |
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| 09:54.19 | gormux | hi all |
| 09:54.51 | gormux | i still fail to find a way to make text messages working in my asterisk |
| 09:54.53 | *** join/#asterisk saftsack (n=saftsack@83.218.162.174) |
| 09:55.25 | gormux | i have a 415 error, and cant figure how to make this work |
| 09:55.38 | gormux | any idea ? |
| 09:57.42 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 09:58.24 | gormux | dont understant, should be simple, i'ts such a simple use |
| 10:10.37 | *** join/#asterisk af_ (n=getsmart@88-149-240-60.dynamic.ngi.it) |
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| 10:47.19 | jeanmi___ | hi |
| 10:47.40 | jeanmi___ | after a call has been completed (one side did hangup) I'd like to know the duration of the call |
| 10:47.55 | jeanmi___ | is there a variable that would contain such an information ? |
| 10:47.55 | kaldemar | gormux: have you bothered to find out what 415 is? |
| 10:48.18 | JT | gormux: what text messages, what are you talking about? |
| 10:48.24 | gormux | yes, its an unsupported media type |
| 10:48.30 | gormux | the instant messaging |
| 10:48.47 | JT | gormux: so why should it be easy? |
| 10:48.56 | gormux | for soft phones |
| 10:49.07 | kaldemar | if it's easy to use, it's easy to configure. |
| 10:49.47 | JT | gormux: you didn't answer my question |
| 10:50.06 | gormux | once the connection is made, i suppose its not hard to transmit simple text messages |
| 10:50.09 | jeanmi___ | also I'd like to find out which side did hang up |
| 10:50.27 | gormux | so i imagined that it would be not too hard |
| 10:50.29 | JT | gormux: i suppose you can also imagine a lot of things |
| 10:50.37 | JT | many of them aren't true |
| 10:50.44 | gormux | it seems |
| 10:55.11 | gormux | so, seems that the only way to make these functions work is patching asterisk, recompile the whole thing, if i have understand |
| 10:55.27 | gormux | source : http://www.voip-info.org/wiki/view/Asterisk+SIP+Messaging |
| 10:55.45 | JT | most of us use asterisk to make and receive phone calls |
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| 10:59.35 | X-Filez | FlatFoot: there ? |
| 11:04.25 | *** join/#asterisk _ys (n=yuri@80.70.236.69) |
| 11:04.35 | gormux | JT: yeah, but it can sometimes be useful, and my boss wants it working |
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| 11:23.45 | jeanmi___ | in an extension, I have a Wait(). The caller is hanging up during the wait. I am then getting a "Spawn extension .... exited non-zero". How can I catch that ? |
| 11:24.19 | jeanmi___ | because there a few lines after the wait which I'd liek to be exectuted even though the caller hung up |
| 11:35.41 | ai-a[dead] | looking for asterisk door phone. we have an old system that works. has an ext, and a switch. looking for an asterisk network device that has a PSTN and a solenoid to switch the 'switch' when the internal guy (from PBX ONLY) enters a DTMF code. |
| 11:40.40 | *** join/#asterisk harpal (n=Harpal@124.125.255.24) |
| 11:41.17 | X-Filez | ~bristuff |
| 11:41.18 | jbot | i guess bristuff is a patch collection with a hfc-pci driver supporting NT and TE mode ISDN. See http://junghanns.net/ for more info |
| 11:46.24 | *** join/#asterisk bantu (n=Miranda@p54A32F18.dip0.t-ipconnect.de) |
| 11:49.33 | *** join/#asterisk Faustov (n=faustov@unaffiliated/faustov) |
| 11:51.50 | tzafrir | ~bristuff |
| 11:51.51 | jbot | methinks bristuff is a collection of patches to asterisk to support BRI in zaptel with HFC-based cards. See http://junghanns.net/ and http://bristuff.org/ |
| 11:53.00 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 11:53.38 | __freedom__lover | hi all |
| 11:54.01 | __freedom__lover | can anyone help me about res_snmp? |
| 11:54.16 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 11:59.08 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.80.10) |
| 11:59.52 | Faustov | quick question on some terms: agent = registered extension for a physical phone? |
| 12:02.04 | tzafrir | __freedom__lover, not sure. maybe someone can if you ask a specific question |
| 12:02.47 | tzafrir | Faustov, basically yes |
| 12:03.04 | tzafrir | It doesn't have to be a physical phone technically. But yes |
| 12:03.11 | Faustov | k |
| 12:03.15 | *** join/#asterisk harpal (n=Harpal@124.125.255.24) |
| 12:03.34 | __freedom__lover | i'm trying to install asterisk mib into net-snmp. i've configured the snmpd.conf but when a use snmpwalk, it return nothing |
| 12:04.00 | jochieng | hello ppl |
| 12:04.07 | Faustov | ok, one more thing: could anyone point me to some manual or examples of incoming call configuration? like, call from sip provider a go to extension A |
| 12:04.09 | Faustov | and so on |
| 12:04.29 | jochieng | i am running etch and i am getting this error wheni dial from CLI> |
| 12:04.30 | jochieng | -- Executing Wait("SIP/1001-0819c930", "1") in new stack |
| 12:04.30 | jochieng | <PROTECTED> |
| 12:04.30 | jochieng | <PROTECTED> |
| 12:04.30 | jochieng | <PROTECTED> |
| 12:04.30 | jochieng | <PROTECTED> |
| 12:05.05 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 12:05.12 | tzafrir | Faustov, in the user/friend configuration you set context=xyz . In extensions.conf, do whatever you need under [xyz] |
| 12:06.35 | jochieng | #flood -- Executing Wait("SIP/1001-0819c930", "1") in new stack |
| 12:06.36 | jochieng | <PROTECTED> |
| 12:06.36 | jochieng | <PROTECTED> |
| 12:06.36 | jochieng | <PROTECTED> |
| 12:06.36 | jochieng | <PROTECTED> |
| 12:06.51 | JT | ~pb |
| 12:06.51 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 12:07.36 | Faustov | tzafrir: what do you mean by user/friend configuration? sip.conf and type=friend? (i got peer there) |
| 12:08.07 | tzafrir | yes |
| 12:08.28 | __freedom__lover | does anyone have a good manual for res_snmp? |
| 12:09.24 | Faustov | tzafrir: so, type=peer lets me only make calls, while type=friend also allows receiving? |
| 12:12.52 | *** join/#asterisk Winkie (n=urmom@general-kt-195.t-mobile.co.uk) |
| 12:13.50 | Faustov | ok found it in the manual |
| 12:14.15 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 12:18.35 | jochieng | #pastbin http://www.pastebin.com [app_dumpchan.so] => (Dump Info About The Calling Channel) |
| 12:18.37 | jochieng | <PROTECTED> |
| 12:19.20 | *** join/#asterisk jochieng (n=jochieng@217.194.147.193) |
| 12:27.34 | Faustov | tzafrir: so in extensions.conf, i got [xyz] exten => DID,1,Dial(agent) - am i getting this right? |
| 12:27.48 | Faustov | and i'd distinguish sip accounts by DID this way |
| 12:27.55 | tzafrir | Faustov, basically |
| 12:28.23 | tzafrir | instead of "agent", you should probably have "channel of agent". e.g: SIP/007 |
| 12:28.36 | Faustov | k |
| 12:28.49 | Faustov | and DID would be the username of that sip account, right? |
| 12:29.20 | tzafrir | The DID depends on your provider |
| 12:31.08 | jochieng | hello |
| 12:31.08 | Faustov | tzafrir: so maybe a better idea to make separate contexts for each sip provider? |
| 12:32.07 | tzafrir | probably |
| 12:32.18 | tzafrir | and use include => ; for the common parts |
| 12:33.08 | Faustov | k |
| 12:33.29 | Faustov | how would a exten => line look in each of them for incoming calls then? |
| 12:33.54 | jochieng | can some body help -- i m running debian etch and i am having problems with asterisk |
| 12:34.48 | FlatFoot | X-Filez: back now |
| 12:36.41 | ice_croft | hi all |
| 12:36.52 | *** join/#asterisk Maliuta (n=nikolai@119.11.100.130) |
| 12:37.34 | Faustov | tzafrir: exten => s,2,Dial(SIP/0007,,) <--- something like this? |
| 12:38.17 | tzafrir | Yes |
| 12:39.26 | tzafrir | You can use simply: Dial(SIP/0007) |
| 12:39.34 | tzafrir | if you don't pass the other arguments |
| 12:40.33 | Faustov | oh so they are not mandatory |
| 12:40.35 | Faustov | k |
| 12:40.53 | jochieng | please some1 help.. |
| 12:41.34 | Faustov | awesome |
| 12:41.43 | Faustov | looks like it's complete! |
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| 13:42.08 | jochieng | hi everyone -- can some one tell me why this error is showing up (pbx.c:1741 pbx_extension_helper: Cannot find extension context 'default') I am using bebian etch |
| 13:44.45 | davevg-btwtech | do you have a [default] context in extensions.conf? |
| 13:50.20 | [TK]D-Fender | davevg-btwtech: Clearly he DOESN'T |
| 13:50.50 | [TK]D-Fender | jochieng: You've got something in your config pointing to a context that does not exist. |
| 13:50.52 | blitzrage | [TK]D-Fender: honestly... first thing I see you say today has a CAPITALIZED word? |
| 13:51.17 | [TK]D-Fender | blitzrage: *sigh* |
| 13:51.24 | [TK]D-Fender | blitzrage: tgif! |
| 13:52.06 | [TK]D-Fender | (should have been capitalized, but I wouldn't want it to be taken the wrong way) |
| 13:52.37 | blitzrage | yes, thank goodness it's friday... although that doesn't really mean much :) |
| 13:53.02 | tzafrir | jochieng, if you used Debian Etch, this wouldn't have happened ;-) |
| 13:53.20 | [TK]D-Fender | ..... |
| 13:53.23 | [TK]D-Fender | *cough* |
| 13:53.24 | tzafrir | jochieng, seriously, when you see such a message, run from the CLI: |
| 13:53.41 | tzafrir | dialplan show default |
| 13:53.56 | *** join/#asterisk PepOSX (n=pepOSX@201.248.215.16) |
| 13:54.00 | tzafrir | that would show the dialplan context [default] |
| 13:54.21 | tzafrir | It also has tab completion for context names |
| 13:54.29 | [TK]D-Fender | tzafrir : Whats the point of that? It doesn't exist. You know it doesn't exist, and Asterisk just finished telling him that to his face. |
| 13:54.59 | [TK]D-Fender | tzafrir : Something else is referencing it. |
| 13:55.19 | tzafrir | Or it can be a fallthrough |
| 13:55.35 | FlatFoot | an aside :- anyone read this ... http://bofh.ntk.net/Bastard.html ... ? |
| 13:55.40 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 13:55.43 | [TK]D-Fender | tzafrir : Either way that CLI command won't give him anything |
| 13:55.52 | [TK]D-Fender | (of value) |
| 13:56.10 | cappiz | I hva a queue wehere all my calls should be placed in by default... (no IVR, just straight to queue). I got one "problem", when i call the phone rings straight away, but the caller doesnt here the music/announcement before the 'agent-timeout' periode has passed. |
| 13:56.48 | [TK]D-Fender | cappiz: Pastebin the calls CLI output from beginning to end at verbose 10 please |
| 13:56.52 | [TK]D-Fender | ~pb |
| 13:56.52 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 13:57.08 | [TK]D-Fender | cappiz: And have you verified through some other means that MoH is functional at all? |
| 13:57.28 | [TK]D-Fender | cappiz: And in your pastebin, please include your queues.conf & musiconhold.conf |
| 13:58.02 | tzafrir | jochieng, so in short, give some more context. What messages do you see immedietly before that one? |
| 13:58.10 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 13:58.13 | cappiz | [TK]D-Fender, let me take a look |
| 13:58.54 | [TK]D-Fender | jochieng: Please pastebin the complete call where that occurs as well. |
| 14:04.20 | cappiz | [TK]D-Fender, you want the outout of /var/log/asterisk/full ? |
| 14:04.41 | [TK]D-Fender | cappiz: No, I want CLI output at verbose 10 as stated |
| 14:04.53 | cappiz | k |
| 14:08.13 | cappiz | here is the CLI sip debug |
| 14:08.14 | cappiz | http://pastebin.com/d33fc463d |
| 14:09.10 | cappiz | queue: http://pastebin.com/d97cc28e |
| 14:10.02 | cappiz | moh: http://pastebin.com/d30adba17 |
| 14:12.07 | *** join/#asterisk ELBunce (n=erik@kde/developer/bunce) |
| 14:12.22 | *** join/#asterisk egypcio (n=vinicius@200.150.142.210) |
| 14:13.49 | [TK]D-Fender | cappiz: FreePBX is not supported here |
| 14:13.55 | anonymouz666 | how do we suppose to answer an incoming SIP OPTIONS request? |
| 14:14.04 | cappiz | Okey |
| 14:14.34 | [TK]D-Fender | cappiz: Executing Macro("Local/101@from-internal-0096,2", "dial|25|trM(auto-blkvm)|101") in new stack <-- and this line that gets called to actually ring your agent is using "r" to provide forced ringing. This would override MoH |
| 14:15.28 | cappiz | could you explain :) ? what does r stand for? |
| 14:16.02 | *** join/#asterisk vetetix (n=vetetix@eclip3.ec-lille.fr) |
| 14:16.26 | [TK]D-Fender | cappiz: "Ring". But this is moot. This isn't a system under your control, and this isn't a place to try and debug it. |
| 14:16.37 | cappiz | okey |
| 14:16.55 | cappiz | very strange that it works this way though |
| 14:17.01 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 14:17.35 | *** join/#asterisk phillipk (n=pkey@fw.datafax.net) |
| 14:17.45 | [TK]D-Fender | anonymouz666: * doesn't support responding to OTIONS IIRC |
| 14:18.24 | *** join/#asterisk w3n0y (n=chatzill@241644hfc229.tampabay.res.rr.com) |
| 14:19.01 | anonymouz666 | at least I don't know a way to do that. |
| 14:19.38 | file | chan_sip responds to incoming OPTIONS requests, happens automatically |
| 14:21.40 | anonymouz666 | so why you must have an extension for that? My box is answering a 404... |
| 14:22.21 | file | you don't *have* to... depending on the remote SIP stack it might be perfectly happy with a 404 |
| 14:23.34 | file | but it does check for the presence of the extension in the OPTIONS packet |
| 14:23.35 | anonymouz666 | I don't think so. Because it retransmits the request so fast. |
| 14:27.18 | anonymouz666 | Ok |
| 14:27.27 | anonymouz666 | Very easy to fix. |
| 14:27.39 | anonymouz666 | just s,1,NoOp and it makes * answer a 200 OK. |
| 14:28.16 | anonymouz666 | thanks for clarifying. |
| 14:33.38 | *** join/#asterisk adjohn (n=adjohn@p5087-ipad408marunouchi.tokyo.ocn.ne.jp) |
| 14:35.49 | *** join/#asterisk phillipk (n=pkey@fw.datafax.net) |
| 14:41.05 | *** join/#asterisk errr (n=errr@fedora/errr) |
| 14:41.08 | *** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-a54ac96b51598bfb) |
| 14:42.55 | *** join/#asterisk bkw_ (n=brian@adsl-64-149-47-251.dsl.tul2ok.sbcglobal.net) |
| 14:43.53 | *** join/#asterisk oppassum (n=op@adsl-76-249-13-6.dsl.klmzmi.sbcglobal.net) |
| 14:45.17 | oppassum | does anyone know an easy way to "highlight a directory name" on a phone. i.e. i send a signal to a phone, it reads that signal and highlights a directory name as a response |
| 14:45.37 | oppassum | i've got the premise done, i just need to know how to highlight the name, or make some kind of notice that something has been done |
| 14:46.18 | [TK]D-Fender | oppassum: That would be extremely phone specific. |
| 14:46.37 | *** join/#asterisk Skarmeth (n=Skarmeth@201.9.74.32) |
| 14:46.39 | oppassum | Fender: polycom phones |
| 14:46.40 | [TK]D-Fender | oppassum: And I know of no phone that can do anything like that. |
| 14:46.47 | oppassum | ... |
| 14:46.56 | oppassum | you don't know a whole lot then, i've seen it on many phones |
| 14:47.12 | ai-a | contact polycom. ask for software addition :) |
| 14:47.19 | [TK]D-Fender | oppassum: Closest thing to that on Polycom is to use the MicroBrowser, but that means the user has to go LOOKING for that information. |
| 14:47.29 | [TK]D-Fender | oppassum: Oh, like which? |
| 14:47.35 | ai-a | oppassum: what phone supports it ? |
| 14:48.02 | jwh | it'll be nasty proprietary stuff |
| 14:48.09 | oppassum | not specifically what i'm doing...but any phone that can transfer someone can do what i'm talking about. |
| 14:48.09 | [TK]D-Fender | oppassum: And this one aspect is a poor one for trying to judge my experience |
| 14:48.42 | oppassum | Fender: you may help people here, but you do it in the most conceited way. Rest assured you aren't being judged on this one instance. |
| 14:48.43 | [TK]D-Fender | oppassum: Give me a specific model and a reference to a document that explains this functionality |
| 14:49.07 | oppassum | well as I'm creating the functionality by hand, I obviously can't do that. |
| 14:49.19 | file | what does transferring have to do with looking up a directory name? the phone probably does it as part of the transfer but outside of that you can't control it |
| 14:49.31 | file | at least in the SIP world. |
| 14:49.48 | [TK]D-Fender | oppassum: So far noone else here seems to think this is anything mainstream and standardized. Beyond SIP messaging, which * doesn't really do. |
| 14:50.14 | [TK]D-Fender | oppassum: You can't show me a an example of this functionality yuo say is suported all over the place? |
| 14:50.18 | *** join/#asterisk sts (n=sts@mia.ono.at) |
| 14:50.26 | oppassum | exactly what i'm trying to do isn't done |
| 14:50.31 | oppassum | similar things are. |
| 14:50.49 | oppassum | changing the soft key functions, for instance. |
| 14:50.56 | ai-a | "<oppassum> you don't know a whole lot then, i've seen it on many phones" <-does NOT say similar things. |
| 14:51.09 | [TK]D-Fender | oppassum: So first you tell me I'm wrong and it can be, and then you tell me that it doesn't exist. Which one is it now? |
| 14:51.10 | ai-a | you claim you've seen it on many phones. |
| 14:51.39 | arctanx | tzafrir: The issues I described are in asterisk-1.4.16.1. My fear is that things are set up in a sort of "hackish" manner and that asterisk 1.4 is actually dealing with things more elegantly |
| 14:51.47 | oppassum | look, stop attacking me here. I'm not looking for a concrete answer, I'm looking for a way to do this. excuse me for not using the word "similar" |
| 14:52.05 | [TK]D-Fender | oppassum: No, you attacked me, and I'm asking you to validate it. |
| 14:52.07 | Qwell | oppassum: and at least 3 people have told you that exactly what you want isn't possible automatically |
| 14:52.19 | ai-a | :) |
| 14:52.41 | arctanx | tzafrir: resulting in things not working how they do in 1.2. Still, it's bizarre. It's at work, so I can't guarantee that I'll get the chance to stuff around on there getting the right info for a bug report, but I'll see |
| 14:52.58 | tzafrir | arctanx, sorry. I don't recall the exact problem |
| 14:53.27 | sts | hello folks. i seem to experiance some interrupt problems with my digium card. doesn anybody know how i could debug this issue? |
| 14:53.36 | Qwell | sts: call Digium support |
| 14:53.37 | sts | s/doesn/does/ |
| 14:53.41 | arctanx | tzafrir: No worries, it was an SPA3000 set up in a particular way working with 1.2, then getting 302s and 503s when trying to do particular things in 1.4. |
| 14:53.42 | tzafrir | ah, the problem with "1.4" earlier this morning? |
| 14:53.45 | Qwell | sts: they're best equipped to help you |
| 14:54.51 | arctanx | tzafrir: As I understand it, the SPA3000 is deprecated and has other issues (such as with RFC2833 DTMF) anyway |
| 14:54.56 | tzafrir | arctanx, if you can afford a short downtime I suggested you a way to get some debugging trace... |
| 14:55.10 | arctanx | Yes I did see your message, I have it saved, and I will do so if I get the opportunity |
| 14:55.15 | arctanx | Thanks for that |
| 14:55.25 | tzafrir | a device that works with SIP of asterisk 1.2 and not of 1.4? |
| 14:55.27 | sts | well, the problem is that i cannot use a dedicated interrupt for my card since the BIOS of my HP server only allows me to share IRQ 7 with the first usb controller. |
| 14:55.45 | sts | did anybody have the same issue? |
| 14:55.58 | *** join/#asterisk mog (n=mog@nat/digium/x-4c59099e66eb4c80) |
| 14:55.58 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:56.00 | tzafrir | sts, have you tried support@ ? |
| 14:56.04 | [TK]D-Fender | sts: What version of * & zaptel are you using? |
| 14:56.11 | sts | what about APIC? |
| 14:56.12 | arctanx | tzafrir: That's correct. However, I was using an odd configuration. Actually, I believe my work has another SPA3000 sitting around... I'll see if I can borrow it and experiment at home. I should be able to get some definitive results |
| 14:56.41 | sts | [TK]D-Fender: the last stable version |
| 14:56.42 | *** join/#asterisk clandmeter (n=Carlo@81.175.82.2) |
| 14:56.52 | [TK]D-Fender | arctanx: SPA-3000 seems to work good with "AVT INFO" matching "dtmfmode=info" in sip.conf |
| 14:56.53 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 14:56.56 | Qwell | sts: which is? |
| 14:57.01 | [TK]D-Fender | sts: ^^^ |
| 14:57.05 | dominic1 | which codec is better G722 or G711? |
| 14:57.12 | Qwell | dominic1: depends |
| 14:57.15 | [TK]D-Fender | dominic1: G.722 clearly. |
| 14:57.20 | tzafrir | g722, if it is actually supported |
| 14:57.21 | sts | [TK]D-Fender: gimme a sec... |
| 14:57.22 | arctanx | [TK]D-Fender: That's right. I use info to avoid that rfc2833 issue |
| 14:57.40 | file | if you're on a 56k modem, both suck! >:D |
| 14:58.11 | arctanx | They work nicely over gigabit lan though I've found :) |
| 14:58.41 | file | that gigabit ethernet stuff will never catch on! token ring I say. |
| 14:58.47 | sts | [TK]D-Fender: zaptel-1.2.22.1, libpri-1.2.7, asterisk-1.2.26 |
| 14:59.11 | [TK]D-Fender | sts: Please read the channel topic. |
| 14:59.26 | [TK]D-Fender | sts: We've been on 1.4 for over a year now. |
| 14:59.53 | dominic1 | Thank you for the information. Do you know where I can get more information about the differences G711 vs G722? |
| 14:59.59 | [TK]D-Fender | sts: Development on 1.2 is completely dead and no bug fixes are coming out. |
| 15:00.09 | tzafrir | Except for Zaptel |
| 15:00.19 | jochieng | Hi -- |
| 15:00.24 | arctanx | It's interesting, debian and ubuntu both only have 1.2 in their stable repository -- does anybody know why? |
| 15:00.26 | Qwell | [TK]D-Fender: unless we screw up...twice |
| 15:00.30 | [TK]D-Fender | sts: And there have been major changes to the Zaptel driver in 1.4 to help with IRQ sharing |
| 15:00.35 | tzafrir | Where bug fixes and support for new hardware are included |
| 15:00.40 | jochieng | i am getting this -- pbx.c:1741 pbx_extension_helper: Cannot find extension context 'default' on my debian etch |
| 15:00.52 | jochieng | i am using version 1.2 |
| 15:00.56 | [TK]D-Fender | arctanx: Because glaciers move faster than Debian packagers |
| 15:00.56 | jochieng | of asterisk |
| 15:00.58 | tzafrir | jochieng, and I asked you a followup question on that one |
| 15:01.24 | arctanx | [TK]D-Fender: good point. they still have gaim for that matter. Anyway, it's 2am here and time for bed |
| 15:01.27 | [TK]D-Fender | jochieng: And I asked your for complete CLI output of the failed call attempt at verbose 10 over an hour ago and you have not provided it |
| 15:01.28 | arctanx | cheers and night all :) |
| 15:01.31 | tzafrir | arctanx, what Ubuntu? |
| 15:01.44 | arctanx | tzafrir: I think it's dapper LTS which probably explains it |
| 15:01.57 | tzafrir | http://packages.debian.org/asterisk , http://packages.ubuntu.com/asterisk |
| 15:02.27 | tzafrir | That package in Ubuntu LTS is in Universe, and hence not supported as part of LTS |
| 15:02.48 | sts | [TK]D-Fender: well, we are forced to use the old version, tho |
| 15:02.57 | sts | [TK]D-Fender: since the upgrade would need to much time ATM.. |
| 15:03.15 | [TK]D-Fender | sts: Its a really bad thing when you ask for help and can't follow through with what you need to do. |
| 15:03.28 | tzafrir | The Ubuntu package is based on a Debian package. And Dapper was frozen at about the same time Etch was frozen |
| 15:04.05 | arctanx | tzafrir: The package is in universe for all versions, going by that list. And gutsy's a mess as far as I'm concerned, which is why I haven't encountered 1.4 there |
| 15:04.06 | tzafrir | If you use Ubuntu, I'd recommend a newer version. If you use Etch: that package is reasonable, and has been updated with security fixes |
| 15:04.54 | jochieng | tzafrir: i tried the command you gave but still no luck .. jochieng*CLI> dialplan show default No such command 'dialplan' (type 'help' for help) |
| 15:05.06 | arctanx | Anyway, actually going to bed now. night all |
| 15:06.27 | sts | [TK]D-Fender: well the problem is the factor time... so currently i'm forced to stay at this version.. |
| 15:06.43 | tzafrir | jochieng, if you use 1.2: show dialplan whatever |
| 15:07.06 | [TK]D-Fender | sts: Doesn't sound like you have much of a choice. Digium cards + IRQ issues + * 1.2 = bad |
| 15:07.20 | tzafrir | jochieng, but as TK noted there, it is interesting to see what led to that message |
| 15:07.34 | sts | [TK]D-Fender: ok. thank you for that information. |
| 15:07.36 | [TK]D-Fender | jochieng: And I told you that there was no point to that and for you to show us the complete call attempt where that error message occurs |
| 15:07.37 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 15:07.44 | tzafrir | An incoming call of soe sort? An attempt to call out some where? |
| 15:07.49 | *** join/#asterisk anthm (n=anthm@70-9-142-245.area4.spcsdns.net) |
| 15:07.49 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:15.04 | jochieng | [TK]D-Fender: & tzafrir: please take a look at these http://pastebin.com/m2b23b363 and http://pastebin.com/d572af76e |
| 15:15.34 | *** part/#asterisk sts (n=sts@mia.ono.at) |
| 15:16.55 | [TK]D-Fender | jochieng: And I asked you to pastebin the complete call attempt that generated that message, and if its due to a SIP call, do so with SIP debug enabled. |
| 15:17.18 | [TK]D-Fender | jochieng: And please pastebin the output of "ls -l /etc/asterisk" as well |
| 15:17.23 | *** join/#asterisk _ys (n=yuri@80.70.236.69) |
| 15:17.41 | *** join/#asterisk mikecx (n=mikecx@pool-68-238-4-252.chi.dsl-w.verizon.net) |
| 15:17.49 | jochieng | [TK]D-Fender: i am a newbie i am not well versed with asterisk this is my very first time to instal it |
| 15:18.15 | jochieng | <PROTECTED> |
| 15:18.44 | mikecx | i can't get my Playtones(ring) to stop ringing. http://pastebin.ca/826964 is my extensions.conf |
| 15:19.44 | *** join/#asterisk gardo (n=gardo@121.97.108.189) |
| 15:22.39 | [TK]D-Fender | mikecx: I might wonder because of the way that SLA uses meetme that this might be the issue |
| 15:23.48 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:23.50 | mikecx | [TK]D-Fender: sounds about right. Is there a way to do two things during one step? |
| 15:24.17 | mikecx | [TK]D-Fender: i think if I had the stop tones get called in the goto it might work |
| 15:24.48 | [TK]D-Fender | mikecx: I'm not sure why you'd have to calling ringing yourself anyways... |
| 15:25.07 | mikecx | [TK]D-Fender: i pick up the call to check for faxes and the boss doesn't want an IVR |
| 15:25.24 | jochieng | [TK]D-Fender:take a look at this http://pastebin.com/ddc3d201 and http://pastebin.com/d73faf9ef |
| 15:25.28 | [TK]D-Fender | mikecx: Yes, but I thought that SLATrunk would generate ringing... |
| 15:25.57 | [TK]D-Fender | jochieng: -rw-r----- 1 root root 17706 2007-12-21 12:17 extensions.conf |
| 15:26.16 | mikecx | [TK]D-Fender: it does so long as the line hasn't been answered. once that happens all the person dialing in hears is silence. The phone themselves still ring, but the outside user hears nothing |
| 15:26.22 | [TK]D-Fender | jochieng: All of your other configs are owned by Asterisk. You have a files permissions issue and its not getting loaded because of this |
| 15:27.00 | jochieng | [TK]D-Fender:i saw that and i have corrected it immediately |
| 15:27.10 | jochieng | [TK]D-Fender: but still problem looomes |
| 15:27.46 | jochieng | [TK]D-Fender: unless i have to restart asterisk i think |
| 15:28.02 | [TK]D-Fender | jochieng: You might |
| 15:28.09 | [TK]D-Fender | jochieng: but a simple "reload" might do |
| 15:28.38 | jochieng | [TK]D-Fender: done let me try and make a call now |
| 15:31.26 | jochieng | [TK]D-Fender: after reload i see this http://pastebin.com/d36870aa4 |
| 15:31.49 | *** join/#asterisk amessina (n=amessina@h-66-166-108-202.chcgilgm.covad.net) |
| 15:31.54 | jochieng | [TK]D-Fender: cld this be with my configs |
| 15:32.21 | amessina | putnopvut: thanks for the fix: http://bugs.digium.com/view.php?id=11589 |
| 15:32.51 | putnopvut | amessina: your welcome. |
| 15:34.55 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 15:36.06 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 15:36.06 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:38.56 | *** part/#asterisk piper69 (n=haiger@unaffiliated/piper69) |
| 15:38.58 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 15:41.10 | mikecx | [TK]D-Fender: do you think switching to nvfaxdetect might help at all? |
| 15:44.13 | *** join/#asterisk FireMac (n=firemac@CPE000d88ae88b9-CM0011ae8bb0ee.cpe.net.cable.rogers.com) |
| 15:46.39 | *** part/#asterisk [gnubie] (n=[gnubie]@cm71.gamma182.maxonline.com.sg) |
| 15:48.21 | *** join/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 15:48.45 | mikecx | looks like that goto is useless |
| 15:50.15 | jochieng | [TK]D-Fender: Now i have this other problem http://pastebin.com/d4691c9e7 :-( ;) |
| 15:51.52 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-24-106-241-121.central.biz.rr.com) |
| 15:52.38 | ZPertee | when I configure zapata.conf and I split an 8 fxo port card into 8 channels does the numbering start left to right or right to left? |
| 15:53.29 | *** join/#asterisk bgat (n=bgat@adsl-75-23-66-208.dsl.peoril.sbcglobal.net) |
| 15:55.13 | ZPertee | nevermind I figured it out |
| 15:55.22 | rob0 | :) I was going to say TIAS |
| 15:55.49 | tzafrir | ZPertee, well, what is it? Don't keep us in the dark |
| 15:56.58 | *** join/#asterisk gerphimum (n=trekkie@cpe-70-125-148-108.satx.res.rr.com) |
| 15:57.49 | jochieng | tzafrir: kindly help with my situation for [TK]D-Fender seems not to be available at the moment |
| 15:58.04 | mikecx | any idea what options SLATrunk takes? |
| 15:58.27 | jochieng | tzafrir: with this http://pastebin.com/d4691c9e7 |
| 15:59.29 | *** join/#asterisk infernixx (n=nix@unaffiliated/infernix) |
| 16:00.38 | tzafrir | jochieng, what did you expect to happen there? What has happened? |
| 16:01.39 | ZPertee | tzafrir sorry I figured since I was such a noob everyone already knew the answer. I found in an article that they are numbered left to right |
| 16:01.47 | [TK]D-Fender | jochieng: #### WITH NO DAIL TONE EITHER AND 2000 DOESN'T RING <--- 200 isn't designed to ring. You can see that your dialplan has it answer and hangup immediately |
| 16:02.24 | [TK]D-Fender | jochieng: It waits, answers , and hangs up. |
| 16:02.31 | tzafrir | [TK]D-Fender, ok. clam down :-) |
| 16:02.46 | [TK]D-Fender | jochwhich is exactly what lines 27-29 of your pastebin tell it to do |
| 16:03.03 | tzafrir | s/clam/calm/ |
| 16:03.09 | [TK]D-Fender | tzafrir :I am perfectly calm. The capitalization was his as written in his pastebin :) |
| 16:03.15 | jochieng | LET ME TRY AND CHANGE IT |
| 16:03.26 | [TK]D-Fender | ^^^ see? its not me :) |
| 16:03.28 | tzafrir | jochieng, calm down ;-) |
| 16:03.50 | *** join/#asterisk Mugatu (n=mugatu@unaffiliated/Mugatu) |
| 16:03.59 | jochieng | tzafrir: being a newbie is not so easy ;) |
| 16:04.21 | tzafrir | Just don't use the caps-lock key |
| 16:04.22 | *** join/#asterisk oppassum (n=op@adsl-76-249-13-6.dsl.klmzmi.sbcglobal.net) |
| 16:04.29 | *** join/#asterisk E-bola (n=bola@cpe-76-179-4-233.maine.res.rr.com) |
| 16:04.35 | oppassum | is there a way to remotely set DND on a phone? |
| 16:04.36 | E-bola | Hey can some1 give me some help with gotoiftime |
| 16:04.41 | mikecx | [TK]D-Fender: could i possibly use AGI Background to play the ringing until they enter? |
| 16:04.47 | [TK]D-Fender | oppassum: Not in any phone I've ever seen. |
| 16:04.49 | E-bola | GotoIfTime("SIP/35101085-082fb720", "16:31-23:59|*|21|Dec*?35101085|400") in new stack |
| 16:04.49 | E-bola | [Dec 21 16:58:35] WARNING[27174]: pbx.c:4063 get_range: Invalid month 'Dec*', assuming none |
| 16:04.59 | oppassum | sweet. |
| 16:05.00 | oppassum | lol |
| 16:05.04 | E-bola | i originally used dec as month, but neither dec nor Dec works, whats the correct syntax? |
| 16:05.15 | oppassum | is there a way to remotely change group rings? |
| 16:05.16 | tzafrir | A locale problem? |
| 16:05.17 | bgat | anyone here tried to run asterisk on a Dreambox? |
| 16:05.18 | E-bola | i copied dec from the wiki so asumed that was valild..... |
| 16:05.20 | [TK]D-Fender | mikecx: I can't see how you could control it to make it stop. this is ugly. You mean to say when you use SLA to ring tied-in phones that it doesn't ring for the caller? |
| 16:05.47 | [TK]D-Fender | oppassum: Can you clarify what you mean by "group rings"? |
| 16:05.48 | mikecx | [TK]D-Fender: since i'm already picking up the line with Answer() on the way in to fax detect, no |
| 16:05.54 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 16:06.14 | [TK]D-Fender | mikecx: It would make sense that SLA should have to option of answered and forcing a ring.... |
| 16:06.23 | oppassum | *sigh* sadly i'm still stuck using trixbox..freepbx has a group rings module. From what I can tell it simple using a dial macro to dial multiple extensions |
| 16:06.24 | [TK]D-Fender | mikecx: there's no parameter option for it? |
| 16:06.39 | [TK]D-Fender | oppassum: Those terms are meaningless unfortunately. |
| 16:06.48 | mikecx | [TK]D-Fender: not if the options are the same as the MeetMe options. The SLATrunk options aren't listed in the book |
| 16:07.14 | [TK]D-Fender | oppassum: And FreePBX isn't a supported thing around here. |
| 16:07.24 | oppassum | well then don't answer fender |
| 16:07.28 | oppassum | no one's forcing you to |
| 16:08.08 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 16:08.08 | *** mode/#asterisk [+o anthm] by ChanServ |
| 16:08.13 | [TK]D-Fender | oppassum: As for what can be changed remotely the typical answer is "sure, you can do the majority of whatever you can think of, just get coding". However you have given all control of your setup to a Toaster-grade GUI |
| 16:08.17 | *** join/#asterisk |dennis| (n=dennis@200.32.233.84) |
| 16:08.36 | oppassum | i know, it's causing dilemmas all over the place. |
| 16:10.21 | E-bola | Could somebody please tell me how to specify a month in gotoiftime? |
| 16:10.22 | [TK]D-Fender | oppassum: Well you know where to go for support of what you've got and that iff you head the other way perhaps we can help |
| 16:11.02 | oppassum | what pbx do you use fender? |
| 16:11.24 | [TK]D-Fender | oppassum: Asterisk |
| 16:11.48 | Mugatu | is it possible to transcode ibound in-band DTMF to RFC2833/SIP INFO messages? (this dtmf is coming in-band from a SIP peer over ulaw) |
| 16:11.53 | oppassum | ...*bonks self on head*, duh. |
| 16:11.56 | [TK]D-Fender | E-bola: GotoIfTime("SIP/35101085-082fb720", "16:31-23:59|*|21|Dec*?35101085|400") in new stack <-- you have a very obvious typo here. You put a "*" after Dec. |
| 16:12.14 | [TK]D-Fender | oppassum: Perhaps you should rethink your question. |
| 16:12.29 | oppassum | no no, i understand |
| 16:12.38 | [TK]D-Fender | Mugatu: * naturally translates those between legs of a bridged call. |
| 16:12.45 | oppassum | freepbx, which is in use on my system, does nothing more than generate code for asterisk |
| 16:12.47 | [TK]D-Fender | oppassum: think about it and try again |
| 16:12.54 | oppassum | you do it yourself, correct? |
| 16:13.02 | oppassum | or am i missing something? |
| 16:13.02 | E-bola | [TK]D-Fender: ohh its not a typo hehe, i actualy thought u ended with a * |
| 16:13.03 | E-bola | thanks |
| 16:13.08 | [TK]D-Fender | oppassum: Yes. and I generate my own code for *, like everyone else here |
| 16:13.17 | Mugatu | [TK]D-Fender: OK, I'll keep looking didn't appear to be happening for me here, wanted to make sure it was possible |
| 16:13.23 | oppassum | so you did actually answer my question |
| 16:13.24 | [TK]D-Fender | E-bola: Not according to the app's instructions. |
| 16:14.06 | [TK]D-Fender | oppassum: This channel is for everyone who is actually using Asterisk. You are not. Trixbox is using Asterisk and you have little to do with it. |
| 16:14.32 | E-bola | [TK]D-Fender: Do you have a tip for me for the future where to get the best information? I tried the wiki, which didnt help |
| 16:14.41 | E-bola | and i did core show application GotoIfTime |
| 16:14.43 | Qwell | [TK]D-Fender: ab.. |
| 16:14.51 | E-bola | which didnt reveal much but just told me to look at examples |
| 16:14.55 | E-bola | which didnt use months... |
| 16:15.02 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 16:15.05 | *** join/#asterisk hfb (n=hfb@pool-71-106-219-180.lsanca.dsl-w.verizon.net) |
| 16:15.20 | [TK]D-Fender | Qwell: Yes? |
| 16:15.35 | Qwell | used |
| 16:15.39 | Qwell | or, using |
| 16:15.40 | [TK]D-Fender | E-bola: "show application gotoiftime" <-- |
| 16:16.00 | [TK]D-Fender | Qwell: Don't follow you... |
| 16:16.07 | Qwell | "Trixbox is using Asterisk" |
| 16:16.14 | Qwell | s/u/abu/ |
| 16:16.19 | Qwell | jbot++ |
| 16:16.22 | [TK]D-Fender | Qwell: Whats wrong with that statement |
| 16:16.37 | Qwell | nevermind... |
| 16:16.38 | drmessano | Qwell |
| 16:16.47 | Qwell | drmessano |
| 16:16.48 | syzygyBSD | lol |
| 16:16.50 | drmessano | Asterisk isn't Trixbox? |
| 16:17.04 | drmessano | But.... but... |
| 16:17.16 | E-bola | :P |
| 16:17.48 | syzygyBSD | when is asterisk 1.6 comming out? |
| 16:18.26 | jochieng | tzafrir: ok it has now got to demo-congrats and hangs up, its not easy making a dailed phone ring after all.. :-( |
| 16:18.58 | *** join/#asterisk iratik (n=itariki@12-226-116-3.client.mchsi.com) |
| 16:18.58 | iratik | http://pastie.caboo.se/131352 |
| 16:18.59 | file | syzygyBSD: yes. |
| 16:19.02 | iratik | Whats going on herE? |
| 16:19.03 | syzygyBSD | dial(sip/myextension) |
| 16:19.10 | syzygyBSD | file: ;) |
| 16:19.23 | tzafrir | jochieng, the demo is on extension s in the demo context |
| 16:19.56 | jochieng | tzafrir: i just want to ame phones rings each other |
| 16:20.12 | jochieng | tzafrir: any sccripts somewhr u can direct me to learn more |
| 16:21.03 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net) |
| 16:21.19 | tzafrir | exten => 400,1,Dial(SIP/2000) |
| 16:21.21 | *** part/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 16:21.36 | tzafrir | This means that if you dial 400 you get to SIP/2000 |
| 16:22.05 | jochieng | tzafrir: does exten => 2000,1,Dial(SIP/2000) make more sense? |
| 16:22.14 | Mugatu | [TK]D-Fender: you are correct on dtmf transcoding, of course. Either I was completely crazy, but after fiddling with dtmfmode, forcing ulaw to my peer, and a few iterations of 'sip reload', restarting asterisk finally seemed to do the trick |
| 16:22.29 | tzafrir | jochieng, yes. Should work. |
| 16:22.31 | syzygyBSD | Time to go have more fun rewiring :9 |
| 16:22.55 | jochieng | tzafrir: thanks alot now i can begin from somewhr -- thanks alot for the info |
| 16:23.00 | iratik | Whats going on here? http://pastie.caboo.se/131352 ... ... when i dial directly to the trunk its working...?!? |
| 16:23.23 | tzafrir | jochieng, also: exten => 200,1,Dial(SIP/${EXTEN}) |
| 16:23.48 | tzafrir | oops: 2000, not 200 |
| 16:24.01 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 16:24.14 | jochieng | tzafrir: same thing with the solution i asked you about earlier about calling the same ext.? |
| 16:24.26 | mikecx | would there be any problem in having people answer the phone for faxes and then manually transfer them to exten => 999,1,Dial(Zap/1)? |
| 16:25.14 | tzafrir | mikecx, that has to happen fast enough |
| 16:25.26 | tzafrir | before the sending fax gives up |
| 16:26.01 | iratik | I'm getting "Everyone is busy/congested at this time " from every dial request |
| 16:26.03 | mikecx | tzafrir: we do it now for older fax machines that don't send the fax tone but that's with our old system |
| 16:26.05 | iratik | what might be wrong? |
| 16:26.34 | [TK]D-Fender | mikecx: No, thats perfectly viable... except I'm not sure how SLA will handle that as it passes through meetme for part of it |
| 16:26.41 | [TK]D-Fender | mikecx: In theory it might work. |
| 16:26.49 | tzafrir | iratik, please be more specific |
| 16:27.08 | mikecx | [TK]D-Fender: that's my concern too but I guess I won't know until I try |
| 16:27.22 | tzafrir | That message is FreePBX's dialplan way of telling you that "something was wrong" |
| 16:27.28 | tzafrir | set verbose 3 |
| 16:27.36 | tzafrir | and look at the generated trace |
| 16:27.44 | [TK]D-Fender | tzafrir : No, that is a normal * message |
| 16:28.06 | *** join/#asterisk teknoprep (n=tekno@unaffiliated/teknoprep) |
| 16:28.07 | tzafrir | hmm... mixing messages, I guess |
| 16:28.16 | [TK]D-Fender | iratik: Please pastebin the entire output of your call at verbose 10. |
| 16:28.23 | *** join/#asterisk Maliuta (n=nikolai@119.11.96.227) |
| 16:29.30 | [TK]D-Fender | tzafrir : a nearly useless one mind you :) |
| 16:29.40 | iratik | <PROTECTED> |
| 16:29.46 | iratik | but... i think i found out whats wrong |
| 16:30.55 | [TK]D-Fender | iratik: Can't trust anything there until you enable SIP debug |
| 16:31.17 | *** join/#asterisk Le_Vert (n=gandalf@adsl02.metz.linbox.com) |
| 16:31.25 | Le_Vert | hi :) |
| 16:31.25 | *** part/#asterisk Mugatu (n=mugatu@unaffiliated/Mugatu) |
| 16:31.40 | Le_Vert | could someone give me a little help to figure out why call deflection doesn't work ? |
| 16:31.50 | Le_Vert | i'm using a 4 channel eicon diva card |
| 16:32.10 | Le_Vert | "/usr/lib/divas/divactrl mantool -c 1 -r"Config/Supplementary Values/SSFeatures"" gives |
| 16:32.15 | Le_Vert | -w------hit-[Config\Supplementary Values\SSFeatures ] = 0x10 |
| 16:32.27 | Le_Vert | so I guess call deflection is enable on the hardware side |
| 16:32.44 | Le_Vert | but capiinfo | grep -i 'Defle' doesn't return anything |
| 16:33.32 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-24-106-241-121.central.biz.rr.com) |
| 16:36.08 | Le_Vert | any pointer would be greatly appreciated :) |
| 16:36.52 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 16:38.28 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 16:41.21 | *** join/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/Zeeek) |
| 16:41.40 | *** join/#asterisk inv_arp[work] (n=junya@c-76-26-24-197.hsd1.fl.comcast.net) |
| 16:42.32 | Zeeek | http://VoipUsersConference.org is starting on the hour. Kerry Garrison will be on to talk about what they're doing at Fonality about the script issues that made the scandal sheets last week. |
| 16:43.09 | Zeeek | All asterisk users are welcome to join us on IRC #voip-users-conference and post questions and comment there or join the live call |
| 16:43.09 | Qwell | Zeeek: 15m? |
| 16:43.43 | Qwell | Zeeek: tell Kerry to get on IRC |
| 16:43.59 | Qwell | please :D |
| 16:44.01 | Zeeek | The IRC channel is open, the Talkshoe call in server opens in a few minutes and can accept hundreds of callers |
| 16:44.10 | Zeeek | I'm not Kerry's boss :) |
| 16:44.41 | Qwell | no, but you should totally suggest it :) |
| 16:46.03 | Zeeek | I emailed just now |
| 16:48.54 | Zeeek | To be in on this call |
| 16:48.55 | Zeeek | Call in now: SIP 123@66.212.134.192 |
| 16:49.03 | Zeeek | enter 22622# 1# |
| 16:49.10 | Zeeek | and you're there with us |
| 16:49.34 | Zeeek | the room is non-smoking, but drugs are freely available |
| 16:50.27 | *** join/#asterisk Winkie (n=urmom@general-kt-195.t-mobile.co.uk) |
| 16:52.16 | *** join/#asterisk ice_croft (n=nolan@213.132.86.246) |
| 16:55.34 | *** part/#asterisk ice_croft (n=nolan@213.132.86.246) |
| 16:56.35 | jochieng | hello -- i am running asterisk 1.2 on debian etch and i wld like to upgrade to asterisk 1.4 with my configs intact, how is this possible? |
| 16:57.00 | iratik | is there a free text to speech engine? |
| 16:57.02 | Qwell | jochieng: you'll need to read UPGRADE.txt in the 1.4 source dir |
| 16:57.16 | iratik | a free alternative to cepestral? |
| 16:58.19 | Corydon76-dig | Festival |
| 16:58.39 | Qwell | festival works, but it isn't great. It's certainly acceptable though |
| 16:58.48 | Le_Vert | nobody could confirm me that capiinfo should return Call Deflection |
| 16:59.00 | Le_Vert | It's really hard to figure out what's wrong |
| 16:59.15 | Le_Vert | that would really help me to understand what's ok and what's wrong |
| 16:59.32 | Le_Vert | I enabled call deflection at the hardware level with divactrl |
| 16:59.42 | Le_Vert | that looks like being ok |
| 17:00.05 | Le_Vert | but capiinfo doesn't tell me it supports call deflections then.... |
| 17:02.04 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 17:02.31 | *** part/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/Zeeek) |
| 17:08.36 | [TK]D-Fender | jochieng: Your configs are all in 1 folder, and you need only copy them. |
| 17:08.51 | [TK]D-Fender | jochieng: And keep in mind config file changes between versions |
| 17:13.47 | *** part/#asterisk amessina (n=amessina@h-66-166-108-202.chcgilgm.covad.net) |
| 17:23.58 | mikecx | shizer, i forgot which port on my card is fxo and which is fxs |
| 17:24.04 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 17:27.23 | *** join/#asterisk dkatz333 (i=121mhz@66.sub-75-223-121.myvzw.com) |
| 17:27.41 | mikecx | the fxs ones are the closest to the back of the computer, the fxo's are farthest away, but which is which on the back connector thingy |
| 17:27.46 | dkatz333 | Afternoon all. |
| 17:28.08 | dkatz333 | Any zaptel experts onboard today? |
| 17:33.02 | *** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 17:38.09 | *** join/#asterisk nny_1 (n=scott@64.203.239.83.static-pool-4.pool.hargray.net) |
| 17:38.51 | *** join/#asterisk ldsjohn (n=darksage@216-70-238-106.static-ip.telepacific.net) |
| 17:39.44 | ldsjohn | can anyone point me in the right direction, I want to have one person record a message and 30 people login with a pin and listen, it used to be done on the old hardware pbx with virtual mailboxes but I can't get that to work with asterisk |
| 17:42.06 | *** join/#asterisk dkatz334 (i=121mhz@157.sub-75-222-25.myvzw.com) |
| 17:42.33 | *** join/#asterisk joelsolanki (i=joelsola@220.224.75.145) |
| 17:42.41 | ldsjohn | im resorting to trying to use Record( ) to record a file and then Playback to play it but playback isnt working for me? |
| 17:42.59 | dkatz334 | Okay, sorry for the disconnect... My question is for a zaptel expert, anybody want to field this one? -> |
| 17:44.48 | dkatz334 | I couldn't compile zaptel 1.4.6 with hdlc support because line 6399 of zaptel-base.c shows "skb->mac.raw = skb->data;" apparently mac doesn't exist in sk_buff anymore. I don't think mac.raw is used anywhere else. Any danger in commenting this line out? It compiles without it. |
| 17:45.35 | dkatz334 | kernel version is 2.6.23 bundled kernel with fc8 |
| 17:47.28 | *** join/#asterisk ar3dam (n=fl3pix@189.156.231.173) |
| 17:47.56 | *** join/#asterisk mihinomenest (i=31iI@66.255.220.17) |
| 17:47.57 | *** join/#asterisk MacWinner (n=chatzill@74-33-175-68.dsl1-merch.roc.ny.frontiernet.net) |
| 17:49.18 | MacWinner | so tmobile has a plan with unlimited incoming/outgoing to a specific number. if setup an asterisk box with a DID, and have it autoforward to my cell phone, can i have the callerid of the caller forwarded even though the incoming call is from my DID (and presumably 0 rated)? |
| 17:49.43 | MacWinner | yes, i'm trying to game the system |
| 17:50.30 | dkatz334 | MacWinner, whats your outbound method? |
| 17:50.50 | dkatz334 | I assume your inbound did is SIP, right? |
| 17:50.54 | MacWinner | i'll setup a SIP or IAX peer |
| 17:51.10 | MacWinner | i'm flexible.. whichever works |
| 17:51.23 | dkatz334 | if the SIP/IAX peer allows you to set your CID to an arbitrary number, sure. |
| 17:51.29 | jwh | you'll need a telco that will propogate your cid |
| 17:51.31 | dkatz334 | I do the same with an 888 |
| 17:52.00 | MacWinner | Do you think tmobile will use the CID for billing purposes? |
| 17:52.01 | dkatz334 | users call the 888, it rings at my desk, if I don't answer it rings my cell with the CID of the original caller. |
| 17:52.12 | jwh | *nod* |
| 17:52.28 | dkatz334 | Billing, how? |
| 17:52.42 | ar3dam | hi there, what is named for example, if i want call long distante... i live in usa, if i wanna call to mexico i need type 01152867, but i dont wanna use this manned, i prefered type 7 and add the phone on mexico. |
| 17:53.20 | MacWinner | so let's say my DID is 1234, and somecalls 1234 and then 1234 rings my cell (but sets the CID), tmobile will probably use 1234 as the originating caller right? |
| 17:53.37 | dkatz334 | exten => _7X.,1,Dial(METHOD/01152867${EXTEN}) should do the trick |
| 17:53.39 | jwh | if they trust the telco enough, yes |
| 17:53.51 | dkatz334 | Make sure you replace METHOD with however you usually dial |
| 17:54.32 | dkatz334 | Yeah TMO just gets the CID info from your telco. |
| 17:54.34 | MacWinner | cause with tmobile i've configured the 1234 as the number i want to be unlimited incoming/outgoing |
| 17:54.51 | joelsolanki | anybody's g729 working on asterisk 1.4 on celeron ? |
| 17:54.52 | dkatz334 | OIC |
| 17:54.53 | *** join/#asterisk Maliuta_ (n=nikolai@119.11.96.189) |
| 17:55.04 | dkatz334 | I get it now. |
| 17:55.12 | MacWinner | yep, so assuming tmobile passes the CID, will they still use 1234 as the caller for unlimited purposes? |
| 17:55.22 | MacWinner | (sorry, i didn't explain clearly at first) |
| 17:55.27 | joelsolanki | i installed g729 p3 version on celeron but when shutdown process it tells kernel panic |
| 17:55.33 | joelsolanki | any hints plz |
| 17:55.47 | ar3dam | dkatz334, how find more information or how is nammed this? |
| 17:56.14 | dkatz334 | It's part of the dial plan. |
| 17:56.24 | MacWinner | dkatz334: did you use anything like freepbx to do this 888 configuration you have? |
| 17:56.29 | ar3dam | Oh, tks :D .. |
| 17:56.37 | dkatz334 | _7X. means anything which has a 7 starting it and followed by any number of digits. |
| 17:56.45 | joelsolanki | if i remove g729 codec then it shutdown normally. |
| 17:56.59 | dkatz334 | no Asterisk and I edited the dialplan... did use PICO to edit istead of VI :) |
| 17:57.28 | dkatz334 | No one has an answer for my zaptel question, eh? |
| 17:58.02 | MacWinner | dkatz334: what's your zaptel question? (not that i would know the answer anyway :) |
| 17:58.34 | dkatz334 | I couldn't compile zaptel 1.4.6 with hdlc support because line 6399 of zaptel-base.c shows "skb->mac.raw = skb->data;" apparently mac doesn't exist in sk_buff anymore. I don't think mac.raw is used anywhere else. Any danger in commenting this line out? It compiles without it. |
| 17:58.58 | MacWinner | if it compiles, it must work ;) |
| 17:59.12 | jwh | not really ;) |
| 17:59.55 | dkatz334 | yeah right |
| 18:00.16 | dkatz334 | My CS students in "Intro to Programming" try to argue that with me every week! |
| 18:01.11 | *** join/#asterisk michael-i (n=michael-@host-170-68-220-24.midco.net) |
| 18:02.42 | MacWinner | yeah.. it's classic |
| 18:06.02 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 18:06.04 | *** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 18:06.25 | *** join/#asterisk Winkie (n=urmom@general-ld-220.t-mobile.co.uk) |
| 18:08.37 | MacWinner | is there a good site with lots of example asterisk dialplans? |
| 18:10.51 | dkatz334 | Asteriskguru.com is very good |
| 18:13.42 | *** join/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 18:14.11 | MacWinner | my zttest on ztdummy consistently gives the following accuracies: 99.963379% 99.938965% |
| 18:14.37 | mocker | MacWinner: That's not good. |
| 18:14.38 | mocker | ;) |
| 18:14.43 | MacWinner | i was reading that there may be some rtc bug or something with the linux kernel? |
| 18:15.01 | MacWinner | or something with the code in ztdummy that is giving inaccurate info? |
| 18:15.21 | MacWinner | using a 2.6.15 kernel SMP |
| 18:15.33 | MacWinner | zaptel 1.4.5 |
| 18:16.46 | MacWinner | anyone run into this issue? |
| 18:17.22 | *** join/#asterisk curtn (n=curtis@cl-451.trn-01.it.sixxs.net) |
| 18:17.32 | jwh | 99.975586% here |
| 18:17.35 | jwh | ztdummy, works fine |
| 18:17.38 | curtn | hi all |
| 18:18.43 | MacWinner | jwh, which kerrnel and zaptel versions are u running? |
| 18:18.54 | jwh | MacWinner: |
| 18:19.01 | jwh | uk0# uname -sr |
| 18:19.01 | jwh | FreeBSD 6.2-STABLE |
| 18:19.14 | jwh | zaptel-1.4.6_2 |
| 18:19.59 | jwh | probably not applicable ;) |
| 18:20.59 | curtn | it seems to be very difficult to obtain good voice quality on SPA3102.. I still have a very bad echo |
| 18:23.07 | WilliamK | interesting, I wonder if Linksys is using the cisco designs or the sipura designs in that box |
| 18:23.13 | WilliamK | I know the sipura's work well |
| 18:24.27 | curtn | WilliamK: my PSTN line is from France Telecom |
| 18:24.47 | curtn | I stil have a doubt on impedance/capacity tuning... |
| 18:25.17 | curtn | input/output gain is difficult to tune... |
| 18:25.31 | curtn | echo cancellation doesn't work very well... |
| 18:27.14 | curtn | I understand why only ISDN is used on all "professional" VoIP installation.. |
| 18:27.30 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 18:28.06 | WilliamK | saves alot of time, but I prefer to use ISDN PRI over T1s |
| 18:29.12 | curtn | through asterisk, the quality with my SIP provider (keyyo) is very good |
| 18:29.13 | *** join/#asterisk r0d3nt (n=astrutt@foster.stonedcoder.org) |
| 18:29.21 | dkatz334 | Anybody have zaptel working with a split data/voice t1 with kernel 2.6??? |
| 18:29.42 | dkatz334 | everytime I run ifconfig on the interface is seg faults. |
| 18:32.34 | WilliamK | curt... have you seen this doc yet? http://linksys.custhelp.com/cgi-bin/linksys.cfg/php/enduser/std_adp.php?p_faqid=5167&lid=6870369263B13 |
| 18:33.11 | curtn | WilliamK: yes |
| 18:33.53 | WilliamK | k, just figured I'd mention it :) |
| 18:36.05 | curtn | WilliamK: do I have another solution to cancel echo with asterisk ? |
| 18:37.26 | syzygyBSD | echo is caused by bad lines a lot of the time |
| 18:37.35 | *** part/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 18:37.40 | tzafrir | dkatz334, do you get something in the kernel logs? |
| 18:38.59 | [TK]D-Fender | curtn: Analog can work fine with normal HWEC. ISDN sucks as bad on the EC side if you're not covered |
| 18:39.19 | [TK]D-Fender | curtn: www.voxilla.com <= go check out their forums on tweaking the SPA-3102 for echo |
| 18:39.47 | curtn | [TK]D-Fender: what is HWEC ? |
| 18:40.00 | [TK]D-Fender | curtn: And you may want to pick some specific firmware as different revisions have different performance |
| 18:40.14 | [TK]D-Fender | curtn: HardWare Echo Cancellation |
| 18:40.25 | WilliamK | curtn, asterisk itself has ways to cancel but if you're going off hook using the SPA-3102 it kinda defeats the purpose |
| 18:40.57 | *** join/#asterisk holiday_42 (n=no@spike.wcta.net) |
| 18:41.01 | *** join/#asterisk leanshen (n=bob@ool-43546a71.dyn.optonline.net) |
| 18:41.09 | curtn | [TK]D-Fender: i have thhe last firmware.. maybe not the best one |
| 18:45.52 | [TK]D-Fender | curtn: Exactly. Latest != best |
| 18:46.03 | [TK]D-Fender | curtn: Go over the forums carefully |
| 18:47.14 | curtn | [TK]D-Fender: on the forums I can see some people having good result with very high gains... I'm not convinced |
| 18:47.36 | curtn | their conditions seems to be different than mine |
| 18:48.44 | curtn | in general, is the "Disconnect Tone" usefull ? or not ? |
| 18:48.45 | WilliamK | the other thing I noticed that was odd is it seems that Sipura doesn't manufacturer the boxes with the right ring frequence/voltage (on the US boxes anyway); looks like they recently corrected their "defaults" to match bellcore |
| 18:49.19 | nny_1 | god why? why must i be forsaken |
| 18:49.46 | nny_1 | wheres my sword, i must die with honor |
| 18:50.45 | WilliamK | you could always downgrade to Win3.1 |
| 18:50.56 | WilliamK | I keep all the files on a server nowdays just for fun :) |
| 18:51.06 | holiday_42 | only 6 disks :) |
| 18:51.28 | curtn | WilliamK: ring frequence/voltage is fully configurable on my SPA-3102.. it should match France Telecom.. |
| 18:52.01 | [TK]D-Fender | nny_1: Here... you can borrow mine :) - http://gallery.aocomputing.net/index.php?album=2007-03-02+Oni+Forge+Bushi |
| 18:52.10 | coppice | people who use high gains are usually operating their system in almost permanent clipping. many do this, despite it sounding bloody awful |
| 18:52.22 | Qwell | coppice: I set my gains to 30 |
| 18:52.28 | WilliamK | curtn, my comment was to the fact that they didn't match bellcore standards by default prior |
| 18:53.08 | WilliamK | hey coppice, have you ever gotten an SPA-2002 to work on faxes? |
| 18:53.23 | coppice | never used one |
| 18:53.43 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 18:53.43 | *** mode/#asterisk [+o russellb] by ChanServ |
| 18:54.07 | MacWinner | are my zttest result problems something to do with 1000khz clock vs 1024khz clock? |
| 18:54.21 | WilliamK | k, was just curious... I keep seeing the unknown codec 100 messages and when I think I had that resolved it doesn't work at all.... gave up and ordered an SPA-2102 to hope that T.38 works better than just pass-through |
| 18:54.31 | MacWinner | like is that skewing the accuracy of zttest? |
| 18:54.43 | MacWinner | i vaguely remember a problem with this before |
| 18:58.10 | coppice | WilliamK: there are numerous ATAs, many with something in their config menus about FAX, which have zero chance of ever working with FAX due to weird things they do. See http://www.soft-switch.org/foip-with-real-atas.html for some weird effects |
| 18:59.04 | iCEBrkr | haha |
| 18:59.12 | coppice | I must be getting famous. www.soft-switch.com is a mass of links to other people's FAX software :-) |
| 18:59.39 | iCEBrkr | No sane person should be trying to send FAXes over VoIP channels. However, in development work we do some insane things as part of our system investigations. |
| 18:59.43 | iCEBrkr | awesome |
| 18:59.56 | iCEBrkr | coppice: Micro-celebrity |
| 19:01.38 | MacWinner | am i correct in saying that ztdummy is not usef if i'm setting up a basic call forwarder where someone calls my DID and then it rings my cell? |
| 19:01.57 | [TK]D-Fender | MacWinner: Nothing to do with that |
| 19:02.20 | MacWinner | what's the general rule where ztdummy is used? when sounds are being mixed? |
| 19:02.34 | [TK]D-Fender | MacWinner: This is documented to death already |
| 19:02.48 | [TK]D-Fender | MacWinner: A zaptel timingf source is needed for MeetMe, and IAX2 Trunking |
| 19:02.49 | iCEBrkr | To death! |
| 19:03.26 | iCEBrkr | [TK]D-Fender: I guess I don't have room to talk, I didn't know a timing source was needed for IAX trunking. |
| 19:03.29 | iCEBrkr | :( |
| 19:03.56 | MacWinner | i heard about meetme.. didn't know iax required it too |
| 19:04.44 | *** join/#asterisk emist (n=emist@unaffiliated/emist) |
| 19:05.02 | jer | anyone recommend a reliable place to buy did's ? (particularly DIDs for NA) |
| 19:05.16 | iCEBrkr | :'( |
| 19:05.56 | iCEBrkr | I musta had ztdummy working the last time I trunked 2 asterisk boxes. |
| 19:06.50 | MacWinner | iCEBrkr: are you using IAX trunking now? |
| 19:07.04 | iCEBrkr | MacWinner: No, I mean.. well. I want to |
| 19:07.16 | iCEBrkr | This was years ago when I first started tinkering with Asteriks. |
| 19:07.29 | MacWinner | do you mind sharing your zttest results? |
| 19:07.37 | MacWinner | (are u using digium hardware?) |
| 19:07.51 | iCEBrkr | Now that my primary asterisk box is located at home behind nat on a dynamic IP.. It's difficult to have a softphone on my laptop when I'm remote. |
| 19:08.27 | iCEBrkr | So I was going to run a small instance of Asterisk on my hosted virtual machine which has a public IP |
| 19:08.53 | iCEBrkr | So then I just connect both asterisk machines via IAX and be done with it :) |
| 19:09.06 | iCEBrkr | MacWinner: I haven't touched this stuff in years. |
| 19:09.24 | MacWinner | yeah, i wanted to be able to trunk with a provider while i'm behind NAT |
| 19:09.55 | MacWinner | my zttest results don't look good enough though supposedly |
| 19:09.55 | _x86_ | MacWinner: IAX2 is your friend |
| 19:10.18 | MacWinner | _x86_: but ztdummy seems to be my enemy :) |
| 19:10.20 | iCEBrkr | IAX is NAT friendly.. Which is pretty much the only reason for this trunking |
| 19:10.21 | _x86_ | zttest has nothing at all to do with NAT |
| 19:10.23 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:10.26 | _x86_ | MacWinner: why do you need timing? |
| 19:10.38 | _x86_ | MacWinner: you trying to do conferencing or something? |
| 19:10.49 | MacWinner | _x86_: TK-fender says you need zaptel timing for IAX2 trunking |
| 19:12.01 | _x86_ | as long as you have ztdummy you'll be fine |
| 19:12.01 | MacWinner | conferencing not such a big deal.. just trying to get some call forwarding going |
| 19:12.01 | _x86_ | if you have problems, get a X110P ;) |
| 19:12.42 | _x86_ | i've done IAX2 without ztdummy before (not in trunk mode though) |
| 19:12.56 | _x86_ | how many calls are you going to have going out at once to your ITSP? |
| 19:13.07 | _x86_ | at any one given time |
| 19:14.23 | *** join/#asterisk apocn (n=htejeda@unaffiliated/apocn) |
| 19:14.43 | MacWinner | maybe 2-3 |
| 19:15.11 | _x86_ | how much bandwidth do you have? |
| 19:15.16 | apocn | Hello, Im trying to register my asterisk with a softswitch and specify the user/pass/host and reload asterisk I get the following message : Got SIP response 423 "Interval Too Brief". Any help? |
| 19:15.22 | *** join/#asterisk fnordus (n=dnall@24.84.160.227) |
| 19:15.27 | MacWinner | i found this article about ztdummy.. the guy seems to have the same exact numbers as i do for zttest.. he seems to claim that the zttest numbers are not quite accurate: http://www.opensubscriber.com/message/asterisk-bsd@lists.digium.com/7252809.html |
| 19:15.34 | _x86_ | MacWinner: trunking with only 2 or 3 calls is not going to save you much bandwidth |
| 19:15.50 | _x86_ | MacWinner: you can do iax2 without trunking you know :) |
| 19:15.51 | MacWinner | _x86_: not sure if the exact amount of bandwidth, but it's a lot (dedicated server with 1and1) |
| 19:15.51 | Qwell | bsd? don't bother |
| 19:16.04 | _x86_ | yeah, just dont do trunking |
| 19:16.07 | _x86_ | you'll be fine |
| 19:16.13 | _x86_ | even without ztdummy |
| 19:16.22 | MacWinner | _x86_: oh, shiz, i forgot :) |
| 19:16.24 | MacWinner | thanks! |
| 19:16.34 | MacWinner | trunking just for saving bandwidth? |
| 19:16.39 | _x86_ | yes |
| 19:16.53 | MacWinner | coolio |
| 19:17.11 | _x86_ | trunks multiple calls along the same signalling path |
| 19:18.11 | apocn | can someone help me? |
| 19:19.15 | michael-i | I have several incoming DIDs defined for my ISDN connection and I want to catch all other numbers with an X! pattern. Currently calls to the defined DIDs/MSNs work, but the catch-all is failing rejecting the calls due to an unfound extension. My logic is here (http://pastebin.ca/827408). Any feedback would be greatly appreciated. |
| 19:22.38 | MacWinner | _x86_: thanks for the info.. i ran a patched version of the zttest which seems to give me better results. if i understand correctly, even though my short term zttest results are not very good, my long term ones are very good. does that guys post make sense to you? |
| 19:23.00 | *** join/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
| 19:23.37 | redder86 | How does one debug these? : Dec 21 14:21:53 WARNING[25378]: channel.c:780 channel_find_locked: Avoided initial deadlock for '0x9078128', 9 retries! |
| 19:32.16 | holiday_42 | Michael-i: I cant see your pastebin... anyway does the pattern matching in your dialplan start with _? |
| 19:32.32 | nny_1 | lol so if i have a network, millions of miles away, that is obviously disparate, is there any temp fix to still getting calls through even with packet loss? |
| 19:32.52 | nny_1 | right now it thinks the sip peers are unavailable.. my luck it's some god awful 90s hub between them all |
| 19:33.19 | michael-i | holiday_42: http://pastebin.ca/827408 : yes |
| 19:34.32 | *** join/#asterisk apocn (n=htejeda@unaffiliated/apocn) |
| 19:35.08 | apocn | Hello, Im trying to register my asterisk with a softswitch and specify the user/pass/host and reload asterisk I get the following message : Got SIP response 423 "Interval Too Brief". Any help? |
| 19:37.53 | *** join/#asterisk angom (n=angom@200.38.31.239.dsl.dyn.telnor.net) |
| 19:38.20 | _x86_ | MacWinner: doesn't matter, you dont even need ztdummy if you dont trunk |
| 19:39.42 | *** join/#asterisk chisefu|afk (n=brett@24.68.237.193) |
| 19:39.55 | chisefu|afk | Is there anyone available for help? |
| 19:40.00 | [TK]D-Fender | michael-i: Pastebin a failed call at verbose 10 |
| 19:40.01 | [TK]D-Fender | ~pb |
| 19:40.02 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 19:40.03 | [TK]D-Fender | ^^^^^^^ |
| 19:40.26 | [TK]D-Fender | apocn: The registration period is unacceptably low for one side or the other. |
| 19:42.07 | *** part/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
| 19:42.41 | chisefu|afk | what's the AsteriskNOW command line network tool? do we have to use ifconfig? |
| 19:44.59 | michael-i | [TK]D-Fender: this is someone else's box, so I can't do this myself. Just making sure my little bit of logic there was correct. |
| 19:46.21 | tzafrir | chisefu|afk, #asterisknow or #rpath |
| 19:46.32 | chisefu|afk | ok sorry |
| 19:56.43 | [TK]D-Fender | michael-i: And you could be showing us stuff that never comes into play. |
| 19:56.54 | [TK]D-Fender | michael-i: Ask when you have a chance to do something about it. |
| 19:59.12 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 20:00.58 | michael-i | [TK]D-Fender: outgoing works, incoming works for the defined destinations. The only failures are with destinations falling into the _X! extension. Just making sure nothing obvious was wrong there. |
| 20:02.02 | *** join/#asterisk jmesquita (n=jmesquit@201.7.117.114) |
| 20:02.36 | *** part/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 20:03.26 | [TK]D-Fender | michael-i: I have no proof it even uses the contexts you showed us. When you say things aren't working and I can't see everything my trust level reaches 0 almost instantly |
| 20:04.55 | michael-i | [TK]D-Fender: sorry you're so untrusting, thought I was making things easier by narrowing down the problem instead of throwing my hands in the air and posting 20 pages of logs and .conf files |
| 20:05.39 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:05.39 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 20:08.47 | [TK]D-Fender | michael-i: Oh no... showing the configs is a GREAT thing... seeing that they MATTER almost supercedes that however :) |
| 20:09.26 | [TK]D-Fender | michael-i: Sorta like when Rome set out 5000 soldiers to guard the city from the Huns.... they shouldn't have been looking the OTHER WAY |
| 20:10.13 | chisefu|afk | that's so FUNNY haha 5000 soldiers OH MAN they totally should have TURNED around! |
| 20:10.48 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:10.48 | *** mode/#asterisk [+o lmadsen] by ChanServ |
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| 20:46.48 | _x86_ | lmadsen: heya :) |
| 20:48.17 | lmadsen | howdy :) |
| 20:48.57 | apocn | Hello, Im trying to register my asterisk with a softswitch and specify the user/pass/host and reload asterisk I get the following message : Got SIP response 423 "Interval Too Brief". Any help? |
| 20:49.56 | dacs | i am havining problem with ATA config |
| 20:52.55 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 20:53.01 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 20:53.33 | michael-i | apocn: some info, http://bugs.digium.com/view.php?id=7254 |
| 20:53.45 | *** join/#asterisk Cresl1n (n=matt@nat/digium/x-94caefee76138fea) |
| 20:53.45 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 20:53.55 | Qwell | Cresl1n: ! |
| 20:54.03 | Cresl1n | Qwell: !!! |
| 20:59.17 | twisted | Cresl1n |
| 20:59.27 | Cresl1n | no way! it's twisted! |
| 20:59.42 | twisted | yup |
| 21:01.06 | *** join/#asterisk craigk (n=ckowald@58.174.150.119) |
| 21:01.36 | Qwell | twisted: I can start drinking again on the 1st :p |
| 21:01.40 | twisted | oh? |
| 21:01.43 | Qwell | s/can/will/ |
| 21:01.44 | twisted | i didn't know you couldn't |
| 21:02.03 | Qwell | no, not really "couldn't", just "wouldn't |
| 21:02.07 | twisted | heh |
| 21:02.23 | twisted | jbot: fkuc you |
| 21:02.25 | twisted | s/fkuc/fuck |
| 21:02.26 | Corydon76-dig | Qwell: why'd you stop? |
| 21:02.33 | Qwell | twisted: trailing / |
| 21:02.41 | twisted | you don't need trailing / |
| 21:02.45 | Qwell | Corydon76-dig: no particular reason |
| 21:02.48 | Qwell | twisted: you do with jbot |
| 21:02.48 | twisted | only if you are doing operations |
| 21:02.49 | *** join/#asterisk BCS-Satori (n=somewher@75.148.21.113) |
| 21:03.02 | twisted | qwell: o reeallie? |
| 21:03.07 | twisted | s/reallie/really |
| 21:03.11 | twisted | s/reallie/really/ |
| 21:03.13 | twisted | guess so |
| 21:03.26 | twisted | but see what i really meant? :P |
| 21:03.43 | BCS-Satori | I having having isssues doing a make install no asterisk-addons 1.4.5 i keep recieving this error: http://rafb.net/p/jwpkdn67.html |
| 21:04.16 | Qwell | BCS-Satori: there's an open bug |
| 21:04.43 | BCS-Satori | Qwell: ah any more information I can find on it, or should i use an older verison? |
| 21:04.56 | Qwell | there's a workaround patch on the bug |
| 21:05.18 | dacs | twisted: have you ever worked on Cisco ATA 186 |
| 21:05.38 | BCS-Satori | Qwell: would you happen to have the link or able to point me in the correct direction? |
| 21:05.51 | Qwell | no, you'll have to search for it on bugs.digium.com |
| 21:07.50 | mikecx | on a card exactly like this one: http://www.openvox.com.cn/productsFile/A800P2.jpg (fxo and fxs are in the same spots) is the one nearest or furthest from the empty spots where the lines from the phone company come in? |
| 21:14.28 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 21:14.49 | *** part/#asterisk dacs (n=haiger@unaffiliated/dacs) |
| 21:17.17 | BCS-Satori | Qwell: i see the bug report from 5-1-07 but i see no workaround =/ |
| 21:19.14 | BCS-Satori | Qwell: if we did a ln -s libchan_h323.1.0.1 libchan_h323.so.1.0.1 it would resolve it? are they the same files? |
| 21:22.49 | BCS-Satori | Qwell: yep, that totally worked |
| 21:23.27 | MacWinner | can you match an incoming callerID in sip.conf or iax.conf and then send to a specific context based upon CID? or is this not standard? ie, should all CID matching happen in the dialplan? |
| 21:23.45 | Qwell | MacWinner: do CID matching in the dialplan |
| 21:23.48 | Qwell | you can do something like |
| 21:24.13 | Qwell | exten => 5551212/5552233,Goto(somecontext) |
| 21:24.23 | Qwell | if 5552233 calls 5551212 |
| 21:25.14 | MacWinner | sorry, i'm not familiar with that notation... 5551212/5552233 matches if 5552233 calls 5551212? |
| 21:25.21 | MacWinner | that's pretty damn cool |
| 21:25.24 | Qwell | that's what I said, yeah :p |
| 21:25.39 | MacWinner | awesome, thanks so much |
| 21:25.51 | MacWinner | is it even possible to do it in the sip.conf? |
| 21:25.57 | MacWinner | out of curiousity |
| 21:26.10 | Qwell | no ways I can think of. I'm sure you *could* do it though |
| 21:31.01 | MacWinner | qwell, i have a tmobile phone which allows allows unlimited incoming/outgoing to 5 specific DIDs.. if i setup my asterisk box with one of the DID and let other people call that DID to autoforward to my cell, i should get free incoming calls.. however, I wanted to set the CID so that I can know who is calling. do you think this will interfere with tmobile's billing? ie, do you think they... |
| 21:31.03 | MacWinner | ...depend on the CID to determine whether the call is one of my 5 free DIDs? |
| 21:31.15 | MacWinner | or is CID distinct from the originating caller number |
| 21:31.33 | Qwell | WELL |
| 21:31.38 | Qwell | it is |
| 21:31.40 | Qwell | BUT |
| 21:31.46 | MacWinner | (yes, i'm trying to work the system) ;) |
| 21:31.46 | Qwell | tmobile doesn't care |
| 21:31.52 | Qwell | ymmv |
| 21:32.07 | Qwell | I know at one point, they didn't bother looking at ANI, but looked at CID instead |
| 21:32.15 | Qwell | (same with vmail auth...scary, eh?) |
| 21:32.24 | MacWinner | wow |
| 21:32.33 | Qwell | yeah, so, there's a story that goes with that... |
| 21:32.53 | Qwell | at AstriCon one year, people "hacked" Mark Spencer's voicemail, and had Allison Smith record a new greeting for him |
| 21:33.12 | MacWinner | hehe |
| 21:33.25 | Qwell | I think twisted could probably tell the full version :p |
| 21:33.35 | MacWinner | he was tmobile customer? |
| 21:33.39 | Qwell | yeah, iirc |
| 21:33.50 | Qwell | most of the providers used to have that flaw |
| 21:34.11 | Qwell | MacWinner: here's another trick you might consider though |
| 21:34.23 | Qwell | myFaves costs more than "in-network" calling... |
| 21:34.28 | Qwell | *hint*hint* |
| 21:34.46 | Qwell | assuming you're only doing one direction |
| 21:35.11 | Qwell | of course, ymmv - don't try this at home - the usual disclaimers apply |
| 21:35.19 | MacWinner | hook up cell to pbx? |
| 21:35.34 | Qwell | nah, cid of another t-mobile cell |
| 21:35.57 | MacWinner | heh, so no need for fav5 |
| 21:36.01 | Qwell | maybe |
| 21:36.22 | MacWinner | so you think they use CID for the billing then? |
| 21:36.27 | Qwell | they used to |
| 21:40.15 | MacWinner | oki, so i got the the call coming in on the sip peer, and then to the dial plan to Dial() my cell.. how do you set the CID so that my cell sees the CID of the original caller? |
| 21:40.23 | MacWinner | is a parameter of Dial() ? |
| 21:41.23 | MacWinner | nm, setcallerid :P |
| 21:44.39 | *** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
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| 21:56.59 | MacWinner | Qwell: looks like you can set the ANI as well |
| 21:57.00 | *** join/#asterisk disposable (i=disposab@blackhole.sk) |
| 21:58.33 | disposable | how many concurrent phonecalls will a core2duo 2.33GHz with 4GB ram be able to handle? (no recoding, everything in g711) rough guesstimate is enough. |
| 21:59.20 | *** join/#asterisk ELBunce (n=erik@kde/developer/bunce) |
| 21:59.29 | carrar | 1,000,000 |
| 22:00.14 | disposable | carrar: at the same time i mean :) |
| 22:00.31 | *** join/#asterisk VJFROMGT (n=vjfromgt@user-387g9sf.cable.mindspring.com) |
| 22:00.52 | disposable | carrar: hence the 'concurrent' |
| 22:00.55 | VJFROMGT | what calls macro-record-enable .. i want to disbale it |
| 22:02.54 | *** join/#asterisk gerphimum (n=trekkie@cpe-70-125-148-108.satx.res.rr.com) |
| 22:04.37 | VJFROMGT | can someone tell me how to disable a marcro?> |
| 22:07.45 | *** join/#asterisk `Sauron (n=sauron@dsl001-130-033.aus1.dsl.speakeasy.net) |
| 22:08.57 | Nugget | what does "disable" mean in that context? A macro is either present in the dialplan or it isn't. |
| 22:09.12 | Nugget | if you invoke the macro it runs, if you don't it doesn't. |
| 22:09.49 | Nugget | are you sure you're really using asterisk and not freepbx or trixbox or something like that? |
| 22:11.31 | *** join/#asterisk outtolunc (n=me@63.197.134.218) |
| 22:11.51 | *** join/#asterisk mikecx (n=mikecx@pool-70-104-112-166.chi.dsl-w.verizon.net) |
| 22:13.25 | mikecx | what's the CLI command to show what an application can do/it's flags |
| 22:17.07 | Nugget | show application <appname> |
| 22:17.29 | mikecx | Nugget: thanks |
| 22:18.03 | mikecx | any ideas as to why the dundi stuff is still getting loaded and iaxtel is getting loaded even though it's not in the dialplan |
| 22:18.46 | *** join/#asterisk matsk (n=mk@83.233.97.210) |
| 22:19.08 | Nugget | add noload lines to modules.conf if you don't want those apps loaded. |
| 22:19.50 | mikecx | i can't really think of a good reason to have them running |
| 22:21.57 | VJFROMGT | sorry, i am back, i am using trix, i want to know what would normally call macro-record-enable so i can disable it from calling |
| 22:22.12 | VJFROMGT | guys at #freepbx dont know taht stuff |
| 22:22.37 | Qwell | ~trixbox |
| 22:22.38 | jbot | trixbox is, like, a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org |
| 22:23.48 | VJFROMGT | qwell, under regular asterisk, what would call such a file? |
| 22:23.53 | mikecx | i wonder if there is a way to stop the slaline macro from hanging up on transfer |
| 22:24.01 | Qwell | VJFROMGT: asterisk would |
| 22:24.19 | *** join/#asterisk drmessano (n=nonya@c-76-125-26-150.hsd1.ga.comcast.net) |
| 22:24.35 | VJFROMGT | correct, what conf file, i am trying to disable it from been called |
| 22:24.49 | Qwell | VJFROMGT: I don't know - trixbox and freepbx do things differently |
| 22:25.05 | MacWinner | ok, so is ANI spoofable or not? the asterisk application indicates so, but info on the web seems to indicate otherwise |
| 22:25.08 | VJFROMGT | qwell, please tell me how regular asterisk would do it |
| 22:25.17 | Qwell | MacWinner: nope |
| 22:25.21 | Qwell | not over the pstn |
| 22:25.32 | MacWinner | spoofable over VoIP? |
| 22:25.44 | Qwell | voip doesn't really use "ani" |
| 22:25.46 | file | VJFROMGT: if you go modifying extensions.conf it'll get overwritten by trixbox, you have to disable it there |
| 22:25.56 | MacWinner | err, from my VoIP provider then to their PSTN connection |
| 22:25.59 | VJFROMGT | thanks file, that is all i was asking for |
| 22:26.07 | Qwell | MacWinner: nope, it still goes over the pstn |
| 22:26.18 | Qwell | so, no ani spoofing for you |
| 22:27.07 | MacWinner | not really looking to spoof ANI.. just CID.. so i can get my free unliited incoming if tmobile uses ANI for billing, but get my cell to display the CID for my convenience |
| 22:27.58 | drmessano | hmmm |
| 22:28.25 | VJFROMGT | file, just a fyi trix does not write to extension.conf, it calls a file called extension_additional.conf which it does writing to |
| 22:28.25 | MacWinner | Qwell: what's this then: Set(CALLERID(all|name|num|ANI|DNID|RDNIS)=_CALLER NAME_<_CALLER NUMBER_>) |
| 22:28.31 | MacWinner | ANI is listed there |
| 22:28.46 | Qwell | because you can set it over some technologies |
| 22:28.58 | file | MacWinner: that doesn't mean it'll actually get transported... |
| 22:28.58 | Qwell | doesn't mean it'll carry over to the pstn though |
| 22:28.59 | Qwell | VJFROMGT: wrong |
| 22:29.11 | drmessano | Yeah, what the hell would the developers know |
| 22:29.12 | Qwell | VJFROMGT: it 100% definitely overwrites extensions.conf |
| 22:29.28 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 22:29.30 | VJFROMGT | tehre must be a glitch in the one i am using then |
| 22:29.31 | MacWinner | got it, thanks! |
| 22:31.27 | *** part/#asterisk bkw_ (n=brian@adsl-64-149-47-251.dsl.tul2ok.sbcglobal.net) |
| 22:33.13 | *** join/#asterisk mosty (n=mostyn@60-241-198-194.static.tpgi.com.au) |
| 22:33.17 | *** join/#asterisk vetetix (n=vetetix@83.222.33.25) |
| 22:33.42 | *** join/#asterisk adorah (n=Michael@87.69.57.246.cable.012.net.il) |
| 22:37.02 | *** join/#asterisk implicit (i=implicit@gateway/tor/x-21a74fc27bbfd208) |
| 22:44.00 | *** join/#asterisk piper69 (n=haiger@unaffiliated/piper69) |
| 22:45.26 | *** join/#asterisk outtolunc (n=user@63.197.134.218) |
| 22:45.31 | piper69 | guy please i need help setting up Cisco ATA 186 , every time i try to upgrade to H.323 using the tftp i get updgrade failed |
| 22:45.37 | piper69 | any idea please |
| 22:47.54 | *** part/#asterisk outtolunc (n=user@63.197.134.218) |
| 22:48.37 | *** join/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
| 22:49.32 | redder86 | Would someone please comment on how to troubleshoot/debug these: Dec 21 17:45:56 WARNING[26785]: channel.c:780 channel_find_locked: Avoided initial deadlock for '0x845f3b8', 9 retries! |
| 22:50.16 | piper69 | redder86: wait in line :) |
| 22:50.30 | redder86 | is there a line? |
| 22:50.39 | mikecx | no |
| 22:50.48 | redder86 | is there help? |
| 22:50.48 | piper69 | no , jk |
| 22:51.04 | mikecx | redder86: depebds on if someone knows the answer or not |
| 22:51.10 | piper69 | well i have been asking for help , but seems all are busy |
| 22:51.21 | piper69 | ^^that too |
| 22:51.24 | mikecx | piper69: gotta wait longer than a few minutes |
| 22:51.48 | mikecx | as the people that normally help here could either be out to dinner or living lives :-P |
| 22:51.51 | piper69 | mikecx: nope sir, i have been here since last 2 days , but i logoff and on |
| 22:52.08 | mikecx | piper69: hrrm, i've kinda idled the last few days and not seen that message |
| 22:52.19 | redder86 | well, I've been here off and on for the last few years |
| 22:52.23 | redder86 | :-D |
| 22:52.45 | piper69 | mikecx: just go back 3 hrs ago |
| 22:52.57 | mikecx | wait until you see people like bkruse, Qwell, and [TK]D-Fender on, they can usually help with most stuff |
| 22:53.07 | mikecx | *talking, not just on |
| 22:53.11 | piper69 | troy was helping me the other night but i am stuck now |
| 22:53.15 | piper69 | mikecx: yes |
| 22:53.42 | mikecx | redder86: have you emailed Cisco about it? |
| 22:53.46 | drmessano | piper69, call Cisco |
| 22:53.52 | drmessano | lol |
| 22:53.58 | mikecx | great minds... |
| 22:53.59 | redder86 | Cisco? |
| 22:54.00 | piper69 | funnyyyy |
| 22:54.02 | piper69 | hahah |
| 22:54.04 | piper69 | no |
| 22:54.26 | drmessano | 1-800-R-U-CCNP |
| 22:54.34 | mikecx | yeah, who would call the people that make the phones ( or buy them and rebrand them) for help, that would be silly |
| 22:54.41 | drmessano | ROFL |
| 22:54.56 | mosty | redder86, is that just a warning, or do you have an actual problem? |
| 22:55.03 | piper69 | i am a 1/2 CCNA |
| 22:55.06 | piper69 | lol |
| 22:55.15 | redder86 | mosty: it slows down the call progress |
| 22:55.26 | redder86 | mikecx: why would I e-mail Cisco? |
| 22:55.37 | piper69 | redder86: it was a typo |
| 22:55.42 | drmessano | Surely an IRC channel would know more than the manufacturer |
| 22:55.53 | piper69 | drmessano: you where here last night right |
| 22:55.55 | mikecx | redder86: i musta hit the wrong person with that one |
| 22:55.59 | mosty | ReD-MaN, what version of asterisk? |
| 22:56.02 | mikecx | redder86: sorry |
| 22:56.06 | redder86 | mosty: 1.2.26 |
| 22:56.09 | drmessano | Im always here... Im built on Asterisk |
| 22:56.29 | mikecx | bkruse: that was me, i was telling peoples of your genius |
| 22:56.34 | drmessano | I'm a perl script |
| 22:56.35 | bkruse | oh, awesome :P |
| 22:57.22 | piper69 | mikecx: so do you think you could help me |
| 22:57.52 | piper69 | i am tryiing to upgrade from SCCP to H.323 |
| 22:57.53 | mikecx | piper69: nope, i've never worked with the cisco stuff |
| 22:58.29 | chisefu|afk | hm I'm having problems getting my phones to register the lines with the server |
| 22:58.33 | chisefu|afk | is anyone good with this kind of thing/ |
| 22:59.03 | mikecx | chisefu|afk: be more specific. Can your phones not register? Can you not dial out? What exactly is not working |
| 22:59.06 | mosty | chisefu|afk, sip phones? pastebin your sip.conf and the console error message? |
| 22:59.29 | piper69 | drmessano: have you seen troy here latly |
| 23:00.16 | chisefu|afk | when I go on my polycom soundpoint ip 430 to status->lines |
| 23:00.22 | chisefu|afk | it says the extension isn't registered |
| 23:00.27 | chisefu|afk | I can't call other extensions or call out |
| 23:00.52 | *** join/#asterisk MohShami (n=mohshami@86.108.40.150) |
| 23:00.55 | chisefu|afk | I can connect to the phones with a browser though |
| 23:00.56 | mikecx | chisefu|afk: have you setup your dial plan and looked at the output from the cli? |
| 23:01.17 | MohShami | hey guys, does * support qsig? if it does, can someone please point me to a document that I can start with? |
| 23:01.52 | chisefu|afk | Asterisk 1.4.9 built by admin @ kyoto on a x86_64 running Linux on 2007-07-25 21:11:37 UTC |
| 23:03.21 | *** part/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
| 23:03.51 | chisefu|afk | I can't seem to get my dial plan to save in the gui |
| 23:03.56 | mosty | chisefu|afk, pastebin your sip.conf and the error message in the console/logs when you attempt to register |
| 23:04.15 | chisefu|afk | ok |
| 23:04.15 | mosty | chisefu|afk, are you using trixbox? we don't really support that here, try #trixbox |
| 23:04.23 | chisefu|afk | I'm using asteriskNOW |
| 23:04.58 | mosty | see the /topic - i think #asterisk-gui is the place to try |
| 23:06.02 | *** join/#asterisk PepOSX (n=pepOSX@201.248.215.16) |
| 23:15.31 | mikecx | i can't tell if i'm getting closer or father away from getting SLA + Fax Detection working |
| 23:19.00 | *** part/#asterisk Cresl1n (n=matt@nat/digium/x-94caefee76138fea) |
| 23:30.35 | MacWinner | is caller id spoofing illegal? |
| 23:32.22 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 23:35.04 | mikecx | MacWinner: http://en.wikipedia.org/wiki/Caller_ID_spoofing |
| 23:35.17 | mikecx | first result in google for called id spoofing |
| 23:40.19 | carrar | Oh see, now you are expecting people to know what google is |
| 23:40.21 | MacWinner | seems to be only if intending fraud or harm |
| 23:40.35 | MacWinner | but CID spoofing doesn't seem illegal in general |
| 23:41.53 | *** join/#asterisk RoyK (n=roy@ip-133-27-149-91.dialup.ice.no) |
| 23:55.31 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 23:56.54 | chisefu|afk | WARNING[3531] chan_sip.c: Probably a DNS error for registration to 456@dynamic, trying REGISTER again (after 20 seconds) |
| 23:56.54 | chisefu|afk | [Dec 21 00:00:35] NOTICE[3531] chan_sip.c: -- Registration for '420@dynamic' timed out, trying again (Attempt #35064) |
| 23:56.54 | chisefu|afk | [Dec 21 00:00:35] WARNING[3531] chan_sip.c: No such host: dynamic |
| 23:57.10 | chisefu|afk | this is the error that keeps coming up on my command line |
