| 00:04.01 | kippi | ok |
| 00:04.02 | kippi | thanks |
| 00:04.30 | kippi | just seems to hang me up :( |
| 00:04.40 | kippi | is there anything on the wiki? |
| 00:04.47 | Qwell | mog_work: If you remind me to bring my VPN stuff, I'll show you the GUIs I've been writing for work |
| 00:05.24 | kippi | found out why |
| 00:05.34 | kippi | woo :) |
| 00:05.38 | Qwell | kippi: You need to have something after Dial() |
| 00:05.47 | Qwell | check on ${DIALSTATUS} |
| 00:05.48 | kippi | I put in the wrong DDI |
| 00:07.17 | mog_work | 2ooh thats hot Qwell |
| 00:10.30 | *** part/#asterisk mattwj2005 (n=Matt@dialup-4.159.62.213.Dial1.Chicago1.Level3.net) |
| 00:13.38 | Lino` | woah what i extremely hate |
| 00:13.42 | Lino` | is theft |
| 00:13.49 | Qwell | indeed |
| 00:14.09 | *** part/#asterisk bkw__ (n=bkw_@adsl-70-142-62-199.dsl.tul2ok.sbcglobal.net) |
| 00:14.33 | Lino` | at the fair booth we were exibiting a new scanner which scans and sorts mail |
| 00:14.46 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 00:14.57 | Lino` | suddenly a chinese guy appears who wants to buy such a scanner, but only one and only the sorter module |
| 00:15.24 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 00:15.35 | Lino` | well, that was not for sale, obviously. he just grabbed his cell, yelled some incomprehensible stuff into it and 5 minutes later some army arrived like 20 persons or so |
| 00:15.47 | Lino` | looked at the scanner, made photos, etc |
| 00:16.06 | Lino` | i bet there will be a chinese rebuild version in 6 months or so -.- |
| 00:16.08 | Lino` | scum |
| 00:17.29 | Qwell | hmm |
| 00:17.52 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 00:18.38 | Qwell | Lino`: not fun |
| 00:18.50 | Lino` | no definately not |
| 00:19.02 | Lino` | cameras are prohibited in that area |
| 00:19.07 | brookshire | qwell: is too |
| 00:19.25 | Qwell | Qwell is prohibited from many places |
| 00:19.30 | Lino` | but they care nothing about patents etc |
| 00:20.00 | Lino` | well but the sangoma guy was worth the trip lol never seen anybody being that agressive for NOTHING |
| 00:21.29 | Qwell | sangoma guys are newbs |
| 00:21.38 | bugz | lol |
| 00:21.50 | Lino` | whenever you see one ask him wether he uses HFC chips |
| 00:21.53 | Qwell | I heard somebody *cough*cough* rant about sangoma for a while, at Astricon |
| 00:22.14 | Lino` | its better than ANYTHING... thats something you dont even get in the zoo |
| 00:22.19 | *** join/#asterisk Abydos313 (i=justcall@72.20.3.66) |
| 00:22.37 | Qwell | Lino`: Was Mark at this thing, where sangoma was next to the Digium booth? |
| 00:22.50 | Lino` | dunno I talked to kenny |
| 00:22.55 | Qwell | brookshire: ? |
| 00:23.06 | Lino` | it was not directly next to it, it was like diagonal opposite |
| 00:23.10 | Qwell | oh...still |
| 00:23.15 | Lino` | but pretty near |
| 00:23.20 | Lino` | extremely near |
| 00:23.36 | Qwell | unless it was a tiny arena, I'll bet they planned it |
| 00:23.36 | *** join/#asterisk Aurs (n=Aurs@a217-118-41-157.bluecom.no) |
| 00:23.36 | Lino` | way too near |
| 00:23.36 | Lino` | :D |
| 00:23.45 | Lino` | maybe next time digium will bring some serious stuff, like guns or tanks |
| 00:23.49 | Lino` | :D |
| 00:24.03 | Qwell | heh |
| 00:24.10 | Lino` | or they simply use their cleverness to defeat the stupid monkey |
| 00:25.50 | *** join/#asterisk sag_ich_nicht (i=bitch2k@85-124-39-30.dynamic.xdsl-line.inode.at) |
| 00:26.12 | Lino` | omg the enemy |
| 00:26.16 | sag_ich_nicht | lol |
| 00:26.26 | sag_ich_nicht | hi there ;P |
| 00:26.33 | *** join/#asterisk rehash (n=Nando@unaffiliated/nand0) |
| 00:26.38 | *** join/#asterisk SibRphrek (i=SibrPhre@user-12lccke.cable.mindspring.com) |
| 00:26.57 | Lino` | hmmm i got an old HFC based card here with two jumpers on it |
| 00:27.05 | Lino` | sitecomm |
| 00:27.42 | Lino` | i believe they are used to emulate a crossover cable for directly going into the line |
| 00:28.37 | Lino` | well, i guess it wont explode right into my face if i try it *g* |
| 00:29.33 | sag_ich_nicht | wouldn't it be easier to control it directly trough software? |
| 00:31.09 | sag_ich_nicht | i mean, why put a jumper for it there if you could control it directly from what ever OS you use |
| 00:31.26 | Lino` | nono thats not the NT mode |
| 00:31.27 | kippi | with the sample files, is there one that says please enter an extension? |
| 00:31.42 | Lino` | hmmm |
| 00:31.48 | Lino` | i guess mom i'll have a look |
| 00:31.57 | Lino` | theres something like that for the company directory |
| 00:32.44 | *** join/#asterisk glm2k (n=glm@rrcs-24-199-11-46.west.biz.rr.com) |
| 00:33.20 | Lino` | try |
| 00:33.20 | Lino` | # please-enter-the.gsm :Please enter the |
| 00:33.20 | Lino` | # extension.gsm :extension |
| 00:33.31 | Lino` | ior |
| 00:33.32 | Lino` | # |
| 00:33.32 | Lino` | # enter-ext-of-person.gsm :enter the extension of the person you are trying to reach |
| 00:33.46 | Lino` | alternatively: # |
| 00:33.46 | Lino` | # pls-entr-num-uwish2-call.gsm :please enter the number you wish to call |
| 00:33.46 | kippi | brill |
| 00:33.48 | kippi | thanks |
| 00:33.59 | Lino` | there are more than enough choices |
| 00:34.04 | Lino` | http://www.voip-info.org/wiki/index.php?page=Asterisk+sound+files+additional <= |
| 00:34.10 | Lino` | you'll have to install the additional sounds |
| 00:34.14 | kippi | ah cool |
| 00:35.13 | Lino` | http://www.voip-info.org/wiki/index.php?page=Asterisk+sound+files <= those are "default" ;) |
| 00:35.14 | *** join/#asterisk TUplink (n=Tommy@68-232-82-147.chvlva.adelphia.net) |
| 00:35.26 | TUplink | whats the new version number? |
| 00:35.28 | TUplink | 1.2? |
| 00:35.34 | Qwell | TUplink: see topi |
| 00:35.35 | Qwell | c |
| 00:35.45 | TUplink | topic |
| 00:35.53 | Qwell | yes, that |
| 00:36.40 | TUplink | how do i c that? |
| 00:36.43 | TUplink | tryed /topic |
| 00:36.44 | Lino` | omg |
| 00:36.56 | Lino` | /topic #asterisk |
| 00:37.15 | TUplink | thanks |
| 00:37.16 | Lino` | [01:37:39] * Topic is 'Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- Asterisk 1.2.5 released! (March 3, 2006) -=- Asterisk-addons 1.2.2 now available (March 6, 2006)' |
| 00:37.29 | TUplink | now here is a question |
| 00:37.42 | Lino` | just ask |
| 00:37.44 | TUplink | i have a CVS version how do i find the ver number there? |
| 00:37.53 | Lino` | hmm |
| 00:38.00 | Lino` | installed? |
| 00:38.02 | Lino` | or the source? |
| 00:38.07 | TUplink | instaled |
| 00:38.11 | Lino` | asterisk -r |
| 00:38.16 | TUplink | show version |
| 00:38.30 | Lino` | if you start the CLI the version number is displayed |
| 00:38.42 | TUplink | Asterisk CVS-v1-0-10/27/05-22:10:44 built by root@Server.KAK-Networks.net on a i386 running FreeBSD |
| 00:38.59 | Lino` | yeah |
| 00:39.07 | Lino` | that looks like 1.0.10 *g* |
| 00:39.08 | TUplink | well.. that translates to? |
| 00:39.14 | TUplink | ok... |
| 00:39.19 | TUplink | well here is a q |
| 00:39.24 | TUplink | i want to upgrade |
| 00:39.29 | Lino` | yeah |
| 00:39.32 | TUplink | will all of my conf files still work? |
| 00:39.37 | Lino` | huh |
| 00:39.42 | Lino` | ask a pro ;) |
| 00:39.50 | TUplink | whos a pro? |
| 00:39.57 | Lino` | anybody but be :D |
| 00:40.03 | TUplink | haha |
| 00:40.05 | *** join/#asterisk remmo_ (n=chatzill@adsl-1180.isp.net.au) |
| 00:40.28 | Lino` | ~seen possible |
| 00:40.31 | jbot | possible <n=Babbel@23.255-136-217.adsl-fix.skynet.be> was last seen on IRC in channel #asterisk, 1d 12h 21m 10s ago, saying: 'I guess not'. |
| 00:40.33 | TUplink | Qwell will my conf files from 1.0.10 work on the new version? |
| 00:40.51 | Qwell | TUplink: there is a file called UPGRADE.txt that you should read |
| 00:40.54 | remmo_ | does anyone know why svn asterisk does not autoload modules. seems like i onyl have 5 modules loaded? |
| 00:41.04 | Lino` | hmmm |
| 00:41.07 | TUplink | ok... |
| 00:41.10 | Qwell | remmo_: Do you have autoload=no in modules.conf? |
| 00:41.10 | Lino` | look into /etc/asterisk/modules.conf |
| 00:41.25 | remmo_ | no its autoload=yes |
| 00:41.32 | Qwell | remmo_: odd |
| 00:41.37 | Lino` | apart from autoload=no it maybe noload somethign |
| 00:41.39 | Lino` | something |
| 00:41.41 | remmo_ | svn version @ jan 03 works fine |
| 00:41.54 | Qwell | remmo_: I don't think the default behavior has changed |
| 00:42.14 | remmo_ | yeah that what i though |
| 00:42.38 | remmo_ | its like its not even looking at the modules.conf. perms are all right |
| 00:43.22 | remmo_ | the jan version was from cvs, looks like i'm having issues with the svn version. everything compiles and installs right |
| 00:43.34 | remmo_ | i dont even have stop or start command in cli |
| 00:43.35 | *** join/#asterisk joat (n=joat@ip70-160-213-117.hr.hr.cox.net) |
| 00:45.47 | *** join/#asterisk |omni| (i=cathode@c-67-185-96-86.hsd1.wa.comcast.net) |
| 00:48.10 | Lino` | hmmm |
| 00:49.33 | *** join/#asterisk Zilasb (i=Zilasb@c-67-191-160-216.hsd1.ga.comcast.net) |
| 00:49.39 | Zilasb | hello |
| 00:50.15 | Zilasb | I need to add h323 to my * does anbody have good step by step instructions or smth that would help me? |
| 00:50.23 | remmo_ | in which modules would stop and start be located in? |
| 00:52.41 | Zilasb | i see h323 comes already as asterisk addon 1.2.2 |
| 00:53.16 | *** join/#asterisk diclophis (n=diclophi@65.203.37.58) |
| 00:53.21 | diclophis | howdy all |
| 00:53.32 | diclophis | can asterisk Record MP3 files? |
| 00:53.38 | diclophis | like thoguh the AGI RECORD FILE command? |
| 00:53.46 | Qwell | diclophis: should be able to. Just install asterisk-addons, to get format_mp3 |
| 00:55.23 | diclophis | rockin |
| 00:55.48 | ambriento | I don't think so diclophis |
| 00:56.02 | ambriento | You'll have to record and then convert to mp3 |
| 00:56.04 | diclophis | mmm i think i just crashed it |
| 00:56.46 | diclophis | Mar 11 16:54:43 ERROR[15466] format_mp3.c: I Can't write MP3 only read them. |
| 00:56.47 | diclophis | Damn |
| 00:56.49 | TUplink | mv -f .libs/SQLAllocStmt.lo SQLAllocStmt.lo |
| 00:56.49 | TUplink | source='SQLBindCol.c' object='SQLBindCol.lo' libtool=yes depfile='.deps/SQLBindCol.Plo' tmpdepfile='.deps/SQLBindCol.TPlo' depmode=gcc3 /bin/sh ../../depcomp /bin/sh ../../libtool --mode=compile cc -DPACKAGE_NAME=\"\" -DPACKAGE_TARNAME=\"\" -DPACKAGE_VERSION=\"\" -DPACKAGE_STRING=\"\" -DPACKAGE_BUGREPORT=\"\" -DPACKAGE=\"unixODBC\" -DVERSION=\"2.2.11\" -DYYTEXT_POINTER=1 -DSTDC_HEADERS=1 -DHAVE_SYS_TYPES_H=1 -DHAVE_SYS_STAT_H=1 -DHAV |
| 00:56.51 | TUplink | oops |
| 00:56.54 | TUplink | sorry |
| 00:57.01 | ambriento | :) |
| 00:57.08 | ambriento | get used to pastebin.com :) |
| 00:57.39 | diclophis | so.. how much work is involved in encoding 16-bit 8000Hz mono wav files into mp3? |
| 00:58.07 | diclophis | and or does lame work on BSD? |
| 00:58.18 | ambriento | are you using BSD? |
| 00:58.27 | diclophis | i am thinking i will just encode them before i let my users download them |
| 00:58.45 | diclophis | ambriento: for the web-side of my product yes |
| 00:58.52 | ambriento | I see |
| 00:59.01 | diclophis | i want to have a inline flash player for their voicemails |
| 00:59.11 | ambriento | so you'll have an asterisk box and a web box |
| 00:59.11 | diclophis | but the damn flash code only supports streaming mp3s |
| 00:59.23 | diclophis | like.. wavs are too easy to decode or something |
| 00:59.30 | diclophis | ambriento yes |
| 00:59.45 | ambriento | let the encode thing to the web box |
| 01:00.44 | diclophis | yea.. but i need to find lame for BSD |
| 01:00.54 | Zilasb | Q: once I download asterisk addon and make clean and make install I am good to go? should it be h323.conf file in /etc/asterisk???? |
| 01:01.08 | diclophis | and the thing is.. i can get it to work fine, but i need to reveal this knowledge to my production team so they can whiteroom the system |
| 01:01.33 | ambriento | zilasb, there's more to do with h.323 |
| 01:01.39 | diclophis | hence... it would be sweet if there was a uber simple wav2mp3.c ... or if lame was in portage. |
| 01:02.12 | TUplink | just download and install LAME |
| 01:02.38 | Zilasb | ambriento: what else to do? maybe you know any place I can see instructions pls? |
| 01:03.18 | ambriento | diclophis, "LAME compiles on Windows, DOS, GNU/Linux, MacOS X, *BSD..." |
| 01:03.33 | diclophis | rock... but can they install it using bsd ports... |
| 01:03.46 | TUplink | lame is in freebsd ports |
| 01:03.56 | diclophis | interesting |
| 01:04.04 | TUplink | cvsup |
| 01:04.11 | TUplink | cvssup* |
| 01:05.05 | ambriento | zilasb, I had. try this one: http://www.inaccessnetworks.com/projects/asterisk-oh323 |
| 01:05.16 | ambriento | brb guys |
| 01:05.50 | diclophis | of course lame isnt in the fedora core 2 repos |
| 01:06.12 | TUplink | i have lame on my FC4 |
| 01:06.19 | TUplink | just download the rpm |
| 01:06.22 | *** join/#asterisk Lino` (n=Lino@i577BD751.versanet.de) |
| 01:06.31 | diclophis | mmm but yum is so much easier |
| 01:06.37 | diclophis | is it by chance called liblame? |
| 01:07.20 | TUplink | "rpm http://dag.wieers.com/packages/lame/lame-3.96.1-3.1.fc2.rf.i386.rpm" |
| 01:07.24 | TUplink | try that command |
| 01:08.06 | Zilasb | ambriento: Thanks |
| 01:08.12 | TUplink | and do http://dag.wieers.com/packages/lame/lame-devel-3.96.1-3.1.fc2.rf.i386.rpm |
| 01:08.25 | TUplink | i belive thats how i did it |
| 01:10.26 | *** join/#asterisk Delmar (n=delmar@203-114-178-231.inspire.net.nz) |
| 01:12.35 | Lino` | what would you prefer? bristuff or misdn? |
| 01:13.41 | *** join/#asterisk stuartcw (n=chatzill@softbank221025056004.bbtec.net) |
| 01:25.59 | *** join/#asterisk zotz (n=zotz@24.231.32.85) |
| 01:31.16 | diclophis | TUplink... ok i have it working however my 16bit 8000hz wav files are getting sample WAY too fast |
| 01:31.47 | TUplink | i dont know any thing about it... just know it was a dependicie i had to install |
| 01:32.29 | TUplink | dose the wav play rite? |
| 01:33.31 | TUplink | all i know is is that "Lame Aint a Mp3 Encoder" |
| 01:33.41 | diclophis | haha |
| 01:33.44 | diclophis | yea the wavs are ok |
| 01:34.07 | TUplink | whats the command you are using to convert? |
| 01:34.20 | *** join/#asterisk file (n=joshnet@mctnnbsa24w-142167032200.pppoe-dynamic.nb.aliant.net) |
| 01:34.37 | diclophis | i have tried a bunch |
| 01:34.45 | diclophis | <PROTECTED> |
| 01:34.55 | diclophis | did i mention my sound player was a flash thingy |
| 01:35.52 | TUplink | try "lame original.wav new.mp3" |
| 01:36.15 | TUplink | well c if the mp3 plays fine when its raw |
| 01:37.21 | diclophis | yes it does |
| 01:37.38 | diclophis | but in the flash player its WAY too fast |
| 01:37.57 | TUplink | well.. then thats a flash problum and thats way out of my reach |
| 01:38.57 | sevard | <PROTECTED> |
| 01:39.00 | sevard | shit. |
| 01:39.11 | Lino` | almost @ sevard |
| 01:39.12 | Lino` | :D |
| 01:39.27 | sevard | signs it's been a long day and I'm getting tired ^^^ |
| 01:40.04 | Lino` | lol |
| 01:40.08 | Lino` | its not that late |
| 01:40.14 | Lino` | 2.40.55 |
| 01:40.21 | Lino` | 2 o clock in the morning |
| 01:44.17 | diclophis | here is the magic flag |
| 01:44.18 | diclophis | --resample 11.025 |
| 01:46.25 | *** join/#asterisk bkw__ (n=bkw_@adsl-70-142-62-199.dsl.tul2ok.sbcglobal.net) |
| 01:58.47 | *** join/#asterisk jeebusroxors (n=jeebusro@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
| 01:59.27 | *** join/#asterisk littleball (n=littleba@cm67.epsilon168.maxonline.com.sg) |
| 01:59.44 | *** join/#asterisk Octothorpe (n=octothor@unaffiliated/octothorpe) |
| 02:00.01 | littleball | hi, anyone here is interested in JAIN SLEE? |
| 02:00.24 | diclophis | only if you got some nudes |
| 02:01.09 | TUplink | ok.... this si wierd... |
| 02:01.24 | TUplink | just instaled 1.2.4 freebsd.... cant get it to start |
| 02:01.32 | TUplink | afraid to restart box |
| 02:01.47 | littleball | it seems that JAIN SLEE promise a lot of features and functions |
| 02:02.16 | Octothorpe | JAIN SLEE? |
| 02:02.41 | littleball | yes |
| 02:03.00 | littleball | asterisk also can be integrate into it |
| 02:03.17 | *** join/#asterisk coppice (n=chatzill@140.155.17.210.dyn.pacific.net.hk) |
| 02:03.58 | diclophis | is it an OSS pbx? |
| 02:04.20 | *** join/#asterisk jeebusroxors (n=jeebusro@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
| 02:04.25 | littleball | it is much more than that |
| 02:05.57 | littleball | currently, for me its value is that it make prividing new value added service more easy and migration also become easy |
| 02:06.11 | diclophis | what is the URL? |
| 02:06.55 | littleball | java.sun.com/products/jain/ |
| 02:07.20 | littleball | service logic execution enviroment |
| 02:07.22 | littleball | SLEE |
| 02:08.19 | justinu | boy, you can't talk about java here |
| 02:08.21 | justinu | people get all edgy |
| 02:08.24 | littleball | i have done a few asterisk projects. But there is no clues to handle/pattern business logic. |
| 02:08.40 | littleball | this is not java but telecommunication |
| 02:08.51 | diclophis | a java PBX |
| 02:08.56 | diclophis | ... thatll be the day |
| 02:09.12 | littleball | no, it is not a java pbx. actually, it has nothing directly related to pbx. |
| 02:09.49 | diclophis | mmm if its not a pbx... then its either a IM server, or a Mail server or a Web server |
| 02:09.54 | diclophis | perhaps a file server it is? |
| 02:10.17 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 02:10.54 | diclophis | well, it uses SIP.. so it might have some potential |
| 02:10.55 | littleball | no, service logic execution enviroment. Do you have the same experience of how to code /organize your business logic when doing asterisk projects? And how to migrate/add new service? |
| 02:11.16 | diclophis | well.. i can't say my AGI code is perfect |
| 02:11.48 | littleball | i am using AGI also.but i have the feeling |
| 02:12.05 | diclophis | oh.. it could def. be better |
| 02:12.10 | jeebusroxors | exit |
| 02:12.22 | *** join/#asterisk ms345 (n=mike_sim@dsl027-163-193.atl1.dsl.speakeasy.net) |
| 02:12.47 | *** join/#asterisk jeebusroxors (n=jeebusro@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
| 02:14.47 | diclophis | is there a way i can set the uniqueid of the CDR myself? |
| 02:16.20 | diclophis | and is the uniqueid unique out of every row in the cdr table (i am using mysql_cdr) |
| 02:16.21 | diclophis | ? |
| 02:19.01 | *** join/#asterisk r0d3nt|m (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 02:27.01 | littleball | in my current projects, i move all business related issues to external system. such as billing, user management, some parts of routing. asterisk act as service adapters to networks. Because lackiing of reference implementation, i am wondering wheether my strategy is the proper way |
| 02:27.47 | zigman | diclophis no and yes |
| 02:28.19 | zigman | no you can't set the uniqueid yourself, and yes its unique (hense the name) |
| 02:28.39 | diclophis | ok cool |
| 02:30.41 | diclophis | do you know the format of the field? |
| 02:35.59 | littleball | http://www.mobicents.org/ |
| 02:43.43 | *** join/#asterisk tainted_ (n=identd@adsl-71-129-41-176.dsl.irvnca.pacbell.net) |
| 02:43.58 | tainted_ | what does 'CHANUNAVAIL' dialstatus mean exactly |
| 02:44.04 | tainted_ | i get it in so many scenarios |
| 02:44.40 | tainted_ | sometimes it is because of a bad phone number, sometimes provider's fault |
| 02:44.43 | *** join/#asterisk austinnichols101 (n=austinni@dsl-10-169.cofs.net) |
| 02:50.58 | *** join/#asterisk coppice (n=chatzill@251.203.17.210.dyn.pacific.net.hk) |
| 02:51.18 | *** join/#asterisk project_2501 (n=project-@S01060004e2929dc9.br.shawcable.net) |
| 02:51.34 | Qwell | project_2501: What is that from? |
| 02:51.34 | *** join/#asterisk brockj49464 (n=brockj49@63.87.56.235) |
| 02:52.01 | project_2501 | it's a reference the the anime 'ghost in the shell' |
| 02:52.05 | Qwell | right |
| 02:52.43 | project_2501 | Qwell: so what are you up to tonight? |
| 02:52.51 | Qwell | project_2501: not a lot |
| 02:53.15 | project_2501 | Qwell: I'm just working on my asterisk server, trying to get voivemail etup |
| 02:53.38 | Faithful | Wondering if I can get some help with my dial plan... I have a GSM gateway that I dial in with and get * to answer, I then want to be able to punch in a phone # and dial out of * VOIP |
| 02:53.47 | diclophis | app_voicemail sucks |
| 02:53.59 | Faithful | any suggestions? |
| 02:54.01 | diclophis | i had to make a brand new one out of AGI to do anything useful |
| 02:54.26 | austinnichols101 | faithful: which gateway you using? |
| 02:55.11 | glm2k | Faithful: lemme guess, sam's gateway? |
| 02:55.15 | Faithful | austinnichols101: a doc-n-talk connected to a sipura... because my provider i 3G and lock sims to phones |
| 02:56.04 | Faithful | otherwise I would have a GSM SIP gateway |
| 02:56.14 | Faithful | like blue voice |
| 02:56.40 | austinnichols101 | how does the doc-n-talk handle multiple calls (if at all)? |
| 02:57.00 | austinnichols101 | I'm getting ready to set up something similar to what you're describing |
| 02:57.32 | Faithful | austinnichols101: I have it working to the point I can dial out ok and dial in... but all I can figure to do is redirect the incomming call... but I need (once connected to asterisk) to be able to dial another number to connect |
| 02:58.09 | austinnichols101 | faithful: do you have it hitting a specific context when it gets to asterisk? |
| 02:58.10 | Faithful | austinnichols101: multiple calls... does call waiting but it is only a single sim card device |
| 02:58.34 | Faithful | austinnichols101: Yes, I have set up it's own context |
| 02:58.44 | austinnichols101 | faithful: not sure I understand how that would actuall show up. What happens if two asterisk users tried to go out to the dnt? |
| 02:59.06 | Faithful | austinnichols101: one gets a busy signal |
| 02:59.09 | austinnichols101 | k |
| 02:59.17 | austinnichols101 | so not really call waiting |
| 02:59.18 | Faithful | you need a multi-channel device for that |
| 02:59.21 | austinnichols101 | k |
| 02:59.33 | austinnichols101 | does such an animal exist? |
| 02:59.52 | Faithful | http://www.2n.cz/products/gsm_gateways.html |
| 03:00.07 | austinnichols101 | yes - saw that one |
| 03:00.08 | Faithful | and ericsson etc... |
| 03:00.17 | *** part/#asterisk littleball (n=littleba@cm67.epsilon168.maxonline.com.sg) |
| 03:00.20 | austinnichols101 | they don't seem to have anyone in the US, but still looking |
| 03:00.30 | Faithful | but I am stuck because of my telco locking sims to phones |
| 03:00.45 | austinnichols101 | I may be in the same spot because I'm probably going to have to go CDMA |
| 03:01.12 | austinnichols101 | I'm checking my configs because I'm doing something similar with a dialback that you might be able to use |
| 03:01.26 | Faithful | doc-n-talk works ok with sipura spa-3000 with a bit of teaking |
| 03:02.01 | Faithful | austinnichols101: tnx |
| 03:02.17 | austinnichols101 | faithful - in my case all my internal users are in the from-internal context |
| 03:02.33 | austinnichols101 | except for my cell user who's call ends up in the from-callback context |
| 03:03.00 | austinnichols101 | in from-callback I'm just doing 'include => from-internal' |
| 03:04.01 | austinnichols101 | of course, I do have some security built in to verify that the from-callback user |
| 03:04.25 | austinnichols101 | not sure if that's of any help or not |
| 03:05.12 | diclophis | so what does a gsm gateway do? |
| 03:05.19 | diclophis | isnt gsm cellular type stufgf> |
| 03:05.29 | austinnichols101 | gsm on one side, sip on the other |
| 03:05.30 | Qwell | yes |
| 03:05.50 | diclophis | so... why not just dial into the * machine usng a pstn number? |
| 03:06.15 | austinnichols101 | usual answer is that by going gsm to gsm you end up paying less |
| 03:06.18 | Faithful | I tried to get MIAX going but found it problematic and there is very little documentation |
| 03:06.53 | diclophis | so.. a gsm gateway is basically a cellphone/.. that plugs into a pc, that exposes channels like a digium card? |
| 03:07.08 | Faithful | if you have a free gsm-gsm plan then you can call from mobile at VOIP prices |
| 03:07.13 | austinnichols101 | diclophis: for example, verizon users in the us can get the 'IN' plan and call each other as much as they want for one rate |
| 03:07.19 | austinnichols101 | diclophis: right |
| 03:07.27 | diclophis | hmm |
| 03:07.29 | austinnichols101 | a phone brick |
| 03:07.39 | diclophis | so why not just wire up someting to a cheap phone |
| 03:07.39 | diclophis | ? |
| 03:07.49 | diclophis | how much could there be to a fxo port? |
| 03:07.51 | diclophis | there is only 2 wires |
| 03:07.54 | diclophis | ... |
| 03:07.55 | austinnichols101 | you just described the dock-n-talk |
| 03:08.01 | diclophis | oh.. haha |
| 03:08.10 | diclophis | man way to not pay attenteion eh |
| 03:08.16 | austinnichols101 | http://www.voip-info.org/wiki/view/VOIP+GSM+Gateways |
| 03:08.47 | Faithful | but that only gives you fxs you need sipura to turn it into sip or use an fxo port in asterisk |
| 03:09.12 | diclophis | yea... arnt there like 15 dollar fxo boards out there? |
| 03:09.21 | diclophis | like... some sort of cheap winmodem that you can hack |
| 03:09.32 | diclophis | x100p clones or something |
| 03:09.42 | *** join/#asterisk justnulling2 (n=justnull@ool-18bcc906.dyn.optonline.net) |
| 03:09.48 | Faithful | So what I need is that * answers the call and the lets me punch in a phone number |
| 03:10.07 | diclophis | yea.. thats like 10 lines of extensions.conf programming |
| 03:10.15 | austinnichols101 | I have 'free unlimited incoming' with nextel so I have a dialback setup now, but it's a bit of a pain to have to dial and then wait for the callback and then dial the number you really wanted |
| 03:10.59 | diclophis | so your basically trying to hack it so you can make pstn calls from your celly using your free gsm2gsm plan |
| 03:11.04 | austinnichols101 | faithful: did you try including your internal context? |
| 03:11.21 | austinnichols101 | that's my goal - 'stick it to the man' |
| 03:11.27 | diclophis | haha |
| 03:11.30 | Faithful | I will try |
| 03:11.38 | Faithful | I did before tho |
| 03:11.57 | diclophis | hell.. you could even use something like DUNDI over sip and not need a pstn on the other side of your * machine |
| 03:12.13 | Faithful | Hey, 1c/min mobile calls is all right by me... just have to hit speed dial before dialing the number |
| 03:12.40 | austinnichols101 | I was thinking of that. If I could get a message from a smartphone back to the * box then it could handle the whole call setup |
| 03:12.52 | diclophis | text-message through email? |
| 03:12.54 | austinnichols101 | us cellular suck |
| 03:12.55 | austinnichols101 | s |
| 03:13.19 | Faithful | I pay 25c flag fall and 60c/min on my mobile |
| 03:13.39 | austinnichols101 | I'm just trying to get it as seamless as possible. I don't want to have to go through much more than just dialing a number... |
| 03:13.57 | Faithful | gsm-gsm 1c/min is attractive in AU |
| 03:14.24 | diclophis | well.. you could txt the number to a pop account, then have your * machine check that and then initiate the call to that number and call you on the other side |
| 03:14.39 | Faithful | heh... |
| 03:14.55 | austinnichols101 | diclophis: sure - but what's involved in txting the number (ie how many steps to place a call) |
| 03:15.18 | diclophis | well... just txt it, and the asterisk machine will call you |
| 03:15.25 | austinnichols101 | on my nextel I have to hit about a dozen things before I can send the stupid txt |
| 03:15.28 | diclophis | then when you pick up.. it calls the number you txted to it |
| 03:15.32 | diclophis | oh |
| 03:15.38 | diclophis | well.. yea that could be a pay |
| 03:15.43 | diclophis | er pain |
| 03:15.49 | austinnichols101 | so the nextel has a soon-to-be date with ebay |
| 03:16.02 | austinnichols101 | buh-bye |
| 03:16.21 | Faithful | that's exactly what you do with the TAPI interface or * manager |
| 03:17.43 | austinnichols101 | yeah - would be nice to have a web-based directory of my numbers that I access through a smartphone. Then when I click * would do it's thing |
| 03:18.23 | diclophis | you have unlimted data? |
| 03:18.27 | austinnichols101 | so I can write that part, but I'm still trying to work out the best way to connect the phone. apparently gsm is the best which narrows down which carrier I can use here in FL |
| 03:18.51 | austinnichols101 | tmobile and cingular, both of which kinda suck for reception |
| 03:19.09 | austinnichols101 | then there's verizon but they're cdma (bye-bye gsm gw) |
| 03:19.14 | diclophis | perhaps there are free public gsm gateways in existance? |
| 03:19.22 | diclophis | sorta like anon-proxies |
| 03:19.37 | austinnichols101 | that might be an interesting service offering |
| 03:19.43 | *** join/#asterisk coppice (n=chatzill@182.194.17.210.dyn.pacific.net.hk) |
| 03:21.03 | diclophis | i say go oldschool, get a 2 ham radios, on the * side hook it into a fxo port/channel that is always 'dialed in'... then some speach recognition to detect tyhe numbers you radio to it via your other ham radio |
| 03:21.27 | diclophis | once it gets the numbers the * machine dials you over a free DUNDI/SIP pstn dial-out service to your cell |
| 03:21.40 | diclophis | then dials the number the other half of the connection |
| 03:21.44 | austinnichols101 | any idea how much the 2n gateway goes for? |
| 03:21.45 | diclophis | bridges the two channels and your set |
| 03:22.13 | austinnichols101 | yeah - and then we could bounce the signal off the moon just for fun :) |
| 03:22.17 | diclophis | haha |
| 03:24.20 | austinnichols101 | faithful: does multi-channel require multi-sim? |
| 03:28.13 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
| 03:29.42 | coppice | don't bounce signals off the moon. latency, man, latency :-) |
| 03:29.59 | *** join/#asterisk Vyeperman (n=Vye@ip68-8-242-220.sd.sd.cox.net) |
| 03:31.09 | coppice | use the UTF-8 codec - if its worth saying, write an e-mail :-) |
| 03:31.21 | *** join/#asterisk Nag (n=NetAdmin@LSt-Amand-152-31-11-135.w82-127.abo.wanadoo.fr) |
| 03:31.24 | Nag | Hi ! |
| 03:34.24 | Nag | is someone use asterisk with wengo here please ? |
| 03:35.00 | project_2501 | when calling my phone, how do I set asterisk to drop out of voicemail and jump to voicemailmain so I can check my messages? |
| 03:35.21 | Qwell | project_2501: check out the stdexten macro. it has that |
| 03:38.57 | project_2501 | Qwell: I looked up stdexten on voip-info.org, but I'm not sure how to implement a macro like that |
| 03:39.21 | project_2501 | Qwell: any tips |
| 03:40.21 | project_2501 | Qwell: would I just copy that into my extensions.conf? I'm pretty sure thats what I have to do |
| 03:42.05 | *** join/#asterisk mogorman (n=mogorman@68.62.237.103) |
| 03:42.39 | Qwell | You could |
| 03:42.43 | Qwell | or you could create a macro |
| 03:42.59 | mogorman | macro-qwell |
| 03:43.03 | Qwell | yep |
| 03:43.29 | Qwell | <PROTECTED> |
| 03:52.33 | *** join/#asterisk bmg505 (n=leon@dsl-146-19-108.telkomadsl.co.za) |
| 03:54.21 | Nag | how can i do for when i receive a call on my wengo number, the asterisk server answer and give an holdon music while he call on my gsm please ? |
| 04:03.06 | *** join/#asterisk brettnem (n=brettnem@72.29.102.158) |
| 04:08.45 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 04:16.35 | file | Nag: yes |
| 04:16.51 | Nag | yes.. :-) |
| 04:16.56 | Nag | but how please :-) |
| 04:17.02 | file | show application dial |
| 04:17.06 | Nag | i 'm totally newbie with asterisk.. |
| 04:17.13 | Nag | i have mad the install this night |
| 04:17.18 | file | there's an option that provides musiconhold while trying to call the destination |
| 04:17.27 | Nag | yes, the m ? |
| 04:17.27 | Nugget | You are basically asking us to tell you how to do everything |
| 04:17.35 | file | Nag: yes |
| 04:17.36 | Nugget | Why not read the documentation first? |
| 04:17.58 | Nag | no Nugget --> i don't ask everything |
| 04:17.58 | file | Nugget: documentation is overrated, everyone just wants a personal support drone |
| 04:18.17 | Nag | i have the doc, and i am on the configuration since 6 hours.. |
| 04:18.20 | file | Nugget: and what's today's sauce? |
| 04:18.32 | Nag | i have make many many test configuration |
| 04:18.43 | Nugget | homemade sauce. butter and brown sugar. |
| 04:18.53 | file | Nugget: ooh exciting |
| 04:19.01 | Nugget | it's the weekend! cut loose! ;) |
| 04:21.43 | Nag | this is my error : |
| 04:21.44 | Nag | -- Executing Dial("SIP/morgan-hamart-dbb7", "SIP/morgan|60|tr") in new stack |
| 04:21.44 | Nag | Mar 12 05:20:27 NOTICE[23124]: app_dial.c:1011 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 04:21.44 | Nag | == Everyone is busy/congested at this time (1:0/0/1) |
| 04:21.44 | Nag | -- Executing Hangup("SIP/morgan-dbb7", "") in new stack |
| 04:21.45 | Nag | == Spawn extension (from-Wengo, morgan, 2) exited non-zero on 'SIP/morgan-dbb7' |
| 04:21.54 | Qwell | ~pb |
| 04:21.57 | jbot | methinks pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 04:21.58 | Qwell | not that loose |
| 04:24.32 | *** join/#asterisk gandhijee (i=HydraIRC@ip72-192-222-181.dc.dc.cox.net) |
| 04:24.53 | gandhijee | has anyone been able to get a hold of the Miomstone WP150? |
| 04:25.12 | gandhijee | *Moimstone |
| 04:28.16 | *** join/#asterisk Delmar (n=delmar@203-114-178-231.inspire.net.nz) |
| 04:29.40 | Damin | Oooooo...kkkkkk.... |
| 04:29.56 | Damin | I just heard calmriver.mp3 on a Vagisil commercial.. |
| 04:30.08 | Nugget | heh |
| 04:30.09 | Qwell | heh |
| 04:30.13 | Qwell | that's great |
| 04:30.29 | Qwell | gotta call it VoH then, eh? |
| 04:30.45 | Nugget | hah |
| 04:33.55 | project_2501 | Qwell: is there a way to tell voicemail to quit and jump to the next instruction in the dialplan, I have stdexten setup almost perfect, but this is my only problem |
| 04:34.36 | |ryan| | are there any USB based phone jacks that can be used with asterisk? |
| 04:34.44 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 04:34.53 | |ryan| | (i'm specificaly interested in running under linux) |
| 04:35.20 | project_2501 | when I push #, it's skips allison and then beeps and starts recording a message, then if I push # again, then it will skip to the next instruction, but it leaves a 2 second blank message before skipping! |
| 04:35.37 | gandhijee | ryan: http://x100p.com/products_2.htm |
| 04:35.44 | gandhijee | someone makes an 8 port one too |
| 04:35.58 | |ryan| | ugh |
| 04:36.08 | |ryan| | wat//// |
| 04:36.15 | Qwell | project_2501: needs to be * I believe |
| 04:36.19 | |ryan| | gandhijee: I asked about USB..... |
| 04:36.38 | Qwell | project_2501: use the a extension |
| 04:36.39 | gandhijee | umm its USB to phone |
| 04:36.47 | Qwell | a,1,VoicemailMain(something) |
| 04:36.59 | project_2501 | i tried pressing * and it didn't do anything, do I have to set it up in voicemail.conf? |
| 04:37.13 | *** part/#asterisk diclophis (n=diclophi@65.203.37.58) |
| 04:37.17 | |ryan| | gandhijee: that page does not mention USB anywhere. |
| 04:37.44 | gandhijee | oh wait. |
| 04:37.48 | gandhijee | thats the IAX box |
| 04:37.52 | project_2501 | Qwell: what do you mean by 'a' extension? |
| 04:40.51 | gandhijee | why not just use a network one? |
| 04:41.07 | gandhijee | ryan: do you not have a network card or something? |
| 04:41.15 | *** join/#asterisk coppice (n=chatzill@225.168.17.210.dyn.pacific.net.hk) |
| 04:45.05 | gandhijee | ryan: ???? |
| 04:52.13 | Nag | is it possible to JUST answer and get music (No Dial ? No Hangup) please ? |
| 04:54.10 | *** join/#asterisk pauldy (n=pauldy@m190e36d0.tmodns.net) |
| 04:54.15 | gandhijee | i guess if u have it just pickup the line and run the music |
| 04:54.41 | gandhijee | run it as background |
| 04:55.54 | gandhijee | exten => 1,s,Answer() |
| 04:56.43 | gandhijee | exten =>2,s,Background(path to music file) |
| 04:57.05 | gandhijee | wait that should be s,1 |
| 04:57.07 | gandhijee | and s,2 |
| 04:57.47 | gandhijee | that way they can imput something if they want and change the music |
| 04:58.10 | Nag | ok |
| 04:58.15 | Nag | i try thanks a lot |
| 05:09.36 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 05:13.45 | *** join/#asterisk tecnico (n=tecnico@user-24-236-120-2.knology.net) |
| 05:22.44 | *** join/#asterisk KaBewM (n=DA-MAN@66-215-7-106.dhcp.psdn.ca.charter.com) |
| 05:23.10 | *** join/#asterisk mattwj2005 (n=Matt@dialup-4.159.59.25.Dial1.Chicago1.Level3.net) |
| 05:23.22 | mattwj2005 | hi everyone |
| 05:23.50 | mattwj2005 | I got a VoIP question (not necessary Asterisk) but I thought you guys might know :) |
| 05:24.40 | KaBewM | ok |
| 05:24.48 | mattwj2005 | do you guys know of any sip chatrooms? |
| 05:25.03 | KaBewM | like Gizmo |
| 05:25.20 | KaBewM | or like irc.freenode.net in #SIP |
| 05:25.35 | mattwj2005 | like Gizmo |
| 05:25.39 | KaBewM | or like an irc/im like program that uses SIP |
| 05:25.43 | KaBewM | hehe yeah, Gizmo |
| 05:25.53 | Nugget | I'm envisioning 100 fat geeks, paralyzed, breathing into their mics waiting to hear if a girl calls and says anything. |
| 05:26.07 | mattwj2005 | lol :P |
| 05:26.46 | KaBewM | anything in gizmo that starts with 1222 is a chatroom with unlimited number of users |
| 05:27.02 | mattwj2005 | do you know any such rooms that don't require Gizmo? |
| 05:27.30 | KaBewM | yuou can go to sipphone.com and create an account, use any sip software with the gizmo network |
| 05:27.48 | KaBewM | not entirely sure what you want to do . . . |
| 05:28.51 | mattwj2005 | well I have always enjoyed voice chat like Yahoo IM....but frankly Yahoo Chat has a bit to be desired |
| 05:29.13 | Nag | can someone can just help me with this problem please : http://pastebin.ca/45301 ?? |
| 05:29.51 | KaBewM | any sip client + asterisk box will do |
| 05:29.58 | KaBewM | or sip client + sipphone acct |
| 05:30.50 | KaBewM | Nag, do you have mpg123 |
| 05:31.59 | Nag | KaBewM -> yes |
| 05:37.31 | mattwj2005 | so how do you dial a number using a sip software? |
| 05:40.21 | KaBewM | depends on the software |
| 05:41.24 | mattwj2005 | nm I figured out |
| 05:41.33 | mattwj2005 | anyone up for a good conference call? :P |
| 05:43.30 | KaBewM | naw its ok |
| 05:46.46 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 05:47.11 | *** join/#asterisk mattwj2005 (n=Matt@dialup-4.159.59.25.Dial1.Chicago1.Level3.net) |
| 05:47.48 | mattwj2005 | now is there a list of the rooms with people talking in them? |
| 05:48.55 | KaBewM | no |
| 05:48.57 | KaBewM | lol |
| 05:49.15 | mattwj2005 | :P |
| 05:50.00 | mattwj2005 | sip:12220000001@proxy01.sipphone.com if anyone is interested |
| 05:50.27 | *** join/#asterisk gandhijee (i=HydraIRC@ip72-192-222-181.dc.dc.cox.net) |
| 05:52.29 | *** join/#asterisk MGSsancho (n=user@adsl-67-126-143-33.dsl.irvnca.pacbell.net) |
| 05:56.56 | KaBewM | i cant seem to get on that one |
| 05:56.56 | KaBewM | hehe |
| 05:57.44 | mattwj2005 | try again kabewm |
| 05:57.45 | mattwj2005 | :) |
| 05:57.51 | KaBewM | hmm |
| 05:57.59 | mattwj2005 | I am using SJphone |
| 05:58.00 | KaBewM | it doesnt like that one 0 when i try |
| 05:58.13 | mattwj2005 | 0? |
| 05:58.50 | KaBewM | yeah if I cal 12222000001 |
| 05:58.58 | KaBewM | it works, but not 12220000001 |
| 05:59.43 | KaBewM | wait i think i got it to work |
| 05:59.59 | KaBewM | no guess not |
| 06:00.05 | KaBewM | cant seem to talk |
| 06:00.25 | mattwj2005 | I think I had something for a second there |
| 06:00.35 | KaBewM | maybe |
| 06:00.51 | mattwj2005 | it went silent for a sec |
| 06:00.58 | KaBewM | i gotta get newer software |
| 06:01.02 | KaBewM | i use old school firefly |
| 06:01.24 | mattwj2005 | got you |
| 06:01.31 | KaBewM | huh |
| 06:01.33 | KaBewM | no you dont |
| 06:01.33 | KaBewM | hehe |
| 06:01.39 | mattwj2005 | lol |
| 06:01.44 | mattwj2005 | haha :P |
| 06:02.39 | mattwj2005 | I really think this channel needs its own private conference number |
| 06:02.52 | KaBewM | its very possible |
| 06:03.07 | KaBewM | that it has one, but we just arent told what it is |
| 06:03.07 | KaBewM | haha |
| 06:04.07 | mattwj2005 | well if you want to try again....I am going down a number |
| 06:04.19 | mattwj2005 | there are people there....but one of them is a jerk |
| 06:04.25 | KaBewM | ah |
| 06:04.34 | KaBewM | i may try again shortly, wanna get sjphone though |
| 06:04.36 | KaBewM | is it good |
| 06:05.36 | mattwj2005 | FWD is good too....but I am on dialup |
| 06:05.46 | KaBewM | ah |
| 06:05.48 | KaBewM | wow |
| 06:05.59 | KaBewM | thats patience |
| 06:05.59 | KaBewM | hehe |
| 06:06.39 | mattwj2005 | I am actually moving tomorrow....I got a new job on Monday....the place I am moving to has 8 Meg service |
| 06:06.53 | mattwj2005 | but it is $55 a month |
| 06:07.34 | KaBewM | cool |
| 06:07.57 | KaBewM | i hate moving, but i must move soon as well |
| 06:09.07 | mattwj2005 | that is cool |
| 06:09.13 | mattwj2005 | do they have broadband? |
| 06:09.14 | mattwj2005 | :P |
| 06:11.07 | KaBewM | not sure where i M moving to yet |
| 06:11.26 | mattwj2005 | oh okay got you |
| 06:11.45 | giesen | has anyone managed to get a Cisco 7970 working with asterisk yet (with the new SIP firmware) |
| 06:11.55 | Qwell | giesen: sccp all the way. ;) |
| 06:11.59 | Qwell | giesen: actually, ask [av]bani later |
| 06:13.05 | giesen | the issue right now is I need a .cnf.xml file example |
| 06:13.36 | giesen | I have a call manager kicking around that I can use, but it looks like CM5.0 is unavailable as yet |
| 06:13.44 | giesen | so I cant even generate a config |
| 06:13.58 | Qwell | giesen: checked the wiki yet? |
| 06:14.04 | Qwell | I bet there is a sample 8.0 sip config up there |
| 06:14.08 | Qwell | or will be soon |
| 06:14.17 | giesen | Ive looked all over |
| 06:14.59 | KaBewM | i got a 7960 working with asterisk |
| 06:15.46 | giesen | yeah |
| 06:15.48 | giesen | Ive got several |
| 06:15.57 | giesen | two 7960s, two 7940s |
| 06:16.07 | giesen | the config for the 7970 is xml |
| 06:16.16 | giesen | and different from the other configs |
| 06:16.48 | giesen | Qwell: how do you find sccp compares in functionality to sip? |
| 06:16.53 | KaBewM | http://www.cisco.com/en/US/products/hw/phones/ps379/products_qanda_item09186a00801c963a.shtml <== says 7970 only supports sccp |
| 06:17.55 | giesen | KaBewM: cisco released sip firmware for it a couple days ago |
| 06:18.02 | KaBewM | oh |
| 06:18.06 | giesen | cisco just announced everything is going sip |
| 06:18.08 | Qwell | giesen: I use it, and love it |
| 06:18.11 | KaBewM | well then, i am mistaken |
| 06:18.12 | giesen | CallManager |
| 06:18.18 | giesen | and all its phoens |
| 06:18.46 | giesen | there's 8.2 SIP firmware for the 7940/7960, and 8.0 SIP firmware for the 7970 |
| 06:19.15 | giesen | Qwell: do you find SCCP deals with nat traversal issues better than SIP? |
| 06:19.37 | Qwell | giesen: We're 100% behind nat, so nat isn't an issue |
| 06:20.52 | giesen | ah |
| 06:21.03 | giesen | my issue is dealing with a bunch of different users |
| 06:21.08 | giesen | behind a bunch of different NATs |
| 06:21.18 | giesen | only really one problem user |
| 06:21.30 | mattwj2005 | Join the party!!!! sip:12220000000@proxy01.sipphone.com |
| 06:21.44 | mattwj2005 | now jerk free! :) |
| 06:21.45 | giesen | I would if I could get this blasted phone to work :/ |
| 06:23.18 | *** join/#asterisk Delmar (n=delmar@203-114-178-231.inspire.net.nz) |
| 06:28.32 | mattwj2005 | so what is wrong with this one user? |
| 06:28.38 | mattwj2005 | not connecting? |
| 06:29.10 | Nag | someone can just tell me if this line is ok in extensions.conf please : exten => _0144556677.,1,Dial(SIP/wengo/${EXTEN},240,T) |
| 06:31.03 | giesen | mattwj2005: firewall issues, phone goes unreachable constantly |
| 06:32.18 | giesen | that and he gets breakup and echo |
| 06:32.27 | giesen | which I havent experienced with mine yet |
| 06:32.44 | giesen | we just got these 7970s |
| 06:32.52 | giesen | and he couldnt get em to even show anything on the screen |
| 06:33.01 | giesen | and I had one booting inside of a half hour |
| 06:33.07 | giesen | and almost spooged when I saw the display light up |
| 06:33.34 | giesen | okay is anyone actually on the party line? |
| 06:33.47 | mattwj2005 | right now there is only music |
| 06:33.56 | giesen | yeah |
| 06:33.57 | giesen | that's all I hear |
| 06:34.01 | giesen | sounds like someone just joined |
| 06:34.04 | mattwj2005 | but I am hear just listening |
| 06:34.36 | giesen | is that you? |
| 06:34.40 | Nag | nobody can just answer to my little question please ? |
| 06:34.58 | mattwj2005 | the loud noises is some other guy |
| 06:35.06 | mattwj2005 | I think jerk is back |
| 06:38.07 | *** join/#asterisk liew123 (n=goh_mail@211.25.8.74) |
| 06:38.18 | giesen | I should order a pizza before the place closes |
| 06:38.54 | liew123 | hey who have done auto-dial out before? I have no luck doing so |
| 06:39.01 | *** join/#asterisk tsume (n=tsume@zanshin.tsumelabs.com) |
| 06:39.27 | tsume | I've a small problem :) This phone has BootROM 3.0.1, and I can't find where the SIP application version for this phone is anymore :) |
| 06:39.37 | tsume | can someone please clue me in? :) |
| 06:39.49 | Qwell | What phone? |
| 06:39.50 | tsume | its a Polycom 501 of course :) |
| 06:39.57 | Qwell | no clue |
| 06:40.01 | mattwj2005 | giesen........the firewall I am thinking DHCP, firewall port problem, and echo makes me think low bandwidth....that is just an educated guess |
| 06:40.11 | tsume | I can't seem to upgrade to version 3.1, else I'd just use the other SIP app |
| 06:42.13 | Nag | please , can you tell me the line for dial an external number in asterisk please ? (i receive a call on asterisk, and i transfer this to a GSM for exemple) |
| 06:42.37 | giesen | depends on what your device is for outbound calls |
| 06:42.41 | giesen | for example |
| 06:44.16 | Nag | Giesen --> excuse my english, but i dont understand your answer...is this line is ok ? --> exten => _0144556677.,1,Dial(SIP/wengo/${EXTEN},240,T) |
| 06:44.47 | giesen | exten => 5555551212,n,Dial(IAX2/user:password@iax.server.com/5555551213) |
| 06:45.04 | giesen | for IAX |
| 06:45.16 | Nag | i am in SIP with Wengo |
| 06:45.28 | giesen | and you have Wengo setup as a friend in sip.conf? |
| 06:45.49 | giesen | and you have that extension in your incoming context? |
| 06:46.03 | Nag | i have wengo in sip.conf |
| 06:46.14 | Nag | but no extension in incoming context |
| 06:46.16 | Nag | ?? |
| 06:46.26 | giesen | well then that's probably your problem |
| 06:47.54 | Nag | wich extension must i put in context ? |
| 06:48.23 | giesen | Nag you have to have a context to which all your incoming calls are routed |
| 06:48.39 | giesen | and then setup an extension in that context to do what you want |
| 06:49.02 | Nag | this is a part of my sip.conf --> http://pastebin.ca/45313 |
| 06:49.21 | Nag | and i will give you my extensions.conf |
| 06:49.35 | Nag | [wengo-in] |
| 06:49.35 | Nag | exten => _.,1,Goto(moncontexte,100,1) |
| 06:49.39 | Nag | that's it |
| 06:50.00 | giesen | where is the call coming in from? |
| 06:50.38 | Nag | this is wengo-in |
| 06:50.40 | Nag | ? |
| 06:51.41 | giesen | wait like 10-15 minutes and Ill come back and help you if I can |
| 06:51.45 | giesen | I gotta go pick up my pizza |
| 06:51.55 | Nag | ok, no problems |
| 06:52.00 | Nag | thansk a lot giesen |
| 06:52.20 | giesen | but if your incoming context is wengo-in |
| 06:52.26 | giesen | then you need to add the extension there |
| 06:53.20 | Nag | extension is my number where i want to forward the call .? |
| 06:53.28 | giesen | no |
| 06:53.36 | giesen | extension is _blah |
| 06:54.39 | Nag | [wengo-in] --> exten => _{I MUST PUT SOMETHING THERE}.,1,Goto(moncontexte,100,1) ? |
| 07:04.12 | justnulling2 | how can i find out what version 7960 is running? |
| 07:06.41 | giesen | justnulling2: it's under status |
| 07:06.57 | giesen | Settings > Status > Firmware Versions |
| 07:07.50 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 07:07.51 | justnulling2 | giesen: well i get protocol application invalid and none of the buttons work |
| 07:08.13 | giesen | probably corrupted firmware |
| 07:08.21 | giesen | try booting the phone without connecting it to the network |
| 07:10.56 | justnulling2 | giesen: ok now it gives me ethernet disconnected and still no bottons work |
| 07:14.33 | *** join/#asterisk riksta (n=rick@213.121.151.210) |
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| 07:19.34 | *** join/#asterisk Fedoracore6 (n=deddd@60.50.132.131) |
| 07:19.38 | Fedoracore6 | hai all |
| 07:22.08 | justnulling2 | giesen: that didn't help |
| 07:30.32 | giesen | you'll probably have to setup a tftp server then |
| 07:30.39 | giesen | with some firmware image |
| 07:30.50 | giesen | there's lots of info on the wiki |
| 07:33.10 | justnulling2 | giesen: ok and from what version should i start? |
| 07:33.25 | giesen | I'd start with the 5.x SCCP |
| 07:35.07 | tsume | I think this polycom is dead :) |
| 07:35.09 | tsume | its dead jim |
| 07:35.39 | *** part/#asterisk Nag (n=NetAdmin@LSt-Amand-152-31-11-135.w82-127.abo.wanadoo.fr) |
| 07:36.28 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
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| 07:55.51 | *** part/#asterisk mattwj2005 (n=Matt@dialup-4.159.59.25.Dial1.Chicago1.Level3.net) |
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| 08:00.18 | tuxinator_linux | busy night here in #asterisk |
| 08:00.37 | glm2k | hehe |
| 08:01.25 | tuxinator_linux | 1 day left till VON |
| 08:01.53 | tuxinator_linux | battery dead |
| 08:01.55 | tuxinator_linux | night |
| 08:08.26 | *** join/#asterisk konfuzed (n=Konf@H211.C18.B96.tor.eicat.ca) |
| 08:08.54 | *** part/#asterisk KaBewM (n=DA-MAN@66-215-7-106.dhcp.psdn.ca.charter.com) |
| 08:18.18 | [av]bani | tsume: killed a phone? |
| 08:19.07 | *** join/#asterisk justnulling2 (n=justnull@ool-18bcc906.dyn.optonline.net) |
| 08:23.57 | *** join/#asterisk axscode (n=asdfasdf@203.213.217.123) |
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| 08:32.11 | tsume | [av]bani: yeah :( |
| 08:32.50 | tsume | [av]bani: I loaded 3.0.1 on the phone because I couldn't get any sip I currently had to work. Now I can't load any of the sip apps from polycom, I downloaded them all from the polycom site and tried them all |
| 08:39.59 | *** part/#asterisk SwK_ (n=Silik0nJ@12-219-147-107.client.mchsi.com) |
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| 09:18.18 | HamYaI | where's everyone? |
| 09:19.33 | *** join/#asterisk liew123 (n=goh_mail@211.25.8.74) |
| 09:21.37 | liew123 | hi |
| 09:24.25 | liew123 | any one know why channel syntax is not recogine in .call file? |
| 09:32.01 | *** part/#asterisk HamYaI (i=HamYai@125.24.11.54) |
| 09:32.11 | tecnico | what are you typing to channel ? I have "Channel: IAX2/peer_name/numeric_caller_id" |
| 09:33.15 | tecnico | sorry.. not caller_id, but number_to_call |
| 09:33.54 | DaveHope | Would someone be kind enough to take a look at my dial plan and point out why I'm getting "pbx.c:1330 pbx_extension_helper: Cannot find extension context 'Internal'" errors ? - http://pastebin.com/596813 |
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| 09:43.59 | liew123 | Channel: Zap/g1/number to call |
| 09:58.29 | Fedoracore6 | hello all .. i try do code for access system, that user must enter same password in databases ,like this http://pastebin.com/597585 ... but fill |
| 09:58.53 | Fedoracore6 | cos when student put the wrong password, still can access the system ... |
| 10:08.57 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
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| 10:21.13 | tzafrir | Fedoracore6, where do you actually check the password? |
| 10:22.19 | tzafrir | the database query should involve the expected username, right? |
| 10:23.19 | tzafrir | anyway, have you tried using DISA? |
| 10:35.18 | Fedoracore6 | oic |
| 10:35.29 | Fedoracore6 | i must use DISA |
| 10:35.30 | Fedoracore6 | ok ok |
| 10:35.35 | Fedoracore6 | tzafrir thank allot |
| 10:35.43 | Fedoracore6 | i search disa |
| 10:37.48 | *** join/#asterisk saftsack (n=saftsack@p54A7D60A.dip.t-dialin.net) |
| 10:44.51 | tzafrir | It's not a must. But if you can easily set up a list of access codes, DISA will be just fine for you. Alternatively, lookup the expected access code in the database, and run DISA |
| 10:51.16 | Fedoracore6 | tzafrir its i must install disa |
| 10:51.19 | Fedoracore6 | or not |
| 10:54.03 | tzafrir | DISA is a standard part of astrerisk. if you don't autoload all modules you may need to load app_disa.so or whatever |
| 10:56.49 | Fedoracore6 | oic hemm i mast luad in modules.conf right |
| 11:00.23 | *** part/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
| 11:01.38 | DaveHope | tzafrir: You're evidently pretty knowledgeable regarding asterisk, fancy helping me with an idiot-level dialplan problem ? :) |
| 11:07.29 | tzafrir | If it's a short one... |
| 11:08.15 | DaveHope | I'm getting "pbx.c:1330 pbx_extension_helper: Cannot find extension context 'Internal'" errors and can't work out why as 'Internal' is listed in extensions.conf, http://pastebin.com/596813 |
| 11:10.43 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 11:18.18 | Fedoracore6 | tzafrir where i can find simple code for disa ... other from this--http://www.voip-info.org/wiki-Asterisk+cmd+DISA |
| 11:20.56 | tzafrir | DISA in itself does basically all the work. You just have to know the pin if you identify the caller by extension. Alternatively, you can use a file/db with a list of pins |
| 11:22.19 | Fedoracore6 | wakaaak tzafrir i dont understand what u say ...:( |
| 11:22.30 | Fedoracore6 | what mean pin |
| 11:22.59 | tzafrir | PIN: access code. call it "password" |
| 11:24.02 | Fedoracore6 | ahhahaha ok ok |
| 11:24.07 | Fedoracore6 | sorry |
| 11:24.08 | *** join/#asterisk saftsack (n=oliver@p54A7F03F.dip.t-dialin.net) |
| 11:24.08 | saftsack | hi |
| 11:24.28 | saftsack | i was on cebit yesterday and i heart that digium will release a bri isdn card :) |
| 11:26.03 | DaveHope | tzafrir: Any ideas ? :) - |
| 11:29.02 | *** join/#asterisk astra^^ (n=muhajir_@59.145.104.74) |
| 11:31.04 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 11:35.52 | astra^^ | hello all |
| 11:50.44 | DaveHope | hrm. this is _really_ weird, 'show dialplan' doesn't show the dial plan either. It just shows the parking one. |
| 11:50.48 | DaveHope | *sigh* |
| 11:56.58 | RoyK | load pbx_config.so :P |
| 11:57.22 | RoyK | DaveHope: you probably haven't load pbx_config.so |
| 11:57.46 | DaveHope | RoyK: Will take a look, thanks :) |
| 11:58.33 | DaveHope | RoyK: Bingo! - Thanks. |
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| 12:03.13 | riksta | hey, i'm just in the process of converting to realtime for extensions, but have a problem... i have a line in extensions.conf "exten => 4484XXXXX,4,Dial(${MYPHONE},20)" but when i put the data "Dial" into app and "${MYPHONE},20" in appdata asterisk grumbles that it doesnt like the ",20" in the appdata |
| 12:03.16 | riksta | how do i solve that please? |
| 12:04.14 | riksta | <PROTECTED> |
| 12:04.14 | riksta | Mar 12 12:01:06 WARNING[24630]: chan_sip.c:1970 create_addr: No such host: 1000,20 |
| 12:06.30 | riksta | ahh gotta use pipe not comma |
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| 12:37.23 | riksta | sipgate broken againnnn yawn |
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| 13:08.16 | freat | <yawn> |
| 13:08.30 | CMike | hi all |
| 13:08.43 | freat | Hi CMike |
| 13:08.49 | freat | brb... |
| 13:09.35 | riksta | is there a web gui i can use to edit extensions in realtime mysql db? can't find one |
| 13:09.56 | Aurs | phpmyadmin |
| 13:11.10 | riksta | hehe |
| 13:11.22 | riksta | not that, you know what i mean, something more managable |
| 13:30.59 | *** join/#asterisk saftsack (n=oliver@IP-213188106101.dialin.heagmedianet.de) |
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| 13:31.11 | saftsack | hi i search for a music on hold file (mp3) |
| 13:31.26 | saftsack | i couldnt find anything in google :( just for paying |
| 13:37.30 | fugitivo | www.freeplaymusic.com |
| 13:44.42 | saftsack | thanks :) |
| 13:45.04 | *** join/#asterisk linuxgeekuk (n=paul@host217-41-10-28.in-addr.btopenworld.com) |
| 13:46.26 | linuxgeekuk | hi all anyone know about iaxy |
| 13:47.35 | linuxgeekuk | as I have a cracker of a problem |
| 13:54.09 | freat | hi linuxgeekuk |
| 13:54.15 | freat | what's up w/ your iaxy? |
| 13:54.55 | linuxgeekuk | got one way speach but only on in coming calls |
| 13:55.29 | freat | the iaxy is remote? |
| 13:55.43 | linuxgeekuk | no same network same switch |
| 13:56.22 | freat | if you call it it rings and you can answer? but no audio at all? |
| 13:56.53 | freat | then if it calls... you have 1 way audio? |
| 13:56.57 | linuxgeekuk | it works fine calling from a sip extension |
| 13:57.04 | freat | ahh |
| 13:57.20 | freat | put notransfer=yes into the context for it in iax.conf |
| 13:57.29 | linuxgeekuk | but from a pstn call you get in coming audio but no out going |
| 13:58.00 | freat | yeah you'll need to use that notransfer=yes... asterisk is trying to hand off the call. I'm assuming your pstn is actually an IAX provider? |
| 13:58.12 | *** join/#asterisk coppice (n=chatzill@33.192.17.210.dyn.pacific.net.hk) |
| 13:58.26 | linuxgeekuk | no its on a tdm card |
| 13:58.32 | freat | ooh... |
| 13:59.11 | freat | well, since the iaxy is on your network, this is odd. I would probably add: |
| 13:59.15 | freat | notransfer=yes |
| 13:59.27 | freat | forcejitterbuffer=no |
| 13:59.31 | freat | jitterbuffer=no |
| 14:00.11 | freat | I've got some iaxy's out there.... logging into machine |
| 14:00.41 | linuxgeekuk | just trying those changes |
| 14:01.31 | *** join/#asterisk antoniojulio (n=antonioj@unaffiliated/antoniojulio) |
| 14:02.10 | linuxgeekuk | those changes dont change it |
| 14:02.12 | freat | here's my config from iax.conf |
| 14:02.12 | freat | http://pastebin.ca/45341 |
| 14:03.34 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 14:07.08 | Zeeek | astricon |
| 14:07.36 | freat | linuxgeekuk: this also has my general context in it: http://pastebin.ca/45342 |
| 14:08.23 | *** join/#asterisk stuartcw (n=chatzill@softbank221025056004.bbtec.net) |
| 14:09.50 | Lino` | hmmm |
| 14:09.51 | linuxgeekuk | that great it now works looks like there was a problem with amportal |
| 14:10.03 | Lino` | i have an HFC sitecom card running with bristuff |
| 14:10.06 | linuxgeekuk | many thanks freat |
| 14:10.23 | Lino` | where can i change the mode it is running in? |
| 14:10.32 | Lino` | NT mode / etc |
| 14:10.43 | freat | linuxgeekuk: cool! |
| 14:11.03 | freat | linuxgeekuk: amportal restart eh? |
| 14:11.06 | freat | :) |
| 14:11.16 | *** join/#asterisk Flauto (n=zhao@71.194.194.48) |
| 14:11.29 | linuxgeekuk | you know I could not find any real info on the iaxy on google just digiums pdf |
| 14:11.50 | Zeeek | what d'you wanna know about iaxy? |
| 14:11.54 | Flauto | good morning |
| 14:12.30 | linuxgeekuk | zeeek thanks but freat has fixed it |
| 14:12.44 | Flauto | i am using vbuzzer but since yesterday, it started failing registering |
| 14:12.49 | Flauto | any idea about it |
| 14:12.55 | Zeeek | I thought you wanted to know something about it |
| 14:13.00 | freat | linuxgeekuk: yeah the iaxy is kind of a pain. I do have a few hooked up though. I just wish they would put the MAC address on the damn things |
| 14:13.26 | linuxgeekuk | yes would be good I found it with nmap |
| 14:13.43 | Zeeek | can't you see it on your router? |
| 14:13.44 | freat | then at least with I could do: nmap -sP | grep daMac |
| 14:14.48 | freat | Zeeek: there's lots of times you may go to deploy something like that, and not have access to the DHCP server on the network |
| 14:14.56 | linuxgeekuk | I use nmap -sU -p4569,5036,5060 192.168.x.x-x.x |
| 14:15.11 | fugitivo | -sU is udp only |
| 14:15.38 | linuxgeekuk | yep iaxy is a udp device |
| 14:15.42 | freat | fugitivo = nmap guru |
| 14:15.42 | fugitivo | -sP is ping scan |
| 14:16.12 | ambriento | jejeje |
| 14:16.47 | Flauto | what is new in 1.2.5 |
| 14:17.16 | ambriento | flauto, http://ftp.digium.com/pub/telephony/asterisk/ChangeLog-1.2.5 |
| 14:17.20 | Zeeek | echo has been totally eliminated |
| 14:17.38 | fugitivo | yeah, right |
| 14:17.49 | freat | ambriento beat /me to the link |
| 14:18.15 | nettie | Hey guys anyone know the right syntax to dial a sip extension defined in a different context please? |
| 14:19.08 | ambriento | You need to go to the other context first |
| 14:19.10 | ambriento | then you dial |
| 14:19.11 | freat | usually one would include the context... in this case you don't want to I guess? |
| 14:19.46 | Zeeek | usually the point of having contexts is to forbid the dialing of the other |
| 14:19.52 | nettie | yeah exactly.. I'm trying to configure an automated attendant |
| 14:20.07 | nettie | I just want to point the extensions I need |
| 14:20.35 | freat | well, I include my internal extensions in the auto attendant context |
| 14:20.37 | nettie | so when the caller push "1" the sip phone in office context rings |
| 14:20.38 | fugitivo | use include |
| 14:21.03 | nettie | I dont want him to be able to exploit other extensions |
| 14:21.06 | Flauto | anyone is using vbuzzer, |
| 14:21.13 | Flauto | i have time out problem with them |
| 14:21.23 | Flauto | is it my problem or their server |
| 14:21.27 | Curus | nettie: You can just goto(office, 1234567, 1) |
| 14:21.32 | Flauto | i really did not do anything to it |
| 14:21.34 | Zeeek | nettie you may want to tread the pages about dialplan on the wiki |
| 14:21.38 | Flauto | i still can make phone calls |
| 14:21.45 | fugitivo | Curus solution will work |
| 14:21.51 | Curus | I love goto |
| 14:21.52 | freat | nettie: then you should split out the exploitable extensions into a separate context |
| 14:21.55 | Curus | Screw Dijkstra |
| 14:22.03 | nettie | Curus, Zeeek thanx I'll look at tit |
| 14:22.20 | Zeeek | http://www.voip-info.org/wiki/index.php?page=Asterisk+config+extensions.conf |
| 14:22.58 | Zeeek | the idea of the contexts (one idea) is that you have them partitioned in areas like office, bedroom, harem... |
| 14:23.05 | kippi | how can you get asterisk to pick up what number is rining your system? |
| 14:23.12 | Zeeek | you can include the harem in the bedroom but not the office :) |
| 14:23.25 | freat | Zeeek: nice one |
| 14:23.47 | ambriento | maybe Dial(SIP/Channell,30,GOTO:<context>^<extensio>n^<priority>) |
| 14:23.58 | Zeeek | so if you have a privileged user his context includes ALL others. The limited user's context includes only a few like "inhouse" or something |
| 14:24.15 | Zeeek | that's what I do, anyway |
| 14:24.37 | Zeeek | so some people can dial any service and some can only dial inhouse extensions |
| 14:24.53 | Curus | I used to use lots of includes, but these days I mostly prefer to use Goto |
| 14:24.55 | Zeeek | also, you want to make sure the default context is limited |
| 14:25.07 | Zeeek | goto is fine if that's what you like |
| 14:25.39 | Zeeek | test |
| 14:25.46 | Curus | test test |
| 14:25.50 | Zeeek | test test |
| 14:25.54 | Curus | Heh |
| 14:25.56 | linuxgeekuk | test test |
| 14:25.58 | Zeeek | color |
| 14:26.33 | nettie | guys another qustion.. what's the best way to name sip users? |
| 14:26.34 | nettie | I mean |
| 14:26.59 | Zeeek | I used to name them for their extension. Then I ran into problems with that |
| 14:27.07 | Zeeek | now I name them for phones |
| 14:27.16 | nettie | the test inside square brackets in sip.conf |
| 14:27.21 | nettie | test=text |
| 14:27.26 | Zeeek | if I had extername users I'd name them with their nbames or user names |
| 14:27.34 | Curus | nettie: If you name them for some part of the phone number, you can sometimes get away with _1235XX,1,Dial(SIP/${EXTEN:-2}) |
| 14:27.57 | Zeeek | that's not a name it's an extension |
| 14:28.07 | Curus | Oh |
| 14:28.07 | Zeeek | oh I see what you meant |
| 14:28.28 | Curus | It just gets nasty if someone decides they want two phones on one number |
| 14:28.49 | Zeeek | I have multiple phones on most extensions |
| 14:28.49 | nettie | yeah |
| 14:29.01 | nettie | I have this problem too |
| 14:29.03 | Zeeek | which is why I now use phone names |
| 14:29.10 | nettie | zeek |
| 14:29.13 | DaveHope | Anyone fancy going through why sipgate calls arn't working for me ? - Spent ~3hrs playing with it and so far, no luck :( - http://pastebin.com/597838 |
| 14:29.29 | nettie | you just give the phone a fantasy name? |
| 14:29.31 | Zeeek | incoming or outgoing? |
| 14:29.39 | Zeeek | DaveHope ^^^ |
| 14:29.40 | DaveHope | Zeeek: Both |
| 14:30.02 | Curus | By the way, Snom phones like to put their user name in the display, and this is rather annoying in my case |
| 14:30.12 | Curus | Is there a way to make them put something else there? |
| 14:30.26 | Zeeek | why no alpha LED? |
| 14:30.43 | Curus | ? |
| 14:31.35 | Zeeek | DaveHope remove nat=yes |
| 14:31.44 | DaveHope | Zeeek: Okies...Removing now. |
| 14:32.02 | Lino` | do you do the register correctly?` |
| 14:32.09 | Lino` | register => sip-ID:sippasswort@sipgate.de/sip-ID |
| 14:32.25 | DaveHope | Lino`: Yep. I register correctly. |
| 14:33.18 | DaveHope | Zeeek: Ooh. Nice. I now get a notice I kinda understand :) - 'Cannot find extension context 'default' |
| 14:33.21 | Zeeek | DaveHope also you have no codec specified? |
| 14:33.34 | nettie | can I specify more extensions is a goto line? does the & works there? |
| 14:33.41 | Zeeek | so you have no default? |
| 14:34.00 | Zeeek | nettie just create an extension that does the dial |
| 14:34.04 | DaveHope | Zeeek: I have what I put on pastebin :) |
| 14:34.20 | Zeeek | exten => heheh,1,Dial(something&somebody_else) |
| 14:34.29 | nettie | and then point to it |
| 14:34.30 | nettie | clean |
| 14:34.32 | nettie | thanx |
| 14:34.52 | Zeeek | DaveHope you have no default context it would seem |
| 14:35.06 | DaveHope | Zeeek: Writing one now :P |
| 14:35.20 | Zeeek | or put context=incoming-sipgate |
| 14:35.30 | Zeeek | and then write the extension logic there |
| 14:35.44 | Zeeek | [incoming-sipgate] (put the stuff underhere) |
| 14:36.12 | Zeeek | this is especially useful if you have a bunch of incoming providers and you want to do different stuff |
| 14:36.16 | Zeeek | with each one |
| 14:36.34 | Zeeek | like maybe FWD goes to one phone and sipgate to another |
| 14:36.37 | *** join/#asterisk puzzled (n=yeahrigh@puzzled.xs4all.nl) |
| 14:36.38 | DaveHope | Zeeek: Okies, will have a go ) |
| 14:36.50 | Zeeek | I have about 20 provoders |
| 14:36.51 | DaveHope | Zeeek: Thanks. Will no doubt ask for some more help in a few mins :P |
| 14:37.25 | *** join/#asterisk TUplink (n=Tommy@68-232-82-147.chvlva.adelphia.net) |
| 14:37.37 | TUplink | how do i find where my sound files are suposed to be? |
| 14:37.40 | puzzled | hi |
| 14:37.52 | Zeeek | TUplink which ones |
| 14:38.09 | Zeeek | they all should be under /var/lib/asterisk/sounds |
| 14:38.09 | TUplink | all the gsm's |
| 14:38.21 | Zeeek | ^^^^ |
| 14:38.46 | TUplink | well i do background(welcome) and it says Mar 12 09:35:57 WARNING[802]: file.c:509 ast_openstream_full: File welcome does not exist in any format |
| 14:39.11 | TUplink | its there |
| 14:39.23 | Zeeek | and you looked in the above directory? |
| 14:40.12 | DaveHope | Zeeek: Still no luck, It seems to see the incommiung call now though. (tries to do update_user_counter: 9448153) |
| 14:40.51 | Zeeek | DaveHope it sees it but the call doesn't come in? Or what? |
| 14:41.32 | DaveHope | Zeeek: When I call my sipgate number from my mobile, asterisk sees the incomming call but my mobile gets disconnected instantly. |
| 14:41.58 | Zeeek | what does the console say? |
| 14:42.26 | *** join/#asterisk Phoenix[Y-LAN] (n=phoenix@80.237.162.90) |
| 14:42.29 | DaveHope | Zeeek: I'll re-post my config files and log on pastebin, sec :) |
| 14:42.30 | Phoenix[Y-LAN] | hi |
| 14:42.47 | *** join/#asterisk TUplink (n=Tommy@68-232-82-147.chvlva.adelphia.net) |
| 14:42.52 | Zeeek | what does you dial line look like also? |
| 14:42.53 | TUplink | Server# ls -alg /var/lib/asterisk/sounds/wel* |
| 14:42.53 | TUplink | -rw-r--r-- 1 root wheel 1584 Apr 12 2005 /var/lib/asterisk/sounds/welcome.gsm |
| 14:43.04 | TUplink | its there |
| 14:43.19 | TUplink | i just upgraded from 1.0.3 |
| 14:43.34 | nettie | Zeeek I tried what you suggested the problem is that I'n using macros to handle extensions and voicemail so if I point directly to the extension I lose the macro configuration. |
| 14:44.42 | Zeeek | I use macros too, but I gues I have specific ones for each need |
| 14:44.54 | *** join/#asterisk ToTo (n=ToTo@host10-138.pool872.interbusiness.it) |
| 14:45.10 | Zeeek | also, you can pass an arg to the macro with who to dial |
| 14:45.32 | nettie | ok |
| 14:45.34 | nettie | got it |
| 14:45.40 | nettie | you're right |
| 14:45.46 | Zeeek | exten => s,1,Macro(mymac,SIP/1&IAX2/2,45,opt) |
| 14:46.29 | Zeeek | in the macro Dial(${ARG1},${ARG2} etc |
| 14:47.06 | Zeeek | TUplink are you using a language other than english? |
| 14:47.19 | TUplink | no... |
| 14:47.26 | linuxgeekuk | freat the critical lines that where needed where disallow=all allow=ulaw |
| 14:47.33 | TUplink | i dont understand it |
| 14:47.38 | TUplink | it worked b4 |
| 14:48.02 | Zeeek | I hate when that happens |
| 14:48.04 | DaveHope | Zeeek: http://pastebin.com/597865 |
| 14:48.27 | *** join/#asterisk FarrisG (n=jrush@65.91.221.131) |
| 14:48.37 | *** join/#asterisk willt (n=willt@wifi-napanet-static-206-81-99-68.napanet.net) |
| 14:48.57 | DaveHope | Zeeek: the messages is what is written to /var/log/.../messages when I call my sipgate number from my mobile |
| 14:49.11 | Zeeek | hold on... no |
| 14:49.12 | FarrisG | anybody know where the hell I can find AT&T's business support line? Our friggin' PRI got cut last night, and none of my contacts are answering the damn phone |
| 14:49.24 | Zeeek | not [default] context = |
| 14:49.48 | willt | FarrisG what area are you in? |
| 14:49.56 | Zeeek | you need a context that contains the extension you want sipgate to dial |
| 14:50.13 | TUplink | how do i find the sound dir? |
| 14:50.15 | FarrisG | willt: "Southwest". Dallas |
| 14:50.29 | willt | hmm ill give you a number I have it may work just a sec |
| 14:50.32 | TUplink | the demo-echotest works |
| 14:50.34 | Zeeek | for example [sipgate-in] then add s,1,Dial(SIP/phone) |
| 14:50.58 | FarrisG | Hey, I'll start anywhere, this is getting ridiculous |
| 14:51.14 | Zeeek | DaveHope also sipgate looks like it's not working (on their end) |
| 14:51.41 | DaveHope | Zeeek: hrm, ok. Thanks. Will try and follow the other bits you said to do :) |
| 14:51.50 | DaveHope | Zeeek: (not entirely sure what I'm doing though :P) |
| 14:51.54 | Zeeek | don't they have a page with instructions? |
| 14:52.01 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 14:52.09 | Zeeek | I use sipgate.de - well I did at one time |
| 14:52.09 | willt | 1-800-332-1321 is the number I call for all my AT&T lines |
| 14:52.26 | *** join/#asterisk Parallax_ (n=pbl@85.155.10.145) |
| 14:52.43 | willt | anyone have any recomendations on good soft phones? |
| 14:52.43 | FarrisG | willt: thanks |
| 14:52.46 | DaveHope | Zeeek: The example I used from their site didn't work for me either, but since they say they don't support asterisk configuration I figured I'd ask the asterisk community :) |
| 14:52.46 | willt | np |
| 14:53.13 | Zeeek | DaveHope what you have can't work. Also the console message says you're not registered with them |
| 14:53.16 | TUplink | they moved them... i found them /usr/local/share/asterisk/sounds |
| 14:53.23 | Zeeek | oh my! |
| 14:53.45 | Zeeek | DaveHope sip show registery |
| 14:54.04 | Zeeek | sip show reg<tab> |
| 14:54.09 | DaveHope | Zeeek: State = Registered |
| 14:54.18 | Zeeek | not according to the console message though |
| 14:54.36 | Zeeek | Registration for '9448153@sipgate.co.uk' timed out, trying again |
| 14:54.54 | DaveHope | Zeeek: will try calling again then. |
| 14:54.59 | Parallax_ | what Can asterisk do with a standart PCI Modem ???? |
| 14:55.13 | Zeeek | nothing |
| 14:55.23 | Flauto | after i upgraded the 1.2.5, it is telling me app_saycountpl.so and format_mp3.so are not compatible |
| 14:55.28 | Parallax_ | not even receving fax? |
| 14:55.38 | Zeeek | DaveHope listen, go find out what a context is or you'll never get there |
| 14:55.51 | Zeeek | check out http://asteriskdocs.org |
| 14:55.53 | DaveHope | Zeeek: Ok. |
| 14:55.58 | DaveHope | Zeeek: Thanks for your help. |
| 14:56.02 | Zeeek | np |
| 14:56.21 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 14:56.35 | Zeeek | Flauto you may have to delete modules before the make |
| 14:56.51 | Zeeek | I've had that problem before |
| 14:56.53 | Flauto | zeeek, thanks |
| 14:57.18 | Zeeek | or rename the modules dir and create a new empty one before the make |
| 14:57.55 | Flauto | it is under /usr/lib/asterisk/modules right? |
| 14:58.33 | tclark | way long shot here any body have an e&m wink config for a dialogic board in windoze box, eg the paraneters file with all thiose settting number that ahex flag value ? |
| 14:58.37 | Zeeek | Flauto ya |
| 14:58.43 | Flauto | thanks |
| 14:59.08 | Flauto | zeeek, i also use vbuzzer, i had no problem with it but now, it is timed out |
| 14:59.11 | Flauto | i dont' know why |
| 14:59.17 | Flauto | i checked the setting |
| 14:59.20 | Flauto | nothing was wrong |
| 14:59.32 | Flauto | and i can still make outbound calls through the service |
| 15:00.03 | Zeeek | what is that? |
| 15:00.18 | Zeeek | a service, ok |
| 15:00.38 | Flauto | yes |
| 15:01.24 | Flauto | the service does not really support asterisk officially |
| 15:01.36 | Flauto | but it is listed under voip-info |
| 15:02.14 | Flauto | i have used it for a few months now |
| 15:02.14 | Flauto | and it has been working well |
| 15:02.14 | Flauto | only since yesterday |
| 15:02.14 | Flauto | it stopped working |
| 15:02.14 | Flauto | only inbound |
| 15:02.17 | Flauto | and showing timed out |
| 15:02.37 | Flauto | do you thing that they changed something from their end? |
| 15:03.23 | *** join/#asterisk saftsack (n=oliver@IP-213188106101.dialin.heagmedianet.de) |
| 15:03.25 | saftsack | hi |
| 15:03.54 | saftsack | is it possible to enable overlapdial within extensions.conf? |
| 15:05.27 | DaveHope | Zeeek: I've read up on dialplans, contexts, priorities and applications and am redoing what I've done. Mind if I re-ask you in a few moments ? :) |
| 15:06.41 | *** part/#asterisk Parallax_ (n=pbl@85.155.10.145) |
| 15:07.15 | RoyK | what is netsec? |
| 15:11.15 | Zeeek | I'll be here in a fe, I have to finish ironing! |
| 15:12.06 | willt | whats better sjphone or x-lite? |
| 15:13.19 | *** join/#asterisk Tamarisk (n=adrian@user-5659.l6.c4.dsl.pol.co.uk) |
| 15:14.30 | DaveHope | Zeeek: Still getting similar problems, if you (or anyone else) could point out why it'd be greatly appreciated! - http://pastebin.com/597904 |
| 15:17.41 | Zeeek | what is disable? It's disallow |
| 15:17.44 | *** part/#asterisk mover (n=dlu@gw-dus-net.dus.de.ncore.net) |
| 15:18.04 | *** join/#asterisk pengyong (n=lala@218.93.159.168) |
| 15:18.24 | Zeeek | where is this coming from? 9448153 is not a local user |
| 15:18.55 | tclark | doers any one have any dialogic spandti.prm that will talk to an asterisk e&m,pri any thing :) ? |
| 15:19.03 | DaveHope | Zeeek: It's me calling my sipgate number from my mobile phone., |
| 15:20.14 | Zeeek | well it's trying to tell yyou that there's no such number |
| 15:20.33 | Zeeek | I told you before that default stuff is wrong |
| 15:20.59 | Zeeek | # |
| 15:21.02 | Zeeek | [Sipgate-Outgoing] |
| 15:21.05 | TUplink | i need help asterisk wont start and wont output any messages |
| 15:21.08 | Zeeek | <PROTECTED> |
| 15:21.09 | DaveHope | Zeeek: In which case, I've obviously not understood you. sorry. |
| 15:21.32 | TUplink | what is this Asterisk Dynamic Loader Starting: |
| 15:21.32 | TUplink | <PROTECTED> |
| 15:21.32 | TUplink | /libexec/ld-elf.so.1: /usr/lib/asterisk/modules/res_adsi.so: Undefined symbol "ast_load" |
| 15:21.33 | Zeeek | then add a context [incoming-sipgate] to extensions.conf |
| 15:21.58 | Zeeek | then add exten => 9876543,1,Dial(number,time,opt) in that context |
| 15:22.17 | Zeeek | incoming has only ONE m btw |
| 15:23.08 | DaveHope | Ok, done all that. |
| 15:23.14 | DaveHope | (and corrected the typos :P) |
| 15:23.34 | TUplink | what is res_adsi |
| 15:23.34 | Zeeek | http://pastebin.com/597919 |
| 15:23.50 | Zeeek | to send stuff to adsi phones |
| 15:24.55 | saftsack | if i type 012345678 on my phone just the 0 is dialed |
| 15:25.10 | DaveHope | Zeeek: You'd like me to replace my sip.conf with that, I persume ? |
| 15:25.21 | Zeeek | http://pastebin.com/597923 |
| 15:25.38 | Zeeek | comment the old one out and put that one in maybe |
| 15:25.53 | Zeeek | then add thre above to extensions.conf. Then reload sip and extensions |
| 15:26.16 | rpm | anyone use link2voip.com? |
| 15:26.42 | Zeeek | saftsack sounds like a digitmap issue |
| 15:26.51 | Zeeek | what phone? |
| 15:26.57 | saftsack | isdn phone |
| 15:27.16 | saftsack | i do a goto first and then dialing |
| 15:27.22 | Zeeek | the phone is sending the 0 too soon? |
| 15:27.27 | saftsack | <PROTECTED> |
| 15:27.36 | saftsack | [outgoing] |
| 15:27.40 | saftsack | <PROTECTED> |
| 15:27.58 | Zeeek | why are you doing thazt? |
| 15:28.17 | Zeeek | the desired result is what exactly? |
| 15:28.30 | saftsack | calling another party |
| 15:28.41 | saftsack | i did an extra extension for internal and external dialing |
| 15:28.47 | saftsack | <PROTECTED> |
| 15:28.54 | saftsack | this here waits for the decision |
| 15:29.16 | Zeeek | what is the 0 for? |
| 15:29.30 | saftsack | i must do this because i want to play a different dialtone after taking the handset |
| 15:29.36 | saftsack | for dialing out |
| 15:29.56 | fugitivo | Zeeek: he wants to press 0 and get dialtone |
| 15:30.01 | Zeeek | yeah |
| 15:30.08 | saftsack | and become a pseudo dialtone before |
| 15:30.09 | riksta | Mar 12 15:28:44 WARNING[22729]: chan_zap.c:8162 zt_pri_error: 1 updating callstate, ourcallstate 19 to 12 anyone seen this with a sangoma card? |
| 15:30.19 | TUplink | what dose this mean /libexec/ld-elf.so.1: /usr/lib/asterisk/modules/res_adsi.so: Undefined symbol "ast_load" |
| 15:30.27 | Zeeek | dooesn't ignore pat do that? |
| 15:30.29 | fugitivo | TUplink: you're missing a libray |
| 15:30.31 | fugitivo | library |
| 15:30.44 | saftsack | Zeeek, ignorepat does the half |
| 15:30.47 | TUplink | witch libray |
| 15:30.56 | fugitivo | i don't know |
| 15:30.56 | saftsack | because the first sound after taking the handset must be a pseudo dialtone |
| 15:30.59 | Zeeek | saftsack post the dialplan part to pastebin |
| 15:31.37 | fugitivo | TUplink: undefined symbol means that it can't find where "ast_load" is defined, usually it's defined inside a library, and maybe you don't have that library |
| 15:31.58 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 15:32.03 | saftsack | http://pastebin.com/597931 |
| 15:33.11 | Zeeek | So you dial 0, then hear a dialtone, then you dial your number? |
| 15:33.34 | saftsack | hearing a dialtone, then dial a 0 then hearing a normal dialtone and then dialing the number |
| 15:33.48 | Zeeek | and what happens? |
| 15:33.54 | fugitivo | why you need to do that? |
| 15:33.54 | saftsack | the first dialtone should be different |
| 15:34.12 | Zeeek | the first dialtone is the phone, no? |
| 15:34.14 | TUplink | so what do i do bout the ast_load? |
| 15:34.25 | fugitivo | TUplink: find out what library is missing |
| 15:34.33 | TUplink | how do i do that? |
| 15:34.44 | TUplink | [res_adsi.so] => (ADSI Resource) |
| 15:34.47 | saftsack | because it is normal, that theres another dialtone before getting a trunk |
| 15:35.08 | TUplink | that line comes up before the ast_load |
| 15:35.33 | saftsack | fugitivo, is it possible to enable overlapdial within the dialplan? |
| 15:36.08 | Zeeek | saftsack is the waitexten working? |
| 15:36.21 | saftsack | yes |
| 15:36.30 | saftsack | the extension 0 is working too |
| 15:36.32 | TUplink | i have an idea |
| 15:36.41 | Zeeek | so I'd dial 0, then 0123456789 ? |
| 15:36.41 | saftsack | but then it dials just 0 in the dialstring |
| 15:36.46 | saftsack | and then i get congestion |
| 15:36.57 | saftsack | because i have no chance for further dialing |
| 15:37.11 | Zeeek | yes because EXTEN is 0 ! how about that! |
| 15:37.23 | saftsack | exactly |
| 15:37.36 | fugitivo | saftsack: i think you should do this |
| 15:37.44 | saftsack | HOW? ^^ |
| 15:37.54 | riksta | it's not rocket science |
| 15:38.02 | TUplink | errrr how o i find out witch module is fobar |
| 15:38.04 | fugitivo | saftsack: exten => 0,1,Dial(Zap/g2,t) |
| 15:38.11 | fugitivo | saftsack: that'll give you normal tone |
| 15:38.15 | fugitivo | and you can dial the number |
| 15:38.24 | Zeeek | did you look at this? http://www.voip-info.org/wiki-Asterisk+cmd+WaitExten |
| 15:39.43 | saftsack | didnt get you fugitivo |
| 15:40.08 | fugitivo | fugitivo: if i understand what you need, exten => 0,1,Dial(Zap/g2,t) should work |
| 15:40.29 | Zeeek | "This command is particularly interesting as it pushes the received extension back on to the stack and then restarts the current context." |
| 15:40.30 | fugitivo | you pick up the phone and get asterisk dialtone, press 0, and get pstn dialtone |
| 15:40.35 | saftsack | ok |
| 15:40.43 | fugitivo | then you dial the number |
| 15:40.49 | fugitivo | i don't know if that's what you need |
| 15:41.18 | fugitivo | problem is that you'll not have control over that call |
| 15:41.29 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 15:41.37 | freat | any of you all using the ackcall for agents? |
| 15:41.50 | freat | I would like to play a message _before_ the # key... |
| 15:42.05 | saftsack | digit timeout would help |
| 15:43.48 | *** join/#asterisk axscode (n=asdfasd@203.213.217.123) |
| 15:43.59 | TUplink | any one know what /libexec/ld-elf.so.1: /usr/lib/asterisk/modules/res_adsi.so: Undefined symbol "ast_load" means? |
| 15:44.05 | TUplink | or how to fix it |
| 15:44.09 | axscode | astbill.. |
| 15:44.25 | fugitivo | TUplink: i'd already told you what's the problem |
| 15:44.31 | TUplink | even on a cleaninstall its broke |
| 15:44.37 | Zeeek | TUplink just don't load that module |
| 15:44.39 | TUplink | well how do i fix it? |
| 15:44.42 | TUplink | how? |
| 15:44.47 | saftsack | fugitivo, Mar 12 16:49:10 WARNING[18700]: app_dial.c:1163 dial_exec_full: Invalid timeout specified: 't' |
| 15:45.04 | TUplink | i edited modules.conf and did a noload=res_adsi.so |
| 15:45.12 | fugitivo | saftsack: when did that happen? |
| 15:45.13 | Zeeek | noload => res_adsi.so |
| 15:45.29 | saftsack | it happened at exten => 0,1,Dial(Zap/g2,t) |
| 15:45.29 | fugitivo | saftsack: create a t default extension |
| 15:45.34 | TUplink | even with that it dont work |
| 15:46.08 | saftsack | and what to do in the default one? |
| 15:46.09 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 15:46.31 | fugitivo | saftsack: what should do if it reaches timeout |
| 15:46.40 | Zeeek | saftsack the t is not what Dial wants there, it wants a number of seconds |
| 15:46.40 | TUplink | Zeeek it had no efect |
| 15:46.51 | saftsack | ok |
| 15:46.54 | *** join/#asterisk [ProB]CrazyMan (n=Tobias@p549F25E1.dip0.t-ipconnect.de) |
| 15:46.56 | Zeeek | Dial(ZAP/g2,45) |
| 15:47.00 | saftsack | so one second would be gread or? |
| 15:47.09 | saftsack | or is that the timeout for ringing? |
| 15:47.23 | Zeeek | At this point you maybe want to read a doc or two |
| 15:47.26 | fugitivo | timeout for ringing, but you don't need that |
| 15:47.37 | Zeeek | show application dial for example |
| 15:47.38 | fugitivo | because 0 will give you instant pstn dialtone |
| 15:47.43 | Zeeek | or http://asteriskdocs.org |
| 15:48.03 | TUplink | ok.. i renamed /usr/lib/asterisk/modules/res_adsi.so to /usr/lib/asterisk/modules/res_adsi.so.temp and tryed again then i get /libexec/ld-elf.so.1: /usr/lib/asterisk/modules/res_features.so: Undefined symbol "ast_load" |
| 15:48.05 | saftsack | Zeeek, i know it thanks but i dont need a manual atm because i have another problem |
| 15:48.12 | TUplink | so its somthing higherup |
| 15:48.32 | Zeeek | saftsack you don't appeat to understand what dial dores so I'd recommend reading a little |
| 15:48.53 | Zeeek | but you can keep asking here if that's your thing |
| 15:49.13 | saftsack | i dont know which thing in dial you mean |
| 15:49.19 | *** join/#asterisk rene- (n=rene-@201.137.84.151) |
| 15:49.26 | Zeeek | that's why I recommend you read |
| 15:49.35 | *** join/#asterisk coppice (n=chatzill@89.202.17.210.dyn.pacific.net.hk) |
| 15:49.41 | Zeeek | learn how Dial works |
| 15:49.48 | Zeeek | it's all there |
| 15:50.12 | Zeeek | show application dial, read the wiki on it, read docs, read something |
| 15:50.12 | saftsack | i know how it works but i have a question and the question is. howto say dial that it should wait for digits with a timeout of 1? |
| 15:50.21 | saftsack | i am on the wiki |
| 15:50.33 | Zeeek | you know how it works but you put Dial(ZAP/g2,t) ? |
| 15:50.45 | Zeeek | sorry, you do NOT know |
| 15:50.52 | Phoenix[Y-LAN] | hi |
| 15:50.57 | Zeeek | you can easily find out by reading though |
| 15:51.03 | saftsack | i put it without thinking and i know that there must be a THIRD option |
| 15:51.11 | Phoenix[Y-LAN] | has anybody the sip.conf in a mysql database? |
| 15:51.14 | Zeeek | stop guessing and read |
| 15:51.33 | saftsack | i guess nothing |
| 15:51.42 | Zeeek | going to ignore mode |
| 15:51.59 | fugitivo | saftsack: did you try what i said? it should work |
| 15:52.01 | Flauto | Zeeek, would you have any idea about the vbuzzer problem? |
| 15:52.27 | rene- | how does one have a softphone (e.g. SJPhone) connect to an asterisk box seamlessly inside the office and outside of it, i have been told to put asterisk on the lan external IP, but what about NAT, i would need to enable it to use it when i am outside of the office, will the phone work reliably on the LAN with NAT enabled in asterisk? |
| 15:52.27 | Zeeek | Sorry, no, but if it worked before there is obviously some chnage in the new version |
| 15:52.43 | saftsack | fugitivo, you mean the t default extension? |
| 15:52.46 | Phoenix[Y-LAN] | and my second problem is how to use my isdn fritzcard |
| 15:52.51 | fugitivo | saftsack: the dial string |
| 15:53.01 | Zeeek | rene - you could use an IAX client to avoid NAT problems |
| 15:53.10 | saftsack | t: Allow the called user to transfer the call by hitting # |
| 15:53.14 | saftsack | or what do you mean? |
| 15:53.22 | fugitivo | the dialstring |
| 15:53.40 | saftsack | Dial(Zap/g2,t) do you mean this dialstring? |
| 15:53.47 | fugitivo | exten => 0,1,Dial(Zap/g2,whateverparamyouwant,too) |
| 15:53.50 | fugitivo | yes |
| 15:54.14 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 15:54.18 | saftsack | so you mean a dialstring without a number to dial? |
| 15:54.26 | rene- | Zeeek, IAX will know that i am inside a NAT by itself so no configuration here right? |
| 15:54.35 | fugitivo | saftsack: that'll give you pstn dialtone |
| 15:54.37 | Zeeek | no, IAX doesn't care about NAT at all |
| 15:54.45 | Zeeek | it uses the same port for everything |
| 15:54.49 | fugitivo | saftsack: but you won't have control over that call (cdr, etc) |
| 15:55.14 | saftsack | i want to have control over the call with my telephone |
| 15:55.22 | saftsack | i want to add a digit whenever i want |
| 15:55.38 | core-ix | you have to configure your NAT box to forward UDP port 4569 to the appropriate host |
| 15:55.40 | fugitivo | maybe i didn't understand what you want to do |
| 15:55.48 | Zeeek | core-ix not really |
| 15:55.53 | saftsack | just calling another party ^^ |
| 15:56.05 | rene- | it is an option, but right now i have several softphones deployed, and they have to be configured manually and people always forget to switch profiles |
| 15:56.12 | Zeeek | IAX2 doesn't need prot forwarding as long as it registeres and keps the port open |
| 15:56.20 | fugitivo | saftsack: if you want to pick up your phone, have asterisk dialtone, press 0, get pstn dialtone, and then dial a phone number, that'll work |
| 15:56.22 | x86 | whats a prot? |
| 15:56.22 | core-ix | Zeeek at least this is what i read in Spencer's doc on IAX2 |
| 15:56.34 | Zeeek | you can try using qualify with SIP to keep the NAT open |
| 15:56.37 | fugitivo | saftsack: if you want to do another thing, i don't really know what you want to do |
| 15:56.44 | saftsack | fugitivo, ok :) |
| 15:56.48 | rene- | core-ix sounds right, if you are putting an IAX server |
| 15:56.56 | Zeeek | core-ix you need to forward port only if you want to provision |
| 15:57.00 | saftsack | and do you mean t or r? |
| 15:57.11 | fugitivo | saftsack: just try it and tell me if that's what you want |
| 15:57.15 | Zeeek | yes, on the server end you do, sorry |
| 15:57.20 | core-ix | Zeeek you're right |
| 15:57.23 | Zeeek | I was referring to the client end |
| 15:57.35 | Zeeek | I have both client AND server behind NAT |
| 15:57.46 | Zeeek | so I do use port forwarding on both ends |
| 15:57.48 | rene- | how does that work Zeeek? |
| 15:57.49 | rene- | ahh |
| 15:57.51 | rene- | damm |
| 15:57.53 | rene- | damn |
| 15:58.07 | rene- | try that with sip |
| 15:58.09 | Zeeek | but my son installed a SIP phone and it worked without forwarding! |
| 15:58.33 | saftsack | fugitivo, do you have my pastebin entry? because i dont know where to add |
| 15:58.39 | Zeeek | Maybe because I use qualify on his account? |
| 15:58.40 | rene- | so weird |
| 15:58.56 | rene- | i have no idea |
| 15:59.19 | rene- | but i would think that sip would be next to impossible to make it work with double nat |
| 15:59.22 | fugitivo | saftsack: add that in your outgoing context |
| 15:59.30 | rene- | unless you were a killer network ninja |
| 15:59.45 | Curus | Double NAT shouldn't be any worse han single NAT... |
| 15:59.47 | TUplink | sip is next to imposable with 2xNAT |
| 15:59.47 | Zeeek | rene- who? |
| 16:00.01 | Zeeek | TUplink bzzzzt wrong. I have used it for 2 years |
| 16:00.06 | TUplink | i tryed for 2 weeks to get it to work |
| 16:00.07 | Zeeek | no problems at all |
| 16:00.17 | Zeeek | it may depend on the routers involved? |
| 16:00.27 | Curus | As well as the devices |
| 16:00.29 | Zeeek | on the asterisk side you MUST use prot forwarding |
| 16:00.36 | TUplink | well Zeeek when i get me server back up i want you to help me set it up |
| 16:00.37 | Zeeek | or even port forwarding |
| 16:01.00 | Zeeek | forward 5060 and 10000-11000 to the * box ip |
| 16:01.13 | Zeeek | use nat=yes, externip= in spi.conf |
| 16:01.17 | TUplink | how bout just put it in the MZ |
| 16:01.19 | TUplink | DMZ* |
| 16:01.29 | Zeeek | I've never gotten stuff to work right in the DMZ |
| 16:01.40 | Zeeek | well, yes, a webcam, but no voip |
| 16:01.55 | core-ix | OT: any channel in FreeNODE for SER (SIP Express Router) ??? |
| 16:01.57 | TUplink | DMZ just forwards every thing to that host |
| 16:02.17 | Zeeek | look at this: http://willypick.mindsay.com/ |
| 16:02.37 | rene- | Zeeek, what does asterisk uses externip for? |
| 16:02.38 | saftsack | fugitivo, no it isnt that what i want :( sry |
| 16:02.42 | Zeeek | http://willypick.mindsay.com/?skip=10 |
| 16:02.44 | saftsack | i want to dial out with my phone |
| 16:02.55 | Zeeek | rene- it adds the correct ip to the SIP header |
| 16:03.00 | *** join/#asterisk zotz (n=zotz@24.231.32.85) |
| 16:03.03 | fugitivo | saftsack: i don't understand what you need then |
| 16:03.13 | rene- | i see |
| 16:03.20 | rene- | so client will know what to connect bak |
| 16:03.22 | rene- | back |
| 16:03.27 | saftsack | i want to hear the asterisk dialtone. and then dial out. not more |
| 16:03.37 | Zeeek | other wise astrisk calls out with 192.168.1.10 or whatever - makes it hard to find :) |
| 16:03.50 | TUplink | but what bout DHCP |
| 16:03.53 | fugitivo | saftsack: then why don't you use a normal dial string for that? |
| 16:04.10 | saftsack | i have a normal one but it doesnt work |
| 16:04.12 | Zeeek | http://www.voip-info.org/wiki/view/Asterisk+SIP+externip |
| 16:04.22 | saftsack | <PROTECTED> |
| 16:04.29 | saftsack | this is a normal one i thinks |
| 16:04.30 | saftsack | -s |
| 16:04.47 | fugitivo | saftsack: _0.,1,Dial(Zap/g2/${EXTEN:1,30,t) |
| 16:04.54 | fugitivo | saftsack: _0.,1,Dial(Zap/g2/${EXTEN:1},30,t) |
| 16:05.13 | fugitivo | saftsack: you'll need to dial 0 and the phone number |
| 16:05.20 | saftsack | this doesnt work :( because it doesnt leave the waitexten application then |
| 16:05.20 | TUplink | errr i hate compilingf |
| 16:05.33 | ambriento | saftsack, are u using an analog phone? or a FXO device? |
| 16:05.39 | saftsack | isdn telephone |
| 16:05.46 | fugitivo | saftsack: what do you mean? |
| 16:05.47 | Phoenix[Y-LAN] | has anyone asterisk + mysql |
| 16:05.52 | saftsack | an bri telephone |
| 16:05.56 | ambriento | got it |
| 16:06.03 | saftsack | but it is handled as an analog device |
| 16:07.09 | fugitivo | saftsack: maybe if you explain correctly what you need to do, i can help you |
| 16:07.16 | saftsack | ok |
| 16:07.40 | Zeeek | fugitivo I gave up long ago |
| 16:08.00 | saftsack | if i take the handset i want to hear an congestion dialtone and if i press a 0 i want to be able to dial to an external party |
| 16:08.05 | fugitivo | Zeeek: i'm starting to fell the same |
| 16:08.25 | Zeeek | people come here and refuse to read a few lines that have the answers |
| 16:08.39 | Zeeek | I'm not gonna spend that day discussing shit that's well documented |
| 16:08.40 | ambriento | saftsack, do u have ignorepat => 0 inside your context in extensions.conf? |
| 16:08.46 | saftsack | yes |
| 16:09.31 | *** join/#asterisk Eggplant (i=No@dsl-827.cascadeaccess.com) |
| 16:09.43 | fugitivo | saftsack: _X is wrong |
| 16:09.50 | saftsack | yes ;) |
| 16:09.50 | ambriento | saftsack, when you lift the hook, what happens now? no tones of any kind? |
| 16:10.07 | x86 | Phoenix[Y-LAN]: i'm doing CDR with mysql |
| 16:10.14 | saftsack | ambriento, hook = handset? |
| 16:10.28 | ambriento | I'm not used to bri, but as far as I know, its digital, so no dial needed |
| 16:10.32 | ambriento | yes saftsack :) |
| 16:10.42 | ambriento | dial tone* |
| 16:10.47 | saftsack | there is a pseudo dialtone generated from asterisk |
| 16:10.49 | Phoenix[Y-LAN] | cdr is only logging? |
| 16:11.00 | saftsack | generated by the line Playtones(Congestion) |
| 16:11.38 | saftsack | fugitivo, i cant do _X. because of the goto application |
| 16:11.40 | ambriento | what asterisk says when you lift the handset? |
| 16:11.58 | fugitivo | saftsack: why are you doing it so complicated? you only need one line for dialing out |
| 16:11.58 | Phoenix[Y-LAN] | i want to have the users in a mysql database so you can add and remove users by a website |
| 16:12.06 | saftsack | <PROTECTED> |
| 16:12.10 | saftsack | fugitivo, yes i know |
| 16:12.26 | saftsack | but i need it after pressing the 0 |
| 16:12.29 | saftsack | and NOT before |
| 16:12.34 | saftsack | thats complicated :( |
| 16:12.45 | ambriento | indeed |
| 16:13.17 | ambriento | how asterisk sees it? as a Zap Channel? |
| 16:14.01 | Zeeek | rene- I am looking at something that says X-Lite send stayalive packets - thazt should work behind NAT |
| 16:14.16 | saftsack | ambriento, yes as a zap channel |
| 16:14.35 | ambriento | zeeek, its on sip.conf sample |
| 16:14.48 | DaveHope | Zeeek: Just want to say a big thanks, with some minor tweaks to the last pastebin entry you made I've been able to get things working. |
| 16:14.48 | Zeeek | yeah |
| 16:14.52 | saftsack | fugitivo, howto say goto that there is a _X. in the other context? |
| 16:14.57 | fugitivo | saftsack: it's not before |
| 16:14.58 | Zeeek | DaveHope hey that's great! |
| 16:15.04 | saftsack | fugitivo, ? |
| 16:15.17 | *** join/#asterisk Kernel_Core (i=Kernel_C@217.218.80.165) |
| 16:15.19 | Kernel_Core | hi all |
| 16:15.22 | Kernel_Core | helo me ! |
| 16:15.31 | Kernel_Core | rtp.c:911 ast_rtcp_new: Unable to allocate socket: Too many open files |
| 16:15.36 | fugitivo | saftsack: if you press 0 and then the number, and your dialstring is correct, it'll work |
| 16:15.38 | Zeeek | rm *.core |
| 16:15.41 | Kernel_Core | Mar 12 09:57:37 WARNING[16340]: res_agi.c:240 launch_script: unable to create fromast pipe: Too many open files |
| 16:15.51 | Kernel_Core | Mar 12 09:57:39 ERROR[14138]: ast_h323.cxx:169 void PAssertFunc(const char*): Assertion fail: Function ::close failed, file tlibthrd.cxx, line 761, Error=9 |
| 16:15.59 | Kernel_Core | after that asterisk crashes !!!!!!!!! |
| 16:16.00 | fugitivo | saftsack: you just need something like exten => _0.,1,Dial(...) |
| 16:16.08 | saftsack | fugitivo, yes thats true |
| 16:16.12 | saftsack | but if i have something like this |
| 16:16.17 | saftsack | it doesnt leaves waitexten |
| 16:16.35 | fugitivo | why you want that? |
| 16:16.39 | saftsack | <PROTECTED> |
| 16:16.39 | saftsack | <PROTECTED> |
| 16:16.54 | saftsack | fugitivo, i want to play another dialtone before a digit is pressed |
| 16:17.19 | saftsack | if there isnt waitexten there will be autofallthrough |
| 16:18.10 | wunderkin | Kernel_Core: google |
| 16:21.32 | Kernel_Core | wunderkin: got nothing useful ! |
| 16:22.03 | saftsack | fugitivo, is it possible to enable overlapdial within the dialplan? |
| 16:23.21 | wunderkin | google "too many open files" that is a common problem and not specific to asterisk |
| 16:26.35 | *** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfmt2.dialup.mindspring.com) |
| 16:26.49 | *** part/#asterisk mhnoyes (n=mhnoyes@user-2ivfmt2.dialup.mindspring.com) |
| 16:27.33 | *** join/#asterisk Nukemizer (n=Nuke@160.7.249.15) |
| 16:28.26 | *** part/#asterisk axscode (n=asdfasd@203.213.217.123) |
| 16:28.43 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 16:30.34 | willt | Kernel Core: what os are you usiNG? |
| 16:31.10 | Abydos313 | win 3.1 |
| 16:31.19 | Phoenix[Y-LAN] | hehe |
| 16:31.20 | willt | nice! |
| 16:31.24 | Zeeek | otherwise know as "Stable" |
| 16:31.29 | Abydos313 | haha |
| 16:31.31 | RoyK | Abydos313: that is not an OS |
| 16:31.34 | willt | exactly! |
| 16:31.42 | Abydos313 | it was a sad attempt |
| 16:31.50 | Zeeek | Stay with WIn 2.0 |
| 16:32.08 | willt | I prefer win 3.11 :) |
| 16:32.21 | Zeeek | nah, all that networking stuff is bloat |
| 16:32.27 | Abydos313 | damn it's been awhile since i seen good ol' win 3.11 |
| 16:32.27 | *** join/#asterisk Nukemizer (n=Nuke@160.7.249.15) |
| 16:35.21 | *** join/#asterisk coppice (n=chatzill@89.202.17.210.dyn.pacific.net.hk) |
| 16:35.52 | *** join/#asterisk _MartinCabrera_ (n=_MartinC@litigaractivos1.att.net.co) |
| 16:36.00 | Kernel_Core | wunderkin: do you have solution for this error ? |
| 16:36.04 | fugitivo | it was the last windows i used |
| 16:36.36 | Abydos313 | i make my money off widows users.. no choice |
| 16:37.10 | willt | Kernel_Core: what os are you using? |
| 16:37.36 | saftsack | where to change the standard zaptel tone if i raise the handset? |
| 16:37.47 | fugitivo | Abydos313: that sucks |
| 16:38.25 | _MartinCabrera_ | Hi, Anyone knows if there is anny issue installing both TE210 and TE110P digium cards in the same box? |
| 16:38.57 | *** part/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 16:39.05 | Abydos313 | i still run linux in vmware at work... how else can i do my scans |
| 16:40.31 | Kernel_Core | willt: Linux amirrazavi 2.4.27-2-386 #1 Wed Nov 30 21:38:51 JST 2005 i686 GNU/Linux ( Debian ) |
| 16:40.59 | Abydos313 | why the older kernel? |
| 16:41.56 | *** join/#asterisk project_2501 (n=project-@S01060004e2929dc9.br.shawcable.net) |
| 16:43.04 | Kernel_Core | Abydos313: I have to recompile it .... I don't have local access to this machine |
| 16:43.16 | Kernel_Core | Abydos313: do you have any solution ? |
| 16:43.55 | Abydos313 | kewl... i personally don't see any diff between 2.4 and 2.6 for the things i do except zaptel timing.. :) |
| 16:45.49 | Abydos313 | just out of curiousity i installed asterisk@home and it installs ztdummy and 2.6 kernel. i read you don't need ztdummy with 2.6 |
| 16:49.03 | *** join/#asterisk mexuar-tim (n=mexuar-t@host-212-158-206-61.bulldogdsl.com) |
| 16:52.08 | freat | Abydos313: you DO need ztdummy w/ 2.6. It is just a completely different version |
| 16:52.28 | freat | Abydos313: you need ztdummy if you don't have a zaptel card (for timing device) |
| 16:52.53 | freat | Abydos313: in 2.6, the kernel has new features for timer resolution. 2.4 stuff more more of a hack, using USB device for timing |
| 16:53.03 | willt | Kernel_Core: what user is asterisk running as? |
| 16:56.15 | Kernel_Core | willt: ROOT |
| 16:56.51 | Phoenix[Y-LAN] | is there anywhere a howto for pci ftitzcard A ? |
| 16:57.19 | *** join/#asterisk SWAT (n=SWAT@dsl159-68-100.fastxdsl.nl) |
| 16:57.30 | Phoenix[Y-LAN] | i get an error in capiinfo when i install isdn with yast |
| 16:57.51 | Phoenix[Y-LAN] | capi not installed - No such device or address (6) |
| 16:58.48 | SWAT | Is Asterisk capable/designed to be able to accept LAN-only software SIP clients with optional conferencing? |
| 17:00.08 | *** part/#asterisk jluk (n=jon@80-235-135-92.cable.ubr07.nail.blueyonder.co.uk) |
| 17:04.22 | *** join/#asterisk bkw_ (n=bkw_@adsl-70-142-62-199.dsl.tul2ok.sbcglobal.net) |
| 17:04.36 | Abydos313 | thx i was curious about that |
| 17:05.51 | *** join/#asterisk Tamarisk (n=adrian@user-5659.l6.c4.dsl.pol.co.uk) |
| 17:07.48 | Tamarisk | Some thimes the solution can be so simple it hurts to think of all the time spent finding it |
| 17:07.53 | Tamarisk | Times |
| 17:08.51 | Tamarisk | If Justinu and RoyK are on channel and listening thanks for your previous time and effort |
| 17:12.58 | project_2501 | when I start asterisk with the /etc/init.d/asterisk script on my gentoo box, it behaves differently than when I run asterisk -vvvvc |
| 17:13.13 | project_2501 | does anyone else have that problem? |
| 17:14.52 | project_2501 | my voicemail is totally botched when I start asterisk at boot with /etc/init.d/asterisk, but works fine when I run asterisk -vvvvc |
| 17:15.11 | RoyK | me? |
| 17:15.25 | *** part/#asterisk SWAT (n=SWAT@dsl159-68-100.fastxdsl.nl) |
| 17:16.28 | Abydos313 | project_2501 you're leaving out the -r |
| 17:16.56 | project_2501 | which command requires a '-r' ? |
| 17:17.06 | Abydos313 | astersik -rvvvvvc |
| 17:17.37 | Abydos313 | atleast that's how i view a verbose console |
| 17:18.29 | project_2501 | yeah, I guess I could do that so I don't have to keep starting and sopping asterisk, and then I can just alter it on the fly |
| 17:19.56 | austinnichols101 | has anyone seen the 'goldenGate C2V' before? http://www.qiiq.com/products/productsGoldenGateCellToVoIP.htm |
| 17:20.17 | austinnichols101 | wasn't listed on voip-info.org and not sure it actually exists in the wild |
| 17:20.36 | austinnichols101 | supposed to let you gateway sip to gsm or cdma |
| 17:20.56 | Tamarisk | RoyK |
| 17:21.36 | project_2501 | Abydos313: thanks, using that -r switch allowed me to see what was causing the problem |
| 17:21.40 | RoyK | Tamarisk |
| 17:21.52 | Tamarisk | Hi RoyK |
| 17:22.03 | Tamarisk | Sorry just chatting to the wife |
| 17:22.31 | Tamarisk | Chinese for Tea |
| 17:22.44 | project_2501 | Abydos313: it looks like there is a permission problem when app_voicemail goes to create a new directory for the message |
| 17:22.44 | RoyK | wot? |
| 17:22.45 | *** join/#asterisk horvath (i=horvath@efnut.com) |
| 17:22.49 | *** join/#asterisk iq (n=iq@71-214-5-20.omah.qwest.net) |
| 17:23.08 | Tamarisk | The issue I had for thew past few weeks? with no audio between phones is now solved |
| 17:23.24 | project_2501 | Abydos313: the reason I would't get that when just running asterisk -vvvvc is because I ran that command as root |
| 17:23.36 | Tamarisk | The most simple of things and not * it was the firewall on the * box |
| 17:23.42 | iq | Hi All |
| 17:24.27 | Tamarisk | The network card was set as an externic NIC |
| 17:24.47 | Tamarisk | External NIC |
| 17:24.56 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 17:24.59 | PakiPenguin | evening |
| 17:25.27 | project_2501 | Abydos313: do you have any idea what I need to change permission's on so that I don't get this conflict? |
| 17:25.33 | Tamarisk | <PROTECTED> |
| 17:25.37 | Tamarisk | TA |
| 17:27.48 | project_2501 | does anybody have any idea what I need to change permission's on so that I don't get this conflict? |
| 17:29.02 | RoyK | ~seen zoa |
| 17:29.06 | jbot | zoa <n=kkk@pirus.securax.be> was last seen on IRC in channel #asterisk, 1d 53m 37s ago, saying: ':)'. |
| 17:30.03 | saftsack | hi my telephone can do direct redirection with using the moh function but it doesnt works |
| 17:30.18 | saftsack | asterisk destroys the parked channel if i hang up the host telephone |
| 17:30.48 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 17:34.29 | *** join/#asterisk Vyeperman (n=Vye@ip68-6-130-118.sd.sd.cox.net) |
| 17:36.35 | *** join/#asterisk ioerror (n=ioerror@mail.lostinthenoise.net) |
| 17:48.00 | project_2501 | haha, I got my problem fixed, I had to change the user setting in /etc/conf.d/asterisk |
| 17:50.00 | *** part/#asterisk antoniojulio (n=antonioj@unaffiliated/antoniojulio) |
| 17:51.26 | Abydos313 | ahh |
| 17:54.49 | *** join/#asterisk sevard (i=sev@24-179-181-160.dhcp.dlth.mn.charter.com) |
| 17:56.28 | project_2501 | Abydos313: it's amazing what someone can do when they RTFM !!! |
| 17:58.22 | *** join/#asterisk ToTo (n=ToTo@host10-138.pool872.interbusiness.it) |
| 18:00.13 | Abydos313 | i know what you mean |
| 18:00.43 | Abydos313 | i'm on my second pass thru the asterisk book. i understand most of it now and soon will be working on my dialplan |
| 18:01.20 | RoyK | good boys |
| 18:01.40 | Abydos313 | :) |
| 18:02.55 | tsume | its easy to learn just by looking at it, but I'm one of those people runnning FreeBSD-HEAD and NetBSD-HEAD on his home Desktop machines |
| 18:03.20 | Abydos313 | quit bragging..hee |
| 18:03.21 | Abydos313 | heh |
| 18:03.22 | tsume | BSD is so ready for the desktop :P |
| 18:03.39 | Abydos313 | anyone running asterisk on freebsd? |
| 18:03.44 | RoyK | lots |
| 18:03.52 | tsume | Abydos313: yes,but not with zaptel cards |
| 18:04.02 | Abydos313 | do you have to run linux binary compat? |
| 18:04.07 | tsume | what? no |
| 18:04.13 | tsume | I build asterisk from source |
| 18:04.21 | *** join/#asterisk razu_ (n=razu@80-235-91-173-dsl.prn.estpak.ee) |
| 18:04.34 | tsume | I build everything from source :) except I don't use -O99 like all the genpoo'ers |
| 18:04.49 | tsume | breaking compat layers and bloating binary footprints, yuck! |
| 18:04.49 | russellb | hey, that doesn't do anything :-p |
| 18:04.50 | Abydos313 | haha genpoo'ers |
| 18:05.24 | tsume | also optimizing vms with anything above 2 is a bad idea. Usually breaks the GC |
| 18:06.28 | tsume | Abydos313: Genpoo is how Sylvester would pronounce it ;) |
| 18:06.37 | Abydos313 | i like it |
| 18:06.52 | Abydos313 | just downloaded 2006.0 two days ago |
| 18:07.09 | tsume | Abydos313: all my asterisk boxes are debian |
| 18:07.27 | *** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin) |
| 18:07.27 | Abydos313 | debian is my fav.. but ihaven't tried * on it |
| 18:07.45 | tsume | its simple, I download the asterisk stuff, I have a script on my servers to where I can update, download source packages, build zsaptel drivers, etc. |
| 18:08.08 | tsume | then its al ready.. with gentoo.. you'll have to wait a couple days for everythign to build :) |
| 18:08.11 | Abydos313 | so you're runnign it all from svn or cvs |
| 18:08.29 | tsume | Abydos313: a stable release though :) |
| 18:08.32 | Lino` | gentoo is something for people who love pain |
| 18:08.58 | Abydos313 | i had good success with cvs..but svn gives me issues when i try to upgrade and recompile |
| 18:09.01 | DaveHope | Lino`: Hey! - I don't like pain. |
| 18:09.02 | tsume | Lino`: they should just all use FreeBSD :D they are blind users who need a path to enlightenment. |
| 18:10.16 | tsume | Lino`: gentoo users like to waste time. Gentoo could never be used in real world applications. Saving tim eis essential. Hence Debian. |
| 18:10.43 | DaveHope | tsume: I use Gentoo and debian on my machines :P |
| 18:10.54 | Abydos313 | centos and debian here |
| 18:11.06 | tsume | centos is a RHEL rip with very late security patches :) |
| 18:11.09 | DaveHope | tsume: Gentoo for my desktop, debian for servers. |
| 18:11.20 | Abydos313 | yeah but we are cheap and use it to run oracle at work ;) |
| 18:11.32 | tsume | Abydos313: what is wrong with debian? |
| 18:11.42 | tsume | Abydos313: use alien and convert the rpm to deb |
| 18:11.43 | Abydos313 | nothing but oracle isn't supported on it |
| 18:12.07 | Abydos313 | really? nice |
| 18:12.12 | tsume | I've even ran oracle on my netbsd box |
| 18:12.16 | tsume | with Linux compat, works fine. |
| 18:12.58 | Abydos313 | i have to say centos has been rock solid.. but so has running oracle on windows.. i know that doesn't sound right, but it's true. |
| 18:13.09 | tsume | hehe.. |
| 18:13.21 | Abydos313 | only time we reboot is for windows updates |
| 18:13.52 | Abydos313 | our webserver is MS also.. eek! |
| 18:14.02 | Abydos313 | asp/.net site |
| 18:14.03 | tsume | its not MS software |
| 18:14.06 | tsume | its just asp.net.. :) |
| 18:14.23 | tsume | asp.net isn't bad at all though. Its better than PHP for sure. |
| 18:14.24 | giesen | I dont suppose anyones got a 7970 with the SIP 8.0(2) firmware to work with asterisk yet? (I'd even settle for an example SEP<mac>.cnf.xml file setup for CCM5 |
| 18:14.24 | Abydos313 | i know your's isn't, i was referring to ours |
| 18:18.55 | Lino` | ... |
| 18:19.25 | *** join/#asterisk justinu (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 18:27.53 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 18:34.27 | *** join/#asterisk mogorman (n=mogorman@68.62.237.103) |
| 18:38.41 | *** join/#asterisk newmember[laptop (n=newmembe@S010600036d1139bb.cg.shawcable.net) |
| 18:39.49 | *** join/#asterisk InfraRed (n=subhi@iwastedmoneyonadomainnamejustforadifferenthostmaskonirc.co.uk) |
| 18:46.05 | kippi | what ports do I need to allow on my firewall to let my SIP phoned connect to my asterisk box |
| 18:48.52 | *** join/#asterisk riddlebox (n=james@24-171-11-166.dhcp.stls.mo.charter.com) |
| 18:49.54 | kippi | would it be ports 5060/5004/5062/5006 ? |
| 18:50.16 | InfraRed | 5060 udp is standard sip port |
| 18:50.27 | kippi | and thats all I need to be open? |
| 18:50.28 | InfraRed | but since it's udp you have to open a wide range of ports |
| 18:50.47 | kippi | is there somewhere that tells me what ports i need? |
| 18:51.03 | InfraRed | tried voip-info.org? |
| 18:52.02 | kippi | yep |
| 18:52.05 | kippi | can't see it |
| 18:52.35 | InfraRed | http://www.google.co.uk/search?num=100&hl=en&q=asterisk+firewall+ports+configuration&btnG=Search&meta= |
| 18:52.42 | InfraRed | first hit on voip-info |
| 18:52.43 | InfraRed | :P |
| 18:55.36 | kippi | cool. thanks :) |
| 18:55.42 | InfraRed | np |
| 18:55.51 | InfraRed | I am your google bitch for today |
| 18:55.56 | *** join/#asterisk xevo (n=bob@c-67-182-205-227.hsd1.ut.comcast.net) |
| 18:56.11 | tsume | LAMP < *. ASP.NET(C#) using mono and Ruby > *. ;) |
| 18:56.38 | InfraRed | heh |
| 18:57.26 | kippi | hope this works |
| 18:59.37 | willt | Is there any comparisions on linux and freebsd runing asterisk? |
| 19:00.52 | InfraRed | depends if you're using hardware or voip only setup |
| 19:00.56 | InfraRed | zapata is better on linux |
| 19:01.52 | willt | voip only no hardware |
| 19:02.14 | InfraRed | I use fbsd on that |
| 19:02.16 | InfraRed | it's fine |
| 19:02.21 | InfraRed | no issues |
| 19:02.42 | tsume | willt: freebsd works great with asterisk |
| 19:02.59 | tsume | willt: using 6.x, you have a better firewall than netfilter of course ;) |
| 19:03.05 | tsume | Wonderful pf |
| 19:03.10 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 19:03.25 | willt | yeah im justing wondering about call volumes if one would handle it better or not |
| 19:03.38 | tsume | freebsd? sure it will. |
| 19:03.42 | tsume | the power to serve! :) |
| 19:04.28 | willt | all my servers are freeBSD except my chilisoft server |
| 19:04.35 | willt | and a few suns :) |
| 19:06.51 | astra^^ | how much b/w does a g711 call take in asterisk..? |
| 19:07.01 | InfraRed | 85Kbps |
| 19:07.15 | astra^^ | g729..? |
| 19:07.24 | astra^^ | g723? |
| 19:07.43 | InfraRed | use the bandwidth calculator |
| 19:08.52 | kippi | is anyone using the lastest version of firmware on the GXP-200? |
| 19:11.27 | InfraRed | i only use 5 channels on the fbsd box |
| 19:11.36 | InfraRed | ATA->fbsd->out |
| 19:11.46 | InfraRed | i run out of bandwith before i run out of cpu/channels |
| 19:11.47 | InfraRed | :) |
| 19:15.39 | willt | what kind of connection do you have? |
| 19:15.43 | *** join/#asterisk bkw__ (n=bkw_@adsl-70-142-62-199.dsl.tul2ok.sbcglobal.net) |
| 19:15.54 | InfraRed | adsl |
| 19:16.03 | *** join/#asterisk RoyK (n=roy@a217-118-45-74.bluecom.no) |
| 19:16.22 | willt | get a bigger pipe :) |
| 19:16.30 | InfraRed | 5 channels is all i need |
| 19:16.44 | InfraRed | maximum peak was 4 channels |
| 19:16.46 | PakiPenguin_ | hi RoyK |
| 19:16.47 | InfraRed | so it's enough |
| 19:16.59 | InfraRed | i'm on ADSL2 here 12Mbps/768 |
| 19:17.12 | willt | Where is this bandwidth calculater? |
| 19:17.12 | InfraRed | it's the uplink that's the limiting factor |
| 19:17.17 | RoyK | PakiPenguin_: hi |
| 19:17.20 | InfraRed | willt: use teh google |
| 19:17.28 | willt | wow 12mbs is that verizon? |
| 19:17.34 | RoyK | PakiPenguin_: hm. it should be pretty late down there now, shouldn't it? |
| 19:17.34 | InfraRed | ukonline |
| 19:17.40 | InfraRed | i'm in the uk |
| 19:17.44 | willt | cool |
| 19:17.51 | InfraRed | it's 22Mbps service |
| 19:17.59 | PakiPenguin_ | RoyK, 12:17am |
| 19:18.00 | InfraRed | but it syncs at 12 |
| 19:18.09 | willt | thats crazy! |
| 19:18.10 | InfraRed | distance from exchange, line quality etc |
| 19:18.37 | willt | wish I could get that here in the US |
| 19:19.42 | willt | I guess I should upgrade my wireless equipment :) |
| 19:19.58 | InfraRed | heh |
| 19:20.02 | willt | on thing nice is I have about 3Mbs upstream |
| 19:20.08 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 19:20.29 | PakiPenguin_ | 3Mbps upstream to the internet :o |
| 19:20.29 | tsume | 800USD Dlink outide APs are nice :) |
| 19:20.32 | PakiPenguin_ | hey shido6 |
| 19:20.40 | PakiPenguin_ | oh tsume |
| 19:20.48 | tsume | they pump out 5.4 and 2.4 signals |
| 19:20.54 | PakiPenguin_ | i did a 4+KM link with D-Link indoor APs |
| 19:21.05 | PakiPenguin_ | and 18db antennas |
| 19:21.08 | PakiPenguin_ | its still up and working fine atm :) |
| 19:21.23 | tsume | well.. I don't know what the signal is on these antennas, but they are huge :) |
| 19:21.30 | *** join/#asterisk heison (n=heison@CPE000625e6c664-CM00122570a518.cpe.net.cable.rogers.com) |
| 19:21.34 | InfraRed | radiation! |
| 19:21.35 | InfraRed | :) |
| 19:21.55 | PakiPenguin_ | tsume, you'r talking about DL-AP2700 right? |
| 19:22.23 | Damin | WOWOWOWOWOWOW!!! |
| 19:22.24 | Damin | echo -n "64" > /sys/block/sda/queue/iosched/queued |
| 19:22.24 | Damin | echo -n "32768" > /sys/block/sda/queue/nr_requests |
| 19:22.29 | Damin | THAT IS WHAT I AM TALKING ABOUT!!! |
| 19:22.52 | Damin | Read 18000 MBs | 87.1 s | 206.601 MB/s | 39.0 % | 260.7 % |
| 19:22.57 | Damin | YEAH BABY! :) |
| 19:27.01 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 19:27.42 | InfraRed | sda? |
| 19:27.45 | InfraRed | isnt that scsi |
| 19:27.52 | RoyK | or sata |
| 19:27.53 | PakiPenguin_ | yup |
| 19:27.56 | InfraRed | ah |
| 19:28.06 | InfraRed | sick sick person |
| 19:28.07 | InfraRed | :) |
| 19:28.32 | PakiPenguin_ | :p |
| 19:41.54 | *** join/#asterisk mrempire (n=trefpunt@mrempire.demon.nl) |
| 19:45.37 | *** join/#asterisk freezer (i=freeze@66.29.46.127) |
| 19:45.39 | freezer | hi |
| 19:47.44 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 19:52.04 | *** part/#asterisk project_2501 (n=project-@S01060004e2929dc9.br.shawcable.net) |
| 19:54.17 | *** join/#asterisk rajiv (n=irc@gentoo/developer/rajiv) |
| 20:00.15 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 20:01.43 | *** join/#asterisk tsume (n=tsume@zanshin.tsumelabs.com) |
| 20:02.45 | *** join/#asterisk chr|s_ (n=chris@217.171.53.239) |
| 20:02.45 | *** join/#asterisk Tamarisk (n=adrian@user-5659.l6.c4.dsl.pol.co.uk) |
| 20:05.10 | *** join/#asterisk Rowter (n=Silver@201.145.5.59) |
| 20:07.24 | backblue | Damin: mount it with -o sync and test again, that's fake, and test write, read its fast... writing its the problem. |
| 20:08.51 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-166-193.red.bezeqint.net) |
| 20:09.18 | RoyK | Damin: there is no such drive on the market that can sustain anything about ~110MB/s, and that is outer rim only |
| 20:10.30 | chr|s_ | RoyK, unless you are talking rocket drives |
| 20:10.47 | RoyK | rocket? |
| 20:10.54 | Phoenix[Y-LAN] | has anyone experiences with asterisk + isdn? |
| 20:10.56 | chr|s_ | (large memory arrays interfaced to an SATA controller) |
| 20:11.19 | chr|s_ | RoyK, 20Gb of DDR2 ram that acts like a hard disk |
| 20:11.31 | RoyK | heh |
| 20:11.33 | RoyK | yes |
| 20:11.42 | RoyK | I was talking of rotating disks |
| 20:12.18 | RoyK | normal 7k2 ATA/SATA drives get up to something like 65 MB/s outer rim, about half of that inner rim |
| 20:12.46 | RoyK | s/inner rim/inner area/ |
| 20:12.48 | RoyK | perhaps |
| 20:12.57 | RoyK | thanks, jbot |
| 20:13.11 | chr|s_ | yeah I understood, just dreaming of ultra high performance. I don't keep up with CPU, but I realised CPU frequencies haven't done anything in the last 2 years, and I read an anandtech artlce (and tomshardware) quite insightful on the issue. I want 100 fold of todays performance. Anything that takes 10 seconds for me today, or indeed, 10 minutes, I want to take milliseconds. |
| 20:14.23 | *** join/#asterisk zotz (n=zotz@24.231.32.85) |
| 20:15.31 | chr|s_ | =] |
| 20:15.52 | chr|s_ | suprisingly after the first 37 pages that asterisk book starts getting damn useful!! :-) |
| 20:16.16 | Phoenix[Y-LAN] | whicj?h book do u have? |
| 20:16.31 | chr|s_ | O'Reilly one |
| 20:16.37 | RoyK | it's interesting how traffic volume jumps 30-50% at about 21:00 :P |
| 20:17.35 | Phoenix[Y-LAN] | Asterisk. The Future of Telephony |
| 20:17.38 | Phoenix[Y-LAN] | this one |
| 20:17.54 | chr|s_ | ja, |
| 20:18.36 | Phoenix[Y-LAN] | du sprichst deutsch? |
| 20:19.16 | RoyK | nein! |
| 20:19.30 | chr|s_ | *tries to switch to german mode* |
| 20:19.33 | chr|s_ | genau |
| 20:20.13 | chr|s_ | aber, jetz ich kannst mit ein anderer sprache gedenke |
| 20:20.27 | tsume | ewww german |
| 20:20.29 | tsume | evil |
| 20:20.42 | RoyK | american english is FAR more evil than german |
| 20:20.43 | RoyK | :P |
| 20:20.58 | tsume | RoyK: exactly why I'm learning Japanese ;) |
| 20:21.09 | RoyK | hehe |
| 20:21.21 | RoyK | my gf studied japanese for three years or so |
| 20:21.35 | RoyK | now she is getting a job as a it support consultant |
| 20:21.47 | tsume | its a fun language |
| 20:21.48 | RoyK | seems japanese isn't really relevant in .no |
| 20:22.10 | tsume | move to japan, to the funny side :) |
| 20:22.24 | RoyK | it seems to me, yes, it is. stuff like women's rights or laws against racism does not exist at all |
| 20:22.27 | tsume | move to tokyo and be fried like an egg in the summer at a wonderful human 36+F |
| 20:22.32 | tsume | *36+C |
| 20:22.57 | tsume | s/funny/sunny/ |
| 20:23.14 | tsume | RoyK: don't you like all the butter dog pr0no? :P |
| 20:23.22 | chr|s_ | ;-) lol |
| 20:23.41 | chr|s_ | raining |
| 20:23.44 | RoyK | bloody winter |
| 20:23.46 | tsume | http://www.five-d.co.jp/sbd/ |
| 20:23.50 | *** join/#asterisk Phoenix[1-LAN] (n=phoenix@www.netzwerk-profis.de) |
| 20:23.54 | Phoenix[1-LAN] | ree |
| 20:23.58 | tsume | oh wait, that is the rock band :P |
| 20:24.03 | RoyK | alaska is probably worse than oslo, but this is bloody bad anyway |
| 20:24.12 | RoyK | tsume: http://engrish.com/ |
| 20:24.17 | chr|s_ | I hope it is snowing up in the mountains, I desperately need to ski this weekend |
| 20:24.27 | chr|s_ | I almost took a job in Japan, about 4 years ago |
| 20:24.40 | chr|s_ | went to Germany instead |
| 20:24.54 | RoyK | chr|s_: where're you from? |
| 20:24.55 | chr|s_ | RoyK, I coudl win a gold medal in the snowball olympics if only they had them |
| 20:25.08 | RoyK | heh |
| 20:25.10 | chr|s_ | United Kingdom of Great Brirtain and Northern Island |
| 20:25.31 | chr|s_ | in welsh though ;-) |
| 20:25.51 | RoyK | Prost! |
| 20:26.09 | chr|s_ | Prost!! |
| 20:26.13 | chr|s_ | Skol! |
| 20:26.15 | RoyK | where in .de do you live? |
| 20:26.17 | RoyK | skål! |
| 20:26.22 | chr|s_ | Giamas! |
| 20:26.26 | RoyK | Kippis! |
| 20:26.30 | chr|s_ | I lived in Stuttgart |
| 20:26.33 | RoyK | ah |
| 20:26.34 | RoyK | ok |
| 20:26.36 | chr|s_ | Cheers! |
| 20:26.43 | RoyK | don§t think i've been there.... |
| 20:26.59 | chr|s_ | Quite nice, I f&*&*&king love the German way of life |
| 20:27.04 | Phoenix[1-LAN] | hey stuttart is about 1 h from me |
| 20:27.13 | chr|s_ | The first time someone told me off for washing my car on a sunday outside my house I was so happy :-) |
| 20:27.33 | RoyK | sláinte! |
| 20:27.42 | Phoenix[1-LAN] | hehe |
| 20:27.42 | chr|s_ | man, beer fest... long summer afternoons drinking with chicks, then frolicking in the saunas :-) |
| 20:28.10 | chr|s_ | those damn naughty czech girls who used to come with us, lol, damn *wipes tear* |
| 20:28.20 | RoyK | brb |
| 20:29.28 | chr|s_ | heh, I swapped saunas for Greek beaches now... aah topless sunbathing, man any Americans here who live near a beech where topless bathing is not allowed... take your second ammendment and bust a couple of caps in someone ass until you are allowed to gawk at fit chicks |
| 20:29.36 | Phoenix[1-LAN] | what have you done with asterisk |
| 20:29.47 | Phoenix[1-LAN] | which projects |
| 20:29.52 | chr|s_ | I am setting it up now... |
| 20:30.02 | chr|s_ | I havent seen anyone doing exactly what I want to do |
| 20:30.05 | chr|s_ | but that doesnt dissuade me |
| 20:30.06 | Phoenix[1-LAN] | okay |
| 20:30.18 | Phoenix[1-LAN] | and what u want to do with |
| 20:30.49 | chr|s_ | well, normal vanilla setup, 10 or so soft phones taking calls from incoming lines |
| 20:30.59 | chr|s_ | but, the softphone lookups need to be database driven |
| 20:31.15 | chr|s_ | so, depending on who is calling (or what number is called) dynamically the lookup needs to take place |
| 20:31.16 | Phoenix[1-LAN] | hmmm ok |
| 20:31.20 | Phoenix[1-LAN] | which database? |
| 20:31.22 | Phoenix[1-LAN] | mysql? |
| 20:31.27 | chr|s_ | yep yep |
| 20:31.42 | Phoenix[1-LAN] | nice |
| 20:31.52 | chr|s_ | oracle express won't cut it (long term) and I am sick of oracle at my paid job anyway |
| 20:32.01 | Phoenix[1-LAN] | hehe |
| 20:32.15 | chr|s_ | I will look, pehraps you have heard of some sort of dynamic dial plan? or it support this with script calling? |
| 20:32.38 | Phoenix[1-LAN] | hmmm |
| 20:32.45 | chr|s_ | I had to use 8.2, the last version before column renaming, in a project where there was heaving schema development... on a live application... the numebr of work arounds was a nightmare |
| 20:32.46 | Phoenix[1-LAN] | i was just starting |
| 20:33.00 | Phoenix[1-LAN] | oh thats bad |
| 20:33.38 | Phoenix[1-LAN] | i want to buy a book |
| 20:33.47 | chr|s_ | Attack of the Bacon Robots |
| 20:33.51 | Phoenix[1-LAN] | but in german there is no book |
| 20:34.01 | chr|s_ | you mean an asterisk book? |
| 20:34.05 | Phoenix[1-LAN] | yes |
| 20:34.18 | InfraRed | 8.2? |
| 20:34.36 | InfraRed | oh oracle |
| 20:35.52 | chr|s_ | Yes indeed! anyone used either of the two Java APi wrappers for asterisk? anyone done heavy custom programming with asterisk? |
| 20:36.10 | InfraRed | programing yes |
| 20:36.12 | InfraRed | java, no |
| 20:36.15 | InfraRed | java is evil |
| 20:36.38 | chr|s_ | ahem |
| 20:36.46 | InfraRed | * is a movinf target |
| 20:36.50 | chr|s_ | Java is like oiled up lesbian wrestling |
| 20:36.55 | InfraRed | whatevr you write wont work in the next release |
| 20:37.18 | chr|s_ | InfraRed, that is sad to hear, but luckily I doubt I will upgrade tooo much (look at forth code upgrade since I got started lol) |
| 20:41.48 | Phoenix[1-LAN] | im searching for a asterisk project in the web |
| 20:41.57 | Phoenix[1-LAN] | but i dont find anything |
| 20:42.03 | chr|s_ | :S |
| 20:42.34 | shido6 | ?? |
| 20:42.46 | Phoenix[1-LAN] | with all config files |
| 20:43.03 | Phoenix[1-LAN] | an example what somebody have done with asterisk |
| 20:43.14 | Phoenix[1-LAN] | and what experiences he had |
| 20:43.48 | chr|s_ | would be nice |
| 20:44.02 | chr|s_ | ok, I have a question regarding PSTN and ISDN - about caller ID and redirected calls |
| 20:44.41 | chr|s_ | Phone A calls Phone B, the provider has been instructed to reroute all phone B to phone C (which is asterisk as it happens) |
| 20:44.41 | Phoenix[1-LAN] | oh i try to use isdn but is doesnt work |
| 20:45.00 | *** join/#asterisk RoyK (n=roy@a217-118-45-74.bluecom.no) |
| 20:45.05 | chr|s_ | Imagine A could be any number, and in fact, B could be any number. with PSTN and or ISDN can asterisk (C) know both A and B when the call arrives? |
| 20:45.48 | chr|s_ | if) B forwards to C using a unique number (ie each B phone has a unique number to that line, 555-12345 > 555-9999) or |
| 20:46.05 | chr|s_ | each B phone redirects to the same number (but B's number is somehow detectable from the phone network) |
| 20:47.14 | *** join/#asterisk heison (n=heison@CPE000625e6c664-CM00122570a518.cpe.net.cable.rogers.com) |
| 20:47.42 | chr|s_ | InfraRed, do you really think Java is Evil? :S |
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| 20:55.05 | InfraRed | chr|s_: i am lost in what you're trying to do |
| 20:55.12 | InfraRed | say again |
| 20:55.18 | InfraRed | java *IS* evil |
| 20:56.41 | chr|s_ | o_O |
| 20:57.10 | chr|s_ | InfraRed, out of curiosity are you a Perl programmer InfraRed ? |
| 20:57.14 | InfraRed | you want to pass the CLID? |
| 20:57.19 | InfraRed | c++ |
| 20:57.25 | InfraRed | .net now:) |
| 20:57.29 | InfraRed | c# ftw |
| 20:58.11 | chr|s_ | ok, C# is a bag of balls |
| 20:58.21 | chr|s_ | at least it was for the 8 months I usedit |
| 20:58.52 | chr|s_ | I can categorically say it has less readability than Java, and is more prone to errors, and sucks at threading |
| 20:59.21 | InfraRed | :) |
| 20:59.32 | chr|s_ | aaalso I got the feeling that the whole ASP.NET framework wasn't developed on a solid foundation, it felt hacky, rigid |
| 20:59.34 | InfraRed | i dont like c# either |
| 20:59.54 | InfraRed | but java makes baby jesus cry |
| 21:00.02 | chr|s_ | heh, how so? |
| 21:00.14 | InfraRed | it has the capabilty of turning any reasonable server with 8763487638 processors into a 286 |
| 21:00.23 | chr|s_ | Do a google for square heads demo |
| 21:00.48 | chr|s_ | InfraRed, Java is no different to low level assembly code |
| 21:00.58 | chr|s_ | you confuse start up times with running costs |
| 21:01.09 | chr|s_ | you confuse ovject creation overheads with JVM speeds :p |
| 21:02.19 | chr|s_ | truth: Java can easily run cycles faster than C++ because it can do memory page optimisation in real time, so that after a number of itterations code that gets executed ****VERY*** frequently (and we are talking about only a certain cross section of applications, number crunshin etc) will do so quicker after the runtime optimization of byte code (for this machine) |
| 21:02.51 | chr|s_ | of course, that one simple truth doesn't prove an argument, and I hope I don't bore anyone with my java evangelism, the end if night etc ;-) :- |
| 21:02.55 | chr|s_ | *is nigh |
| 21:03.42 | InfraRed | heh |
| 21:04.43 | InfraRed | another 50 greatest <shit> voted by you |
| 21:04.45 | chr|s_ | I feel with Java, you are talking to the machine directly, you can see how memory is allocated, even if you don't have to do it yourself. |
| 21:05.05 | chr|s_ | lol, I have no idea what that means, but i hope it means I get an ascar |
| 21:05.08 | chr|s_ | *oscar, damn typing |
| 21:05.25 | InfraRed | you've got the bullshit thing going for sure: ) |
| 21:05.50 | chr|s_ | InfraRed, I have used Java for 7 years, and it has made me a lot of money =] I have reasons to love it |
| 21:06.05 | chr|s_ | and reasons to hate it (*remembers 1.2/1.3 work arounds *ouch**) |
| 21:06.07 | *** part/#asterisk freat (n=freat@h-72-244-84-46.chcgilgm.covad.net) |
| 21:06.18 | InfraRed | haha |
| 21:06.33 | *** join/#asterisk mover (n=dlu@gw-dus-net.dus.de.ncore.net) |
| 21:06.34 | InfraRed | 1.4 and 1.5 has gone a long way for sure |
| 21:07.14 | chr|s_ | 1.6 isn't far from being out, and you will see man... just google for square heads demo, and have Java web start installed, and click and run, you will be blown away |
| 21:07.43 | chr|s_ | 1 click app installation, on any JVM, imagine 1 click quake 3 install on linux. |
| 21:08.08 | InfraRed | ooh |
| 21:08.12 | InfraRed | trojans central |
| 21:08.21 | InfraRed | YOU NEED THIS [Y] |
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| 21:10.32 | chr|s_ | InfraRed, trojan central? |
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| 21:10.33 | InfraRed | i can see people exploiting the easiness the same way activeX in windows was exploited fro trojans |
| 21:10.56 | chr|s_ | you mean for 1 click installs? Java runs in a secure memory sandbox, you cannot have a buffer overflow 'sploit in Java, hence it is very very secure, 99% of shit that hits the fan today is unchecked buffers |
| 21:11.02 | InfraRed | no |
| 21:11.07 | chr|s_ | you know, the funny thing is, most viruses, when taken at a macro level are 99% microsoft code, and the cirus writer just 'innocently' calls methods in a certain order, and passes certain data |
| 21:11.36 | chr|s_ | so, since Microsoft wrote 99% of what constitutes worms today, shouldn't they be responsible? |
| 21:11.44 | *** join/#asterisk Tamarisk (n=adrian@user-5659.l6.c4.dsl.pol.co.uk) |
| 21:11.55 | chr|s_ | InfraRed, you cannot get root through a Java process |
| 21:12.31 | chr|s_ | InfraRed, banks use java for a very good reason :-) class verification |
| 21:12.36 | InfraRed | I know |
| 21:12.38 | InfraRed | i am saying |
| 21:12.42 | InfraRed | 21:08 < InfraRed> trojans central |
| 21:12.42 | InfraRed | 21:08 < InfraRed> YOU NEED THIS [Y] |
| 21:12.49 | InfraRed | from users visitng websites |
| 21:12.50 | chr|s_ | aaah I didn't understand that |
| 21:12.52 | InfraRed | java install |
| 21:12.58 | InfraRed | of trojans etc |
| 21:13.03 | chr|s_ | I think i still don't :s |
| 21:13.09 | InfraRed | thats why MS dont show activeX install anymore |
| 21:13.18 | InfraRed | because users click yes |
| 21:13.32 | InfraRed | time for a dump |
| 21:13.35 | InfraRed | bbl; |
| 21:13.38 | chr|s_ | InfraRed, Java could be used to download a dangerous application, using a HttpConnection of course |
| 21:13.45 | chr|s_ | too much info |
| 21:13.59 | InfraRed | :) |
| 21:15.50 | sivana | # |
| 21:16.49 | Tamarisk | Is there a package for a PC that would act as a main switchboard allow the operator to answer calls in any priority |
| 21:17.32 | Tamarisk | Basically on hold or treat as some form of call request until Op decides to call back? |
| 21:17.45 | Tamarisk | SIp calls for 99 % of time |
| 21:18.15 | shmaltz | Tamarisk, again what are you trying to get? |
| 21:18.31 | Tamarisk | Ok |
| 21:19.00 | Tamarisk | I will try my best to explain |
| 21:19.10 | I-MOD | Tamarisk wants a queue where you can pull people out at your descretion |
| 21:19.17 | *** join/#asterisk steveaj (n=steve@62.55.147.53) |
| 21:19.27 | Tamarisk | I am hoping to use * over a wireless network |
| 21:19.37 | Tamarisk | Mesh network. |
| 21:19.53 | Tamarisk | Drivers of vehicles have simple VOIP capabilities |
| 21:20.23 | Tamarisk | and are basically programmed tablets to call certain ops |
| 21:20.29 | *** join/#asterisk FaithX (n=FaithX@202-6-145-116.ip.adam.com.au) |
| 21:20.38 | Tamarisk | Yes I-MOD is correct |
| 21:21.07 | Tamarisk | But the op will decide what calls are important what can wait etc. |
| 21:21.26 | *** part/#asterisk steveaj (n=steve@62.55.147.53) |
| 21:21.55 | saftsack | has someone of you an internal dialtone with asterisk? |
| 21:22.23 | JonR800 | i personally think a wireless mesh might cause a problem. |
| 21:22.26 | Tamarisk | The idea is to use * as the PBX of the system handle SIP voicemail etc |
| 21:22.42 | FaithX | Hey guys... how can I dial in from another location and be able to punch in another phone number to take over voip: |
| 21:23.01 | Tamarisk | JonR800. Can I ask why |
| 21:23.23 | *** join/#asterisk RoyK (n=roy@a217-118-45-74.bluecom.no) |
| 21:23.23 | JonR800 | 1. it's wireless 2. unpredictable latency in a mesh |
| 21:23.28 | I-MOD | FaithX: setup an extension for it |
| 21:24.03 | I-MOD | there is a VERY good chance that packets will arrive out of order alot |
| 21:24.19 | FaithX | I-MOD: Yes but what |
| 21:24.53 | Tamarisk | We will be getting hands on the Motorola MESH system, I have seen streaming video demonstrations and |
| 21:25.05 | I-MOD | exten => _NXXXXXX,1,Dial(Sip/${EXTEN}) |
| 21:25.27 | *** join/#asterisk gaiadog (n=josh@cpe-69-203-211-233.nyc.res.rr.com) |
| 21:25.32 | JonR800 | Tamarisk: buffered streaming video? |
| 21:25.43 | JonR800 | or are we talking real time |
| 21:25.45 | Tamarisk | sat in on a course giving examples of latency depending on number of HOPs |
| 21:25.53 | Tamarisk | As far as I know real time |
| 21:26.02 | FaithX | I-MOD: Hmmm. |
| 21:26.09 | Tamarisk | there is a link or two somewhere let me find |
| 21:26.48 | I-MOD | FaithX: thats the general idea, but probably not correct in your situation |
| 21:27.25 | Tamarisk | http://www.motorola.com/governmentandenterprise/northamerica/en-us/public/functions/browsesolution/browsesolution.aspx?navigationpath=id_804i/id_2523i |
| 21:27.39 | Tamarisk | Sorry very long let me try to find another |
| 21:27.49 | gaiadog | Hello, I have a question.... I would like to forward calls to my cell phone, however often i am on the subway or in another location where my phone is out of range. Naturally the cell phones voicemail answers. Is there a way that I might be able catch this and have the call routed back to the voicemail on the asterisk server? |
| 21:28.18 | FaithX | I-MOD: Well my indial includes my outdial context but I seem to get errors the dtmf doesn't seem to pick upthe numbers properly |
| 21:28.35 | Tamarisk | Another link bit shorter http://www.extremecyborg.com/article/Motorolas+Mesh+Network/169777_1.aspx |
| 21:29.17 | I-MOD | FaithX: pastebin your config |
| 21:29.40 | FaithX | gaiadog: turn your voicemail off on your mobile? |
| 21:31.02 | FaithX | giesen: make your mobile and extension and setup a context that includes voicemail after 30s say |
| 21:31.05 | gaiadog | yeah that may work, however I was thinking maybe somehow I can do this with DTMF; if I actually answer the cell I press # the call is connected. If not the call goes back to asterisk vm. Not sure how I would do that though. |
| 21:31.35 | JonR800 | Tamarisk: i don't see any real info about latency or specific mention of voip.. there is one way to find out for sure though :) |
| 21:31.57 | Tamarisk | I guess try it and see? |
| 21:33.18 | JonR800 | Bingo.. In my very limited experience with mid grade wireless equipment, voip is a toss up. |
| 21:33.31 | FaithX | I guess it seems to me that my dtmf is not received from my mobile correctly |
| 21:33.57 | *** join/#asterisk trelane (i=trelane@66.93.203.199) |
| 21:34.10 | gaiadog | hmm |
| 21:36.55 | FaithX | I punch numbers in and it only seems to receive certain of them... maybe I have a silence cancellation prob with my doc-n-talk -> sipura -> * |
| 21:48.25 | *** join/#asterisk weinerk (n=weinerk@88.152.197.222) |
| 21:48.51 | weinerk | ? |
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| 21:51.13 | *** join/#asterisk Tamarisk (n=adrian@user-5659.l6.c4.dsl.pol.co.uk) |
| 21:51.52 | Tamarisk | Sorry for the loss in connection, MY ISP decided to change my INET address so lost everything from 08:33 |
| 21:51.54 | weinerk | ? |
| 21:52.56 | Tamarisk | Sorry 20:33 |
| 21:54.53 | Tamarisk | Even 21.33 |
| 21:56.19 | *** join/#asterisk lemmy (n=lemmy@developer.g2gui.net) |
| 21:56.23 | lemmy | hi |
| 21:56.49 | I-MOD | hey |
| 21:57.42 | lemmy | does anybody know if its possible to use meetme/app_conference to create a conference and then dial another conferencesystem from this conference room? |
| 21:57.51 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 21:58.07 | lemmy | so that i can link two conference rooms with one landline. |
| 21:58.21 | I-MOD | its doable |
| 21:58.51 | gaiadog | does anyone know how you can transfer a call (say to a mobile phone) and not actually complete the call until # is pressed on the mobile? |
| 21:59.06 | lemmy | I-MOD: with meetme or with app_conference or with both? |
| 21:59.12 | gaiadog | There used to be a "c" option to dial cmd |
| 21:59.40 | I-MOD | lemmy: you should be able to do it with either |
| 21:59.58 | lemmy | I-MOD: great, because i dont want to patch my kernel ;) |
| 22:00.06 | I-MOD | lol |
| 22:00.35 | *** join/#asterisk Mother (n=mother@93.Red-80-32-127.staticIP.rima-tde.net) |
| 22:01.34 | *** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 22:04.10 | weinerk | Where can one buy expert help in small chunks (say half hour blocks)? Thanks |
| 22:04.35 | *** join/#asterisk Luke-Jr (n=luke-jr@user-0c93tin.cable.mindspring.com) |
| 22:04.43 | *** part/#asterisk Luke-Jr (n=luke-jr@user-0c93tin.cable.mindspring.com) |
| 22:07.40 | *** join/#asterisk Luke-Jr (n=luke-jr@user-0c93tin.cable.mindspring.com) |
| 22:08.09 | I-MOD | weinerk: digium, usually in hour chunks though |
| 22:15.05 | mover | support asterisk L16? It also known as Linear Wideband L16/16000 ? |
| 22:18.07 | wulfy814 | ok - I've managed to install asterisk and wanpipe drivers for the A200 - does anyone know whether I should use mulaw or alaw for verizon in PA? |
| 22:18.25 | *** join/#asterisk razu_ (n=razu@80-235-91-173-dsl.prn.estpak.ee) |
| 22:21.08 | *** join/#asterisk coppice (n=chatzill@82.143.17.210.dyn.pacific.net.hk) |
| 22:23.03 | *** join/#asterisk ast_freak (n=ast_frea@68-112-130-237.dhcp.stcl.mn.charter.com) |
| 22:23.46 | shido6 | ulaw |
| 22:25.04 | *** join/#asterisk SWAT (n=SWAT@dsl159-68-100.fastxdsl.nl) |
| 22:25.53 | SWAT | can asterisk handle 'conferencing' with 50 people? And what bandwith does it require per user? I'm considering it as an alternative to for example teamspeak |
| 22:26.54 | MikeJ[Laptop] | SWAT, yes, |
| 22:26.57 | MikeJ[Laptop] | and depends |
| 22:27.04 | SWAT | on what? |
| 22:27.12 | MikeJ[Laptop] | codec for one |
| 22:27.26 | MikeJ[Laptop] | there are all kinds of sizing estimate things out there for voip |
| 22:27.40 | SWAT | haven't inquired information about codecs, yet |
| 22:27.58 | MikeJ[Laptop] | depending on your needs, for 50 people app_conference may work better or worse.. you will need to test in your environment |
| 22:28.20 | SWAT | as I said, it should replace TeamSpeak |
| 22:28.46 | SWAT | would TS require a machine as heavy as Asterisk would require for this task? |
| 22:29.04 | SWAT | then my associate and I could talk it over |
| 22:30.38 | *** join/#asterisk weinerk (n=irc@bzq-88-152-197-222.red.bezeqint.net) |
| 22:31.32 | SWAT | thanks for the information though :) |
| 22:31.37 | *** part/#asterisk Tamarisk (n=adrian@user-5659.l6.c4.dsl.pol.co.uk) |
| 22:33.56 | lemmy | has anybody expreriences compiling app_conference for asterisk 1.2.1? i have dozent compiling errors. :o |
| 22:34.20 | lemmy | and there is no real howto/guide. |
| 22:34.53 | SWAT | lemmy, first off, try the newest version? |
| 22:35.58 | lemmy | SWAT: you mean 1.2.4? |
| 22:36.42 | SWAT | read the channel title. 1.2.5 ? |
| 22:37.42 | lemmy | SWAT: i don't want to upgrade to 1.2.5. 1.2.1 works fine for me and debian has no 1.2.5 packages. |
| 22:37.55 | *** join/#asterisk lzhang (n=lewiszha@67.95.13.46) |
| 22:38.26 | lemmy | but you're saying that app_conference isn't working with 1.2.1? |
| 22:38.34 | SWAT | lemmy, then at least start with giving us a pastebin URL to your errors. Be specific |
| 22:39.25 | *** join/#asterisk powerchip (n=dfskjd@197.80-202-229.nextgentel.com) |
| 22:40.28 | powerchip | how can i make so the auto take the phone , and so the coming a voice , "Hey , Will talk to powerchip press one .." |
| 22:40.34 | *** join/#asterisk RoyK (n=roy@ti211310a080-7445.bb.online.no) |
| 22:41.21 | lemmy | SWAT: no problem, here: http://pastebin.com/598689 |
| 22:41.32 | I-MOD | powerchip: use immediate=yes |
| 22:41.56 | FaithX | It seems only certain DTMF tones are correctly interpreted from my cell phone... 123456 don't work, 789 are ok, *0# make it jump straight as if I did a 0# or a ## or a *# |
| 22:42.36 | FaithX | Noisy connection? |
| 22:42.57 | SWAT | lemmy, btw, I'm not able to help you. But you have a shitload of errors |
| 22:42.59 | powerchip | pl |
| 22:43.01 | powerchip | ok |
| 22:43.23 | powerchip | how can i play voice clip then? |
| 22:43.26 | *** join/#asterisk BladeRunner05 (n=feelme@adsl-ull-144-67.44-151.net24.it) |
| 22:43.54 | SWAT | lemmy, just asking, do have all required components installed? |
| 22:44.41 | BladeRunner05 | Hallo, I'm troubling compiling chan capi 0.3.5 with asterisk 1.2.x, many errors while compiling and stop it, who know how to resolve this ? |
| 22:44.44 | I-MOD | powerchip: in the phone's context, do a exten => s,1,Playback("<your filename here>") |
| 22:45.08 | lemmy | swat: not sure, libc6-dev is install though. |
| 22:45.19 | I-MOD | BladeRunner05: pastebin the errors |
| 22:45.45 | lemmy | SWAT: and the * sources are there too. |
| 22:45.53 | BladeRunner05 | I-MOD: that was on another pc ... |
| 22:46.09 | powerchip | ok nice |
| 22:46.26 | BladeRunner05 | I can teel u that this problem is starting 1.2 version of * |
| 22:46.26 | powerchip | can it play .mp3? |
| 22:47.15 | I-MOD | powerchip: it can if you use mpg123, but its better to just convert it to 8khz wav or gsm |
| 22:47.28 | powerchip | ok |
| 22:47.47 | *** join/#asterisk zoa (n=kkk@pirus.securax.be) |
| 22:47.53 | lemmy | SWAT: but then I'm no c hacker so I might miss something. %) |
| 22:51.33 | RoyK | methinks SWAT is such a lame nick :P |
| 22:52.35 | RoyK | powerchip: sox asdf.mp3 asdf.wav |
| 22:54.12 | powerchip | RoyK: ok nice |
| 22:55.36 | RoyK | sox is nice |
| 22:55.47 | FaithX | It seems only certain DTMF tones are correctly interpreted from my cell phone... 123456 don't work, 789 are ok, *0# make it jump straight as if I did a 0# or a ## or a *# , anyone? |
| 22:56.24 | I-MOD | do you have relaxdtmf=yes? |
| 22:56.53 | RoyK | Faithful: inband? what hardware? |
| 22:57.11 | SWAT | RoyK, somehow I really don't care |
| 22:57.57 | SWAT | finally, I'm clean again! Now I don't have to shower again for another year :) |
| 22:58.09 | RoyK | heh |
| 22:58.50 | *** join/#asterisk ibob63 (n=hp@bb-87-82-25-249.ukonline.co.uk) |
| 22:59.01 | SWAT | mental note :) |
| 22:59.19 | *** part/#asterisk ibob63 (n=hp@bb-87-82-25-249.ukonline.co.uk) |
| 22:59.42 | SWAT | ROFL :D |
| 23:00.32 | SWAT | but seriously there are people out there so smell like shit (not me). There was this guy who stood next to me in a bus... I can't take a lot, but his SMELL made me gag... |
| 23:00.55 | I-MOD | RoyK: are you trying to say that the smell of SWAT could not overwhelm a bear? |
| 23:01.18 | RoyK | I've never smelt a bear |
| 23:01.27 | SWAT | imagine an open sewer, and that on a person... |
| 23:01.28 | RoyK | I've smelt beer, though |
| 23:01.30 | RoyK | Prost! |
| 23:01.59 | SWAT | I prefer the smell of vodka |
| 23:02.02 | core-ix | in bulgarian prost means "stupid" |
| 23:02.12 | RoyK | vodka sucks imho |
| 23:02.24 | RoyK | core-ix: hehe |
| 23:02.29 | RoyK | Skål! |
| 23:02.57 | SWAT | I thought the German "Prost!" which means "Cheers!" |
| 23:03.13 | core-ix | :) |
| 23:03.23 | *** part/#asterisk BladeRunner05 (n=feelme@adsl-ull-144-67.44-151.net24.it) |
| 23:07.15 | *** join/#asterisk gaiadog (n=josh@cpe-69-203-211-233.nyc.res.rr.com) |
| 23:10.42 | chr|s_ | eh: cp: cannot stat `.cleancount': No such file or directory |
| 23:11.06 | powerchip | I-MOD: on phone.conf or what file? |
| 23:11.09 | chr|s_ | building 1.2.5 |
| 23:11.44 | I-MOD | what type of line is it? sip? iax? zap? some other tech i know nothing about? |
| 23:11.46 | chr|s_ | I am right in assuming that you can make clean zaptel, then make it normally and it will overight previous version yep? |
| 23:11.57 | I-MOD | yep |
| 23:12.17 | chr|s_ | cheers |
| 23:12.38 | I-MOD | as long as you have a make install somewhere in there |
| 23:14.04 | chr|s_ | ok starting make from top again, see if it goes through |
| 23:14.32 | powerchip | I-MOD: it is sip |
| 23:15.10 | powerchip | and i have 1 pc on the network so are conectet |
| 23:15.48 | chr|s_ | meh, make fails on asterisk: cp: cannot stat `.cleancount': No such file or directory |
| 23:16.08 | chr|s_ | googling gives me 404 on asterisk website |
| 23:17.14 | I-MOD | powerchip: sip.conf under that phone's context |
| 23:18.21 | powerchip | ok |
| 23:18.29 | chr|s_ | anyone know why .cleancount cannot be found? |
| 23:19.32 | I-MOD | no idea |
| 23:19.42 | I-MOD | its not there? |
| 23:19.53 | chr|s_ | yeah, erm, should I put it there? :s |
| 23:20.22 | *** join/#asterisk r0d3nt|m (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 23:20.49 | I-MOD | what user are you doing make as? |
| 23:20.52 | chr|s_ | root |
| 23:21.17 | chr|s_ | I renamed the folder from asterisk-*version* to just asterisk |
| 23:21.23 | chr|s_ | but I did that before and it worked |
| 23:21.29 | chr|s_ | so that is nuts to think it relies on that |
| 23:22.38 | I-MOD | you can compile asterisk regardless of the name of the directory |
| 23:22.53 | I-MOD | latest src? |
| 23:22.53 | chr|s_ | yeah, figures, so, hrm, puzzling |
| 23:23.15 | chr|s_ | I had another console in that directory |
| 23:23.19 | chr|s_ | maybe that wsa holding something |
| 23:23.37 | I-MOD | have you tried the 1.2.5 tarballed src? |
| 23:23.49 | chr|s_ | this is that |
| 23:24.13 | I-MOD | theres a difference between the "latest" and the 1.2.5 tarball |
| 23:24.14 | chr|s_ | it goes past /stdtime |
| 23:24.23 | *** join/#asterisk Poincare (n=jefffnod@195.207.137.89) |
| 23:24.24 | chr|s_ | I-MOD, I got off front page |
| 23:24.36 | chr|s_ | I gaveup on latests from svn |
| 23:24.40 | chr|s_ | I have 1.2.4 here too |
| 23:24.48 | I-MOD | it doesnt work either? |
| 23:24.51 | *** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 23:24.55 | chr|s_ | I didnt bother trying |
| 23:25.06 | chr|s_ | I think nothing else has changed from the other packages |
| 23:25.37 | chr|s_ | one question: when asterisk went from 1.2.4 to 1.2.5 - but the other versiosn stayed the same, did I still haev to redownload them due to erm, linking to newer version? no right) |
| 23:25.58 | *** join/#asterisk ravsi (n=ravsi@pool-71-108-205-127.lsanca.dsl-w.verizon.net) |
| 23:26.17 | ravsi | how can I see incoming connection attemps? |
| 23:26.39 | powerchip | can i use exten => s,1,Playback("/sound/file.wav") ?? |
| 23:26.39 | I-MOD | chr|s_: it should be fine |
| 23:26.53 | I-MOD | powerchip: leave off the .wav |
| 23:26.58 | I-MOD | ravsi: tcpdump |
| 23:27.02 | chr|s_ | w00t |
| 23:27.03 | chr|s_ | noob |
| 23:27.05 | powerchip | ok |
| 23:27.31 | chr|s_ | I dunno what I did, prolly, no, I have no idea, but I reuntarrd it and viola... now I realise I forgot termcap ;-p |
| 23:27.43 | I-MOD | lol |
| 23:27.58 | I-MOD | the noob badge is fun! |
| 23:28.00 | ravsi | no I mean in asterisk cli |
| 23:28.02 | gaupe | powerchip: you are not mentioning that you are using asterisk@home, that's essential |
| 23:28.06 | powerchip | I-MOD: how know asterisk the fil-type then? |
| 23:28.09 | I-MOD | ravsi: for sip? |
| 23:28.10 | chr|s_ | I-MOD, weirdness, I am sure I untarrd the source and mv'd it to asterisk |
| 23:28.13 | chr|s_ | but the was the other day |
| 23:28.23 | ravsi | I-mod: iax2 |
| 23:28.32 | chr|s_ | and I just untarrd from the tar that was sat there, and that dir (with version) worked, but as you said, i highly doubt that |
| 23:28.36 | I-MOD | powerchip: it checks for wav and gsm types and picks the least cost file type available |
| 23:28.44 | chr|s_ | so this must have been 1.2.4, but AFAIK 1.2.4 should have worked |
| 23:28.57 | I-MOD | ravsi: on the console, type iax2 debug |
| 23:33.57 | ravsi | if my provider shows up in my peers list on iax2 show peers does that mean that the connection and user and pass are all good? |
| 23:34.18 | ravsi | or is it just a basic connection test? |
| 23:34.30 | *** join/#asterisk zotz (n=zotz@24.231.32.85) |
| 23:34.32 | *** join/#asterisk pauldy (n=pauldy@24-155-86-154.ip.grandenetworks.net) |
| 23:34.51 | ravsi | I am trying to figure out if the problem is me or them |
| 23:35.28 | *** join/#asterisk Mavvie (n=edwin@252-131-222-203.rev.techex.net.au) |
| 23:35.53 | Mavvie | Anybody from .au here who is willing to test if it's me who is going crazy or that it's something else going on? |
| 23:36.33 | kippi | does SIP not like NAT? |
| 23:36.53 | Mavvie | kippi: it can work around it, but it wasn't designed for a NATted network. |
| 23:37.00 | Mavvie | kippi: just like FTP for example. |
| 23:37.22 | I-MOD | ravsi: no audio, or no connection whatsoever? |
| 23:37.34 | kippi | Mavvie: I have opened all ports that are need for SIP but my phone dosn't seem to be getting to my asterisk box |
| 23:37.47 | ravsi | I-MOD: the provider is rolling over to my failover pstn line |
| 23:38.15 | I-MOD | so no connection whatsoever |
| 23:38.46 | chr|s_ | what it says termcap not found, what is the name of the library / -devel? |
| 23:38.59 | I-MOD | ncurses |
| 23:39.05 | chr|s_ | aah yes |
| 23:39.09 | chr|s_ | and newt something? |
| 23:39.13 | I-MOD | libnewt |
| 23:39.19 | I-MOD | but thats only for zttool |
| 23:39.28 | chr|s_ | ill have it anyway |
| 23:39.51 | chr|s_ | phew there ya go, simple enough |
| 23:39.58 | chr|s_ | thanks |
| 23:40.01 | I-MOD | np |
| 23:40.52 | ravsi | I-MOD: if I do telnet to 4569 on local host it's denied |
| 23:41.01 | ravsi | did I not open somthing? |
| 23:41.26 | I-MOD | ravsi: do a tcpdump, but filter it so that only communication between you and your iax provider is dumped |
| 23:41.47 | I-MOD | ravsi: does it say connection refused or connection timeout? |
| 23:41.59 | ravsi | refused |
| 23:42.05 | chr|s_ | when in the docs it says 10-15 chanels that means simultaneously right? |
| 23:42.17 | I-MOD | yes |
| 23:42.27 | I-MOD | but i've seen boxes go way over that |
| 23:42.44 | chr|s_ | so a 3ghz 1gb ram could do that... I may need upto 100 simultaneous, but from the same system... is that possible? |
| 23:42.45 | justinu | they're also talking about analog zap, i believe |
| 23:42.55 | kippi | anyone got anyideas what I can do to get my phone connect? I am guessing its because of NAT |
| 23:43.01 | chr|s_ | I think I will be using T/E 1 |
| 23:43.06 | justinu | if you're doing sip or pri, you can deal with 100 channels no prob |
| 23:43.24 | justinu | unless you transcode to g729 or other compressed codec |
| 23:43.33 | chr|s_ | justinu, I will only have FXO ports on my cards, all phones will be soft phones (does that make sense?) |
| 23:43.42 | justinu | yes |
| 23:43.48 | I-MOD | kippi: sip doesnt like nat and doesnt work at all when the connection is dual natted |
| 23:43.53 | chr|s_ | justinu, I guess I don't need to compress if it is all internal and I have gigabit ethernet |
| 23:44.01 | justinu | so you're use the 4 span T1 card? |
| 23:44.04 | justinu | you'll |
| 23:44.32 | chr|s_ | justinu, that is what I will find out :s right now I am testing with a 4 FXO card |
| 23:44.36 | kippi | I-MOD: so its possiable that i'll nerver get it to connect? |
| 23:44.47 | justinu | 4 FXO... you said T1 tho |
| 23:45.09 | I-MOD | kippi: double nat, 99.99999% no, single nat works most of the time |
| 23:45.19 | justinu | you 3ghz PC will handle 4 FXOs, or 96 DS0 no problem |
| 23:46.21 | *** join/#asterisk Flauto (n=zhao@71.194.194.48) |
| 23:46.27 | Flauto | hi there |
| 23:46.29 | kippi | I-MOD Is there a test I can do to see if its double NAT? |
| 23:46.30 | I-MOD | ravsi: make sure your port forwarding is right |
| 23:46.41 | Flauto | anyone here is using vbuzzer? |
| 23:46.51 | I-MOD | kippi: you said the asterisk server is behind a nat right? |
| 23:46.57 | ravsi | on my firewall or local? |
| 23:47.16 | I-MOD | ravsi: do you have iptables blocking traffic? |
| 23:47.20 | kippi | I-MOD yeah, and my phone is, so its not going to work? |
| 23:47.33 | chr|s_ | justinu, this is a test box |
| 23:47.43 | chr|s_ | I am using PSTN at home |
| 23:47.46 | Flauto | hello |
| 23:47.53 | Flauto | anyone here is using vbuzzer? |
| 23:47.55 | chr|s_ | it is enough to develop the database backend though |
| 23:48.00 | I-MOD | kippi: most likely not, I have only seen it correctly done on ONE occasion out of several hundred |
| 23:48.03 | *** join/#asterisk RoyK (n=roy@ti211310a080-7445.bb.online.no) |
| 23:48.21 | chr|s_ | I am suddenly very optimistic about asterisk :-) even moreso |
| 23:48.52 | kippi | I-MOD ok, many thanks for your help, going to have to have to have a re-thing, maybe setup a vpn or somthing |
| 23:49.30 | I-MOD | alright, i'm out for a while |
| 23:50.33 | ravsi | I-MOD: so if it is proper, then telnet should make a connection on 4569 right? |
| 23:51.14 | zigman | no iax is udp |
| 23:51.18 | zigman | telnet is tcp |
| 23:51.27 | _chr|s_zzz | thanks peeps |
| 23:52.21 | ravsi | is there a simple way of testing to see if asterisk is making a connection? |
| 23:52.42 | De_Mon | ravsi what do you mean? |
| 23:52.54 | ravsi | I am pretty sure my firewall is passing udp 4569 |
| 23:53.11 | *** join/#asterisk xyklopz (n=xyklopx@216-91-89-21.biltmorecomm.com) |
| 23:53.51 | ravsi | Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE |
| 23:54.04 | ravsi | that is in the debug |
| 23:54.18 | De_Mon | there isn't anyting more siple than dialing an extension |
| 23:54.25 | Flauto | there are 290 people here but nobody is using vbuzzer? |
| 23:54.29 | xyklopz | Could someone tell me if it is possible to write a script (say in python), to do an automatic call where upon answering, an sequence of audio clips (digital voice representations of the numbers 0 - 9) is played and then the call ends |
| 23:55.03 | De_Mon | xyklopz why not do that in extensions.conf |
| 23:55.20 | xyklopz | it's going to be in response to an event through a webpage |
| 23:56.01 | ravsi | De_Mon: I have just exten => s,1,Answer() +playback+hangup |
| 23:56.06 | xyklopz | I'm still fairly new to "hands-on" * but have read quite a bit |
| 23:56.20 | De_Mon | xyklopz that didn't answer my question |
| 23:56.31 | xyklopz | okay ... it's not due to someone calling |
| 23:56.52 | De_Mon | if they arn't calling asterisk, why are you in this channel? |
| 23:57.01 | xyklopz | i need asterisk to make the call |
| 23:57.13 | Mavvie | This is just plain strange. When I call a number in my own PSTN network, it stays in Ring. That means that the user doesn't get a ringing tone. When I call outside my own PSTN network, it goes through to Ringing well euhm... pretty soon enough. |
| 23:57.48 | xyklopz | and I need a php page to tell it when to make the call and provide some mechanism to return success/failure |
| 23:58.19 | xyklopz | that's why I mentioned something like python and just doing it through a command line interface |
| 23:58.23 | De_Mon | ahh, now you're starting to make sense |
| 23:58.40 | xyklopz | ./script <number_to_call> <sequence_to_play> |
| 23:59.12 | xyklopz | return value of the process maps to a table of values specifiy the status of the attempt (busy, PSTN down, success, etc...) |
| 23:59.24 | xyklopz | *specifying |