| 00:04.01 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 00:05.50 | dijungal | what's the difference between safe_asterisk and asterisk? |
| 00:06.47 | fujin | one automatically rsetarts asterisk if it dies |
| 00:07.00 | fujin | the other just dies ;) |
| 00:10.32 | *** join/#asterisk DoDaT69 (n=DoDaT69@internal.digitalson.com) |
| 00:11.14 | DoDaT69 | has anyone successfully installed hudlite server on a flat asterisk install? |
| 00:14.18 | JT | hmmhesays: that wouldn't be a good guess, as it would be slin |
| 00:14.37 | JT | actually it could be slin or alaw |
| 00:15.51 | dijungal | fujin: thanks |
| 00:21.43 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 00:21.58 | DoDaT69 | has anyone successfully installed hudlite server on a flat asterisk install? |
| 00:21.59 | hmmhesays | anyone running any websites with google adsense? |
| 00:22.21 | *** join/#asterisk craigk (n=ckowald@58.174.150.119) |
| 00:22.35 | hmmhesays | oh yeah? what kind of revenue do you get off of that? |
| 00:22.51 | DoDaT69 | damn near nothing-- no one ever clicks the links |
| 00:23.06 | hmmhesays | don't they have "pay per impression" also? |
| 00:23.17 | DoDaT69 | not that I've seen |
| 00:23.30 | DoDaT69 | maybe I should login and check |
| 00:23.54 | hmmhesays | pay per thousand impressions as far as I know |
| 00:24.18 | DoDaT69 | up until the 1st I ran a store that received about 20k/month |
| 00:24.22 | DoDaT69 | in hits |
| 00:24.42 | hmmhesays | and you didn't get paid per impression? |
| 00:25.15 | DoDaT69 | checking now |
| 00:25.34 | DoDaT69 | Hmm thats odd |
| 00:25.42 | DoDaT69 | it says I dont have an account, but the ads display |
| 00:25.46 | DoDaT69 | someone is getting paid then..... |
| 00:26.00 | hmmhesays | google |
| 00:26.16 | DoDaT69 | thats a kicker.. I know I signed up for an account, thats how I have the # string to stick in my page |
| 00:26.21 | DoDaT69 | those bastards! |
| 00:27.26 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 00:33.46 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-b8eb5014af57f78b) |
| 00:36.44 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 00:37.10 | hmmhesays | I'm still waiting for them to approve mine |
| 00:38.37 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 00:44.30 | *** join/#asterisk fskrotzki (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 00:50.29 | *** join/#asterisk mcab (n=mb@mostly-harmless.ca) |
| 00:59.25 | *** join/#asterisk mltlnx (n=mltlnx@rrcs-208-125-29-189.nyc.biz.rr.com) |
| 01:01.22 | *** join/#asterisk test34_ (n=test34@c-67-162-175-187.hsd1.fl.comcast.net) |
| 01:07.05 | *** join/#asterisk SkramX (n=mark@cpe-70-112-25-138.austin.res.rr.com) |
| 01:07.13 | test34_ | Can you buy a phone number without service and use your own server to route calls ? |
| 01:07.28 | SkramX | has anyone been able to install/register/whatever g729 through digium with 1.4.15? |
| 01:07.40 | SkramX | I was able to run the register script but it's now showing up in `show translations` |
| 01:08.35 | *** join/#asterisk tobias (n=tobias@user-0ce2hr3.cable.mindspring.com) |
| 01:08.48 | mosty | test34_, what service would you use to accept calls? |
| 01:09.31 | test34_ | mosty, my ip address ? |
| 01:09.51 | mosty | test34_, many voip service providers will do that |
| 01:10.13 | mosty | SkramX, do you have the g729a module installed, and have you restarted asterisk? |
| 01:10.27 | SkramX | im retrying |
| 01:10.35 | test34_ | mosty, is it alot cheaper? |
| 01:10.37 | SkramX | redownloaded .so and regsiter utility |
| 01:11.14 | mosty | test34_, sometimes |
| 01:12.32 | SkramX | all fixed, woopsies ;) |
| 01:13.03 | test34_ | mosty, do you know where I could find some documention about the process ? I didn't see anything yet on google |
| 01:14.11 | mosty | test34_, the process for what exactly? you have to setup/configure asterisk, and get a sip or iax account with a voip provider that provides the service. if you don't know where to start, read the book |
| 01:14.13 | mosty | ~thebook |
| 01:14.24 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 01:14.49 | test34_ | thanks mosty |
| 01:18.33 | barhom | I dont get it, adding whatever I have in my "include => internal" directly to the context is working, but having it through the include function is not, is there anything special I need to think about when using include ? |
| 01:21.08 | *** join/#asterisk jsoftw (n=Administ@60.234.135.124) |
| 01:23.18 | ardor2 | mvanbaak: just added the extra code to make it work |
| 01:23.36 | *** join/#asterisk mltlnx (n=mltlnx@cpe-72-229-184-242.nyc.res.rr.com) |
| 01:23.39 | ardor2 | mvanbaak: with freepbx/trixbox |
| 01:24.14 | barhom | [internal] deadel,1,Dial(SIP/deadel) deadel,n,Hangup [mycontext] include => internal |
| 01:24.29 | barhom | using mycontext it isnt including the internal for some reason, its not dialing deadel |
| 01:24.37 | *** join/#asterisk tc3driver (n=huh@rrcs-24-199-16-118.west.biz.rr.com) |
| 01:30.51 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 01:32.55 | *** join/#asterisk |omni| (n=rob@c-67-185-144-206.hsd1.wa.comcast.net) |
| 01:37.45 | Yourname` | Hmm, what could it be that's causing this. When I dial a number, on eyebeam, it says Calling .. for about 2 seconds, before I see it on Asterisk CLI.. why the wait? Wasn't there before. |
| 01:43.27 | *** join/#asterisk a1fa (n=zZZ@unaffiliated/a1fa) |
| 01:43.41 | a1fa | is there a way to call two different extensions and have them talk at the same time? |
| 01:43.54 | a1fa | without using a conference room? |
| 01:45.29 | a1fa | <jeopardy music> |
| 01:45.59 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.ITB.ac.id) |
| 01:48.56 | fujin | a conference room |
| 01:49.00 | fujin | meetme or app_conference |
| 01:49.43 | a1fa | you have to drop users into the conference room |
| 01:49.44 | a1fa | right? |
| 01:50.50 | fujin | yes |
| 01:51.04 | fujin | you will have to put channels into the conference |
| 01:51.53 | a1fa | what about if you have 300+ channels |
| 01:53.03 | Yourname` | http://pastebin.ca/812863 -> I want the number of the PSTN caller in the callerid(num) field.. how do I? |
| 01:55.17 | a1fa | fujin: so cmd_conference can handle 300+ channels? |
| 01:59.46 | fujin | I don't know |
| 01:59.46 | fujin | I don't use it |
| 01:59.52 | fujin | read the documentation? |
| 02:00.24 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
| 02:00.28 | Yourname` | Hello errrbody, http://pastebin.ca/812863 -> I want the number of the PSTN caller in the callerid(num) field.. how do I? |
| 02:00.38 | Yourname` | sup fujin |
| 02:00.53 | fujin | howdy |
| 02:01.18 | fujin | post expired Yourname` |
| 02:01.28 | Yourname` | How's it hangin, see that roughstud sam again? lol |
| 02:01.42 | fujin | ha |
| 02:01.43 | fujin | nah |
| 02:01.46 | fujin | haven't seen him for a while. |
| 02:01.54 | Yourname` | oh. oops, http://pastebin.ca/812874 |
| 02:02.26 | fujin | uh |
| 02:02.35 | fujin | the number of the pstn caller *will* be in callerid(num) |
| 02:08.32 | Yourname` | But is isn't. |
| 02:08.37 | Yourname` | it* |
| 02:08.39 | fujin | huh? |
| 02:08.42 | fujin | if a call comes in from the PSTN |
| 02:08.47 | fujin | the value of callerid(num) |
| 02:08.54 | fujin | will be the callerid |
| 02:09.17 | Yourname` | Ok, what about an outbound call made TO a pstn and then transferred to 300? How could I put THAT number in callerid(num)? |
| 02:09.32 | fujin | that'd be a little harder. |
| 02:12.32 | Yourname` | ah |
| 02:16.17 | osiris | YAY~! finally got my trixbox to call in and out over my provider~! |
| 02:16.30 | *** join/#asterisk macros73_ (n=cs@c-24-3-246-27.hsd1.pa.comcast.net) |
| 02:17.02 | *** join/#asterisk justnulling2 (n=menashe@ool-457bcf75.dyn.optonline.net) |
| 02:18.18 | osiris | NGT's broadsoft can be a pita |
| 02:21.18 | *** join/#asterisk switched (i=juanchic@oj.dreamhost.com) |
| 02:22.32 | ManxPower | Calls to the PSTN cannot set their callerid unless you are using ISDN (PRI/BRI) (most VoIP providers use PRIs) |
| 02:22.41 | ManxPower | You can never set the name. |
| 02:22.43 | switched | i suspect that i missed a step when configuring/compiling asterisk - right now in asterisk -r, i can't issue a PRI command like pri debug span 1 |
| 02:23.13 | ManxPower | switched: then you did not have libpri and zaptel installed before you built Asterisk |
| 02:24.15 | switched | ManxPower: should I just blow away everything and start again? I tried just going back and remaking libpri then zaptel then asterisk, but still the same problem |
| 02:24.44 | ManxPower | switched: I have no idea how to fix the problem with 1.4 |
| 02:25.05 | switched | is that a common complaint with 1.4? |
| 02:25.20 | switched | i just want to get it to work so I can see if our D channel is live on the PRI |
| 02:25.21 | ManxPower | most people manage to figure out to install zaptel and libpri before installing Asterisk |
| 02:25.33 | ManxPower | I don't use 1.4, so I can't say. |
| 02:25.47 | switched | yeah, that was going to be my next step, but I just wanted to verify |
| 02:26.04 | switched | i mean, why can't i just re-make what I have? |
| 02:26.08 | switched | becaus apparently that doesn't work |
| 02:26.21 | switched | like make clean, and make for libpri/zaptel/asterisk |
| 02:26.30 | switched | er ..make clean/make/make install |
| 02:26.37 | ManxPower | Oh, I'm sure you can tell the build system to re-compute the installed libraries, but *I* do not know how, as I use 1.2 |
| 02:26.50 | ManxPower | no, make clean won't do it. |
| 02:27.06 | ManxPower | you can do something in menuconfig I hear. |
| 02:27.06 | switched | i guess..just to make sure..then start from scratch and reinstall the os |
| 02:27.16 | ManxPower | no need to reinstall the OS. |
| 02:27.18 | switched | if i don't want to bother thinking about it |
| 02:27.28 | switched | hm |
| 02:28.14 | ManxPower | re-download asterisk if you don't have the original source handy, install asterisk after you have installed zaptel and libpri |
| 02:28.31 | ManxPower | This is not rocket science. |
| 02:28.49 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177581616.dsl.bell.ca) |
| 02:28.49 | *** part/#asterisk DrRighteous (n=DrRighte@ool-44c7ad06.dyn.optonline.net) |
| 02:29.12 | switched | yeah well i did do that, and still the same problem |
| 02:29.29 | ManxPower | then you have some other problem |
| 02:29.34 | switched | but i'm coming from a 1.4 perpsective - but surely it hasn't changed that much |
| 02:29.45 | switched | i mean, it's the same 3 packages |
| 02:29.51 | ManxPower | The entire build system was rewritten for 1.4 |
| 02:30.01 | switched | 1.4 has been out for a while right? |
| 02:30.04 | ManxPower | (at least for asterisk) |
| 02:30.11 | ManxPower | 1.4 has been out a while. |
| 02:31.16 | *** join/#asterisk ZX81 (n=matt@202.49.106.158) |
| 02:32.13 | ZX81 | hi all - got a weird problem here - a machine at a customer's has three iax accounts - one to each of our exchanges. They can ping all three servers but two are unreachable via IAX (not firewall involved) |
| 02:32.50 | ManxPower | The thing is that the Asterisk 1.4 build system, the first time you run it, it looks for zaptel and libpri. It saves that information and does not check again when you run it again, which is why I suggested you start from virgin source. |
| 02:33.01 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 02:33.49 | ZX81 | or do make distclean |
| 02:34.11 | ManxPower | ZX81: As I don't use 1.4, I was not aware of that option. |
| 02:34.29 | ZX81 | yeah kinda annoying though cos it deletes the sound files too :) |
| 02:34.29 | ManxPower | But since that did not work for switched, I suspect he's doing something odd, like using BSD or something |
| 02:34.43 | ZX81 | oy! what's wrong with BSD! |
| 02:34.44 | ZX81 | :) |
| 02:34.47 | *** join/#asterisk pirulo (n=andres_p@70.56.223.76) |
| 02:35.09 | ManxPower | ZX81: Since the problem has not been fixed in the first 15 releases of 1.4, I assume it is a design decision, rather than total and utter stupidity. |
| 02:35.19 | ZX81 | lol yeah |
| 02:35.56 | JT | ZX81: asterisk is only designed to work with linux unfortunately |
| 02:39.57 | ZX81 | asterisk-bsd may disagree :) |
| 02:40.03 | ZX81 | as may luigi :) |
| 02:40.15 | JT | that is not asterisk plain |
| 02:40.29 | switched | the windows admin within me says, "reinstall" .. since I have not much to lose anyway |
| 02:40.30 | JT | goes to show that asterisk really was designed only to work with linux |
| 02:40.41 | JT | if they had to make a modified version :) |
| 02:40.55 | ZX81 | :) |
| 02:41.11 | ZX81 | if it were clean pure c it would work - mostly |
| 02:41.25 | ZX81 | I rewrote our predictive dialer in c on a linux box |
| 02:41.38 | ZX81 | then installed freebsd and recompiled and it worked fine |
| 02:41.48 | ZX81 | obviously not as complicated as asterisk though :) |
| 02:41.57 | JT | pure and clean, and didn't have all this hardware and kernel timers |
| 02:42.05 | ZX81 | correct :D |
| 02:49.51 | switched | when you say "asterisk build system" that includes the MakeFile and what else? |
| 02:50.37 | ManxPower | the ./configure script, the "make menuconfig" (whatever that is) |
| 02:51.22 | *** part/#asterisk switched (i=juanchic@oj.dreamhost.com) |
| 02:53.09 | *** join/#asterisk switched (i=juanchic@oj.dreamhost.com) |
| 03:01.13 | *** join/#asterisk pirulo (n=andres_p@70.56.223.76) |
| 03:02.13 | Yourname` | What kind of config would be good for an openser box to be used as a border controller trying to process around 10,000 channels atleast (coming from various asterisk dialers) and taking care of CDRs ? |
| 03:02.55 | JT | concurrent calls? what about CPS rate? |
| 03:03.16 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-914902f5bb7ef230) |
| 03:05.59 | *** part/#asterisk switched (i=juanchic@oj.dreamhost.com) |
| 03:08.11 | *** join/#asterisk klyrelion (n=Kevin@boyne.demon.co.uk) |
| 03:08.42 | *** join/#asterisk joat (n=joat@ip70-160-147-169.hr.hr.cox.net) |
| 03:12.06 | *** join/#asterisk KeltusEx (i=Keltus@c-76-21-119-166.hsd1.ca.comcast.net) |
| 03:15.38 | *** join/#asterisk alrs (i=non-knav@pozug.com) |
| 03:26.44 | JT | Yourname`: ? |
| 03:27.19 | Yourname` | JT: Yeah, that too. Would like to control CPS/concurrents, etc. |
| 03:27.30 | Yourname` | (Sorry, was afk for a bit.) |
| 03:29.32 | JT | i thought you might have some estimates |
| 03:30.58 | *** join/#asterisk switched (i=juanchic@oj.dreamhost.com) |
| 03:31.51 | switched | if my PRI is not live, should that prevent asterisk from letting me at least issue the command "pri debug span #" ? |
| 03:32.14 | JT | no |
| 03:32.57 | switched | IOW, i type in pri, and it says command not found. Hmm I just reinstalled FC8, then did libpri, zaptel and asterisk in that order. |
| 03:33.10 | switched | lsmod shows zaptel and the kernel module that corresponds to my card |
| 03:33.21 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-cadd21c39dd7ff00) |
| 03:33.59 | switched | is there a separate SIP module? I can't issue sip command either |
| 03:34.26 | switched | i wonder, maybe i skipped a step, that there's some directory of asterisk modules that i need to build |
| 03:34.38 | switched | and this is a 1.4 stuff |
| 03:34.40 | switched | all |
| 03:35.56 | JT | check you've loaded all the relevant modules |
| 03:36.31 | switched | you mean other than the kernel modules? |
| 03:36.40 | switched | how do i do that? |
| 03:36.52 | switched | log file? |
| 03:37.14 | *** join/#asterisk hardwire (n=bip@rdbck-7085.palmer.mtaonline.net) |
| 03:37.20 | hardwire | hello dere |
| 03:37.59 | switched | ah ..modules.conf? |
| 03:42.02 | *** join/#asterisk klyrelion (n=Kevin@icsnet.demon.co.uk) |
| 03:42.07 | *** part/#asterisk klyrelion (n=Kevin@icsnet.demon.co.uk) |
| 03:43.18 | Yourname` | Hi, what could it be that's causing this. When I dial a number, on eyebeam, it says Calling .. for about 2 seconds, before I see it on Asterisk CLI.. why the wait? Wasn't there before. |
| 03:43.51 | *** join/#asterisk Zoolooc (n=fredsiba@p54952B22.dip0.t-ipconnect.de) |
| 03:52.37 | *** join/#asterisk angom (n=Angel@201.170.35.218) |
| 03:56.44 | joat | i'm hearing complaints from friends that FWD is acting up... anyone have issues? |
| 03:57.00 | bjweeks | FWD is always acting up |
| 03:57.24 | joat | supposedly it's refusing registration (clear text and md5) this evening |
| 03:57.48 | bjweeks | wouldn |
| 03:57.59 | bjweeks | 't doubt it, they always have problems |
| 03:58.30 | joat | hmm.. i've never had any problem. is there any equivalent to it available? |
| 04:00.09 | bjweeks | public sip network? gizmo is one |
| 04:02.04 | *** join/#asterisk gardo (n=gardo@121.97.79.51) |
| 04:02.29 | joat | true... they're locked down a bit more though... |
| 04:02.54 | *** join/#asterisk _mm_ (n=mmclain@cpe-75-80-238-180.dc.res.rr.com) |
| 04:02.58 | joat | managed to run a few extra concurrent calls through fwd recentlly |
| 04:05.57 | *** join/#asterisk marcan (i=1337@host215-248.cvf.fit.edu) |
| 04:08.22 | Nivex | I'm registered to FWD via IAX2 right now |
| 04:12.19 | *** join/#asterisk shmaltz (n=mybox@mail2.dmaven.com) |
| 04:12.31 | shmaltz | is there a way to clear all channel variables? |
| 04:12.48 | MrTelephone | yourname? |
| 04:38.13 | *** join/#asterisk neoalex (n=chatzill@user-0ccengj.cable.mindspring.com) |
| 04:38.33 | neoalex | does anyone have a stanaphone account their not using by any chance? |
| 04:40.33 | *** join/#asterisk snafsnaf (n=sdf@159.114.dsl.mel.iprimus.net.au) |
| 04:41.09 | snafsnaf | Hello All |
| 04:41.31 | MrTelephone | what is that i wonder |
| 04:41.49 | snafsnaf | Anyone know if there is any way to disallow zombie channels? or kill then as soon as they appear? |
| 04:42.03 | snafsnaf | They are wreaking havoc with my AGI rating script |
| 04:48.29 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 04:49.13 | hmmhesays | oh what a slow time of night |
| 04:49.24 | hmmhesays | why are you getting zombie |
| 04:49.25 | hmmhesays | s |
| 04:52.19 | hmmhesays | god I want some vodka and oj |
| 04:53.44 | snafsnaf | Im not 100% sure, but I read about a similar issue on a mailing list, which turned out to be reinvites not reaching their destination |
| 04:54.17 | hmmhesays | thats very possible |
| 04:55.40 | snafsnaf | and if that is the case there is not alot I can do to remove the cause |
| 04:55.56 | *** join/#asterisk groogs[h] (n=gregm@cbl-66-102-80-229.wtccommunications.ca) |
| 04:59.54 | hmmhesays | don't reinvite |
| 05:00.27 | *** join/#asterisk hfb (n=hfb@75.80.37.175) |
| 05:11.51 | ManxPower | snafsnaf: I assume you are doing the standard AGI cleanup when a channel hangs up, right? |
| 05:14.18 | snafsnaf | correct |
| 05:23.51 | snafsnaf | ManxPower: Were you just verifying that, or did you have an idea if the above was true? |
| 05:38.58 | ManxPower | My idea was to start dong it if you were not doing it. |
| 05:39.04 | ManxPower | see the asterisk-perl sample programs |
| 05:43.13 | snafsnaf | Yes, I am using it, the problem is zombie channels do not match the 'h' extension. |
| 05:43.51 | ManxPower | snafsnaf: zombies are normally harmless |
| 05:49.32 | *** join/#asterisk ZX81 (n=matt@202.20.97.211) |
| 05:53.24 | *** join/#asterisk sg` (n=saurabh@59.160.224.34) |
| 05:54.35 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-2490172988822374) |
| 06:00.11 | snafsnaf | ManxPower: The issue is the calls are actually passing, we are being billed for them, but our customers are not because the zombies are not rated by AGI script that runs in the 'h' extension. |
| 06:07.29 | ZX81 | snafsnaf: you need to have 2 agi files |
| 06:07.35 | ZX81 | one for routing one for billing |
| 06:07.41 | ZX81 | routing as in: |
| 06:07.50 | ZX81 | exten => _X.,1,AGI(route.php) |
| 06:07.54 | ZX81 | and billing as in: |
| 06:08.07 | ZX81 | exten => h,1,DeadAGI(billing.php) |
| 06:08.16 | ZX81 | route.php needs to set costs etc |
| 06:08.26 | ZX81 | and billing.php checks answered status etc |
| 06:13.06 | *** join/#asterisk timgws (n=LivedTyp@202.172.97.51) |
| 06:13.45 | timgws | Hi, does anyone have any thoughts on a good Open Source billing application for Asterisk? |
| 06:14.13 | bjweeks | http://www.voip-info.org/wiki/view/Asterisk+billing |
| 06:14.15 | denon | I would imagine that someone does |
| 06:14.39 | timgws | I have tried MOR and A2Billing, but both of them have things that I don't like |
| 06:14.43 | timgws | especially MOR :P |
| 06:15.08 | *** join/#asterisk mosty (n=mostyn@60-241-198-194.static.tpgi.com.au) |
| 06:15.32 | snafsnaf | ZX81: That is how it is set up |
| 06:16.32 | snafsnaf | but when dialer.php (route.php) passes to Dial() sometimes this results in a Zombie channel |
| 06:16.49 | snafsnaf | and exten => h does not rate the call |
| 06:17.00 | snafsnaf | timgws: write your own? |
| 06:17.24 | timgws | snafsnaf: Working on it, but just before I spend the time to finish it, I just want to double check :P |
| 06:17.33 | timgws | I mean, why re-invent the wheel? |
| 06:17.53 | snafsnaf | If you can tailor it, why not? |
| 06:18.24 | snafsnaf | there are always going to be things you like and things you dont and things that are missing, but if you write your own, you can avoid all those issues |
| 06:19.05 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 06:30.24 | *** join/#asterisk Maliuta (n=nikolai@ppp214-92.static.internode.on.net) |
| 06:33.16 | awk | anyones brain working this morning? |
| 06:33.18 | awk | http://www.pastebin.ca/813060 |
| 06:33.41 | awk | any idea why these channels don't hangup.. and as you can see, I get avoiding dead lock.. I can't even soft hangup these channels.. |
| 06:34.07 | bjweeks | just wondering, is "17 of 255 max active calls ( 6.67% of capacity)" for g.729? |
| 06:34.26 | awk | and over time I start to get more.. this has been a few days.. and I have 2... but i've had up to 10.. |
| 06:35.48 | awk | hmm, some of their VOIP calls are using g729.. but local calls are using ALAw |
| 06:38.09 | awk | if I try softhangup and tab accross it actually kills my session.. hangs it.. have to re-login... if I try actually hangup the exact channel it says it doesn't exsist? |
| 06:48.21 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 06:48.21 | *** mode/#asterisk [+o denon] by ChanServ |
| 06:52.32 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 06:52.32 | *** mode/#asterisk [+o denon] by ChanServ |
| 06:53.53 | DarKnesS_WolF | anyone knows a good cheap A-Z termination " IAX2 " that accept normal visa no need for paypal or moneybooker? |
| 06:59.21 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 06:59.32 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 06:59.35 | *** join/#asterisk nirz (n=nir@mail2.tikalnetworks.com) |
| 07:03.23 | *** join/#asterisk ussrback (n=MAX@80.241.177.19) |
| 07:04.59 | *** join/#asterisk macros73_ (n=cs@c-24-3-246-27.hsd1.mn.comcast.net) |
| 07:08.17 | *** part/#asterisk ussrback (n=MAX@80.241.177.19) |
| 07:12.46 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 07:16.21 | *** join/#asterisk mog (n=mog@c-68-62-172-83.hsd1.al.comcast.net) |
| 07:16.21 | *** mode/#asterisk [+o mog] by ChanServ |
| 07:25.28 | awk | Gradwell |
| 07:25.39 | awk | hrm, can nobody help me out? |
| 07:25.56 | awk | fancy that a channel filled with asterisk guys who have no clue how to solve this issue.. |
| 07:27.26 | [hC] | probably worth mentioning that most people are asleep |
| 07:27.56 | [hC] | and insulting people to guilt them into helping you probably wont work either |
| 07:29.26 | JT | indeed |
| 07:33.44 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 07:45.07 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-e80e2b9706813c6b) |
| 07:45.32 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 07:48.16 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 08:06.43 | *** join/#asterisk Schumie (n=Steve@cmarfw01.marlow.spinvox.com) |
| 08:06.53 | *** join/#asterisk slima (i=slima@unaffiliated/slima) |
| 08:11.38 | *** join/#asterisk vgster (n=vgster@psc.navonline.net) |
| 08:11.49 | *** join/#asterisk guomi (n=francois@c2cpc3.camptocamp.com) |
| 08:14.57 | *** join/#asterisk harpal (n=Harpal@124.125.255.24) |
| 08:15.46 | *** join/#asterisk ice_croft (n=nolan@213.132.86.246) |
| 08:17.28 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 08:19.12 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 08:19.40 | dominic1 | users with knowledge about realtime and hints here? |
| 08:22.15 | *** join/#asterisk ptblank (n=MURDER1@cpe-75-84-47-107.socal.res.rr.com) |
| 08:24.25 | *** join/#asterisk robeph (n=robf@76.73.206.120) |
| 08:24.46 | robeph | silly question, but how the heck do I quit the bloody cli console for asterisk without killing the service? |
| 08:25.05 | robeph | I couldv'e sworn it was "exit" but apparently vodka has muxxed my mind |
| 08:25.06 | hmmhesays | exit |
| 08:25.16 | hmmhesays | depends on how you start it to |
| 08:25.20 | dominic1 | is it possible to use my snom telephone as headset for my pc? |
| 08:25.21 | robeph | ah |
| 08:25.31 | dominic1 | if you start asterisk as service just exit |
| 08:25.33 | robeph | asterisk -c in this case , cos my init script is taking a dump on it |
| 08:25.38 | robeph | ah that's the problem then ;p |
| 08:25.43 | hmmhesays | whats wrong with your init script? |
| 08:25.48 | robeph | good question |
| 08:25.57 | hmmhesays | are you using the right one for your distro? |
| 08:26.06 | robeph | well, yeah |
| 08:26.08 | hmmhesays | i've never had any problems in debian, fedora or centos |
| 08:26.18 | hmmhesays | bash -X script |
| 08:26.19 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 08:26.21 | robeph | how can I quit the cli console from a -c start |
| 08:26.50 | robeph | i love the error handling... they just lie and say a command doesn't exist, rather than hey, you started it like this, do this to quit isntead heh, |
| 08:27.01 | robeph | i thought I was going crazy cos I've never just -c'd it |
| 08:27.12 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 08:27.16 | robeph | and exit is No such command 'exit' (type 'help' for help) |
| 08:27.26 | hmmhesays | yeah that is odd |
| 08:27.31 | robeph | no joke heh |
| 08:27.44 | robeph | I'd been at a bar and thought I must've totally lost my headpiece |
| 08:27.44 | hmmhesays | asterisk -c & |
| 08:27.45 | hmmhesays | asterisk -r |
| 08:27.46 | hmmhesays | lol |
| 08:27.56 | *** join/#asterisk eserra (i=nobody@89-96-52-24.ip10.fastwebnet.it) |
| 08:27.58 | robeph | yeh, but it won't lemme do anything to quit now :( |
| 08:28.15 | robeph | how can I get outta this without having to ssh like 3 servers deep to the box cos I killed my ssh session ;) |
| 08:28.23 | hmmhesays | ! |
| 08:28.30 | robeph | well hell |
| 08:28.33 | robeph | that was easy |
| 08:28.42 | hmmhesays | now fix your init script |
| 08:28.50 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 08:28.57 | robeph | root@installation:/etc/init.d# ./asterisk start * Starting Asterisk Software PBX.. /usr/sbin/safe_asterisk: 108: Syntax error: Bad fd number |
| 08:29.19 | robeph | it's erroring on "asterisk &" heh |
| 08:29.20 | hmmhesays | are you out of file descriptors? |
| 08:29.53 | robeph | uhm how can I tell =s I'm a wee bit stupid at the mo; prolyl shouldn't be doing this, but I want it done by the morning when I go in |
| 08:30.07 | robeph | never in my 10 years of linux have I run outta fds |
| 08:30.21 | SparFux | robeph: I get segfault on asterisk startup, but it seems to be due to "file not found error". |
| 08:30.35 | SparFux | robeh: "bad fd number sounds like it." |
| 08:30.48 | hmmhesays | are you running asterisk as root or another user? |
| 08:31.00 | robeph | root |
| 08:31.08 | SparFux | user asterisk. |
| 08:31.33 | robeph | well tbh, I'm running the init script as root |
| 08:31.41 | hmmhesays | yeah what distro? |
| 08:31.48 | robeph | ubuntu/deb |
| 08:32.26 | hmmhesays | cat /proc/sys/fs/file-max |
| 08:32.50 | SparFux | Me, debian, too. |
| 08:32.55 | robeph | 101696 |
| 08:33.10 | robeph | when I manually run asterisk & it's just at 25299 |
| 08:33.29 | robeph | it's prolly something goofy with some of this custom stuff we got ;) |
| 08:33.32 | SparFux | It says BAD fd, not OUT OF fd. |
| 08:33.41 | robeph | true |
| 08:33.44 | hmmhesays | ulimit -n |
| 08:33.54 | robeph | 1024 |
| 08:33.56 | hmmhesays | yeah it does, but I don't know what that means haha |
| 08:34.00 | robeph | me neither |
| 08:34.13 | robeph | maybe it wants something divisible by sin(5*21) |
| 08:34.20 | hmmhesays | ulimit -n 16384 |
| 08:34.34 | robeph | well magic |
| 08:34.45 | hmmhesays | hmm? |
| 08:34.56 | robeph | root@installation:/etc/init.d# ./asterisk restart * Restarting Asterisk Software PBX... [ OK ] |
| 08:35.01 | hmmhesays | there you go |
| 08:35.02 | robeph | thats wonky O.o |
| 08:35.06 | hmmhesays | it was running into the per user limit |
| 08:35.06 | robeph | wtf happened lol |
| 08:35.57 | robeph | thanks a ton, I'll have to make note of that so I can determine the problem and why that would occur, thank god its not a prod. machine and just an upgrade test box |
| 08:36.07 | *** join/#asterisk sergee (n=serg@195.94.224.197) |
| 08:36.22 | SparFux | Hm... my ulimit -n is 1024 too. |
| 08:36.33 | hmmhesays | SparFux: he may have more stuff running |
| 08:36.45 | hmmhesays | causing him to run out of file descriptors |
| 08:36.50 | robeph | but that much? |
| 08:36.55 | SparFux | ok. |
| 08:36.57 | hmmhesays | beats the hell out of me |
| 08:37.08 | SparFux | I have gnunet running. |
| 08:37.09 | hmmhesays | ubuntu is a bastard child of debian so maybe |
| 08:37.18 | robeph | ps aux , eg. show like but maybe 30-40 processes |
| 08:37.26 | robeph | yeh I'm no ubuntu natice |
| 08:37.28 | robeph | native* |
| 08:37.30 | robeph | gentoo <3 |
| 08:37.38 | SparFux | My misdn crashed. |
| 08:37.39 | hmmhesays | google, ubuntu why did I run out of file descriptors |
| 08:37.45 | SparFux | I will have to reboot. |
| 08:37.59 | hmmhesays | robeph: my paypal is hmmhesays at gmail dot com, I take donations |
| 08:38.02 | hmmhesays | :D |
| 08:38.42 | robeph | lol |
| 08:39.02 | robeph | yeh, I just started working here, so... i don't even get paid myself for a while :''( |
| 08:39.22 | robeph | I think I wonked up my "attempt" at upgrading rather than base installing our software heh |
| 08:39.45 | robeph | of course I did it all by hand without any instructions and just kinda guessed at how it is sposed to be done lol |
| 08:39.56 | hmmhesays | which software? |
| 08:40.01 | robeph | uhm |
| 08:40.06 | robeph | buncha custom stuff we built |
| 08:40.12 | hmmhesays | ah I see |
| 08:40.21 | robeph | dialers / ivrs / configurator stuff |
| 08:40.25 | hmmhesays | yeah |
| 08:40.37 | hmmhesays | dialers ugh, you damn telephone nazi |
| 08:40.40 | robeph | :'( |
| 08:41.00 | robeph | in my defense, WE don't use them |
| 08:41.08 | robeph | more a supply thing |
| 08:41.16 | hmmhesays | btw when you reboot that box you're going to have to set your FD again |
| 08:41.31 | hmmhesays | check out /etc/security/limits.conf |
| 08:41.41 | robeph | well i'd just have rebooted but I left the ubuntu disc in the drive and it'd hang at the choose boot sequence/install screen |
| 08:41.46 | robeph | yeh already noted that |
| 08:41.50 | robeph | that isn't the issue though, |
| 08:41.57 | robeph | I need to find out WHY its maxing out |
| 08:42.05 | robeph | cos it shouldn't need confing, it shouldn't hit a limit |
| 08:42.08 | remmo | cause it has power |
| 08:42.24 | robeph | yeh, but IT shouldn't do that ;p |
| 08:42.31 | robeph | it's my wonky upgrade i'm sure, |
| 08:42.33 | hmmhesays | yeah, but you don't want some dumb@$$ crashing your asterisk on because he decided he wanted to reboot while you're figuring it out |
| 08:42.47 | robeph | I'm magic at breaking things in ways no one has yet to do |
| 08:42.54 | robeph | ah nah this box is under my desk heh |
| 08:43.11 | robeph | it's just hooked up to some softphones for me to learn on for deployment etc |
| 08:43.19 | robeph | before I go borking up customer ware. |
| 08:43.30 | hmmhesays | ahh I see |
| 08:43.37 | hmmhesays | I don't have any vodka damnit |
| 08:43.40 | hmmhesays | i'm going to have some gin |
| 08:43.43 | robeph | this problem isn't on production systems, so I'm sure its something I wonked up |
| 08:43.45 | robeph | man...gin? |
| 08:43.51 | hmmhesays | its that or morgan |
| 08:43.52 | robeph | that's like....drinking a pine tree |
| 08:43.55 | robeph | go with morgan |
| 08:43.59 | robeph | rum is the stuff |
| 08:44.03 | hmmhesays | no no |
| 08:44.05 | robeph | eh |
| 08:44.06 | hmmhesays | gin to fall asleep |
| 08:44.09 | hmmhesays | stronger |
| 08:44.16 | robeph | gin tastes like sucking on a christmas tree |
| 08:44.17 | *** join/#asterisk leanshen (n=bob@ool-43546a71.dyn.optonline.net) |
| 08:44.24 | hmmhesays | not if you mix it with lime |
| 08:44.27 | hmmhesays | takes the pin out |
| 08:44.35 | robeph | either that or every sip makes me think of that woman from the pinesol commercials telling me I should use it to mop my floor |
| 08:44.38 | hmmhesays | wait.. maybe I have some vodka.... hmm |
| 08:44.55 | hmmhesays | or that slump buster you took home from the bar ... |
| 08:45.01 | hmmhesays | er.. I didn't say that |
| 08:45.02 | robeph | =s |
| 08:45.03 | *** part/#asterisk leanshen (n=bob@ool-43546a71.dyn.optonline.net) |
| 08:45.25 | robeph | I'm at home with 5 terminals open to various boxes.... no one accompanied me, if they did they'd be like, wow...you're geek |
| 08:45.28 | robeph | and leave. |
| 08:45.43 | robeph | even if they were the bottom barral oinklet |
| 08:45.45 | hmmhesays | yeah I have computers everywhere... next to my marshall stack |
| 08:46.01 | *** join/#asterisk SparFux (n=raoul@e182016014.adsl.alicedsl.de) |
| 08:46.05 | hmmhesays | the guitar makes the computers hot |
| 08:46.23 | robeph | http://www.maj.com/gallery/robeph/Misc/random_stuff_023.jpg |
| 08:46.32 | robeph | it heats the room |
| 08:46.37 | hmmhesays | i will click this |
| 08:46.47 | robeph | it's sfw |
| 08:46.48 | *** join/#asterisk tc3driver (n=huh@rrcs-24-199-16-118.west.biz.rr.com) |
| 08:46.49 | hmmhesays | damn |
| 08:46.55 | hmmhesays | nice rack dude |
| 08:46.59 | DarKnesS_WolF | anyone knows a good cheap A-Z termination " IAX2 " that accept normal visa no need for paypal or moneybooker? |
| 08:47.15 | hmmhesays | why do you need iax2? |
| 08:47.21 | hmmhesays | and what is wrong with paypal |
| 08:47.30 | DarKnesS_WolF | hmmhesays: much better than SIP cuz my * will be bbehind nat |
| 08:47.30 | hmmhesays | and why don't i have a drink in my hands, brb |
| 08:47.40 | DarKnesS_WolF | hmmhesays: egypt is banned from paypal |
| 08:47.46 | robeph | really DarKnesS_WolF ? |
| 08:47.52 | robeph | i figured they'd only ban nigeria |
| 08:47.57 | hmmhesays | vitelity does, and they handle sip behind nat pretty well |
| 08:48.00 | robeph | egypt has all the money |
| 08:48.08 | hmmhesays | I can hook my ip phones right up to them with no nat settings on the phones |
| 08:48.10 | DarKnesS_WolF | robeph: yes :( egypt banned from most alll money servcies :-s |
| 08:48.16 | SparFux | Unable to find a codec translation path from unknown to ulaw |
| 08:48.25 | DarKnesS_WolF | hmmhesays: they are cheap ? |
| 08:48.28 | hmmhesays | you're still getting that SparFux |
| 08:48.29 | DarKnesS_WolF | hmmhesays: link please? |
| 08:48.33 | hmmhesays | DarKnesS_WolF: reasonable |
| 08:48.34 | robeph | DarKnesS_WolF: egypt is a money service... between them and the saudis they prolly own 70% of the world currency |
| 08:48.35 | SparFux | hm...hesays: yes! |
| 08:48.36 | hmmhesays | www.vitelity.net |
| 08:48.48 | DarKnesS_WolF | SparFux: are u using G729 ? |
| 08:48.52 | SparFux | hmmhesays: And then segfault. |
| 08:48.57 | hmmhesays | sometimes the termination is a little wonky, but its an itsp what do you expect |
| 08:49.09 | hmmhesays | SparFux: why haven't you submitted a backtrace to the forums yet? |
| 08:49.12 | SparFux | darkness_wolf: it's a capi line calling asterisk. |
| 08:49.13 | ice_croft | hi all |
| 08:49.13 | DarKnesS_WolF | hmmhesays: voipjet sounds nice |
| 08:49.17 | DarKnesS_WolF | and euroiax |
| 08:49.19 | ice_croft | check this out: http://touchmods.blog.com/2399964/ |
| 08:49.27 | hmmhesays | DarKnesS_WolF: I thought voipjet only took paypal |
| 08:49.35 | SparFux | hmmhesays: you mean the output of asterisk -gvvvvvr ? |
| 08:49.36 | DarKnesS_WolF | also voicetradining but all paypal or money booker excpet for euroiax it band egypt |
| 08:50.39 | hmmhesays | SparFux: that and a backtrace |
| 08:50.48 | robeph | DarKnesS_WolF: why is egypt banned anyhow? |
| 08:51.08 | hmmhesays | SparFux: obviously have you a core dump wherever you started asterisk from right? |
| 08:51.34 | *** join/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 08:51.51 | SparFux | hmmmhesays: I should have, it segfaults. |
| 08:51.55 | hmmhesays | find it |
| 08:52.01 | SparFux | ok. |
| 08:52.06 | hmmhesays | should be named core.xxxx |
| 08:52.08 | hmmhesays | something like that |
| 08:52.31 | robeph | we were working on this box for a customer, and they'd done something odd with their ivr recordings, their voicemail was using a .sln for the goodbye message and it was all ridiculously encoded at some wonky rate and it sounded like something from the exorcist and they called up and were like... listen to the voicemail goodbye when you hit #...it's creepy, wee need to change this because it's really scaring us |
| 08:52.55 | hmmhesays | you should record all your prompts in something not lossy |
| 08:52.59 | hmmhesays | and convert from there |
| 08:53.05 | robeph | we didn't do it |
| 08:53.06 | robeph | they did it |
| 08:53.17 | hmmhesays | they are not thinking straight then |
| 08:53.19 | DarKnesS_WolF | robeph: i think foude |
| 08:53.30 | robeph | I have no idea what foude is |
| 08:54.34 | robeph | hmmhesays: I dunno what they did, they also had stuck their custom ivr messages in mp3 format and it was like the lowest volume ever |
| 08:54.53 | robeph | fixed that by unloading the mp3deoder, and itj ust fellback on gsm and sounds fine now |
| 08:54.59 | robeph | I dunno wtf they're doing |
| 08:55.04 | robeph | niether do they i don't think |
| 08:55.12 | hmmhesays | SparFux: instead of my running you through all this look at your source. ~/doc/backtrace.txt |
| 08:55.23 | hmmhesays | that will tell you exactly how to obtain the information you need to get your problem fixed |
| 08:55.33 | SparFux | hmmhesays: ok. |
| 08:55.46 | *** join/#asterisk sergey (n=sergey@91.189.233.71) |
| 08:56.00 | hmmhesays | you will probably have to rebuild without optomizations and with valgrind |
| 08:56.24 | robeph | well I'm off to bed now, i get about 6 hrs of sleep + hangover for tomorrow...early christmas I call it |
| 08:56.43 | hmmhesays | yeah i'm going to go make myself a gin + something now and watch earth final conflict |
| 08:56.51 | robeph | thanks for the help on that hmmhesays I'd have been mad issues trying to figure that lot out.... |
| 08:57.10 | hmmhesays | np, I'm going to post your problem on my wiki |
| 08:58.55 | *** join/#asterisk RoyK (n=roy@fw.fortel.no) |
| 09:04.47 | ZX81 | I've been checking this problem I have with a customers machine and it doesn't register to two of our three exchanges. If I do a tcpdump at both ends I get the packets leaving the machine but only arriving at one machine and yet if I do an IAX ping all machines receive packets - any ideas? |
| 09:05.25 | ZX81 | the iax ping is basically a packet on port 4569 using udp (perl script off the wiki) |
| 09:05.42 | ZX81 | I've restarted all Astricies :) |
| 09:10.38 | mosty | ZX81, pastebin your iax.conf |
| 09:10.41 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 09:11.47 | ZX81 | :) umm can type something up - client is iax.conf but servers are all realtime |
| 09:12.00 | ZX81 | same for every customer though and there are like 500 accounts |
| 09:12.13 | SparFux | Ok, I have the backtrace! |
| 09:12.17 | ZX81 | but will pastebin client side - maybe i've done something weird |
| 09:12.23 | ZX81 | Spar for? |
| 09:12.27 | *** join/#asterisk ronr (n=ron@ip51cdd509.speed.planet.nl) |
| 09:12.29 | SparFux | ZX81: for my coredump. |
| 09:12.38 | SparFux | Which forum should I send it to? |
| 09:14.33 | ZX81 | bugs.digium.com |
| 09:14.36 | ZX81 | what happens? |
| 09:14.49 | ZX81 | oh |
| 09:14.52 | ZX81 | and sparfux |
| 09:15.04 | ZX81 | you should compile with the DONT_OPTIMIZE flag |
| 09:15.17 | ZX81 | to get a proper backtrace on it with gdb |
| 09:15.29 | ZX81 | its in the make menuselect thingy in 1.4.x |
| 09:15.34 | *** join/#asterisk shtoom (n=godson@59.93.114.165) |
| 09:15.35 | ZX81 | or trunk |
| 09:15.42 | ZX81 | btw, |
| 09:15.47 | ZX81 | mosty: my pastebin is |
| 09:15.49 | ZX81 | http://pastebin.ca/813140 |
| 09:15.59 | ZX81 | not very exciting :) |
| 09:16.03 | ZX81 | auto generated |
| 09:16.20 | ZX81 | was working up till 7:30am this morning |
| 09:16.23 | ZX81 | :) |
| 09:18.17 | mosty | ZX81, are all the asterisk machines on static ip's? |
| 09:19.22 | ZX81 | yeah |
| 09:19.32 | ZX81 | city.venturevoip.com |
| 09:19.34 | *** join/#asterisk shtoom (n=godson@59.93.114.165) |
| 09:19.38 | ronr | the o'reilly asterisk book show a working sip.conf that is described as not pretty and not secure, however, it doesn't tell me what to do to get it secure (pretty would be nice, but not required), what do I need to do to make my sip.conf secure? |
| 09:19.46 | ZX81 | city being auckland christchurch or dunedin |
| 09:20.00 | ZX81 | ronr maybe read the wiki |
| 09:20.06 | ZX81 | ~voip-info |
| 09:20.06 | jbot | hmm... voip-info is the Voice Over IP wiki. It is a community resource which will answer all of your questions, from Asterisk to ZTDummy. You can find it over at http://www.voip-info.org - well worth bookmarking |
| 09:20.20 | mosty | ronr, see the bit about guest accounts on the sip.conf page on the wiki |
| 09:21.11 | mosty | ZX81, have you checked the routers at the sites that don't receive the iax packets? |
| 09:21.20 | ronr | thx, I'll be doing some reading |
| 09:23.04 | ZX81 | the thing is, an iaxping gets through |
| 09:23.10 | ZX81 | same port, same protocol |
| 09:23.12 | ZX81 | and |
| 09:23.25 | ZX81 | the same packets to auckland get replied to |
| 09:23.45 | ZX81 | meh |
| 09:23.49 | ZX81 | maybe router |
| 09:23.54 | ZX81 | also |
| 09:23.55 | ZX81 | :) |
| 09:23.59 | ZX81 | the ip changed |
| 09:24.01 | ZX81 | :) |
| 09:24.04 | ZX81 | of the client |
| 09:24.09 | ZX81 | in the middle of the night |
| 09:24.16 | ZX81 | so I thought the whole machine was gone |
| 09:24.18 | *** join/#asterisk clandmeter (n=Carlo@81.175.82.2) |
| 09:24.35 | ZX81 | but the isp had just set up dhcp with a changing ip |
| 09:24.38 | mosty | have you tried using ip addresses instead of dns names? |
| 09:24.42 | ZX81 | yeah |
| 09:24.53 | ZX81 | and tried enabling and disabling dnsmanager |
| 09:25.08 | ZX81 | can't remotely restart the router |
| 09:25.19 | ZX81 | and the linux box isn't storing arp routes |
| 09:25.36 | ZX81 | the only difference with the iaxping |
| 09:25.43 | ZX81 | is that it uses a random source port |
| 09:25.49 | ZX81 | instead of 4569 |
| 09:26.06 | ZX81 | problem is, the customer has got some iax phones |
| 09:26.13 | ZX81 | so I can't change the server port |
| 09:26.17 | ZX81 | oh maybe I could |
| 09:26.22 | mosty | if a packet trace at the source shows the packets going out correctly, then the problem must be somewhere in the middle |
| 09:26.32 | ZX81 | yeah traces work fine |
| 09:26.41 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-c634d4b16a766e62) |
| 09:26.52 | ZX81 | if I find out all the ips first of the phones and then login to them with elinks I could also change the server port |
| 09:26.54 | mosty | then the next thing to try is tests at the boundaries of the networks, ie the routers |
| 09:27.18 | ZX81 | yeah the router is looking suspicios but changing the source port might work around it |
| 09:27.20 | mosty | and if that all looks ok, then something in the middle is wrong |
| 09:27.25 | ZX81 | yeah |
| 09:27.27 | ZX81 | sec brb :) |
| 09:28.18 | ZX81 | HAH!!!!!!!! |
| 09:28.19 | ZX81 | YES |
| 09:28.23 | ZX81 | ROCKIN! |
| 09:28.24 | ZX81 | heh |
| 09:28.30 | ZX81 | I changed the bindport |
| 09:28.32 | ZX81 | in iax.conf |
| 09:28.35 | ZX81 | to 8888 |
| 09:28.40 | ZX81 | and all the regs are up |
| 09:28.49 | ZX81 | so I just need to change the iax phones :) |
| 09:29.03 | ZX81 | something is screwed on the router but this'll sort it for the moment |
| 09:29.19 | ZX81 | pity I can't set the source port per account :) |
| 09:31.49 | ZX81 | ah and finally the mrtg stats are looking happy again :) |
| 09:37.43 | *** join/#asterisk Nugget (i=nugget@carrera.macnugget.org) |
| 09:40.29 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 09:44.06 | *** join/#asterisk joelsolanki (i=joelsola@202.160.161.94) |
| 09:44.16 | joelsolanki | good morning all |
| 09:44.51 | joelsolanki | exten => _011442X.,1,Dial(SIP/${EXTEN:3}@voipinviteuk,90) |
| 09:45.08 | *** join/#asterisk harpal (n=Harpal@124.125.255.24) |
| 09:45.19 | joelsolanki | this i have configured for removin 3 digital means removing 011 |
| 09:45.41 | joelsolanki | it is digit stripping but what if i want to add 3 digits |
| 09:45.46 | joelsolanki | what is the parameter ? |
| 09:46.00 | joelsolanki | i want to add 222 prefix before all number dialed |
| 09:46.42 | kaldemar | just put 222 in there. 222${EXTEN} |
| 09:48.50 | joelsolanki | kaldemar: thanks. let me try that |
| 09:50.22 | *** join/#asterisk Psychobilly (n=Fuzz@online1.ioa.forthnet.gr) |
| 09:51.09 | Psychobilly | hello, anyone knows any softphone (for linux if possible) that supports mgcp? |
| 09:51.47 | *** join/#asterisk af_ (n=getsmart@88-149-241-31.dynamic.ngi.it) |
| 09:55.06 | *** join/#asterisk _ys (n=yuri@236-069.nat.mns.ru) |
| 09:55.45 | *** join/#asterisk vale-ICS (n=vale@icsnet.demon.co.uk) |
| 10:00.12 | cy3o3 | "We are sorry to report that we will not be able to obtain the following item(s) from your order: Jim Van Meggelen (Author), et al "Asterisk Cookbook" http://www.amazon.com/gp/product/059652692X" wtf I ordered that like 9 months ago or some shit :( |
| 10:00.31 | *** join/#asterisk Shaun2222 (n=Shaun222@ip68-4-127-67.oc.oc.cox.net) |
| 10:01.21 | Shaun2222 | Any issues or reasons not to run asterisk on a 64bit linux install with a Sangoma PRI T1 Card? |
| 10:03.15 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 10:07.36 | ronr | I added a secret to my sip.conf, but now my polycom phones no longer work, the logs say the user can't be authenticated, however, the username shown is what I have set as Display Name, not as the UserId? Is this a known problem with the polycom (IP 430) or did I do somehting wrong? |
| 10:13.40 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 10:13.47 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 10:13.50 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 10:14.09 | dominic1 | is it possible to check how many channels on my pri are available? |
| 10:14.23 | dominic1 | available, not in use |
| 10:15.37 | *** join/#asterisk Oerd (n=Oerd@ip-90-187-152-153.web.vodafone.de) |
| 10:15.47 | Oerd | Hi |
| 10:16.20 | mosty | dominic1, you might be able to subtract the number of pri channels in 'show channels' from the number of lines you have |
| 10:19.04 | *** join/#asterisk GerjanT (n=gerjan@frontgate.watchthe.net) |
| 10:19.22 | Oerd | i'm trying to have a snom360 be able to get the mailbox with the retrieve button on the phone. Is it possible to configure asterisk that the user doesnt have to put in his Boxnumber+secret, but only has to put in the secret? |
| 10:19.24 | *** part/#asterisk GerjanT (n=gerjan@frontgate.watchthe.net) |
| 10:33.37 | mosty | set a channel variable in sip.conf for that user |
| 10:34.03 | mosty | call it accountcode or something, then use that variable in the call to VoicemailMain |
| 10:42.30 | *** join/#asterisk linagee (n=linagee@about/linux/staff/linagee) |
| 10:42.59 | *** join/#asterisk MooingLemur (n=troy@unaffiliated/mooinglemur) |
| 10:44.22 | *** part/#asterisk RoyK (n=roy@fw.fortel.no) |
| 10:50.43 | *** join/#asterisk jmesquita (n=jmesquit@201.7.117.114) |
| 10:52.39 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-51612644075c058d) |
| 10:53.27 | *** join/#asterisk alejandro (n=asanchez@kde/developer/alejandro) |
| 10:54.04 | alejandro | someone wants to participate in the next Umeet (celebrated as always in IRC) talking about Asterisk or a success case with Asterisk ? |
| 10:54.11 | alejandro | the event is the next week |
| 10:55.13 | *** join/#asterisk marklar (n=marklar@unaffiliated/marklar) |
| 10:55.53 | marklar | how would I go about dialling a group of numbers one by one, checking for dead ones? |
| 10:59.33 | *** join/#asterisk dijungal (n=kdaniel@69.73.217.203) |
| 11:00.21 | mosty | what do you mean by dead? |
| 11:01.22 | dominic1 | thank you mosty |
| 11:01.47 | marklar | disconnected numbers, for example |
| 11:02.38 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 11:03.39 | mosty | as far as i know, you can't distinguish between disconnected numbers and congested numbers |
| 11:03.53 | mosty | without listening to the message played by the provider |
| 11:04.43 | *** join/#asterisk aboyousif (n=aboyousi@drupal.org/user/90522/view) |
| 11:06.23 | *** join/#asterisk d00gster (n=doughant@bas1-toronto12-1128684591.dsl.bell.ca) |
| 11:07.06 | Shaun2222 | anybody using a sangoma card here? |
| 11:09.29 | mosty | yes |
| 11:10.04 | Shaun2222 | what OS you runnin? |
| 11:12.51 | mosty | debian |
| 11:13.04 | Shaun2222 | did you have to build a driver against the kernel? |
| 11:13.25 | mosty | yes |
| 11:13.36 | Shaun2222 | where did you get the driver? |
| 11:13.52 | mosty | see the instructions on the sangoma wiki site |
| 11:13.58 | Shaun2222 | does zaptel source have it or did you download the wanpipe tarball and run there ./Setup install bs.. |
| 11:14.21 | mosty | their Setup script builds a .deb package |
| 11:15.00 | Shaun2222 | ya, same thing just apt knows about it now... |
| 11:15.12 | mosty | what problem are you having with wanpipe? |
| 11:15.16 | *** join/#asterisk _ys (i=yuri@91.151.196.254) |
| 11:15.48 | Shaun2222 | nothing, just trying to find out if the driver was in the vanila kernel or not yet and if so what ver |
| 11:16.25 | *** join/#asterisk nirz (n=nir@89.1.160.121.dynamic.barak-online.net) |
| 11:17.49 | mosty | i'm not sure it ever will be, depending on the licence of the firmware it uses |
| 11:17.55 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 11:28.56 | *** join/#asterisk stimpie (n=stimpie@84-104-5-115.cable.quicknet.nl) |
| 11:32.47 | dominic1 | is it possible to use G.722 via isdn? |
| 11:32.57 | dominic1 | or in europe only G.721? |
| 11:34.41 | mosty | the regular telephone network uses g711 |
| 11:34.58 | mosty | for isdn lines, i believe. ulaw in usa and alaw most everywhere else |
| 11:36.19 | dominic1 | okay, thank you very much, just read, that some isdn provider uses G722 too |
| 11:36.26 | dominic1 | but I think only in france... |
| 11:38.29 | tzafrir | Shaun2222, the wanpipe driver in the vanila kernel is obsolere and should not be used for any recent Sangoma cards |
| 11:39.03 | mosty | tzafrir, do you know if the wanpipe Setup script patches the kernel source you point it at? |
| 11:39.24 | tzafrir | mosty, run it as non-root :-) |
| 11:39.37 | tzafrir | IIRC it used to |
| 11:39.49 | tzafrir | I think later versions don't. But I'm not sure |
| 11:39.54 | mosty | hmm ok |
| 11:41.01 | mosty | i'm trying to build a kernel deb that doesn't conflict with the wanpipe deb |
| 11:41.23 | mosty | i'd like it to be as close as possible to the standard debian package |
| 11:43.28 | *** join/#asterisk morex (n=m@91.84.56.12) |
| 11:43.30 | morex | Hi there |
| 11:43.54 | morex | We've had a bug with SIP reinvite, and our supplier has suggested a patch which has fixed it |
| 11:44.09 | morex | I'm thinking of posting a bugreport with the patch to bugs.digium.com |
| 11:44.25 | morex | but neither of us are sure whether the patch is fully compliant with the SIP protocol |
| 11:44.33 | morex | Am I gonna get like totally flamed? |
| 11:45.21 | tzafrir | mosty, what's the conflict with the wanpipe deb? |
| 11:45.47 | stimpie | morex, not likely unless you code the patch in VB |
| 11:45.50 | mosty | tzafrir, the wanpipe deb wants to overwrite the wanrouter module that's in the debian kernel packages |
| 11:46.11 | morex | loud even :-) |
| 11:46.17 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 11:46.21 | morex | OK I'll get the thing posted then |
| 11:48.19 | *** join/#asterisk ice_croft (n=nolan@213.132.86.246) |
| 11:52.59 | *** join/#asterisk harpal (n=Harpal@124.125.255.24) |
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| 12:02.49 | *** join/#asterisk SagaZ- (n=danilo@unaffiliated/dbaio) |
| 12:05.57 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 12:07.42 | *** join/#asterisk Dovid (n=Dovid@bzq-79-180-45-64.red.bezeqint.net) |
| 12:14.05 | Dovid | hello ev1 |
| 12:15.51 | *** join/#asterisk ReDLe (n=rdl@200.138.29.170) |
| 12:17.38 | Dovid | anyone know of issues with ACPI on dell box's |
| 12:18.26 | mosty | yes. ztdummy doesn't like it |
| 12:19.33 | tzafrir | Dovid, which specific Dell box? |
| 12:20.20 | Dovid | I dont know. just started working on a box and my coder wants me to turn it off but I want to know what it is before I do |
| 12:20.38 | Dovid | I want to make sure that if I do turn it off I wont have other issues |
| 12:21.08 | Dovid | I am using CentOS |
| 12:21.22 | mosty | acpi is power management stuff. without it you may need to press the power button after shutdown to power down completely |
| 12:22.11 | Dovid | mosty: So I may not be able to reboot it remotely ? |
| 12:23.02 | mosty | you will need to test it, but reboot would probably work |
| 12:23.40 | bobkare | reboot shouldn't be a problem, only a complete shutdown |
| 12:23.44 | Dovid | mostry: may I PM ? |
| 12:24.02 | Dovid | bobkkare: so for a compete shutdown I would need to hit the button on the box ? |
| 12:24.13 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 12:24.13 | bobkare | yes, to actually power it off |
| 12:24.36 | mosty | dovid: i prefer to talk here unless there's a special reason not to |
| 12:24.43 | Dovid | ah ok. I woubt I would wana do that on a live box. |
| 12:24.54 | Dovid | mosty: didn't wana flood the room wit all my stupid questions ;) |
| 12:24.54 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 12:25.06 | bobkare | yeah, not usually something you want to do with a server |
| 12:25.16 | Dovid | my concern is just making problems on the box since I dont know it well |
| 12:25.26 | Dovid | how ever it seems to be making problems with ztdummy |
| 12:25.41 | Dovid | is it something that people do if they have ztdummy issues ? |
| 12:25.49 | mosty | dovid: you wil have to choose between ztdummy and acpi |
| 12:27.45 | Dovid | mosty: from my google searches it seems to control the fans in the PC etc. |
| 12:27.50 | Dovid | i dont want to burn out the box |
| 12:28.25 | mosty | Dovid, it's only used to slow the fans down when they don't need to be at full blast |
| 12:28.46 | Dovid | ah ok. now is there anything in the BIOS that should be changed as well ? |
| 12:28.49 | mosty | no |
| 12:29.32 | Dovid | mosty: SO to sum it up, its a package that should be enabled on the machine unless you have core preformance issues |
| 12:32.00 | mosty | i use it on every machine unless it conflicts with something |
| 12:32.55 | mosty | on my machines i tend to scrap ztdummy rather than acpi |
| 12:33.41 | Dovid | mosty: can you have a look at this ? |
| 12:33.42 | Dovid | http://pastebin.ca/813281 |
| 12:33.53 | Dovid | i am in Israel now and the box is in ireland :( |
| 12:34.00 | Dovid | any other hardware that I can use for timing ? |
| 12:34.16 | Dovid | other than zap hardware (hard to get it here) |
| 12:35.06 | mosty | not really |
| 12:35.27 | Dovid | eh. ok |
| 12:35.38 | Dovid | gona be ireland next week. gona make the client pick up a card |
| 12:35.40 | mosty | you don't need a zaptel timer if you don't use meetme, iax trunking, and a few other commands |
| 12:35.49 | Dovid | using meetme ;) |
| 12:36.00 | Dovid | also there is an issue with the voicemails |
| 12:36.38 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 12:47.33 | mosty | tzafrir, btw wanpipe does modify the kernel source |
| 12:47.57 | mosty | or the headers at least, i guess |
| 12:48.01 | *** join/#asterisk znndrp (n=alex@qwoot.net) |
| 12:48.57 | *** join/#asterisk ming_zym (n=ming_zym@124.14.234.65) |
| 12:49.10 | *** join/#asterisk cbx33 (n=pete@ubuntu/member/cbx33) |
| 12:49.14 | cbx33 | hi all |
| 12:49.19 | cbx33 | got a question about asterisk |
| 12:50.20 | cbx33 | at a particular site, we have a few offices scattered around linked with a wireless bridge. The phone lines in these other buildings are on completely different systems, is it possible to use asterisk in the main building to provide a sip line and link an extension at the main building to that?? |
| 12:51.11 | tzafrir | mosty, oops. Problems, I guess |
| 12:51.15 | mosty | cbx33, yes |
| 12:51.32 | cbx33 | mosty is this going to be the best way to do this? |
| 12:51.44 | mosty | tzafrir, the wanpipe build script is messy, this is just one more messy bit :/ oh well |
| 12:52.24 | mosty | cbx33, it's hard to tell, your description is not very clear |
| 12:52.36 | cbx33 | sorry.....I'll try to rephrase |
| 12:53.33 | cbx33 | At the moment we have a number of lines come in, and I believe there must already be a pbx of some sort to do call transferring etc. There are other workers in other buildings that would like to be connected to this system, would like their phone number to be the same as the main office. So we could transfer calls to them |
| 12:53.45 | cbx33 | We have a wireless bridge to them |
| 12:54.00 | cbx33 | I was wondering if asterisk box could be plugged into a spare extension |
| 12:54.06 | *** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 12:54.20 | cbx33 | and then the "remote" workers could connect to asterisk and it would backend the call onto the extension? |
| 12:54.34 | mosty | something like that would work |
| 12:54.51 | mosty | assuming the wireless connection is good enough, not too loaded etc |
| 12:54.55 | cbx33 | what hardware would I need for the asterisk box? |
| 12:54.58 | cbx33 | yeh it's not too loaded |
| 12:55.06 | cbx33 | to connect it to the extension |
| 12:55.18 | mosty | depends what the existing pbx has |
| 12:55.26 | cbx33 | hmm |
| 12:55.34 | cbx33 | how can I find out? |
| 12:55.48 | mosty | look in the manual for the existing pbx? |
| 12:55.53 | cbx33 | hehehe |
| 12:55.54 | cbx33 | ko |
| 12:56.09 | znndrp | Hi, I'm trying to call multiple phones from a callfile. I tried 'SIP/21&SIP/22' in the channel variable but that doesn't work :/ |
| 12:56.12 | znndrp | any clues? |
| 12:56.44 | mosty | znndrp, if that doesn't work, maybe you can call chan_local, and setup the extension to call multiple sip clients |
| 12:57.29 | znndrp | what's chan_local? |
| 12:57.43 | mosty | look it up on the wiki |
| 12:58.13 | mosty | you can dial a n extension/context as if it were a channel |
| 12:58.22 | znndrp | oh, cool |
| 12:58.24 | znndrp | thnx |
| 12:58.31 | znndrp | gonn try that |
| 12:58.34 | znndrp | +a |
| 12:58.56 | *** join/#asterisk J4zen (n=Jvan4zen@a213-84-139-244.adsl.xs4all.nl) |
| 12:59.42 | *** join/#asterisk lsodi (n=lsodi@195.80.124.193) |
| 13:02.13 | lsodi | greetings, I would like to hire administrator for asterisk, whos job is to keep asterisk alive and up to date, asterisk has 200 sip users, what is sensible sum per month for that job? |
| 13:03.17 | riddlebox | lsodi, where is the company? |
| 13:03.30 | lsodi | in estonia |
| 13:04.33 | J4zen | no clue, depends how many hours you'd want him to be available |
| 13:05.51 | riddlebox | yeah would he be an onsite guy, or remote? |
| 13:06.08 | tzafrir | J4zen, for a momen I parsed what you wrote as: s/him/it/ :-) |
| 13:07.04 | J4zen | i'd say thats a : #PARSE ERROR ON LINE 14:06 |
| 13:07.20 | J4zen | ;) |
| 13:07.25 | lsodi | remote and max downtime 3h |
| 13:07.55 | riddlebox | lsodi, 3h a day? |
| 13:07.57 | J4zen | max downtime of three hours, that means he has to be on call 24/7 and at the job site within the hour? |
| 13:08.05 | J4zen | that's going to cost you a fortune i believe |
| 13:10.21 | *** join/#asterisk myiagy (n=Jose@200.215.59.133) |
| 13:12.19 | riddlebox | crap, I took a sick day today and there is nothing on tv |
| 13:12.29 | riddlebox | guess I will check out the mythbox and see what I have recorded |
| 13:12.42 | stimpie | riddlebox, thats why they invented usenet |
| 13:12.45 | lsodi | 100EUR per month for administration and 35EUR prer/h on fixing/upgrading asterisk? |
| 13:13.12 | riddlebox | stimpie, lol |
| 13:13.48 | znndrp | mosty: works like charm now ;) |
| 13:14.10 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 13:14.39 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.125.196) |
| 13:15.37 | J4zen | lsodi: 35E per hour seems a bit low for such delivery times |
| 13:15.57 | J4zen | 3 hours max downtime |
| 13:16.00 | J4zen | i'd say 45 |
| 13:16.14 | J4zen | perhaps 40 during business hours |
| 13:16.17 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 13:16.29 | dominic1 | I always get this error: [Dec 12 16:19:51] WARNING[4409]: chan_sip.c:3182 update_call_counter: Inringing for peer '100' < 0? |
| 13:16.29 | dominic1 | <PROTECTED> |
| 13:16.32 | dominic1 | what does it mean? |
| 13:16.33 | J4zen | and 50~60 inside of business hours |
| 13:16.42 | J4zen | Thats no error |
| 13:16.45 | J4zen | Thats a warning :) |
| 13:17.06 | dominic1 | What does the warning mean? |
| 13:17.10 | dominic1 | :-D |
| 13:17.16 | J4zen | hehe |
| 13:17.24 | stimpie | lsodi, I agree with j4zen |
| 13:17.27 | J4zen | no clue to be honost, not using ODBC controls for anything |
| 13:17.52 | J4zen | 50~60 outside of business hours* my bad. |
| 13:18.19 | stimpie | the 24/7 availibility is the 'problem' |
| 13:18.49 | J4zen | i work for a datacenter; Customers with SLA's already pay 30E per hour in business hours and 40E per hour outside.. and thats just regular System administration support |
| 13:19.03 | J4zen | Asterisk -administrators are generally more expensive |
| 13:19.06 | J4zen | as there are less of them |
| 13:19.59 | *** join/#asterisk mltlnx (n=mltlnx@64.3.170.41.ptr.us.xo.net) |
| 13:25.51 | lsodi | ok thank you all |
| 13:31.12 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 13:42.54 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkgroup.com.au) |
| 13:43.17 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 13:50.57 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 13:54.17 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:57.17 | *** join/#asterisk Buhntz (i=Boones@port-212-202-42-25.dynamic.qsc.de) |
| 13:59.47 | *** part/#asterisk myiagy (n=Jose@200.215.59.133) |
| 14:01.54 | *** join/#asterisk PepOSX (n=pepOSX@190.79.246.105) |
| 14:07.01 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:18.54 | *** join/#asterisk ussrback (n=MAX@80.241.177.19) |
| 14:18.59 | ussrback | hi |
| 14:19.21 | riddlebox | hi |
| 14:19.34 | ussrback | is it possible to stream music from shoutcast in meetme conference room background? |
| 14:22.54 | [TK]D-Fender | ussrback: Yes, there is clearly a way, go sit and think about it for a little bit |
| 14:24.04 | ussrback | [TK]D-Fender: AMI ? |
| 14:24.29 | [TK]D-Fender | ussrback: keep thinking. Ask youself how audio gets into meetme. |
| 14:25.21 | ussrback | [TK]D-Fender: MOH |
| 14:25.34 | ussrback | but when other user enters its stopped |
| 14:26.06 | ussrback | other way is to write some SIP linux console client, connect it to chatroom and stream |
| 14:26.29 | ussrback | but its a long story |
| 14:26.41 | ussrback | may be u know better way for this? |
| 14:27.47 | [TK]D-Fender | ussrback: So have MoH JOIN the conference |
| 14:28.29 | ussrback | [TK]D-Fender: moh join???????? |
| 14:28.43 | cpm | yup |
| 14:28.47 | [TK]D-Fender | ussrback: This is so remarkably easy its pathetic..... |
| 14:29.18 | ussrback | ok but let say i have 2 rooms 1. rock 1 pop 1. rap |
| 14:29.31 | ussrback | and i want to stream different shoutcast channels there |
| 14:29.44 | ussrback | rock music in rock conference room |
| 14:29.47 | ussrback | and so on |
| 14:29.50 | [TK]D-Fender | ussrback: exten => 12345,1,Musiconhold(sectionwithstreaming) <- call this, then call into the conference and TRANSER THE CALL |
| 14:29.56 | J4zen | Is anyone using A2Billing? |
| 14:30.52 | [TK]D-Fender | J4zen: No, its a dead project that NOBODY uses. |
| 14:31.18 | J4zen | [TK]D-Fender: What do people use? |
| 14:31.40 | [TK]D-Fender | J4zen: Yes people use it, no this isn't a support channel for it. |
| 14:32.06 | J4zen | [TK]D-Fender: Nowhere did i ask for support, i was merely wondering if its a common method of billing your customers. |
| 14:32.31 | J4zen | [TK]D-Fender: Obviously, No it isn't. Hence my second question; What is the most commonly used software used to bill customers? |
| 14:32.49 | [TK]D-Fender | J4zen: It is fairly popular. Requires a lot of dialplan hacking. Makes a real mess, but it ""works" more or less |
| 14:33.10 | J4zen | Any altnernatives? |
| 14:33.15 | J4zen | i like to keep things tidy |
| 14:33.21 | ussrback | ok 10xs |
| 14:34.07 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 14:34.39 | [TK]D-Fender | J4zen: Go take a look at it. There aren't a lot of OSS billing platforms out there. See if you can make it manageable |
| 14:35.02 | J4zen | Will do, Thanks. |
| 14:38.07 | *** join/#asterisk fbnts (n=thomas@vds2-lon.vidicom.co.uk) |
| 14:38.29 | fbnts | hi, just a quick question, is the MeetMe application included in Asteridk 1.4? |
| 14:38.57 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:38.57 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:45.56 | *** part/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
| 14:45.56 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
| 14:46.56 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 14:47.13 | *** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-e25931f5febceee8) |
| 14:48.05 | *** join/#asterisk tc3driver (n=huh@rrcs-24-199-16-118.west.biz.rr.com) |
| 14:49.16 | *** join/#asterisk thedonvaughn (n=jayson@unaffiliated/printk) |
| 14:50.16 | [TK]D-Fender | fbnts: Yes, along with every other version since the start |
| 14:51.54 | Dr-Linux | anybody tried RHEL 5 with asterisk? |
| 14:52.22 | [TK]D-Fender | Dr-Linux: Plenty of us. |
| 14:52.35 | *** join/#asterisk cygar (n=cygar@200.26.191.3) |
| 14:53.07 | cygar | Hello |
| 14:53.17 | Dr-Linux | [TK]D-Fender: it's stable right? |
| 14:53.49 | [TK]D-Fender | Dr-Linux: Why do you think it carries the name Enterprise Linux....... |
| 14:54.40 | matmoj | not that i would say that using another distribution means its unstable |
| 14:54.44 | Dr-Linux | from the web "Red Hat Enterprise Linux 5, now available in Beta 2" |
| 14:54.46 | matmoj | its still using the linux kernel... |
| 14:55.26 | Juggie | Dr-Linux, RHEL5 is production |
| 14:55.27 | Juggie | not beta |
| 14:55.30 | Juggie | has been for months |
| 14:55.38 | matmoj | unless you are after the support, id say you might as well go for something else.. |
| 14:56.05 | Juggie | like, centos5. |
| 14:56.13 | matmoj | i prefer debian |
| 14:56.19 | matmoj | mostly for the package system i guess |
| 14:56.21 | [TK]D-Fender | Juggie: Which when you get right down to it... IS RHEL 5. |
| 14:56.26 | matmoj | also redhat (for me) feels so bloated... |
| 14:56.34 | Juggie | [TK]D-Fender, shhhh :) |
| 14:56.38 | matmoj | im administering a rhel4 and 5 ... |
| 14:56.55 | Dr-Linux | Juggie: thanks. actually we only use RHEL for our 12 asterisk servers, currently it's RHEL 4 |
| 14:56.57 | Juggie | matmoj, perhaps who ever did the install installed everything. |
| 14:57.01 | Dr-Linux | hhm.. |
| 14:57.07 | matmoj | dell preinstalled actually |
| 14:57.11 | Dr-Linux | lemme test it first |
| 14:57.14 | awk | can somebody tell me, does hyperthreading still affect quality with asterisk and should it still be turned off? |
| 14:57.15 | matmoj | i dont like x on my servers... |
| 14:57.19 | Juggie | Dr-Linux, great, i have about 6 all on centos4 |
| 14:57.25 | Juggie | (same os, no support) |
| 14:57.49 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 14:57.50 | Juggie | awk, it could only if the kernel was broken |
| 14:58.02 | Dr-Linux | Juggie: i had support but nomore, but i never need for it |
| 14:58.15 | *** join/#asterisk Dovid (n=Dovid@bzq-79-180-45-64.red.bezeqint.net) |
| 14:58.16 | Juggie | i think it had problems originally but if you are running a later kernel it should support HT ok |
| 14:58.29 | Dovid | hi, anyone have problems with ACPI and ztdummy ? |
| 14:58.30 | Juggie | but i guess it still could cause issues, i run HT on all my boxes though with no issue |
| 14:58.37 | Juggie | they are xeons w/ HT. |
| 14:58.47 | mosty | Dovid, do you have amnesia or something? |
| 14:58.56 | Dr-Linux | Juggie: actually i wanna swtich from asterisk 1.2 to 1.4.x as well |
| 14:59.00 | Dovid | mosty: i am starting allllllllllllllll over again |
| 14:59.29 | mosty | dovid: don't use ztdummy and acpi at the same time on dells. pick one. |
| 14:59.31 | Dovid | i was unable to stop it from loading in grub.conf no one in centos knew what the issue was so I wanted a fresh aproach |
| 14:59.42 | Dovid | mosty: just found out it is not a dell |
| 15:00.05 | Dovid | mosty: I guess this is more of a CentOS question (since that is what I use) but I cant seem to stop acpi |
| 15:00.10 | mosty | dovid: there is a kernel command line arg that will disable acpi. acpi=off or noacpi, i forget. ask google |
| 15:00.30 | Dovid | mosty: I tried both (one at a time) and that didnt worn |
| 15:00.40 | Dovid | in grub.conf acpi=no and noacpi |
| 15:01.00 | mosty | dovid: i recommend that you ask a centos channel then |
| 15:01.28 | Dovid | mosty: I am coming here after banging my head against the wall there ;) |
| 15:02.56 | mosty | i looked it up, it's acpi=off |
| 15:04.20 | [TK]D-Fender | Dovid: http://www.google.ca/search?hl=en&q=centos+grub+disable+acpi&btnG=Google+Search&meta= |
| 15:04.24 | [TK]D-Fender | Dovid: http://www.centos.org/docs/5/html/Virtualization-en-US/ch-config-grup.html |
| 15:04.26 | *** part/#asterisk iratik (n=itariki@12-226-116-3.client.mchsi.com) |
| 15:04.31 | [TK]D-Fender | Dovid: First friggen link. |
| 15:04.52 | [TK]D-Fender | Dovid: Google + 5 second search > you |
| 15:05.14 | [TK]D-Fender | Oh.. and I suck at linux. |
| 15:05.37 | [TK]D-Fender | *sigh* |
| 15:07.15 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 15:10.18 | Dovid | TK: I am horrible with google |
| 15:10.25 | *** join/#asterisk beek (n=klinebl@65.211.106.243) |
| 15:10.47 | Dovid | TK: I always try google b4 coming here. |
| 15:11.15 | J4zen | How often do you actually encounter hacks against SIP/VOIP/Asterisk, i've been reading about it and there are quite a lot of known and unpatched exploits available versus softphones/hardphones/asterisk/SIP in general. How much of a security risk is VOIP really? |
| 15:11.45 | J4zen | For example; I just read about a PoC where a hacker managed to have any recieving SIP phone automatically answer a call silently |
| 15:11.56 | J4zen | turning it into a bugged phone |
| 15:12.01 | J4zen | only.. without hardware required |
| 15:12.35 | J4zen | Are attacks on VOIP/SIP common? |
| 15:12.43 | *** join/#asterisk rcphq (n=rllibre@200.42.219.109) |
| 15:12.52 | *** join/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 15:12.52 | *** part/#asterisk rcphq (n=rllibre@200.42.219.109) |
| 15:13.14 | mosty | J4zen, that's a 'feature' of some sip phones |
| 15:13.21 | J4zen | I'm aware of that |
| 15:13.22 | mocker | J4zen: I don't think it's common yet, but I think it'll increase soon. |
| 15:13.27 | J4zen | But with the feature disabled |
| 15:13.44 | mocker | J4zen: Because most people don't even know that SIP is unencrypted.. |
| 15:13.46 | *** join/#asterisk cesar_CR (n=cesar@201.192.86.6) |
| 15:14.02 | mocker | So launching wireshark you can record the whole conversation. |
| 15:14.03 | J4zen | heh |
| 15:14.06 | J4zen | its basically like HTTP |
| 15:14.08 | lirakis | mocker: no |
| 15:14.15 | J4zen | the Protocl that is |
| 15:14.15 | lirakis | mocker: rtp is unencrypted |
| 15:14.20 | lirakis | mocker: sip is not media |
| 15:14.21 | *** join/#asterisk callguy (n=callguy@pool-71-162-97-18.bstnma.east.verizon.net) |
| 15:14.27 | mocker | lirakis: sematics. :) |
| 15:14.34 | lirakis | mocker: not really |
| 15:14.53 | nestAr | tomato tomato |
| 15:15.10 | mocker | lirakis: I think everyone here knew what I meant, but you're right. |
| 15:15.10 | nestAr | doesn't really work online, i guess |
| 15:15.20 | mocker | nestAr: heh. |
| 15:16.18 | lirakis | mocker: im sure they did.. but there is no need to further propegate what many people misunderstand... |
| 15:16.28 | lirakis | mocker: thats all |
| 15:16.48 | J4zen | heh, inthe end |
| 15:16.56 | J4zen | both the protocol and media are unencrypted |
| 15:17.06 | J4zen | leaving them amazingly vulnerable |
| 15:17.10 | mocker | J4zen: Indeed. |
| 15:17.39 | J4zen | Is there a modification available to encrypt RTP-traffic at least? |
| 15:17.47 | lirakis | J4zen: there are clients that do it |
| 15:17.48 | J4zen | or tokenize the protocol |
| 15:17.50 | mocker | It's going to be a blow PR-wise when someone says a CC # and it get's intercepted. |
| 15:17.51 | lirakis | J4zen: zphone |
| 15:18.03 | J4zen | i see |
| 15:18.11 | J4zen | but |
| 15:18.17 | lirakis | <PROTECTED> |
| 15:18.18 | lirakis | http://www.openser.org/docs/modules/ |
| 15:18.20 | J4zen | that would still allow it to be hijacked |
| 15:18.20 | lirakis | arg |
| 15:18.29 | J4zen | Imagine DNS-poisening in combination with VOIP |
| 15:18.30 | lirakis | ** http://zfoneproject.com/ |
| 15:18.45 | J4zen | lilalinux: Thanks |
| 15:19.09 | lirakis | <PROTECTED> |
| 15:19.14 | lirakis | well .. on a lan |
| 15:19.45 | J4zen | agreed |
| 15:20.25 | J4zen | So many companies aspiring in providing VOIP, so few providing security for VOIP |
| 15:20.39 | J4zen | *and a light flashes on* |
| 15:20.52 | mocker | I tend to tunnel everything through a VPN. |
| 15:21.00 | J4zen | yeah i was thinking of that |
| 15:21.13 | mocker | It's easy.. |
| 15:21.14 | J4zen | but in alarge enviroment |
| 15:21.15 | mocker | ;) |
| 15:21.22 | J4zen | i'd have to set up dedicated RAS servers |
| 15:21.24 | lirakis | mocker: you tunnel through a vpn to your carriers? ... |
| 15:21.30 | J4zen | which doesnt seem that viable |
| 15:21.39 | mocker | lirakis: PRI is my carrier. |
| 15:21.39 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 15:21.53 | J4zen | hehe for the carrier i hope he snot |
| 15:21.55 | J4zen | oh |
| 15:21.58 | lirakis | mocker: so you are talking internal only.. you have no voip carriers |
| 15:21.58 | J4zen | that works |
| 15:22.04 | mocker | lirakis: Jah. |
| 15:22.11 | mocker | I don't trust VoIP carriers for business stuff. |
| 15:22.17 | mocker | Because if they fuck up, I get fired. :P |
| 15:22.29 | lirakis | mocker: ookay |
| 15:22.33 | J4zen | you're still using a telcom that can fuck up .. |
| 15:22.43 | J4zen | just as likely as the voip carrier |
| 15:23.01 | J4zen | and it'll get you fired .. just as likely ;) |
| 15:23.25 | mocker | J4zen: I doubt that, using an ITSP for business grade stuff introduces lots of problems that using a dedicated PRI eliminates. |
| 15:23.33 | lirakis | not to mention.. that huge amounts of carrier traffic transits in voip anyway.. which is why TDM/VoIP conversion exists |
| 15:23.43 | lirakis | and .. vice verssa |
| 15:23.59 | mocker | lirakis: The difference is they control the network there. |
| 15:24.24 | J4zen | mocker: It also removes a lot of problems that carries bring, i suppose it depends on the situation |
| 15:24.33 | lirakis | from what i understand... verizon, for all its broadwing voip customers... actually convert and switch everything TDM b/c its the only "standard" that everything can convert to/from |
| 15:25.10 | lirakis | mocker: .. i think you just need none sh** carriers and some routing |
| 15:25.54 | lirakis | mexico is going off the hook now with gray routes since the rate change |
| 15:26.13 | J4zen | "gray routes" ? |
| 15:26.18 | mocker | lirakis: I trust old world telcoms PRI stuff for reliability much more than an ITSP that's been in business for 1-2 years. |
| 15:26.22 | mocker | But this is all personal preference. |
| 15:26.26 | mocker | And $$$. |
| 15:26.35 | mocker | My way isn't the cheapest route. |
| 15:27.00 | lirakis | mocker: it depends on your operation .. i dont know what kind of minutes you run. And it isnt about cheap... I have a pri failover incase some thing goes wrong with my carriers as well. |
| 15:27.37 | *** join/#asterisk BCS-Satori (n=somewher@75.148.21.113) |
| 15:27.45 | mocker | lirakis: Who do you use for VoIP? |
| 15:27.59 | fbnts | hi, just a quick question, is the MeetMe application included in Asteridk 1.4? |
| 15:28.04 | lirakis | <PROTECTED> |
| 15:28.49 | J4zen | Never heard of that |
| 15:28.51 | J4zen | Cool |
| 15:29.07 | J4zen | I'm off for today, have a great newyear(my holiday starts) and christmas! |
| 15:29.08 | J4zen | bye |
| 15:30.37 | lirakis | mocker: lots of carriers, but I work at a clec so I have lots of options for wholesale routing. |
| 15:30.43 | tzanger | hahaha |
| 15:30.45 | tzanger | "w00t" crowned word of year by U.S. dictionary by the merriam-webster |
| 15:31.22 | mocker | lirakis: Hah, you work at a CLEC, that's cheating. :) |
| 15:33.55 | *** join/#asterisk techie (n=techie@adsl-76-247-10-106.dsl.lsan03.sbcglobal.net) |
| 15:34.14 | BCS-Satori | Are there any adjustments I need to make to have a SIP Phone and a soft client registered to the same extension in asterisk. Example: the site has a phone which will be running 24 hours a day with a private extension. If the end user travels and launchs a software client, but wants the phone extension to stay the same, do i need to make an adjusts or use SLA, or can asterisk realize 2 registrations on same extensions |
| 15:34.48 | *** join/#asterisk moprilo (n=jjohn@190.10.0.84) |
| 15:35.02 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:35.02 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 15:36.37 | [TK]D-Fender | BCS-Satori: You cannot have 2 phones registered to the same SIP entry. Period. And stop calling SIP accounts/devices "EXTENSIONS". |
| 15:37.53 | lirakis | BCS-Satori: it will break .. or rather .. have very strange and unreliable behaviour. Each account will attempt to keep registering. One will get registered, then it will unregister.. etc. |
| 15:38.45 | BCS-Satori | lirakis: understand thank you. |
| 15:40.30 | *** join/#asterisk BugsiE (n=Stef@isfw.jhb.24-7online.co.za) |
| 15:41.39 | BugsiE | hi guys |
| 15:42.10 | BugsiE | just installed server with asterisk |
| 15:42.38 | BugsiE | how do i log into the box |
| 15:43.36 | mosty | BugsiE, are you talking about trixbox? |
| 15:43.38 | mort_gib | BugsiE: ssh root@yourasterisk :-) |
| 15:44.26 | BugsiE | is there a web interface ? |
| 15:45.39 | mosty | bugsie: not in the standard asterisk. how did you install linux? |
| 15:46.17 | *** join/#asterisk ManxPower (n=manxpowe@194.sub-75-201-101.myvzw.com) |
| 15:46.34 | *** join/#asterisk Victor_Yure (n=aaa@postfix.tradein.com.br) |
| 15:47.12 | BugsiE | like the website told me under support and so forth download, make install file.... |
| 15:47.28 | *** join/#asterisk rcphq (n=rllibre@200.42.219.109) |
| 15:47.52 | *** join/#asterisk pirulo (n=andres_p@65.19.28.123) |
| 15:51.14 | mosty | well you probably don't have a gui, you will have to edit config files |
| 15:51.16 | *** join/#asterisk jameswf (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 15:52.53 | javb | does anyone here have taken the Digium dCAP express certification mode (without the bootcamp, just going the last day of the bootcamp and paying $300) ? |
| 15:52.54 | [TK]D-Fender | BugsiE: What website? |
| 15:53.50 | mocker | javb: I did the bootcamp then test. |
| 15:54.18 | javb | mocker, can i msg you? |
| 15:55.20 | mocker | javb: Sure, but if it's asterisk related you might as well say it in channel. :) |
| 15:55.28 | mocker | After all, it's why everyone's here! |
| 15:58.20 | jameswf | When building asterisk via script is it best to patch to compile app_mp3 or is there like an --enable flag i can run on configure |
| 16:01.22 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 16:01.53 | *** join/#asterisk russellb (i=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 16:01.53 | *** mode/#asterisk [+o russellb] by ChanServ |
| 16:02.33 | *** join/#asterisk astCoderX (n=root@200.93.195.132) |
| 16:02.36 | mosty | you can save the menuselect options file |
| 16:02.48 | astCoderX | hi need help with asterisk. |
| 16:03.12 | astCoderX | i want to know if a channel is avaliable for make an outgoing call |
| 16:03.41 | [TK]D-Fender | astCoderX: Try, and lok at the dialstatus when it fails. |
| 16:04.08 | astCoderX | for example if my trunk can handle 30 concurrent outgoing calls .. i want to know if there's is chance to make the 31th call |
| 16:04.13 | *** join/#asterisk d-k-t (n=dt@125.120.133.104) |
| 16:04.45 | mosty | astCoderX, what kind of trunk? |
| 16:04.45 | [TK]D-Fender | astCoderX: You won't be able to get a count easily. You'd have to write an AGI or something that would parse the output of something like "show channels concise" |
| 16:05.46 | astCoderX | D-Fender: i've been reading about dialstatus variable. Which status i must to evaluate to know if the channel is avalaible? BUSY or CONGESTION? |
| 16:06.30 | astCoderX | i'm programming with perl |
| 16:06.42 | *** join/#asterisk MindTheGap (n=MindTheG@201.80.194.113) |
| 16:07.28 | ManxPower | astCoderX: you don't know of the channel is available or not from DIALstatus, you can only know if a call using that line worked. If it's analog, that status will be ANSWERED, for all others it will be ANSWERED or BUSY. Check "show application dial" to make sure there are not any others you would want to check for. |
| 16:07.37 | [TK]D-Fender | astCoderX: First, what is this "trunk" you're dialing out of? |
| 16:07.43 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 16:07.50 | ManxPower | ~siptrunk |
| 16:07.50 | jbot | from memory, siptrunk is Asterisk does not support SIP Trunks. Set trunk=no in sip.conf and then set up the device normally in sip.conf. |
| 16:08.46 | ManxPower | astCoderX: you just do the Dial and check the result. If you want to do it the way a girlyman does it, you could also use ChanIsAvail. |
| 16:09.23 | astCoderX | well i'm not a pro in asterisk.. i call trunk the channel i'm using to make an outgoing call..... |
| 16:09.24 | De_Mon | any way to run a replace or substitute command in asterisk short of using func_odbc? |
| 16:09.35 | De_Mon | or agi -- shutter |
| 16:09.47 | De_Mon | shudder? |
| 16:09.54 | blitzrage | De_Mon: you mean a SQL command? you need func_odbc or agi |
| 16:09.58 | blitzrage | func_odbc is *easy* |
| 16:10.03 | astCoderX | ManXPower: i've reading about chanisavail and think it always says the channel is avalaible for outgoing calls |
| 16:10.20 | *** join/#asterisk kand (n=kanderso@139.148.188.72.cfl.res.rr.com) |
| 16:10.30 | De_Mon | blitzrage I just want to REPLACE(string,word,replacement) steems silly to call func_odbc and use an external database to do something like that |
| 16:10.46 | blitzrage | replace what? |
| 16:11.40 | [TK]D-Fender | astCoderX>well i'm not a pro in asterisk.. i call trunk the channel i'm using to make an outgoing call..... <-- not an answer. What technology are you calling out on? |
| 16:11.48 | *** join/#asterisk thansen|laptop (n=thansen@19.243.sfcn.org) |
| 16:11.49 | astCoderX | oh sorry |
| 16:11.50 | astCoderX | SIP |
| 16:12.22 | mosty | astCoderX, the only way to find out if you can make a call, is to make a call and see if it worked |
| 16:12.40 | [TK]D-Fender | astCoderX: then you'll have to parse out "show channels concise" or similar to get the usage count. Or insert a bunch of GROUPCOUNT function calls throughout your dialplan. |
| 16:13.05 | De_Mon | blitzrage I've storing the name of channels that need to be bridged to my queue in DB(Qeueues/queue-name/bridge) and since that key could contain multiple channels (; delimited) I want to remove "SIP/somechannel-ayxz;" from astdb once it is bridged. |
| 16:13.11 | astCoderX | well i wanto to do this. I must to do a script to dialing out. But before to dial i need to know if the outgoing channel is free. The asterisk admin told me he configured for manage 30 concurrent outgoing calls |
| 16:13.20 | blitzrage | De_Mon: CUT() |
| 16:13.41 | blitzrage | although the ; delimiter might screw with you |
| 16:13.47 | blitzrage | (and it probably will) |
| 16:13.49 | mosty | astCoderX, use GROUP_COUNT and GROUP |
| 16:14.12 | astCoderX | mosty: i will check the doc of group_count |
| 16:14.14 | De_Mon | blitzrage that means looping through each delimiter and testing for a match |
| 16:14.14 | *** join/#asterisk dadbee (n=josh@66.207.134.74) |
| 16:14.21 | blitzrage | De_Mon: pretty much ya |
| 16:14.50 | blitzrage | I don't know of any other functions to do that -- might want to double check the list of functions |
| 16:14.55 | *** join/#asterisk pirulo (n=andres_p@65.19.28.123) |
| 16:15.06 | blitzrage | De_Mon: patches accepted! :) |
| 16:15.18 | kand | de_mon: I have already writen such a dialplan macro like blitzrage is discussing if you want it. |
| 16:15.44 | De_Mon | I thought about looking at improving func_regex() and ading substitution support, but that, would be a lot of work |
| 16:15.55 | *** join/#asterisk pirulo (n=andres_p@65.19.28.123) |
| 16:16.59 | De_Mon | kand that'd be cool, might save me some time if we go that route |
| 16:17.46 | kand | de_mon: here you go: http://pastebin.ca/813507 |
| 16:22.06 | De_Mon | hey kand why is it depriciated? |
| 16:22.43 | Nugget | s/depriciated/deprecated/ :P |
| 16:23.08 | kand | de_mon: lol, I am the worst speller ever, but it works fine I just dont use it anymore in my dialplan |
| 16:24.22 | De_Mon | its actually depreciated, I just got carried away with the i's (you spelled it right kand!) |
| 16:24.24 | *** join/#asterisk bmcghee (n=brentmcg@d66-183-250-149.bchsia.telus.net) |
| 16:24.34 | De_Mon | kand why did you stop using it |
| 16:24.58 | De_Mon | you started using func_odbc and a real database didn't you |
| 16:25.14 | file | it's deprecated, not depricated or depriciated or dipricated or even diprecated |
| 16:25.17 | kand | de_mon: It was slow and yes........ but not for find replace |
| 16:25.42 | Nugget | "depreciated" is a completely different word. |
| 16:25.48 | Nugget | "deprecated" is the word you think you're using. |
| 16:25.59 | kand | "no longer needed" |
| 16:26.28 | De_Mon | ooh... I forgot about that other word we both fail |
| 16:26.38 | FlatFoot | afternoon all , i am trying to set a VAr within different contexts BUT i need it to be declared not within a dial routine is this possible ? |
| 16:27.49 | *** join/#asterisk pirulo (n=andres_p@65.19.28.123) |
| 16:28.32 | mosty | FlatFoot, where do you want it declared? |
| 16:28.53 | FlatFoot | basically just after the context name so ... |
| 16:28.59 | FlatFoot | [mycontext] |
| 16:29.04 | FlatFoot | myvar = xxxx |
| 16:29.19 | mosty | so just use the Set dialplan command |
| 16:29.32 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-6b05d96d8e20199b) |
| 16:29.33 | *** mode/#asterisk [+o bkruse] by ChanServ |
| 16:29.39 | FlatFoot | thats what i wasn't sure about |
| 16:29.54 | FlatFoot | i cant't use exten => 1,1,Set etc |
| 16:30.00 | file | bkruse: you. |
| 16:30.01 | De_Mon | FlatFoot no, you have to set global variables to make the accessable |
| 16:30.03 | *** join/#asterisk mwilson-cobasys (n=mwilson@70.90.142.202) |
| 16:30.08 | mosty | FlatFoot, why can't you do that? |
| 16:30.10 | bkruse | file: Hi! |
| 16:30.11 | mwilson-cobasys | hello all |
| 16:30.14 | bkruse | file: drink? |
| 16:30.15 | De_Mon | FlatFoot try something like CONTEXTNAME_VARIABLE |
| 16:30.29 | file | bkruse: I *suppose* so |
| 16:30.43 | bkruse | what kind!? |
| 16:30.49 | file | mountain dew! |
| 16:30.50 | FlatFoot | because i have a set of outgoing dial routines that i will include within diff context's and need to be able to change a few values within per context |
| 16:30.57 | bkruse | file: done. done. done. |
| 16:31.20 | file | lies. |
| 16:32.17 | mwilson-cobasys | anyone here used Grandstream FXO gateways before |
| 16:32.30 | mosty | FlatFoot, organise your includes so that they don't include these Set calls |
| 16:32.54 | FlatFoot | mosty: not quite with you |
| 16:32.56 | riddlebox | mwilson-cobasys, I used the ht488, which was one fxo and 1 fxs |
| 16:33.37 | FlatFoot | ~pb |
| 16:33.37 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 16:33.47 | mwilson-cobasys | riddlebox, I keep getting this error on incoming calls... chan_sip.c:13774 handle_request_invite: Call from '6201' to extension '6000' rejected be |
| 16:33.48 | mwilson-cobasys | cause extension not found |
| 16:34.13 | riddlebox | mwilson-cobasys, which fxo is it? |
| 16:34.15 | mosty | FlatFoot, i don't see why you can't just use Set to set these variables in each context you need it |
| 16:34.40 | mwilson-cobasys | have you seen that before?... 6201 is the 8 port FXO gateway setup as a peer |
| 16:35.03 | FlatFoot | mosty thats the bit i'm after being able to set a var within a context BUT not in a dial routine. |
| 16:35.34 | mwilson-cobasys | I have the sip debug captured, but I dont understand it |
| 16:35.43 | mosty | FlatFoot, what do you mean by "within a dial routine" ? |
| 16:35.49 | FlatFoot | mosty: http://pastebin.com/m5b68233a |
| 16:36.02 | riddlebox | mwilson-cobasys, just a dumb question, but is there an extension 6000? |
| 16:36.09 | FlatFoot | thats part of the dial routines that i need to include within context's |
| 16:36.13 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:36.41 | FlatFoot | but i need to change the callerid and other bits , so wanted to declare these at the beginning of the context |
| 16:37.21 | mwilson-cobasys | yes 6000 exists & is a softphone that does register fine |
| 16:38.02 | mosty | flatfoot: so set the cdr variables etc in other contexts that the call passes through before this context |
| 16:38.53 | riddlebox | mwilson-cobasys, can you pastebin your dialplan? |
| 16:39.33 | FlatFoot | mosty: i shall investigate thanks |
| 16:40.26 | mwilson-cobasys | ok probably gonna sound dumb, but pastebin?? |
| 16:40.40 | riddlebox | ~pb |
| 16:40.41 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 16:41.32 | mwilson-cobasys | ok 1 sec |
| 16:41.36 | [TK]D-Fender | mwilson-cobasys: it is looking for "6000" in your DIALPLAN in the targeted context. A SIP device is NTO an extension. |
| 16:41.45 | [TK]D-Fender | mwilson-cobasys: this is a DIALPLAN ERROR |
| 16:42.11 | riddlebox | you had to steal my thunder ;) |
| 16:43.09 | *** join/#asterisk tobias (n=tobias@user-0ce2hr3.cable.mindspring.com) |
| 16:43.16 | nestAr | USER ERRAH! |
| 16:44.20 | mwilson-cobasys | what would I put in the dialplan so that all incoming calls route to the 6000 ext? sorry about this I'm still new, gui wouldnt let me setup a dialplan without a service provider, and gui doesnt accept the service provider as the sip device config |
| 16:44.48 | riddlebox | ERRAH ERRAH I swear she drowned in the lake |
| 16:45.01 | mosty | mwilson-cobasys, what gui? |
| 16:45.05 | *** join/#asterisk bjweeks (n=bjweeks@unaffiliated/bjweeks) |
| 16:45.09 | mwilson-cobasys | asterisk-gui |
| 16:45.11 | [TK]D-Fender | mwilson-cobasys: your 6000 is the name of your SIP DEVICE, not an EXTENSION. |
| 16:45.36 | riddlebox | mwilson-cobasys, I believe that grandstream even has examples on their site |
| 16:45.50 | [TK]D-Fender | mwilson-cobasys: An extension is number you can dial in your DIALPLAN. You can have 1,000,000 extensions in your dialplan taht have NOTHING to do with making a phone ring. |
| 16:47.12 | *** join/#asterisk rcphq (n=rllibre@200.42.219.109) |
| 16:47.19 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 16:48.31 | *** join/#asterisk hfb (n=hfb@pool-71-106-219-180.lsanca.dsl-w.verizon.net) |
| 16:49.33 | astCoderX | i've been checking the GROUP and GROUP_COUNT docs. i think i couldn't use groups because i can't modify the current dialplan for outgoing calls |
| 16:49.40 | mwilson-cobasys | ok let me go through some of this stuff... I may be back |
| 16:50.09 | ManxPower | [TK]D-Fender: what about that horrid little calllimit= option? |
| 16:50.13 | [TK]D-Fender | astCoderX: can't modify the dialplan?!?! WTF? |
| 16:50.22 | [TK]D-Fender | ManxPower: You jsut said it all... |
| 16:50.39 | bmcghee | Hey All |
| 16:50.41 | bmcghee | hows it going |
| 16:51.26 | astCoderX | well i don't think the admin is going to modify the dial plan to put on every dial or anwer for setting the group |
| 16:51.31 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 16:51.31 | ManxPower | mwilson-cobasys: You would use exten => _X.,1,Goto(6000,1) That would route all calls to the line exten => 6000,1,Whatever |
| 16:51.39 | astCoderX | so i couldn't use GROUP_COUNT |
| 16:51.57 | ManxPower | astCoderX: You really should not be on this channel if you can't change the Asterisk config./ |
| 16:52.46 | [TK]D-Fender | ManxPower: He doesn't HAVE an exten => 6000 ...... another person who can't differentiate a SIP device from an extension. |
| 16:52.47 | ManxPower | An extension starts with exten => There is nothing else in asterisk that is called an "extension" except for lines in extensions.conf that behind with exten =? |
| 16:52.49 | astCoderX | well like i say.. i'm just a programmer not a asterisk guru. And it's not like i can't do the changes but i don't have the privbileges to do it. |
| 16:53.04 | [TK]D-Fender | astCoderX: Don't ask for help when you can't follow through with it. |
| 16:53.14 | ManxPower | [TK]D-Fender: Not really my problem. I answered his question. |
| 16:53.35 | ManxPower | astCoderX: then get someone to this channel that CAN make changes. |
| 16:53.39 | astCoderX | thats why i wanted to know another ways to do it with programming via Asterisk Manager Interface or something like that |
| 16:53.40 | [TK]D-Fender | ManxPower: :/ |
| 16:53.42 | riddlebox | mwilson-cobasys, did you say it was the 6201? I dont even see a 6201 in their lineup? |
| 16:54.22 | [TK]D-Fender | astCoderX: You can't change your call flow without changing your dialplan. |
| 16:54.45 | astCoderX | ok got it. |
| 16:54.48 | [TK]D-Fender | 11:33]<mwilson-cobasys>riddlebox, I keep getting this error on incoming calls... chan_sip.c:13774 handle_request_invite: Call from '6201' to extension '6000' rejected be |
| 16:54.51 | [TK]D-Fender | ^^^^^ |
| 16:55.13 | mwilson-cobasys | 6201 is just what I called it.. this Grandstream GXW4108, 8 port FXO gateway |
| 16:55.19 | [TK]D-Fender | riddlebox: SIP/6201 is dialing 6000@somecontext. |
| 16:55.32 | [TK]D-Fender | riddlebox: And therein lies the dialplan failure |
| 16:55.57 | mwilson-cobasys | From: "unknown"<sip:unknown@69.220.229.51>;tag=35134a2525914677 |
| 16:55.57 | mwilson-cobasys | To: <sip:6000@69.220.229.51>;tag=as16a6b7fc |
| 16:56.05 | riddlebox | yeah I figured it was where the problem was, but I asked which grandstream model it was and he said 6201 |
| 16:56.06 | mwilson-cobasys | thats what I see |
| 16:58.29 | ManxPower | mwilson-cobasys: too bad you give your SIP user ID's 4 digit numbers that look like extension numbers. |
| 16:58.56 | [TK]D-Fender | mwilson-cobasys: pastebin the ENTIRE CALL. Those single -line pastes are not doing us any good. And make sure to include your entire dialplan while you're at it |
| 16:58.58 | ManxPower | making the SIP userid be bobs-phone or 4o57834o would be less confusing for you. |
| 16:59.38 | riddlebox | mwilson-cobasys, http://grandstream.com/user_manuals/GXW410x_User_Manual.pdf, it talks about asterisk in it |
| 16:59.53 | mort_gib | ManxPower: Whats wrong with calling SIP (users) the extension they will get?? |
| 16:59.55 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 17:00.24 | *** join/#asterisk mog (i=mog@nat/digium/x-888e9566e0b92c7c) |
| 17:00.24 | *** mode/#asterisk [+o mog] by ChanServ |
| 17:00.26 | mwilson-cobasys | ok let me make some changes that have been suggested so far & them will pastebin in a few |
| 17:04.03 | *** part/#asterisk rcphq (n=rllibre@200.42.219.109) |
| 17:10.20 | *** join/#asterisk Nuxi (n=david@host-72-175-248-154.static.bresnan.net) |
| 17:10.39 | *** join/#asterisk s0lid (n=_freq@210.213.199.146) |
| 17:10.50 | *** join/#asterisk jhb (n=joerg@81-5-139-2.dsl.eclipse.net.uk) |
| 17:11.04 | Nuxi | After I answer, I am trying to stream a gsm file. The first part of the gsm is chopped off. What am I doing wrong? |
| 17:12.27 | jhb | Hi *. I have an "exten => 1,1,Curl(http://www.baach.de/)" (asterisk 1.4.15), but get pbx.c:1816 pbx_extension_helper: No application 'Curl' for extension. |
| 17:12.29 | jhb | any ideas |
| 17:12.48 | Kobaz | Nuxi: are you doing something like Answer(xx).. Play()... |
| 17:13.11 | blitzrage | Nuxi: Playback(silence/1&file_you_want_to_play) |
| 17:13.36 | Kobaz | Nuxi: if your play is right after your answer, the answer may be too short |
| 17:13.37 | blitzrage | or do: Answer(), Wait(1), Playback() |
| 17:13.40 | Nuxi | I am using agi. agi->answer then agi->stream_file |
| 17:13.47 | [TK]D-Fender | jhb: Clearly CURL is not an application that exists |
| 17:14.00 | Nuxi | I put a delay before the steaming and it waits, but then still chops it off. |
| 17:14.23 | Kobaz | Nuxi: maybe your gsm file itself has the beginning chopped off? |
| 17:14.24 | blitzrage | do the silence/1 trick |
| 17:14.38 | awk | ok help please.. going to copy and paste my question.. not going to write it out again.... |
| 17:14.40 | awk | can somebody shed some light on a iptable query.. I want to do 1:1 NAT aswell as some port forwarding.. does the |
| 17:14.40 | awk | <PROTECTED> |
| 17:14.40 | awk | <PROTECTED> |
| 17:14.40 | awk | <PROTECTED> |
| 17:14.43 | awk | <PROTECTED> |
| 17:14.46 | *** join/#asterisk mltlnx (n=mltlnx@cpe-68-173-11-113.nyc.res.rr.com) |
| 17:14.53 | Kobaz | awk: wrong channel |
| 17:14.55 | Nuxi | I can play the gsm. it is not chopped off. |
| 17:14.56 | [TK]D-Fender | awk: Here's an idea, NEVER SPAM LIKE THAT AGAIN |
| 17:15.04 | awk | fuck off [TK]D-Fender |
| 17:15.09 | Nuxi | Seems odd to have to use a silence trick to play a gsm. |
| 17:15.12 | Kobaz | awk: join #networking |
| 17:15.13 | awk | it wasn't supposed to spread over those lines |
| 17:15.28 | awk | Kobaz, nobody in iptables or linuxhelp can answer this.. |
| 17:15.43 | Kobaz | have you tried #networking ? |
| 17:15.50 | awk | let me try there.. |
| 17:15.56 | jhb | [TK]D-Fender, ok, but 'core show function CURL' means its there |
| 17:16.12 | [TK]D-Fender | jhb: well thats a FUNCTION, not an APPLICATION. |
| 17:16.30 | [TK]D-Fender | jhb: So call it the way you're supposed to for it being a function. |
| 17:16.59 | jhb | [TK]D-Fender, thx a lot, I see what you mean |
| 17:21.20 | jhb | great, works. Now, if Dial(sip/123&sip/456), and 123 picks up, is there a way to trigger e.g. a set(foo=${Curl(...)}) on SIP/456, the 'failing' call? |
| 17:22.25 | blitzrage | don't think so.... only 2 ways I can think of, and they are both kinda messy |
| 17:23.27 | [TK]D-Fender | jhb: Probably using the Macro option in dial to see which channel answered. |
| 17:24.15 | jhb | [TK]D-Fender, thx again, will read on this. Cheers |
| 17:24.40 | [TK]D-Fender | jhb: np, let us know how it works out for you. |
| 17:24.58 | [TK]D-Fender | jhb: What are you looking to do to that guy who didn't make it first? |
| 17:28.13 | *** join/#asterisk ibob63 (n=james@88-97-143-14.dsl.zen.co.uk) |
| 17:29.15 | *** join/#asterisk mbranca (n=matteo@81.208.92.210) |
| 17:30.38 | *** join/#asterisk Victor_Yure (n=aaa@postfix.tradein.com.br) |
| 17:31.15 | ibob63 | hi guys. can anyone recommend a good iax provider in Germany? |
| 17:31.19 | *** join/#asterisk rcphq (n=rllibre@200.42.219.109) |
| 17:31.28 | *** part/#asterisk rcphq (n=rllibre@200.42.219.109) |
| 17:32.29 | [TK]D-Fender | ibob63: Taht might be a fair bit more difficult. Any specific reason for IAX? |
| 17:34.47 | mwilson-cobasys | ok D-Fender, here is the SIP debug http://pastebin.com/m6df31404 |
| 17:37.27 | *** join/#asterisk klyrelion (n=Kevin@boyne.demon.co.uk) |
| 17:38.21 | ibob63 | Hi D-Fender. I was thinking of using iax rather than sip because I understand this works better with asterisk |
| 17:39.18 | *** part/#asterisk klyrelion (n=Kevin@boyne.demon.co.uk) |
| 17:39.28 | [TK]D-Fender | mwilson-cobasys: Now after all that time you took for a simple SIP debug, you still didn't include your DIALPLAN like I asked. But here is the problem clear as day : Looking for 6000 in incoming (domain 69.220.229.51) - SIP/2.0 404 Not Found - . Quite clearly you don't have an exten in [incoming] that can match "6000" |
| 17:39.43 | [TK]D-Fender | ibob63: No, I'd generally say no to that. |
| 17:39.47 | mwilson-cobasys | and here is the dialplan |
| 17:39.49 | mwilson-cobasys | http://pastebin.com/d524e0efd |
| 17:40.21 | [TK]D-Fender | mwilson-cobasys: Your device is using a context called [incoming] and it doesn't even exist. |
| 17:40.29 | *** join/#asterisk Defraz (n=t0tal@fw.fuzecore.com) |
| 17:41.10 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 17:41.13 | mwilson-cobasys | That contest came from the mfg notes on asterisk, sorry |
| 17:41.30 | mwilson-cobasys | context |
| 17:41.45 | ManxPower | mwilson-cobasys: http://pastebin.com/m21836e94 |
| 17:41.48 | ManxPower | see my additions |
| 17:41.49 | [TK]D-Fender | mwilson-cobasys: Don't apologize to me, just go deal with it. You should know exactly what to add |
| 17:42.15 | [TK]D-Fender | ManxPower++ |
| 17:42.39 | [TK]D-Fender | ManxPower: For comical yet still technically mean assistance :) |
| 17:42.52 | ManxPower | mwilson-cobasys: also I'm surpized Asterisk loads at all. You don't have a [general] section a [globals] section |
| 17:43.19 | mwilson-cobasys | thanks, I will add in a few |
| 17:43.26 | [TK]D-Fender | ManxPower: See that [numberplan-custom-1] ? You KNOW what that means, don't you? |
| 17:43.34 | ManxPower | mwilson-cobasys: you need to read The Book. |
| 17:44.00 | ManxPower | [TK]D-Fender: No, but I'll bet it means we can feed mwilson-cobasys to the alligators |
| 17:44.05 | mwilson-cobasys | I know I do need to read the book, I can admidt that |
| 17:44.10 | [TK]D-Fender | ManxPower: *-GUI |
| 17:44.15 | [TK]D-Fender | ManxPower: And yup. |
| 17:44.31 | ManxPower | mwilson-cobasys: Did you know that you are on the wrong channel? |
| 17:44.40 | ManxPower | You should be on #asterisk-gui |
| 17:44.58 | ManxPower | no wonder it took so long to be able to help you. |
| 17:46.21 | *** join/#asterisk shido6 (n=shido6@74-130-53-46.dhcp.insightbb.com) |
| 17:46.57 | mwilson-cobasys | that worked |
| 17:47.07 | mwilson-cobasys | thank for helping the noob |
| 17:47.13 | ManxPower | mwilson-cobasys: now you need to go read the book. |
| 17:47.38 | mwilson-cobasys | will do |
| 17:50.17 | ManxPower | [TK]D-Fender: I whined about a couple of these things on #asterisk-dev just a few moments ago. |
| 17:54.14 | *** join/#asterisk mltlnx (n=mltlnx@cpe-68-173-11-113.nyc.res.rr.com) |
| 17:55.13 | *** join/#asterisk moprilo (n=nada@190.10.0.84) |
| 17:56.53 | *** join/#asterisk mltlnx (n=mltlnx@cpe-68-173-11-113.nyc.res.rr.com) |
| 17:57.42 | *** join/#asterisk gardo (n=gardo@61.14.191.140) |
| 18:00.45 | ManxPower | ~seen bkruse |
| 18:00.47 | jbot | bkruse is currently on #asterisk-dev (1h 30m 57s) #asterisk (1h 30m 57s) #openmoko (1h 30m 57s). Has said a total of 4 messages. Is idling for 1h 29m 50s, last said: 'file: done. done. done.'. |
| 18:01.21 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 18:03.45 | *** join/#asterisk thansen|laptop (n=thansen@19.243.sfcn.org) |
| 18:06.04 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 18:07.39 | bkruse | ManxPower: whats up |
| 18:07.41 | bmcghee | ~seen bmcghee |
| 18:07.42 | jbot | bmcghee is currently on #asterisk (1h 43m 18s). Has said a total of 3 messages. Is idling for 1s, last said: '~seen bmcghee'. |
| 18:07.42 | bmcghee | lol |
| 18:08.00 | bmcghee | w00 3 messages |
| 18:08.07 | ManxPower | hekko, bkruse |
| 18:08.14 | ManxPower | and hello too. |
| 18:08.25 | *** join/#asterisk |omni| (n=rob@c-67-185-144-206.hsd1.wa.comcast.net) |
| 18:08.54 | ManxPower | ~seen manxpower |
| 18:08.55 | jbot | manxpower is currently on #asterisk (23m 35s). Has said a total of 46 messages. Is idling for 1s, last said: '~seen manxpower'. |
| 18:10.05 | bkruse | ManxPower: whats up? |
| 18:10.34 | ManxPower | bkruse: Was just trying to find stats to backup my rant on -dev |
| 18:10.50 | bkruse | oh hheh :P |
| 18:10.59 | ManxPower | my rant was "Digium developers don't spend enough time answering newbie questions." |
| 18:11.27 | ManxPower | If they answered newbie questions, I'll bet the issues that newbies always ask about would fixed pretty fast. |
| 18:11.31 | bkruse | I do, but on my focus in #asterisk-gui |
| 18:12.01 | ManxPower | bkruse: thank Dog SOMEONE is helping the poor WIMPS (window icon mouse pointer system) people. |
| 18:12.29 | bkruse | I do :] |
| 18:12.49 | *** part/#asterisk ibob63 (n=james@88-97-143-14.dsl.zen.co.uk) |
| 18:12.50 | ManxPower | the #asterisk-gui people keep coming here because nobody helped them on #asterisk-gui |
| 18:13.07 | bkruse | ManxPower: then say my name, so my irc client goes blinky. |
| 18:13.24 | bkruse | I have no problem with helping, but im working on the products that they are asking questions about, which is sometimes hard to balance. |
| 18:13.31 | ManxPower | They make a total mess, tracking in dirt, spilling their drinks, leaving empty beer cans all over the place. Better to keep them in their squalid trailer called #asteriskgui |
| 18:13.45 | bkruse | I hate to be "that guy" but we DO have an awesome appliance tech support |
| 18:13.47 | ManxPower | bkruse: I understand. |
| 18:14.10 | bkruse | ManxPower: but if you say 'bkruse' in a sentence or message in any chat, I will response within a couple seconds, most likel |
| 18:14.19 | bkruse | s/likel/likely/g |
| 18:14.34 | ManxPower | bkruse: I'll keep that min mind for the -GUI people. |
| 18:15.07 | *** join/#asterisk HCEvan (i=edude@pool-72-77-226-91.tampfl.fios.verizon.net) |
| 18:16.00 | ManxPower | bkruse: Here's another suggestion (mostly for other developers): once per day, spend 30 mins reviewing the logs of #asterisk, looking for items that would make the user experience better and have fewer frequently asked questions |
| 18:16.26 | *** part/#asterisk techie (n=techie@adsl-76-247-10-106.dsl.lsan03.sbcglobal.net) |
| 18:16.48 | bkruse | ManxPower: that might work, and yes, please do for gui people |
| 18:17.23 | ManxPower | If Me, [TK]D-Fender, and JT all get burnt out at the same time, this channel is pretty much screwed. |
| 18:17.27 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net) |
| 18:18.22 | ManxPower | [TK]D-Fender: I still think all three of us should not be on #asterisk for a week, |
| 18:18.46 | blitzrage | you're addicted, and you know it :) |
| 18:18.47 | [TK]D-Fender | ManxPower: Call a union meeting to see if we're clear to strike... |
| 18:18.53 | blitzrage | heh |
| 18:19.57 | *** join/#asterisk af_ (n=getsmart@88-149-241-31.dynamic.ngi.it) |
| 18:20.18 | ManxPower | [TK]D-Fender: people that work for free have unions? 8-) |
| 18:20.40 | [TK]D-Fender | ManxPower: Not mutually exclusive. |
| 18:21.28 | ManxPower | I suppose not, but the union platform would be a tad lean. Higher Wages, um, no. Better benefits, um, no. Shorter work week, uh no. |
| 18:21.30 | ManxPower | 8-) |
| 18:22.05 | ManxPower | Piss people off, well you and I are already pretty good at that 8-) |
| 18:23.11 | ManxPower | anyway, I have things to do |
| 18:23.12 | *** part/#asterisk ManxPower (n=manxpowe@194.sub-75-201-101.myvzw.com) |
| 18:24.12 | *** join/#asterisk asr33 (n=asr33@ppp-RAS1-5-233.dialup.eol.ca) |
| 18:24.16 | *** join/#asterisk Fisix (i=sbk@i.have.30.efnetsluts.com) |
| 18:24.44 | [TK]D-Fender | blitzrage: Piss people off? Me? |
| 18:25.11 | [TK]D-Fender | [12:15]<awk>fuck off [TK]D-Fender <--- Feelt he love! |
| 18:25.38 | riddlebox | yeah I was suprised you didnt have anything to say about that? |
| 18:25.57 | Yourname`` | What's a good STUN server I can use? |
| 18:26.25 | [TK]D-Fender | riddlebox: I figured if he couldn't take the comment silently realizing the trusth of my point, there's no point in "retaliating". |
| 18:26.34 | [TK]D-Fender | Yourname`I think FWD has one. |
| 18:26.38 | riddlebox | yeah |
| 18:26.55 | [TK]D-Fender | And I can't type for beans today. |
| 18:26.59 | [TK]D-Fender | :/ |
| 18:28.03 | Yourname`` | [TK]D-Fender: Ok. |
| 18:28.36 | Yourname`` | Another question, when I call an extension of a queue, and the queue rings a logged in agent. It just says Called 301, but not 301 is ringing right after that like it usually does. What does that mean? |
| 18:28.54 | [TK]D-Fender | Yourname``: Here's a big list : http://www.voip-info.org/wiki-STUN |
| 18:28.55 | *** join/#asterisk |omni| (n=rob@c-67-185-144-206.hsd1.wa.comcast.net) |
| 18:29.18 | [TK]D-Fender | Yourname`Pastebin the complete CLI output. |
| 18:29.47 | Yourname`` | One sec.. |
| 18:30.39 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 18:30.43 | *** join/#asterisk robeph (n=robf@24.214.206.254) |
| 18:31.46 | robeph | hmmhesays: it was a problem with sysctl and the asterisk init script, I think it was for dapper and not feisty so it had some incongruities, I think the whole issue stemmed from the svn not having the correct scripts when I updated the box... thus the bad fds |
| 18:32.07 | robeph | but now it works, thanks for the help lastnight/morning |
| 18:32.29 | Yourname`` | [TK]D-Fender: Parsing it.. one sec |
| 18:34.57 | hmmhesays | how come increasing the file descriptor limit worked then? |
| 18:35.36 | *** join/#asterisk rcphq (n=rllibre@200.42.219.109) |
| 18:35.38 | *** join/#asterisk mltlnx (n=mltlnx@cpe-68-173-11-113.nyc.res.rr.com) |
| 18:35.39 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 18:35.44 | *** part/#asterisk rcphq (n=rllibre@200.42.219.109) |
| 18:37.26 | hmmhesays | what exactly what the problem? |
| 18:38.58 | Yourname`` | [TK]D-Fender: http://pastebin.ca/813692 |
| 18:40.29 | tzafrir | robeph, sysctl? |
| 18:40.38 | tzafrir | huh? |
| 18:40.40 | hardwire | blah |
| 18:40.57 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 18:41.36 | *** join/#asterisk mltlnx (n=mltlnx@cpe-68-173-11-113.nyc.res.rr.com) |
| 18:42.23 | Yourname`` | [TK]D-Fender: This is really weird too. It just stays at Called 301, not being immediately followed by something like SIP/301-45ef23 is ringing. Which seems to be the case with everybody else. Now, 301 is on a remote PC. But when I try doing the same to 301 on my PC, it works? |
| 18:46.56 | *** join/#asterisk annedonaldson17 (n=ryan@magic.skylab.org) |
| 18:47.04 | [TK]D-Fender | Yourname``: So the remote PC won't actually ring, the local will? |
| 18:47.24 | annedonaldson17 | Has anyone every owned a TE210P and upgraded it with the Octasic echo cancellation card? |
| 18:47.33 | annedonaldson17 | There are _zero_docs_ on this ritual. |
| 18:47.51 | Yourname`` | [TK]D-Fender: Yes :S |
| 18:48.45 | [TK]D-Fender | Yourname``: First guess.. they other guy's behind NAT and your account wasn't set up to deal with it. It then tries to contact the loacl IP if got on register to no avail... |
| 18:50.06 | Yourname`` | [TK]D-Fender: It just worked when I included a couple other contexts to it, which is weird. But meanwhile, this is the scnario, a call comes into the queue1 (sandiego).. and then that call is transferred to another queue (exten 300, called united) by dialing a quicknumber 7. Then agents logged into the second queue called united, will hear their eyebeams ring. Currently, eyebeam shows "Live transfer" as callerid(name) which is intend |
| 18:50.06 | Yourname`` | w the phone number of the person instead. But it shows 35@65.xx.xx.xx |
| 19:05.54 | *** join/#asterisk hohum (n=dcorbe@h-74-1-66-114.lsanca54.covad.net) |
| 19:06.15 | hohum | How do I get my asterisk box to NOT strip headers out of an incoming message when I Dial() something else |
| 19:08.32 | blitzrage | annedonaldson17: I'm assuming you purchased the card, thus you should be entitled to technical support |
| 19:09.00 | blitzrage | hohum: strip what out? Asterisk is a B2BUA, not a proxy |
| 19:09.41 | hohum | I'm sending it a call with an Alert-Info header |
| 19:09.50 | hohum | and it's getting stripped |
| 19:10.04 | hohum | I realize it's a B2bUA but it shouldn't be arbitrarily stripping headers either |
| 19:10.33 | *** join/#asterisk nirz (n=nir@bzq-79-181-116-158.red.bezeqint.net) |
| 19:11.59 | Yourname`` | [TK]D-Fender: Ok, so what ports should be forwarded? |
| 19:12.01 | Yourname`` | 5060 |
| 19:12.04 | Yourname`` | eyebeam being used. |
| 19:12.28 | [TK]D-Fender | ~sipnat |
| 19:12.29 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 19:13.22 | *** part/#asterisk moprilo (n=nada@190.10.0.84) |
| 19:13.23 | blitzrage | hohum: how are you doing this? If the variable is set on one leg of the channel, it won't go out on the 2nd leg of the channel unless you tell it to be transitive |
| 19:13.25 | [TK]D-Fender | hohum: Pastebin your complete call's CLI output at verbose 10 |
| 19:13.26 | [TK]D-Fender | ~pb |
| 19:13.27 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 19:13.28 | [TK]D-Fender | ^^^^^^^^^^^^^^^ |
| 19:13.35 | blitzrage | i.e. Set(__MY_TRANSITIVE_VARIABLE=value) |
| 19:14.02 | hohum | blitzrage: the call is being originated from an OpenSER box, so it's being SENT to Asterisk with the Alert-Info: header |
| 19:14.21 | hohum | the OpenSER endpoint knows it wants to page the target phone, which just happens to live on an asterisk box |
| 19:14.21 | blitzrage | hohum: ok... so it gets to Asterisk w/ the Alert-Info header? |
| 19:14.31 | hohum | icorrect |
| 19:14.33 | blitzrage | but it's not going to the phone on the OTHER channel leg |
| 19:14.44 | hohum | correct |
| 19:14.52 | blitzrage | those are two separate variable spaces -- asterisk will not pass that variable through unless it is transitive like I said above |
| 19:15.04 | blitzrage | they are independent calls |
| 19:15.17 | hohum | so how do I tell it to be transitive? |
| 19:15.22 | blitzrage | I said how to do it above |
| 19:15.33 | blitzrage | <PROTECTED> |
| 19:15.45 | hohum | ah |
| 19:16.13 | hohum | it's not a variable though, it's a header |
| 19:16.15 | *** join/#asterisk poor_man (n=chatzill@213.63.2.234) |
| 19:16.17 | bjweeks | "WARNING[8649] chan_sip.c: sip_xmit of 0x2aaab0043f20 (len 567) to 192.168.1.117:5060 returned -2: Bad file descriptor" just started getting this with trunk, bug? |
| 19:16.19 | hohum | so I don't quite understand |
| 19:16.27 | poor_man | Hello everyone! |
| 19:16.36 | hohum | and the asterisk box actually has know way of knowing what the header should be set to |
| 19:17.10 | annedonaldson17 | Hi Poor_man! |
| 19:17.13 | blitzrage | hohum: SIP_HEADER() |
| 19:17.55 | Yourname`` | This is SO weird. When I call that extension directly, it rings. But when I transfer to it .. it doesn't ring. |
| 19:17.56 | hohum | ah |
| 19:17.57 | hohum | I see |
| 19:18.13 | hohum | so I call SIP_HEADER() before Dial, and then SipAddHeader(), right? |
| 19:18.27 | hohum | seems simple enough |
| 19:18.37 | [TK]D-Fender | hohum: No, SipAddHeader only, and BEFORE your dial. |
| 19:18.59 | poor_man | is there any possibility to dial some code and make asterisk to private my number? |
| 19:20.42 | *** join/#asterisk ThatKidKel (n=Kelvin@cm-64-221-171-186.dhcp.southerncoastalcable.net) |
| 19:21.07 | [TK]D-Fender | poor_man: what are you dialing after this that you don't want to see your #? |
| 19:21.32 | hohum | like this, right? |
| 19:21.33 | hohum | exten => _X.,1,Set(foo=${SIP_HEADER(Alert-Info)}) |
| 19:21.33 | hohum | exten => _X.,2,SipAddHeader(Alert-Info: ${foo}) |
| 19:21.33 | hohum | exten => _X.,3,Dial(SIP/${EXTEN}@core) |
| 19:21.39 | Yourname`` | [TK]D-Fender: Call placed from an asterisk server with no NAT, transferred to a queue behind NAT, that agent from that queue transfers to another queue/agent behind another NAT. For which I have canreinvite=no, and both those agents are nat=yes. Correcto? |
| 19:21.56 | *** join/#asterisk Schumie (n=Steve@cpc1-rdng2-0-0-cust441.winn.cable.ntl.com) |
| 19:22.26 | poor_man | i'm not dialing nothing more that the number i want |
| 19:22.31 | [TK]D-Fender | hohum: might do. Pastebin how it works in run-ime |
| 19:22.43 | hohum | yeah |
| 19:22.45 | hohum | that worked perfectly |
| 19:22.46 | hohum | thanks |
| 19:22.48 | [TK]D-Fender | Yourname`and qualify=yes |
| 19:23.05 | hohum | you guys are awesome dudes |
| 19:23.08 | ThatKidKel | I've got a unique problem with dropped calls.. Dec 12 13:38:41 WARNING[5784] chan_sip.c: Hanging up call fabe60a73271e6348165b40161ce893b-47602a9d@ip.ad.dr.ess - no reply to our critical packet... I'm assuming the critical packet is an ACK.. The call is Answered() ACK sent, according to Logs.. But then queued.. When the call is answered from the queue, I get this error and hten the call drops.. It is not consiste |
| 19:23.30 | Yourname`` | [TK]D-Fender: Ok.. and under general for the asterisk server nat=yes, correct? |
| 19:24.03 | Yourname`` | [TK]D-Fender: I mean nat=no, since the asterisk server is not behind NAT |
| 19:24.23 | blitzrage | hohum: ya, you got the idea, nice :) |
| 19:24.50 | hohum | I appriciate the gentile massage in the correct direction |
| 19:24.52 | blitzrage | [TK]D-Fender: he needed the SIP_HEADER() because the Alert-Info was set outside of Asterisk originally |
| 19:25.05 | blitzrage | no problem -- I don't mind people who can actually help themselves |
| 19:25.17 | blitzrage | don't mind helping* |
| 19:25.37 | [TK]D-Fender | Yourname``: nat=yes for your PHONE'S entry |
| 19:26.09 | Yourname`` | [TK]D-Fender: But what about general? |
| 19:26.16 | [TK]D-Fender | Yourname``: that isn't just for [general] you know... read the guide! |
| 19:26.27 | [TK]D-Fender | Yourname`its all explained in full |
| 19:26.35 | blitzrage | ThatKidKel: yes, that usually means there was no ACK to the 200 OK that asterisk sent (usually a NAT problem........) |
| 19:27.08 | annedonaldson17 | So, most of you have just purchased TE212P's out of the box, yes? |
| 19:27.16 | Yourname`` | [TK]D-Fender: That's what I said! lol this was the second statement, I said "under general" |
| 19:27.46 | Yourname`` | [TK]D-Fender: The sip peers are nat=yes, but the [general] nat was set to yes, even though the asterisk server was NOT behind NAT |
| 19:28.03 | ThatKidKel | blitzrage.. that's the unique problem.. these phoens are directly on net, no nat involved.. |
| 19:28.26 | [TK]D-Fender | Yourname``: Well if we're to do more we'll need a comprehensive pastebin. |
| 19:28.42 | ThatKidKel | blitzrage.. does the phone send a 200 OK that would be sent back to the proxy when it answers a call in the queue |
| 19:28.43 | ThatKidKel | ?? |
| 19:28.46 | blitzrage | ThatKidKel: not much we can do without a complete SIP trace of the problem put into a pastebin |
| 19:29.08 | ThatKidKel | that's the thing, i'm gonna have to trace a pretty busy server for a problem that "MAY" happen :) |
| 19:29.10 | ThatKidKel | its inconsident |
| 19:30.17 | *** join/#asterisk poor_man (n=chatzill@213.63.2.234) |
| 19:30.21 | Yourname`` | [TK]D-Fender: It works now, thank you. But new thing now is even though an agent is logged into the queue, it still doesn't go into the queue and ring that logged in agent at times. |
| 19:30.28 | poor_man | sorry my wireless felt |
| 19:30.31 | ThatKidKel | blitzrage.. i just started a trace.. i'm hoping it will happen again |
| 19:30.37 | poor_man | Fender, as i was saying, i'm not dialing nothing more that the number i want |
| 19:31.15 | [TK]D-Fender | Yourname``: Well you showed it dialing what looks like a static device, not an agent.... |
| 19:31.58 | [TK]D-Fender | poor_man: I asked you what you were dialing out ON? What technology? What hardware, etc. |
| 19:32.00 | Yourname`` | [TK]D-Fender: But the agent is logged in though.. |
| 19:32.36 | [TK]D-Fender | Yourname``: You're talking oranges, and showing me applies and no proof that the oranges even EXIST. |
| 19:32.44 | Yourname`` | LOL |
| 19:32.50 | Yourname`` | How can I reload the queues.conf from CLI? |
| 19:33.00 | Yourname`` | [TK]D-Fender: Gimme two seconds.. trying something out with the queues. |
| 19:33.17 | [TK]D-Fender | Yourname``: "reload" usually does it. |
| 19:33.17 | Yourname`` | got it |
| 19:33.27 | Yourname`` | [TK]D-Fender: For some reason reload is killing current calls.. |
| 19:34.34 | davevg-btwtech | Yourname, if you just want to reload queues, try "reload app_queue.so" |
| 19:35.07 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 19:35.20 | Yourname`` | Hey dave, |
| 19:35.23 | Yourname`` | yeah, it worked, thank you |
| 19:35.37 | poor_man | fender, sorry ;) i'm using trixbox, with Asterisk 1.2.23-BRIstuffed-0.3.0-PRE |
| 19:35.39 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 19:35.49 | Yourname`` | This is again something weird. If I call the direct extension that is supposed to put the person in a queue.. it doesn't. It just goes to the voicemail of that queue exten, lol |
| 19:36.26 | Yourname`` | http://pastebin.ca/813746 |
| 19:36.30 | Yourname`` | This is what it is.. |
| 19:36.36 | Yourname`` | And there are agents logged in that queue. |
| 19:36.55 | *** join/#asterisk Oerd (n=Oerd@ip-90-187-135-80.web.vodafone.de) |
| 19:36.55 | Yourname`` | But when I call that exten 300 directly, it goes to the voicemail of that queue, instead of trying to go inside the queue. |
| 19:37.07 | Yourname`` | How can I test this queue if this is like a normal behaviour? |
| 19:37.30 | *** join/#asterisk bmg505 (n=leon@196.209.180.166) |
| 19:37.49 | poor_man | fender, with a Junghanns.NET PCI BASED ISDN INTERFACE CARD |
| 19:38.26 | Yourname`` | Perfect, I'm agent 30. And I'm not logged into the queue and I call 300, and it doesn't work.. and sends me to 300's voicemail. But when I log into the queue and THEN call 300, it works and puts me in the queue. What am I doing wrong again! |
| 19:39.41 | davevg-btwtech | what is your joinempty parameter for the queue in queues.conf set at? |
| 19:40.09 | Yourname`` | Before it was set to strict |
| 19:40.14 | Yourname`` | But now I set it to no |
| 19:40.16 | Yourname`` | And did a reload |
| 19:40.24 | Yourname`` | Still nogo |
| 19:40.25 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 19:40.42 | davevg-btwtech | Yourname, so if there are no agents logged in, it will skip the queue.. |
| 19:40.55 | Yourname`` | I know, but the agent is logged in. |
| 19:41.41 | putnopvut | Yourname`, have you tried not putting '0' as the timeout for the queue? |
| 19:41.43 | [TK]D-Fender | Yourname``: Looks like you are calling the Queue with a "0" timeout = instan failure |
| 19:42.07 | [TK]D-Fender | poor_man: ... |
| 19:42.10 | [TK]D-Fender | ~trixbox |
| 19:42.11 | jbot | [~trixbox] Trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org |
| 19:42.12 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^ |
| 19:42.15 | davevg-btwtech | i *think* he also has the pipe in the wrong place for the options |
| 19:42.30 | Qwell | caio1982: you realize that Digium sells t-shirts on the digium.com web store, right? |
| 19:42.48 | caio1982 | Qwell: actually I didnt know that |
| 19:42.50 | Yourname`` | [TK]D-Fender: timeout is set to 15 |
| 19:42.55 | Qwell | will link, sec |
| 19:43.02 | [TK]D-Fender | Yourname`not in the line that calls QUEUE |
| 19:43.21 | [TK]D-Fender | Yourname`those are completely DIFFERNT timeouts. |
| 19:43.24 | Qwell | http://store.digium.com/products.php?category_id=22 |
| 19:43.36 | Qwell | I really want one of the laptop backpacks :( |
| 19:43.37 | Yourname`` | [TK]D-Fender: Are you talking about the one in extensions.conf? |
| 19:43.45 | davevg-btwtech | try this Queue(unitedq|Tt||15) |
| 19:43.55 | [TK]D-Fender | Yourname``: one is for agent-dial, the on you call in the dialplan is for TOTAL time allowed to sit arount |
| 19:43.59 | [TK]D-Fender | arount. |
| 19:44.04 | [TK]D-Fender | around |
| 19:44.05 | caio1982 | Qwell: that's sweet! thanks for the address |
| 19:44.05 | [TK]D-Fender | asdfasldallasfdfd |
| 19:44.10 | Yourname`` | [TK]D-Fender : exten=> 300,n,Queue(unitedq||Tt|0) |
| 19:44.29 | Qwell | caio1982: word of warning - if you get the geek shirt, pay attention to the washing instructions |
| 19:44.34 | [TK]D-Fender | Yourname``: 0 = why don't we just hangup now. |
| 19:44.47 | poor_man | Fender, sorry for my english, what do you mean more specificly? |
| 19:44.53 | Qwell | the orange can easily bleed through the black - that's been my experience at least |
| 19:45.05 | Yourname`` | [TK]D-Fender: Let me try |
| 19:45.20 | caio1982 | Qwell: are they all made with the same material? |
| 19:45.30 | [TK]D-Fender | poor_man: I mean Trixbox is NOT supported here. We cannot help you. |
| 19:45.31 | Qwell | like...cotton? :p |
| 19:45.54 | Qwell | no, I think the geek one is a little more "fluffy"...or something |
| 19:46.09 | poor_man | ok, but it has asterisk also |
| 19:46.13 | Qwell | that, combined with the big logo makes for interesting issues... |
| 19:46.20 | Yourname`` | [TK]D-Fender: I changed it, and now it seems to be going in. BUT, when I call the exten 7, it doesnt RING the agent. |
| 19:46.21 | Qwell | the other two are quite nice |
| 19:46.22 | caio1982 | sometimes I stumble upon those "sports" thing, no sweatting etc, no cotton... |
| 19:46.32 | caio1982 | Qwell: great |
| 19:46.35 | [TK]D-Fender | poor_man: That does not matter. you are not in control of your dialplan. FreePBX (the GUI trixbox uses) controls EVERYTHING and you play by ITS rules. |
| 19:46.43 | [TK]D-Fender | poor_man: You are completely out of luck here. |
| 19:46.54 | poor_man | ok |
| 19:47.01 | poor_man | thanks for the info |
| 19:47.20 | [TK]D-Fender | Yourname`And you STILL aren't showing me anything of value... |
| 19:49.18 | hardwire | ok ok ok |
| 19:49.38 | hardwire | Yourname` to your side of the ring. |
| 19:52.59 | Yourname`` | [TK]D-Fender: That's because I'm still trying to figure out WHY when I call the queue the agent logged in from the other network doesn't get any rings.. :S |
| 19:53.10 | Yourname`` | Just said Called SIP/301 |
| 19:53.17 | Yourname`` | But doesn't give anything else after :S |
| 19:54.11 | [TK]D-Fender | Yourname``: rhetoric++ |
| 19:56.25 | Yourname`` | ok ok |
| 19:56.32 | Yourname`` | Alright, let me do the port forwarding. |
| 19:56.44 | Yourname`` | 5060 8000-9000 will be forwarded, right? |
| 19:57.07 | caio1982 | Qwell: have you seen this one to know if it's good looking in fact? looks like something back from the 70's: http://store.digium.com/productview.php?product_code=8ORANGETEE |
| 19:57.09 | [TK]D-Fender | ..... |
| 19:57.20 | [TK]D-Fender | Yourname``: read the guide again. |
| 19:57.26 | Yourname`` | And I'm thinking it's their sonicwall |
| 19:57.28 | Qwell | caio1982: I like it.. |
| 19:57.37 | [TK]D-Fender | Yourname``: And then after all that try and think of something useful to show me. |
| 19:58.07 | [TK]D-Fender | Yourname``: I'm running my work system behind a SonicWALL TZ170 myself. try again... |
| 19:58.27 | Yourname`` | [TK]D-Fender: So, I don't need to do anything on sonicwall? |
| 19:58.33 | caio1982 | Qwell: take action shots of that to show us :D |
| 19:58.33 | Yourname`` | [TK]D-Fender: No port forwarding? |
| 19:58.45 | Qwell | caio1982: it looks like the images there |
| 19:58.47 | [TK]D-Fender | Yourname`tell it to kepp the hell away from SIP NAT transform |
| 19:59.04 | Yourname`` | So disable it? |
| 19:59.07 | Yourname`` | uncheck it i mean? |
| 19:59.24 | [TK]D-Fender | Yourname``: OFF |
| 19:59.34 | *** join/#asterisk holiday_42 (n=no@spike.wcta.net) |
| 19:59.48 | Yourname`` | [TK]D-Fender: There is something called "Enable SIP transformations", which was checked. I just unchecked it. And then it also had "Enable consistent NAT" as checked. |
| 20:00.34 | Yourname`` | [TK]D-Fender: So, now I unchecked consistent nat and sip transformations. |
| 20:03.23 | Yourname`` | Ok, so basically I'm going to be giving pastebins. |
| 20:03.34 | Yourname`` | Since I call that agent and it doesn't ring.. it's a NAT issue. |
| 20:03.40 | Yourname`` | What would you like me to paste to you [TK]D-Fender? |
| 20:04.07 | [TK]D-Fender | Yourname``: Configs, calls out to the device with SIP debug, etc. |
| 20:05.54 | *** join/#asterisk fbnts (n=thomas@host86-141-143-173.range86-141.btcentralplus.com) |
| 20:06.44 | Yourname`` | k |
| 20:06.49 | fbnts | Hi, just a quick question - Does Asterisk 1.4 have support for the MeetMe application? |
| 20:07.37 | blitzrage | of course |
| 20:07.49 | blitzrage | you need zaptel installed though before you compile asterisk, or it won't compile |
| 20:08.00 | blitzrage | (you need a timing source, ala hardware, or ztdummy) |
| 20:08.16 | fbnts | ah right, I compiled asterisk 1st then ztdummy afterwards |
| 20:08.26 | blitzrage | just run, "make install" in Asterisk again |
| 20:09.00 | blitzrage | errr... thats wrong -- you need to run: ./configure ; make menuselect ; make install |
| 20:09.11 | blitzrage | ./configure needs to be run so Asterisk knows about the ztdummy |
| 20:10.25 | *** join/#asterisk angom (n=angom@200.38.31.239.dsl.dyn.telnor.net) |
| 20:11.09 | fbnts | ah great, just logging in to try it now! |
| 20:12.32 | jhb | hi *. I originate calls sip/123@foo and sip/456@foo, and if 123 connects would like to hangup 456. any ideas? |
| 20:12.48 | jhb | softhangup(sip/123@foo) does not do it |
| 20:14.05 | [TK]D-Fender | jhb: try to originate "sip/123@foo&sip/456@foo" |
| 20:14.08 | rajiv | where can i get high quality eps files of the asterisk and digium logos ? |
| 20:14.25 | [TK]D-Fender | rajiv: You don't |
| 20:14.48 | [TK]D-Fender | rajiv: Not without real contact with their marketing dept for trademark reasons. |
| 20:14.54 | rajiv | http://www.digium.com/en/company/digium-identity-guidelines.pdf has usage guidelines but not downloads |
| 20:15.46 | jhb | [TK]D-Fender, ok, in reality 123 has to press a certain key, and it depends on that key press wether to hangup 456 |
| 20:16.01 | rajiv | [TK]D-Fender: hmm. i am doing a presentation that falls within the usage guidelines |
| 20:16.15 | rajiv | well i emailed licensing@digium ... we'll see |
| 20:16.53 | [TK]D-Fender | jhb: Taht gets tricky. What I'd do is that when you orginate those 2 channels you set a variable in each call and inside the acknowledge macro, call an AGI that will scan for a channel with the same unique key set and issue the hangup for that channel. |
| 20:18.24 | jhb | [TK]D-Fender, how do you trigger the acknowledge macro? |
| 20:18.28 | jhb | (from originate) |
| 20:18.37 | jhb | put them into an extension that does it? |
| 20:19.16 | [TK]D-Fender | jhb: I'd dial a local channel that would Dial your SIP entry for you with the M option. |
| 20:20.09 | jhb | [TK]D-Fender, another great tip. Really good, helps a lot |
| 20:20.39 | [TK]D-Fender | jhb: NP... the really tricky part is the AGI to parse the open channels to try and kill the other channels... |
| 20:21.00 | [TK]D-Fender | jhb: OR..... I'm not sure it this'll work but... |
| 20:21.18 | *** join/#asterisk |omni| (n=rob@c-67-185-144-206.hsd1.wa.comcast.net) |
| 20:22.51 | [TK]D-Fender | jhb: I belive that if you use the "M" option taht the call is not infact truly considered answered until "acknowledged. you could maybe do a single originate of "Local/123@foo/n&Local/456@foo/n", and each of their "M"'s would not fallow through till answered. |
| 20:22.55 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 20:23.01 | [TK]D-Fender | jhb: Which saves you all that scripting mess |
| 20:23.09 | [TK]D-Fender | (race conditions, etc) |
| 20:24.18 | [TK]D-Fender | Follow* |
| 20:24.33 | Yourname`` | [TK]D-Fender: It worked! I did the sonicwall thing you asked me to do.. Enable sip .. and it worked! :) Thank you. |
| 20:25.31 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177581616.dsl.bell.ca) |
| 20:25.38 | fujin | </3 sonicwall |
| 20:27.02 | *** join/#asterisk Greek-Boy (n=email@41.221.58.5) |
| 20:28.43 | jhb | [TK]D-Fender, was just thinking about that |
| 20:29.34 | jhb | [hC], I think.... |
| 20:30.40 | outtolunc | ...... |
| 20:31.41 | outtolunc | i think, this week has been long enought and vote we call it friday and continue from there |
| 20:31.45 | outtolunc | er -t |
| 20:32.09 | hardwire | deal |
| 20:32.41 | outtolunc | sweet, now where is my friday newspaper at.. hmm |
| 20:32.44 | jhb | [TK]D-Fender, is there something like sip show channels for agi, so that I could see how they are mapped? |
| 20:33.48 | jhb | [hC], my app is a bit like followme. If somebody takes up the phone, and does not want to take the call, the other called parties should still have a chance |
| 20:33.59 | jhb | sorry, that was for [TK]D-Fender |
| 20:34.09 | jhb | [TK]D-Fender,, my app is a bit like followme. If somebody takes up the phone, and does not want to take the call, the other called parties should still have a chance |
| 20:34.11 | [hC] | oh i was gonna say... |
| 20:34.11 | [hC] | heh |
| 20:34.15 | jhb | ;-) |
| 20:34.23 | jhb | sorry |
| 20:34.35 | [TK]D-Fender | Yourname``: Nothing says "thank you!" quite like PayPal ;) |
| 20:35.11 | [TK]D-Fender | fujin: Yeah... they're "toasters" |
| 20:35.25 | fujin | lol |
| 20:35.35 | jhb | [TK]D-Fender, in that sense originate 123&456 would not be enough |
| 20:35.48 | fujin | I fully agreed with nothing says thank-you like Paypal, until someone from here frauded me nearly $1000 usd and got my paypal account frozen. |
| 20:35.52 | fujin | which fucked up all my christmas plans. |
| 20:36.07 | [TK]D-Fender | jhb: Sure, as long as each was a local channel, and no SIP/123 as I mentioned |
| 20:36.14 | jameswf | I like BevMO gift cards |
| 20:37.10 | holiday_42 | fujin: what happened? |
| 20:37.16 | fujin | I did some work for a guy in here |
| 20:37.24 | fujin | writing a perl autodialer and configuring a few asterisk systems |
| 20:37.31 | fujin | and he transferred me about 1000 usd |
| 20:37.35 | fujin | from two accounts |
| 20:37.40 | fujin | paypal froze both |
| 20:37.59 | kand | [TK]D-Fender: you are correct about the M option, I have something similar. I also us MACRO_RESULT to controll the original dial |
| 20:38.10 | jhb | [TK]D-Fender, ok, will try to do the Local&Local and see what happens (and learn and understand) |
| 20:38.34 | holiday_42 | fujin: but why did paypal freeze them? Did the guy use accounts that were not his or something? |
| 20:38.41 | fujin | It'd appear so, yeah. |
| 20:38.45 | fujin | they believe they're fraudulent. |
| 20:41.02 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 20:41.06 | hmmhesays | oh boy I haven't install mysql in forever |
| 20:41.29 | fujin | apt-get install mysql-server-5? |
| 20:41.30 | fujin | ;] |
| 20:41.44 | hmmhesays | doing the initial config lol |
| 20:42.50 | jhb | [TK]D-Fender, not finished trying, thinking: but if I do Local&Local, I have not way to hang them up manually before they connect? |
| 20:43.03 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 20:43.10 | errr | is it possible to use patterns when using realtime mysql?? |
| 20:43.28 | [TK]D-Fender | jhb: clairfy what you mean by "manually". |
| 20:43.49 | jhb | [TK]D-Fender, I mean, if I call a and b: a takes, b should be hung up. a denies, b should be able to take the call. if a does not answer, b should be able to take the call |
| 20:43.59 | [TK]D-Fender | jhb: Each end should stay up till acknowledged by 1 end, and the other dropped when one does. |
| 20:44.16 | [TK]D-Fender | jhb: Taht should work by default |
| 20:45.11 | jhb | [TK]D-Fender, its the second case: a takes the call, but presses 2 to deny it, not wanting to talk |
| 20:45.27 | jhb | [TK]D-Fender, sorry, forget that |
| 20:45.46 | [TK]D-Fender | jhb: Should work fine, just go try it :) |
| 20:46.06 | jhb | [TK]D-Fender, will do |
| 20:46.10 | jhb | [TK]D-Fender, thx |
| 20:47.59 | *** join/#asterisk harpo_marx (n=harpo_ma@78-0-134-91.adsl.net.t-com.hr) |
| 20:48.41 | mocker | fujin: That sucks |
| 20:48.50 | fujin | indeed |
| 20:49.08 | fujin | I hope paypal unfreeze it all before christmas |
| 20:49.10 | fujin | so I can withdraw it out |
| 20:49.14 | mocker | Who was it, so we can all watch out? |
| 20:49.24 | holiday_42 | yeah |
| 20:49.34 | jhb | [TK]D-Fender, I get a Unable to request channel Local/004952144694640@de-calling |
| 20:49.34 | jhb | Local/00447726761210@de-calling |
| 20:49.36 | mocker | any maybe an op can /mode +b :) |
| 20:49.37 | holiday_42 | not that I do anything worth any $ :) |
| 20:49.37 | *** join/#asterisk apocn (n=htejeda@unaffiliated/apocn) |
| 20:49.44 | fujin | He was on the username 'andrieu_x' |
| 20:49.59 | *** join/#asterisk _ys (n=yuri@80.70.236.69) |
| 20:50.00 | jhb | [TK]D-Fender, did 'Local/004952144694640@de-calling\nLocal/00447726761210@de-calling\n' |
| 20:50.09 | fujin | (03:45:56) • Joins: andreiu_x:#asterisk (n=andreiu_@84.126.96.217.dyn.user.ono.com) |
| 20:50.10 | fujin | (13:20:56) • Quits: andreiu_x (n=andreiu_@84.126.96.217.dyn.user.ono.com) : [ ] |
| 20:50.15 | holiday_42 | ah |
| 20:50.16 | jhb | [TK]D-Fender, also adding a & does not do it |
| 20:50.43 | apocn | is it possible to have more than one periodic-announce? for example I want the users while in queue hear 4 comercials one after the other. |
| 20:51.07 | fujin | apocn: use MoH? |
| 20:51.13 | holiday_42 | that ip belongs to an amsterdam ip netblock, not that you can really rely on it tho |
| 20:51.26 | fujin | holiday_42: he claimed to be in spain |
| 20:51.51 | apocn | fujin : Im using MoH for normal music playing in background. |
| 20:52.08 | apocn | but I want to "announce" some stuff using time intervals |
| 20:52.18 | fujin | then no, I don't believe it's possible |
| 20:52.24 | fujin | multiple periodic announces I mean |
| 20:52.28 | apocn | like every 10 seconds to play a new stuff |
| 20:52.29 | apocn | ok |
| 20:52.30 | holiday_42 | ripe.net says spain |
| 20:52.56 | [TK]D-Fender | jhb: 'Local/004952144694640@de-calling\nLocal/00447726761210@de-calling\n' <-- missing & |
| 20:53.36 | blitzrage | [TK]D-Fender: you mean /n |
| 20:53.36 | fujin | holiday_42: probably serves me right for freelancing off IRC |
| 20:54.05 | apocn | fujin : maybe only with this patch: http://bugs.digium.com/view.php?id=6681 |
| 20:54.09 | holiday_42 | no, i don't think you did anything wrong. you're the victim, not the culprit |
| 20:54.25 | holiday_42 | but maybe use a separate paypal accout for freelance work? |
| 20:54.29 | fujin | heh |
| 20:54.30 | [TK]D-Fender | blitzrage: I was just cut& pasting his line to highly the first error. I was going to let himcome to the others in priority sequence :p |
| 20:54.31 | fujin | yeah; |
| 20:54.33 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 20:54.39 | [TK]D-Fender | highlight* |
| 20:54.41 | holiday_42 | i'll have to keep that in mind myself |
| 21:00.35 | Yourname`` | [TK]D-Fender: LOL paypal is the shit! |
| 21:01.46 | [TK]D-Fender | Yourname``: Its how I'm payed for most of my contracting work, how I pay my internet bill, and even that custom sword I commissioned :) |
| 21:02.17 | hmmhesays | oh mysql is so foreign |
| 21:02.17 | holiday_42 | o_O |
| 21:03.22 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 21:04.42 | Yourname`` | As long as you don't use the sword on me! :D I think I should keep your paypal addr with me just in case. |
| 21:04.57 | fujin | keep mine too |
| 21:05.20 | Yourname`` | l0olol |
| 21:05.41 | Yourname`` | come ye come all.. its xmas 2008 |
| 21:09.21 | Qwell | Yourname`: where's my gift? |
| 21:09.52 | Qwell | s/gift/bribe/ |
| 21:09.59 | Qwell | jbot said it - not me |
| 21:11.55 | Qwell | outtolunc: it just means he owes us gifts for 2 years |
| 21:12.00 | Qwell | s/gifts/bribes/ |
| 21:12.02 | outtolunc | sweetness |
| 21:12.34 | outtolunc | i want a n810 |
| 21:15.44 | fujin | mm |
| 21:15.46 | fujin | n810 looks nice. |
| 21:16.05 | *** join/#asterisk pirulo (n=andres_p@65.19.28.123) |
| 21:17.24 | Qwell | I want like an E61 or something |
| 21:17.34 | fujin | oh man |
| 21:17.36 | fujin | that n810 is awesome |
| 21:17.39 | fujin | i have a jasjam right now |
| 21:17.40 | fujin | htc tytn |
| 21:18.08 | fujin | although it's surprising to me that the n810 runs Maemo |
| 21:19.00 | outtolunc | Qwell: only 64meg of user space |
| 21:19.05 | Yourname`` | You guys are going to drown my bank! |
| 21:19.08 | Qwell | so? |
| 21:19.11 | outtolunc | hehe |
| 21:19.25 | Qwell | that's 63.5mb more than my current phone |
| 21:19.28 | outtolunc | i can barely live with the extra 1g my n770 has <G> |
| 21:19.47 | fujin | outtolunc: dude! it doesn't have a phone |
| 21:19.49 | Yourname`` | Everyone here should have their paypal address on a webpage so they can be 'thanked' via paypal. So, people can 'thank' their helpers just by going to that webpage. :D |
| 21:20.19 | outtolunc | fujin: i already have a 'phone' |
| 21:20.19 | Yourname`` | [TK]D-Fender: My friend got an HTC touch, don't get it! The touch sensitivity on it sucks currently. |
| 21:20.34 | fujin | two devices vs. 1 = fail |
| 21:20.35 | Yourname`` | It also makes calls when you 'slide' the phone into your pocket, lol |
| 21:20.52 | [TK]D-Fender | Yourname``: I'm getting the CDMA version because of my carrier, never heard anything negative about it. |
| 21:21.10 | [TK]D-Fender | Yourname`and thats what the "off" button is for :) |
| 21:21.17 | outtolunc | thats the other thing i want, 1yr unlimited data |
| 21:21.27 | Yourname`` | [TK]D-Fender: I'm talking about the one from TELUS. |
| 21:21.30 | [TK]D-Fender | outtolunc: $7md unlimited :) |
| 21:21.44 | [TK]D-Fender | mo* |
| 21:21.51 | outtolunc | eh? |
| 21:22.05 | Yourname`` | Nothing is ever unlimited. Just like Gafachi. They say you have unlimited, and when you truly go unlimited, they slap fees left and right. |
| 21:23.08 | [TK]D-Fender | Yourname`Lets say "exremely reasonable" then :) |
| 21:23.25 | [TK]D-Fender | Yourname``: And sane amount of browsing & e-mail = fine |
| 21:23.29 | Yourname`` | Yes.. |
| 21:24.44 | outtolunc | unlimited = you want fries with that |
| 21:25.33 | [TK]D-Fender | Either way, having most standard browsing, Google Maps, Youtube on demand = me happy |
| 21:26.15 | outtolunc | for $7/mo ? <G> |
| 21:26.38 | outtolunc | i wanna play <G> |
| 21:30.33 | [TK]D-Fender | alrighty, check-out time, heading home... |
| 21:37.01 | *** join/#asterisk Qwell_ (i=north@pdpc/sponsor/digium/Qwell) |
| 21:37.01 | *** mode/#asterisk [+o Qwell_] by ChanServ |
| 21:40.40 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 21:43.33 | lirakis | ok.. gtg later all |
| 21:43.43 | *** part/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 21:44.17 | *** join/#asterisk cesar_CR (n=cesar@201.192.86.6) |
| 21:46.09 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 21:46.29 | *** part/#asterisk annedonaldson17 (n=ryan@magic.skylab.org) |
| 21:46.57 | *** join/#asterisk |dennis| (n=dennis@200.32.236.18) |
| 21:47.10 | jameswf | wink |
| 21:47.30 | jameswf | ~wink |
| 21:47.31 | jbot | ACTION winks at jameswf |
| 21:47.32 | mvanbaak | n/3 |
| 21:47.43 | jameswf | ~human |
| 21:47.44 | jbot | rumour has it, human is the mind of an angel in the body of an animal |
| 21:49.28 | *** join/#asterisk dbtid (i=qjgns0eb@cpe-71-72-252-171.columbus.res.rr.com) |
| 21:49.55 | holiday_42 | ~pinky |
| 21:49.56 | jbot | And what are we going to do tomorrow night, Pinky? |
| 21:49.59 | holiday_42 | :) |
| 21:50.03 | dbtid | hello; i understand asterisk 1.4.x will run on the wrt54gl. the links on the voip-info.org site aren't clear as to WHERE to get the distribution to do this. |
| 21:50.05 | holiday_42 | ~brain |
| 21:50.06 | jbot | brain is probably a wonderful organ; it starts working the moment you get up |
| 21:50.14 | holiday_42 | ~the brain |
| 21:50.15 | dbtid | can someone point me to a clear source?? |
| 21:50.20 | dbtid | (please) |
| 21:50.38 | dbtid | "Asterisk 1.4.x is also available from the OpenWRT forums and the link above." (which link???) |
| 21:50.48 | fujin | go to the openwrt forums? |
| 21:50.49 | mocker | dbtid: Might check the unslung stuff. |
| 21:50.50 | holiday_42 | the generic broadcom |
| 21:53.11 | holiday_42 | gah, sorry dbtd, i was thinking dd-wrt |
| 21:53.14 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net) |
| 21:53.57 | holiday_42 | http://www.dd-wrt.com/dd-wrtv2/downloads.php is dd-wrt.v24_voip_generic.bin <-- is that running asterisk? |
| 21:54.05 | hmmhesays | let me know when i've done wrong, and iv'e know this all along |
| 21:55.15 | *** join/#asterisk techie (n=techie@adsl-76-247-10-106.dsl.lsan03.sbcglobal.net) |
| 21:57.11 | *** join/#asterisk techie (n=techie@adsl-76-247-10-106.dsl.lsan03.sbcglobal.net) |
| 21:57.13 | *** part/#asterisk techie (n=techie@adsl-76-247-10-106.dsl.lsan03.sbcglobal.net) |
| 21:58.17 | *** join/#asterisk tobias (n=tobias@user-0ce2hr3.cable.mindspring.com) |
| 21:58.37 | Yourname`` | Now this is a very interesting thing for you guys. When I get a call to a DID that's puts the call in a queue, somehow the agents are getting thrown "off" the queue. And the phones (Vertical 9133i) says "Service not available". |
| 21:58.40 | Yourname`` | What's that? |
| 21:59.35 | Yourname`` | And when I do show queues, I see the agent that reports a "Service not available" as "Unavailable" on the show queues.. |
| 22:03.54 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 22:04.12 | hardwire | dookie or ducky? |
| 22:04.56 | *** join/#asterisk remmo (n=junk@203.32.47.250) |
| 22:11.17 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 22:13.51 | *** join/#asterisk fbnts (n=thomas@host86-141-143-173.range86-141.btcentralplus.com) |
| 22:15.28 | holiday_42 | hmmhesays: what are you rambling on about? is that a line from a song? |
| 22:16.00 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 22:16.01 | fbnts | hi, I have just recompiled Asterisk but now when calling in and asterisk runs playback() i hear nothing. Any ideas? |
| 22:16.45 | mocker | fbnts: Error? |
| 22:17.29 | fbnts | nope, the console looks normal - running through the dialplan and logging the playback command |
| 22:18.06 | fbnts | If I call another number in that just simply dials a SIP phone in the office i hear the ringing tones but I presume that my telco playing them to me |
| 22:22.32 | [TK]D-Fender | In Soviet Russia, file deletes YOU! |
| 22:22.35 | fbnts | just doubled checked: Executing [0@ivr:4] Playback("IAX2/********-1", "ivr-welcome") in new stack |
| 22:22.53 | fbnts | <IAX2/***********-1> Playing 'ivr-welcome' (language 'en') |
| 22:23.11 | fbnts | strange, theres no error |
| 22:24.40 | fbnts | ahh, the reason I recompiled was to enable meetme after installing ztdummy |
| 22:24.50 | fbnts | if I rmmod ztdummy it plays fine |
| 22:25.13 | fbnts | if i then quit and do modprobe ztdummy and start asterisk again its silent again |
| 22:33.30 | [hC] | Heres an interesting question for anyone who knows anything about how asterisk does codec selection.. |
| 22:34.29 | [hC] | If i have an IAX (or SIP) trunk that is forced to g729, and i have an FXO port in some card (which wants to speak ulaw) -- If i put deny=all allow=g729 allow=ulaw in my sip.conf for the desk phones, will asterisk pick according to what codec it needs to speak (and not transcode?) |
| 22:34.51 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 22:34.53 | [hC] | Say, if a call comes in the g729 IAX trunk to desk phone, it will use g729... but if a call comes in (or goes out) the FXO port, it will just 'pick' ulaw, instead of transcoding? |
| 22:35.14 | [TK]D-Fender | [hC], if your itsp says G.729 only, then thats it. |
| 22:35.38 | [hC] | [TK]D-Fender: i am the itsp, i'm trying to figure out how to avoid transcoding on sites with small cpu (soekris) but also have analog ports. |
| 22:35.41 | *** join/#asterisk dijungal (n=kdaniel@205.244.149.157) |
| 22:35.44 | dijungal | hi guy |
| 22:35.57 | [TK]D-Fender | [hC], then no G.729. |
| 22:36.30 | [hC] | [TK]D-Fender: im trying to figure out if asterisk is smart enough so that if a phone in sip.conf is told that it allows both g729 and ulaw, if it will pick g729 when trying to make a call out an IAX2 trunk that is set to use g729, and will also pick ulaw when trying to call out the FXO port (zap) |
| 22:36.41 | [hC] | [TK]D-Fender: rather than forcing the phone to a particular codec and needing to transcode. |
| 22:37.22 | [hC] | If i understand how SIP works, it should negotiate the best codec and pick the right one based on which technology the caller is dialing out on. |
| 22:37.35 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 22:37.42 | dijungal | this is not an asterisk question but i'm hoping someone can help me with a polycom provisioning... I have a windows dhcp server on my network and just configured a linux dhcp server for my polycom phones, but the phone keeps getting it's ip and and settings from the windows dhcp server. How do i make the linux dhcp server take precedence for the mac addresses of the phones??? |
| 22:37.42 | hmmhesays | yeah should |
| 22:37.44 | [TK]D-Fender | [hC], as long as Zap never needs to hit G.729, disallow=all, allow=ulaw, allow=g729, might do |
| 22:38.02 | [hC] | [TK]D-Fender: yeah thats what im asking... that exact case. Zap and IAX will never cross |
| 22:38.20 | jameswf | 2 dhcp servers on 1 domain is bahbahbahdddd |
| 22:38.21 | [TK]D-Fender | dijungal, You can't run 2 DHCP servers on the same network segment. Thats nuts |
| 22:38.23 | [hC] | [TK]D-Fender: its only Joe Deskuser calling NXXNXXXXXX and it either goes out IAX(g729) or Zap(ulaw) |
| 22:38.37 | [TK]D-Fender | [hC], try as I suggested and see |
| 22:38.45 | dijungal | they're on diff subnets |
| 22:38.54 | [hC] | [TK]D-Fender: yeah thats what im just gonna do now. Thanks. just wanted some affirmation. |
| 22:39.04 | JT | [hC]: was that just wishful thinking, sip choosing the best codec? |
| 22:39.11 | Yourname`` | [TK]D-Fender: http://pastebin.ca/813990 |
| 22:39.12 | dijungal | [TK]D-Fender: how do u segment the network? |
| 22:39.14 | jameswf | the computers dont have subnets untell the DHCP server assigns |
| 22:39.35 | jameswf | ~arp |
| 22:39.35 | jbot | hmm... arp is the address resolution protocol, which converts IP addresses to MAC addresses |
| 22:39.46 | dijungal | uhuh |
| 22:40.16 | [TK]D-Fender | Yourname``, networking issues clearly |
| 22:40.20 | JT | dijungal: you're crazy, as [TK]D-Fender said |
| 22:40.26 | lesouvage | What does <unowned> mean when doing a local show channels: asterisk -rx "local show channels" <unowned> -- 316@default-agent |
| 22:40.33 | JT | dijungal: you cannot run 2 DHCP servers on the same segment |
| 22:40.40 | JT | at a minimum you need VLANs |
| 22:40.40 | Qwell | JT: sure you can |
| 22:40.48 | dijungal | JT: once again.. how do u segment the network??? |
| 22:40.52 | jameswf | dijungal, set up 1 DHCP server with two buckets and assign IPs based on MAC addresses |
| 22:40.56 | JT | VLANs, different switches |
| 22:40.58 | Yourname`` | [TK]D-Fender : lol, it usually happens when someone ELSE sends those calls to us. But when we dial, and send the resultant call to the queue it's all good. |
| 22:41.07 | JT | dijungal: make them be different networks. |
| 22:41.28 | JT | Qwell: sure you can... |
| 22:41.59 | jameswf | I assume you used webmin on the linux box... read up it will allow you to set this up windows style ( without thinking) |
| 22:42.09 | *** join/#asterisk switched (i=juanchic@oj.dreamhost.com) |
| 22:42.21 | dijungal | no webmin on the box |
| 22:42.26 | dijungal | all cli |
| 22:42.43 | jameswf | perhaps a network admin is in order.... |
| 22:42.55 | switched | when i do 'pri show span 1' i get - "no PRI running on span 1" ... now I know that Verizon hasn't "totally" turned on the PRI yet, but shouldn't the D channel be active? |
| 22:43.02 | dijungal | he's out of office... and i can't get him on the phone |
| 22:43.28 | [hC] | switched: if they havent turned up the pri, the dchannel will most likely be dead. |
| 22:43.30 | switched | but then again, if asterisk can't see the PRI itself, then I can't see D channel - so either asterisk is misconfigured, or there really isn't a PRI setup? |
| 22:43.39 | switched | aw wtf |
| 22:43.41 | JT | dijungal: the easiest way to run 2 DHCP servers is to use 2 physically seperate networks (different switches), or 2 virtually seperated networks (different VLANs) |
| 22:43.51 | jameswf | switched, if your settings are correct pri show span1 will say down your settings are incorrect |
| 22:44.11 | switched | oh well, learning experience i guess. i was under the impression that they can set it so D channel is up, but not the rest of them? |
| 22:44.22 | switched | or maybe i misunderstood them |
| 22:44.33 | Qwell | switched: sure, they *could*, but this is verizon we're talking about |
| 22:44.36 | JT | switched: they can set it up so the D channel is up but calls don't work |
| 22:44.37 | jameswf | swiched your not configured for pri |
| 22:44.44 | jameswf | nothing to do with them |
| 22:44.46 | JT | switched: but they are all just timeslots |
| 22:44.56 | JT | so the whole link is either up or not |
| 22:45.59 | switched | i think i've checked multiple places: 1) /etc/zaptel.conf looks right 2) /etc/asterisk/zapata-channels.conf looks right 3) /etc/asterisk/modules.conf looks right (had to had chan_zap.o to it) |
| 22:46.17 | jameswf | pastebin zaptel and zapata |
| 22:46.33 | jameswf | swiched do you have libpri |
| 22:46.39 | switched | for some reason i had a hard time figuring out i had to add chan_zap in - just that I know I didn't have the pri command in the asterisk CLI |
| 22:46.43 | switched | yeah I got libpri installed |
| 22:46.48 | switched | k imma pastebin those |
| 22:48.17 | switched | my zapata: http://pastebin.com/m5add2c6a and my zaptel.conf: http://pastebin.com/m375bdc3d |
| 22:48.58 | switched | also, my digium card has a amber light - which to me is good - better than no light. according to docs, means it can't see the far end. |
| 22:49.12 | switched | which makes sense - the pri isn't totally live yet |
| 22:49.15 | JT | span=1,1,1,esf,b8zs should probably be span=1,1,0,esf,b8zs not that it will really make a difference |
| 22:49.27 | jameswf | try span=1,1,0,esf,b8zs |
| 22:49.37 | *** join/#asterisk Beirdo (n=gjhurlbu@unaffiliated/beirdo) [NETSPLIT VICTIM] |
| 22:49.43 | switched | you mean the LBO setting? |
| 22:49.50 | JT | yes |
| 22:49.55 | switched | but isn't the CSU ..built into the digium card? |
| 22:50.06 | switched | er wait i mean then zero it should be. |
| 22:50.12 | switched | so.. genzaptelconf sometimes is wrong? |
| 22:50.32 | JT | umm, why has your zapata.conf got zaptel.conf stuff in it? or did you pb wrong? |
| 22:50.37 | jameswf | genzaptelconf is 50/50 you should tweak the final output |
| 22:50.55 | *** join/#asterisk remmo (n=junk@203.32.47.250) [NETSPLIT VICTIM] |
| 22:50.55 | *** join/#asterisk ectospasm (n=ectospas@c-68-62-219-11.hsd1.al.comcast.net) [NETSPLIT VICTIM] |
| 22:51.21 | switched | JT: sorry, I pb'd wrong, so lines 1-20 are the zaptel, the rest is a cat of the zapata.conf |
| 22:51.48 | jameswf | I see zapata-channels and zaptel no zapata |
| 22:51.50 | JT | oh which one |
| 22:52.05 | jameswf | /etc/asterisk/zapata.conf |
| 22:52.13 | switched | i thought zapata-channels is the same? i have no zapata.conf |
| 22:52.17 | JT | ... |
| 22:52.19 | switched | er.. i guess i should make one. lol |
| 22:52.22 | jameswf | there is the issue |
| 22:52.24 | JT | then how will asterisk read the configuration? |
| 22:52.42 | JT | zapata-channels is non standard |
| 22:52.56 | switched | ok - thing is.. if you don't have zapata.conf does that get logged somewhere? |
| 22:53.06 | jameswf | zapata-channels should be a #include |
| 22:53.14 | switched | in my trixbox machine, everything goes to /var/log/asterisk/full |
| 22:53.19 | JT | ... |
| 22:53.22 | Qwell | ~trixbox |
| 22:53.23 | jbot | [~trixbox] Trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org |
| 22:53.24 | JT | trixbox is not asterisk |
| 22:53.24 | switched | on my current test machine |
| 22:53.41 | switched | er.. yeah - on my current non-trixbox test box (asterisk by source) - i dunno where the logs are |
| 22:53.48 | switched | maybe i misconfigured it |
| 22:53.51 | jameswf | probably missed make configs... |
| 22:54.01 | switched | oh goddammit that makes sense. |
| 22:54.04 | *** join/#asterisk etix (n=etix@ram94-3-82-224-49-128.fbx.proxad.net) |
| 22:54.09 | switched | i blew through it being lazy |
| 22:54.33 | jameswf | I script my asterisk/zaptel/libpri... installs |
| 22:54.43 | jameswf | that is lazy :) |
| 22:54.52 | switched | ok i was being stupid then |
| 22:55.01 | lesouvage | No one familiar with "unknowned channels"? |
| 22:55.22 | jameswf | ~unknowned |
| 22:55.32 | jameswf | got me |
| 22:55.37 | Qwell | ~unknowned channels |
| 22:56.03 | jameswf | jbot unknowned is WTF? |
| 22:56.05 | jbot | jameswf: okay |
| 22:56.08 | jameswf | ~unknowned |
| 22:56.09 | jbot | unknowned is, like, WTF? |
| 22:56.13 | jameswf | lol |
| 22:56.20 | lesouvage | I googled but nothing usefull showed up |
| 22:57.02 | *** join/#asterisk optize (i=tyler@ip70-176-254-41.ph.ph.cox.net) |
| 22:57.32 | optize | I have this one customer, who sends us an INVITE (while he's registered) we send back a 'Proxy Auth Required', he re-sends the INVITE, and then I came back with '403 Forbidden' What would cause that? Him not senidng auth details? |
| 22:57.34 | fujin | unknownennnnedddd |
| 22:57.37 | optize | I think he is, I have no idea what the issue is. |
| 22:57.46 | jameswf | unknowned.com is a spam site.... |
| 22:58.05 | JT | lesouvage: possibly because there is no such english word as unknowned. |
| 22:58.19 | Qwell | JT: past tense verbed |
| 22:58.25 | jameswf | psh english... who speaks that crap |
| 22:58.51 | Qwell | definition: "Made to be unknown" |
| 22:58.59 | lesouvage | And it is actually not that funny because it causing real problems in a production envirement. |
| 22:59.10 | JT | lesouvage: and yet there is still no such word. |
| 22:59.11 | jameswf | I would like to formally reintroduce the word chode to the daily vocabulary... |
| 22:59.13 | Qwell | lesouvage: how about you explain the problem? |
| 22:59.40 | jameswf | pastebin the error |
| 22:59.58 | jameswf | maybe an asterisk dev was drunk one day |
| 23:00.44 | jameswf | ~app_monkeys |
| 23:01.00 | Qwell | res.. |
| 23:01.05 | jameswf | doh |
| 23:01.06 | lesouvage | jameswf: <unowned> -- 316@default-agent Yes I wrote it wrong, its unowned and not unknowned |
| 23:01.12 | jameswf | thats funny as hell :) |
| 23:01.13 | Qwell | un OWNED |
| 23:01.14 | Qwell | :p |
| 23:01.28 | Yourname`` | I sersly hate these Vertical 9133is (rebranded from Aastra). |
| 23:02.05 | fujin | get new phones |
| 23:02.07 | fujin | && doen |
| 23:02.09 | fujin | profit++ |
| 23:02.09 | fujin | .; |
| 23:02.14 | jameswf | agent:excuse me sir did you hear the screaming monkeys: caller: What? |
| 23:02.33 | fujin | hahaha |
| 23:02.35 | hardwire | haw |
| 23:02.36 | Yourname`` | lolol |
| 23:03.06 | Yourname`` | fujin: Looking for a Polycom IP330 provider. Got a couple quotes so far, but none offer good discount for bulk orders, lol. |
| 23:03.18 | fujin | ip330's cost me $250nzd~ |
| 23:03.45 | jameswf | maybe replace tt-monkeys with a whisper of "kill, kill em all" |
| 23:04.31 | lesouvage | The problem is that there is a strange behaviour of queues. When an agents tranfer the call to another extension and the agents is freed for other calls after the call he transfered has ended. The call isn't tranfered but bridged. When examining the logs I found the unowned message that might be part of the problem or an indication of the problem. |
| 23:04.50 | jameswf | have allison do subliminal messages for crazy people/// |
| 23:05.40 | [TK]D-Fender | jameswf, No need... crazy people already imagine hearing Allison say all sorts of things.... |
| 23:05.58 | jameswf | I know I do lol |
| 23:06.02 | Yourname`` | fujin: Costing me $109.99 each for 50 phones. |
| 23:06.17 | fujin | usd? |
| 23:06.23 | Yourname`` | Yeah |
| 23:06.26 | fujin | that's about right man.. |
| 23:06.30 | fujin | get what you pay for though ;) |
| 23:06.34 | Yourname`` | I want it to be more like $70 |
| 23:06.53 | jameswf | s/hi/his/ |
| 23:07.31 | fujin | i'm told the ip330 |
| 23:07.35 | fujin | are much better than 501 |
| 23:07.39 | fujin | if not only for PoE support :P |
| 23:08.19 | *** join/#asterisk Qwell_ (i=north@pdpc/sponsor/digium/Qwell) |
| 23:08.19 | *** mode/#asterisk [+o Qwell_] by ChanServ |
| 23:10.27 | *** join/#asterisk AJaymn (n=mypocket@71-82-218-158.dhcp.mdsn.wi.charter.com) |
| 23:10.30 | *** join/#asterisk anthm (n=anthm@mbf0736d0.tmodns.net) |
| 23:10.30 | *** mode/#asterisk [+o anthm] by ChanServ |
| 23:11.41 | fbnts | Has anyone had a problem that when the ztdummy module is loaded, applications like playback() and musiconhold() don't play any sound? |
| 23:11.49 | *** join/#asterisk craigk (n=ckowald@58.174.150.119) |
| 23:11.59 | AJaymn | Has anyone found a decent Billing system other then A2billing for Asterisk? I need to be able to bill monthly, and min usage of tollfree service |
| 23:12.35 | Yourname`` | AJaymn: http://tinyurl.com/37u6vx |
| 23:13.05 | AJaymn | Thanks |
| 23:13.28 | JT | fujin: that's crazy, the 501s are clearly better |
| 23:13.44 | fujin | bigger screen eh? |
| 23:13.48 | fujin | i haven't used either :) |
| 23:13.58 | *** join/#asterisk barhom (n=barhom@h-89-233-196-214.wholesale.rp80.se) |
| 23:14.07 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 23:14.38 | barhom | how do I match all non-existing extens? Ive tried s,i,h nothing is working as expected.. I would like to not use "_." |
| 23:15.09 | jameswf | t |
| 23:15.12 | lesouvage | I understand that "<unowned> -- 316@default-agent is a rare message". No one out there who could give an explanation or give some direction what causes this message. |
| 23:15.19 | barhom | thanks james, Ill try |
| 23:15.19 | JT | bigger and better screen, 3 line appearances, line appearances show on screen |
| 23:15.31 | [TK]D-Fender | barhom, You're going to have to. |
| 23:15.33 | JT | fujin: more buttons too |
| 23:15.46 | fujin | handy |
| 23:15.57 | jameswf | that sells me I like buttons |
| 23:16.43 | [TK]D-Fender | All hard buttons on Polycom phones that have soft-keys are wasted |
| 23:17.05 | [TK]D-Fender | IP 330 = much cheaper. In most cases I'd suggest the IP 330 over the 501. |
| 23:17.29 | [TK]D-Fender | IP 501 has a very limited scope of suggestability in their lineup |
| 23:17.59 | jameswf | ~!~ |
| 23:18.00 | fujin | hard buttons are bertter for most users (read: idiots) |
| 23:18.10 | jameswf | ~idiots |
| 23:18.11 | jbot | so far only bleck |
| 23:18.21 | outtolunc | lesouvage: look in res_features of whatever you are using |
| 23:19.16 | lesouvage | outtolunc: what should I look for? |
| 23:19.22 | [TK]D-Fender | fujin, Idiots will try hitting hard-button when they're not allowed. |
| 23:19.30 | outtolunc | didn't you ask where the 'unowned' came from? |
| 23:19.32 | fujin | that's true, also |
| 23:19.43 | outtolunc | or i misunderstood |
| 23:20.48 | lesouvage | outtolunc: yes i'm looking for an explanation for the unowned message. |
| 23:21.33 | *** join/#asterisk pirulo (n=andres_p@65.19.28.123) |
| 23:22.03 | *** join/#asterisk techie (n=techie@adsl-76-247-10-106.dsl.lsan03.sbcglobal.net) |
| 23:22.08 | *** part/#asterisk techie (n=techie@adsl-76-247-10-106.dsl.lsan03.sbcglobal.net) |
| 23:23.25 | *** join/#asterisk pirulo (n=andres_p@65.19.28.123) |
| 23:27.45 | jameswf | ~fuck |
| 23:27.46 | jbot | Now where did I put the lube...? Eh, no matter, dry it is tonight! |
| 23:29.54 | *** join/#asterisk andreiu_x (n=andreiu_@2.128.219.87.dynamic.jazztel.es) |
| 23:30.00 | *** part/#asterisk andreiu_x (n=andreiu_@2.128.219.87.dynamic.jazztel.es) |
| 23:30.32 | [TK]D-Fender | lol |
| 23:30.37 | [TK]D-Fender | ~sex |
| 23:30.38 | jbot | updatedb; locate; talk; date; cd; strip; look; touch; finger; unzip; uptime; gawk; head; apt-get install condom; mount; fsck; gasp; more; yes; yes; yes; more; umount; apt-get remove --purge condom; make clean; sleep, or super extractor, http://sf.net/projects/sex/ |
| 23:30.40 | switched | in zapata.conf what's the group setting for? |
| 23:30.49 | JT | the zaptel group |
| 23:30.50 | [TK]D-Fender | ~wikis |
| 23:30.51 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 23:30.52 | [TK]D-Fender | ^^^^^^^^^^ |
| 23:31.20 | *** join/#asterisk Adolph-testing (n=andreiu_@2.128.219.87.dynamic.jazztel.es) |
| 23:31.26 | Adolph-testing | hello |
| 23:31.50 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 23:32.37 | Adolph-testing | i got a voip account who default use ATA device i speak with customer support and they give me a SIP account, i configured asterisk with that account without register => line |
| 23:33.22 | *** join/#asterisk Maliuta (n=nikolai@203.201.152.211) |
| 23:33.41 | switched | k so my zapata.conf is now: http://pastebin.com/m59329045 and zaptel.conf is: http://pastebin.com/m5add2c6a .. restarted zaptel and asterisk and still says no pri detected |
| 23:34.09 | switched | which i guess.. means...the PRI is not there. of course. i just want to make super sure it's not a config problem when we call verizon again |
| 23:34.32 | JT | switched: pri show span 1 |
| 23:34.40 | jameswf | did you restart asterisk |
| 23:34.45 | Adolph-testing | can anyone help me ? |
| 23:34.53 | switched | No PRI running on span 1 |
| 23:35.05 | JT | Adolph-testing: you didn't even say what the problem was |
| 23:35.09 | switched | yeah did a `service asterisk restart` after a `service zaptel restart` |
| 23:35.48 | JT | switched: what happens when you do ztcfg -vv ? |
| 23:35.57 | Adolph-testing | so i configured sip.conf and try to make an call but i get this error > Channel SIP/voiceral-08d58090 was never answered. |
| 23:36.03 | [TK]D-Fender | switched, that had better not be your entire zapata.conf.... |
| 23:36.17 | switched | D-Fender: it can see my card on span #1 |
| 23:36.48 | switched | I thought that would be the minimal I could get away with...but sounds like not. |
| 23:36.49 | Yourname`` | fujin: |
| 23:37.06 | Adolph-testing | when i try to register sip on voip server i recive this error Got 404 Not found on SIP register to service 1651293@64.128.190.111 |
| 23:37.11 | [TK]D-Fender | switched, You telling me THIS is your entire zapata.conf ? -> http://pastebin.com/m59329045 |
| 23:38.09 | alrs | click: SQUAREPUSHER! |
| 23:38.15 | alrs | wrong window, sorry |
| 23:38.34 | Yourname`` | I place a call from a dialer, and when the callee rcvs the call, the call is placed back into a queue.. and the agent of the queue sees the callerid of that callee. THEN, this agent transfers this callee to another queue. But the agents of the other queue only see the callerid of agent that transferred the callee.. how do I get the callerid of the callee? |
| 23:39.02 | switched | D-Fender yes, I was just going off of this http://www.voip-info.org/wiki/view/Asterisk+PRI#etcasteriskzapataconf ... hoping that would be the minimal config I can use to at last successfully see feedback from "PRI debug span 1" |
| 23:39.14 | outtolunc | cat /etc/asterisk/zapata-channels.conf ... intended to be 'included' from zapata.conf .. is it? |
| 23:39.46 | [TK]D-Fender | switched, you don't have a [channels] section header in there and "zapata-channels.conf" is not used ANYWHERE in your setup. Fix your header |
| 23:40.10 | Adolph-testing | JT any ideea what to try to search to ask voip provider supprt ? |
| 23:40.14 | [TK]D-Fender | outtolunc, No, thats broken leftovers. |
| 23:40.37 | outtolunc | why is it always get the broken-leftover shit <G> |
| 23:40.42 | outtolunc | sheesh |
| 23:41.44 | *** join/#asterisk ExplodingLemur (n=change@72.29.166.30) |
| 23:42.38 | fujin | Yourname`: set it into a __variable |
| 23:42.46 | fujin | one of those magical ones that doesn't disappear, even down a macro chain |
| 23:42.57 | *** join/#asterisk macros73_ (n=cs@c-24-3-246-27.hsd1.pa.comcast.net) |
| 23:43.02 | fujin | then before sending the call to someone else, re-set callerid to that variable |
| 23:43.42 | [TK]D-Fender | Adolph-testing, register error tells you you're using the wrong username |
| 23:43.43 | jameswf | http://www.voip-info.org/wiki/index.php?page=Asterisk+config+zapata.conf.sample |
| 23:44.36 | JT | Adolph-testing: |
| 23:44.39 | JT | ~question |
| 23:44.40 | jbot | well, question is If you have a question and want people to give useful answers, make sure you have read this first: http://www.catb.org/~esr/faqs/smart-questions.html |
| 23:44.57 | *** join/#asterisk codefreeze (i=steve_mu@nat/digium/x-298af643db995eb0) |
| 23:44.57 | *** mode/#asterisk [+o codefreeze] by ChanServ |
| 23:45.02 | *** join/#asterisk etix (n=etix@2a01:5d8:52e0:3180:216:4242:cafe:cafe) |
| 23:45.27 | jameswf | sed '/^\;/d' zapata.conf.sample > zapata.conf |
| 23:45.54 | [TK]D-Fender | jameswf : or he could just add the single line I told him to. |
| 23:46.23 | jameswf | he probably should read the book |
| 23:46.26 | jameswf | ~book |
| 23:46.27 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 23:47.12 | blitzrage | can someone double check that tfot.leifmadsen.com is available from the outside world? (and not just from my laptop?) :) |
| 23:47.47 | outtolunc | Asterisk™: The Future of Telephony |
| 23:47.50 | jameswf | http://tfot.leifmadsen.comloads |
| 23:47.59 | Qwell_ | comloads? |
| 23:48.03 | Qwell_ | interesting tld |
| 23:48.06 | jameswf | dirty :) |
| 23:48.18 | outtolunc | This book uses RepKover™, a durable and flexible lay-flat binding. |
| 23:48.22 | outtolunc | hehe |
| 23:48.32 | barhom | what is so special with "_", I mean I can write exten => 555,etc and I can write exten => _555,etc - difference? |
| 23:48.57 | *** join/#asterisk gardo (n=gardo@121.97.194.130) |
| 23:49.00 | Adolph-testing | i will check [TK]D-Fender |
| 23:49.12 | outtolunc | _X55,etc |
| 23:49.23 | [TK]D-Fender | barhom, "_" is to indicate a pattern which that ISN'T |
| 23:50.09 | barhom | _ as a pattern, got it. This is quick learning, thanks |
| 23:50.38 | [TK]D-Fender | barhom, exten => _5XX,1, <--- this takes any 3-digit # starting with "5". exten => 5XX,1, <--- this takes EXACTLY "5XX" alphanumeric. |
| 23:51.18 | [TK]D-Fender | barhom, Which means you'd better be able to dial the alphabet.... |
| 23:51.27 | outtolunc | basically, _ means look for swapable wildcard ... such as X or N |
| 23:51.34 | [TK]D-Fender | (only good as an initial dial from a softphone for example) |
| 23:51.46 | Adolph-testing | [TK]D-Fender: the voipral support confirmed, username and password are good |
| 23:51.59 | Adolph-testing | voiceral* |
| 23:52.07 | JT | _ enables pattern matching. |
| 23:52.12 | JT | no _ no pattern matching. |
| 23:52.17 | barhom | what is N outtolunc? |
| 23:52.24 | barhom | numberonly? |
| 23:52.26 | outtolunc | 2-9 iirc |
| 23:52.28 | JT | ~thebook |
| 23:52.29 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 23:53.07 | [TK]D-Fender | Adolph-testing, 404 = I don't know you. |
| 23:53.09 | *** join/#asterisk Olobola (n=sfsdsdfs@74.95.13.57) |
| 23:53.28 | JT | or i don't know the extension you're trying to reach |
| 23:53.41 | Olobola | my call log is 8 hours ahead for some reason |
| 23:53.42 | [TK]D-Fender | JT, we're dealing with a REGISTER here |
| 23:53.52 | JT | ah ok |
| 23:55.16 | JT | Adolph-testing: have you tried using a softphone to the provider instead of asterisk/ |
| 23:55.42 | Adolph-testing | yes |
| 23:55.47 | Adolph-testing | not working |
| 23:55.52 | JT | i see |
| 23:57.14 | Adolph-testing | i sayd to the voiceral support to remove user and password and to try to authenticate by ip address |
| 23:57.20 | Adolph-testing | and still not working |
| 23:57.52 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 23:59.24 | Adolph-testing | and softphone i used Express Talk from NCH software |
| 23:59.26 | outtolunc | man, i'm bored out of my freakin mind |
| 23:59.52 | JT | use xlite |
| 23:59.57 | JT | or ekiga |
