| 00:03.59 | Nate9939 | what function of application would i use to playback a precorded message after a dial () command? when connected, would i use the playback function? |
| 00:06.37 | voiper1 | any one had any experience using asterisk behind a cisco router? Im having a problem were I can register the phone externally but there is no audio. I tested with wireshark and there is no rtp traffic but i keep getting told by the company who manages the router that 10000-20000UDP has been forwarded. Any ideas? |
| 00:06.44 | ManxPower | Nate9939: the dialplan stops until the call is finished |
| 00:07.00 | ManxPower | Nate9939: there are several announcements available to the Dial command. "show application dial" |
| 00:07.14 | ManxPower | voiper1: turn off sip fixup |
| 00:07.28 | ManxPower | no service sip 5060 |
| 00:07.32 | ManxPower | or something like that |
| 00:07.40 | voiper1 | manxpower: within the router you mean? |
| 00:07.47 | ManxPower | voiper1: yes. |
| 00:07.57 | ManxPower | also remember to forward 5060 UDP as well. |
| 00:08.09 | ManxPower | I assume you have localnet and externip set as well, right? |
| 00:08.29 | voiper1 | manxpower: yes localnet, externip, nat=yes and rtp.conf is set correctly |
| 00:08.49 | ManxPower | nat=yes is only for remote devices behind nat |
| 00:09.11 | voiper1 | yep thats what im after as the client wants remote extensions |
| 00:09.19 | Nate9939 | thanks manxpower |
| 00:11.11 | ManxPower | voiper1: many cisco routers have SIP NAT supports and as your mother always told you "two nat fixups make a wrong" |
| 00:11.11 | ManxPower | and if you have asterisk's SIP fixup (localnet, externip, nat=yes, etc) and then the cisco goes thru and screwes that all up when it tries to fixup the packet |
| 00:11.11 | J4k3 | hrm, ever since our local telco switched to GPS trunk timing for their crap, I've been seeing 10-12 second outages on a PTP T1 that runs from another telco... this sucks :| |
| 00:11.11 | *** join/#asterisk ptiggerdine (n=ptiggerd@203-219-14-182.static.tpgi.com.au) |
| 00:11.24 | ManxPower | J4k3: document them, report to telco, then report to the PSC/PUC |
| 00:11.44 | J4k3 | yeah, the documentation is what I'm working on now |
| 00:12.47 | J4k3 | anyone know of a network uptime monitoring app for 'nix thats designed around say, 5 tests per second vs 1 test per 5 minutes? :) |
| 00:12.48 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 00:13.11 | *** join/#asterisk Aces1Up (n=really@ip68-227-41-148.lv.lv.cox.net) |
| 00:13.54 | ManxPower | J4k3: On MANY routers you can tell them to send their logs to syslog, and most of them send interface up / interface down messages |
| 00:14.05 | ManxPower | Then just grep thru the syslog |
| 00:14.21 | J4k3 | ManxPower: unluckily the router isn't documenting the event. |
| 00:14.23 | J4k3 | :| |
| 00:14.25 | ManxPower | remmeber to use the -r option to syslog on the logging host so it will accept remote syslog messages |
| 00:14.42 | J4k3 | I may have the csu/dsu set to "ignore everything and BS the router to think its connected" mode. |
| 00:14.45 | ManxPower | J4k3: then you have a couple of options 1) buy a router that does not suck. |
| 00:14.50 | ManxPower | 2) find another job |
| 00:14.53 | ManxPower | 3) live with it. |
| 00:15.12 | J4k3 | ManxPower: well, I was thinking of using something a little higher level, like ping. |
| 00:15.14 | ManxPower | J4k3: well THAT would be silly, wouldn't it? |
| 00:15.30 | ManxPower | J4k3: telcos really only care about interfaces doing up and down. |
| 00:15.42 | J4k3 | well, if its a timing slip, the interface will never cycle |
| 00:15.54 | CoolGuy21 | ok i have a wakeup script but it doesnt allow me to tell it what number it should call for the wakeup |
| 00:15.56 | J4k3 | and thats what it appears to be... the lights on the csu/dsu never show anything happening |
| 00:15.58 | ManxPower | the timing slip should not make the link go down tiehr |
| 00:16.01 | CoolGuy21 | it automatically uses the extension |
| 00:16.07 | ManxPower | CoolGuy21: looks loike you'll have to modify it. |
| 00:16.15 | CoolGuy21 | i dont know how to code php |
| 00:16.34 | *** part/#asterisk bapril (n=bapril@pool-70-20-40-8.man.east.verizon.net) |
| 00:16.35 | ManxPower | on audio a time slip sounds like a click. on data it should show as a crc or other error on the router. |
| 00:17.47 | ManxPower | J4k3: and you realize that timing on a T-1 has nothing to do with actual time, right. |
| 00:17.54 | ManxPower | it would be better to call it a "sync source". |
| 00:18.25 | ManxPower | think of timing on a T-1 like a timing belt on a car. It just keeps everything in sync and has nothing to to with actual time. |
| 00:19.17 | J4k3 | ManxPower: yeah I know... A few months ago the telco did something completely nutty, breaking the local phone traffic for 3 days (hardcore audio screwups, lost calls, getting other people's calls, it was nuts) to which I wrote a letter to the PUC about. Since that event, the T1's been weird... a few times a day I notice about 10-12 seconds of lag.. ping sees it. |
| 00:19.41 | J4k3 | but... this is a PTP T1 that runs *outside* of this local telco's turf. |
| 00:19.48 | ManxPower | J4k3: so all you really know is that you have weird ping times. |
| 00:19.57 | ManxPower | define that |
| 00:19.58 | J4k3 | ManxPower: correct... |
| 00:20.05 | ManxPower | Point to Point T-1s are all the same. |
| 00:20.18 | ManxPower | If it's not the same, then it's not a point to point T-1. |
| 00:20.32 | J4k3 | going along, traffic goes to zero, data stops moving. most of it appears to buffer |
| 00:20.46 | J4k3 | as I end up with 8-9 second pings, anything longer than that is dropped. |
| 00:21.04 | ManxPower | Like this piece of crap "3Mbps bonded T-1" crap a telco bamboozled the IT manager into buying. They handed me a damn ETHERNET connection. |
| 00:21.35 | J4k3 | so if its pinging once per second... I see normal pings, then it stops scrolling, 10 seconds later I see 8k, 7k, 6k, 5k, 4k, 3k, 2k, 1k msec pings, then it goes back to normal |
| 00:21.48 | J4k3 | the traffic isn't discarded as I'd expect it would be. |
| 00:21.52 | ManxPower | That telco then understood when they say "Bonded T-1" it had better be a bonded T-1 or they get ripped a new one. |
| 00:22.12 | ManxPower | J4k3: you are going to have problems diagnosing this stuff with the router you have. |
| 00:22.30 | J4k3 | yeah. I think I'll attempt to break out an old cisco. |
| 00:23.19 | ManxPower | We have 583275 CRC errors on the T-1 between 11am and 5pm. Telco fix it! |
| 00:23.23 | ManxPower | that is what you can tell them |
| 00:23.31 | J4k3 | hmm, I also have a csu/dsu that keeps loop quality stats.. at least according to the inventory. |
| 00:23.51 | J4k3 | but I think its only monitoring the local loop |
| 00:23.56 | J4k3 | as its just a csu/dsu, not a router. |
| 00:24.41 | ManxPower | the csu/dsu might give you interface up/down and CRC info |
| 00:25.27 | J4k3 | yeah... its an oldold tylink ons400 |
| 00:25.34 | ManxPower | I like my CSU/DSUs like I like my checkout clerk at the store. sumb, quiet, and beige. |
| 00:25.46 | ManxPower | ..er..dumb, quiet, and beige. |
| 00:25.55 | J4k3 | yeah, same here |
| 00:26.01 | J4k3 | hence the stable little kentrox :| |
| 00:26.07 | Aces1Up | man you guys are so cool. |
| 00:26.16 | J4k3 | the tylink has a fan.... a really loud one |
| 00:26.17 | J4k3 | haha |
| 00:26.33 | ManxPower | Aces1Up: I'm an assole. Get it right. |
| 00:26.41 | ManxPower | one that can't type too! |
| 00:26.46 | J4k3 | haha |
| 00:27.14 | J4k3 | assole! (insert little accent mark above the e) |
| 00:29.59 | ManxPower | J4k3: Just remember that the telco NEVER EVER has a problem. |
| 00:30.02 | J4k3 | ooh, this csu/dsu is nice... and loud |
| 00:30.27 | ManxPower | It's just that problems sometimes fix themselves after you report them to the telco |
| 00:30.27 | J4k3 | haha yeah, I've been in the ISP game for a while... the telco is ALWAYS innocent |
| 00:30.27 | J4k3 | yeah |
| 00:30.27 | J4k3 | haha |
| 00:30.46 | J4k3 | Hyundai Motors of America is the same way... you take your car in, they fix it, you sign for service performed, then they act like nothing was broken. |
| 00:30.49 | J4k3 | ;) |
| 00:31.18 | *** part/#asterisk CoolGuy21 (n=77889789@cpe-76-173-56-41.socal.res.rr.com) |
| 00:31.39 | J4k3 | wow, this thing keeps a lot of stats |
| 00:32.24 | ManxPower | I think I'll go socialize. |
| 00:33.13 | J4k3 | wow, this thing has a printer and console port |
| 00:33.52 | J4k3 | of course, its big enough to have an xeon server inside, along with a T1 interface card ;) |
| 00:34.07 | J4k3 | haha... email ;) |
| 00:34.37 | ManxPower | J4k3: if it has a printer port..... |
| 00:34.55 | J4k3 | well, its just serial |
| 00:35.00 | *** join/#asterisk monstertruck (n=monstert@c-75-74-251-82.hsd1.fl.comcast.net) |
| 00:35.01 | J4k3 | I'll cap it |
| 00:36.37 | *** join/#asterisk DaveCanoe (n=Dave@H6.C30.B96.tor.eicat.ca) |
| 00:39.48 | *** join/#asterisk chronos_ (n=chronos@adsl-68-252-236-45.dsl.chcgil.ameritech.net) |
| 00:39.48 | J4k3 | alright, time to insert the battleship. |
| 00:45.30 | acidchild | anyone know any comapnys that offer SMS on pc's with a return address and everything, so you can use your computer as a phone to send txts :/ |
| 00:47.20 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 00:48.29 | J4k3 | acidchild: there are some but you have to pay for it. |
| 00:48.37 | J4k3 | per message. |
| 00:48.52 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.163.72) |
| 00:49.48 | *** join/#asterisk sharp (n=sharp@dsl092-234-217.phl1.dsl.speakeasy.net) |
| 00:49.50 | acidchild | you know the names of them? |
| 00:59.34 | rob0 | I just don't know any reason why I'd answer someone called "acidchild" :) |
| 01:04.22 | tzanger | damn, chan_mobile's segfaulting here |
| 01:06.53 | *** join/#asterisk cambocambo (n=cambo@202.162.177.83) |
| 01:07.38 | cambocambo | hi, quick question, how do i tell which version of asterisk is installed? |
| 01:09.29 | *** join/#asterisk roe_ (n=keith@216-164-160-36.c3-0.eas-ubr10.atw-eas.pa.static.cable.rcn.com) |
| 01:10.04 | cambocambo | hi, quick question, how do i tell which version of asterisk is installed? |
| 01:10.21 | roe_ | asterisk -V |
| 01:11.21 | cambocambo | thank you. |
| 01:14.06 | *** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-118-241.ph.ph.cox.net) |
| 01:14.40 | roe_ | anyone have experience with call files? |
| 01:15.04 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.139.144) |
| 01:15.06 | roe_ | I'm getting a permission denied, unable to open error |
| 01:15.21 | roe_ | even though the file is rw all and owned by the asterisk user |
| 01:15.28 | roe_ | and it is able to delete the call file |
| 01:17.30 | cambocambo | hi, is there another document anywhere that tells me how to upgrade asterisk to the latest version? |
| 01:17.46 | cambocambo | current version is 1.2.10 |
| 01:17.52 | roe_ | what distro? |
| 01:17.56 | cambocambo | debian |
| 01:18.05 | cambocambo | i just did an apt-get update and dist-upgrade. |
| 01:18.19 | roe_ | what revision? |
| 01:18.29 | roe_ | i'm using etch, and I have 1.2.13 |
| 01:18.43 | cambocambo | sarge |
| 01:18.44 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 01:18.49 | roe_ | ah, that's why |
| 01:18.56 | *** join/#asterisk larrywww (n=larryRI@ool-44c6e4b6.dyn.optonline.net) |
| 01:19.06 | roe_ | that means 1.2.10 is the newest version in sarge |
| 01:19.09 | cambocambo | should i be updating to the lastest version of asterisk? |
| 01:19.12 | cambocambo | gotcha. |
| 01:19.42 | cambocambo | the latest version of asterisk is 1.2.18 though? |
| 01:19.51 | roe_ | that's what it says in the topic |
| 01:20.20 | roe_ | you could add unstable as an alternate source for apt |
| 01:20.28 | roe_ | and install asterisk from unstable (or testing) |
| 01:20.42 | roe_ | but that's a question for #debian ;-) |
| 01:20.47 | cambocambo | is that reccommended in a production environment. |
| 01:20.49 | cambocambo | ok thanks. |
| 01:20.58 | roe_ | probably not |
| 01:21.04 | roe_ | but etch is stable now |
| 01:21.34 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 01:22.27 | cambocambo | yeah, but i dont really have time to upgrade the box... |
| 01:23.25 | roe_ | then you must not need any of the features that have been fixed from 1.2.10 to 1.2.18 ;-) |
| 01:23.32 | cambocambo | slack excuse i know. |
| 01:23.33 | *** join/#asterisk tessier (n=treed@kernel-panic/sex-machines) |
| 01:23.43 | JT | i recommend compiling asterisk, always |
| 01:23.50 | cambocambo | i just want bug fixes :) |
| 01:23.53 | JT | distros are way too slow |
| 01:24.25 | cambocambo | thanks for your help |
| 01:24.29 | roe_ | good luck |
| 01:24.34 | cambocambo | cheers |
| 01:25.09 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 01:28.02 | roe_ | anyone use call files? |
| 01:28.21 | J4k3 | why is it that linux kids are always having to do so much work to optimize performance? I've been installing FreeBSD for years and theres never been more than maybe 5% performance to be found tweaking everything to the max. |
| 01:28.39 | JT | what are you talking about? |
| 01:28.53 | roe_ | what performance tweaks? rolling a custom kernel? |
| 01:29.00 | J4k3 | roe_: exactly. |
| 01:29.06 | J4k3 | 20:23 < JT> i recommend compiling asterisk, always |
| 01:29.10 | J4k3 | 20:23 < JT> distros are way too slow |
| 01:29.13 | JT | zomg custom |
| 01:29.22 | JT | J4k3: slow to come out with latest releases |
| 01:29.26 | J4k3 | ohhhh! |
| 01:29.32 | JT | not the software actually running any slower |
| 01:29.34 | J4k3 | thats an aspect I hadn't considered at all |
| 01:29.36 | J4k3 | ok |
| 01:29.37 | J4k3 | hahaha |
| 01:29.42 | roe_ | he |
| 01:29.43 | roe_ | h |
| 01:29.52 | acidchild | rob0: ha. |
| 01:29.54 | roe_ | well, it's not like asterisk development is lightning fast |
| 01:29.58 | JT | i also don't trust distros to compile asterisk in a non crack affected way :P |
| 01:30.00 | J4k3 | yeah, I know all about recompiling stuff daily... I used to run sendmail |
| 01:30.02 | J4k3 | ;) |
| 01:30.07 | roe_ | when did sarge come out? 2 years ago? |
| 01:30.10 | roe_ | and 1.2.10 was in sarge? |
| 01:30.10 | JT | roe_: a new release comes out roughly once a month |
| 01:30.17 | roe_ | up to 1.2.18 now? |
| 01:30.23 | roe_ | that's not a *huge* change |
| 01:30.37 | roe_ | plus, they backport big fixes and security issues to debian stable |
| 01:30.47 | roe_ | but debian is notorious for a slow stable release cycle |
| 01:30.53 | snuffy22 | hmm.. anyone know if www.thevoipconnection.com sells/ships to AU? |
| 01:30.57 | JT | releases have a lot of changes in asterisk |
| 01:31.07 | roe_ | i'm taking my asterisk box from stable to testing as we speak |
| 01:31.08 | JT | snuffy22: what are you trying to buy? |
| 01:31.19 | snuffy22 | the aussie stores for the g729 card is like double what it should be |
| 01:31.27 | *** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com) |
| 01:31.46 | roe_ | cheaper to ship from US? |
| 01:31.46 | snuffy22 | $3k AU my ass.. when the exchange rate is 80c in the dollar |
| 01:32.03 | JT | yes, most australian stores are rip offs on asterisk hardware |
| 01:32.05 | JT | roe_: usually is |
| 01:32.27 | JT | 1 AUD is about 82 US cents atm |
| 01:32.40 | snuffy22 | yer but i like round figures :) |
| 01:32.43 | JT | usd is worthless |
| 01:32.47 | J4k3 | yeah |
| 01:32.54 | J4k3 | usd is worthless, so if the card is made in asia expect to pay more |
| 01:33.09 | roe_ | heh |
| 01:33.34 | J4k3 | for example, the Ubiquiti SR9 is built for an american company by somebody somewhere in asia... and the wholesale price has gone up 15% in the last 2 months |
| 01:33.41 | jebba | is the "known good" with asterisk mpg123 still the old 0.59c? Because those docs also reference like redhat 7.3.... |
| 01:35.22 | JT | snuffy22: http://www.telephonydepot.com/product_p/105-050-tc400b.htm |
| 01:35.28 | JT | telephonydepot ships to au |
| 01:36.34 | ManxPower | jebba: I believe so. no recent release that I know of will work |
| 01:36.55 | jebba | ManxPower, thx |
| 01:37.13 | jebba | actually 0.59r |
| 01:37.25 | ManxPower | that's the one that works |
| 01:38.08 | ManxPower | jebba: for the most part the only reason I use mpg with Asterisk anymore is because I'm too lazy to change it over to native MoH support |
| 01:38.52 | roe_ | i've been using mpg123 0.61 |
| 01:38.54 | ManxPower | or have not had a chance to get a copy of the original CD that the customer provided. |
| 01:39.10 | roe_ | going to 0.65 now, apparently |
| 01:39.15 | ManxPower | roe_: interesting. |
| 01:39.30 | ManxPower | I've never ever ever heard of a non 0.59r working. |
| 01:40.01 | roe_ | are you using it for something other than MoH? |
| 01:40.01 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-155-5-67.red.bezeqint.net) [NETSPLIT VICTIM] |
| 01:40.01 | *** join/#asterisk Avochelm (n=damo@gw-morphett.koalatelecom.com.au) [NETSPLIT VICTIM] |
| 01:40.01 | *** join/#asterisk antlers (n=antlers@ip70-173-89-173.lv.lv.cox.net) [NETSPLIT VICTIM] |
| 01:40.01 | *** join/#asterisk Know1 (i=know1@creep.bur.st) [NETSPLIT VICTIM] |
| 01:40.05 | jebba | ManxPower, this is for MPlayer of a radio stream, not MoH |
| 01:40.15 | roe_ | ah |
| 01:51.03 | snuffy22 | thanks jt |
| 01:51.28 | JT | snuffy22: be prepared for at least a 10% tax hit from customs |
| 01:54.15 | *** join/#asterisk lee_is_me (n=chatzill@12-201-102-196.client.mchsi.com) |
| 01:58.56 | lee_is_me | Curious, is a "Barg in" accomplished by sending each end of the call to a conference along with the person barging in? |
| 01:59.42 | lee_is_me | Assuming my idea of bargin in is correct...which I understand to be breaking into the call to interact or just listen... |
| 02:01.27 | jebba | Hmm. mpg123 0.59r doesnt compile on fc6 :( 0.61 appears to work, but i just get silence. I dont see a way to (alternatively) stream ogg or similar either |
| 02:02.16 | ptiggerdine | doesn't mpg123 have some big security holes? |
| 02:02.44 | *** part/#asterisk larrywww (n=larryRI@ool-44c6e4b6.dyn.optonline.net) |
| 02:03.53 | jebba | ptiggerdine, well i'm having it connect to my own server and it's not crossing the 'net, so it's safe in this case at least. |
| 02:04.13 | jebba | ptiggerdine, but if you have a better suggestion, i'm all for it ;) |
| 02:05.06 | snuffy22 | they should mark it as GIFT :P |
| 02:05.12 | snuffy22 | i've done that before |
| 02:06.05 | JT | snuffy22: it makes no difference afaik |
| 02:06.31 | JT | snuffy22: if the total value of an import via postal service exceeds AUD$1000, it's taxed |
| 02:06.37 | JT | $250 for courier service |
| 02:06.48 | JT | telephony depot sends via courier service |
| 02:06.58 | ptiggerdine | jebba, I'm just surpised that asterisk needs mpg123 for MoH |
| 02:07.10 | Corydon76-home | It doesn't anymore |
| 02:07.31 | Corydon76-home | Native format support is actually the recommended way now |
| 02:07.49 | ptiggerdine | okay so it's just or mp3's then? |
| 02:08.02 | Corydon76-home | Not even |
| 02:08.20 | Corydon76-home | You can get asterisk-addons and get format_mp3 for native mp3 support |
| 02:08.24 | Corydon76-home | (decoding only) |
| 02:08.58 | snuffy22 | hmm Corydon76-home, with format_mp3 does it restart the music everytime you put someone back on hold? |
| 02:09.05 | Corydon76-home | It's the old deprecated way, still supported for the time being |
| 02:09.16 | Corydon76-home | snuffy22: no, it does not |
| 02:09.17 | snuffy22 | had the issue a while ago when i used 'files' mode |
| 02:09.30 | snuffy22 | k |
| 02:09.43 | JT | snuffy22: pretty sure you will pay tax |
| 02:10.10 | snuffy22 | yer will factor that in JT :) |
| 02:11.46 | jebba | ptiggerdine, it's not for MoH. It's for playing an icecast2 stream |
| 02:11.54 | ptiggerdine | ah ok... |
| 02:12.10 | jebba | Corydon76-home, but as far as i know native support isn't for streaming, but only for MoH, or no? |
| 02:12.28 | snuffy22 | hmm anyone know where to get the memory info on a linux machine.. aka DDR/DDR2 / 400/533 MHz etc ? |
| 02:12.57 | jebba | i have it working with mpg123 0.59r (compiled with `make linux-nas`) but it's a bit "underwater"... |
| 02:13.05 | snuffy22 | proc/meminfo gives me how much is in there.. no detail of what specs its running at |
| 02:13.17 | jebba | snuffy22, check out the application `lshw` |
| 02:15.18 | snuffy22 | k |
| 02:15.58 | *** join/#asterisk N9URK (n=leonard@rrcs-70-63-204-32.midsouth.biz.rr.com) |
| 02:16.39 | N9URK | anyone in here using 3com 3300 switches for their * network? |
| 02:18.31 | snuffy22 | mm very nifty app jebba.. thanks :) |
| 02:22.35 | N9URK | anyone in here using 3com 3300 switches for their * network? |
| 02:23.25 | Hmmhesays | time to find out if wine will work with dreamweaver cs3 |
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| 02:26.00 | *** join/#asterisk [Latre] (n=chatzill@189.153.84.90) |
| 02:28.19 | [Latre] | hi people.......i try to setup asterisk 1.4.4. with unicall, but i have a problem with a patch........a guy make a new patch for unicall and he said that works ...........the patch is http://pastebin.ca/538594 but i never seen a patch like this....only kind +++ --- , can anyone helpe to how apply this patch???? |
| 02:30.07 | blitzrage | patch -p0 < my_patch.txt |
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| 02:30.54 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 02:32.13 | *** join/#asterisk bbryant (n=Brett@user-24-214-124-177.knology.net) |
| 02:32.53 | [Latre] | are you sure that that patch must works of that way? |
| 02:33.10 | *** join/#asterisk cr4z3d (n=cr4z3d@168.158.222.2) |
| 02:33.54 | [Latre] | is like this: a15 6 |
| 02:33.56 | [Latre] | #define AST_MODULE “Unicall” |
| 02:34.09 | [Latre] | a17 4 |
| 02:34.15 | [Latre] | #ifdef __NetBSD__ |
| 02:34.44 | [Latre] | i didn't see any +++ or ---- |
| 02:39.44 | roe_ | anyone use call files? |
| 02:40.02 | *** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com) |
| 02:40.25 | [TK]D-Fender | roe_, Plenty of us. Now ask your REAL question. |
| 02:41.02 | roe_ | [TK]D-Fender, I remember you! |
| 02:41.11 | roe_ | you helped me through a fax detection problem |
| 02:41.17 | roe_ | (i think) |
| 02:41.19 | Hmmhesays | I hate faxing |
| 02:41.33 | JT | asterisk hates faxing |
| 02:41.40 | Hmmhesays | I use openSER for faxing |
| 02:42.03 | ManxPower | I use PSTN for faxing |
| 02:42.05 | roe_ | when I create a call file, and move it into /var/spool/asterisk/outgoing, it says "unable to open, permission denied, deleting" |
| 02:42.17 | Hmmhesays | are you running asterisk as root? |
| 02:42.21 | roe_ | nope |
| 02:42.25 | roe_ | as asterisk |
| 02:42.28 | Hmmhesays | there you go |
| 02:42.28 | roe_ | file owned by asterisk |
| 02:42.37 | ManxPower | roe_: does the user asterisk is running as have permission to access the fgile. |
| 02:42.37 | roe_ | should I be? |
| 02:42.46 | Hmmhesays | ManxPower: obviously not |
| 02:42.51 | roe_ | yes, I set it to rw-rw-rw |
| 02:42.52 | ManxPower | one might imagine that if it can't access the file, then it can't delete the damn thing either. |
| 02:42.59 | roe_ | yah, exactly |
| 02:43.01 | roe_ | it's very strange |
| 02:43.17 | roe_ | scan_service: Unable to open /var/spool/asterisk/outgoing/call123.call: Permission denied, deleting |
| 02:43.37 | [TK]D-Fender | roe_, How are you getting the file into that folder? |
| 02:43.41 | roe_ | cp |
| 02:43.45 | roe_ | but it deletes it! |
| 02:43.52 | roe_ | oh, and this: scan_thread: Failed to scan service '/var/spool/asterisk/outgoing/call123.call' |
| 02:43.52 | [TK]D-Fender | CP = BAD |
| 02:43.59 | [TK]D-Fender | roe_, mv it |
| 02:44.01 | Hmmhesays | mv |
| 02:44.08 | roe_ | ok |
| 02:44.30 | roe_ | arg!!!! |
| 02:44.38 | roe_ | are you f'ing kidding me?! |
| 02:44.59 | roe_ | that's it |
| 02:45.18 | roe_ | anyone have a technical explanation for why that is? |
| 02:45.21 | [Latre] | so....nobody? |
| 02:45.32 | Hmmhesays | because mv moves the whole file at once |
| 02:45.37 | Hmmhesays | is basically what it boils down to |
| 02:45.38 | roe_ | oh damn |
| 02:45.43 | roe_ | and cp creates a new one |
| 02:45.50 | roe_ | which asterisk tries to parse before it's finished writing it? |
| 02:45.54 | Hmmhesays | bingo |
| 02:46.55 | roe_ | yay |
| 02:46.57 | roe_ | you guys rock |
| 02:47.32 | JT | err |
| 02:47.36 | Hmmhesays | I know |
| 02:47.45 | Hmmhesays | feel free to express your feelings with cash ;) |
| 02:47.58 | roe_ | how about a beer? |
| 02:48.04 | roe_ | are you anywhere near pennsylvania? |
| 02:48.20 | roe_ | ;-) |
| 02:48.22 | JT | mv moves the descriptor for the inode, not the file, usually |
| 02:48.31 | JT | which is the advantage |
| 02:48.47 | JT | cp copys the whole thing, and makes the file available before it may be completed copying |
| 02:49.12 | JT | so asterisk may read the partial file |
| 02:49.12 | Hmmhesays | now I have to figure out how to make dreamweaver cs3 with wine |
| 02:49.46 | roe_ | makes it tough to debug call files if they keep getting deleted... |
| 02:50.54 | roe_ | so does that mean that moving a call file across filesystems would have the same problem as using cp? |
| 02:50.55 | Hmmhesays | you make a script to duplicated it then move it |
| 02:51.08 | Hmmhesays | thats how I've always done troubleshooting with callfiles |
| 02:51.22 | JT | roe: cp file.call file.call.tmp;mv file.call.tmp /var/spool/asterisk/outgoing/file.call |
| 02:51.29 | roe_ | yah, yah |
| 02:51.30 | roe_ | alright |
| 02:51.35 | roe_ | good point |
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| 03:55.36 | n00dle | Here's a question: if I gosub(), do I get an implied return() at the end of the priorities where I've gone, or must I explicitly return()? |
| 03:55.36 | n00dle | (That was a dialplan question, specifically) |
| 03:55.50 | n00dle | Oh, heck... I'll just try... |
| 03:57.03 | roe_ | that might depend if you have auto fall through or not? |
| 03:59.45 | [TK]D-Fender | n00dle, explicit |
| 04:00.29 | n00dle | Ok, tanx. :) |
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| 04:03.47 | roe_ | [TK]D-Fender: I'm halfway to getting Nagios to call me and report what the problem is |
| 04:04.07 | roe_ | king of like this: http://www.mail-archive.com/nagios-users@lists.sourceforge.net/msg04341.html |
| 04:05.12 | [TK]D-Fender | roe_, If you worked half this hard at Asterisk itself... you wouldn't HAVE errors to report ;) |
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| 04:12.47 | roe_ | details, details |
| 04:13.27 | roe_ | i read all that I could find about call files, and everyone says "mv the call file", but nobody emphasises that, or explains why |
| 04:13.49 | roe_ | and when you don't think about it, cp is as good as mv |
| 04:13.56 | Qwell | roe_: because if you write directly to the spool dir, it could read half-way through the write |
| 04:13.59 | Qwell | cp or mv are fine |
| 04:14.14 | Qwell | ack! |
| 04:14.47 | russellb | i was wondering the same thing |
| 04:14.52 | *** part/#asterisk InHisName (n=Administ@c-68-80-56-212.hsd1.pa.comcast.net) |
| 04:15.11 | JT | Qwell: i've heard it said many a time that cp is not recommended |
| 04:15.18 | Qwell | JT: eh? |
| 04:15.35 | JT | Qwell: as asterisk can read the file when it's only partially written |
| 04:15.56 | Qwell | hmm, I'm not sure how cp works... I guess it doesn't just add an inode entry |
| 04:17.02 | Qwell | JT: yeah, I'll give you that one |
| 04:17.59 | JT | i thought it was strange when i first heard it |
| 04:18.05 | JT | but it makes sense when you think about it |
| 04:18.09 | Qwell | yeah |
| 04:19.08 | Qwell | cp somefile /tmp/somefile.call; mv /tmp/somefile.call /var/spool/asterisk/outgoing/ |
| 04:19.22 | Qwell | of course...mv isn't right either |
| 04:19.31 | Qwell | If it's cross-filesystem, you run into the same problems |
| 04:20.43 | roe_ | correct |
| 04:20.45 | roe_ | cp does not work |
| 04:20.51 | roe_ | and mv across file systems does not work either |
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| 04:21.10 | Qwell | symlink ;) |
| 04:21.17 | Qwell | (soft) |
| 04:21.46 | roe_ | hmm, now there's an idea |
| 04:22.36 | Qwell | now, the odds of this happening with such a small file are really low - the biggest one is just don't open a file, and try to write to it slowly, or something |
| 04:22.59 | roe_ | i tried and tried and tried |
| 04:23.04 | mkl1525 | Hi, I've got a beronet B4S0 isdn card with misdn that works but I'm not able to use DTMF - has anybody any suggestions where to look for? and is it possible to generate DTMF tones using the web interface of a snom300|360? |
| 04:23.04 | roe_ | with a 5 line call file |
| 04:23.08 | roe_ | cp never worked |
| 04:23.15 | Qwell | funky |
| 04:23.20 | Qwell | but mv did? |
| 04:23.20 | JT | roe_: what hardware? |
| 04:23.34 | roe_ | uhhhh |
| 04:23.37 | roe_ | P3 600? |
| 04:23.40 | JT | mkl1525: dtmf going where to where? |
| 04:23.41 | roe_ | dual |
| 04:23.43 | JT | roe_: hmm, okay |
| 04:23.50 | roe_ | CF based |
| 04:23.54 | Qwell | ahh |
| 04:23.59 | Qwell | that explains a lot |
| 04:24.10 | Qwell | CF writes are cached... |
| 04:24.10 | JT | cf, only a SLIGHT detail :P |
| 04:24.20 | roe_ | mkl1525 I have a snom 300 if you want me to check |
| 04:24.33 | roe_ | well, I just remember that i converted it to CF a few weeks ago |
| 04:24.36 | roe_ | ;-) |
| 04:25.04 | Qwell | yeah, mv or ln -s then :p |
| 04:25.12 | roe_ | Qwell, shouldn't matter if writes are cached |
| 04:25.14 | mkl1525 | JT, dtmf going from snom300|360 -> * -> beronet B4S0 -> some phone number that uses an DTMF ivr |
| 04:25.20 | roe_ | it's not like * is reading at the block level |
| 04:25.32 | JT | mkl1525: last i've heard, the dtmf support in misdn was pathetic |
| 04:25.41 | Qwell | well, off to bed |
| 04:25.47 | roe_ | thanks again, guys |
| 04:26.02 | JT | mkl1525: but i guess you should ensure that the correct dtmf mode is in uses on asterisk and the snoms |
| 04:26.14 | JT | mkl1525: you should try with an endpoint other than isdn |
| 04:28.14 | mkl1525 | roe_, thanks do you know (or try) if there's a way to call a number from web interface and then when the call is running give some numbers fro an dtmf ivr? |
| 04:28.17 | mkl1525 | JT, thanks for the info |
| 04:29.45 | JT | mkl1525: i always recommend bristuff over misdn |
| 04:31.07 | mkl1525 | JT, I'm running bristuff myself but customer already had the beronet card... |
| 04:33.12 | *** join/#asterisk mightnare (n=mike@s230165.ppp.asahi-net.or.jp) |
| 04:35.52 | JT | mkl1525: bristuff works with beronet |
| 04:37.51 | mkl1525 | JT, didn't know this - any drawbacks you know of against misdn? |
| 04:38.50 | JT | misdn is a pile of rubbish |
| 04:38.59 | JT | dtmf issues |
| 04:39.05 | JT | useless NT mode support |
| 04:39.14 | JT | inability to access zap features |
| 04:39.23 | JT | unstable |
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| 04:41.40 | [hC] | weird, even with ztdummy loaded on this box, /dev/zap/pseudo doesnt exist. |
| 04:49.24 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:56.36 | tzafrir_laptop | this means zaptel isn't loaded, probably |
| 04:57.13 | tzafrir_laptop | lsmod | grep zaptel |
| 04:57.54 | tzafrir_laptop | bye now |
| 04:58.21 | flenders | guys, there's a clicking noise on our PRI, any ideas what it might be? card is a sangoma 101 |
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| 05:44.41 | Sargun | Anyone know of any SIP providers that support ANI/ANI2 (receiving) |
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| 05:56.10 | tzafrir | flenders, calls from PRI to what exactly? |
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| 05:59.56 | jebba | snuffy22, what |
| 06:00.05 | jebba | snuffy22, what's better, crack or `lshw` ? |
| 06:03.06 | snuffy22 | mmm.. really do like lshw.. |
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| 07:14.28 | flenders | tzafrir: calls to anywhere |
| 07:14.36 | flenders | or calls to the PRI |
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| 07:24.03 | ingenio | any sed guru want to fix this statement for me? i'm tired. :/ |
| 07:24.06 | ingenio | sed -i 's!^#!/bin/sh!#!/bin/bash!' /usr/sbin/safe_asterisk |
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| 07:27.59 | Nugget | using ! as the regex delimeter and then also having a ! in the match text is probably your problem. |
| 07:28.46 | Nugget | don't need to be a guru to spot that one, and since you asked for gurus I probably shouldn't have helped. :) |
| 07:29.03 | Nugget | seems to me, though, you don't want a guru, you just want someone who knows what the problem was. |
| 07:29.08 | ingenio | haha pretty much |
| 07:29.25 | ingenio | appreciate the help. im not enjoying asterisk thus far, i must admit. :P |
| 07:29.52 | ingenio | long night |
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| 07:41.40 | ingenio | so i have a soho with 2 analog lines, 2 sip phones, and a tdm02b. i'd like to greet callers, ring sip phone for 20 seconds, then use second line to three-way a cell phone. each line gives caller a different message, and both recognize faxes. also during closed hours sends direct to voicemail |
| 07:41.56 | ingenio | would i be better off using an out of the box solution such as trixbox instead of trying to configure asterisk from scratch? |
| 07:42.23 | mosty | ingenio, for a simple setup, maybe |
| 07:43.01 | mosty | except i would try to eliminate fax from the setup |
| 07:43.14 | ingenio | why's that? |
| 07:44.06 | Aces1Up | ingenio i had that same question about 3 days ago, i have had great satisfaction in not going with an out-of-box solution. |
| 07:44.16 | Aces1Up | as i would not understand anything. |
| 07:44.25 | mosty | ingenio, because i don't trust asterisk's fax support |
| 07:44.43 | ingenio | Aces1Up: that makes sense. |
| 07:44.52 | mosty | and i'm not sure if asterisk can even share fax/phone calls on the same line in a sane manner |
| 07:44.53 | ingenio | mosty: ah, I didn't know. trixbox advertises it as a feature.. |
| 07:45.11 | ingenio | otherwise, i was told my needs aren't difficult to configure |
| 07:45.29 | ingenio | but alas after numerous recompiles i'm unable to get asterisk even stable |
| 07:45.45 | JT | asterisk doesn't have much fax support |
| 07:45.49 | JT | but may work with spandsp |
| 07:45.53 | JT | or hylafax |
| 07:46.00 | Aces1Up | if i have a 2 DID's from two different countries inbound to my asterisk box, and are being recieved on their designated channels, what is the command or proper way of tieing these two connections together? |
| 07:46.18 | Trevor_b | share fax/phone calls?? |
| 07:46.54 | Trevor_b | asterisk+spandsp is pretty darn stable. |
| 07:47.21 | Trevor_b | ingenio what OS you compiling it on? |
| 07:47.25 | JT | that would depend on the versions i'd think |
| 07:47.26 | ingenio | ubuntu :/ |
| 07:47.33 | ingenio | not by choice |
| 07:47.41 | JT | spandsp is also no longer maintained for asterisk |
| 07:47.49 | JT | ubuntu is a linux distribution, not an os |
| 07:48.06 | Aces1Up | is there a command or technology i should look into accomplishing my task as mentioned above? |
| 07:48.41 | JT | Aces1Up: i'm not sure what you really want to do |
| 07:48.42 | ingenio | ah i'm tired and i read it as distro. i didn't realize anyone was building asterisk toaster boxes. |
| 07:49.07 | ingenio | actually how does asterisk run on bsd? |
| 07:49.25 | mosty | Aces1Up, calls coming in on those DID's go to a particular dialplan extension/context, use Goto at that point, direct them both to the same place |
| 07:49.37 | Aces1Up | JT, just this, I have 1 DID being forwarded to my asterisk box, and 1 other DID being forwarded to my asterisk box. So I have a person on both incoming lines, how do i tie them together. |
| 07:49.39 | JT | it runs, but zaptel is unsupported, and there is a 3rd party zaptel version, some cards work |
| 07:49.50 | *** join/#asterisk s0ck (n=m@unaffiliated/s0ck) |
| 07:50.15 | Aces1Up | I would like to tie them together so they can talk to each other. |
| 07:50.24 | Trevor_b | Use a conference room |
| 07:50.32 | JT | Aces1Up: you want a call in on one to call out on the other? |
| 07:50.36 | Trevor_b | checkout the meetme options |
| 07:51.08 | mosty | Aces1Up, does your phone have a conference/3-way calling function? |
| 07:51.15 | mosty | if not, use meetme |
| 07:51.44 | JT | meetme requires zaptel timing |
| 07:51.57 | Aces1Up | jt, no, i little more complicated, they are both incoming to my box, i don't want to use a phone connected to my box, i just want the voice stream to comeinto my box and out the other incoming line. my box acting as a liason... |
| 07:51.58 | JT | app_conference may be a better option if no such timing is available |
| 07:52.16 | JT | Aces1Up: so a conference then :) |
| 07:52.32 | Trevor_b | Meetme requires a USB interface for software timing, or a TDM interface to use zaptel timing. |
| 07:52.59 | Aces1Up | jt well ok if thats what its called in asterisk lol.. i'll check it out, just thought a conference was more than 2 users. |
| 07:54.00 | JT | that's what it's called in anything |
| 07:54.12 | JT | you described 2 inbound calls talking to each other |
| 07:54.21 | JT | as opposed to one dialling the other :) |
| 07:54.31 | Aces1Up | jt yes. |
| 07:54.36 | cy303 | sup JT |
| 07:54.46 | Aces1Up | i see where you are going, heh.. |
| 07:54.47 | JT | not much |
| 07:54.48 | Aces1Up | thanks. |
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| 07:55.23 | Zeeek | anyone here using centOS ? |
| 07:55.41 | Trevor_b | Yep |
| 07:55.42 | Zeeek | and if so, what version |
| 07:56.11 | Trevor_b | 4.4 havet updated to 4.5 yet, use 4.3 on trixbox (pretty sure thats what is used) but i dont run the current trixbox's anymore. |
| 07:56.24 | Zeeek | what about centOS 5 ? |
| 07:56.32 | s0ck | uname -a doesn't tell me what ver i have |
| 07:56.39 | Trevor_b | Should be fine, we just haven't built for it yet./ |
| 07:56.41 | s0ck | 4.4ish |
| 07:56.43 | JT | Trevor_b: ztdummy is often not sufficient |
| 07:56.43 | Zeeek | I'm about to download and install linux, looking at the best options for * |
| 07:57.00 | JT | heh |
| 07:57.04 | JT | stuff rpm based distros :P |
| 07:57.24 | s0ck | quite like slackware myself |
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| 07:57.36 | JT | debian or similar gets my vote |
| 07:57.45 | Zeeek | Yeah, I'm running a very old slackware on my main box now |
| 07:57.54 | s0ck | can use swarez to update it too |
| 07:57.59 | s0ck | so it aint completely neanderthal |
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| 07:58.08 | Zeeek | JT please exand on the RPM hatred :) |
| 07:58.18 | s0ck | <3 slackware |
| 07:58.19 | *** join/#asterisk lokkju_wrk (n=lokkju@unaffiliated/lokkju) |
| 07:58.28 | JT | one of the worst package management formats to become popular |
| 07:58.31 | JT | dependancy hell |
| 07:58.33 | Zeeek | I've never used RPM before |
| 07:58.35 | s0ck | trixbox does indeed install it's own distro tho |
| 07:58.46 | JT | s0ck: yeah, it's called centos :P |
| 07:58.51 | s0ck | that's the one :P |
| 07:59.11 | Aces1Up | jt, i kinda didn't want to get into conferencing as i heard it reduces the amount of concurrent calls i can handle on my box. man i just thought it would be easy to connect to calls together, wouldn't what i want be more like a transfer? |
| 07:59.20 | Zeeek | I was looking at playing with the LumenVox voice recognition stuff. Doesn't look like it will work with slack |
| 07:59.37 | JT | Aces1Up: transfers ring extensions |
| 07:59.39 | Zeeek | centOS does work with it tho |
| 07:59.40 | Aces1Up | it comes into my box, i handle it by transfering it to the other call that is maybe in a queue or something. |
| 07:59.45 | *** join/#asterisk af_ (n=getsmart@81-174-46-17.f5.ngi.it) |
| 07:59.54 | JT | Zeeek: i can't recommend slack to a beginner |
| 08:00.00 | mosty | aces1up: transfer is when you send a call to a line that isn't already connected, usually |
| 08:00.11 | Zeeek | I will have the free space on the drive in just a few minutes |
| 08:00.12 | JT | Aces1Up: hmm, you can give it a go, but i dunno |
| 08:00.29 | Zeeek | JT been running slack for 4 years since asterisk 0.? |
| 08:00.47 | Zeeek | or was it 1.0.? |
| 08:00.48 | JT | oh ok |
| 08:01.07 | Zeeek | gotta admit, not using X makes things a lot easier |
| 08:01.07 | JT | i thought you implied you were new to linux |
| 08:01.30 | Zeeek | the truth: I know sheisse about linux and more about FreeBSD |
| 08:01.52 | Zeeek | but I am pretty good with google and have a friend who knows slack in case I get really stuck |
| 08:02.14 | Zeeek | The last slack install went without a hitch |
| 08:02.32 | Zeeek | like I say, not worrying about a desktop simplifies things a lot :) |
| 08:02.48 | JT | lumenvox only cares about distro if it has a binary kernel module or similar |
| 08:03.00 | JT | i never use GUIs on servers |
| 08:03.10 | Zeeek | JT no, no one should |
| 08:04.16 | Zeeek | supposedly centOS is very stable, that's why I was considering it. However, I know zilch about the various distribs, since once you get something that runs, who cares? |
| 08:04.36 | Zeeek | hence my 3 versions behind slack |
| 08:06.07 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 08:06.38 | JT | some versions of the kernel shipped with centos have a spinlock bug with asterisk |
| 08:06.47 | Zeeek | ewww |
| 08:07.08 | JT | just use debian and be done with it ;) |
| 08:07.17 | JT | it's pretty academic if you compile asterisk anyway |
| 08:07.36 | Zeeek | I just remembered I did install Debian on a laptop. But my wife wasn't going for it, had to resore winbloze |
| 08:07.44 | jql | sad |
| 08:08.44 | Zeeek | doesn't digium use fedora for dev ? |
| 08:08.44 | JT | who cares what they use? linux is linux, mostly |
| 08:08.44 | JT | fedora is a joke |
| 08:08.51 | Zeeek | I think it's of interest what they use |
| 08:09.53 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 08:11.58 | JT | more interesting what's used in production sites, if anything |
| 08:12.02 | *** join/#asterisk lokkju_wrk_ (n=lokkju@unaffiliated/lokkju) |
| 08:12.08 | JT | but as i said, it really matters little which distro |
| 08:12.13 | Zeeek | true. I wonder what the majority use? |
| 08:12.42 | Zeeek | not that that would nec be my choice |
| 08:12.55 | JT | heh |
| 08:12.55 | *** join/#asterisk plasmid (n=noway@c-68-46-97-136.hsd1.pa.comcast.net) |
| 08:12.58 | JT | you said nec |
| 08:13.12 | JT | i'd run asterisk on NEC servers if i had a choice :) |
| 08:13.21 | Zeeek | as in "necessarily" a word I was loath to type |
| 08:13.27 | JT | unfortunately they're quite pricy |
| 08:15.57 | jql | nec? |
| 08:15.57 | *** join/#asterisk saftsack (n=oliver@p54a7fae2.dip.t-dialin.net) |
| 08:15.58 | JT | jql: yes, NEC make servers |
| 08:15.58 | JT | very sweet ones |
| 08:15.58 | Zeeek | better than my homebrew JunkPile (tm) ? |
| 08:15.59 | JT | yes, lockstep dual motherboard/everything units |
| 08:16.06 | *** join/#asterisk friedrich| (n=friedric@85.177.253.250) |
| 08:16.06 | Zeeek | with its patented "useorthrowaway" hardwxare |
| 08:16.07 | JT | 2 motherboards do an operation in parallel |
| 08:16.19 | *** join/#asterisk waptaxi (n=cahe@45.151-224-87.telenet.ru) |
| 08:16.21 | JT | if one fails, the hardware lockstep circuitry disconnects it |
| 08:16.37 | Zeeek | way too anal! |
| 08:16.58 | jql | whoa |
| 08:16.59 | JT | it helps you get close to 5 9s of uptime |
| 08:17.24 | JT | you can also split the servers, like splitting a raid1 array |
| 08:17.29 | JT | to upgrade software or what not |
| 08:17.31 | Zeeek | my Pentium III 800 has been up for four years |
| 08:17.55 | JT | you can upgrade one whilst the other handles requests |
| 08:18.01 | s0ck | send your electricity company a christmas card this year :P |
| 08:18.14 | Zeeek | it runs on steam |
| 08:18.29 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 08:18.34 | Zeeek | but the steam is created via atomic energy |
| 08:18.50 | JT | so yeah, they're some of the few x86 servers i consider getting close to telco grade |
| 08:19.36 | jql | I may hit 5 9s in aggregate, but to have a single server for that is pretty nice |
| 08:19.58 | Zeeek | well, I decided to dedicate a couple of hours to centOS 5, downloading the ISO DVD now. I can always go back at this stage |
| 08:19.59 | JT | i think nec guarantee it as well |
| 08:20.25 | JT | with some obvious conditions of course, they will give you your money back if the hardware isn't 99.999% available |
| 08:20.42 | jql | plug a 5-nines server into a 6-nines san and a 5-nines switch, and what do you get? a poor-man's telco. whee |
| 08:21.20 | JT | losing your nines due to an asterisk crash, priceless ;) |
| 08:21.49 | Zeeek | JT what is your favorite SIP phone? |
| 08:22.37 | JT | polycom |
| 08:23.00 | jql | JT: ever use a polycom plugged into AC power? |
| 08:23.03 | Zeeek | they're good. This DVD download is gonna take a while. I wish the fibre would get here |
| 08:23.12 | jql | with a headset? |
| 08:23.20 | Zeeek | jql - hum? |
| 08:23.49 | jql | I'm getting a damn ac hum on the phone when using a headset. the phones are plugged into wall power -- doesn't happen with PoE |
| 08:24.10 | JT | jql: oh, so that's what the hum is |
| 08:24.10 | JT | which power brick |
| 08:24.10 | Zeeek | yeah I get it too |
| 08:24.10 | JT | the new little one? |
| 08:24.10 | jql | 501 or 601 |
| 08:24.10 | JT | switchmode |
| 08:24.10 | JT | 430 |
| 08:24.15 | jql | yeah, the little switching power thing |
| 08:24.24 | JT | the inline one, not wallwart |
| 08:24.38 | jql | 500mA/12v or 250mA/24v inline power thingies |
| 08:24.55 | JT | well that sucks |
| 08:25.02 | Zeeek | I think mine is 1A so it must have more to do with regulation than current? |
| 08:25.05 | JT | need PoE at home now i guess ;) |
| 08:25.12 | jql | pooh |
| 08:25.13 | JT | yes regulation |
| 08:25.27 | Zeeek | I'm using a wall worat from a musical instrument |
| 08:25.38 | *** join/#asterisk tsurko (n=tsurko@77.70.24.142) |
| 08:25.45 | jql | Zeeek: err, I'm off by one |
| 08:26.14 | jql | 1000/12v, 500/24, 250/48 (PoE) |
| 08:26.31 | Zeeek | usually the Roland 1A supplies are well-regulated |
| 08:26.47 | Zeeek | I guess poe is expensive tho |
| 08:26.54 | JT | wall warts are usually shit |
| 08:27.01 | jql | $250 d-link is enough to fix it |
| 08:27.13 | Zeeek | I couldn't even get a polycom ps. Couldn't find them in Europe |
| 08:27.18 | jql | I'm sure cheapers ones would too, but that was on sale |
| 08:27.50 | JT | Zeeek: the new switchmode unit is universal input |
| 08:27.51 | Zeeek | someone must make a better ps |
| 08:28.13 | Zeeek | JT yeah, I CURSE all companies that make 110v hardware in this day and age |
| 08:28.36 | jql | yeah, we ripped one open. the whole thing is a funky bit of circuitry. the hum waveform it puts out is seriously funky |
| 08:28.58 | Zeeek | just generate an equivalent hum 180 out |
| 08:29.16 | Zeeek | use a $5,000 signal generator |
| 08:30.18 | jql | one smooth 60hz sine wave, along with a 120hz spike thing |
| 08:30.32 | jql | very annoying |
| 08:30.49 | JT | i guess there's no room for filter caps inside it |
| 08:30.53 | Zeeek | yes it's audible at the other end, too bad |
| 08:31.20 | Zeeek | but why only with headset? |
| 08:31.30 | jql | at my office, the headsets themselves don't really have a very audible hum (without a headset amplifier). it's the caller who gets buzzed to death |
| 08:31.39 | Zeeek | yep |
| 08:31.50 | JT | my headset doesn't work without a headset amp |
| 08:32.10 | Zeeek | there's one Plantronics that will, but with the hum |
| 08:32.22 | jql | yeah, we have plantronics everywhere |
| 08:32.44 | jql | but the amplifier just makes the hum audible to the polycom user, afaik |
| 08:32.52 | JT | this is plantronics i'm speaking of :) |
| 08:32.53 | Zeeek | that sucks |
| 08:33.11 | Zeeek | the amp should relieve the need for the phone to work harder |
| 08:34.41 | jql | yeah, I've been puzzling on this for weeks. I could never replicate it myself, because I have a PoE switch on my desk. duh |
| 08:34.41 | Zeeek | the hum is obnoxious but it isn't that bad. I've recorded it. |
| 08:34.41 | jql | a couple people in my office have much worse hum than the others |
| 08:34.41 | Zeeek | it's just there and can be a distraction |
| 08:34.41 | jql | I don't know what changes the volume like that |
| 08:34.41 | Zeeek | maybe they're American Idol fans! |
| 08:34.45 | jql | mmmmmmmmmmmmmmmm |
| 08:34.47 | Zeeek | humming |
| 08:34.52 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 08:35.07 | Nate9939 | i'm getting this after i hang up on the called part end on my sip channel, my intuition is telling me the sip channel isn't getting detecting the hang-up? Maximum retries exceeded on transmission |
| 08:35.24 | Nate9939 | i get that error after i hang up my phone which is a cell phone. |
| 08:36.02 | jql | Nate9939: Is that on a BYE message? |
| 08:36.22 | Nate9939 | [Jun 5 01:33:06] WARNING[2554]: chan_sip.c:1900 retrans_pkt: Maximum retries exceeded on transmission 78e6e31c58880e1511594b5053ef5880@gw3.sip.telasip.com for seqno 102 (Critical Response) |
| 08:36.35 | Nate9939 | i get as soon as i hang up my cell-phone. |
| 08:36.43 | jql | you should turn on 'sip debug' |
| 08:36.47 | Nate9939 | ok doke. |
| 08:36.56 | jql | that will log the messages being resent |
| 08:38.38 | Nate9939 | <-------------> |
| 08:38.38 | Nate9939 | --- (9 headers 0 lines) --- |
| 08:38.38 | Nate9939 | SIP Response message for INCOMING dialog BYE arrived |
| 08:38.39 | Nate9939 | Really destroying SIP dialog 'MmExNGE1OWE4MTc5YWNkMTY3YWEzNDJhNzAzNWE1MTM.' Method: ACK |
| 08:38.39 | Nate9939 | Retransmitting #1 (no NAT) to 4.79.19.56:5060: |
| 08:38.39 | Nate9939 | SIP/2.0 200 OK |
| 08:38.44 | woolbeo | how do I check the new jitterbuffer status when used on chan_zap? |
| 08:38.46 | Nate9939 | thats on a bye message. |
| 08:38.55 | Nate9939 | then it re-transmits for like 7 times. |
| 08:41.15 | woolbeo | Zeek, is your ps on your polycom linear or switching? |
| 08:41.52 | JT | <PROTECTED> |
| 08:42.02 | jbot | somebody said pb was a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org |
| 08:42.02 | Zeeek | woolbeo not sure, but it is for a Roland synthesizer, usually they're very well regulated |
| 08:42.20 | JT | Nate9939: yes it basically means you have nat problems usually |
| 08:42.29 | woolbeo | Zeek, is it small and cool, or larger and warm/hot? |
| 08:42.45 | Zeeek | what is this, the pr0n channel? |
| 08:43.04 | woolbeo | Zeek, lol.... |
| 08:43.04 | Zeeek | I think it's warm but cool |
| 08:43.15 | Zeeek | It isn't here I'm at the office |
| 08:43.45 | Zeeek | come to think of it, it's a 110v 60hz ps hung on a transformer. That may have sthing to do with it |
| 08:44.09 | Zeeek | It isn't a problem that has bothered me a lot but someday I should get a proper ps for the phone |
| 08:45.11 | woolbeo | Zeek, I see... We had some problems with IP 430, becasue they came with a switching ps, we switched it for a linear ps, and the hum went away. |
| 08:45.30 | Zeeek | the hum is only with headset |
| 08:45.46 | woolbeo | Zeek, yeah same here.. plantronics with amp. |
| 08:45.55 | Zeeek | mine is without amp |
| 08:45.59 | woolbeo | without amp it is there, just not as bad |
| 08:46.12 | JT | woolbeo: the problem is not that it's a switching ps, but that it seems to be basically an unfiltered switching ps |
| 08:46.27 | Zeeek | well, I learned something today: don't buy an amplified headset to get rid of hum. Noted |
| 08:47.08 | Zeeek | yeah the point is, the hum is a residual AC component. SHould be filtered out |
| 08:47.21 | JT | well |
| 08:47.35 | JT | everything needs filtering on a power supply, especially switchmode |
| 08:47.41 | JT | no filtering == no DC |
| 08:47.46 | woolbeo | JT, good to know.. we couldn't find a filtered swithcing ps, but then again we didn't look hard, because we knew of some linear ones that didn't creat the hum |
| 08:47.54 | Zeeek | is the polycom input DC or pulsating DC ? |
| 08:48.15 | JT | it'd have to be pulsating dc going off what jql said |
| 08:48.24 | JT | with their switchmode ps |
| 08:48.30 | *** join/#asterisk smurf (n=smurf@debian/developer/smurf) |
| 08:49.30 | woolbeo | All I cared about was fixing the problem, not which ps was more efficient. I would have gone with the POE, if I could have. |
| 08:49.31 | Nate9939 | jt, hrmm those nat problems are probably cause i haven't set the externip and localnet stuff? |
| 08:49.47 | Zeeek | Nate9939 that would be the first thing to do |
| 08:49.55 | JT | Nate9939: quite probable, then again i don't know your network setup is |
| 08:50.07 | Nate9939 | Reliably Transmitting (no NAT) to 4.79.19.56:5060: |
| 08:50.20 | Nate9939 | hrmm, and i do have nat. |
| 08:50.25 | JT | that tells me nothing of your network setup |
| 08:50.36 | Nate9939 | i am behind a nat though. |
| 08:50.41 | Zeeek | Nate=1 |
| 08:50.42 | Nate9939 | basic soho router. |
| 08:51.03 | JT | explain what's connecting to what over what or it's not worth our effort to try and help :) |
| 08:52.03 | Nate9939 | ok sip channel connecting to 4.79.19.56 from internat 192.168.1.50. my public ip is 68.227.41.148 |
| 08:53.52 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:54.35 | JT | words are more important than ips... i'll guess that you have a phone on your lan where asterisk is, trying to connect to an ITSP, and your asterisk server is also on a private ip behind nat |
| 08:55.15 | Zeeek | Nate9939 I think the main thing is to know whether the phone and the pbx are on the same side of NAT |
| 08:56.45 | Nate9939 | ahh ok yes the phone and pbx are. |
| 08:57.07 | Nate9939 | it is a softphone to pbx to outside sip channel. |
| 08:57.12 | Zeeek | any ports forwarded? |
| 08:57.46 | JT | is asterisk behind nat? |
| 08:58.14 | Zeeek | are the phone black or grey? |
| 08:58.14 | *** join/#asterisk denke (n=denke@mehess.adsl.datanet.hu) |
| 08:59.48 | *** join/#asterisk fbffff (n=fbffff@c-67-167-98-42.hsd1.il.comcast.net) |
| 09:06.41 | *** join/#asterisk saftsack (n=oliver@p54A7EFE4.dip.t-dialin.net) |
| 09:09.09 | Zeeek | long story short... |
| 09:13.34 | Zeeek | anyone here in France? |
| 09:15.07 | *** join/#asterisk Xen^ (n=linux@unaffiliated/lnux/x-10290) |
| 09:22.17 | Zeeek | ClubAsteriskParis may be having a meeting soon (early July) that would be of interest |
| 09:46.19 | Zeeek | after a brief flurry... nada |
| 09:47.03 | Uatec_ | lol |
| 09:47.05 | Uatec_ | paris |
| 09:47.06 | Uatec_ | ? |
| 09:47.54 | *** join/#asterisk Bladerunner05 (n=feelme@81-174-56-54.f5.ngi.it) |
| 09:47.56 | Zeeek | what about it? |
| 09:48.00 | Uatec_ | i've been there |
| 09:48.05 | Zeeek | beautiful day here now |
| 09:48.21 | Zeeek | lousy day to be in a room talking about voip |
| 09:48.33 | Uatec_ | beautiful day here too, in england |
| 09:48.40 | Uatec_ | again, lousy day to be in a room ircing about voip |
| 09:48.46 | Zeeek | england? Must be the one day of the year, eh? |
| 09:48.58 | J4k3 | Uatec_: thats why you get some wifi... so you can IRC from outside (or the crapper) |
| 09:49.59 | Uatec_ | OI |
| 09:50.10 | Uatec_ | it's been nice here all week |
| 09:50.13 | Zeeek | what? huh? |
| 09:50.16 | Uatec_ | and last time i was in france, 2 weeks ago |
| 09:50.17 | Uatec_ | it rained |
| 09:50.26 | Zeeek | it never rains in France |
| 09:50.27 | Uatec_ | J4k3, stuck at my desk |
| 09:50.28 | Uatec_ | PFFF |
| 09:50.30 | Uatec_ | it rained |
| 09:50.34 | Zeeek | except when foreigners coe |
| 09:50.35 | Uatec_ | cherboug was away |
| 09:50.41 | Uatec_ | lol |
| 09:50.44 | Uatec_ | s/away/awash |
| 09:50.54 | Zeeek | umbrellas of cherbourg and all |
| 09:51.10 | Zeeek | actually a few weeks ago it snowed in the south |
| 09:58.02 | *** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca) |
| 10:01.31 | Uatec_ | lol |
| 10:01.36 | Uatec_ | but it's mountainous there |
| 10:01.43 | Uatec_ | so that's ok |
| 10:03.06 | Uatec_ | i'm going to go to france |
| 10:03.15 | Uatec_ | if i see you Zeeek, i'll wave |
| 10:03.18 | Uatec_ | although i don't know who you are |
| 10:03.22 | Uatec_ | so i'll just wave at everybody |
| 10:03.54 | Zeeek | vacation? |
| 10:05.08 | Zeeek | [10:19] <Zeeek> well, I decided to dedicate a couple of hours to centOS 5 |
| 10:05.18 | Zeeek | the DVD is nearly burned |
| 10:05.43 | Zeeek | oops that's almost two hours already |
| 10:08.44 | Zeeek | DVD burned and boots |
| 10:12.09 | Uatec_ | yeah, i guess |
| 10:12.10 | Uatec_ | why not? |
| 10:12.13 | Uatec_ | france is a nice place |
| 10:12.17 | Uatec_ | beautiful place |
| 10:12.20 | Uatec_ | well, some of it |
| 10:12.47 | *** join/#asterisk andyd (n=andyd@host90-152-23-30.ipv4.regusnet.com) |
| 10:13.50 | Zeeek | I have never been anywhere in the UK aside from London and that only for about one day |
| 10:14.43 | Zeeek | in 5 minutes I'll have spent two hours on the centOS preparations |
| 10:15.24 | Zeeek | it would have taken days in the 19th century |
| 10:15.41 | Zeeek | just finding enough horses would be a major pain |
| 10:15.57 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 10:16.53 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 10:20.04 | Uatec_ | i am fortunate enough to live in a really pretty part of the country |
| 10:20.08 | Uatec_ | but it is a bit wooded |
| 10:20.24 | Uatec_ | it's not like a suffer from claustrophobia, but it always just seems enclosed |
| 10:20.39 | Uatec_ | i like big open fields, and valleys and mountains in the distance and stuff.. |
| 10:21.01 | Uatec_ | it would have taken days to install centos in the 19th century |
| 10:21.02 | Uatec_ | ? |
| 10:23.43 | Zeeek | I'm looking at the partition table wondering if it knows to keep the existing ones |
| 10:24.24 | Zeeek | should I just hit ok and see? |
| 10:24.24 | *** join/#asterisk yassaccan (n=yassacca@admin131.hgo.se) |
| 10:35.51 | Uatec_ | ift there's nothing important there |
| 10:37.50 | Zeeek | he, no there's a lot of data |
| 10:38.02 | Zeeek | I'll have to do the pt manually |
| 10:38.07 | Zeeek | not sure I want to do that right now |
| 10:38.18 | Uatec_ | oop |
| 10:38.47 | Uatec_ | not necessarily a useful answer but, considered using separate system and data disks? |
| 10:39.49 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 10:39.58 | Zeeek | yeah I was just considering throwing in an old disk. I've got plenty of these |
| 10:40.36 | Zeeek | in fact I have one 30gb at home that's already in a drawer |
| 10:41.43 | Uatec_ | what i need to find is a PCI SATA RAID controller that poundsign linux can access so i can install Asterisk Business Edition directly on to the mirrored array |
| 10:42.24 | *** join/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
| 10:43.33 | Zeeek | well it's LUUUUNCH time |
| 10:43.36 | *** part/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 10:50.37 | *** join/#asterisk dec0y (n=decoy@LReunion-151-17-29.w193-253.abo.wanadoo.fr) |
| 10:52.31 | dec0y | bonjour, est-ce qu'il y a des français ici? |
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| 11:08.18 | *** join/#asterisk JT (n=jon@unaffiliated/jt) |
| 11:10.21 | *** join/#asterisk jmls (n=jmls@62.49.235.130) |
| 11:10.42 | jmls | morning ladies and gentlemen, geeks and hackers |
| 11:10.55 | jmls | anyone using cepstral TTS in 1.4 ? |
| 11:11.54 | *** join/#asterisk Splat (n=splat@home.heehawhills.com) |
| 11:12.04 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:17.21 | *** join/#asterisk bapril (n=bapril@pool-70-20-40-8.man.east.verizon.net) |
| 11:17.27 | *** join/#asterisk achu (n=achu@122.167.61.20) |
| 11:17.48 | achu | I have setup brodvoice connection on my asterisk server |
| 11:18.25 | achu | when I look at the asterisk cli it shows registered status |
| 11:18.30 | FuriousGeorge | i simply cannot get asterisk running stable on this server of mine... 1.2.x would deadlock once a week.. i have identical hardware elsewhere that runs fine. every now and then i'll swap some parts, clone the hd of the well-behaved server, and sure enough it would deadlock in a week anyway. |
| 11:18.44 | achu | but the problem is I can't hear anything |
| 11:18.53 | achu | for both incoming and outgoing |
| 11:19.43 | FuriousGeorge | finally i gave up on 1.2.x, since 1.4 is now up to the .4 point release, and i decided to just upgrade. it couldnt get worse than deadlocking once a week, right? well, after 36 hours this time it crashed instead |
| 11:20.22 | FuriousGeorge | i took care to build with DONT_OPTIMIZE and DEBUG_THREADS , but alas it didnt even have the decency to dump a core |
| 11:20.43 | achu | when I look at the logs the incoming calls is accepting and it get into the IVR |
| 11:21.07 | achu | but I can't hear anything , the same result with outgoing |
| 11:21.08 | FuriousGeorge | achu: you probably have NAT issues |
| 11:21.10 | FuriousGeorge | ~nat |
| 11:21.32 | jbot | [nat] Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 11:21.32 | *** join/#asterisk snook3r (n=ariel@bzq-219-46-202.isdn.bezeqint.net) |
| 11:22.03 | FuriousGeorge | ~docs |
| 11:22.08 | jbot | from memory, docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com |
| 11:22.40 | FuriousGeorge | jbot forgot to mention that you also need to forward some ports |
| 11:22.44 | FuriousGeorge | its all in the docs |
| 11:23.07 | achu | k, but I am not using any firewalls |
| 11:23.29 | FuriousGeorge | then i dunno |
| 11:23.56 | achu | I have connected the cable modem directly to a linux router and have no firewalls |
| 11:24.04 | *** join/#asterisk kkeil (n=kkeil@p54978bcb.dip0.t-ipconnect.de) |
| 11:24.23 | FuriousGeorge | firewall != nat, necessarily |
| 11:24.26 | *** part/#asterisk jmls (n=jmls@62.49.235.130) |
| 11:24.38 | *** join/#asterisk jmls (n=jmls@62.49.235.130) |
| 11:24.40 | FuriousGeorge | ~firewall |
| 11:24.42 | jbot | somebody said firewall was This is a form of Internet security that stands between a private network and the Internet. It is like a wall in that it can prevent unwanted traffic from passing either way. Some firewalls have proxy functions built in. In fact, the distinction between a firewall and a proxy is often blurry. Add in the differences and similarities ... |
| 11:25.23 | FuriousGeorge | the point is nat means that you take one public ip and have many private ip |
| 11:25.44 | FuriousGeorge | a firewall can just be software that blocks ports on one computer, like "windows firewall" |
| 11:25.55 | achu | k |
| 11:28.38 | achu | If I call the number from outside it always rings |
| 11:29.46 | Uatec_ | dec0y, zeeek habite en paris, mais il a depart a 12.45 |
| 11:30.37 | achu | I used nat=1 in sip trunk's peer details |
| 11:30.44 | achu | but the result is same |
| 11:30.55 | FuriousGeorge | achu: you said you plugged ur modem into a linux router |
| 11:30.59 | Uatec_ | je suis desole pour ma pauvre francais |
| 11:31.14 | achu | yes |
| 11:31.23 | FuriousGeorge | <PROTECTED> |
| 11:31.33 | achu | no |
| 11:31.41 | achu | its on another machine |
| 11:31.43 | FuriousGeorge | so then you must have NAT correct |
| 11:31.47 | FuriousGeorge | ? |
| 11:32.28 | achu | the asterisk server is on local network |
| 11:32.30 | FuriousGeorge | the answer is almost certainly "yes" |
| 11:32.55 | FuriousGeorge | so, this brings us to some important questions: |
| 11:33.08 | FuriousGeorge | #1 do you know how to forward a port to your asterisk server |
| 11:33.39 | achu | using ssh ? |
| 11:33.55 | FuriousGeorge | the answer is almost certainly "no" |
| 11:34.42 | FuriousGeorge | achu: if you dont have some nice web front end to do it through |
| 11:35.21 | *** join/#asterisk A[s]H (n=Birba@host5-205.pool80183.interbusiness.it) |
| 11:35.29 | A[s]H | people have problem please |
| 11:35.31 | achu | I am using vyatta(debian based) in my router |
| 11:35.42 | A[s]H | I have 3 trunks and 3 extensions |
| 11:35.44 | FuriousGeorge | then join #iptables and ask in there, explain you need to forward port 5060 udb and 10000 - 20000 udb to your computer and you dont know how |
| 11:36.00 | FuriousGeorge | or see support for your debian based router |
| 11:36.10 | A[s]H | i want to associate an extension whit a trunk |
| 11:36.12 | A[s]H | how i can? |
| 11:36.37 | FuriousGeorge | A[s]H: i put zap trunks in a group |
| 11:36.54 | FuriousGeorge | i assume PRI works the same way |
| 11:37.16 | FuriousGeorge | can anyone tell me why * 1.4 stops running and doesnt dump a core? |
| 11:37.18 | A[s]H | example ext 201 musti use trunk 1 (outbound) |
| 11:37.26 | A[s]H | 202 must use trunk 2 (outbound) ... |
| 11:37.49 | FuriousGeorge | then put extension 202 in a context so that when someone dials from there they must use trunk 1 |
| 11:38.23 | FuriousGeorge | ~s/202/201 |
| 11:38.37 | FuriousGeorge | but you get the idea |
| 11:38.37 | A[s]H | :( |
| 11:38.42 | A[s]H | i havent understand |
| 11:39.09 | FuriousGeorge | you are asking a basic question that is not easy to explain |
| 11:39.15 | A[s]H | :( |
| 11:39.17 | jacq | hey.. i have a customer with CID set for all his extensions, plus a CID set for its outgoing trunk. Seems like the CID set for extensions override the CID of the trunk. Any know way to give trunk CID priority? |
| 11:39.31 | jacq | trunk = peer in sip conf |
| 11:39.53 | A[s]H | can u explain me |
| 11:39.55 | FuriousGeorge | jacq: before you dial out set(CALLERID(num)) |
| 11:39.55 | A[s]H | please |
| 11:39.59 | A[s]H | i came back soon |
| 11:40.02 | A[s]H | exuce se |
| 11:40.31 | *** join/#asterisk JT (n=jon@unaffiliated/jt) |
| 11:40.43 | *** join/#asterisk appletizer (n=erktjgek@77-100-109-37.cable.ubr04.hawk.blueyonder.co.uk) |
| 11:40.53 | jacq | FuriousGeorge: thanks |
| 11:41.15 | appletizer | is Asterisk a good software for T.38 faxing service to be setup on a VoIP server, or would it be an overkill? |
| 11:41.31 | JT | no |
| 11:41.48 | JT | asterisk has no T.38 endpoint support |
| 11:42.20 | *** join/#asterisk e-milio (n=emilio@adsl-3-252-172.mia.bellsouth.net) |
| 11:42.30 | appletizer | JT, ah... i must have misread, thanks |
| 11:42.34 | e-milio | hello |
| 11:44.51 | FuriousGeorge | around 4pm yuesterday asterisk 1.4, after 36 hours installed, stopped logging/running, and didnt dump a core. anything i can do to at least take a step toward filing a bug report, or should i switch from crashing version back to the weekly deadlocks of asterisk 1.2.X |
| 11:44.54 | *** join/#asterisk Fieldy (i=wyyvSnGs@gentoo/contributor/Fieldy) |
| 11:45.01 | *** part/#asterisk appletizer (n=erktjgek@77-100-109-37.cable.ubr04.hawk.blueyonder.co.uk) |
| 11:45.22 | FuriousGeorge | ive run out of components to swap, so any suggestions would be appreciated |
| 11:46.47 | e-milio | hello all |
| 11:47.25 | e-milio | and Asterisk 1.4, extensions.conf i created a frompstn context to receive calls, but everytime it says invalid extensions |
| 11:48.24 | e-milio | it is like everycall is dialing 305 (the areacode) as an extension |
| 11:59.23 | e-milio | but i am just calling in |
| 11:59.23 | e-milio | is there a way to "clean" the extensions dialed when the calls comes in? |
| 11:59.24 | *** join/#asterisk docelmo (n=vircuser@c-76-99-157-112.hsd1.de.comcast.net) |
| 11:59.24 | JT | FuriousGeorge: 1.4.0? |
| 11:59.24 | FuriousGeorge | 1.4.4 |
| 11:59.24 | *** join/#asterisk KeltusEx (n=Keltus@c-76-21-119-166.hsd1.ca.comcast.net) |
| 11:59.24 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 11:59.24 | FuriousGeorge | i have to say, crashing is worse than when 1.2.X would deadlock. at least then "the phones would go haywire", as the client describes it, and you know you need to restart |
| 11:59.25 | *** join/#asterisk mihinomenest (n=argh@cerebus.clandestineresearch.com) |
| 11:59.25 | FuriousGeorge | how can i verify the compiler flags i used to build asterisk? |
| 11:59.25 | JT | is this the same system that had deadlock trouble in 1.2? |
| 11:59.25 | Uatec_ | FuriousGeorge, is it often that it crashes on you? |
| 11:59.25 | *** join/#asterisk drrt (n=junior@ip242-64.baltnet.ru) |
| 11:59.25 | drrt | ~pb |
| 11:59.38 | jbot | rumour has it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org |
| 11:59.38 | FuriousGeorge | JT: yeah, you remembered, im impressed |
| 11:59.38 | *** join/#asterisk Fieldy (n=toon@gentoo/contributor/Fieldy) |
| 11:59.38 | JT | FuriousGeorge: what kernel? |
| 11:59.38 | FuriousGeorge | i never resolved that and ran out of components to swap. thing is the identical hardware runs elsewhere with no issues |
| 11:59.38 | drrt | hi. post me a link to pastebin plz |
| 11:59.38 | JT | did you swap the motherboard and cpu and ram |
| 11:59.39 | FuriousGeorge | JT: 2.6.18 |
| 11:59.39 | JT | and is it performing an identical function to other machines? |
| 11:59.40 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 11:59.40 | drrt | i`d like to introduce some debug |
| 11:59.40 | FuriousGeorge | JT: not the cpu, but the other two i have swapped, ram recently and mb about 2 months ago |
| 11:59.41 | JT | hrm ok |
| 11:59.41 | FuriousGeorge | JT: not identical no, thats where the divergence begins |
| 11:59.41 | FuriousGeorge | one place parks calls all the time the other transfers |
| 11:59.41 | *** join/#asterisk znoG_ (n=gs@201.235.180.235) |
| 11:59.42 | FuriousGeorge | i upgraded to 1.4.4 because i was using the metermaid patch |
| 11:59.42 | FuriousGeorge | i thought that might have been causing it |
| 11:59.43 | FuriousGeorge | and it still might be, like i said, 1.4.4 doesnt deadlock, it just crashes |
| 11:59.43 | FuriousGeorge | ive run memtest and prime95 on the server |
| 11:59.43 | FuriousGeorge | i think im gonna try a priest next |
| 11:59.44 | *** join/#asterisk hijacked (n=argh@66.255.220.17) |
| 11:59.44 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 11:59.44 | FuriousGeorge | no indication in the logs as to why. i was getting a lot of "Maximum retries exceeded", i read that sip peers configured but not interfaced could cause that, so i commented them out |
| 11:59.44 | FuriousGeorge | we'll see if that helps, but i am way less than optimistic |
| 11:59.45 | *** join/#asterisk floppp (n=flop@nat-staff.b3g-telecom.com) |
| 11:59.47 | *** join/#asterisk docelmo (n=vircuser@c-76-99-157-112.hsd1.de.comcast.net) |
| 11:59.47 | *** join/#asterisk msetim (n=msetim@200.195.161.164) |
| 11:59.48 | msetim | good morning :) |
| 12:01.07 | *** join/#asterisk JT_ (n=jon@unaffiliated/jt) |
| 12:01.07 | *** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com) |
| 12:01.08 | drrt | http://pastebin.mozilla-russia.org/15143 here is some interesting thing. cdr record is |
| 12:01.08 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 12:01.08 | *** join/#asterisk Fieldy (i=rrZvFZQ6@gentoo/contributor/Fieldy) |
| 12:01.09 | drrt | the system pastes cdr record after timeout of 20secs |
| 12:01.09 | FuriousGeorge | is it safe to just attach gdb to asterisk all day? |
| 12:01.09 | *** join/#asterisk MooingLemur (n=troy@unaffiliated/mooinglemur) |
| 12:01.36 | FuriousGeorge | could that somehow lead to it crashing more often? |
| 12:01.37 | JT_ | probably |
| 12:02.02 | *** join/#asterisk `Sauron (n=sauron@dsl001-130-033.aus1.dsl.speakeasy.net) |
| 12:02.28 | FuriousGeorge | JT_: would that invalidate my bug report, i wonder |
| 12:02.35 | *** join/#asterisk mindCrime_ (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 12:02.43 | JT_ | who knows |
| 12:02.49 | FuriousGeorge | i mean, who really cares if it goes 18 or 26 hours between crashes |
| 12:03.00 | FuriousGeorge | JT_: no one listening to me right now, thats for sure |
| 12:03.14 | JT_ | FuriousGeorge: try changing the whole computer? |
| 12:03.29 | FuriousGeorge | i basically did |
| 12:03.35 | FuriousGeorge | it would be the second time |
| 12:03.38 | JT_ | you left the cpus |
| 12:03.52 | FuriousGeorge | yeah, but in my experience CPUs either work or dont |
| 12:03.55 | JT_ | your cpus could be damaged by heat |
| 12:04.50 | JT_ | not so |
| 12:04.50 | FuriousGeorge | and i ran prime95 |
| 12:04.50 | FuriousGeorge | for 16 hours |
| 12:04.50 | JT_ | cpus have redundant transistor gates |
| 12:04.50 | JT_ | when all the redundancy burns out |
| 12:04.50 | JT_ | you start getting unexplained crashes and errors |
| 12:04.50 | *** join/#asterisk FlatFoot (n=simon@80.88.192.83) |
| 12:04.51 | *** join/#asterisk jkiff (n=jkiffmey@unaffiliated/vorondil) |
| 12:04.58 | FuriousGeorge | JT_: im not saying i dont believe you, but wouldnt those crashed and errors manifest when proving mersene primes on both cores |
| 12:05.13 | ManxPower | I don't believe you! |
| 12:05.32 | ManxPower | FuriousGeorge: Do you get a coredump with the crash? |
| 12:05.37 | FuriousGeorge | nope |
| 12:05.48 | FuriousGeorge | i did build with DONT_OPTIMIZE and DEBUG_THREADS |
| 12:05.48 | JT_ | possibly, but i dunno, i wouldn't trust software, it's the only hardware variable left, is it not? |
| 12:05.59 | JT_ | dont optimise? |
| 12:06.00 | ManxPower | FuriousGeorge: just a hardlock of Asterisk? |
| 12:06.41 | FuriousGeorge | ManxPower: i have no idea cuz they just reboot it and dont even call me anymore, but where 1.2.X would deadlock, this just stopped logging about 15 minutes befoer the rebooted it |
| 12:07.07 | ManxPower | FuriousGeorge: you are using the latest 1.4.x release? |
| 12:07.08 | FuriousGeorge | ManxPower: so my new strategy is attach gdb to * and let it run and see if nothing blows up or what |
| 12:07.11 | FuriousGeorge | ManxPower: yes |
| 12:07.16 | FuriousGeorge | 1.4.4 |
| 12:07.46 | ManxPower | FuriousGeorge: What options are shown with Asterisk in a "ps -axwww | grep asterisk" |
| 12:07.52 | ManxPower | i.e. command line params |
| 12:09.07 | FuriousGeorge | ManxPower: none... /usr/sbin/asterisk |
| 12:09.25 | FuriousGeorge | ManxPower: i should mention that i have an identical machine with different configs that never crashes |
| 12:09.39 | FuriousGeorge | good machine's users transfer mostly, others park |
| 12:09.49 | drrt | FuriousGeorge, do u ve any E1 int cards in the system ? |
| 12:09.55 | ManxPower | FuriousGeorge: add -g to it. That should force a coredump |
| 12:09.56 | FuriousGeorge | drrt: no |
| 12:10.04 | FuriousGeorge | or rite |
| 12:10.07 | FuriousGeorge | oh* |
| 12:10.12 | JT_ | drrt: "u ve"? |
| 12:10.20 | ManxPower | FuriousGeorge: Oh, I think it is hardware too, but 1.4.x does not have a good track record of stability. |
| 12:10.31 | drrt | JT_, do you have. such dirty slang |
| 12:10.35 | ManxPower | so you can't ignore that. |
| 12:10.43 | JT_ | FuriousGeorge: changed power supply? |
| 12:11.11 | FuriousGeorge | ManxPower: most recently i swapped ram for new ecc ram, and tdm400p for sangoma a200. i just wanna know for sure at this point |
| 12:11.31 | ManxPower | We had a standby mail server that randomly hard locked. turned out to be the power supply. |
| 12:12.00 | drrt | FuriousGeorge, can you unplug new hardware |
| 12:14.18 | FuriousGeorge | JT_: not in the last 3 months, but yes |
| 12:14.25 | FuriousGeorge | i just caught it crashing |
| 12:16.00 | FuriousGeorge | http://pastebin.ca/539903 |
| 12:16.22 | FuriousGeorge | that is all greek to me, anyone know if its enough to file a bug report? |
| 12:16.32 | FuriousGeorge | or did i just crash it by attaching gdb to it |
| 12:18.46 | ManxPower | FuriousGeorge: if you get a core file I can MAYBE understand enough to recommend a bug report or not. |
| 12:19.40 | ManxPower | I'm afraid of 1.4 |
| 12:22.32 | *** join/#asterisk penguinFunk (n=penguin@87.224.86.46) |
| 12:22.47 | FuriousGeorge | well i started asterisk myself with safe asterisk. my distros init.d scripts seem fubar. ill have a core dump by the end of the day with any luck |
| 12:23.04 | ManxPower | Cool. |
| 12:23.19 | ManxPower | I'll be here off and on until about 5pm central time |
| 12:23.30 | ManxPower | I have to climb under a house today to run some cable. |
| 12:23.30 | FuriousGeorge | ManxPower: thanks for the time |
| 12:23.38 | ManxPower | I hate running wire. |
| 12:23.44 | FuriousGeorge | dont we all :) |
| 12:25.00 | ManxPower | I don't normally do it, but the owner's house at the campground only has 2 pair red/green/yellow/black cable and one of those pairs is broken. |
| 12:25.20 | ManxPower | hard to install 2nd extension with that cable |
| 12:26.23 | *** join/#asterisk snook3r (n=ariel@bzq-219-46-202.isdn.bezeqint.net) |
| 12:27.05 | *** join/#asterisk VJFROMGT (n=vjfromgt@user-387g9ui.cable.mindspring.com) |
| 12:27.27 | VJFROMGT | iax2 trunk keeps becomming unreachable, any one know how to fix this? |
| 12:27.55 | ManxPower | VJFROMGT: Turn off qualify or turn on qualify smoothing |
| 12:28.31 | LeddyHM | get a better prodiver works too ;) |
| 12:29.06 | *** join/#asterisk DrukenLPY (n=jdumais@CPE000e08cb2a29-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 12:29.25 | VJFROMGT | maxpower,, what is the line for qualify smoothing |
| 12:29.34 | VJFROMGT | leddy,, this is between 2 * boxes |
| 12:30.27 | ManxPower | VJFROMGT: you would have to check the sample config file. |
| 12:32.17 | VJFROMGT | will google |
| 12:33.44 | *** join/#asterisk oej (n=olle@guest-rocq-135234.inria.fr) |
| 12:35.05 | *** join/#asterisk JT (n=jon@unaffiliated/jt) |
| 12:35.15 | VJFROMGT | is qualify = 60 000 the same as smoothing?> |
| 12:36.15 | ManxPower | qualify smoothing tells Asterisk not to consider the remote side unreachable just because 1 packet was lost. |
| 12:36.40 | VJFROMGT | what does that line look like? |
| 12:36.53 | JT_ | if his 2 boxes are on a lan, they should not be losing packets |
| 12:36.58 | VJFROMGT | is 1 packet literal or not? |
| 12:37.21 | VJFROMGT | boxes are over vpn in different continents there is 2% loss at all time |
| 12:37.37 | ManxPower | Without qualify smothing, if 1 qualify packet is lost then the host is considered down |
| 12:37.38 | *** join/#asterisk _omer (n=omer@lhr-mp-dig-p11-81.brain.net.pk) |
| 12:37.38 | DrukenLPY | why loss? |
| 12:37.46 | _omer | hello |
| 12:38.23 | VJFROMGT | manx,, i am not finding a good example what the line should look like, please tell me |
| 12:38.28 | _omer | Trying to install Asterisk in CENTOS ..... getting this error when I do "make" |
| 12:38.29 | _omer | /usr/bin/ld: cannot find -lncurses |
| 12:38.29 | _omer | collect2: ld returned 1 exit status |
| 12:38.40 | VJFROMGT | drunken isps in third world sux,, |
| 12:39.09 | drrt | _omer, try to find libncurses-dev package for you distro |
| 12:39.11 | A[s]H | how can to associate extension whit trunks in outboud? |
| 12:39.16 | ManxPower | VJFROMGT: the default iax.conf.sample included in the Asterisk source code has an example of qualify smoothing |
| 12:39.30 | _omer | drrt: I can try it with yum |
| 12:39.43 | A[s]H | furiousgeorge |
| 12:39.45 | A[s]H | can u help me |
| 12:40.11 | ManxPower | A[s]H: we can't help with FreePBX hwere. |
| 12:40.12 | msetim | VJFROMGT: http://svn.digium.com/view/asterisk/branches/1.4/configs/iax.conf.sample?revision=62371&view=markup |
| 12:40.13 | ManxPower | here. |
| 12:40.17 | JT_ | VJFROMGT: then LeddyHM's suggestion in the first places was correct, get a better provider, since the 2 boxes are on other sides of the Internet |
| 12:40.22 | ManxPower | ~freepobx |
| 12:40.40 | _omer | Yum found nothing .. |
| 12:40.41 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 12:40.41 | A[s]H | why manx? |
| 12:40.41 | msetim | VJFROMGT: Looking for qualifysmoothing |
| 12:40.59 | Zeeek | for anyone following the long and boring Polycom power supply discussion of this morning |
| 12:40.59 | VJFROMGT | no options with providers |
| 12:41.00 | ManxPower | A[s]H: because Freepbx uses very, very, very complicated config files, scripts, and AGIs |
| 12:41.12 | Zeeek | I found a linksys 12VDC 1A supply |
| 12:41.13 | DrukenLPY | ~freepbx |
| 12:41.17 | jbot | from memory, freepbx is unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 12:41.17 | ManxPower | Zeeek: no. What's the issue? |
| 12:41.33 | *** join/#asterisk j23 (n=jerms@static-213-115-44-90.sme.bredbandsbolaget.se) |
| 12:41.44 | Zeeek | ManxPower it was about hum in headset using NOT poe but wallwart type supplies |
| 12:41.46 | LeddyHM | whoot |
| 12:41.50 | JT_ | VJFROMGT: perhaps, but you shouldn't have dismissed LeddyHM's suggestion |
| 12:41.51 | A[s]H | k |
| 12:41.52 | LeddyHM | Leddy: 1 |
| 12:41.55 | ManxPower | Zeeek: Ah. |
| 12:41.59 | j23 | hello hello |
| 12:42.03 | _omer | drrt: whats the correct package name?? libncurses-dev ?? |
| 12:42.20 | Zeeek | So I tried the Linksys router supply which works and does make the hum way less |
| 12:42.31 | ManxPower | _omer: you must have missed the install script someone posted to the mailing list yesterday. |
| 12:42.39 | tzanger | Zeeek: oh? bad power |
| 12:42.45 | JT_ | Zeeek: now try with a 12v battery :) |
| 12:42.48 | ManxPower | Zeeek: so you will be getting a real Polycom PS now? |
| 12:42.48 | VJFROMGT | if i use qualifysmoothing = yes do i still keep qualify=yes ? |
| 12:42.52 | _omer | I am not in mailing list.. |
| 12:42.58 | Zeeek | ironically, the ps that came with the ip500 is only rated at 400ma |
| 12:43.02 | ManxPower | _omer: It sucks to be you. |
| 12:43.09 | JT_ | Zeeek: that's all that's required |
| 12:43.19 | JT_ | polycoms are rated at 5W max or so |
| 12:43.26 | Zeeek | ManxPower this was furnished witht he phone but does not say Polycom on it |
| 12:43.38 | Zeeek | Well, my 1AM supply works so far |
| 12:43.42 | ManxPower | Zeeek: I don't think any of them do. |
| 12:43.46 | j23 | _omer:whats your problem? |
| 12:44.07 | Zeeek | I'll have to record a call to see about the hum but I think it's lower, much lower |
| 12:44.12 | ManxPower | j23: he doesn't know how to install the ncurses devel package on CENTOS |
| 12:44.21 | _omer | getting this error....when I issue make to asterisk in centos |
| 12:44.22 | _omer | /usr/bin/ld: cannot find -lncurses |
| 12:44.22 | _omer | collect2: ld returned 1 exit status |
| 12:44.25 | JT_ | Zeeek: well a higher power psu would work fine |
| 12:44.33 | Zeeek | funny I was installing centOS this morning |
| 12:44.34 | j23 | haha |
| 12:44.37 | drrt | _omer, libncurses5-dev |
| 12:44.49 | _omer | I wanted the actual name of Package duffer :p |
| 12:45.04 | Zeeek | JT the actual gain is that I don't need the 220-110 xfromer anymore |
| 12:45.05 | _omer | thanks drrt |
| 12:45.09 | ManxPower | _omer: you will have SERIOUS issues unless you understand your distro |
| 12:45.51 | ManxPower | JT_: I've setup like 10 430s, but I've never actually seen one. |
| 12:46.02 | JT_ | Zeeek: who knows why you used a 220 to 110v transformer... |
| 12:46.02 | ManxPower | all the ones I've worked with in person are 30x and 50x |
| 12:46.06 | LeddyHM | asterisk is easy as pie on centos |
| 12:46.10 | JT_ | heh ok |
| 12:46.15 | Zeeek | Mine says "Ele$* shock, Made in China" |
| 12:46.15 | Uatec_ | LeddyHM, same on PSL |
| 12:46.28 | JT_ | and debian |
| 12:46.35 | JT_ | again: distribution does not matter |
| 12:46.40 | LeddyHM | nope |
| 12:46.47 | LeddyHM | just as long as you know how to use it |
| 12:46.48 | ManxPower | LeddyHM: Maybe so, but you still have to understand your distro or Asterisk will be hell. |
| 12:46.55 | Zeeek | JT because the input says 110vac ? |
| 12:47.02 | Zeeek | remember this is last century |
| 12:47.19 | JT_ | Zeeek: it makes no sense to buy a 220v to 110v to make 12v, instead of buying a 12v psu |
| 12:47.21 | ManxPower | JT_: does your power supply say it supports 220/110 50/60 ? |
| 12:47.38 | _omer | drrt: u mean ncurses-5 ? |
| 12:47.46 | _omer | or libncurses? |
| 12:47.48 | Zeeek | JT I was so anxious to plug the phone in when I got it, I worked with what I had |
| 12:47.55 | JT | ManxPower: 100-240V 50/60Hz |
| 12:48.00 | Zeeek | then left it for three years since it worked |
| 12:48.07 | ManxPower | JT_: cool. |
| 12:48.15 | JT | it makes 24VDC @ 500mA |
| 12:48.43 | ManxPower | JT_: I'm pretty sure older phones PSU do not have polyom on the name. |
| 12:48.47 | Zeeek | I use poR |
| 12:48.53 | *** join/#asterisk jkiff (n=jkiffmey@unaffiliated/vorondil) |
| 12:48.56 | ManxPower | polycom on the PSU |
| 12:49.04 | JT | right |
| 12:49.07 | *** part/#asterisk jmls (n=jmls@62.49.235.130) |
| 12:49.15 | JT | probably aren't switchmode either |
| 12:49.23 | drrt | _omer, think so |
| 12:50.20 | Zeeek | Power on ReeferNet |
| 12:50.25 | ManxPower | Doesn't whatever package manager CENTOS uses have a search function? |
| 12:50.30 | ManxPower | Reefer. |
| 12:50.38 | ManxPower | Sorry, I just had a flashback. |
| 12:50.47 | Zeeek | HTTP_REEFER |
| 12:50.54 | drrt | ManxPower, yum has i thik |
| 12:50.56 | drrt | think |
| 12:51.42 | _omer | yes...Yum |
| 12:51.48 | _omer | but Yum found nothing... |
| 12:51.49 | *** join/#asterisk echo--- (n=echo@64.184.118.232) |
| 12:51.52 | _omer | trying to do google |
| 12:52.03 | Uatec_ | *sigh* |
| 12:52.04 | Uatec_ | digium |
| 12:52.29 | drrt | _omer, you should visit redhat repository |
| 12:53.02 | ManxPower | No matter what distro you use, you need to know it well before you use Asterisk |
| 12:53.03 | tzafrir | yum has a "search" function, but it is not as nice as apt-get / aptitude |
| 12:53.16 | drrt | there are many more packages you can imagine |
| 12:53.21 | ManxPower | urpmi -y curses |
| 12:53.28 | ManxPower | is how you do it on Mandrake/Mandriva |
| 12:54.20 | tzafrir | be aware of rpmfind.net |
| 12:54.27 | _omer | :-o |
| 12:54.44 | tzafrir | or similar repositories. You don't want to use too many third-parties RPMS . If at all |
| 12:55.30 | _omer | that's bloody Google ... who sent me to them ... :-/ ...anyhow.. |
| 12:56.10 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 12:56.55 | tzafrir | _omer, voip-info.org has a reasonable page on installing asterisk on centos. Don't follow it to the letter. But you can get package names from there |
| 12:57.09 | tzafrir | On debian: apt-get build-dep asterisk |
| 12:57.16 | _omer | alright.. |
| 12:57.19 | ManxPower | THE LIBRARY IS INCLUDED IN ALL DISTROS |
| 12:57.41 | ManxPower | this isn't some oddballd lib requirement. zillions of apps use curses |
| 12:58.15 | LeddyHM | this is just too comical |
| 12:58.21 | LeddyHM | I almost don't want to help ;) |
| 12:58.40 | drrt | ManxPower, any distro has several install levels which cannt have preinstalled ncurses library |
| 12:58.44 | LeddyHM | with a 3rd part rpm |
| 12:59.11 | ManxPower | Zeeek: I hate how practically all linux audio/video conversion apps all use the same libffmpeg, with the same bugs. |
| 13:00.51 | ManxPower | drrt: Yes, but you should at least know how to use the package manager of your distro |
| 13:02.59 | drrt | ManxPower, do you remember your 1st day with linux ? |
| 13:03.00 | j23 | ManxPower:its so frustrating to install the discover and install the dependencies for a package that you want to install. |
| 13:03.08 | j23 | oops! |
| 13:03.09 | drrt | ManxPower, or even 1st month |
| 13:03.35 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 13:03.49 | drrt | j23, package manager is the only way ! |
| 13:03.50 | Katty | morning |
| 13:04.28 | floppp | I search a good T.38 gateway. My config is "phone <--> ATA T.38 <- SIP T.38 -> a good T.38 gateway :) <- SIP or TDM -> Asterisk". Have you got some idea ? |
| 13:04.32 | j23 | Katty:have some mercy |
| 13:05.11 | Katty | j23: i have a pretty good feeling you have no clue what i look like. |
| 13:06.45 | j23 | floppp:so whats your problem |
| 13:07.38 | tzafrir | j23, right. apt-get install asterisk |
| 13:07.39 | floppp | j23 : I don't find a good gateway T.38 for my fax. |
| 13:07.53 | *** join/#asterisk irule (n=irule@189.164.43.19) |
| 13:07.59 | floppp | j23 : I have many ata T.38 gateway. |
| 13:08.42 | j23 | j23:so how can #asterisk help you? |
| 13:08.50 | tzafrir | _omer, for the sake of completeness, the name of the voip-info page: http://www.voip-info.org/wiki/view/Asterisk+Linux+CentOS , and the name of the package on centos4: ncurses-devel |
| 13:08.53 | floppp | j23 : I have understand that Asterisk can't made transcoding between T.38 and tdm (E1) or sip alaw |
| 13:09.01 | j23 | j23:did you explore spandsp with some patches? |
| 13:09.10 | j23 | hmm |
| 13:09.33 | floppp | j23: yes |
| 13:09.37 | tzafrir | floppp, Asterisk has no native support of T.38 . The best it can do is not to get in the way of a T.38 RTP transport |
| 13:10.09 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 13:10.13 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 13:10.13 | *** mode/#asterisk [+o mog] by ChanServ |
| 13:12.40 | floppp | tzafrir: I known but faxes i very important for many company |
| 13:13.21 | tzafrir | so? use an analog fax |
| 13:13.41 | j23 | tzafrir:you mean asterisk could treat t.38 as a pass through by default |
| 13:13.48 | tzafrir | or hylafax/iaxmodem |
| 13:13.51 | j23 | i may not be expressing it correctly |
| 13:14.28 | j23 | tzafrir:explain "not to get in the way" how? where we configure this behaviour? |
| 13:14.59 | tzafrir | j23, not sure exactly. But do search around a bit... |
| 13:15.29 | floppp | j23: t38pt_udptl=yes ?? |
| 13:16.07 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 13:16.32 | *** join/#asterisk jm|work (n=jm@sentry.flags.co.uk) |
| 13:19.39 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 13:20.08 | irule | why not using iaxmodem? |
| 13:20.31 | *** join/#asterisk ertyu (i=left@S010600d0b7928a07.wp.shawcable.net) |
| 13:20.37 | irule | j |
| 13:20.40 | irule | j23 |
| 13:21.09 | j23 | floppp:there is a bounty on t.38 and you know this. what are you complaining for then |
| 13:21.20 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:22.05 | *** join/#asterisk s0ck (n=m@unaffiliated/s0ck) |
| 13:22.23 | cy303 | damn, is there not an easy way to specify a min and max number of digits to get in read()? |
| 13:22.41 | cy303 | read(number,,5) to get 5 digits for example.. but I want to get min of 5 digits as well.. |
| 13:24.12 | *** join/#asterisk `Sean (i=Un1x@CPE000c258d147c-CM000a73a94167.cpe.net.cable.rogers.com) |
| 13:24.57 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 13:25.42 | drrt | tzafrir, do u ve a moment to checkout part of debug i got |
| 13:26.36 | tzafrir | drrt, not really. But pastebin it here, and Someone will |
| 13:26.46 | irule | j23 http://www.callweaver.org/wiki/CallWeaver T.38 Fax over IP (pass-through, termination and gateway) |
| 13:27.08 | drrt | http://pastebin.mozilla-russia.org/15143 |
| 13:28.13 | irule | j23 http://www.callweaver.org/wiki/T38+Compatibility+List |
| 13:28.34 | drrt | the call was still present but the system put cdr record with mess data |
| 13:30.48 | Uatec_ | hey, does anybody know any software for windows that would serve as a Soft Modem, preferably using sip. i.e. appear to the PC as a modem, but actually dial out through Asterisk over SIP over the LAN? |
| 13:31.06 | ManxPower | Uatec_: no |
| 13:32.45 | *** join/#asterisk extr3m (n=cl@213.134.125.3) |
| 13:32.47 | Uatec_ | does anybody else? |
| 13:33.23 | extr3m | Scenario: A user has call-forwarded he's extension to a cellular phone, can that user transfer calls received into some other user ? |
| 13:33.29 | extr3m | with DTMF |
| 13:33.30 | *** join/#asterisk mindCrime_ (n=chatzill@66.83.208.219.nw.nuvox.net) |
| 13:34.06 | ManxPower | extr3m: yes |
| 13:34.09 | extr3m | how? |
| 13:34.21 | ManxPower | "show application dial" also the Wiki |
| 13:34.25 | ManxPower | ~wikis |
| 13:34.37 | jbot | wikis is probably http://www.voip-info.org |
| 13:34.39 | ManxPower | and the mailing list archives. |
| 13:34.41 | ManxPower | ~mailinglist |
| 13:34.44 | jbot | Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.htmm |
| 13:35.25 | ManxPower | Uatec_: since modems works so poorly over VoIP, I doubt anyone will make such a thing. |
| 13:35.52 | Uatec_ | ok, in that case i shall ask about the actually problem i have, not just about the considered solution |
| 13:35.58 | Uatec_ | we have a Credit card processing machine |
| 13:36.05 | *** join/#asterisk De_Mon (i=de_mon@fl-71-55-184-242.dhcp.embarqhsd.net) |
| 13:36.19 | extr3m | ok, ill try |
| 13:36.20 | extr3m | thanx |
| 13:36.26 | Uatec_ | which currently dials up to the CC company with a modem on an analogue line |
| 13:36.31 | ManxPower | Uatec_: Most people install a PSTN line for fax and credit card machines. |
| 13:36.57 | ManxPower | Then you spend time doing faxes and CC transactions, not spend all your time trying to figure out why it does not work. |
| 13:37.17 | Uatec_ | well we do have a line for this |
| 13:37.42 | *** join/#asterisk Stephnie (i=Stephnie@u15157627.onlinehome-server.com) |
| 13:37.45 | Stephnie | hi |
| 13:37.47 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.139.144) |
| 13:37.53 | Uatec_ | but we're going to be removing the old PBX, since it's clumsy, crap and taking up space and being replaced by asterisk |
| 13:38.02 | ManxPower | Trying to run Data over Voice over IP will cost you money, not save you money |
| 13:38.07 | Stephnie | [codec_g723-icc-pentium.so]Jun 5 16:41:27 WARNING[11217]: loader.c:326 __load_resource: /usr/lib/asterisk/modules/codec_g723-icc-pentium.so: cannot restore segment prot after reloc: Permission denied |
| 13:38.07 | Stephnie | Jun 5 16:41:27 WARNING[11217]: loader.c:555 load_modules: Loading module codec_g723-icc-pentium.so failed! |
| 13:38.13 | Uatec_ | how will it cost money? |
| 13:38.34 | ManxPower | Stephnie: we cannot help you with pirated codec software |
| 13:38.44 | Stephnie | it is for education purposes |
| 13:38.51 | ManxPower | Uatec_: How much is your time worth? |
| 13:39.11 | ManxPower | Stephnie: Where exactly does it there is an exemption for "eduational use" |
| 13:39.23 | Uatec_ | depends |
| 13:39.24 | ManxPower | ..e.r.. Where does it say there is an exemption |
| 13:40.03 | ManxPower | A phone line costs $50/month. If you will spend more than $50/month of your time because you will always be trying to figure out why it does not work all the time. |
| 13:40.22 | ManxPower | Yes, it is good for you, but is not good for your customer |
| 13:40.29 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 13:40.47 | Uatec_ | it depends if it keeps breaking or not |
| 13:40.56 | ManxPower | Uatec_: It will. |
| 13:41.02 | Uatec_ | have you tried this before? |
| 13:41.19 | ManxPower | With fax, yes. and fas is really a 9600 baud modem connection. |
| 13:41.37 | Uatec_ | what kind of problems did you come across? |
| 13:41.42 | ManxPower | We removed Fax from Asterisk moved it back to a PSTN line and never had a problem since. |
| 13:41.56 | ManxPower | It prolly cost my customer $2,000 to try faxing. |
| 13:42.53 | ManxPower | Uatec_: %50 of the faxes simply did not complete. The more the pages, the higher the chance of the fax failing. Once you got above 5 pages or so almost all of those faxes failed. |
| 13:43.11 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:43.32 | Katty | why is moving so complicated :< |
| 13:43.34 | Stephnie | http://www.pastebin.ca/540091 ?? anyone? |
| 13:43.40 | Uatec_ | ok |
| 13:43.42 | Uatec_ | did you use http://www.voip-info.org/wiki/index.php?page=Asterisk+IAXmodem ? |
| 13:43.53 | ManxPower | Stephnie: nobody will help you. |
| 13:43.59 | ManxPower | Uatec_: no. |
| 13:44.48 | ManxPower | Stephnie: it says right in the Intel lib license that the SOFTWARE is free, but the CODEC is NOT free and that you must arrange a licence with the patent holders of the codecs. |
| 13:44.50 | tzafrir | centos folks: please pick up: http://www.voip-info.org/wiki/view/Asterisk+Linux+Centos |
| 13:45.04 | ManxPower | ~centos |
| 13:45.23 | jbot | centos is a rebuild of the Red Hat Enterprise Linux RPMs by the community. Check it out at http://www.centos.org/projects/centos |
| 13:45.23 | tzafrir | I don't intend to touch that page much more |
| 13:46.18 | ManxPower | ~centos |
| 13:46.22 | jbot | centos is a rebuild of the Red Hat Enterprise Linux RPMs by the community. Check it out at http://www.centos.org/projects/centos, or http://www.voip-info.org/wiki/view/Asterisk+Linux+Centos |
| 13:46.26 | Stephnie | is it just because you are the god father here? ;) |
| 13:46.38 | JT | Uatec_: i don't see what the old pbx has to do with it |
| 13:46.43 | ManxPower | Stephnie: nobody wants to piss off Digium. |
| 13:46.48 | JT | Uatec_: there is no reason to keep an old pbx to do faxing |
| 13:47.01 | Uatec_ | not just the faxing JT |
| 13:47.05 | ManxPower | Discussions of the pirated intel stuff is the only thing I have ever ever seen Digium remove from the mailing list archives. |
| 13:47.15 | Stephnie | 50% of this room are already pissing off |
| 13:47.27 | JT | Uatec_: what else then? |
| 13:47.33 | Uatec_ | there is a PC in the office with a CC machine and a modem that dials through the analog PBX to the CC company |
| 13:47.49 | JT | Uatec_: you can get rid of the pbx |
| 13:47.50 | ManxPower | Uatec_: why can't it just connect directly to a PSTN line? |
| 13:48.04 | *** join/#asterisk shido6 (i=shido6@d221-68-200.commercial.cgocable.net) |
| 13:50.33 | ManxPower | I suspect Digium has a contractual obligation to try to prevent usage of unlicensed codecs with Asterisk. That is pretty common with licenses for that sort of stuff. |
| 13:50.50 | Uatec_ | the state of the lines and the isdn lines... |
| 13:51.02 | ManxPower | Uatec_: use a POTS line |
| 13:51.26 | Uatec_ | we don't have any |
| 13:51.35 | ManxPower | Then get one. |
| 13:51.53 | Uatec_ | we've got enough isdn lines |
| 13:52.09 | Uatec_ | there's no point getting a pots line when there are isdn lines available |
| 13:52.11 | Uatec_ | infact |
| 13:54.53 | Uatec_ | the fax at the moment bypasses the pbx and goes straight to the isdn, so i might see if i can put the CC machine on the same isdn line |
| 13:54.53 | *** join/#asterisk seanwg (n=sean@216.249.37.157.ppp.northrock.bm) |
| 13:54.53 | ManxPower | Uatec_: Honestly I don't care how you do it. You will have problems if you try to use Fax/CC over VoIP. |
| 13:54.53 | ManxPower | The solution is to connect the fax machine direct to the telco. |
| 13:54.53 | ManxPower | Use an ISDN/POTS converter. |
| 13:54.53 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 13:55.37 | Uatec_ | i don't really know enough about isdn, but... |
| 13:55.38 | JT | they exist for a pri? |
| 13:55.39 | Uatec_ | i reckon i can do that |
| 13:56.04 | ManxPower | JT: No, but if he is too cheap to get a POTS line, he prolly has BRI. |
| 13:56.14 | Uatec_ | it's not me |
| 13:56.16 | Uatec_ | it's my boss |
| 13:56.18 | seanwg | anyone here using noka e-series phones? |
| 13:56.27 | *** join/#asterisk rdb_ (n=rdb@gw.avila.edu) |
| 13:56.30 | *** join/#asterisk waptaxi (n=cahe@45.151-224-87.telenet.ru) |
| 13:56.40 | ManxPower | seanwg: Your extensive search of the mailinglists was not helpful. |
| 13:56.47 | JT | Uatec_: there are faxing solution, but not credit card, for ip |
| 13:56.52 | Sweeper | anyone have a queue_log file with plenty of data in it? |
| 13:56.53 | seanwg | yah i search - i have the phones working perfectly just MWI issue |
| 13:56.58 | Sweeper | I need something for testing! |
| 13:57.01 | j23 | openpbx supports t.38 |
| 13:57.12 | Uatec_ | at the moment the IDSN comes in to a box with two RJ45 sockets, one of which has a cable coming out and leading in to the back of the fax server |
| 13:57.22 | JT | s/openpbx/callweaver/ |
| 13:57.31 | Uatec_ | i reckon i could get away with puttin the CC machine in to the other RJ45 socket |
| 13:57.42 | JT | Uatec_: sounds like BRI |
| 13:57.59 | JT | depends if it's an ISDN fax or not |
| 13:58.07 | JT | and if the box converts to analogue |
| 13:58.37 | Uatec_ | it is bri |
| 13:58.42 | Uatec_ | hmm |
| 13:58.56 | seanwg | is there a specific mailing list i could search? |
| 13:59.42 | Uatec_ | unfortunately the guy who knows about the fax system isn't in work todya |
| 13:59.55 | Uatec_ | although... |
| 14:00.39 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 14:01.40 | ManxPower | seanwg: not that I know of |
| 14:01.41 | ManxPower | ~mailinglist |
| 14:01.55 | jbot | Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.htmm |
| 14:01.56 | *** join/#asterisk yannj_fr (n=yannj@vpn.intelunix.fr) |
| 14:02.28 | seanwg | cool thanks |
| 14:03.25 | seanwg | nah same as before no responses |
| 14:03.42 | seanwg | funny more people are not checking these phones out |
| 14:03.47 | seanwg | nokia e-series that is |
| 14:04.17 | Uatec_ | ah, it's an eicon diva server BRI-2M 2.0 |
| 14:05.51 | Uatec_ | figures |
| 14:06.40 | *** part/#asterisk seanwg (n=sean@216.249.37.157.ppp.northrock.bm) |
| 14:07.02 | *** join/#asterisk seanwg (n=sean@216.249.37.157.ppp.northrock.bm) |
| 14:07.13 | floppp | seanwg : thomson st2030 |
| 14:08.20 | seanwg | cool looking phone |
| 14:08.36 | cy303 | seanwg: those e-series do sip or something? |
| 14:08.47 | cy303 | sip/802.11 |
| 14:08.49 | floppp | j23: I've testing callweaver but it not reassuring that this feature is not support by Asterisk |
| 14:08.49 | cy303 | ? |
| 14:08.59 | seanwg | yah the e-series are a regular gsm smartphone |
| 14:09.02 | seanwg | with wifi & sip |
| 14:09.05 | cy303 | hm |
| 14:09.09 | cy303 | do they work well tho? :P |
| 14:09.13 | seanwg | they do blackberry connect, activesync |
| 14:09.33 | seanwg | yah work super well. i had them running on another sip server but moved to asterisk this weekend for more features |
| 14:09.38 | seanwg | everything works except for MWI now |
| 14:09.40 | floppp | I'm curently testing s60 |
| 14:09.49 | seanwg | s60 runs the same software |
| 14:09.55 | seanwg | or some kind of same software |
| 14:10.02 | seanwg | does your s60 handle stun etc? |
| 14:10.23 | floppp | I don't use stun |
| 14:10.34 | ManxPower | <laugh>STUN</laugh> |
| 14:10.37 | seanwg | haha |
| 14:10.46 | seanwg | in my setup here i need stun |
| 14:11.00 | seanwg | wife travels and its the only way i know of to make it work |
| 14:11.16 | ManxPower | STUN is actually a french word. It translates into english as "almost nobody needs this" |
| 14:11.25 | cy303 | everyone has been telling me that wifi sip phones are horrible failure |
| 14:11.39 | seanwg | i think some of the previous phones were junk |
| 14:11.40 | ManxPower | It could also translate to "Only used when you don't know about all the other NAT options of Asterisk" |
| 14:11.48 | seanwg | i bought a cisco (linksys) wifi phone |
| 14:11.53 | seanwg | before |
| 14:11.59 | seanwg | it was complete and utter junk |
| 14:12.12 | seanwg | worked fine in the house but as soon as you left the house, no service |
| 14:12.29 | cy303 | hm |
| 14:12.37 | seanwg | some of the newer phones with wifi, gsm really kill though |
| 14:12.38 | j23 | seanwg:tell me about its battery life |
| 14:12.42 | j23 | seanwg:talk time? |
| 14:12.48 | seanwg | standby about 2 weeks |
| 14:12.55 | seanwg | if wifi, it runs about 8 hours |
| 14:12.58 | floppp | j23: really good |
| 14:12.58 | seanwg | gsm about 20 hours |
| 14:13.14 | cy303 | damn |
| 14:13.15 | seanwg | wifi burns batteries i have no idea why |
| 14:13.28 | seanwg | gsm i guess it can adjust its power output |
| 14:13.41 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 14:13.45 | Katty | file: come help me move. |
| 14:14.17 | seanwg | so how can asterisk run with terminals out on the greater internet behind nat routers without stun |
| 14:14.19 | j23 | Katty:the ugly girl is back? ahhhh |
| 14:14.36 | Katty | j23: i stay here. kthx. |
| 14:14.37 | ManxPower | seanwg: Asterisk has special support for NAT traversal |
| 14:15.02 | seanwg | yah it basically brings all the audio back and bridges it back |
| 14:15.22 | seanwg | sorry it tells the terminals to send audio back to the asterisk box, and then hairpins the audio back out |
| 14:15.35 | ManxPower | STUN just moves the audio bridge somewhere else, it still has to bridge |
| 14:15.51 | seanwg | the audio should flow from terminal direct to other terminal |
| 14:15.59 | seanwg | whys a bridge required? prompts etc? |
| 14:16.01 | ManxPower | seanwg: not with NAT |
| 14:16.05 | yannj_fr | well, I thought that adding thing like : 5,1,action would match while pressing 5 |
| 14:16.46 | ManxPower | Unless you do special port forwarding on the NAT router, it will not accept incoming connections from unknown IP addresses and ports. |
| 14:17.03 | ManxPower | and the NAT router does now know about the IP/port of the other terminal |
| 14:17.24 | ManxPower | ...does NOT know.... |
| 14:18.02 | floppp | Do you known where I can find an Asterisk roadmap ? |
| 14:18.03 | seanwg | yah |
| 14:18.07 | ManxPower | This problem cannot be fixed without changing the way NAT does in a way that will break pretty much everything out there. |
| 14:18.53 | *** join/#asterisk bdunn (n=bdunn@cpe-76-186-190-98.tx.res.rr.com) |
| 14:19.03 | seanwg | i got the phones working with stun now succesfully |
| 14:19.17 | seanwg | what complicates things is my asterisk server itself is behind nat |
| 14:19.22 | seanwg | another quick question - |
| 14:19.30 | ManxPower | Results 1 - 10 of about 1,070,000 English pages for asterisk road map.AND asshole too lazy to use google (0.14 seconds) |
| 14:19.42 | blitzrage | LOL |
| 14:19.44 | seanwg | is there anyway to run asterisk on a linksys router (with the linux on it) |
| 14:19.47 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 14:19.48 | ManxPower | seanwg: it never complicated things for me. |
| 14:20.05 | ManxPower | blitzrage: my google.foo is strong |
| 14:20.16 | blitzrage | seanwg: Asterisk behind NAT works fine if you know how to set it up :) |
| 14:20.28 | blitzrage | hint: externip and localnet |
| 14:20.34 | blitzrage | ManxPower: it sure is! |
| 14:20.37 | blitzrage | breakfast time!!!! |
| 14:20.38 | ManxPower | I guess shutting down all my equipment finally made the campground owners enclose my equipment area and air condition it. |
| 14:20.49 | blitzrage | I'm gonna make and eat breakfast before 2pm for the first time in a week and a half |
| 14:20.56 | seanwg | i got it working behind nat |
| 14:21.09 | ManxPower | blitzrage: you are in danger of losing your geek card if you keep doing that |
| 14:21.21 | blitzrage | ManxPower: I know! but I gotta do it at least once... I'm starving |
| 14:21.25 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:21.39 | *** join/#asterisk rgsteele (n=rgsteele@nat-pool.agora-net.com) |
| 14:23.54 | rgsteele | Hey folks. I've got a weird problem with asterisk 1.2.13 and zaptel 1.2.11. I'm behind a ChannelBank given to us by the phone company, and whenever anybody hangs up, I get either A) 3:00 dial tone messages or B) I get the fast-busy (a bunch of really fast, sequential beeps), which causes asterisk to go into a voicemail loop, logging 10 second voicemail messages until we restart the asterisk box. |
| 14:24.45 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 14:26.38 | rgsteele | I think the CB has bad hangup detection support, and the busydetect option isn't stable enough to use. Anybody have any suggestions as to what the problem might be, and/or how to fix it? I've come up empty so far. |
| 14:26.38 | JT | change to pri? :) |
| 14:26.38 | rgsteele | JT: pri? |
| 14:26.38 | JT | primary rate interface |
| 14:26.38 | JT | as in common channel signalling |
| 14:26.38 | *** part/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 14:27.15 | Mercestes | rgsteele, Is this happening when internal users hangup, or external users hangup? |
| 14:27.15 | Mercestes | rgsteele, or both? |
| 14:27.16 | *** join/#asterisk Cresl1n (i=matt@nat/digium/x-c66e22bde76f711d) |
| 14:27.17 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 14:27.39 | *** join/#asterisk TimTF (n=mlske@87-237-10-201.office-bru2.benesol.be) |
| 14:29.17 | *** join/#asterisk mirco (n=mirco@88.128.29.86) |
| 14:31.41 | TimTF | I'm trying to get more then one trunk to work (one provider, multiple account). Yet once there is more then one defined (and properly registered) only one inbound trunk can be used. |
| 14:31.55 | TimTF | How can I get around this? |
| 14:34.07 | *** join/#asterisk eeos (n=eeos@86.53.50.16) |
| 14:34.56 | eeos | hi everybody |
| 14:35.20 | *** join/#asterisk _VoicePulse (n=contact@unaffiliated/voicepulse) |
| 14:36.22 | *** join/#asterisk s0ck (n=m@unaffiliated/s0ck) |
| 14:36.24 | rgsteele | Mercestes: external |
| 14:36.51 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:36.51 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:36.52 | Katty | Mercestes: moving is too complicated :< |
| 14:37.10 | Katty | anthm: do you think i should leave early on thursday after the conference, or go home fridayy? |
| 14:37.41 | rgsteele | JT: Sorry, I suppose I'm a bit green on that subject. Any docs that would point to where I can learn about what pri is and how to change my * box to use it instead of - well, whatever it's using now? |
| 14:37.52 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 14:37.56 | anthm | well it really ends thu eve |
| 14:38.03 | rgsteele | (is it a substitute for zaptel?) |
| 14:38.07 | anthm | so friday would be better if you dont want to miss any |
| 14:38.14 | De_Mon | how do I remove all members from a queue? |
| 14:38.24 | anthm | shut the server off |
| 14:39.19 | Katty | anthm: yeah, but amtrak leaves at 4 |
| 14:39.22 | Katty | anthm: on thursday |
| 14:39.39 | Katty | anthm: and i don't really wanna go to amtrak by myself |
| 14:40.21 | Katty | anthm: if i leave friday, everyone will already be gone :< |
| 14:40.24 | anthm | i think there are a lot of ppl there still fri |
| 14:40.28 | Katty | k |
| 14:40.44 | anthm | for sure we will be so if all else fails we will escort ya =D |
| 14:40.50 | Katty | kk |
| 14:41.08 | Katty | i think i might need an escort to the hotel too |
| 14:41.13 | Katty | maybe i can bribe someone. |
| 14:43.12 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 14:46.19 | anthm | yah it's pretty close |
| 14:46.26 | anthm | we can arrange that |
| 14:46.43 | Katty | mkay |
| 14:49.49 | tzanger | where are you moivng to? |
| 14:49.51 | tzanger | moving even |
| 14:51.34 | JT | rgsteele: you'd need to get a pri card, and get the telco to change the service from CAS/RBS signalling to PRI |
| 14:51.36 | *** join/#asterisk mik3 (n=43b8ee33@alcor.lunarpages.com) |
| 14:51.39 | JT | it may or may not be practical |
| 14:51.41 | mik3 | good morning/day |
| 14:51.51 | JT | but it'd take the channel bank out of the equation |
| 14:51.55 | JT | and analogue signalling |
| 14:52.36 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 14:52.37 | mik3 | how difficult is it to set up an asterisk box for someone who's never used it, and has moderate knowledge in unix/linux? |
| 14:52.43 | tzanger | mik3: not really |
| 14:52.48 | mik3 | ok cool. |
| 14:52.48 | tzanger | there's some stuff ot learn and lots of head scratching |
| 14:53.02 | tzanger | but so long as you don't do something stupid like try to deploy into production wihtout having screwed with it for a couple weeks, you're fine |
| 14:53.15 | mik3 | well i'll have a week for trial and error and a lot of red bull, anyone recommend this asterisknow instead of centos with asterisk? |
| 14:53.24 | JT | nup |
| 14:53.26 | tzanger | mik3: I don't use GUIs |
| 14:53.29 | JT | recommend the book |
| 14:53.31 | JT | ~thebook |
| 14:53.36 | jbot | from memory, thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 14:54.03 | cy303 | I pre-orderd the asterisk cookbook |
| 14:54.11 | cy303 | not out until like mid august or something :/ |
| 14:54.12 | Uatec_ | cy303, url? |
| 14:54.19 | Mercestes | Katty, *hugs* I agree. |
| 14:54.22 | cy303 | I just found it on amazon.. it's o'reilley |
| 14:54.24 | *** join/#asterisk coppice (n=chatzill@10.198.17.210.dyn.pacific.net.hk) |
| 14:54.51 | cy303 | http://tinyurl.com/yu3rz3 |
| 14:55.02 | puzzled | hi |
| 14:55.18 | mik3 | well i was thinking of using the asterisknow distro because i'll be revamping a small ecommerce companies old asterisk setup and they don't have any linux knowledge |
| 14:55.23 | *** join/#asterisk heh_v_water (n=heh_v_wa@70-57-205-181.hlna.qwest.net) |
| 14:55.27 | mik3 | so maybe the gui will be better for them once i am off the project? |
| 14:55.38 | JT | but they shouldn't be administering a pbx really |
| 14:56.00 | mik3 | h,. |
| 14:56.23 | mik3 | well once i get a centos/asterisk setup how difficult would it be for me to administrate it remotely? |
| 14:56.36 | JT | ever used ssh? |
| 14:56.40 | mik3 | of course |
| 14:56.44 | mog | then easy |
| 14:56.50 | JT | then there's your answer :) |
| 14:56.52 | mik3 | i mean, is it feasible to manage remotely? |
| 14:57.02 | JT | sure, unless hardware catches fire |
| 14:57.06 | mik3 | right |
| 14:57.31 | mik3 | alright i'll start reading this doc, thanks guys. |
| 14:57.35 | mog | you dont have fire extinguishers of serial jt |
| 14:57.55 | JT | mog: i expect the datacentre NOC to do they job :P |
| 14:57.59 | JT | their |
| 14:59.05 | mosty | mik3, try it yourself first, see what you think |
| 14:59.26 | rgsteele | JT: Yeah, Not practical unfortunately. |
| 14:59.45 | *** join/#asterisk andyd (n=andyd@host90-152-23-30.ipv4.regusnet.com) |
| 14:59.46 | JT | rgsteele: you checked? |
| 15:01.04 | rgsteele | Unless the cost of a PRI card is cheap. We're a pretty small business with a modest budget. |
| 15:01.23 | rgsteele | And, I'm not sure the telco would help us out there, even if we had a pri card. |
| 15:01.34 | JT | the fact you have a channel bank implies you already have a T1 |
| 15:01.40 | JT | it just needs conversion to PRI T1 |
| 15:01.42 | rgsteele | We do. |
| 15:01.52 | JT | but they may charge a lot for this |
| 15:02.52 | rgsteele | JT: So, we'd need that, as well as a pri card? |
| 15:02.58 | rgsteele | And, then ditch the CB? |
| 15:03.15 | mik3 | mosty: we use asterisk at work, the person that set it up is an idiot and we have a plethora of problems with the phone every week, but if someone wants to pay me well to set it up for their ecommerce company sure why not. |
| 15:03.38 | mik3 | i guess it will be a good learning experience |
| 15:04.04 | mosty | heh well expect to have your own plethora of problems, in your first install |
| 15:04.11 | JT | rgsteele: yes |
| 15:05.11 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 15:08.06 | *** join/#asterisk snook3r (n=ariel@bzq-219-46-202.isdn.bezeqint.net) |
| 15:08.20 | Uatec_ | mik3, i manage our phone system remotely, infact, i'm developing our phone system remotely |
| 15:08.41 | Uatec_ | the only times i touch the hardware are when i'm stepping up to the next phase of development, i.e. putting in new cards, or connecting new lines |
| 15:10.45 | tzanger | Katty: where are you moving to? |
| 15:11.16 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 15:12.22 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 15:12.44 | rob0 | On that midnight train ... to Jawjuh |
| 15:14.01 | *** part/#asterisk oej (n=olle@guest-rocq-135234.inria.fr) |
| 15:15.53 | mik3 | and you just use a voip phone of some sort to test it? |
| 15:16.16 | Uatec_ | yeah |
| 15:16.24 | Uatec_ | well, we ahve 4 voip phones |
| 15:16.29 | Uatec_ | and a B410p |
| 15:16.37 | Uatec_ | but it just sits down the stairs behind me |
| 15:16.38 | mik3 | i don't know what that is |
| 15:16.40 | Uatec_ | i never touch it |
| 15:16.45 | Uatec_ | it's an ISDN card |
| 15:16.48 | mik3 | ah |
| 15:17.56 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 15:17.59 | *** join/#asterisk ivanfm (n=ivanfm@c93481ec.virtua.com.br) |
| 15:20.53 | *** join/#asterisk bbryant (i=brett@nat/digium/x-7b9936eed2cd52f8) |
| 15:21.43 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 15:22.59 | *** join/#asterisk jm|work (n=jm@sentry.flags.co.uk) |
| 15:23.38 | *** join/#asterisk osiris (n=osiris@c-71-205-27-131.hsd1.mi.comcast.net) |
| 15:25.15 | De_Mon | I need to get at the list of dynamic queue members in a queue |
| 15:25.34 | De_Mon | or some way of removing all memebers from a queue |
| 15:26.09 | *** join/#asterisk Cas1noGuy (n=shinklej@oh-76-5-244-253.dhcp.embarqhsd.net) |
| 15:28.00 | *** join/#asterisk hfb (n=hfb@pool-72-87-254-188.lsanca.dsl-w.verizon.net) |
| 15:28.10 | *** join/#asterisk [hC] (n=hardcore@190.10.13.145) |
| 15:29.23 | *** join/#asterisk vader-- (n=me@c-71-226-197-0.hsd1.nj.comcast.net) |
| 15:29.24 | mosty | de_mon: show queue <name>, i believe |
| 15:29.27 | vader-- | hola |
| 15:29.43 | vader-- | any of you guys ever pay for a VOIP or network infrastructure audit? |
| 15:30.24 | Corydon76-work | Why pay, when we can do it ourselves? |
| 15:30.34 | De_Mon | mosty right.. through the dialplan though |
| 15:30.45 | vader-- | im looking at a main office and 8 branch locations that are all experiencing VOIP quality issues. |
| 15:31.09 | mosty | de_mon: use realtime queue memberships, then you can write AGI scripts to modify the db |
| 15:31.15 | vader-- | they use VOIP between the branch and main office through microsoft VPN and then they do VOIP out to their main teleco carriar |
| 15:31.21 | De_Mon | ahhh |
| 15:31.22 | vader-- | over 2xT1 |
| 15:31.23 | *** join/#asterisk [GuS] (n=gdnet@unaffiliated/gus/x-663402) |
| 15:32.13 | De_Mon | id rather keep this simple |
| 15:32.14 | Qwell[] | vader--: MS VPN? There's your problem |
| 15:32.14 | Corydon76-work | vader--: set up a monitoring port on one of the main managed switches and log the traffic, for starters |
| 15:32.14 | [GuS] | Hi people! |
| 15:32.14 | vader-- | they are going to get charged like 2Kish |
| 15:32.14 | vader-- | is 2000 too much or not enough for that type of audit? |
| 15:32.14 | sevard | I'd charge one billion dollars. |
| 15:32.58 | [GuS] | I have a doub... i am installing AsteriskNOW, and now configuring, i have a doub about the Service Provider part... can i be my own service provider with that server? |
| 15:32.58 | *** join/#asterisk jm|work (n=jm@sentry.flags.co.uk) |
| 15:34.42 | mosty | [GuS], you should try #asterisknow for asterisknow specific questions |
| 15:34.46 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 15:35.04 | rob0 | Ugh, disconnect supervision on an X101P :( |
| 15:35.44 | *** join/#asterisk davidcsi (n=davidcsi@82.158.35.53.static.user.ono.com) |
| 15:35.44 | [GuS] | yeah sorry.. i forgot |
| 15:35.59 | [GuS] | but indeed mosty, installing only asterisk |
| 15:36.02 | [GuS] | is the same question... |
| 15:36.32 | *** join/#asterisk [TK]D-Fender (n=joe_blow@modemcable089.225-70-69.mc.videotron.ca) |
| 15:36.56 | davidcsi | hello all, is there a way to get the hostname in the dialplan? I tried ${ENV{HOSTNAME}} but it doesn't work, as apparently $HOSTNAME is a USER env variable, and not a regular env var |
| 15:38.55 | mosty | [GuS], depends what you mean by service provider. that probably refers to some remote server that you will send calls to. you don't have to do that if you don't want to |
| 15:39.37 | mosty | davidcsi, you could get the output from the hostname command, using System ? |
| 15:39.55 | sevard | davidcsi: there's ${SYSTEMNAME} which is defined in asterisk.conf |
| 15:40.01 | [GuS] | ok, if i only want to comunicate VoIP via internet only, do i need that service provider? (i mean iaxtel.com example) |
| 15:40.41 | [TK]D-Fender | [GuS]: Who are you looking to connect with? |
| 15:40.45 | davidcsi | i put the systemname variable but ${SYSTEMNAME} doesn't return anything... although i only reloaded, i did not stop and start... do it have to sopt and start? |
| 15:41.00 | [GuS] | [TK]D-Fender: i am installing my own server.. |
| 15:42.15 | mosty | [GuS], if you don't need to connect with another server, then you aren't using a service provider |
| 15:42.17 | [TK]D-Fender | [GuS]: You say you want to do "voip over the internet". I'm asking with WHOM. |
| 15:42.27 | *** join/#asterisk zeeesh (i=zeeesh@14-237-154-202.wol.net.pk) |
| 15:42.28 | zeeesh | hi |
| 15:42.42 | vader-- | so would 2K be too much for that type of audit? |
| 15:42.44 | *** join/#asterisk macTijn (i=martijn@linda.net.insecure.nl) |
| 15:42.52 | [GuS] | [TK]D-Fender: i want only to calls PC-PC |
| 15:43.02 | [GuS] | so i dont need that service provider right? |
| 15:43.25 | [TK]D-Fender | [GuS]: Don't even need * |
| 15:43.42 | [TK]D-Fender | [GuS]: But correct, you sure don't need a provider |
| 15:44.12 | [TK]D-Fender | [GuS]: I'm gueesing you just want to have some remote phones (soft/hard) at another place and chat for free with them? |
| 15:44.22 | [GuS] | ok, so i inly need that VoIP service provider for land lines, cell phones calls right? |
| 15:44.23 | [GuS] | yep |
| 15:44.26 | [GuS] | only that |
| 15:44.29 | [GuS] | example, wengophone |
| 15:44.31 | [TK]D-Fender | Correct them |
| 15:44.38 | [TK]D-Fender | then* |
| 15:44.49 | [GuS] | Thanks, this clarify my doub :) |
| 15:44.53 | [TK]D-Fender | Forget Wengo and get a stadard SIP phoone |
| 15:45.02 | [GuS] | yeah, just example |
| 15:45.14 | [TK]D-Fender | [GuS]: Fair enough. |
| 15:45.19 | [GuS] | :) |
| 15:45.30 | [TK]D-Fender | [GuS]: I suggest Idefisk and x-lite for windows |
| 15:45.59 | [GuS] | i just use linux sorry :P |
| 15:46.34 | [TK]D-Fender | [GuS]: KIAx or Ekiga then ;) |
| 15:46.58 | [GuS] | why not wengo? for me works great :) |
| 15:47.05 | [GuS] | yeah.. i know those clients too |
| 15:47.22 | [TK]D-Fender | [GuS]: Does some proprietary stuff. If you've got it working with * already, fine |
| 15:48.05 | [GuS] | mmm, so wengo does some propietary stuff.. like? if you dont mind to tell me |
| 15:48.09 | [GuS] | i didnt know that :S |
| 15:48.23 | *** join/#asterisk wunderkin (i=wunderki@ip68-104-149-97.ph.ph.cox.net) |
| 15:50.18 | [TK]D-Fender | [GuS]: last I checked there were some minor annoyance differences like PhoneGnome had. I could be mistaken however. |
| 15:51.15 | [GuS] | mmm. |
| 15:51.40 | *** join/#asterisk Xen^ (n=linux@unaffiliated/lnux/x-10290) |
| 15:51.56 | rob0 | Okay, this might be a FAQ, but ... my junky X101P clone isn't detecting ring on incoming calls. I thought I had zaptel.conf right; ztcfg is happy. |
| 15:52.21 | mosty | rob0, is it set to the correct country mode? |
| 15:52.39 | *** join/#asterisk Strom_M (n=strom@192.41.247.50) |
| 15:53.08 | De_Mon | bah, theres a QUEUE_MEMBER_LIST function on voip-info, but it doesn't exist in my version |
| 15:53.09 | rob0 | loadzone = us, yes |
| 15:53.45 | [TK]D-Fender | rob0: Check your zapata |
| 15:53.51 | *** join/#asterisk monstertruck (n=monstert@c-75-74-251-82.hsd1.fl.comcast.net) |
| 15:54.24 | monstertruck | hi, on a spa3102, how do I set a password to the setup? |
| 15:54.42 | monstertruck | the unit is unlocked, im the itsp |
| 15:55.01 | rob0 | This is a new 1.4.4, is the zapata library still required? If so that's what I missed. |
| 15:55.49 | [TK]D-Fender | rob0: zapata.conf....... |
| 15:56.04 | rob0 | ok |
| 15:57.32 | krdian_ | hello |
| 15:58.11 | krdian_ | anybody sending video streaming through asterisk ? |
| 15:58.53 | krdian_ | for ex. 3gpp movies ? |
| 15:59.12 | Qwell[] | 3 gigabytes per pixel? |
| 15:59.50 | *** join/#asterisk andyd (n=andyd@213-228-240-161.dsl.prodigynet.co.uk) |
| 15:59.52 | *** join/#asterisk gardo (n=gardo@121.97.193.142) |
| 16:00.16 | rob0 | <PROTECTED> |
| 16:00.34 | [TK]D-Fender | rob0: You should know better... pastebin the whole mess |
| 16:00.34 | Qwell[] | rob0: You can't change those on a reload - you have to restart |
| 16:00.42 | Qwell[] | iirc |
| 16:00.55 | rob0 | ah ok |
| 16:01.07 | rob0 | yes, I'll bbl, sorry [TK]D-Fender |
| 16:01.22 | [GuS] | [TK]D-Fender: one last question, i am trying to configure the user i've created in asterisk, the extension number is the one user name i should set up in the VoIP client? |
| 16:03.02 | *** join/#asterisk snook3r (n=ariel@bzq-219-46-202.isdn.bezeqint.net) |
| 16:03.02 | [TK]D-Fender | [GuS]: those terms don't mix |
| 16:03.29 | [GuS] | so? |
| 16:03.34 | [TK]D-Fender | [GuS]: "extensions" are just patterns in extensiosn.conf. If your referring to the [something] named sectins in sip.conf, then yes |
| 16:03.44 | [GuS] | i mean, |
| 16:04.13 | [GuS] | i've created a user in asterisk, so now i am trying to set up in my VoIP client |
| 16:04.29 | *** join/#asterisk Strom_M (n=strom@192.41.247.50) |
| 16:04.56 | [TK]D-Fender | [GuS]: ....o.....k.... |
| 16:05.24 | *** join/#asterisk Y0da^ (n=Deb@70.159.118.70) |
| 16:08.10 | [GuS] | i will have to read more... |
| 16:12.04 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 16:12.46 | Uatec_ | hey, i'm trying to setup distinctive ring styles on my sip phone, i'm using: exten => s,n,SIPAddHeader(Alert-Info: something) |
| 16:13.31 | Uatec_ | that is all very well, but my hardware provider doesn't appear to give any hint as to how to use distinctive ringing with it, what kind of alert info am i supposed to send (is it standard? is there a common theme to the info sent to most phones?) |
| 16:13.36 | ManxPower | Uatec_: What you add is TOTALLY dependent on the phone |
| 16:13.47 | Uatec_ | damn |
| 16:13.50 | Uatec_ | i was fearful of that |
| 16:13.56 | *** join/#asterisk Bladerunner05 (n=feelme@81-174-56-54.f5.ngi.it) |
| 16:14.04 | Zeeek | don't many phones allow you to add the ring in the directory? |
| 16:14.21 | *** join/#asterisk Cyon (n=cyon@216.179.31.170) |
| 16:14.27 | Uatec_ | in the in phone directory? |
| 16:14.28 | Uatec_ | possibly |
| 16:14.29 | *** join/#asterisk iwannadoit (i=iwannado@196.211.34.2) |
| 16:14.33 | Zeeek | contact dir |
| 16:14.34 | Uatec_ | but i need asterisk to tell it what ring to use |
| 16:14.36 | Uatec_ | some how |
| 16:14.39 | ManxPower | Uatec_: Are you using any brand phone we might have heard of it. |
| 16:14.44 | ManxPower | Uatec_: NO! |
| 16:14.55 | iwannadoit | I need a programer to import to south africa |
| 16:15.12 | ManxPower | Uatec_: Asterisk simply adds a header to the SIP message, it is up to the phone to do something with it. |
| 16:15.12 | sevard | iwannadoit: why? |
| 16:15.16 | De_Mon | exten => _00!,n,Set(member=${CUT(members,\|,${COUNT})}) |
| 16:15.20 | iwannadoit | wil pay plane ticket and accommodation and arange for traveling. |
| 16:15.22 | De_Mon | I get an error on this line |
| 16:15.27 | ManxPower | iwannadoit: you'll need more derails that that if you want someone to respond |
| 16:15.29 | iwannadoit | msg me to negosiate salary |
| 16:15.45 | Zeeek | and more derails mean higher insurance costs |
| 16:15.45 | De_Mon | iwannadoit what kind of programmer? |
| 16:16.00 | ManxPower | De_Mon: and that error message is...? |
| 16:16.32 | De_Mon | nothing useful: Jun 5 12:14:45 ERROR[17068]: app_cut.c:397 acf_cut_exec: Usage: Splits a variable's contents using the specified delimiter |
| 16:16.40 | ManxPower | Depending on the project, the scope, the contract length, and the design specs, I might be interested. |
| 16:16.48 | De_Mon | it doesnt like the | delimiter |
| 16:17.05 | ManxPower | De_Mon: Don't use ${} around the variable name |
| 16:17.43 | De_Mon | set(member=cut(members,\|,${count})})})})}) |
| 16:17.51 | De_Mon | ? |
| 16:18.08 | ManxPower | How about exten => _00!,n,Set(member=${CUT(members,\|,COUNT)}) |
| 16:18.08 | De_Mon | the variable name (members) isn't using {}'s |
| 16:19.00 | ManxPower | Sorry, COUNT would be a number. |
| 16:19.06 | ManxPower | De_Mon: What is the CLI output of that line. |
| 16:19.07 | De_Mon | umm, |
| 16:19.07 | *** join/#asterisk ectospasm (i=Spasm@nat/digium/x-bdbe432d0c269f26) |
| 16:19.11 | ManxPower | you'll see what it evaluates as. |
| 16:19.12 | Qwell[] | ectospasm: ? |
| 16:19.20 | Qwell[] | oh |
| 16:19.21 | De_Mon | ${COUNT} is 1 |
| 16:19.35 | Qwell[] | ectospasm: More support people should come here... |
| 16:19.38 | ManxPower | have you tried using a 1 instead. |
| 16:19.59 | ManxPower | It is possible you cannot use | as a delimiter. |
| 16:20.01 | De_Mon | no, I changed the delimiter to ; and it work sifne though |
| 16:20.13 | ManxPower | Ah. There is your solution |
| 16:20.18 | Qwell[] | I think you have to doubly escape |, or something |
| 16:20.21 | Uatec_ | ManxPower, i'm using a snom 190 |
| 16:20.26 | De_Mon | 12:16PM <De_Mon> it doesnt like the | delimiter |
| 16:20.27 | Qwell[] | like \\| |
| 16:20.28 | ManxPower | Remember , are converted to | during evaluation |
| 16:20.39 | Qwell[] | I forget exactly |
| 16:20.40 | De_Mon | yeah, I tried \\| too with the same error |
| 16:20.46 | Qwell[] | \\\| ? :p |
| 16:20.50 | ManxPower | Uatec_: Are you too lazy to check the wiki? |
| 16:20.52 | Qwell[] | escape the escape too, heh |
| 16:20.54 | ManxPower | \\\\\\\\\\\\\\\\\\\| |
| 16:21.04 | Qwell[] | There was actually one case where I had to use \\\ |
| 16:21.08 | Uatec_ | yes, i am ManxPower |
| 16:21.09 | Qwell[] | I forget why |
| 16:21.11 | De_Mon | i'm cutting a database key/value pair that is using | as delimiter |
| 16:21.12 | Uatec_ | cos i'm a cunt like you |
| 16:21.15 | Strom_M | "make" for asterisk 1.4 on ubuntu is crapping out with the error "autoconf: no input file" |
| 16:21.21 | Uatec_ | of course i'm looking the shitting wiki |
| 16:21.26 | Uatec_ | and reading the doucmentation |
| 16:21.26 | Strom_M | and google provides no answers, of course |
| 16:21.46 | Qwell[] | Strom_C: try a make distclean |
| 16:21.46 | De_Mon | same error with \\\| |
| 16:21.48 | Uatec_ | but i haven't found nything yet, which is why i asked |
| 16:21.50 | Strom_M | am I overlooking something basic? |
| 16:21.51 | Strom_M | ok |
| 16:22.15 | ManxPower | Results 1 - 10 of about 184 English pages for snom alert-info AND asshole that can't use google. (0.44 seconds) |
| 16:22.23 | tzanger | hahahahahahahha |
| 16:22.23 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 16:22.25 | ManxPower | the first result looks promising |
| 16:22.38 | Zeeek | ok |
| 16:23.18 | Strom_M | Qwell: that worked |
| 16:23.52 | De_Mon | maybe I can replace | with something else and cut on that |
| 16:25.41 | Qwell[] | Strom_C: did you doubt me? ;) |
| 16:25.41 | De_Mon | this is 1.2.14 .. waiting for pastebin |
| 16:25.41 | De_Mon | http://pastebin.ca/541408 |
| 16:25.41 | Strom_M | no, but I wanted to at least give closure to my problem :) |
| 16:25.42 | *** join/#asterisk seanwg (n=sean@216.249.37.157.ppp.northrock.bm) |
| 16:25.49 | seanwg | is there a quick way to see active sip registrations? |
| 16:25.49 | De_Mon | show registrar |
| 16:25.49 | De_Mon | show <tab> one of those |
| 16:25.54 | ManxPower | seanwg: asterisk registrations of devices TO asterisk or registrations of devices asterisk is registered to |
| 16:25.56 | seanwg | mythtv*CLI> show registar |
| 16:25.57 | seanwg | No such command 'show registar' (type 'help' for help) |
| 16:25.57 | seanwg | mythtv*CLI> sip show registry |
| 16:25.57 | seanwg | Host Username Refresh State Reg.Time |
| 16:25.57 | seanwg | mythtv*CLI> |
| 16:26.05 | seanwg | i just want to see what sip devices are registered |
| 16:26.13 | ManxPower | sip show peers |
| 16:26.39 | seanwg | great that work |
| 16:26.40 | seanwg | s |
| 16:26.46 | De_Mon | is registry devices is registered TO? |
| 16:26.52 | De_Mon | * is |
| 16:27.41 | ManxPower | sip show registry shows the status (more or less) of register => lines in sip.conf |
| 16:29.39 | *** join/#asterisk xuser (i=1000@unaffiliated/xuser) |
| 16:30.45 | *** join/#asterisk mrdigital (n=mrdigita@207-172-228-21.c3-0.tlg-ubr2.atw-tlg.pa.cable.rcn.com) |
| 16:31.38 | xuser | Hi, is there any cdr billing software for linux? |
| 16:32.29 | iwannadoit | yes quemetrics works wonders xuser |
| 16:33.16 | iwannadoit | xuser hard and soft designs have a working one where you can set billing for coutry local ect |
| 16:34.25 | xuser | iwannadoit: thanks, gonna check it. |
| 16:35.08 | De_Mon | eeeh I dont see func_cut in funcs |
| 16:36.09 | Uatec_ | ahah, excellent, i have distinctive ringing working |
| 16:36.17 | Uatec_ | now all i need is some decent ring tones |
| 16:37.14 | *** join/#asterisk CVirus (n=GoD@196.205.192.129) |
| 16:38.09 | *** join/#asterisk marcan (i=1337@14.Red-88-27-160.staticIP.rima-tde.net) |
| 16:38.58 | *** join/#asterisk DrukenHME (n=jdumais@CPE000e08cb2a29-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 16:40.50 | *** join/#asterisk seele_ (n=seele@dns.datawareltda.com) |
| 16:40.51 | *** join/#asterisk putnopvut (n=putnopvu@user-24-214-124-177.knology.net) |
| 16:40.54 | iwannadoit | Uatec web.ukonline.co.uk/freshwater/hic120.htm this guy has all legasy pbx listed and in his link pages have some good links to phone tones and ringtones |
| 16:41.01 | seele_ | I have IVR configured, when I calls to the IVR and dial a extension the IVR redirect me to the first option, the extensions has 4 digits and begins with 1 ... the number of my first option in the IVR |
| 16:41.08 | seele_ | how can I make a direct extension dial with IVR options??? |
| 16:41.56 | iwannadoit | Goto(IVR,s,1) seele where IVR is your ivr context |
| 16:42.14 | Uatec_ | WTF? |
| 16:42.28 | Uatec_ | i mean cheers iwannadoit |
| 16:42.40 | Uatec_ | WTF?, when i dial out now, i just get a ring tone |
| 16:42.40 | Uatec_ | *sigh* |
| 16:42.41 | iwannadoit | nice site hey |
| 16:42.51 | seele_ | iwannadoit, I don't know |
| 16:43.07 | iwannadoit | what do you want seele? |
| 16:43.30 | iwannadoit | dcc me you extension.conf seele |
| 16:43.55 | iwannadoit | are you using asterisk source or trixbox seele? |
| 16:44.17 | iwannadoit | whats up uatec? |
| 16:44.24 | ManxPower | seele_: are the extensions in the same context as the IVR, or is the context the extensions in include =>'d in the context of the IVR |
| 16:44.50 | Uatec_ | it's ok |
| 16:44.52 | Uatec_ | i was being stupid |
| 16:45.01 | Uatec_ | now i have another problem |
| 16:45.06 | seele_ | ManxPower, yes the extensions are in the same context |
| 16:45.13 | iwannadoit | happens to all of us ! |
| 16:45.17 | *** join/#asterisk greendisease (n=jack@fedora/greendisease) |
| 16:45.45 | seele_ | ManxPower, some times works ... and some time doesn't |
| 16:45.51 | CVirus | What do you guys think of the Grandstream GXW4004 ? |
| 16:46.10 | Qwell[] | ~phones |
| 16:46.20 | jbot | somebody said phones was http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. |
| 16:46.27 | ManxPower | seele_: do you have any relaxdtmf options set? You should not set that optuon |
| 16:46.32 | *** join/#asterisk steliosk (n=Stelios@62.169.217.209) |
| 16:46.35 | Qwell[] | CVirus: I think that about sums it up |
| 16:46.50 | [TK]D-Fender | ~gs |
| 16:46.52 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 16:46.54 | CVirus | Qwell[]: is it crappy to that extend ? |
| 16:46.55 | [TK]D-Fender | ^^^^^^^^^^^^ |
| 16:47.05 | ManxPower | GS phones don't even make good doorstops or bookends |
| 16:47.08 | ManxPower | they are too light |
| 16:47.10 | seele_ | ManxPower, nop |
| 16:47.13 | Qwell[] | too light |
| 16:47.14 | Qwell[] | heh |
| 16:47.17 | CVirus | LOL |
| 16:47.23 | greendisease | yo Qwell, what up |
| 16:47.25 | Strom_M | React to GS how you'd react to the Killer Rabit: RUN AWAY!!!!!!!!!!!!!!!!!!!!!!!! |
| 16:47.28 | seele_ | ManxPower, where can I enable this option |
| 16:47.29 | Qwell[] | greendisease: hey |
| 16:47.41 | Qwell[] | greendisease: any bugs in FC7 we need to be warned about? ;) |
| 16:47.51 | greendisease | haha, many, can you msg me for a second? |
| 16:47.57 | Qwell[] | uh oh |
| 16:47.57 | ManxPower | seele_: you do NOT wait to enable that option. |
| 16:48.23 | iwannadoit | does anyone have experience with patton bri gateways? |
| 16:48.31 | ManxPower | seele_: do other calls, like into voicemail, work correctly? |
| 16:48.47 | seele_ | ManxPower, yes |
| 16:48.57 | ManxPower | seele_: I don't have any more ideas |
| 16:49.15 | CVirus | What is recommended as a 4 port FXS ? |
| 16:49.20 | *** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-118-241.ph.ph.cox.net) |
| 16:49.31 | ManxPower | CVirus: Adtran channel bank |
| 16:49.52 | CVirus | what else ? |
| 16:50.52 | ManxPower | I think you'll find that most people recommend a channel bank of some form |
| 16:51.15 | CVirus | ManxPower: why not the TDM400P ? |
| 16:51.43 | ManxPower | CVirus: Would you like the honest response or the polite, politically correct answer? |
| 16:52.00 | [TK]D-Fender | CVirus: MediaTrix or AudioCodes gateway. Or 2 ATA's |
| 16:52.03 | Uatec_ | how about an honest polite correct response? |
| 16:52.18 | ManxPower | Uatec_: I don't have one when it comes to the TDM400P. |
| 16:52.21 | CVirus | ManxPower: the honest |
| 16:52.31 | *** join/#asterisk MrTelephone (n=MrTeleph@h697179-171.picriverisp.net) |
| 16:52.41 | MrTelephone | hey |
| 16:52.48 | ManxPower | Aha!. "We had about 8 TDM400Ps scattered across 6 offices. Every single one has been replaced by a T-1 card and a channel bank." |
| 16:53.18 | MrTelephone | what are the tdm cards missing ? |
| 16:53.28 | MrTelephone | why do they work so poorly :( |
| 16:53.44 | iwannadoit | echo cancelation |
| 16:54.12 | [TK]D-Fender | TDM = PCI issues, higher cost,have to wire the phone DIRECT to the server, etc |
| 16:54.15 | [TK]D-Fender | lose/lose |
| 16:54.20 | MrTelephone | Im getting dropped calls with the error- packet retrans exceeded, hanging up call.. Thats due to poor network conditions I'm assuming? |
| 16:54.24 | [TK]D-Fender | ATA/gateways are cheaper and better |
| 16:54.36 | ManxPower | MrTelephone: It can be. |
| 16:54.43 | [TK]D-Fender | MrTelephone: Or due to packets not having a way back to respond. |
| 16:54.44 | MrTelephone | can't we make a card that does more processing onboard? |
| 16:55.03 | ManxPower | MrTelephone: Yes. They are all $500/port |
| 16:55.09 | [TK]D-Fender | MrTelephone: Why bother? Buy the proper hardware for this sort of thing that already exists and is CHEAPER |
| 16:55.16 | ManxPower | Dialogic would be an example of a board with built in DSP |
| 16:55.48 | ManxPower | [TK]D-Fender: I think it is more the TDM400P driver than anything else. |
| 16:56.04 | ManxPower | MANY MANY people have been happy with the TDM400P had not had problems. I am not one of those people. |
| 16:57.04 | ManxPower | MrTelephone: that message basically says there is a problem with network communications. Doesn't give a cause, but network issues would be a good bet. |
| 16:57.06 | iwannadoit | ManxPower tried onle to link to lecasy pbx only got calls on one channel |
| 16:57.35 | ManxPower | iwannadoit: that is a config problem |
| 16:58.50 | iwannadoit | nope |
| 16:59.15 | *** join/#asterisk [hC] (n=hardcore@190.10.12.97) |
| 16:59.15 | *** join/#asterisk ectospasm (i=Spasm@nat/digium/x-5de6187d08fc125c) |
| 16:59.19 | *** join/#asterisk nettie (n=nettie@ns.coolgadgets.it) |
| 16:59.27 | iwannadoit | qsig on my pbx out of dat have to pay $9200 to get new lics from Philips South Africa |
| 16:59.28 | ManxPower | iwannadoit: I have never never heard of that problem and the it not being a config issue |
| 17:00.00 | justdave | If I have both a T1 card and a TDM card, is it possible to direct-bridge a channel on the PRI to an FXS port on the TDM card? |
| 17:00.16 | ManxPower | iwannadoit: we went with a T-1 card for our Nortel off of eBay. PRI support requires a license, but T-1 E&M Wink does not for the Nortel |
| 17:00.24 | justdave | via the zap driver config.. or do I need to do funky routing in asterisk? |
| 17:00.26 | ManxPower | justdave: yes. |
| 17:00.43 | ManxPower | justdave: that depends. do you need asterisk to have ANY access to those channels? |
| 17:00.46 | CVirus | thank you guys |
| 17:01.10 | justdave | yeah, asterisk would get 22 of the PRI channels, the FXS port would get 1 of them |
| 17:01.36 | ManxPower | justdave: no. I meant do you want Asterisk to have access to the channel that is bridged to the FXS? |
| 17:01.49 | *** join/#asterisk ecoleman (n=eric@24.75.47.98) |
| 17:01.50 | justdave | ManxPower: no, it wouldn't need it |
| 17:01.58 | *** join/#asterisk sudhir492 (n=sudhir@c-71-62-102-201.hsd1.va.comcast.net) |
| 17:02.05 | ManxPower | You can use DACS to cross connect any channel to any channel using Zap only, no asterisk. |
| 17:02.13 | justdave | although if I have to use asterisk to do the bridging, I have no objection |
| 17:02.13 | ecoleman | is it possible to play some sound while an agi script is executing and then have it stop right away? |
| 17:02.20 | ManxPower | channels that are not DACS'd act as normal |
| 17:02.22 | *** join/#asterisk nettie (n=nettie@ns.coolgadgets.it) |
| 17:02.24 | justdave | ok |
| 17:02.37 | Zeeek | ManxPower what distro do you use? |
| 17:02.43 | ManxPower | justdave: see the sample config file in the zap source |
| 17:02.48 | ManxPower | Zeeek: Mandriva |
| 17:02.53 | Zeeek | k |
| 17:03.21 | justdave | right now we have a T1 with 5 channels of voice traffic and the remaining channels used for data. We have data from another source now (dedicated fiber :) ) and looking to get the T1 converted to a full PRI |
| 17:03.44 | justdave | 1 of those voice channels is a fax line, the other 4 are a rollover sequence on the main phone number. |
| 17:04.07 | justdave | so the 4 on the main phone number would be the main DID going into the PRI, and the fax we still want to split out if we can pull it off. :) |
| 17:04.17 | ManxPower | justdave: We used to have mixed voice and data T-1s. We would take the data channels and use DACS RBS to cross connect the data channels to channels on another card in the server, then plugged our router into that port. |
| 17:04.43 | justdave | since everything I read says fax is still flakey in asterisk :) |
| 17:06.58 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-82-81-91-140.red.bezeqint.net) |
| 17:07.59 | ManxPower | justdave: It is nice to see someone that does not ignore all the advice about asterisk and fax. |
| 17:08.05 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-82-81-91-140.red.bezeqint.net) |
| 17:08.06 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 17:08.46 | *** join/#asterisk Daejeo1 (n=chatzill@124.62.150.49) |
| 17:09.04 | nettie | Hi guys, I'm having some issues with chan_sip. We've zap (ISDN) and sip trunks, when the DSL link is down chan_sip tries to register again the various carriers but when it does it internal sip clients (polycom phones) cant call eachother. sip show peers shows that the phones are correctly registered. Anyone know if there's a configuration value to tell chan_sip to continue registering without taking offline the phones? Thanx. |
| 17:10.47 | sudhir492 | I have an asterisk box, which talks to the gateway (another asterisk box) through iax. However, when I press digits, it is not passed to the gateway. Any suggestions? |
| 17:10.49 | Daejeo1 | is it possible to use GPS system with Mobile? |
| 17:11.23 | Daejeo1 | is it possible to use GPS system with astersik server |
| 17:12.27 | Daejeo1 | asterisk* |
| 17:12.28 | *** join/#asterisk diclophis-work (n=jbardin@65.203.37.58) |
| 17:13.23 | diclophis-work | hello all |
| 17:13.25 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 17:13.38 | diclophis-work | I am wondering what options there are for load balancing incoming SIP requests? |
| 17:13.46 | diclophis-work | between two asterisk machines |
| 17:15.53 | sulex | hello, in a Background() command I can state the full path to the sound I have to play omitting the file extension. Can you tell me if the same works for Record(). For example: Record(/var/lib/asterisk/sounds/foo/bar/foobar:gsm)? |
| 17:17.25 | diclophis-work | sulex: I do that through AGI all the time |
| 17:17.33 | diclophis-work | RECORD FILE /wang/chung WAV |
| 17:18.05 | ecoleman | is it possible to play some sound while an agi script is executing and then have it stop right away? |
| 17:18.27 | sulex | diclophis-work: it does, cool. thanks |
| 17:18.36 | sulex | diclophis-work: I mean it does it directly in the dp |
| 17:19.09 | diclophis-work | ecoleman: try the background command, it might return control to your agi script |
| 17:19.19 | diclophis-work | or perhaps exec(background) |
| 17:19.39 | diclophis-work | ecoleman: though you might end up having to work some multiprocess or thread magic |
| 17:21.30 | ecoleman | diclophis-work: thanks |
| 17:21.56 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 17:24.51 | Hmmhesays | I dislike it when people don't pay me |
| 17:25.28 | ecoleman | Hmmhesays: don't we all? |
| 17:26.16 | nettie | anyone have issues with internal sip phones when asterisk cant register a trunk please? |
| 17:26.53 | sevard | Hmmhesays: story of my life. |
| 17:27.01 | Hmmhesays | hey stranger |
| 17:27.05 | sevard | hey budddddddy. |
| 17:27.12 | Hmmhesays | ok pauly |
| 17:27.20 | sevard | how you been, mandinga |
| 17:27.31 | Hmmhesays | not bad |
| 17:27.42 | Hmmhesays | playing guitar, wakeboarding, trying to build this postgres module for openser |
| 17:27.51 | sevard | (decided to carry the shore) |
| 17:28.18 | sevard | postgres ftw, but why are you playing with openser? I think that's #asterisk's bible's satan |
| 17:28.57 | Hmmhesays | openser can do things asterisk can't |
| 17:29.44 | sevard | like? (never played with it) |
| 17:29.55 | Hmmhesays | make my faxing work |
| 17:29.58 | Hmmhesays | lol |
| 17:30.28 | sevard | heh |
| 17:30.41 | Hmmhesays | more calls per second on similar hardware than asterisk |
| 17:30.47 | Trevor_b | hmm, thats funny i have spandsp+asterisk running a 24 line fax system on a TDM2400 |
| 17:31.02 | Qwell[] | SER doesn't do "calls" though... |
| 17:31.13 | sevard | annnnnnnnd here it comes |
| 17:31.14 | Qwell[] | it doesn't really have to do a whole heck of a lot with the channel ;) |
| 17:31.40 | Hmmhesays | Qwell[]: yeah, what I meant was more it can process more incoming invites per second |
| 17:31.52 | Qwell[] | sure, it'd be silly if it couldn't |
| 17:32.06 | Hmmhesays | what it is not though... is pbx software, which is why I use both |
| 17:32.39 | sevard | oh jesus, why can't somebody just shoot john cusack |
| 17:32.39 | Hmmhesays | I use asterisk for my pbx needs and pass it off to openSER when the call is a fax call |
| 17:32.47 | Daejeo1 | is it possible to use GPS system with asterisk server? |
| 17:33.00 | sevard | Daejeo1: with OSS anything is possible! |
| 17:33.06 | sevard | ;) |
| 17:33.15 | Qwell[] | It's just software |
| 17:33.26 | Hmmhesays | Hello Katty |
| 17:33.29 | Daejeo1 | sevard: any documentation? |
| 17:33.36 | sevard | Daejeo1: got notepad? |
| 17:33.52 | Daejeo1 | sevard: yes |
| 17:33.58 | sevard | Excellent. type some up |
| 17:34.10 | Daejeo1 | type what? |
| 17:34.24 | sevard | ;) |
| 17:34.26 | sevard | http://asteriskvoip.blogspot.com/2005/12/project-implementing-bluetooth.html |
| 17:35.12 | Daejeo1 | :) |
| 17:35.24 | iwannadoit | can anyone help me with the best softphone? |
| 17:35.45 | sevard | iwannadoit: the _best_ softphone? |
| 17:35.48 | Daejeo1 | :( |
| 17:36.01 | Sweeper | iwannadoit: I'd suggest trying eyebeam |
| 17:36.06 | iwannadoit | which one is it? |
| 17:36.17 | Sweeper | solid piece of software, customizable, remote provisioning capable |
| 17:36.21 | Daejeo1 | eyebeam is not free |
| 17:36.26 | Katty | anthm: up! |
| 17:36.28 | Sweeper | it is not! |
| 17:36.33 | iwannadoit | but is it good? |
| 17:36.35 | Sweeper | neither are polycoms |
| 17:36.38 | Sweeper | iwannadoit: it is |
| 17:36.40 | Daejeo1 | it is good |
| 17:36.44 | Daejeo1 | I have one |
| 17:36.46 | iwannadoit | thx sweeper |
| 17:36.48 | Daejeo1 | do you want to try? |
| 17:36.52 | anthm | ? |
| 17:37.10 | Sweeper | you can write their sales dept and get a 30-timebomb for demo |
| 17:37.16 | sevard | Sweeper: yeah, xlite and eyebeam are great if you like using 80% of your 2.5 ghz cpu for 1 call. |
| 17:37.35 | Sweeper | sevard: well, I don't run it on a 486, so that doesn't happen :o |
| 17:37.42 | sevard | apparently "lite" means feature light while still being a resource hog |
| 17:37.47 | Sweeper | and we use g.729 anyways |
| 17:37.53 | Sweeper | which is even heavier |
| 17:38.05 | Katty | anthm: can you have one of your people call me for billing? |
| 17:38.10 | *** join/#asterisk bcnl (n=mike@S010600131078957c.vc.shawcable.net) |
| 17:38.27 | Sweeper | s/anyways// |
| 17:38.58 | De_Mon | hmm, it looks like DB(/Queue/PersistentMembers/queueName) is where dynamic members are stored |
| 17:39.07 | anthm | mmkay |
| 17:39.12 | Katty | anthm: dankou. |
| 17:39.15 | De_Mon | if I wanted to remove all queue members, can i just set that key to "" to remove them all? |
| 17:39.24 | sevard | Sweeper: or simply s/s$//g |
| 17:39.58 | anthm | if you keep using it enough you will eventually remove them all when it crashes. |
| 17:40.31 | De_Mon | I sense that your not a fan of asterisk queues |
| 17:40.39 | De_Mon | you're |
| 17:40.40 | Daejeo1 | :(( |
| 17:41.19 | anthm | how'd you guess? |
| 17:41.42 | Qwell[] | queues are mostly fine - it's agents that suck |
| 17:41.46 | De_Mon | you're laying it on a little thick |
| 17:42.38 | Daejeo1 | ;) |
| 17:42.51 | anthm | queues are fine if you don't expect any of the "strategies" to actually mean anything and if you *never* reload it. |
| 17:43.14 | De_Mon | so, removing all members from a queue by setting the db key to "" a good solution? My other choice is to get the queue members and remove one and then loop thru again |
| 17:43.22 | De_Mon | anthm queue members are persistant |
| 17:43.22 | Daejeo1 | :'( |
| 17:43.50 | coppice | queues are fine if you are British. They are an innermost part of the national character |
| 17:43.56 | Qwell[] | anthm: are you done trolling? If you are aware of a specific bug, you know where to report it... |
| 17:44.12 | sevard | a trolling op, that's great. |
| 17:44.28 | De_Mon | sounds like hes still complaining about bugs have already been fixed |
| 17:44.57 | anthm | If I were trolling everyone would be in tears. |
| 17:45.10 | sevard | the great and powerful |
| 17:45.12 | De_Mon | s/be in tears/have me on ignore/ |
| 17:45.51 | Hmmhesays | feel the love |
| 17:46.03 | sevard | it's like a warm trashcan |
| 17:46.12 | bcnl | does anyone know if there is a version of Answer Confirmation for SIP/IAX channels that works like the dial feater 'c' for ZAP channels? |
| 17:47.11 | Daejeo1 | (@) |
| 17:48.00 | Daejeo1 | (@)(&) |
| 17:48.04 | Daejeo1 | (&) |
| 17:49.58 | Sweeper | man, I wish the alternative softpbxs weren't all "we hate asterisk, here's our fairly buggy, mostly featureless pbx lo lo lo" |
| 17:50.16 | Hmmhesays | callweaver is ok |
| 17:50.18 | Daejeo1 | :I |
| 17:50.21 | Daejeo1 | :| |
| 17:50.27 | Sweeper | well, callweaver is a fork :P |
| 17:50.35 | Hmmhesays | so? |
| 17:50.42 | Sweeper | in that case it's "we hate digium!" |
| 17:50.52 | Daejeo1 | :p |
| 17:51.15 | tzafrir_laptop | Sweeper, not just |
| 17:51.15 | Sweeper | so I'd like a completely different pbx that was actually good |
| 17:51.19 | De_Mon | I saw a pbxnsip booth at techEd, it looked pretty full featured |
| 17:51.20 | iwannadoit | i take my hat off to all opensource projects i wich i had the brains to write my own! |
| 17:51.25 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 17:52.03 | Sweeper | iwannadoit: your bash login that changes the prompt to [wanker]# doesn't count as an oss project :D |
| 17:52.39 | Qwell[] | it does if it's configurable or random |
| 17:52.40 | Hmmhesays | finally |
| 17:53.24 | iwannadoit | i know |
| 17:53.37 | iwannadoit | but i did read the manual |
| 17:53.51 | iwannadoit | open source manual that is |
| 17:55.25 | iwannadoit | if mark spencer is looking for a free holiday to south africa maybe a free hunting trip or just a sit at the seaside in capetown i will flip the bill because he made me alot of money wit asterisk |
| 17:55.49 | Qwell[] | he's a busy man - I'll go in his place |
| 17:56.12 | file | Qwell[]: pfft |
| 17:56.19 | iwannadoit | the community all helped, but i can not affort to give all you people a holida |
| 17:56.23 | iwannadoit | lol qwell |
| 17:56.24 | Qwell[] | and file too |
| 18:01.06 | mog | Qwell, what about me???? |
| 18:01.30 | mog | fine, ill just go to north africa |
| 18:01.31 | Qwell[] | mog: You'll have to convince iwannadoit :p |
| 18:01.44 | mog | i think he wants to do it Qwell |
| 18:01.45 | Qwell[] | you can totally stow away, and we'll give you a parachute |
| 18:01.51 | file | mog: you can come to Canada |
| 18:01.55 | file | I'll give you a place to stay! |
| 18:02.08 | file | you just have to try not to go crazy from broadway singing and dancing |
| 18:02.21 | mog | oh really |
| 18:02.26 | iwannadoit | where you guys from? |
| 18:02.32 | mog | how far away are you from mcgill? |
| 18:02.41 | file | awhile. |
| 18:03.02 | file | or an hour and a half by plane |
| 18:03.16 | mog | oh well |
| 18:03.26 | mog | want to see em graduate next year |
| 18:04.06 | `Sean | what does cisco recooment people use for voip solloutions |
| 18:04.10 | `Sean | i ment with there ip phones |
| 18:04.17 | `Sean | i assume they must also sell a PBX or something? |
| 18:07.09 | Hmmhesays | call manager |
| 18:11.50 | Bladerunner05 | with tdm400 and asterisk 1.4.4 I have to configure zaptel.conf ? |
| 18:12.26 | *** join/#asterisk andyd (n=andyd@213-228-240-161.dsl.prodigynet.co.uk) |
| 18:13.21 | irule | yes, and sapata.conf too |
| 18:13.28 | iwannadoit | what do you need to know bladerunner05? |
| 18:13.28 | irule | zapata.conf |
| 18:14.55 | Bladerunner05 | <iwannadoit>: what I have to write into zaptel.conf? |
| 18:15.30 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:16.21 | *** part/#asterisk diclophis-work (n=jbardin@65.203.37.58) |
| 18:19.51 | Bladerunner05 | I have tdm400 with 4 fxo, I configure zapata.conf with standard settings more what you see in http://www.pastebin.ca/542061 that was generated by genzaptel command, but when I receive a call * don't answer and no message on cli> |
| 18:20.19 | Bladerunner05 | I use very verbose mode |
| 18:20.35 | Bladerunner05 | I configure also extension.conf e sip.conf in the same context used in zapata.conf |
| 18:21.30 | *** part/#asterisk [TK]D-Fender (n=joe_blow@modemcable089.225-70-69.mc.videotron.ca) |
| 18:22.41 | *** join/#asterisk karlhaines (n=karl@unaffiliated/karlhaines) |
| 18:25.03 | karlhaines | anyone used these motorolla modems that work as fxo cards? |
| 18:25.14 | karlhaines | or are there any other modems capable of this? |
| 18:26.36 | Bladerunner05 | how I have to setup zaptel.conf with 4 fx0 modules ? |
| 18:26.45 | Daejeo1 | :| |
| 18:28.54 | karlhaines | Bladerunner05, trixbox? |
| 18:29.18 | Bladerunner05 | asterisk 1.4.4 |
| 18:29.31 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 18:32.03 | *** join/#asterisk osiris (n=osiris@c-71-205-27-131.hsd1.mi.comcast.net) |
| 18:32.48 | *** join/#asterisk mik3 (n=43b8ee33@alcor.lunarpages.com) |
| 18:33.47 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 18:39.33 | Hmmhesays | i need a telnet client that will let me grep the output |
| 18:39.33 | Nugget | telnet is eeeeeeevil! |
| 18:39.46 | Hmmhesays | does one of those exist? |
| 18:41.01 | SwK | Hmmhesays, ngrep :P |
| 18:41.02 | *** join/#asterisk _VoiceMeUp_COM (n=_VoiceMe@modemcable159.131-56-74.mc.videotron.ca) |
| 18:41.56 | Hmmhesays | not exactly what what I'm looking for |
| 18:42.00 | tzanger | I was going ot say ngrep |
| 18:42.03 | tzanger | nc | grep will do it too |
| 18:42.35 | Hmmhesays | well the problem is I'm logged into the console of a cisco like platform, dumping data in realtime, no grep in that particular terminal |
| 18:42.59 | Hmmhesays | I can log to a text file and grep that |
| 18:43.08 | Hmmhesays | but, what a pain in the @$$ |
| 18:43.29 | Hmmhesays | you smell what I'm stepping in? |
| 18:44.08 | tzanger | ha |
| 18:44.12 | tzanger | smells like poo |
| 18:44.21 | Hmmhesays | yeah pretty much, but you get what I'm saying? |
| 18:48.46 | Hmmhesays | that would be pretty awesome if something like that existed |
| 18:49.12 | tzanger | kind of like nc but where the input from nc can go through grep |
| 18:49.23 | irule | onyone know the default username and password for the latest elastix? |
| 18:49.35 | tzanger | nc over.there 23 | grep foo |
| 18:49.37 | tzanger | should do it |
| 18:49.40 | tzanger | you'll be typing blind though |
| 18:49.51 | _VoiceMeUp_COM | hmmmsays what you trying to parse ? |
| 18:49.54 | Hmmhesays | tzanger: I don't get how that would work |
| 18:50.03 | Bladerunner05 | what can I do the dirver for tdm400 works fine but asterisk don't answer a call !!!!!!!! |
| 18:50.05 | Hmmhesays | _VoiceMeUp_COM: output from a remote terminal |
| 18:50.06 | tzanger | pipe is one way you're not redirecting stdin |
| 18:50.27 | _VoiceMeUp_COM | cisco ? |
| 18:50.32 | Hmmhesays | similar |
| 18:50.51 | _VoiceMeUp_COM | to a securecrt...hmmm |
| 18:50.56 | _VoiceMeUp_COM | ssh /telnet ? |
| 18:51.01 | Hmmhesays | telnet |
| 18:51.08 | *** join/#asterisk AndrewGearhart (n=chatzill@h1.39.213.151.ip.alltel.net) |
| 18:51.14 | tzanger | Hmmhesays: tested it |
| 18:51.16 | tzanger | Hmmhesays: it works fine |
| 18:51.17 | _VoiceMeUp_COM | and you need to parse or just dump ? |
| 18:51.26 | _VoiceMeUp_COM | and waht need to be parse |
| 18:51.27 | _VoiceMeUp_COM | ;) |
| 18:51.30 | Hmmhesays | tzanger: what does over.there represent, i'm not very familiar with netcat |
| 18:51.37 | _VoiceMeUp_COM | ip |
| 18:51.45 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 18:51.48 | tzanger | nc this.is.the.remote.host this.is.the.port.you.want.probably.23 |
| 18:51.55 | _VoiceMeUp_COM | nc -L hostname:port -p port [options] |
| 18:52.01 | tzanger | _VoicePulse: no |
| 18:52.01 | _VoiceMeUp_COM | -L is a tunnel |
| 18:52.03 | _VoiceMeUp_COM | that could help |
| 18:52.03 | tzanger | he's not listening |
| 18:52.09 | tzanger | -L is not tunnel, it's listen |
| 18:52.17 | _VoiceMeUp_COM | listen for inbound: nc -l -p port [options] [hostname] [port] ... |
| 18:52.17 | _VoiceMeUp_COM | tunnel to somewhere: nc -L hostname:port -p port [options] |
| 18:52.21 | *** join/#asterisk catpants (n=catling@12-214-191-244.client.mchsi.com) |
| 18:52.22 | _VoiceMeUp_COM | i guess it dpeends on version mate |
| 18:52.25 | tzanger | _VoicePulse: ahh |
| 18:52.33 | _VoiceMeUp_COM | <PROTECTED> |
| 18:52.33 | _VoiceMeUp_COM | GNU netcat 0.7.1, a rewrite of the famous networking tool. |
| 18:52.46 | _VoiceMeUp_COM | _VoiceMeUp_COM |
| 18:52.47 | _VoiceMeUp_COM | hehe |
| 18:52.50 | _VoiceMeUp_COM | double tabb it |
| 18:53.04 | irule | onyone know the default username and password for the latest trixbox? |
| 18:53.25 | tzanger | yeah I don't have tunneling in my version (1.10) |
| 18:53.40 | _VoiceMeUp_COM | maint password ? |
| 18:53.45 | _VoiceMeUp_COM | and /j #trixcrap |
| 18:53.50 | _VoiceMeUp_COM | cough trixbox i mean |
| 18:54.16 | _VoiceMeUp_COM | well still idnt answer waht do you need to aprse |
| 18:54.17 | _VoiceMeUp_COM | parse |
| 18:54.41 | *** join/#asterisk alrs (n=lars@72.54.121.98) |
| 18:57.25 | *** join/#asterisk joetester (n=joeteste@216.191.34.13) |
| 18:59.45 | Hmmhesays | hmm nc does not seem to be working |
| 19:03.47 | *** join/#asterisk mik3 (n=43b8ee33@alcor.lunarpages.com) |
| 19:03.50 | *** join/#asterisk yardB (n=oats@c-68-39-136-61.hsd1.nj.comcast.net) |
| 19:04.05 | mik3 | what operating system does asterisk work better on typically? |
| 19:04.13 | Katty | Hmmhesays: come help me move |
| 19:04.19 | Trevor_b | I use Linux |
| 19:04.24 | Katty | Hmmhesays: i have a hard time moving my bed and pa gear. |
| 19:04.36 | yardB | i ind linux to be good |
| 19:05.12 | Hmmhesays | nc where are you moving? |
| 19:06.46 | *** join/#asterisk yannj_fr (n=yannj@vpn.intelunix.fr) |
| 19:06.50 | yardB | Help from anyone:!! I have two servers communicating via IAX ould it be better to use a TI (more exensive) or my cable company broadband service |
| 19:07.28 | *** part/#asterisk Cresl1n (i=matt@nat/digium/x-c66e22bde76f711d) |
| 19:07.42 | Hmmhesays | oops |
| 19:07.46 | yannj_fr | depend of the number of simultaneous communication |
| 19:07.49 | Hmmhesays | Katty: where are you moving? |
| 19:08.01 | Katty | Hmmhesays: oh...about 5 miles from my current place. |
| 19:08.14 | yardB | yannj_fr : let say 23 simultaneous calls? |
| 19:09.30 | *** join/#asterisk jebba (n=jebba@220-179-89-200.fibertel.com.ar) |
| 19:10.38 | Mercestes | yardB, T1 would be *better* of course... |
| 19:11.07 | Mercestes | yardB: now whether it is cost effective depends on the communication involved. Is your communication worth $800 a month? |
| 19:11.08 | yannj_fr | which codec do you use? |
| 19:11.33 | mik3 | T1's aren't $800/mo anymore. |
| 19:11.54 | mik3 | well not here at least |
| 19:12.23 | mik3 | anyone running asterisk on a debian box? |
| 19:12.23 | mik3 | or have expereince in doing so |
| 19:12.48 | Mercestes | mik3: He would need 2 T1's. |
| 19:12.48 | Mercestes | mik3: Unless there is some magical T1 to IAX2 gateway device |
| 19:12.50 | mik3 | ah, my apologies |
| 19:12.53 | yardB | Mercestes: This is what i need to understand why T1 is better. The BW is 1.5Mbs while my cable company say they provide me with 6MBp |
| 19:12.54 | Mercestes | :) |
| 19:12.59 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 19:13.25 | Mercestes | yardB, because it is not in yoru cable companies best interest for yoru VoIP to work |
| 19:13.26 | yannj_fr | I use a MPLS link at 2mb for about 30 coms between my 2 sites |
| 19:13.34 | J4k3 | yardB: cable can't TX and RX at the same time, at least no consumer-grade modem I've seen can do that |
| 19:14.48 | J4k3 | yardB: cable also, at best, in my experience, comes with a 24 hour SLA |
| 19:14.52 | yardB | ah, J4K3 ..no duplex communication? |
| 19:14.54 | J4k3 | T1 comes with a 2/4 plan around here. |
| 19:15.04 | J4k3 | yardB: correct, its just fast enough that consumers don't care. |
| 19:15.27 | J4k3 | yardB: but it doesn't scale without a 'conversation merging' technology like iax2 trunking |
| 19:15.34 | J4k3 | where 1 packet may carry 10 calls. |
| 19:15.41 | Trevor_b | your 6Mbps is a connection with hops along the way, and it is only to their core uplink, if you cant PUSH 6Mbps as well as pull it then your connection has a much smaller limit for voice communications. T1 would be 1.5Mbps up and down, and if you had it point to point instead of internet access, you have no extra hops to take, and routes to slow you down on the internet (where your cable would probably have 7-12 hops between your sites unless you had t |
| 19:15.45 | J4k3 | instead of say, 500 pps SIP |
| 19:16.13 | yardB | the difference in price is quite a bit |
| 19:16.28 | J4k3 | a T1 PTP circuit (not frame, not atm) is also gauranteed end to end. You WILL get 1536Kbit to the serial interface of the far end router, no matter what |
| 19:16.31 | *** join/#asterisk tsurko (n=tsurko@77.70.24.142) |
| 19:16.34 | J4k3 | the link either does 1536k, or its dead. |
| 19:16.58 | yardB | So, it seems if I want to provide good service, then bite th ebullet and use T1? ;) |
| 19:17.08 | Trevor_b | THere are other ways |
| 19:17.18 | Trevor_b | if you have low call counts |
| 19:17.21 | *** join/#asterisk tsurko (n=tsurko@77.70.24.142) |
| 19:17.25 | Trevor_b | how many calls between sites you run? |
| 19:17.39 | yardB | max 23 ..PRI |
| 19:17.46 | yardB | number of calls |
| 19:17.49 | *** join/#asterisk Gregabyte (i=wintermu@nat/digium/x-9ace61ce5bfd3f41) |
| 19:18.14 | *** join/#asterisk tsurko (n=tsurko@77.70.24.142) |
| 19:18.21 | Trevor_b | not max, how many do you NORMALLY plan on running, and MAX? |
| 19:18.25 | J4k3 | maybe after today our voip will work better... AT&T and/or our local ILEC had our circuit screwed up for a few months |
| 19:18.58 | yannj_fr | hum yes, 23... I was thinking about here where T1 = 30, 2048MBPS |
| 19:19.01 | Trevor_b | if you normaly have 3 calls going, and it goes up to 23 sometimes, thats alot different then running 23 all day and maybe going down to 20 |
| 19:19.04 | yardB | J4K3 i was thinking of using AT&T ;)) |
| 19:19.33 | AndrewGearhart | howdy folks... Asterisk... how tough is it to setup a "click-to-call" type of arrangement? |
| 19:19.38 | yardB | Trevor_b i expect approx 20 average |
| 19:19.50 | yannj_fr | agi? |
| 19:19.52 | Juggie | andrew, quite easy. |
| 19:20.14 | AndrewGearhart | Juggie: I like your talk! ;-) |
| 19:20.15 | Trevor_b | yeah, then PTP T1 or more. Depends on the location your in, other services are available that could give you much more throughput, and faster internet, but depends on the state your in. |
| 19:20.40 | AndrewGearhart | Juggie: how is it handled if all the folks are busy? |
| 19:20.53 | yannj_fr | andrew : I did it with a php script that create a .call file and mv it in /var/spool/asterisk |
| 19:21.08 | Juggie | yannj_fr, why not use php to connect to the manager interface. |
| 19:21.12 | Juggie | dont bother with .call files. |
| 19:21.30 | J4k3 | yardB: I'm happy with my AT&T service. Its been better than any other 'national' provider I've dealt with (SAVVIS, SprintLink, UUnet [in recent years, uunet USED to rock]) |
| 19:21.45 | Juggie | AndrewGearhart, all i do is a php script using php sockets to connect to the manager interface, and issue an action: originate. |
| 19:21.45 | yannj_fr | you can too, for sure!! but when i did it, I didnt know how to use agi! |
| 19:21.54 | Juggie | then i have dialplan that handles that. |
| 19:21.55 | J4k3 | SAVVIS is the worst ever. the network performs well but my personal experience the reliability SUCKED. |
| 19:22.00 | Juggie | yannj_fr, the manager interface isnt AGI. |
| 19:22.06 | Juggie | its the manager interface, different things. |
| 19:22.08 | J4k3 | we managed to shut down the circuit based on SLA violations. and savvis's sla is pretty crappy. |
| 19:22.37 | AndrewGearhart | Juggie: I've never used php sockets... is it difficult to learn? |
| 19:22.54 | yannj_fr | Juggie : so I still dont know how to use it ;-) |
| 19:22.55 | Juggie | no |
| 19:23.06 | Juggie | let me see if i can find a example i have somewhere, hold on. |
| 19:23.09 | mik3 | is there any programming involved in setting up asterisk? |
| 19:23.18 | J4k3 | they had a defective 'customer feeding' router in houston. Random router failures at 5PM on friday... that got old, quick. |
| 19:24.08 | AndrewGearhart | Juggie: thanks |
| 19:26.02 | yardB | J4K3 so you are using AT&T for for PSTN and also IAX communication between servers 2 separate T1 |
| 19:26.26 | *** join/#asterisk WB0TRA_work (n=WB0TRA_w@64.62.46.146) |
| 19:26.55 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 19:28.05 | Juggie | AndrewGearhart, here is an old piece of code i just dug up, its just a proof of concept i did one time way way ago, but it should help you see how it works. |
| 19:28.26 | Juggie | i wont give you the context for the asterisk dialplan, i'll leave that as an exercise for you, obviously you need to create a [clicktotalk] context. |
| 19:28.33 | Juggie | as per the context defined in the call setup. |
| 19:28.42 | Juggie | you can figure it out from there. |
| 19:28.44 | Juggie | http://www.pastebin.ca/542216 |
| 19:28.51 | *** join/#asterisk karlhaines (n=karl@unaffiliated/karlhaines) |
| 19:29.03 | *** part/#asterisk WB0TRA_work (n=WB0TRA_w@64.62.46.146) |
| 19:29.19 | karlhaines | i found an Ambient MD3200 modem on the shelf and got it to work as an FXO card ;) |
| 19:29.27 | *** join/#asterisk flewid (n=flewid@mail.flewid.ca) |
| 19:29.28 | karlhaines | it actually works pretty well so far! |
| 19:29.31 | flewid | sup |
| 19:29.59 | AndrewGearhart | thanks Juggie. :) |
| 19:30.32 | Juggie | AndrewGearhart, as you can see by this line 'fputs($socket, "Channel: Local/$num1@internals/n\r\n");' i push all my calls through a context to decide if the call is local, long distance, etc. |
| 19:30.56 | Juggie | you can change that to Channel: Zap/g1/$num1 if you like. |
| 19:31.16 | Juggie | but i would recomend using a context w/ local channels to push all your calls through |
| 19:31.20 | Juggie | gives you more flexibility. |
| 19:31.39 | *** join/#asterisk Mad|Cow (n=madcow@74.92.109.205) |
| 19:31.40 | Juggie | that way if a user wants to transfer to a local or longdistance number, you decide how to dial it in asterisk (where it should be) not in your script. |
| 19:32.51 | AndrewGearhart | cool |
| 19:33.09 | *** join/#asterisk galeras (n=root@200.31.204.42) |
| 19:33.50 | AndrewGearhart | Juggie: I can't wait for the boss man to spend the money on the equipment. |
| 19:34.03 | Juggie | so, in that case Local/$num1@internals |
| 19:34.11 | Juggie | you would need an internals context |
| 19:34.17 | Juggie | like [internals] |
| 19:34.20 | Juggie | with say |
| 19:34.28 | *** join/#asterisk [Mr_X] (n=mrx@88.118.57.195) |
| 19:34.42 | Juggie | exten=> _613NXXXXXX,1,Dial(Zap/g1/${EXTEN}) |
| 19:34.51 | Juggie | exten=> _NXXNXXXXXX,1,Dial(Zap/g1/1${EXTEN}) |
| 19:35.04 | Juggie | just as a rough example, presuming all of 613 was local (which it wont be of course) |
| 19:35.29 | Juggie | then your local channel calls this context, parses your dialed number, and connects your call. much better to do it this way. |
| 19:35.33 | Juggie | gives * the control it should have. |
| 19:35.59 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 19:36.42 | *** join/#asterisk bapril (n=bapril@pool-70-109-158-237.cncdnh.east.verizon.net) |
| 19:37.08 | AndrewGearhart | Juggie: the thinking on this... is that we have TONS of out of state customers... and I've found an analog CLEC that can do unlimited for a flat fee... but it only counts on our regular lines... we would still get charged for our inbound 800#s... |
| 19:37.42 | karlhaines | AndrewGearhart: thats almost always the case |
| 19:37.43 | karlhaines | sucks |
| 19:37.49 | Juggie | ya, so with a context, and pushing all your calls through a local channel through that context |
| 19:38.02 | Juggie | you can put all your decision making in one place re, user dials this number, go over this trunk, etc. |
| 19:38.10 | *** join/#asterisk _omer (n=_omer@DSL-202-59-92-141.nexlinx.net.pk) |
| 19:38.55 | *** join/#asterisk jeffik (n=Valued@206-248-152-65.dsl.teksavvy.com) |
| 19:39.49 | iwannadoit | im azpped out |
| 19:39.56 | iwannadoit | im zapped out |
| 19:40.25 | iwannadoit | chhers poeple i am going to leave while I can still type |
| 19:40.44 | *** join/#asterisk grantm (n=grantm@kolob.wingateservices.com) |
| 19:40.55 | *** join/#asterisk Stephnie (i=Stephnie@u15157627.onlinehome-server.com) |
| 19:41.32 | AndrewGearhart | brb... ph |
| 19:43.57 | *** join/#asterisk mvanbaak (n=mafkees@vanbaak.xs4all.nl) |
| 19:46.34 | AndrewGearhart | Juggie: that would be really nice... |
| 19:47.00 | AndrewGearhart | karlhaines: yeah. I know that after much research. ;-) |
| 19:47.09 | AndrewGearhart | karlhaines: so the goal was to reduce the number of incoming calls. |
| 19:47.10 | *** join/#asterisk tsurko (n=tsurko@77.70.24.142) |
| 19:47.30 | AndrewGearhart | karlhaines: if we could get people to use our software that they have installed to originate a call... so much the better! ;-) |
| 19:47.42 | Juggie | AndrewGearhart, its not hard at all, and i do accept contract work :) |
| 19:48.01 | Juggie | you dont need software, just pump your calls through *. |
| 19:48.26 | AndrewGearhart | Juggie: well... we're a software company. We market two particular pieces of software. |
| 19:48.54 | AndrewGearhart | Juggie: my point was that we could give them a "click to receive a call" from within the software. |
| 19:49.08 | Juggie | and you want this for your company? or you want to support it within your software. |
| 19:49.42 | AndrewGearhart | well... the software is Home Health Care agency software... when folks run into problems... they call us for support. |
| 19:50.02 | AndrewGearhart | We want to originate as many minutes as we can though... to reduce our outrageous phone bills |
| 19:50.21 | Juggie | ah, ok i understand, so you want a button in your software (or on your website) |
| 19:50.31 | AndrewGearhart | Juggie: exactly |
| 19:50.33 | AndrewGearhart | :) |
| 19:50.36 | Juggie | so the client would fill in their phone number, extension, press submit. |
| 19:50.46 | Juggie | it calls you, then one of your agents picks up, and it connects to the client. |
| 19:50.50 | AndrewGearhart | right... and in the software... their information is already there. |
| 19:51.13 | yardB | Mercestes can I talk to you off line? |
| 19:51.23 | AndrewGearhart | so... it sends the magic information to * ... connects the call to the agent... and makes the call to the people. |
| 19:51.32 | ManxPower | Mercestes: charge him! charge him! |
| 19:51.34 | Juggie | well, you would be best off just doing it over HTTP... so your software does a HTTP request to your remote server, your remote server handles the request... and places the originate. |
| 19:52.20 | iwannadoit | php |
| 19:52.22 | Juggie | the originate calls your agent, and when they pick it plays a message connecting to client (with perhaps a client id as well) |
| 19:52.28 | Juggie | and then calls the clients number. |
| 19:52.43 | AndrewGearhart | Juggie: right. gotcha. |
| 19:52.56 | Juggie | AndrewGearhart, thats really no trouble at all. |
| 19:53.05 | Juggie | using very basic funtionality. |
| 19:53.07 | AndrewGearhart | Juggie: how much more effort to make it run into a queue? |
| 19:53.50 | Juggie | not much, in that case you would call the user first and place them into a queue. |
| 19:55.19 | Juggie | AndrewGearhart, do you already have a queuing system? or did you want to use * for that? |
| 19:55.54 | Mercestes | ManxPower: Bwahahaa |
| 19:58.36 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 19:59.29 | AndrewGearhart | Juggie: honestly... without having the hardware necessary... I haven't even begun to delve into * except to ask questions... :) |
| 19:59.43 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 20:00.44 | *** join/#asterisk flewid (n=flewid@mail.flewid.ca) |
| 20:02.05 | flewid | hey, am i crazy for trying to get this to work --- asterisk -- pfsense -- cable modem -- internet -- cable modem -- router -- polycom phone |
| 20:02.06 | flewid | ? |
| 20:02.06 | Juggie | AndrewGearhart, what you are asking would be super simple to do, you need the hook in your software, the script on the http server. and the context within the asterisk dialplan. |
| 20:02.06 | flewid | the phone registers, I see it, I can call it, it rings, but user can't pick up, nor can he check voicemail or dial out |
| 20:02.06 | Hmmhesays | mozart px time out |
| 20:02.14 | Hmmhesays | anyone know what the hell that means in regards to payphones |
| 20:03.48 | AndrewGearhart | Juggie: that's what I like to hear. ;-) |
| 20:08.53 | *** join/#asterisk sasch (n=sasch@host102-30-static.107-82-b.business.telecomitalia.it) |
| 20:09.10 | *** join/#asterisk mitcheloc (n=mitchelo@titaniumsoft.net) |
| 20:09.42 | *** join/#asterisk casimir (n=casimir@rrcs-71-43-154-55.se.biz.rr.com) |
| 20:16.58 | *** join/#asterisk CVirus (n=GoD@196.219.179.87) |
| 20:16.58 | *** join/#asterisk tsurko (n=tsurko@77.70.24.142) |
| 20:18.05 | *** join/#asterisk RobH (n=RobH@rrcs-24-73-86-239.se.biz.rr.com) |
| 20:18.50 | RobH | Does anyone have a resource to point me at for having the display on a linksys 941/942 display the parked call line when transferring (a blind transfer, instead of hearing it say where it was transferred to)? |
| 20:21.33 | *** join/#asterisk mindCrime_ (n=chatzill@66.83.208.219.nw.nuvox.net) |
| 20:22.48 | *** join/#asterisk galeras (n=root@200.31.204.42) |
| 20:23.36 | galeras | is here the voip heaven? |
| 20:23.36 | *** join/#asterisk zeeesh (n=aadilism@202.125.143.70) |
| 20:23.36 | zeeesh | hi |
| 20:24.08 | iwannadoit | you just arived galeras |
| 20:24.32 | galeras | i'm lost |
| 20:24.49 | iwannadoit | what do you need? galeras |
| 20:30.54 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 20:31.18 | iwannadoit | hi fkuq |
| 20:32.19 | *** join/#asterisk mirco (n=mirco@88.128.39.99) |
| 20:32.29 | iwannadoit | hi mirco |
| 20:33.33 | mirco | HI |
| 20:33.45 | iwannadoit | whats up |
| 20:34.19 | *** join/#asterisk kombi (n=kombi@213.160.14.18) |
| 20:34.22 | iwannadoit | hi kombi |
| 20:34.27 | kombi | hi iwannadoit! |
| 20:34.27 | iwannadoit | whats up? |
| 20:34.35 | kombi | monging on a capi related issue.. |
| 20:34.43 | *** part/#asterisk RobH (n=RobH@rrcs-24-73-86-239.se.biz.rr.com) |
| 20:34.51 | iwannadoit | try me! |
| 20:35.24 | kombi | the param in Dial() is the channel I understand, would that be what capi show channels returns? |
| 20:35.38 | kombi | the first parameter I mean.. |
| 20:36.13 | iwannadoit | are you using hfg od misdn.conf? |
| 20:36.21 | iwannadoit | sorry or |
| 20:36.34 | kombi | using capi.conf |
| 20:36.43 | iwannadoit | what card? |
| 20:36.48 | iwannadoit | not a diguim? |
| 20:36.53 | iwannadoit | digium? |
| 20:36.55 | kombi | b1 from avm actually |
| 20:37.14 | kombi | got a digium but still packaged..;) |
| 20:37.46 | iwannadoit | you should have multipull ports on channel are you using as te or nt? |
| 20:37.58 | kombi | I got as far as the card receiving calls fine, but no outbound calls yet |
| 20:38.44 | kombi | in capi conf it is configured to non nt, allthough it is connected to pstn. Tried changing that but to no avail.. |
| 20:39.06 | iwannadoit | dial(CAPI/1:1/${EXTEN}) |
| 20:39.12 | *** join/#asterisk tdonahue-laptop (n=tdonahue@vonmail.vonworldwide.com) |
| 20:39.20 | iwannadoit | this will apply to channel1 port 1 |
| 20:39.27 | tdonahue-laptop | hi all |
| 20:39.36 | iwannadoit | hi |
| 20:39.38 | kombi | thanks iwannadoit! I have to wait until my girlfriend hangs up until I can try next.. |
| 20:39.47 | kombi | ;) |
| 20:39.54 | iwannadoit | k lol i hope the call is free |
| 20:40.02 | kombi | almost..;) |
| 20:40.03 | tdonahue-laptop | my asterisk 1.4 box just crashed and the only thing I can find in the logs is "localhost kernel: asterisk[3571] general protection rip:2b3768c7d5b0 rsp:400b2678 error:0" |
| 20:40.56 | tdonahue-laptop | does any additional information get generated when a message like that occurs? |
| 20:41.01 | iwannadoit | go to youasterisk source and do a make clean,make ,makeinstall ansterisk |
| 20:41.13 | iwannadoit | do not do a make config |
| 20:41.52 | iwannadoit | this should restore your server i had this a while back reloading asterisk help me |
| 20:42.27 | kombi | iwannadoit: would the dial-out rule have to be in a certain context? (i.e. the same as for inbound calls?) |
| 20:43.14 | iwannadoit | no you can spit the context |
| 20:43.14 | kombi | kewl! |
| 20:43.36 | iwannadoit | got it working kombi? |
| 20:44.20 | kombi | girlfriend is still on the line, talking to her best friend, that can take SOME time.. |
| 20:44.33 | iwannadoit | where you from kombi? |
| 20:44.37 | kombi | germany |
| 20:44.55 | kombi | how about you? |
| 20:44.58 | iwannadoit | we have a make of car here in south africa called a volkwagen kombi |
| 20:45.01 | *** join/#asterisk seanwg (n=sean@216.249.37.157.ppp.northrock.bm) |
| 20:45.07 | iwannadoit | made in germany |
| 20:45.10 | kombi | my favorite! |
| 20:45.17 | iwannadoit | k love bug |
| 20:45.21 | kombi | like a recent one? |
| 20:45.34 | iwannadoit | nice |
| 20:45.44 | kombi | I believe it is a van type model, right? |
| 20:45.49 | tdonahue-laptop | anyone have a clue where i can get information about what caused my asterisk system to crash? |
| 20:45.54 | iwannadoit | how is your ford knowledge? |
| 20:46.03 | iwannadoit | yip thats right |
| 20:46.16 | kombi | had one once, 17m |
| 20:46.30 | kombi | tdonahue-laptop: /var/log/asterisk |
| 20:46.35 | iwannadoit | just got a ford st 2.5 turbo charged muscle cal |
| 20:46.39 | iwannadoit | i love it |
| 20:46.43 | kombi | nice one! |
| 20:47.02 | iwannadoit | payeda fortune to get stuck in trafic every morning |
| 20:47.04 | tdonahue-laptop | kombi, just stops logging... nothing indicating a problem before the crash happens |
| 20:47.36 | J4k3 | I enjoy passing noisy rice-cars while hauling 1500 lbs of bagged concrete mix in the back. |
| 20:47.36 | kombi | iwannadoit: lol, at least you're stuck in style.. |
| 20:47.57 | iwannadoit | good one |
| 20:48.15 | kombi | tdonahue-laptop: the essence of troubleshooting: reproduce, isolate, reproduce, isolate.. |
| 20:48.25 | iwannadoit | 12liter per 100km is not styling |
| 20:48.31 | FuriousGeorge | great news, i caught asterisk deadlocking and now i have a back trace to file a bug report with |
| 20:48.32 | FuriousGeorge | http://pastebin.ca/542420 |
| 20:49.01 | FuriousGeorge | i have to admit it makes no sense to me, can anyone tell me if my bt and bt full is gonna be usefull |
| 20:49.05 | kombi | J4k3: you're on a concrete van? how can you sit by a computer? |
| 20:49.20 | iwannadoit | laptop! |
| 20:49.33 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 20:49.41 | J4k3 | kombi: no, I've been making stepping stones for my house... |
| 20:49.43 | kombi | ok...lol |
| 20:50.00 | iwannadoit | kombi do you speek german? |
| 20:50.07 | kombi | yip! |
| 20:50.20 | iwannadoit | ever played commandos? |
| 20:50.24 | iwannadoit | the game? |
| 20:50.26 | J4k3 | kombi: but when I'm out-and-about, I have Verizon 1xRTT/EVDO CDMA service and a bluetooth capable handset. |
| 20:50.39 | kombi | iwannadoit: actually don't know it.. |
| 20:50.40 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.163.72) |
| 20:50.43 | J4k3 | worst case scenario I get 14.4k, most of the time about 140kbit, and if I'm near a city, 400kbit. |
| 20:50.57 | kombi | J4k3: sounds good.. |
| 20:51.14 | iwannadoit | is daar iemand ombie |
| 20:51.43 | iwannadoit | is someone there? |
| 20:51.45 | kombi | girls are the first and foremost users of telephony, do you guys realize we're doing all this mainly for the girls of this world? |
| 20:51.55 | iwannadoit | lol |
| 20:52.11 | iwannadoit | voip atleast you dont have to pay for it |
| 20:52.13 | ecoleman | you just ruined my motive for working :'( |
| 20:52.44 | kombi | lol.. girls aren't the worst things also.. |
| 20:53.00 | ecoleman | girl + phone = almost the worst thing |
| 20:53.05 | Hmmhesays | ~seen coppice |
| 20:53.45 | jbot | coppice <n=chatzill@10.198.17.210.dyn.pacific.net.hk> was last seen on IRC in channel #asterisk, 3h 9m 55s ago, saying: 'queues are fine if you are British. They are an innermost part of the national character'. |
| 20:53.47 | kombi | heck, they will speak until dawn.. can you try out outbound connections without your bri card plugged in somehow? |
| 20:53.47 | J4k3 | kombi: I pbx for the nookie |
| 20:53.47 | kombi | ecoleman: point taken.. |
| 20:53.52 | iwannadoit | nice to have around(breeding machine_i have wonderfull 2,5year old boy)the 2 is the love of my life |
| 20:54.06 | tdonahue-laptop | kombi, sure, let me tell our client that we need to reproduce the problem to isolate the cause even though i have no reason to suspect that the logs will capture something useful the next time it happens |
| 20:54.22 | tdonahue-laptop | because they don't seem to have caught anything this time.... |
| 20:54.32 | iwannadoit | set verbose to 30 then check |
| 20:54.41 | iwannadoit | set verbose to 30 then check tdonahue |
| 20:55.23 | kombi | tdonahue-laptop: hmm, you might want to set up a feasable testing environment, it is really hard to debug otherwise |
| 20:55.25 | *** join/#asterisk emiquelito (n=evandro@mx.telium.net.br) |
| 20:55.42 | *** part/#asterisk emiquelito (n=evandro@mx.telium.net.br) |
| 20:55.53 | kombi | tdonahue-laptop: what exactly makes the daemon crash? |
| 20:56.25 | tdonahue-laptop | iwannadoit, verbose and debug were both set to 100, but it seems like it was there one second and gone the next |
| 20:56.30 | *** part/#asterisk galeras (n=root@200.31.204.42) |
| 20:56.47 | tdonahue-laptop | the last debug message was " devicestate.c: Notification of state change to be queued on device/channel SIP/PCS4510-b08a50f0" |
| 20:56.57 | kombi | tdonahue-laptop: have you looked into /var/log/syslog? |
| 20:57.33 | tdonahue-laptop | "localhost kernel: asterisk[3571] general protection rip:2b3768c7d5b0 rsp:400b2678 error:0" was the only message in syslog |
| 20:57.48 | iwannadoit | do you use static addresse on your sip client? |
| 20:58.00 | Katty | i know i don't. |
| 20:58.12 | Katty | i'd be skeered. |
| 20:58.17 | *** join/#asterisk dracosilv (n=draco@CPE-65-29-47-173.wi.res.rr.com) |
| 20:58.19 | iwannadoit | do you use static addresse on your sip client?--donahue-laptop |
| 20:58.32 | tdonahue-laptop | iwannadoit, no |
| 20:58.57 | kombi | what's the PCS4510-b00.. bit? |
| 20:59.01 | Katty | anthm: make bkw answer his phone :< |
| 20:59.21 | iwannadoit | what is your reregister time? donahue-laptop |
| 20:59.28 | kombi | Katty: make my girlfriend hang up the phone.. |
| 20:59.36 | iwannadoit | finaly |
| 20:59.43 | Qwell[] | kombi: stop now |
| 20:59.49 | Qwell[] | That'll hang her up ;) |
| 21:00.00 | iwannadoit | lol Qwell |
| 21:00.03 | iwannadoit | nice one |
| 21:00.16 | iwannadoit | not stop when convenient |
| 21:00.27 | *** join/#asterisk darkskiez (n=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 21:00.32 | FuriousGeorge | http://pastebin.ca/542420 |
| 21:00.33 | FuriousGeorge | can anyone tell me if my bt and bt full is gonna be usefull |
| 21:00.39 | FuriousGeorge | should i bother filing a bug report |
| 21:01.01 | tdonahue-laptop | iwannadoit, whatever default time is, i think 3600 seconds |
| 21:01.18 | iwannadoit | set to 300 |
| 21:01.28 | iwannadoit | on client |
| 21:01.41 | iwannadoit | better to use static |
| 21:02.34 | tdonahue-laptop | iwannadoit, what does that have to do with the fact that asterisk crashed? |
| 21:03.10 | tdonahue-laptop | and static addressing is not possible due to the fact that 50% of their staff are road warriors |
| 21:03.49 | iwannadoit | you sip is trying to reregister therefore error rsp:400b2678 error:0 |
| 21:04.16 | karlhaines | anyone used X100P clone cards? |
| 21:05.13 | *** join/#asterisk br4k3r (n=rod@bas9-ottawa23-1128750720.dsl.bell.ca) |
| 21:05.50 | iwannadoit | what do you need to know karlhaines |
| 21:05.55 | J4k3 | karlhaines: they work. |
| 21:06.01 | J4k3 | sometimes, and not well, but they work |
| 21:06.55 | iwannadoit | this is a 1port card yes? |
| 21:07.01 | iwannadoit | fxo? |
| 21:07.30 | karlhaines | J4k3: yeah, i got one to work today (Ambient MD3200), but when i shutdown and the zap fxo module unloads i get a kernel panic |
| 21:07.36 | karlhaines | J4k3: is this normal? |
| 21:07.58 | karlhaines | J4k3: the REAL X100P cards are reliable, though, right? |
| 21:08.23 | tdonahue-laptop | iwannadoit, there were no messages about any of the clients trying to register at the time that asterisk died |
| 21:08.26 | *** join/#asterisk ManxPower (n=manxpowe@102.sub-75-201-209.myvzw.com) |
| 21:08.33 | iwannadoit | they work and are cheap |
| 21:08.46 | *** join/#asterisk jebba (n=jebba@220-179-89-200.fibertel.com.ar) |
| 21:08.57 | karlhaines | iwannadoit: which? clones or real x100p's? |
| 21:09.20 | iwannadoit | which one do you have? |
| 21:09.36 | karlhaines | ambient md3200 |
| 21:09.50 | tdonahue-laptop | iwannadoit, how did you match the error " rsp:400b2678 error:0" to the fact that there was a registration going on? |
| 21:10.36 | iwannadoit | are you using asterisk@home? |
| 21:10.50 | karlhaines | iwannadoit: yeah, trixbox |
| 21:11.48 | iwannadoit | do a lsmod from shell and past what you see in a dcc chat to me |
| 21:12.13 | karlhaines | iwannadoit: the system is shutdown atm, lol |
| 21:12.25 | karlhaines | iwannadoit: are you using a similar card? |
| 21:12.30 | *** join/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
| 21:13.10 | iwannadoit | i had it as a error on my asterisk 1.2 server with iax connection to multiple asterisk iax connections |
| 21:13.29 | iwannadoit | no i love digium |
| 21:13.43 | karlhaines | i only have 1 sip connection atm |
| 21:14.02 | karlhaines | iwannadoit: yeah, me too! in all my voip systems around town i use a TDM400 card |
| 21:14.09 | karlhaines | iwannadoit: it works wonderfully |
| 21:14.18 | iwannadoit | make sure your chipset on card is the Intel 531 <karlhaines> |
| 21:14.51 | iwannadoit | why atm? |
| 21:15.03 | FuriousGeorge | i caught a deadlock and would like to file a bug report. can anyone tell me if my bt and bt full are going to be usefull |
| 21:15.17 | FuriousGeorge | http://pastebin.ca/542420 |
| 21:15.38 | Qwell[] | FuriousGeorge: no |
| 21:16.08 | FuriousGeorge | Qwell[]: why am i not surprised. can you tell me why the heck not. i followed backtrace.txt |
| 21:16.19 | Qwell[] | don't know - it just isn't... |
| 21:16.52 | FuriousGeorge | any tips for how to make it usefull |
| 21:17.11 | iwannadoit | sorry no |
| 21:17.17 | FuriousGeorge | im in a position now where asterisk deadlocks daily and afaik, there is now way to debug it. im out of components to swap |
| 21:18.14 | kombi | they finally hung up.. |
| 21:18.42 | ManxPower | FuriousGeorge: then switch back to 1.2.15 |
| 21:18.57 | kombi | iwannadoit: just tried with your line but no luck, how do I best debug? |
| 21:19.55 | FuriousGeorge | ManxPower: that one deadlocks bi-daily. any reason to suspect my backtraces are gonna be less useless there |
| 21:20.24 | kombi | got exten => _0.,1,DIAL(CAPI/1:1/${EXTEN}) but the bri card won't call outbound.. |
| 21:21.39 | iwannadoit | no do a _X.,1,Dial(CAPI/1:1/${EXTEN}) where EXTEN is the number to dial |
| 21:21.48 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 21:22.40 | iwannadoit | is this in your default context? |
| 21:22.53 | kombi | not yet.. |
| 21:22.57 | ManxPower | FuriousGeorge: deadlocks are software bugs |
| 21:23.02 | J4k3 | fucking pathetic vitelity... All their houston DIDs died. |
| 21:23.05 | ManxPower | backtrackes can help if it generates a crash |
| 21:23.10 | J4k3 | for at least 5 minutes that I know of. |
| 21:24.01 | J4k3 | PRIs are fairly worthless, or have outrageous costs, unless you need at least 10 concurrent calls average. |
| 21:24.28 | J4k3 | what I need is a sip provider that doesn't make goat testicles a part of their breakfast. |
| 21:24.36 | ManxPower | I guess it depends on if you want to trust your business to various ISPs the packets go thru |
| 21:24.57 | kombi | iwannadoit: maybe I didn't quite get what goes into $EXTEN.., you said the number to dial? |
| 21:24.59 | FuriousGeorge | ManxPower: im a little confused by what your telling me because everything ive read on the matter says first a deadlock or a crash happens then you backtrace |
| 21:25.07 | ManxPower | J4k3: most of the ITSPs use Level3 |
| 21:25.08 | FuriousGeorge | then you file a bugreport |
| 21:25.24 | Mercestes | and level 3 kinda blows a little |
| 21:25.28 | ManxPower | FuriousGeorge: is it crashing |
| 21:25.53 | J4k3 | well, when I originally chose vitelity I was straight AT&T to dallas, which was a whole 2 hops |
| 21:25.58 | J4k3 | now they're in denver, and blow goats. |
| 21:26.26 | J4k3 | they use L3 because thats who they colo with, and L3 charges a METRIC SHITLOAD to haul in cable to the colo from a real provider. |
| 21:26.34 | J4k3 | or well, across the colo. |
| 21:27.14 | ManxPower | You pay one way or another. I just prefer to pay for a PRI |
| 21:27.54 | iwannadoit | _X. meens the number you press(meening anything 0-9 until you stop pressing the digits) |
| 21:27.54 | ManxPower | FuriousGeorge: bad memory or memory corruption could cause a deadlock, I guess. |
| 21:27.54 | J4k3 | yeah well... with PRIs you get to buy from a CLEC (jokers) or ILEC (refuse to DID you outside your physical local calling scope) |
| 21:28.00 | J4k3 | I live in a LATA thats about 180 miles wide and 225 miles tall, and the average local calling scope is about 40x40 miles. |
| 21:28.06 | iwannadoit | [isdn-out] |
| 21:28.07 | iwannadoit | exten => _1NXXNXXXXXX,1,Set(CALLFILENAME=/var/spool/asterisk/monitor/outgoing/8452781980/Out-${TIMESTAMP}_${CALLERIDNUM}) |
| 21:28.07 | iwannadoit | exten => _1NXXNXXXXXX,2,Monitor(wav,${CALLFILENAME},) |
| 21:28.07 | iwannadoit | exten => _1NXXNXXXXXX,3,Dial(CAPI/g1/${EXTEN}/B,,) |
| 21:28.10 | ManxPower | Actually, _X. means "Match any digit and match at least 1 more of any character, digits, numbers, symbols, etc" |
| 21:28.29 | ManxPower | at == and |
| 21:28.34 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 21:28.36 | J4k3 | oddly, my toll free DID never, ever, ever dies. |
| 21:28.42 | J4k3 | nor does it sound like shit. |
| 21:28.55 | n0n4m3 | umm.. in case i have a client using iax defined as a friend in iax.conf, let's say [1234] ... and have the "exten => 1234,1,Dial(IAX/1234)" in extensions.conf that should work, right? |
| 21:29.05 | kombi | iwannadoit: I tried those as well.., isn't there somewhere to get an error message from? |
| 21:29.08 | J4k3 | I'm half tempted to contact customers and start shutting down local DIDs. |
| 21:29.13 | ManxPower | J4k3: I'm a fan of REGIONAL CLECs. They are big enough to be decent, small enough not to act like an ILEC |
| 21:29.40 | iwannadoit | what doe it say in youcli console? |
| 21:29.41 | J4k3 | ManxPower: SBC*ahem*AT&T*ahem*broadwing*ahem*level3 ended up eating every single one of them up here. |
| 21:29.50 | ManxPower | We used to be the 2nd largest customer of a regional CLEC. That was nice. |
| 21:30.10 | J4k3 | well, theres cary fitch's personal nightmare... I've purchased wholesale dialup on that CLEC before, it was worthless... I'm not a big fan of Mr. Fitch. |
| 21:30.16 | J4k3 | and no other regionals left in houston afaik |
| 21:30.23 | ManxPower | that sucks. |
| 21:30.50 | ManxPower | A lot of CLECs closed up shop in New Orleans after Katrina. |
| 21:31.04 | ManxPower | They told their customers "We are closing, you have 30 days to find a new provider. Thank you for being a customer." |
| 21:31.21 | kombi | iwannadoit: that's the thing, nothing at all! set verbosity to 99.. |
| 21:32.19 | ManxPower | I use Deltacom as the CLEC at the campground, but only because Deltacom was the ONLY CLEC in my CO |
| 21:32.30 | ManxPower | And they are strictly resale. |
| 21:32.44 | *** join/#asterisk flewid (n=flewid@mail.flewid.ca) |
| 21:32.54 | flewid | hello |
| 21:32.58 | flewid | anyone happen to be using pfsense in here? |
| 21:34.23 | J4k3 | wow, vitel sucks. |
| 21:34.34 | J4k3 | they want to blame the originating carrier for the calls loss... except... |
| 21:34.40 | J4k3 | Thats funny because the problem existed from AT&T calling card, Verizon Cellular, and two Crockett-area Valor telecom originations? |
| 21:34.56 | iwannadoit | [isdn-out] |
| 21:34.56 | iwannadoit | exten => _X.,3,Dial(CAPI/g1/${EXTEN}) |
| 21:34.56 | iwannadoit | [ISDN2] |
| 21:34.56 | iwannadoit | isdnmode=msn |
| 21:34.56 | iwannadoit | incomingmsn=xxxxxxxx;your number |
| 21:34.56 | iwannadoit | controller=1 |
| 21:34.58 | iwannadoit | group=1 |
| 21:35.00 | iwannadoit | callgroup=1 |
| 21:35.02 | iwannadoit | bridge=yes ;native bridging (CAPI line interconnect) if available |
| 21:35.03 | Mercestes | omg |
| 21:35.04 | iwannadoit | context=incoming-isdn |
| 21:35.05 | Mercestes | STOP |
| 21:35.06 | iwannadoit | echocancel=yes |
| 21:35.08 | iwannadoit | devices=1 |
| 21:35.15 | *** mode/#asterisk [+b %iwannadoit!*@*] by Corydon76-work |
| 21:35.18 | Mercestes | ~pastebin |
| 21:35.27 | jbot | somebody said pastebin was a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or http://rafb.net/paste/, or http://pastebin.com is usually painfully too slow and unresponsive to use, use one of the other pastebin sites, or dpaste.com is a very nice pastebin as well |
| 21:35.27 | *** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-118-241.ph.ph.cox.net) |
| 21:35.52 | *** part/#asterisk iwannadoit (i=iwannado@196.211.34.2) |
| 21:35.59 | *** join/#asterisk iwannadoit (i=iwannado@196.211.34.2) |
| 21:36.49 | *** mode/#asterisk [-b %iwannadoit!*@*] by Corydon76-work |
| 21:36.50 | J4k3 | what this world needs is a good PBX performance statistics generator. |
| 21:37.03 | J4k3 | something I can drop on a POTS line, call a PBX #, and find out of the call actually got delivered or not. |
| 21:37.12 | J4k3 | and do that say, 3 times a minute. |
| 21:37.30 | iwannadoit | sorry |
| 21:39.05 | Mercestes | pastebin is your friend. |
| 21:39.06 | J4k3 | hell, I'd drop some money right now for a third party to do it. |
| 21:39.19 | kombi | iwannadoit: no problem, I actually have those settings in capi.conf, it must be something really stupid that I'm missing |
| 21:39.23 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 21:40.16 | iwannadoit | dcc me your extension.conf or past it in pastebin |
| 21:41.51 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 21:41.52 | kombi | iwannadoit: http://pastebin.ca/542591 |
| 21:42.24 | kombi | everything before it is the default conf file as generated by make config |
| 21:43.47 | iwannadoit | are you linking to a pots or a pbx???? |
| 21:44.14 | kombi | iwannadoit: linking to the phone network |
| 21:45.40 | kombi | extensions can call each other fine, outbound calls come in, just can't call out |
| 21:46.04 | iwannadoit | you have your ;[capi-out] take away the ; on tne whole context and reload extension in cli |
| 21:47.05 | kombi | i commented that out to enable the line at the top which is in default context.. |
| 21:47.28 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 21:47.42 | iwannadoit | sorry i see do this to that line exten => _X.,1,DIAL(CAPI/ISDN1/${EXTEN}/B,,) |
| 21:47.55 | kombi | those are just my other tries..;) |
| 21:47.56 | iwannadoit | relaod extension.conf in cli |
| 21:48.25 | kombi | by doing restart gracefully.. done! |
| 21:48.38 | iwannadoit | try now |
| 21:48.44 | kombi | no luck.. |
| 21:48.54 | kombi | if only there was some error message.. |
| 21:48.58 | iwannadoit | past your sip.conf for me please |
| 21:49.07 | kombi | coming right up.. |
| 21:49.52 | *** join/#asterisk TJ` (i=ch220207@tj.shells.crazyhosters.com) |
| 21:49.59 | kombi | http://pastebin.ca/542640 |
| 21:50.11 | TJ` | anyone know how to add a trunk password to a follow me rule? |
| 21:50.21 | kombi | again just the lines after the default conf |
| 21:51.49 | iwannadoit | from which sip phone are you dialing? |
| 21:52.09 | kombi | Zentrale and Konrad |
| 21:52.27 | iwannadoit | pick one to test |
| 21:52.42 | kombi | Zentrale then |
| 21:52.48 | kombi | being x-lite |
| 21:54.35 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 21:56.09 | iwannadoit | check this out http://pastebin.ca/542676 |
| 21:56.25 | iwannadoit | remember to restart |
| 21:56.43 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.139.144) |
| 21:57.16 | kombi | put nothing in sip.conf? |
| 21:57.25 | iwannadoit | sorry wait |
| 21:59.05 | iwannadoit | sip http://pastebin.ca/542682 |
| 22:02.39 | kombi | holy crap, something happened!!! |
| 22:02.54 | iwannadoit | waht? |
| 22:02.58 | iwannadoit | what? |
| 22:03.04 | kombi | it ... dials!!! |
| 22:03.11 | iwannadoit | nice |
| 22:03.23 | russellb | it's amazing how excited you can get over just making a successful phone call :-D |
| 22:03.36 | kombi | not quite gets anywhere but that will easy from now.. |
| 22:03.49 | kombi | russelb: my girlfriend says the same thing right now.. |
| 22:03.55 | *** join/#asterisk thoughtpolice (n=austin@c75-111-136-171.plaicmtc01.tx.dh.suddenlink.net) |
| 22:03.56 | russellb | once you have an understanding of a lot of what goes on behind the scenes, it's a great feeling |
| 22:04.08 | kombi | it's like smoking crack! |
| 22:04.27 | russellb | i wouldn't know what crack is like :) |
| 22:04.28 | iwannadoit | all you have to remember is that your sip.conf context you must make a separate context for outgoing |
| 22:04.32 | russellb | but I do know about getting excited about phones ... |
| 22:04.34 | kombi | only lasts a second though and then: on to the next problem! |
| 22:04.59 | kombi | russelb: nor do I to be honest.. |
| 22:05.04 | iwannadoit | i took 3 e tonight so i am in a very good mood |
| 22:05.16 | kombi | iwannadoit: you are disco! |
| 22:05.22 | *** join/#asterisk drgalaxy (n=drgalaxy@adsl-70-238-195-120.dsl.lbcktx.sbcglobal.net) |
| 22:05.28 | iwannadoit | boom boom |
| 22:06.14 | drgalaxy | is there any way to call your asterisk box with a modem and get a TTY on the system? |
| 22:06.27 | kombi | holy crap, it rings in the office! this is so exciting I must get drunk right now! |
| 22:06.40 | iwannadoit | smoke a joint |
| 22:06.46 | kombi | drgalasy: why not ssh into it? |
| 22:06.53 | kombi | or that.. |
| 22:06.53 | tzanger | ManxPower: DTMF on zaptel in latest trunk is MUCH better |
| 22:07.10 | drgalaxy | kombi: in case the 'net is down |
| 22:07.15 | drgalaxy | kombi: want to dial in and diagnose/use the serial port to the router |
| 22:07.29 | flewid | is it crazy to try and get a ip500 behind nat registering to an asterisk box also behind nat? |
| 22:07.48 | drgalaxy | I know you can set up a PPP channel and with that I could then ssh to the box - but I wanted to skip that step |
| 22:08.41 | *** join/#asterisk data23 (i=data@92.b6.3845.static.theplanet.com) |
| 22:09.07 | Mercestes | <PROTECTED> |
| 22:09.14 | flewid | mer: heh, figured :( |
| 22:11.00 | flewid | it registers, the user just can't check voicemail or make calls, and if he rings me and i pickup, it's still ringing on his end and /or drops the call |
| 22:12.38 | *** join/#asterisk ecoleman (n=ecoleman@cpe-76-50-132-99.buffalo.res.rr.com) |
| 22:12.44 | *** part/#asterisk ecoleman (n=ecoleman@cpe-76-50-132-99.buffalo.res.rr.com) |
| 22:13.44 | iwannadoit | kombi? |
| 22:13.51 | kombi | right here! |
| 22:13.52 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 22:13.52 | *** mode/#asterisk [+o mog] by ChanServ |
| 22:14.01 | kombi | smoking.. |
| 22:14.02 | iwannadoit | any luck? |
| 22:14.05 | iwannadoit | k |
| 22:14.06 | iwannadoit | <PROTECTED> |
| 22:14.18 | iwannadoit | fly over here and share |
| 22:14.37 | kombi | yeah, I can call other people fine now! Just how do I call internally and outside from the same phone? |
| 22:15.09 | iwannadoit | wait i'll sho you |
| 22:16.41 | iwannadoit | post your sip.conf and extension.conf again |
| 22:18.38 | *** join/#asterisk jeffik (n=Valued@206-248-152-65.dsl.teksavvy.com) |
| 22:18.50 | kombi | http://pastebin.ca/542736 |
| 22:19.34 | *** join/#asterisk JunK-Y (n=junky@modemcable105.205-56-74.mc.videotron.ca) |
| 22:19.49 | kombi | sip.conf now altered in Zentrale (context = isdn-out) |
| 22:20.23 | drgalaxy | the answer to my question was iaxmodem.sf.net |
| 22:21.17 | kombi | drgalaxy: interesting, does that come with the distro? |
| 22:21.33 | drgalaxy | don't think so |
| 22:22.47 | kombi | mgetty.. makes sense as a last resort for connecting. I'd be afraid of someone breaking in though.. |
| 22:26.21 | iwannadoit | do this http://pastebin.ca/542756 kombi |
| 22:28.11 | drgalaxy | kombi: my goal is simply to have an out of band method to fix issues - some of my clients are hundreds of miles away |
| 22:28.35 | iwannadoit | vpn dr galaxy |
| 22:29.19 | iwannadoit | drgalaxy do your cients have static ip's? |
| 22:29.19 | drgalaxy | iwwannadoit: out of band (ie, the internet is down and I need to diagnose from a router plugged into my asterisk box's serial port) |
| 22:29.19 | iwannadoit | kombi? |
| 22:31.31 | kombi | hmm, not quite yet.. |
| 22:31.39 | iwannadoit | why not dialup slow but efective |
| 22:33.08 | kombi | nope, not working.. the line makes sense even to me though |
| 22:33.45 | iwannadoit | can i dcc you in private |
| 22:33.56 | kombi | what is dcc? |
| 22:34.07 | iwannadoit | accept the next popup |
| 22:34.11 | kombi | I only know that as a spam filter |
| 22:34.32 | kombi | oh, no, I'm on epic here, no pop ups possible.. |
| 22:34.49 | sevard | Heh. I use epic aswell. |
| 22:34.56 | iwannadoit | k |
| 22:35.01 | iwannadoit | here goes |
| 22:35.18 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 22:35.21 | iwannadoit | why have you used names in your sip.conf? |
| 22:35.32 | iwannadoit | as context names |
| 22:35.42 | kombi | thought that might be senible.. |
| 22:35.46 | kombi | sensible.. |
| 22:35.51 | iwannadoit | this is how i will do it |
| 22:35.58 | _VoiceMeUp_COM | anyone good enoug to tell me how to partern match 2 lines ? |
| 22:36.03 | *** join/#asterisk ptiggerdine (n=ptiggerd@203-219-14-182.static.tpgi.com.au) |
| 22:36.22 | _VoiceMeUp_COM | like explode by : but data is on 2 lines.. then next 2 lines is another dataset |
| 22:36.22 | *** join/#asterisk seanwg (n=sean@216.249.37.157.ppp.northrock.bm) |
| 22:36.27 | iwannadoit | is all the users softphones onsite? |
| 22:36.27 | seanwg | byod sip providers in canada -anyone know anyone decent? |
| 22:36.53 | kombi | iwannadoit: one soft phone, one hard phone, the others sleeping |
| 22:36.56 | drgalaxy | _VoiceMeUp_COM, are you trying to parse a text file? |
| 22:37.00 | _VoiceMeUp_COM | yes |
| 22:37.21 | _VoiceMeUp_COM | var1:var2:var3 |
| 22:37.21 | _VoiceMeUp_COM | var4:var5 |
| 22:37.21 | _VoiceMeUp_COM | var1:var2:var3: |
| 22:37.21 | _VoiceMeUp_COM | var4 etc |
| 22:37.27 | drgalaxy | _VoiceMeUp_COM, sounds like a job for regex and the script language you are most comfortable with |
| 22:37.37 | kombi | iwannadoit: would it make sense to goto from bureau to isdn-out instead? |
| 22:37.53 | _VoiceMeUp_COM | yes im trying preg_split |
| 22:38.26 | JunK-Y | :1,$s/line1\nline2/hey/g |
| 22:39.24 | _VoiceMeUp_COM | ahah |
| 22:39.33 | _VoiceMeUp_COM | well i dont know length of lines |
| 22:39.39 | _VoiceMeUp_COM | 1,$ ? |
| 22:39.45 | _VoiceMeUp_COM | sorry about the bold |
| 22:39.45 | drgalaxy | line one through the end |
| 22:39.51 | drgalaxy | that looks like vim-regex |
| 22:40.05 | JunK-Y | _VoiceMeUp_COM: vim... |
| 22:40.10 | seanwg | voiceme up - can you guys do canadian dids? |
| 22:40.23 | seanwg | i want to move on from vonage |
| 22:40.27 | _VoiceMeUp_COM | yes |
| 22:40.28 | sevard | seanwg: I think didx dips into canada |
| 22:40.31 | _VoiceMeUp_COM | pm me |
| 22:40.34 | sevard | oh |
| 22:40.36 | sevard | nm :) |
| 22:41.01 | JunK-Y | sevard: take a look at unlimitel for an excellent canadian service. |
| 22:42.05 | jeffik | sevard: I use Unlimitel for 2 years, they are good |
| 22:42.14 | sevard | I don't need canadian service. |
| 22:42.22 | JunK-Y | jeffik: i know, they're excellent :) |
| 22:42.58 | *** join/#asterisk VJFROMGT (n=vijay_0@user-387g9ui.cable.mindspring.com) |
| 22:43.14 | VJFROMGT | how can i limit the number of channels per extension? |
| 22:43.15 | seanwg | unlimitel - they will work with me - i only need 1 did from calgary |
| 22:43.23 | *** join/#asterisk DaveCanoe (n=Dave@H6.C30.B96.tor.eicat.ca) |
| 22:43.45 | Trevor_b | do they pay you to write advertisements? |
| 22:43.50 | *** join/#asterisk [hC] (n=hardcore@190.10.12.97) |
| 22:43.54 | Trevor_b | ;) |
| 22:43.57 | sevard | hahahaha |
| 22:45.07 | jeffik | sevard: then look at les.net he's in Winnepeg and has numbers all over Canada |
| 22:45.26 | sevard | jeffik: for the second time, I don't need service in canada, and it's spelled Winnipeg. |
| 22:46.38 | jeffik | sevard: sorry i misunderstood you |
| 22:47.00 | sevard | jeffgus: no problem. |
| 22:47.09 | _VoiceMeUp_COM | yeah but they both limit 2 channels |
| 22:47.15 | _VoiceMeUp_COM | we limit 15 default |
| 22:47.19 | _VoiceMeUp_COM | and increase on demand |
| 22:47.26 | iwannadoit | kombi |
| 22:47.35 | _VoiceMeUp_COM | but this is not an advertismenet channel |
| 22:47.45 | _VoiceMeUp_COM | so abck to asterisk ? VJFROMGT use callimit |
| 22:47.52 | Mercestes | That explains your username. |
| 22:47.57 | _VoiceMeUp_COM | its per peer not extension |
| 22:48.03 | _VoiceMeUp_COM | hey its inderect ;) |
| 22:48.08 | Mercestes | direct |
| 22:48.09 | _VoiceMeUp_COM | indirect i mean.. |
| 22:48.13 | Mercestes | uh huh |
| 22:48.19 | _VoiceMeUp_COM | voicempulse gave me the idea whats wrong witht hat ? |
| 22:48.27 | _VoiceMeUp_COM | Mercetes is almost a nice car ;) |
| 22:48.44 | VJFROMGT | thanks voice,, going and google |
| 22:48.46 | sevard | sevrd is an awesome dude |
| 22:49.22 | sevard | so is sevard |
| 22:49.27 | *** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-118-241.ph.ph.cox.net) |
| 22:49.30 | iwannadoit | kombi????????????? |
| 22:49.44 | VJFROMGT | voice,,, didnt find anything on callimit, is there a different name for it? |
| 22:50.49 | _VoiceMeUp_COM | hold on |
| 22:51.06 | sevard | VJFROMGT: it's call-limit |
| 22:51.09 | _VoiceMeUp_COM | http://www.voip-info.org/wiki-Asterisk+config+sip.conf |
| 22:51.19 | VJFROMGT | thanks sevard |
| 22:51.31 | _VoiceMeUp_COM | but no wiki on it |
| 22:51.41 | _VoiceMeUp_COM | nothing hard.. just remmeber its per leg |
| 22:51.45 | _VoiceMeUp_COM | else .. use groups |
| 22:52.11 | Mercestes | _VoiceMeUp_COM, only because you can't spell it. :P |
| 22:52.39 | sevard | it's in the wiki. http://voip-info.org/wiki/index.php?page=Asterisk+sip+incominglimit |
| 22:52.58 | _VoiceMeUp_COM | they chagned the name |
| 22:52.59 | _VoiceMeUp_COM | i guess |
| 22:53.06 | iwannadoit | sorry i am slowing down 4 e's doing damage |
| 22:53.11 | _VoiceMeUp_COM | http://www.voip-info.org/wiki-Asterisk+config+sip.conf |
| 22:53.19 | _VoiceMeUp_COM | check the line call-limit and click |
| 22:53.24 | _VoiceMeUp_COM | then come back and tell me im wrong |
| 22:53.54 | kombi | iwannadoit: here i am, sorry, the phone rang.. |
| 22:54.38 | _VoiceMeUp_COM | btw junky your vim thing is sooooo wrong |
| 22:54.44 | kombi | I used your method the other way round working fine, now I just need to get rid of the leading zero in $EXTEN |
| 22:54.46 | _VoiceMeUp_COM | but thanks anyway |
| 22:54.50 | sevard | _VoiceMeUp_COM: that's because it's not there. once again: http://voip-info.org/wiki/index.php?page=Asterisk+sip+incominglimit |
| 22:55.10 | kombi | how do I get rid of a leading 0 in ${EXTEN}? |
| 22:55.18 | iwannadoit | copy everthing from your old context(forgot thr name) to [isdn-out] context |
| 22:55.39 | *** join/#asterisk saftsack (n=saftsack@pD9E0493F.dip.t-dialin.net) |
| 22:55.53 | iwannadoit | ${EXTEN:1} |
| 22:55.57 | sevard | kombi: answers to these simple questions can be found in http://www.voip-info.org/wiki/view/Asterisk+config+extensions.conf |
| 22:56.04 | kombi | great! |
| 22:56.12 | kombi | thanks sevard! |
| 22:56.18 | sevard | kombi: no problem. |
| 22:56.23 | saftsack | hi, someone with patton gateways here? |
| 22:56.30 | Mercestes | Sevard 1 VoiceMeUp 0 |
| 22:56.34 | iwannadoit | mmemememe patton |
| 22:56.39 | sevard | Mercestes++ |
| 22:56.51 | Mercestes | :D |
| 22:56.53 | kombi | and thanks very much wannadoitnow, you helped me a lot! |
| 22:57.03 | kombi | off to sleep now.. |
| 22:57.06 | sevard | Mercestes: that makes you, what, 812098.6? |
| 22:57.08 | kombi | ran out of e's.. |
| 22:57.28 | Mercestes | sevard: 1. |
| 22:57.36 | Mercestes | no love for the Mercestes |
| 22:57.39 | Mercestes | ~mercestes |
| 22:57.55 | jbot | mercestes is definitely a total nub |
| 22:57.55 | sevard | oh sweet, we're equal. |
| 22:57.55 | Mercestes | Yea. |
| 22:57.55 | iwannadoit | no prob kombi these e's are going to keep me up all night |
| 22:58.05 | sevard | hahaha |
| 22:58.08 | iwannadoit | 8 left |
| 22:58.15 | sevard | ~sevard |
| 22:58.26 | sevard | Mercestes: that's the only way to do it |
| 22:58.26 | Mercestes | Yea. I thank [TK]D-Fender for that. |
| 22:58.40 | sevard | He says everyone's a nub. no worries. |
| 22:58.55 | Mercestes | heh. except himself. |
| 22:58.56 | saftsack | which company builds the best gateways? |
| 22:59.00 | Mercestes | ..and anyone with a green dot. |
| 22:59.20 | iwannadoit | saftsack patton & multitech |
| 22:59.21 | Mercestes | [TK]D-Fender says, "yes, yes your butt *does* taste like candy mr. #asterisk-op" |
| 23:00.03 | saftsack | iwannadoit: i have a patton gw. the voice quality is superb, but whats about its software? my 4552 seems to have some software bugs |
| 23:00.06 | sevard | Mercestes: green dot, eh? What client are you using? |
| 23:00.12 | Mercestes | xchat |
| 23:00.22 | saftsack | is there a discussion group with many persons in it? |
| 23:00.29 | sevard | ah, xchat: "The exploitable client." |
| 23:00.32 | *** join/#asterisk Know1 (i=know1@creep.bur.st) |
| 23:00.34 | Mercestes | Yea |
| 23:00.35 | iwannadoit | saftsack:wahts happening? |
| 23:00.38 | Mercestes | I probably shouldn't admit that publicly |
| 23:00.42 | sevard | well |
| 23:00.48 | Mercestes | I'm on Microsoft windows tho which is completely secure. >.> |
| 23:00.55 | sevard | !?! CTCP VERSION reply from Mercestes: xchat 2.8.0-1 Windows XP [Intel /1. |
| 23:00.56 | sevard | 83GHz] |
| 23:01.05 | Mercestes | Yup, that's me. |
| 23:01.13 | Mercestes | mostly...except when hackers take me over. |
| 23:01.35 | saftsack | if a call is established beetwen one external line and my ip telephone and a second call gets into my gateway all telephones ring but it is impossible to answer the call on one of my internal isdn telephones .... :( no voice is established |
| 23:01.50 | sevard | Heh |
| 23:02.27 | Mercestes | anyone good with perforce? |
| 23:03.03 | iwannadoit | the two port can ony have one te an nt port the second port will not function for it needs a isdn phone: saftsack |
| 23:03.07 | sevard | i haven't even heard of it. content management? |
| 23:03.15 | sevard | conf managment |
| 23:03.17 | Corydon76-work | Source control |
| 23:03.18 | Mercestes | yea |
| 23:03.25 | sevard | oooh proprietary |
| 23:03.30 | Corydon76-work | Like svn, only crappy |
| 23:03.36 | Mercestes | I would love to be able to use p4 in linux but it keeps giving me "You don't have permission you nub" |
| 23:03.47 | Mercestes | Corydon76-work, Funny, I heard the same thing about svn. :P |
| 23:03.51 | sevard | Give yourself permissions. |
| 23:03.57 | saftsack | iwannadoit: i draw it then it looks more clearly ;) |
| 23:03.58 | saftsack | mom |
| 23:04.00 | Corydon76-work | Mercestes: chmod +x p4 |
| 23:04.23 | Corydon76-work | Mercestes: no, that'd be CVS |
| 23:04.26 | Mercestes | It is +x |
| 23:04.33 | Mercestes | Corydon76-work, Oh, your right, actually. lol |
| 23:04.47 | Mercestes | and p4 is 755. :( |
| 23:04.49 | sevard | Mercestes: if you want to pay a consultant to figure it out for you... look over here! |
| 23:04.53 | saftsack | TELCO -ISDN> Patton -SIP> Asterisk -SIP> Patton -> ISDN-terminal -> WORKS FINE |
| 23:05.10 | saftsack | second case .... |
| 23:05.14 | iwannadoit | chmod +X 775 p4 |
| 23:05.16 | Mercestes | sevard: That'd require me to give you access to everything.. |
| 23:05.24 | sevard | Mercestes: you can watch in screen. |
| 23:05.28 | Mercestes | iwannadoit, why oh why would I want to write to my p4 executable?? |
| 23:05.28 | saftsack | TELCO ISDN -> Patton -SIP -> Asterisk -> SIP -> SIP-phone works fine |
| 23:05.35 | Corydon76-work | Mercestes: is p4d running? |
| 23:05.37 | saftsack | but when this call is established |
| 23:05.44 | Mercestes | it's a remote server.... |
| 23:05.48 | saftsack | then TELCO -ISDN> Patton -SIP> Asterisk -SIP> Patton -> ISDN-terminal -> WORKS FINE breaks everytime |
| 23:05.58 | Corydon76-work | Mercestes: do you have your environmental variables set? |
| 23:06.05 | Mercestes | I don't have a p4d. it's jsut a p4 binary to sync with a remote respository |
| 23:06.07 | iwannadoit | you can only have 1 port dual channel |
| 23:06.14 | Mercestes | Yea, i exported them. That's how I got this far. |
| 23:06.16 | sevard | Mercestes: I ssh in with a normal user, give you permissions to my screen, you connect, we collaborate. |
| 23:06.28 | Corydon76-work | Mercestes: do you have available licenses? |
| 23:06.47 | Mercestes | I can p4 sync |
| 23:07.02 | Corydon76-work | Then what's the problem? |
| 23:07.43 | Mercestes | I can't p4 add |
| 23:07.58 | Corydon76-work | Does the license you're connecting as have permissions to add to that directory? |
| 23:07.58 | Mercestes | I want to upload /etc/asterisk to the repository |
| 23:08.00 | Mercestes | how do I check that? |
| 23:08.04 | saftsack | iwannadoit: could it be that the patton doesnt support more than 2 SIP connections at the same time? |
| 23:08.16 | Corydon76-work | I forget, it's been so long |
| 23:08.24 | Corydon76-work | p4 help will give you the commands |
| 23:09.00 | iwannadoit | you have to have a 2d bri isdn interfase from your port then it will handle 2 sip connections |
| 23:10.00 | iwannadoit | you have to have a 2d bri isdn interfase from your pots then it will handle 2 sip connections |
| 23:10.12 | *** join/#asterisk Vorondil (n=vorondil@unaffiliated/vorondil) |
| 23:10.16 | Corydon76-work | The only thing perforce gets you above and beyond what other repositories do is the ability to add symlinks to your repo |
| 23:10.18 | iwannadoit | you have to have a 2a bri isdn interfase from your pots then it will handle 2 sip connections |
| 23:10.25 | iwannadoit | sorry typos |
| 23:11.24 | saftsack | what do you mean with 2a and 2d? |
| 23:11.34 | Mercestes | Yea. and give you a headache in linux |
| 23:11.40 | Mercestes | windows client seems to work fine. =/ |
| 23:12.04 | Qwell[] | Corydon76-work: eh? svn can do links |
| 23:12.34 | Corydon76-work | Qwell[]: it can? |
| 23:12.48 | Qwell[] | sure |
| 23:12.57 | Corydon76-work | Oh, then there's no advantage to using perforce anymore |
| 23:12.58 | sevard | svn can do everything. |
| 23:13.00 | Qwell[] | try svn add'ing a symlink sometime |
| 23:13.09 | Corydon76-work | Qwell[]: yeah, uh, no. |
| 23:13.11 | Qwell[] | I think that's how, anyhow |
| 23:13.32 | Corydon76-work | Qwell[]: my bosses at bamm thought it was the greatest thing. I thought it was rather error-prone |
| 23:13.39 | Qwell[] | it creates a file with some text in it, which points to the other file, and on checkout, it does that |
| 23:13.57 | *** join/#asterisk toerkeium (i=oo@201.216.206.221) |
| 23:14.40 | Mercestes | my boss is pro perforce. |
| 23:14.55 | iwannadoit | kill your boss |
| 23:14.59 | Mercestes | I'm pro-tar -zcvf before I change something but he says that's bad change management. |
| 23:15.32 | *** join/#asterisk mightnare (n=mike@s230165.ppp.asahi-net.or.jp) |
| 23:18.11 | Corydon76-work | Mercestes: 'p4 protect' edits the permissions |
| 23:18.24 | *** join/#asterisk Strom_M (n=strom@208.47.199.4) |
| 23:19.27 | *** join/#asterisk Stridernzl (n=neville@125-239-176-235.jetstream.xtra.co.nz) |
| 23:19.47 | *** join/#asterisk paolob (n=donpaolo@196.3.84.214) |
| 23:19.58 | Mercestes | 'p4 protect' You don't have permission for this operation. |
| 23:20.37 | snuffy22 | i have a question about the 'm' feature of cmd Dial.. if i use successive dial commands it stars the music from the begining again.. (moh mode=files) on my old 1.2 with mode=mp3 it continues through nicely |
| 23:20.49 | Mercestes | read user * * //... |
| 23:20.57 | Mercestes | from protects |
| 23:21.02 | snuffy22 | using 1.4.4 (mode=files) |
| 23:21.21 | Corydon76-work | Mercestes: you need write or super access |
| 23:21.46 | Mercestes | So i'm notcing. |
| 23:22.03 | Mercestes | Thanks. Your awesome |
| 23:22.55 | *** join/#asterisk lee_is_me (n=chatzill@12-201-102-196.client.mchsi.com) |
| 23:23.21 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 23:24.15 | lee_is_me | Hi all, Asterisk 1.2.14. Is it normal for MP3's to start over again if there is only one caller on hold? |
| 23:24.57 | Mercestes | alright, thanks again. I have to go shut down some major servers now |
| 23:24.57 | Corydon76-work | lee_is_me: upgrade |
| 23:25.28 | lee_is_me | Corydon76-work: do you know which version it was fixed? |
| 23:25.59 | lee_is_me | I should also mention that I have mode=files. |
| 23:26.00 | Corydon76-work | lee_is_me: not offhand, no |
| 23:26.20 | lee_is_me | Corydon76-work: but it was a recognized bug that WAS fixed? |
| 23:28.00 | Corydon76-work | Correct |
| 23:28.10 | snuffy22 | it claimed to be.. but with 1.4.4 with files it still starts over |
| 23:28.38 | lee_is_me | I have a customer 1.2.18 and it exhibits the same behavior (starting over) |
| 23:34.14 | Juggie | try 1.2svn first |
| 23:34.19 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 23:34.21 | Juggie | if its still broken then, theres a problem |
| 23:34.28 | blitzrage | anyone here happen to do termination via Broadvox? |
| 23:34.32 | blitzrage | Damin: ping |
| 23:40.26 | seanwg | will this exten work |
| 23:40.40 | seanwg | exten => _1.,1,Dial(SIP/$ @voicemeup,360) ; remove spaces in the exten part |
| 23:40.41 | seanwg | ie send _1XXXX |
| 23:40.49 | seanwg | sorry |
| 23:40.56 | seanwg | if I dialed 1-403-974-44000 |
| 23:41.04 | seanwg | would this dial(SIP/14039744000) |
| 23:41.21 | seanwg | or would I need dial(SIP/${EXTEN}@ |
| 23:42.57 | blitzrage | ${EXTEN} |
| 23:43.04 | blitzrage | why would just $<space> work? |
| 23:45.38 | JT | imagining syntax maybe? :) |
| 23:45.40 | blitzrage | I guess so.... |
| 23:46.09 | JT | btw it is better to Dial(SIP/voicemeup/${EXTEN},360) |
| 23:46.20 | JT | much neater and in line with other channel technologies |
| 23:46.37 | JT | i don't know why people use the @ method at all in SIP |
| 23:47.17 | *** join/#asterisk niedobry (n=bbrindle@ip24-254-142-122.rn.hr.cox.net) |
| 23:48.05 | saftsack | is there a voip discussion board somewhere online? |
| 23:48.23 | JT | there's quite a lot |
| 23:49.54 | saftsack | isnt there THIS board? :> i just know the german ip-phone-forum.de but there are just SOHO solutions |
| 23:49.55 | blitzrage | lists.digium.com |
| 23:50.34 | JT | yeah mailing lists are superior to forums |
| 23:50.48 | blitzrage | so much easier to navigate |
| 23:50.48 | saftsack | ok .... whats about boards wcich are like forums? |
| 23:50.53 | blitzrage | forums are a great waste of time |
| 23:51.04 | blitzrage | but Digium has one |
| 23:51.06 | blitzrage | www.asterisk.org |
| 23:51.20 | saftsack | i know but there arent many people with sangoma and other cards ;) |
| 23:51.23 | JT | forums attract idiots, but as blitzrage, they exist |
| 23:51.31 | JT | as blitzrage said |
| 23:51.34 | *** join/#asterisk frocos11292 (n=ask@80.172.186.100) |
| 23:51.35 | blitzrage | JT: ouch |
| 23:51.44 | blitzrage | :) |
| 23:51.45 | saftsack | hmm so in forums there are just noobs? :> |
| 23:51.50 | blitzrage | typically |
| 23:51.57 | JT | usually |
| 23:51.59 | blitzrage | smart people use mailing lists |
| 23:52.41 | saftsack | ok ... and is there a mailing list which is about voip hardware in general? |
| 23:53.13 | frocos11292 | anyone has done some experiment with iaxclient or jiaxclient ? |
| 23:56.08 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 23:57.24 | frocos11292 | to <blitzrage> have u ever tried with iaxclient ? |
| 23:57.35 | *** join/#asterisk mocker (n=mocker@198.247.173.227) |
| 23:57.36 | blitzrage | frocos11292: no |
| 23:57.52 | frocos11292 | blitzrage: thanks |
| 23:59.15 | *** join/#asterisk mocker (n=mocker@198.247.173.227) |
| 23:59.15 | *** join/#asterisk Stridernzl (n=neville@125-239-176-235.jetstream.xtra.co.nz) |