| 00:01.48 | *** join/#asterisk _pepo_ (n=Pepo@190.10.187.20) |
| 00:05.26 | tobias | I am getting "/usr/lib/asterisk/modules/chan_zap.so: undefined symbol: ast_pickup_call" |
| 00:05.37 | tobias | I don't have any zap hardware I just need to use ztdummy |
| 00:05.43 | tobias | the module's loaded |
| 00:05.47 | tobias | what else could i be missing? |
| 00:09.29 | *** join/#asterisk tpak (n=tpak@c-24-9-108-141.hsd1.co.comcast.net) |
| 00:17.43 | *** join/#asterisk saftsack (n=oliver@p54A7FDF9.dip.t-dialin.net) |
| 00:18.05 | saftsack | hi, does somebody know where i can find the t38modem svn? no hints on google right there :( |
| 00:20.11 | mvanbaak | tobias: what version of asterisk and zaptel ? what OS ? |
| 00:20.19 | obnauticus | Does asterisk have some sort of embedded http proxy that I can use instead of configuring like Apache to do so for me (for my hardphones) |
| 00:20.45 | mvanbaak | obnauticus: what you want to do ? |
| 00:20.53 | mvanbaak | provision the phone using HTTP ? |
| 00:20.56 | obnauticus | Just setup a simple HTTP Proxy for my Cisco 7960 |
| 00:20.56 | obnauticus | 's |
| 00:21.09 | obnauticus | Uhh so when it requests a GET <url> then it will provide it |
| 00:21.10 | mvanbaak | asterisk == voip, not http |
| 00:21.23 | mvanbaak | you'll have to setup a webserver for that |
| 00:21.23 | tobias | obnauticus: * 1.2.13, zap 1.2.11 |
| 00:21.24 | obnauticus | /etc/asterisk/http.conf |
| 00:21.28 | tobias | obnauticus: this is debian etch |
| 00:21.40 | obnauticus | tobias uhh? |
| 00:21.41 | mvanbaak | if you want something simple, use thttpd |
| 00:21.46 | mvanbaak | or just setup apache |
| 00:22.01 | tobias | obnauticus: whoops, i mean mvanbaak |
| 00:22.11 | mvanbaak | tobias: try 1.4 |
| 00:22.19 | tobias | gah |
| 00:22.27 | mvanbaak | 1.2 is no longer maintained (besides security patches that is) |
| 00:22.27 | tobias | it's worked fine before |
| 00:22.44 | tobias | i don't like installing software that's not in my distro |
| 00:22.56 | tobias | because it quickly spins out of control |
| 00:23.19 | mvanbaak | tobias: if you installed asterisk from debian packages you should file a bug against them. 1.2.13 is working fine for me |
| 00:23.24 | tobias | it's actually loading chan_zap now |
| 00:23.26 | mvanbaak | I installed them from source though |
| 00:23.30 | tobias | i just needed to load res_features first |
| 00:23.44 | mvanbaak | or set 'autoload=yes' in modules.conf |
| 00:23.51 | tobias | yes it's set |
| 00:24.02 | tobias | i don't know why it wasn't finding it |
| 00:24.29 | tobias | i am totally baffled |
| 00:24.36 | mvanbaak | I skipped 1.2 |
| 00:24.53 | mvanbaak | friday I upgraded all our boxen from 1.0.9 to 1.4-svn |
| 00:24.54 | tobias | i'm still having a problem with silent recordings, though |
| 00:25.21 | [hC] | [TK]D-Fender: obviously you are quite familiar with sip.cfg overrides. Ive pastebinned my overrides here http://pastebin.ca/796356 - specifically the sound effects/patterns/miscellaneous sectipn, the way ive done overrides would this lead you to believe it would screw up ringers and such, not having the rest of the sound effects tags in that file, etc? |
| 00:25.24 | tobias | 1.2.13 is working fine for you but you skipped it? |
| 00:25.41 | mvanbaak | tobias: my home laptop has been running 1.2.13 |
| 00:25.45 | tobias | ah |
| 00:26.07 | mvanbaak | tobias: my laptop actually has 1.0, 1.2, 1.4 and svn-trunk on it |
| 00:26.22 | *** join/#asterisk jql (n=jql@12.9a.344a.static.theplanet.com) |
| 00:26.27 | mvanbaak | gotta love xen (virtualization) |
| 00:27.17 | tobias | i wonder what my deal is with asterisk configuration |
| 00:27.29 | tobias | i've tried innumerable setups |
| 00:27.42 | tobias | and they all eventually break, even if they seem to work fine for a week or so first |
| 00:27.59 | mvanbaak | that's why you have to upgrade to 1.4 |
| 00:28.06 | mvanbaak | a lot of stuff has been fixed in there |
| 00:28.16 | tobias | *sigh* |
| 00:28.30 | tobias | that is one combo i haven't tried yet |
| 00:29.16 | tobias | this *used to work* though |
| 00:29.23 | tobias | and i haven't even changed anything in /etc/asterisk |
| 00:29.25 | tobias | i swear |
| 00:29.36 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 00:30.25 | tobias | the great thing is everyone has completely differing opinions on what my problems are :) |
| 00:31.42 | lesouvage | I want to relate the max number of calls waiting in a queue with the number of agents that is logged in. Are there queue related variables of functions that can be used to determine this two items (waiters in call, number of logged in agents) |
| 00:33.16 | mvanbaak | lesouvage: I dont think so |
| 00:33.27 | lesouvage | So I can do something like Gotoif(<waiters>= <nmb_agents>?call_back_later) |
| 00:35.13 | lesouvage | mvanbaak: I couldn't find them but there are more hidden and not discussed variables and functions |
| 00:36.57 | *** join/#asterisk etfonhomey_ (n=chatzill@74-131-136-195.dhcp.insightbb.com) |
| 00:36.58 | De_Mon | lesouvage show function QUEUE_<TAB> |
| 00:38.22 | De_Mon | QUEUE_WAITING_COUNT and QUEUE_MEMBER_COUNT are what you're looking for, but the member count isn't the total number of members, not on/off a call |
| 00:42.11 | lesouvage | De_Mon: thanks, I will try something out. |
| 00:43.58 | *** join/#asterisk etfonhomey_ (n=chatzill@74-131-136-195.dhcp.insightbb.com) |
| 00:46.28 | mvanbaak | 01:35 < lesouvage> mvanbaak: I couldn't find them but there are more hidden and not discussed variables and functions |
| 00:46.33 | mvanbaak | that is not true |
| 00:46.50 | mvanbaak | 'show functions' will show you all functions |
| 00:47.02 | mvanbaak | 'show applications' will show you all applications |
| 00:47.19 | mvanbaak | all functions and applications show the variables they use/set |
| 00:47.38 | mvanbaak | so with 'show function <functionname>' you can see all vars that function uses |
| 00:47.44 | mvanbaak | same with applications |
| 00:48.02 | tobias | mvanbaak: how do you overcome timing issues with xen? |
| 00:48.29 | mvanbaak | tobias: my laptop has HVM so I simply install ztdummy |
| 00:48.45 | tobias | HVM? |
| 00:48.48 | lesouvage | mvanbaak: I didn't mean hidden in the sence of not documented but more hidden in the sence that noboddy seems to no. I have been searching for a way to find out if a file exists yes or no and finding the Stat() function toke me a long time and nobody on the channel seem to know about it. |
| 00:50.35 | mvanbaak | lesouvage: if you come from a *nix background you could have guessed that |
| 00:51.04 | mvanbaak | stat is used in a lot of stuff to find out if a file exists or not |
| 00:51.25 | tobias | mvanbaak: i usually have RTC linkage errors when i try to load ztdummy in a VM environment |
| 00:51.42 | mvanbaak | tobias: there's a patch for that on mantis |
| 00:51.48 | mvanbaak | bugs.digium.com |
| 00:52.22 | lesouvage | I think this proofs that I don't have a thorough linux background. (although I know my way around) |
| 00:52.28 | tobias | mvanbaak: patch for zaptel? |
| 00:52.35 | mvanbaak | tobias: yup |
| 00:52.49 | mvanbaak | search for 'ztdummy xen' |
| 00:53.35 | tobias | http://bugs.digium.com/view.php?id=8896 ? |
| 00:53.38 | tobias | whoops |
| 00:53.43 | tobias | i'll try that search |
| 00:54.08 | tobias | hm doesn't show anything |
| 00:54.32 | mvanbaak | 8896 has some pointers yes |
| 00:55.34 | tobias | mvanbaak: are you using that and does it work? |
| 00:55.42 | *** join/#asterisk jozu (n=torrent@84.120.184.91.dyn.user.ono.com) |
| 00:56.01 | mvanbaak | tobias: have a look at this one (including all the comments): http://bugs.digium.com/view.php?id=9592 |
| 01:17.38 | *** join/#asterisk fujin (n=aj@unaffiliated/fujin) |
| 01:27.28 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 01:38.20 | tobias | interesting |
| 01:38.24 | tobias | i do modprobe ztdummy |
| 01:38.39 | tobias | and whenever asterisk tries to play back a recording, i just hear silence |
| 01:38.42 | tobias | i do rmmod ztdummy |
| 01:38.45 | tobias | and everything works fine |
| 01:38.59 | tobias | voice calls work fine either way |
| 01:40.49 | *** join/#asterisk FireMac (n=firemac@CPE000d88ae88b9-CM0011ae8bb0ee.cpe.net.cable.rogers.com) |
| 02:02.45 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 02:04.26 | *** join/#asterisk Op3r (n=edwin@222.127.83.248) |
| 02:04.36 | Op3r | anyone knows this error message? |
| 02:04.38 | Op3r | loader.c:555 load_modules: Loading module chan_oss.so failed! |
| 02:05.42 | *** join/#asterisk snazm (n=snazm@89.242.152.224) |
| 02:05.43 | *** join/#asterisk tripppy (n=u@60-242-11-223.static.tpgi.com.au) |
| 02:07.10 | tripppy | hi, i just got voip setup on my router and i can make phone calls out but my ISP didnt give me a call in phone number. can i use my IP somehow? |
| 02:09.37 | *** join/#asterisk tobias (n=tobias@nat1.ppckernel.org) |
| 02:10.54 | tobias | sorry i lost connection there |
| 02:11.01 | tobias | any thoughts about my ztdummy issue? |
| 02:11.25 | file | tobias: what kernel version? |
| 02:11.39 | tobias | 2.6.18 |
| 02:11.45 | tobias | the debian etch kernel |
| 02:12.18 | Op3r | tobias: what was your issue with your ztdummy? |
| 02:12.44 | tobias | modprobe ztdummy |
| 02:12.55 | tobias | i can't hear any recordings that asterisk tries to play back (but voice calls work fine) |
| 02:12.58 | Op3r | and what was the error? |
| 02:13.00 | tobias | rmmod ztdummy |
| 02:13.10 | tobias | everything works fine (except for conferences, of course) |
| 02:13.14 | tobias | no error |
| 02:13.18 | tobias | i just get silence |
| 02:13.22 | *** part/#asterisk snazm (n=snazm@89.242.152.224) |
| 02:13.40 | tobias | e.g., when calling the voicemail prompt |
| 02:13.43 | Op3r | check if ztdummy is actually modprobbed |
| 02:13.45 | Op3r | lsmod |
| 02:13.47 | tobias | it is |
| 02:14.00 | tobias | and chan_zap finds it |
| 02:14.09 | tobias | because going into a conf doesn't throw an error anymore |
| 02:14.27 | tobias | (in the logs, that is) |
| 02:14.27 | Op3r | recompile your zaptel (thats what I do when Im lazy) and see if it works |
| 02:14.33 | tobias | just did that |
| 02:14.44 | Op3r | errr |
| 02:14.49 | Op3r | what version? |
| 02:14.52 | tobias | purged /dev/zap, purged all the zaptel packages, rebooted, and rebuilt |
| 02:14.57 | tobias | 1.2.11 i think |
| 02:15.08 | tobias | whatever debian has |
| 02:15.18 | tobias | yeah |
| 02:15.20 | tobias | 1.2.11 |
| 02:15.39 | file | ztdummy uses the kernel timers to generate timing, that timing is used for conferences and file playback when available, if the kernel doesn't generate the timing right then ztdummy won't provide timing and things like what you described won't work |
| 02:15.47 | file | I have heard of issues on later kernels but not with that one |
| 02:15.57 | tobias | hm |
| 02:19.03 | *** join/#asterisk UnixDog (n=unixdog@adsl-69-234-183-148.dsl.irvnca.pacbell.net) |
| 02:19.10 | UnixDog | ok whats going on here |
| 02:20.22 | *** join/#asterisk lemanal (n=lemanal@ip68-14-106-198.no.no.cox.net) |
| 02:20.30 | tobias | file: i see a lot of 'rtc: lost some interrupts at 1024Hz. |
| 02:20.30 | tobias | ' in dmesg |
| 02:20.49 | file | that would be why |
| 02:20.53 | tobias | i have put '1024' in /proc/sys/dev/rtc/max-user-freq though |
| 02:20.56 | tobias | so i'm not sure why |
| 02:22.34 | UnixDog | I need help with how to properly do status checking in a dial plan like for dialing to check if call waiting is active or if call forwording is enabled and set |
| 02:22.54 | tobias | lemanal: yo |
| 02:23.09 | lemanal | hey tobias. |
| 02:23.37 | tobias | hmph |
| 02:23.40 | file | UnixDog: active? on what sort of device? |
| 02:23.59 | tobias | i'm stumped again, heh |
| 02:24.06 | tobias | recompile the kernel? |
| 02:24.16 | rob0 | Check the Thurman unit. |
| 02:24.18 | UnixDog | sip phomes like polycoms |
| 02:24.34 | UnixDog | but most of it is dialplan and db correct |
| 02:25.21 | file | UnixDog: if you force the user to do it by calling numbers and such, then it's done server side... if they do it on the phone itself you can't know |
| 02:25.26 | lemanal | tobias: what's that config file that's missing? |
| 02:26.17 | UnixDog | ok we are just talkoing dial plan then all done on server |
| 02:26.37 | tobias | lemanal: hm dunno |
| 02:26.53 | tobias | lemanal: was it an error on /etc/init.d/zaptel start ? |
| 02:27.04 | file | then you use standard logic... Gotoif, ${DB} |
| 02:27.24 | lemanal | dunno |
| 02:27.36 | UnixDog | ok |
| 02:28.01 | UnixDog | do you have some dialplan I can look at to understand |
| 02:28.30 | file | nope |
| 02:30.09 | UnixDog | ? |
| 02:30.37 | *** join/#asterisk Grnd-Wire (n=grundofw@65.101.128.1) |
| 02:30.43 | Grnd-Wire | Greetings! |
| 02:31.44 | UnixDog | btw I am using asterisk 1.4 |
| 02:32.31 | Grnd-Wire | So I am getting started with speech synthesis - and I'm not sure if I should be learning Festival or Flite? |
| 02:32.37 | UnixDog | is there anydial with this function that I can see file |
| 02:32.49 | [TK]D-Fender | UnixDog, There are some samples on the WIKI if you don't have the insight to build it yourself. |
| 02:32.59 | UnixDog | flite I read is better |
| 02:33.15 | UnixDog | but cepstriel is also suppost to be good |
| 02:33.23 | Grnd-Wire | Is that free? |
| 02:33.43 | UnixDog | no |
| 02:33.46 | Grnd-Wire | Good evening [TK]D-Fender |
| 02:33.49 | Grnd-Wire | oh, ok.. :( |
| 02:33.58 | UnixDog | its like 35 bucks I was looking into it |
| 02:34.05 | file | building it yourself is not that hard... just have to think, just like the rest of Asterisk - all the tools to build what you want are there, you just have to put them together |
| 02:34.19 | Grnd-Wire | oh! Well that's very affordable, especially if it works better.. :P |
| 02:34.24 | [TK]D-Fender | Grnd-Wire, y0 |
| 02:35.03 | UnixDog | ? wiki |
| 02:35.25 | UnixDog | I have read the Book |
| 02:35.26 | Grnd-Wire | [TK]D-Fender: Any input on text to speech stuff? I've got a virgin 1.4.14 install, and I want something I don't have to spend alot of time installing.. ? |
| 02:35.42 | tobias | lemanal: ". Effectively the 2.6 version of ztdummy does the same job as zaprtc does for 2.4 kernels." |
| 02:35.45 | UnixDog | get the flite rpm |
| 02:35.48 | tobias | http://www.voip-info.org/wiki/view/Asterisk+timer+ztdummy |
| 02:35.58 | [TK]D-Fender | Grnd-Wire, never touched TTS |
| 02:36.00 | lemanal | ah |
| 02:36.03 | UnixDog | and app-flite5.0.rpm |
| 02:36.45 | Grnd-Wire | UnixDog: hmm.. ok - and how does that integrate with Asterisk? I don't even know where to go for help/documentation |
| 02:37.32 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 02:38.29 | UnixDog | I know they use it on trixbox aka trashbox |
| 02:38.42 | UnixDog | but I did not like it |
| 02:38.44 | tripppy | hi, i just got voip setup on my router and i can make phone calls out but my ISP didnt give me a call in phone number. can i use my IP somehow? |
| 02:38.55 | UnixDog | and there was no one really using it |
| 02:39.13 | UnixDog | you cant really learn on trashbox |
| 02:39.22 | UnixDog | its more a pbx for dummies |
| 02:40.26 | Grnd-Wire | heh.. So Cepstral looks to be pretty promising.. |
| 02:40.57 | *** join/#asterisk tripppy (n=u@60-242-11-223.static.tpgi.com.au) |
| 02:41.21 | *** join/#asterisk RoyK (n=roy@ip-216-23-149-91.dialup.ice.no) |
| 02:41.50 | [TK]D-Fender | tripppy, "set up on your router"? |
| 02:42.43 | Grnd-Wire | Cepstral charges $50 per "port" (how many channels are actively creating speech at once), and voices are active .. and then $30 per voice you want to use |
| 02:44.32 | [TK]D-Fender | Grnd-Wire, Not exorbitant if its decent quality |
| 02:45.17 | Grnd-Wire | [TK]D-Fender: I'm not complaining! They've got a demo section on their site, so you can get WAV files made for specific pieces of text.. That at least allows me to start working on the logic/flow of my diaplan. :) |
| 02:45.28 | Grnd-Wire | www.cepstral.com/demos |
| 02:46.14 | UnixDog | ?wiki |
| 02:46.23 | UnixDog | !wiki |
| 02:47.33 | tripppy | [TK]D-Fender, yes. it has two phone handsets attached . they can both make calls out using my voip account. I CANT CALL INTO it tho.... |
| 02:47.58 | [TK]D-Fender | tripppy, this very clearly has nothing to do with ASTERISK, so why ask here? |
| 02:48.53 | [hC] | [TK]D-Fender: hey, would you mind taking a peek at a sip.cfg overrides file ive been using and maybe affirm that what I'm doing is right? |
| 02:49.02 | [TK]D-Fender | [hC], sure |
| 02:49.24 | tripppy | because i was told alot of voip nerds hang around here. basically is it possible with a SIP phone to call a IP (netcomm nb9w) with handsets attached? |
| 02:49.24 | [hC] | [TK]D-Fender: http://pastebin.ca/796356 |
| 02:50.03 | [hC] | [TK]D-Fender: one major thing im trying to determine, is if i override things like this, and for example the sound_effects tags, if i skip the rest of the insides and only override a specific tag, will it have a negative effect? |
| 02:50.31 | [hC] | [TK]D-Fender: I'm wondering if my everrides of the contents of sound_effects has for some reason made the phones think that that is the ONLY thing in sound_effects now, screwing up ringing tones |
| 02:50.51 | [TK]D-Fender | tripppy, You'll have to see on you router itself if it lets you add a 2nd provider and supports a dialplan to let you choose it. First guess = no, you're probably screwed |
| 02:51.13 | [hC] | [TK]D-Fender: my <phonemac>.cfg file does x###.cfg, myoverrides.cfg, phone1.cfg, sip.cfg incase you're curious about loading order. |
| 02:53.11 | [TK]D-Fender | [hC], AFAIK all config files act like layers over the total sum of configurable parameters. You should be able to do it in 10 steps if you want, layer by layer. |
| 02:54.54 | *** join/#asterisk jetlagmk2 (n=jetlag@70.17.37.23) |
| 02:55.14 | [hC] | [TK]D-Fender: yeah.. Thats what I understand as well. I'm just curious if what i should be doing is duplicating the entirety of sip.cfg and just making my own changes, or if the way im doing it is okay. I suppose if the point was to duplicate sip.cfg and then make changes, that would hose the upgrade process. |
| 02:56.22 | [TK]D-Fender | [hC], No, that part can be explained by how the "-phone" overrides work. This supports the previous thoeries |
| 02:58.40 | [hC] | [TK]D-Fender: hmm however the-phone overrides are explicitly wrapped in <OVERRIDES>, where as im hijacking entire tag sets and only overriding single tags at a time. |
| 02:58.51 | [hC] | [TK]D-Fender: how do you do it? |
| 02:59.42 | [TK]D-Fender | [hC], I don't :) I jsut use the structure given by the base "<mac>.cfg sip.cfg phoneXXX.cfg (renamed)" |
| 02:59.58 | *** join/#asterisk usam (n=alx@124.157.166.145) |
| 03:00.04 | [hC] | [TK]D-Fender: and every time a new software version comes out you port your changes entirely? :) |
| 03:00.38 | [TK]D-Fender | [hC], Where changes to the configs are required, yes. Few things come down the the phone level. |
| 03:01.23 | [TK]D-Fender | [hC], if I wanted something "smarter" I'd *grep* into key fields and auto-generate everything, but for now, search&replace works well for me :) |
| 03:01.25 | *** join/#asterisk plumbus (n=plumbus@unaffiliated/plumbus) |
| 03:01.36 | [hC] | [TK]D-Fender: interesting. that can be pretty tedious work though, I mean the same tags are not in phone1.cfg and sip.cfg |
| 03:02.24 | [TK]D-Fender | [hC], There is very little you need to mod in sip.cfg and it applies to all. In the phone level there's typically about 10 fields. Not a lot... |
| 03:02.27 | [hC] | [TK]D-Fender: so when a software update comes you have to take the new sip.cfg, and transpose your old settings into the new sip.cfg file so that you dont keep running old settings on new software b y accident and get an undesired outcome |
| 03:02.52 | [TK]D-Fender | [hC], I check the changelog to see if the old is compatible first. |
| 03:02.59 | [hC] | ah. |
| 03:04.55 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@216-110-94-252.static.twtelecom.net) |
| 03:09.26 | [hC] | huh.. looking @ how other people do it, and how trixbox does it |
| 03:09.28 | [hC] | i think im doing it wrong |
| 03:10.45 | [TK]D-Fender | [hC], Structure-monger! :p |
| 03:11.03 | [hC] | Haha |
| 03:11.13 | [hC] | Go figure. going thru the pain of tracing all the parent tags back |
| 03:11.18 | [TK]D-Fender | [hC], You are like a D&D "Min-Max"-er and end up outsmarting yourself |
| 03:11.19 | [hC] | and it seems to now be biting me in the ass! |
| 03:11.28 | [TK]D-Fender | ^^^^^^ |
| 03:11.32 | [hC] | Hahaha. |
| 03:11.43 | [hC] | I'm craftier and sneakier than I think. |
| 03:11.45 | [hC] | Or.... something. |
| 03:11.49 | [hC] | :) |
| 03:12.14 | [hC] | I'm overriding using the <sip> tag. Apparently I should be using the <localcfg> tag first of all |
| 03:12.33 | [hC] | and also i didnt realize i could format tags by inserting carriage returns in the lines |
| 03:14.05 | ido | i don't know what that means but it sounded great in my head |
| 03:20.17 | Speedy2 | Anyone here use Sipuras? |
| 03:22.39 | [TK]D-Fender | Speedy2, Yes, SPA-1001's even..... now maybe you can catch up and ask that real question we were waiting for hours ago :) |
| 03:23.23 | Speedy2 | [TK]D-Fender: Just curious if the Linksys versions were any different than Sipura branded |
| 03:23.49 | [TK]D-Fender | Speedy2, not really. |
| 03:23.54 | [TK]D-Fender | Speedy2, Sipura is no more... |
| 03:24.09 | Speedy2 | [TK]D-Fender: I know that... |
| 03:24.47 | Speedy2 | [TK]D-Fender: Going to buy a used one, wondering if Linksys != Sipura model |
| 03:27.06 | [TK]D-Fender | Speedy2, when Linksys bought them out, the changed the logos & plastic casing, thats about it. |
| 03:27.35 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177583123.dsl.bell.ca) |
| 03:28.26 | coppice | aren't those the key elements of the design? |
| 03:29.16 | [TK]D-Fender | coppice, And the big honking wing on the tail ;) |
| 03:30.34 | coppice | does the main chip in a linksys still say its a VCD chip like the sipuras did? |
| 03:31.32 | [TK]D-Fender | coppice, I don't go out and try open-tech surgery whenever I buy something new :) |
| 03:32.56 | coppice | you shop in the wrong places if they won't let you adequately assess your potential purchase |
| 03:35.23 | [TK]D-Fender | coppice, a sacrifice I'm willing to made given the commute :p |
| 03:42.03 | jameswf-home | this is neat, um get digital copies of books you allready own ;) http://textbooktorrents.com |
| 03:43.27 | Speedy2 | [TK]D-Fender: So the same Sipura firmware can go into the Linksys branded devices? |
| 03:43.57 | Speedy2 | [TK]D-Fender: I updated the firmware of an SPA-1001 to a Linksys branded firmware a bunch of stuff broke, so I reverted back. I wanted to get another SPA-1001 and stick the (working) firmware on to it. |
| 03:44.11 | [TK]D-Fender | Speedy2, That I wouldn't bet on. Do you have an issue with the one its got and others available from Linksys? |
| 03:44.42 | Speedy2 | [TK]D-Fender: See above statement. |
| 03:44.56 | [TK]D-Fender | Speedy2, so NONE of them are good for you> |
| 03:45.32 | Speedy2 | [TK]D-Fender: Well, I found a Sipura firmware image that works so I'm happy with it. |
| 03:45.49 | [TK]D-Fender | Speedy2, And works with your unit? |
| 03:46.40 | *** join/#asterisk fujin (n=aj@unaffiliated/fujin) |
| 03:47.42 | Speedy2 | [TK]D-Fender: Well I have a Sipura (non-Linksys) SPA-1001 and I found a Sipura firmware that works for me without issue. I'm worried if I buy a Linksys I can't use that firmware. |
| 03:48.07 | [TK]D-Fender | Well what is the "bunch of stuff" that broke? |
| 03:48.14 | coppice | what's so special about this firmware? |
| 03:48.25 | Speedy2 | I don't know, Linksys broke a bunch of stu |
| 03:48.26 | Speedy2 | stuff |
| 03:48.32 | [TK]D-Fender | Speedy2, how... generic... |
| 03:48.34 | Speedy2 | So I reverted back to Sipura |
| 03:48.43 | Speedy2 | [TK]D-Fender: Caller ID I think was borked |
| 03:52.22 | Speedy2 | [TK]D-Fender: Basically with the latest Linksys-branded firmware, I've experienced a number of problems. My SIP provider (who uses Asterisk) was troubleshooting with me and traced the issues back to the Linksys firmware. |
| 03:52.57 | [TK]D-Fender | Speedy2, And your own local tests? |
| 03:53.37 | Speedy2 | [TK]D-Fender: I couldn't figure out exactly what changes caused it, but the previous firmware worked OK. That was pretty old, Sipura branded firmware. |
| 03:57.48 | *** join/#asterisk joat (n=joat@ip70-160-147-169.hr.hr.cox.net) |
| 04:01.49 | *** join/#asterisk RoyK (n=roy@ip-216-23-149-91.dialup.ice.no) |
| 04:03.08 | *** join/#asterisk techie (n=techie@adsl-76-214-9-1.dsl.lsan03.sbcglobal.net) |
| 04:21.04 | *** join/#asterisk RoyK (n=roy@ip-216-23-149-91.dialup.ice.no) |
| 04:34.50 | *** join/#asterisk usam (n=alx@124.157.166.145) |
| 04:37.08 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 04:37.26 | usam | <PROTECTED> |
| 04:42.07 | *** join/#asterisk thansen|laptop (n=thansen@19.243.sfcn.org) |
| 04:43.05 | *** part/#asterisk joat (n=joat@ip70-160-147-169.hr.hr.cox.net) |
| 04:47.32 | UnixDog | man dial plan is not easy |
| 04:48.12 | UnixDog | and we are doing gialplan for a new embedded system based on bsd+asterisk+php called askozia |
| 04:51.47 | [TK]D-Fender | UnixDog, dialplan is dialplan, it does not matter what system you're using. |
| 04:55.36 | *** join/#asterisk EnigmaCurry (n=user@67.166.72.245) |
| 05:06.12 | *** join/#asterisk GoldFingaZ (n=wt5@bas13-ottawa23-1177808000.dsl.bell.ca) |
| 05:10.16 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@216-110-94-252.static.twtelecom.net) |
| 05:15.46 | BadHorsie | why is zaptel's wathdog disabled by default? |
| 05:15.54 | BadHorsie | s/wath/watch/ |
| 05:18.36 | EnigmaCurry | Anyone know if the SPA3102 has a delay after dialing before it makes a connection outbound? I'm trying to find out why my softphone is connecting instantly but my SPA3102 takes about 5-10 seconds before a connection is made. |
| 05:20.55 | EnigmaCurry | Or does anyone know the general term I should be searching for? Where a device waits to see if the user types more numbers to the dial string? |
| 05:23.34 | [TK]D-Fender | EnigmaCurry, Think of the time it takes to seize a line and then pass DTMF on to it and await the first progress tone. Thats analog for you. |
| 05:24.43 | EnigmaCurry | huh. It's interesting though that *incoming* calls are noticiably faster. |
| 05:25.51 | [TK]D-Fender | EnigmaCurry, well the very second your line rings you don't have to do anything but pick up. You aren't dialing and CREATING delay and waiting for confirmation. Instead the FXO says "Holy shit the line is RINGING, just answer it!" |
| 05:27.05 | file | unless you do callerid... |
| 05:27.27 | [TK]D-Fender | file, bah.. with all those hackers running *, who can trust it anyways ;) |
| 05:27.55 | file | I tweaked my SPA3102 though, only rings once to get the callerid before passing it on via SIP |
| 05:28.47 | EnigmaCurry | file: Your running an SPA3102? How long does it take for you to dial out to where you see the connection on an asterisk console? |
| 05:29.02 | file | EnigmaCurry: you are talking about the FXS port? |
| 05:29.08 | EnigmaCurry | yes |
| 05:29.21 | EnigmaCurry | The FXO is baren for me at the moment |
| 05:29.25 | file | that's different, it's instant because the dialplan I have configured matches everything I dial so it's passed on instantly |
| 05:29.25 | EnigmaCurry | just doing voip |
| 05:29.34 | [TK]D-Fender | EnigmaCurry, Oh well you probably have 1 extra thing to add... FXS dialplan delay <- |
| 05:30.04 | [TK]D-Fender | EnigmaCurry, So if we're talking using the FXS port to dial a number, going through *, then out the SPA's FXO port there is even MORE in the way. |
| 05:30.26 | jameswf-home | BAM |
| 05:30.43 | [TK]D-Fender | EnigmaCurry, On the FXS you can speed up its "send" so that once it knows you aren't going to dial more digits, that it doesn't jsut wait a few seconds. |
| 05:30.49 | EnigmaCurry | I'm taling SPA FXS -> ethernet -> * -> SIP address on the internet |
| 05:31.14 | [TK]D-Fender | EnigmaCurry, Well FXS.... thats purely your SPA's dialplan then |
| 05:31.26 | EnigmaCurry | [TK]D-Fender: OK, that's exactly what I'm after... to speed up the "send" |
| 05:31.52 | [TK]D-Fender | EnigmaCurry, go fix the dialplan so that it knows when you've dialed a "complete" number without waiting for more digits. |
| 05:31.56 | EnigmaCurry | I'm using a cordless... so it sends all the digits pretty fast, I don't need it to wait at all |
| 05:32.13 | EnigmaCurry | Are we talking about the SPA or *? |
| 05:32.38 | [TK]D-Fender | <[TK]D-Fender> EnigmaCurry, Well FXS.... thats purely your SPA's dialplan then <- Good MORNING! |
| 05:32.47 | *** join/#asterisk RoyK (n=roy@ip-216-23-149-91.dialup.ice.no) |
| 05:32.50 | [TK]D-Fender | ~cluebat EnigmaCurry |
| 05:32.51 | jbot | ACTION pulls out a ClueBat (tm) and thwaps EnigmaCurry. |
| 05:33.16 | EnigmaCurry | I apologize, I came here because I'm a newb |
| 05:33.26 | EnigmaCurry | thanks for the help |
| 05:34.02 | [TK]D-Fender | *thwap* |
| 05:34.25 | [TK]D-Fender | EnigmaCurry, www.voxilla.com <- good place to learn all about configuring your SPA |
| 05:34.43 | EnigmaCurry | cool |
| 05:35.27 | *** join/#asterisk dseeb_ (n=dcb@CPE-124-179-235-116.vic.bigpond.net.au) |
| 05:46.16 | jameswf-home | ~newb |
| 05:46.17 | jbot | Don't bother telling us you're a "newb" or a "n00b". We can tell. |
| 05:46.41 | file | [TK]D-Fender: be nice to the locals, they might all rebel one day |
| 05:47.50 | jameswf-home | ~local |
| 05:47.51 | jbot | local is probably like, is your system local to you? Can you physically touch it from where you're sitting, or maybe by going to another room? As opposed, say, to being 1500km from you, accessible only via air and sea (combined) travel, and installed in a restricted-access facility? This matters if we, say, try to restart your system and it doesn't. |
| 05:48.06 | [TK]D-Fender | file, I was nice, not a swear, "newb", RTFM ( good refferal after certainly is not condescending). |
| 05:48.25 | file | I don't want to have to clean up after they slaughter you |
| 05:48.29 | [TK]D-Fender | file, So all in all, I'd give my performance a solid 8.5 :) |
| 05:48.41 | file | it is very difficult to get blood out of carpet |
| 05:48.46 | file | don't ask how I know this |
| 05:49.08 | [TK]D-Fender | file, like in Lethal Weapon "please stand over there on the plastic sheet. Why? *BLAM*" |
| 05:49.44 | jameswf-home | I like to take people out on accident like the car scene in pulp fiction |
| 05:50.25 | *** join/#asterisk zerocod3r (n=Z3R3CoD3@b208d108.dorm.bilkent.edu.tr) |
| 05:52.03 | [hC] | is "on accident" an american thing? In canada we say "by accident" |
| 05:52.09 | [hC] | on accident sounds weird to me |
| 05:52.51 | jameswf-home | printing stuff in french in an english country by mandate sounds wierd to me |
| 05:52.52 | jameswf-home | :) |
| 05:54.13 | jameswf-home | i mean tell the french to get over it, they dont fight back |
| 05:54.31 | [TK]D-Fender | jameswf-home, Canada is officially bilingual, though you'd have trouble with concensus west of Quebec.... |
| 05:54.36 | [hC] | I agree, I dont like french either. |
| 05:54.48 | [hC] | but yeah, the bilingual thing is why that happens. |
| 05:54.59 | [TK]D-Fender | [hC], Va-t'ens tabarnac! |
| 05:55.00 | [hC] | I am pro-making-quebec-their-own-country |
| 05:55.08 | [hC] | <- Westerner! |
| 05:55.34 | jameswf-home | we have been trying to make mexico its own country but seems they like us too much |
| 05:55.50 | zerocod3r | at least people talk about something and I understand exactly what they talk about :) we have anchestor saying to be french to something mean understand nothing |
| 05:56.19 | [TK]D-Fender | ... huh? |
| 05:56.56 | zerocod3r | Eha I have just start to learn asterisk due to that reason dont understand most of things ;) I imply people asking question |
| 05:57.11 | jameswf-home | in the US all profanity is considered french |
| 05:57.14 | [hC] | [TK]D-Fender: dont feel bad, i dont get it either. |
| 05:58.01 | jql | ne shit pas? |
| 05:58.07 | [TK]D-Fender | jameswf-home, I remember when it was "all Greek to me".... |
| 05:58.40 | *** join/#asterisk techie (n=techie@adsl-76-214-9-1.dsl.lsan03.sbcglobal.net) |
| 05:59.20 | jameswf-home | ~french |
| 05:59.21 | jbot | Vous pouvez obtenir de l'aide sur Debian sur le canal #debian-fr - For help in french, please join #debian-fr |
| 05:59.41 | [hC] | what the hell.. you cant modify dhcp/provisioning server settings via the polycom web interface? |
| 05:59.56 | [TK]D-Fender | ~asterisk-fr |
| 06:00.07 | jql | the polycom web interface... sucks. I turn that shit off |
| 06:00.38 | [hC] | yes of course it does. as I am clearly pointing out :) |
| 06:00.46 | [hC] | i just need to reboot the thing remotely and its not registered to my pbx. |
| 06:01.35 | jameswf-home | ~dropdatabase; |
| 06:01.40 | [TK]D-Fender | [hC], hack the power grid :) |
| 06:01.53 | jameswf-home | jbot dropdatabase; |
| 06:01.53 | jbot | So you think you could kill me, did you. You thought you could come in here, in my own home, and take me down without a fight. I would have given you a chance, really. But now, you have unleashed hell. May god have mercy on your soul. |
| 06:02.47 | jql | hell, the web interface reboots the phone as soon as you click any button |
| 06:03.07 | jql | you can't make it NOT reboot the stupid damn thing |
| 06:03.58 | jameswf-home | ~beer |
| 06:03.58 | jbot | ACTION has disconnected (Read error: 99 (Connection reset by beer)) |
| 06:04.08 | jql | you killed jbot |
| 06:04.21 | jameswf-home | ~~ |
| 06:04.22 | jbot | Every moment in which I'm called upon is torture. |
| 06:04.33 | jameswf-home | ~~~ |
| 06:04.33 | jbot | I HATE YOU, jameswf-home!!! |
| 06:04.38 | jameswf-home | sweet |
| 06:04.48 | jameswf-home | ~~~~ |
| 06:04.49 | jbot | ARGH!!! STOP IT jameswf-home!!! |
| 06:05.02 | jameswf-home | roflmfao |
| 06:05.17 | coppice | ~~~~~~~~~~~~~~ |
| 06:05.23 | jameswf-home | ~botsnack |
| 06:05.23 | jbot | aw, gee, jameswf-home |
| 06:05.35 | jameswf-home | ~~~~~ |
| 06:05.36 | jbot | grrrr |
| 06:05.40 | jameswf-home | ~~~~~~ |
| 06:05.41 | jbot | ACTION takes out a revolver and shoots jameswf-home in the head three times. |
| 06:05.46 | jameswf-home | ~~~~~~~ |
| 06:05.46 | jbot | ACTION lets a freakishly huge killer whale named Hugh eat jameswf-home. |
| 06:05.53 | jameswf-home | ~~~~~~~~ |
| 06:05.54 | jbot | You know, this got old a long time ago. |
| 06:06.01 | jameswf-home | yeah it did |
| 06:06.13 | zerocod3r | ~turkey |
| 06:06.23 | jameswf-home | ~fat |
| 06:06.23 | jbot | But I'm on a diet! |
| 06:06.24 | zerocod3r | just wonder is it only sensitive agains programmed words |
| 06:06.38 | zerocod3r | or just searching and creating something logic |
| 06:07.11 | jameswf-home | ~thanksgiving |
| 06:07.11 | jbot | bitte, jameswf-home |
| 06:07.26 | jameswf-home | wtf is a bitte |
| 06:08.04 | zerocod3r | ~jesus |
| 06:08.05 | jbot | jesus is, like, cool, or apparently now EMACS (see emacs) |
| 06:08.13 | coppice | its bitte about the way you are treating it |
| 06:08.48 | jameswf-home | we say we dont have an instruction manual but we have an instruction juan |
| 06:09.21 | coppice | your juan funny guy |
| 06:09.54 | jameswf-home | oh jose can you see by the dawns early light... |
| 06:13.24 | coppice | its a bugs bunny cartoon where ther use that line, isn'it it? |
| 06:15.08 | Speedy2 | heh |
| 06:15.43 | *** join/#asterisk RoyK (n=roy@ip-216-23-149-91.dialup.ice.no) |
| 06:15.45 | coppice | the world's spanish speaking population is growing. there's juan born every minute |
| 06:17.44 | *** join/#asterisk techie (n=techie@adsl-76-214-9-1.dsl.lsan03.sbcglobal.net) |
| 06:18.15 | *** join/#asterisk robl^ (n=robl@pdpc/supporter/monthlybyte/robl) |
| 06:18.49 | jameswf-home | everytime you send a ping you piss of an asian lady |
| 06:18.58 | jameswf-home | s/of/off/ |
| 06:20.16 | BadHorsie | poor bot. |
| 06:20.36 | jameswf-home | n n n n n n n n |
| 06:20.42 | jameswf-home | s/n/y/ |
| 06:20.49 | jameswf-home | s/n/y/g |
| 06:20.53 | jameswf-home | sweet |
| 06:21.40 | jameswf-home | can i cause ann odd loop hmmmmmmm |
| 06:21.53 | jameswf-home | s/can/~/ |
| 06:24.13 | coppice | maybe jbot can translate |
| 06:24.40 | coppice | ~/maybe/或者 |
| 06:25.05 | jameswf-home | jbot: talk to monty |
| 06:25.06 | jbot | ACTION chatters endlessly to to monty |
| 06:25.17 | coppice | s/maybe/或者 |
| 06:25.35 | coppice | maybe jbot can translate |
| 06:25.41 | coppice | s/maybe/或者/ |
| 06:26.01 | coppice | s/can/會/ |
| 06:26.26 | jameswf-home | jbot generate all permutations of "sarcasm" |
| 06:26.53 | jameswf-home | jbot: be random |
| 06:26.54 | jbot | Chaos! Chaos! Pi! E! Help! Weather! |
| 06:28.11 | tzafrir_home | ~bot abuse |
| 06:28.11 | jbot | ACTION huddles in the corner, whimpering 'please, please stop' |
| 06:28.39 | coppice | s/translate/ç¹™è¯/ |
| 06:29.09 | coppice | s/translate/ç¹™è¯/ s/can/會/ s/maybe/或者/ |
| 06:30.00 | *** join/#asterisk techie (n=techie@adsl-76-214-9-1.dsl.lsan03.sbcglobal.net) |
| 06:30.03 | *** part/#asterisk techie (n=techie@adsl-76-214-9-1.dsl.lsan03.sbcglobal.net) |
| 06:30.37 | jameswf-home | ~cat |
| 06:30.38 | jbot | it has been said that cat is officially used to concatenate files. cat is also used to display the contents of a file on screen. Syntax: cat (file1) (file2) ...(fileN) Where file1 through fileN are the files to display. Example: cat letters/from-mdw displays the file letters/from-mdw. or a clawed walking stomach that meows, or ... |
| 06:31.04 | coppice | ~pussy |
| 06:31.04 | jbot | Read: coppice |
| 06:31.09 | obnauticus | ~vagina |
| 06:31.09 | jbot | from memory, vagina is something that i dont have but i like to suck, SUCK PUSSY YEAH!!! |
| 06:31.17 | obnauticus | ... |
| 06:31.20 | obnauticus | fucking bot. |
| 06:31.21 | obnauticus | lol. |
| 06:32.55 | jameswf-home | jbot: you suck |
| 06:32.56 | jbot | and very well I might add |
| 06:33.15 | jameswf-home | ~god |
| 06:33.16 | jbot | god is, like, a llama, or real unless declared integer |
| 06:33.37 | [hC] | i wonder how much he knows about |
| 06:33.38 | jameswf-home | ~wiki vagina |
| 06:33.39 | [hC] | ~kram |
| 06:33.39 | jbot | methinks kram is a jerk |
| 06:33.44 | [hC] | haha |
| 06:33.59 | SwK | ~jbot jerk |
| 06:34.00 | jbot | i heard jerk is the derivative of acceleration with respect to time |
| 06:34.17 | jameswf-home | ~hasselhoff |
| 06:34.31 | jameswf-home | ~bsd |
| 06:34.32 | jbot | bsd is probably a UNIX operating system. An asterisk port is currently being worked on. |
| 06:34.40 | jameswf-home | ~linux |
| 06:34.41 | jbot | i guess linux is the cure for cancer, AIDS and slavery to corporations |
| 06:34.49 | jameswf-home | ~windows |
| 06:34.49 | jbot | windows is probably a 32 bit hack on a 16 bit operating system, originally designed for an 8 bit CPU, with a 4 bit system bus, made by a 2 bit company that can't stand 1 bit of competition... or the World of Warcraft bootloader, or the most important collection of bugs |
| 06:35.07 | jameswf-home | ~wow |
| 06:35.08 | jbot | I have no life | Lets go raid! |
| 06:35.20 | coppice | ~jbot jerk is also the derivative of politian, without respect to anything |
| 06:35.21 | jbot | ACTION suddenly yanks on the leash around is also the derivative of politian, without respect to anything's neck |
| 06:35.38 | BadHorsie | lol |
| 06:35.49 | jameswf-home | ~porn |
| 06:35.50 | jbot | Porn remains one of the largest problems with Open Source Software. Often causing development delays, flooded links and, in extreme cases, disabling programmers ability to type. |
| 06:35.51 | coppice | s/politian,/politician,/ |
| 06:36.12 | jameswf-home | ~emo |
| 06:36.13 | jbot | /wrists |
| 06:36.15 | jameswf-home | lmao |
| 06:36.27 | *** part/#asterisk dseeb_ (n=dcb@CPE-124-179-235-116.vic.bigpond.net.au) |
| 06:36.28 | BadHorsie | you guys need a life, hahaha |
| 06:37.15 | SwK | obviously we do... |
| 06:37.25 | SwK | its 12:30 on a saturday night and we're home on irc |
| 06:37.27 | jameswf-home | ~grass |
| 06:37.28 | jbot | rumour has it, grass is /me wishes grass was emo so it would cut it's self |
| 06:38.47 | BadHorsie | i'm having a gin while reading TFOT... |
| 06:39.26 | BadHorsie | i was checking some boot camps, but, 3000 USD? |
| 06:40.29 | coppice | isn't the whole point of boot camps to make you suffer? |
| 06:40.51 | BadHorsie | ~curse |
| 06:40.51 | jbot | i guess swearing is Silence, you sock-clucking mother-trucker |
| 06:41.00 | BadHorsie | hehe |
| 06:41.47 | jameswf-home | ~dns 127.0.0.1 |
| 06:41.59 | jameswf-home | ~dns google.com |
| 06:42.09 | obnauticus | ~dns \\ |
| 06:42.13 | obnauticus | damns. |
| 06:42.19 | obnauticus | that would be netbios though :\ |
| 06:42.26 | coppice | jbot curse is also May you live for a thousand installs. |
| 06:42.27 | jbot | May the fleas of a thousand camels infest your most sensitive regions, is also May you live for a thousand installs. ! |
| 06:42.34 | obnauticus | ~dns 10.0.0.0 |
| 06:43.00 | coppice | ~dns 192.168.1.1 |
| 06:43.34 | jameswf-home | ~kick coppice |
| 06:43.35 | jbot | ACTION kicks coppice |
| 06:44.00 | coppice | ~lart jameswf-home |
| 06:44.00 | jbot | judo chops jameswf-home |
| 06:44.14 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 06:50.43 | jameswf-home | ~loboyomy |
| 06:50.53 | jameswf-home | ~lobotomy |
| 06:50.53 | jbot | I feel different somehow. |
| 06:51.21 | jameswf-home | ~bye |
| 06:51.22 | jbot | cya |
| 06:52.47 | robl^ | anyone here have a chance to use an AA50 yet? Any opinions / major shortcomings / gotchas ? I am considering one for a small site. |
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| 07:49.26 | yang | Which is a good VOIP related forums ? |
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| 08:36.02 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 08:45.38 | *** part/#asterisk dudes (n=nixtux@74-60-94-233.stc.clearwire-dns.net) |
| 09:15.19 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 09:43.25 | *** join/#asterisk mkl1525 (n=qwertz@212.204.47.147) |
| 09:46.13 | mkl1525 | Hi, just read that in realtime setup queue agents are updated if a new call comes in - does this mean if I've already a caller in queue and an agent is inserted in the db the agent had to wait till another caller comes in? or does the retry time work in this setup too? |
| 09:51.45 | *** join/#asterisk inv_arp[work] (n=junya@c-76-26-24-197.hsd1.fl.comcast.net) |
| 09:59.27 | *** join/#asterisk RoyK (n=roy@ip-216-23-149-91.dialup.ice.no) |
| 10:07.26 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 10:08.49 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 10:27.34 | *** join/#asterisk xxoxx (n=xxoxx@tor/regular/xxoxx) |
| 10:49.05 | phix | hihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihih |
| 10:50.14 | Speedy2 | Having fun? |
| 10:50.58 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 10:52.59 | phix | Speedy2: nearly |
| 10:53.06 | phix | hihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihihih |
| 10:53.16 | phix | ok fun++; |
| 10:56.31 | *** join/#asterisk d-k-t (n=dt@60.176.197.184) |
| 10:58.01 | ido | hi phix |
| 10:58.41 | ido | so, of all of the free/open source software pbxes out there, how would you rank the top three, in order of best to worst? |
| 11:01.18 | phix | ido: hi |
| 11:01.26 | ido | hi phix |
| 11:01.32 | phix | sup |
| 11:01.37 | phix | how is your day/ |
| 11:01.39 | phix | or night |
| 11:01.41 | phix | or morning |
| 11:01.50 | ido | morning here, 5am |
| 11:01.53 | ido | my day's okay |
| 11:16.49 | *** join/#asterisk briantumor (n=echelon@ool-44c7f686.dyn.optonline.net) |
| 11:16.53 | briantumor | hi |
| 11:18.35 | briantumor | in sip.conf.. where it says. register => 1234567901:password@proxy01.sipphone.com |
| 11:18.44 | briantumor | should password be the actual password? |
| 11:18.52 | ido | quite. |
| 11:19.03 | briantumor | ok, i thought maybe it was a formatting thing |
| 11:19.24 | ido | you may want to double check the permissions on your configuration file to ensure they are the strictest possible |
| 11:19.27 | *** join/#asterisk RoyK (n=roy@ip-216-23-149-91.dialup.ice.no) |
| 11:19.48 | briantumor | i'm just following this.. http://support.gizmoproject.com/index.php?_m=knowledgebase&_a=viewarticle&kbarticleid=166 |
| 11:20.08 | ido | yes, i'm just saying...be careful of putting your passwords anywhere in plain text |
| 11:20.14 | ido | in general, not just in asterisk |
| 11:20.16 | briantumor | oh alright |
| 11:20.36 | briantumor | ah, thanks for reminding me |
| 11:21.03 | briantumor | yup.. only root has read access |
| 11:21.16 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
| 11:21.30 | briantumor | ok.. now that i have sip.conf and extensions.conf configured |
| 11:21.34 | briantumor | what do i do? :S |
| 11:23.51 | briantumor | strange.. when i installed the package there was supposed to have been an /etc/rc.d/rc.asterisk file installed |
| 11:27.32 | briantumor | hmm.. perhaps i should add the user and group |
| 11:38.12 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 11:41.53 | ido | you really ought to read the asterisk installation guide / manual |
| 11:42.17 | tzafrir | ido, which specific one? |
| 11:42.24 | ido | there are a few |
| 11:42.41 | ido | though i didn't have trouble just reading the code and figuring out what goes where |
| 11:43.10 | ido | http://www.voip-info.org/wiki-Asterisk+installation+tips <-- there are links to different ones here |
| 11:43.40 | ido | and there's of course the INSTALL/README/whatever files |
| 11:58.39 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 12:16.06 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 12:23.59 | beasty | anyone knows if it's possible to receive sms'es with asterisk ? |
| 12:30.54 | *** join/#asterisk usam (n=alx@124.157.166.145) |
| 12:50.19 | d-k-t | beasty, where are you wanting to receive them from? |
| 12:54.44 | *** join/#asterisk RoyKa (n=roy@ip-216-23-149-91.dialup.ice.no) |
| 12:55.41 | beasty | d-k-t: my cellphone |
| 13:00.34 | mosty | beasty, how would you connect your cellphone to asterisk? |
| 13:02.07 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 13:03.42 | beasty | i have a sip provider for my asterisk ? |
| 13:03.47 | beasty | so i'll just enter the nr |
| 13:11.55 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 13:16.12 | *** join/#asterisk RoyK (n=roy@ip-216-23-149-91.dialup.ice.no) |
| 13:17.10 | *** join/#asterisk juuva (i=juuva@peili.org) |
| 13:26.41 | *** part/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 13:32.29 | *** join/#asterisk loompek (n=NoName@noname.rula.net) |
| 13:34.12 | loompek | hi.. i've got a little ol question |
| 13:34.20 | loompek | about localization... |
| 13:35.01 | loompek | indestead of "2 hunderd 80 8" i'd like asterisk to say "2 hundred 8 and 80" |
| 13:35.12 | loompek | in my language... |
| 13:35.20 | loompek | where could i set that for 'say numbers' |
| 13:35.29 | tzafrir | loompek, what language is that? |
| 13:35.33 | loompek | slovenian |
| 13:35.56 | tzafrir | Have a look at say.conf |
| 13:36.04 | loompek | i belive germany has the same |
| 13:36.04 | tzafrir | for some languages it is good enough |
| 13:39.07 | *** join/#asterisk Strom_M (n=strom@ip70-170-60-8.lv.lv.cox.net) |
| 13:40.33 | juuva | asterisk-ooh323c seams to crash sometimes |
| 13:42.31 | juuva | tzafrir: which h323 stack you would recommend? |
| 13:43.17 | tzafrir | juuva, not sure |
| 13:43.56 | juuva | isn't one packaged with asterisk 1.4? |
| 13:46.54 | juuva | actually asterisk seams to crash every time when callername or callerid is not set (calling sip -> ooh323c) |
| 13:47.04 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 13:47.11 | loompek | argh |
| 13:47.13 | loompek | gdammnit! |
| 13:47.33 | loompek | even though i copied [de] to [sl] i still keep getting english voices :S |
| 13:47.48 | loompek | do i need to include say.conf anywhere? |
| 13:47.55 | loompek | i set language=sl in sip.conf |
| 13:48.33 | loompek | i mean.. it says the numbers in my language.. but the order is incorrect :S |
| 13:52.36 | tzafrir | loompek, have you set language=sl ? |
| 13:52.47 | tzafrir | in the channel config ? |
| 13:58.32 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 14:00.26 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 14:02.41 | robl^ | morning everyone |
| 14:03.05 | coppice | ooh323c seems to have gone very quiet. has it been abandoned? |
| 14:03.17 | *** join/#asterisk bantu (n=Miranda@p54A32CFD.dip0.t-ipconnect.de) |
| 14:06.27 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 14:06.54 | loompek | tzafrir yes i have.. and i also tried with Set("SIP/1-08211230", "CHANNEL(language)=sl") |
| 14:06.55 | loompek | err |
| 14:07.05 | loompek | exten => _0.,1,Set(CHANNEL(language)=sl) |
| 14:07.10 | loompek | no luck |
| 14:36.36 | juuva | coppice: might be abandoned, propably I got to try another h323 implementation |
| 14:40.37 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 14:40.58 | Siya | <PROTECTED> |
| 14:40.58 | Siya | app_rxfax.c:60: warning: data definition has no type or storage class |
| 14:41.05 | Siya | does that ring a bell with anyone |
| 14:43.04 | tzafrir | Siya, sounds remotely familiar. What version of spandsp? What version of app_rxfax.c (from where?) |
| 14:44.02 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 14:46.26 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 14:48.07 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 14:53.22 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53) |
| 14:58.31 | *** part/#asterisk RoyK (n=roy@ip-216-23-149-91.dialup.ice.no) |
| 14:58.49 | *** join/#asterisk RoyK (n=roy@ip-216-23-149-91.dialup.ice.no) |
| 14:59.10 | juuva | asterisk-h323 seams to tolerate missing values much better than ooh323c, no crashing yet |
| 14:59.23 | robl^ | morning [TK]D-Fender |
| 14:59.53 | Siya | tzafrir: asterfax-1.1-freeb4.i386.rpm |
| 15:00.10 | Siya | trying to follow these instructions: http://asterfax.sourceforge.net/Installing%20AsterFax.html |
| 15:00.20 | Siya | though I'm on Debian (* etc from svn) |
| 15:04.45 | *** join/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 15:04.56 | Siya | I think libspandsp came with asterfax |
| 15:05.27 | Siya | though I installed libspandsp1 and libspandsp-dev through apt |
| 15:06.24 | Siya | I had to symlink /usr/lib/libspandsp.so.0.0.1 to /usr/lib/libspandsp.so |
| 15:15.36 | *** join/#asterisk tobias (n=tobias@nat1.ppckernel.org) |
| 15:19.35 | Siya | tzafrir: it seems to trip over STANDARD_LOCAL_USER; and LOCAL_USER_DECL; |
| 15:20.32 | tzafrir | Siya, that sounds like a mighty old spandsp. What version of spandsp is it? |
| 15:20.52 | Siya | tzafrir: how can I figure that out? |
| 15:20.56 | tzafrir | dpkg -l libspandsp-dev |
| 15:21.09 | Siya | :) |
| 15:21.11 | Siya | doh |
| 15:21.25 | Siya | 0.0.2pre26-1 |
| 15:21.35 | Siya | too old? |
| 15:21.42 | tzafrir | ah, right. that's the version in Etch |
| 15:21.45 | tzafrir | reasonable |
| 15:22.04 | tzafrir | though probably too old for current rx_fax, tx_fax |
| 15:22.36 | tzafrir | more up-to-date debs are available at the pkg-voip.buildserver.net |
| 15:22.49 | tzafrir | but I suspect that spandsp has not had its share of testing |
| 15:26.38 | Siya | tzafrir: i can build from source or would you not advise that? |
| 15:27.50 | tzafrir | Siya, that same repo also has app_rxfax and app_txfax packages... |
| 15:28.34 | tzafrir | but those indeed depend on the specific Asterisk |
| 15:32.14 | *** join/#asterisk bkw_ (n=brian@adsl-64-149-41-151.dsl.tul2ok.sbcglobal.net) |
| 15:32.53 | *** part/#asterisk bkw_ (n=brian@adsl-64-149-41-151.dsl.tul2ok.sbcglobal.net) |
| 15:54.40 | *** join/#asterisk davixx (n=davixx@85.69.181.206) |
| 15:57.25 | hi365 | tzafrir: hi :) |
| 15:58.01 | tzafrir | Hi |
| 15:58.13 | tzafrir | hi356:-) |
| 15:58.15 | hi365 | how are you. ive got a quick questions |
| 15:58.31 | hi365 | question |
| 15:59.07 | hi365 | if i use externapass option in voicemail.conf - when a user changes his password does it get updated imidiatly? |
| 15:59.37 | hi365 | i.e. in asterisks memeory? i presume its updated i the voicemail file ?? |
| 16:00.27 | tzafrir | I don't remember. I'll have to look at the code... |
| 16:01.03 | robl^ | hi365: it *should* be, as long as permissions are set to allow the user running the asterisks process to modify the file |
| 16:01.09 | tzafrir | should update in memory, I guess |
| 16:01.50 | hi365 | robl^: you mean that asterisk wont do it automaticly? |
| 16:02.49 | robl^ | hi365: it depnds on filesystem permissions asterisk needs write permission to the file containing the password |
| 16:03.19 | hi365 | robl^: assuming that asterisk HAS the required perm, and i run a script - will the vm file be updated? |
| 16:03.46 | tzafrir | hi365, you have externpass and externpassnotify |
| 16:04.17 | tzafrir | externpass means: "don't edit voicemail.conf . your command should do that" |
| 16:04.42 | tzafrir | externpassnotify mean "I'll change voicemail.conf, and then run your external command" |
| 16:05.00 | hi365 | cool. thats the one i need then. |
| 16:05.13 | hi365 | are any variables passed to the script? |
| 16:05.56 | robl^ | hi365: ohh.. extern pass. yeah, its the script.. but you still need to make sure the scripts have the correct permissions to allow them o modify the files |
| 16:06.16 | tzafrir | snprintf(buf,255,"%s %s %s %s",ext_pass_cmd,vmu->context,vmu->mailbox,newpassword); |
| 16:06.48 | hi365 | thanks tzafrir ! and you robl^ |
| 16:07.10 | hi365 | one last thing - why in the worl would someone want to edit the vm file "manualy"?? |
| 16:07.22 | hi365 | s/worl/world |
| 16:12.56 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 16:14.01 | tzafrir | then you'd have to reload |
| 16:14.49 | *** join/#asterisk ghento (n=ghento@d216-232-170-111.bchsia.telus.net) |
| 16:25.24 | *** join/#asterisk GoldFingaZ (n=wt5@bas13-ottawa23-1177808000.dsl.bell.ca) |
| 16:28.23 | *** join/#asterisk blq (n=Bl@dslb-088-067-025-155.pools.arcor-ip.net) |
| 16:40.41 | *** join/#asterisk mamep (i=fallen@helios.edu.uoc.gr) |
| 16:42.52 | *** join/#asterisk tobias (n=tobias@nat1.ppckernel.org) |
| 16:44.27 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 16:45.10 | tzafrir | oh323 is probably quite dated by now |
| 16:45.23 | tzafrir | the latest released version will not build. |
| 16:45.30 | tzafrir | latest cvs will |
| 16:46.54 | *** part/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 16:48.58 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 16:49.01 | [TK]D-Fender | CVS ... *snicker* |
| 16:50.31 | tobias | heh |
| 16:50.39 | tobias | [TK]D-Fender: figured out my problem |
| 16:50.50 | tzafrir | [TK]D-Fender, CVS of oh323 . latest released oh323 probably does build with latest CVS of Asterisk |
| 16:51.18 | tobias | [TK]D-Fender: the key was discovering that after rmmod ztdummy recordings played back fine |
| 16:51.23 | [TK]D-Fender | ...CVS... <- |
| 16:51.27 | tobias | (but of course conferences didn't work) |
| 16:51.36 | mamep | anyone experienced with chan_h323? |
| 16:51.58 | tobias | [TK]D-Fender: so that led me to question ztdummy's usage of the RTC |
| 16:52.43 | tobias | after rebuilding ztdummy with RTC disabled and rebuilding my kernel with a timer freq of 1000, everything is happy |
| 16:53.32 | [TK]D-Fender | tobias, I have heard of this before but didn't think of it when you asked... |
| 16:53.57 | *** join/#asterisk Greek-Boy (n=email@41.221.58.5) |
| 16:56.33 | tobias | i wonder why ztdummy uses the RTC at all |
| 16:56.47 | tobias | it seems so finicky in a significant number of configurations |
| 16:57.14 | tobias | i guess recompiling one's kernel is not something that everyone wants to do |
| 16:57.29 | tobias | but it's a hell of a lot less work that debugging the problems when they arise |
| 16:58.01 | tobias | s/that/than |
| 16:58.32 | *** join/#asterisk GoldFingaZ (n=wt5@bas13-ottawa23-1177808000.dsl.bell.ca) |
| 17:11.48 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 17:14.10 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:14.10 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 17:16.25 | *** part/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 17:17.40 | *** join/#asterisk scream_ (n=stas@87.110.230.26) |
| 17:23.01 | *** join/#asterisk Nest0r (n=asd@201.230.44.106) |
| 17:23.05 | Nest0r | Hi |
| 17:23.20 | mamep | anyone experienced with chan_h323? ?? |
| 17:25.09 | moemoe | anybody who can tell me why i can get called, but outgoing calls fail: http://www.noname-ev.de/pastebin/37 |
| 17:40.18 | tzafrir | moemoe, hmm... what version? |
| 17:41.30 | moemoe | asterisk 1.4.13 (debian-unstable-package) |
| 17:41.43 | moemoe | zaptel 1.4.5 |
| 17:45.35 | *** join/#asterisk d-k-t (n=dt@60.176.197.184) |
| 17:49.06 | Siya | hmmm same error |
| 17:49.24 | *** join/#asterisk JayTee52 (n=jforde05@c-69-137-243-25.hsd1.in.comcast.net) |
| 17:50.40 | Siya | tzafrir: would sort it for me? Or would my solution be to find a different source of asterfax as the app_rxfax.c I have now did come with the asterfax rpm |
| 17:51.36 | tzafrir | Siya, I'm looking into http://sourceforge.net/projects/agx-ast-addons |
| 17:53.13 | *** join/#asterisk GoldFingaZ (n=wt5@bas13-ottawa23-1177808000.dsl.bell.ca) |
| 17:54.20 | tzafrir | moemoe, do you happen to have any SIP / IAX phone to test with? |
| 17:54.33 | tzafrir | I wonder if they would show the same problem |
| 17:55.30 | moemoe | tzafrir: yes, just tested with my snom. still the same failure |
| 17:55.57 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 17:58.32 | Kobaz | speaking of h323 |
| 17:58.32 | moemoe | http://www.noname-ev.de/pastebin/38 this is the same problem with the snom |
| 17:58.55 | Kobaz | mamep: what have you found so far in terms of getting h323 to go? |
| 17:58.58 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 18:01.38 | Kobaz | is there a way to reload the voicemail settings without completely restarting asterisk? |
| 18:04.20 | *** join/#asterisk MrParity (n=MrParity@dslb-088-076-195-117.pools.arcor-ip.net) |
| 18:04.26 | MrParity | hello :-) |
| 18:05.11 | MrParity | i'm new to asterisk dialplan programming and i want to execute a local program after a call, but it don't work :-( |
| 18:05.52 | MrParity | i have tow exten lines: "6523771,1,VoiceMail(021516523771,s)" and "6523771,n,System(/root/example.rb &)" |
| 18:06.44 | tzafrir | moemoe, oh, it's dialing outside. |
| 18:07.12 | MrParity | there must be anything wrong, but i don't know what it is :-( |
| 18:07.16 | tzafrir | moemoe, one thing to try: set pridialplan=unknown |
| 18:07.17 | MrParity | any idea? |
| 18:07.20 | tzafrir | in zapata.conf |
| 18:07.34 | tzafrir | It usually just works |
| 18:12.33 | moemoe | tzafrir: great, now it works :D |
| 18:24.17 | *** join/#asterisk salviadud (n=dude@131.178.57.9) |
| 18:30.40 | Kobaz | anyone know how to reload the voicemail config without completely restarting asterisk |
| 18:33.48 | MrParity | Kobaz: reload |
| 18:34.01 | MrParity | (reloads everything) |
| 18:35.14 | Kobaz | yeah that doesn't do it, that's the first thing i did |
| 18:35.31 | Kobaz | i have to kill asterisk and start it again, and then the changes take effect |
| 18:37.05 | juuva | try restart gracefully, it waits until no channels are open before restarting asterisk |
| 18:37.22 | Kobaz | yeah we want to have to not do that |
| 18:37.58 | juuva | if reload won't do what you want, then it's propably not possible without restart |
| 18:39.37 | moemoe | WARNING: chan_sip.c:1938 retrans_pkt: Maximun retriex exceeded on transmission foo@bar for seqno2 (Critical Response) |
| 18:40.28 | moemoe | WARNING: chan.sip.c: 1962 retrans_pkt: Hanging up call foo@bar - no reply to our critical packet |
| 18:40.36 | moemoe | okay, next problem :/ |
| 18:41.14 | moemoe | but before the call works for about 30s |
| 18:41.57 | *** join/#asterisk The_Charlie (n=J_Cutler@131.178.46.136) |
| 18:43.09 | The_Charlie | Wazup guys... i´m new and i´m new at Asterisk i hope you dont get mad if i ask an basic (stupid jeje) question |
| 18:43.17 | The_Charlie | sorry, a basic |
| 18:45.32 | moemoe | but the phone is in the same net as the asterisk, so this cant be a nat problem |
| 18:50.06 | tzafrir | moemoe, what is asterisk trying to do there, exactly? |
| 18:52.05 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 18:53.20 | *** join/#asterisk hardwire (n=bip@rdbck-6096.palmer.mtaonline.net) |
| 18:53.26 | moemoe | tzafrir: this kicks me out of a ongoing phonecall sip -> asterisk -> zaptel -> isdn |
| 18:53.53 | hardwire | tzafrir: so asterisk isn't in debian testing because of other dependencies? or did something else happen? |
| 18:55.04 | tzafrir | hardwire, the automated answer: http://bjorn.haxx.se/debian/testing.pl?package=asterisk |
| 18:55.28 | hardwire | I see it, I just have no idea what that means |
| 18:55.41 | hardwire | its waiting for other packages, eh? |
| 18:56.12 | hardwire | yate maintainers gunking up the works? |
| 18:58.10 | hardwire | tzafrir: I just have no idea where the red flag is in all of this |
| 18:59.02 | robeph | moemoe: does it actually say Maximun retriex exceeded or is that sposed to be retries |
| 18:59.22 | robeph | did the devs typo in the error msgs heh |
| 18:59.28 | tzafrir | moemoe, I don't have any bright ideas |
| 18:59.41 | moemoe | no i had no console running inside screen and so couldnt copy the message |
| 18:59.42 | tzafrir | so maybe a higher debug level, or sip debug |
| 19:00.02 | moemoe | before are just the message that the call was successfully established etc |
| 19:00.05 | robeph | ah ok |
| 19:00.08 | robeph | heh just wondering |
| 19:00.21 | tzafrir | moemoe, most messages also go to the logs |
| 19:00.21 | moemoe | but currently my isdn doesnt work "d-chan on span1 down" so i cant even reproduce the msg |
| 19:00.45 | robeph | you turned on sip debug? |
| 19:00.50 | tzafrir | moemoe, I figure that it is down and is only up when you call out, right? |
| 19:00.56 | tzafrir | or when you call in |
| 19:01.32 | moemoe | tzafrir: no, i cant even call out, service unavailable |
| 19:01.52 | robeph | moemoe: turn on sip debug when ya can that prolly has some decent info in there regarding this |
| 19:02.13 | robeph | `sip debug` to turn it on |
| 19:02.14 | moemoe | okay, im waiting for my d-chan and try again |
| 19:02.21 | tzafrir | try calling in |
| 19:02.34 | tzafrir | from a mobile or whatever |
| 19:02.41 | robeph | ok, majority of the time the debug info will give you a good clue |
| 19:04.51 | moemoe | my mobile terminates the call w/o any msg |
| 19:04.59 | moemoe | and all the time before i didnt get that message |
| 19:05.47 | robeph | moemoe: all your routing set up right as far as port fwd and such? |
| 19:06.39 | moemoe | robeph: yes, asterisk and sip are in the same subnet, only connection to outside is zaptel, so there cant be such a thing like missing port forwardings |
| 19:10.11 | moemoe | http://www.noname-ev.de/pastebin/43 okay, here is the failure again |
| 19:10.15 | robeph | well, just I had an issue where a box had iptables routing my local stuff funky.. |
| 19:10.22 | robeph | from the same box it was on |
| 19:10.54 | moemoe | and for some reasons, my phone still thinks it is connected |
| 19:11.39 | robeph | hmm |
| 19:13.04 | robeph | moemoe: what kinda hardware ya using with that t1? |
| 19:13.12 | moemoe | just one thought - could this be a hardware problem? |
| 19:13.27 | robeph | heh that was what I was going to ask heh |
| 19:13.49 | moemoe | robeph: k6-2 350mhz, 512mb ram, 3c905 network controller, a noname-hfc-card |
| 19:14.28 | robl^ | [TK]D-Fender: you around? |
| 19:17.15 | robeph | moemoe: could it just be you're using an unstable package? you tried using a production release? |
| 19:17.46 | robeph | i dunno whats causing it off hand though. |
| 19:18.07 | moemoe | robeph: i used the package out of debian unstable, becaus i didnt want to install a new asterisk using 1.2 |
| 19:18.39 | robeph | just build it yourself, don't use the packages ;p |
| 19:19.41 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@216-110-94-252.static.twtelecom.net) |
| 19:21.04 | robeph | some things I just can't use packages, too much to worry with how they built it and all |
| 19:21.33 | hardwire | tzafrir: any attempts yet at segregating the asterisk modules into packages? |
| 19:21.40 | hardwire | I'd be willing to take a crack at it |
| 19:22.05 | hardwire | that may help keep it in testing, while allowing only specific modules to be unavailable |
| 19:22.50 | moemoe | hmmm ill try it later after i moved to my new flat ;) |
| 19:23.56 | *** join/#asterisk bkw_ (n=brian@adsl-64-149-41-151.dsl.tul2ok.sbcglobal.net) |
| 19:25.22 | briantumor | hi! |
| 19:25.39 | briantumor | i'm trying to build asterisk with zaptel |
| 19:25.49 | briantumor | but i get this error.. The Zaptel installation on this system appears to be broken. |
| 19:26.22 | briantumor | i thought zaptel is packaged with asterisk? |
| 19:27.00 | tzafrir | hardwire, but asterisk migrates as a source package |
| 19:27.22 | briantumor | hello? |
| 19:28.30 | tzafrir | briantumor, where do you get this error from? what versions of asterisk and of zaptel? |
| 19:28.31 | hardwire | hmm |
| 19:29.32 | briantumor | i tought zaptel is packaged with asterisk source? |
| 19:29.33 | briantumor | for linux |
| 19:30.00 | briantumor | "Zaptel on Linux is available alongside Asterisk from www.asterisk.org or any of the Asterisk mirrors" |
| 19:30.11 | briantumor | oh.. it's not in the same source package? |
| 19:30.17 | tzafrir | no |
| 19:30.20 | d-k-t | there is a seperate zaptel package |
| 19:30.27 | briantumor | where? |
| 19:30.30 | d-k-t | available from the same place |
| 19:31.00 | briantumor | ok.. which should i get? |
| 19:31.03 | *** join/#asterisk SwK (n=SwK@user-24-214-55-149.knology.net) |
| 19:31.08 | briantumor | 1.2 or 1.4? |
| 19:31.17 | d-k-t | latest? |
| 19:34.46 | MrParity | does anyone know how i con do something after VoiceMail() ? |
| 19:35.01 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 19:35.09 | MrParity | are i'm just to stupid to do that? |
| 19:35.35 | *** join/#asterisk businesspartner (n=bb@123.98.181.58.dynamic.max.com.pk) |
| 19:40.14 | CrazyTux[m] | How would I listen on multiple ports in sip.conf for *? |
| 19:40.23 | CrazyTux[m] | i.e. 5060, 5061, etc. |
| 19:45.12 | *** join/#asterisk RoyK (n=roy@ip-216-23-149-91.dialup.ice.no) |
| 19:54.55 | [TK]D-Fender | MrParity, if they hangup DURING Vm to terminate it, the next priority will not get executed. That will fall on the "h" standar extension |
| 19:55.01 | [TK]D-Fender | robl^, back |
| 19:55.12 | blitzrage | CrazyTux[m]: don't think you can do that |
| 19:55.15 | robl^ | [TK]D-Fender: hey hey! welcome back |
| 19:56.09 | CrazyTux[m] | blitzrage, yea google, returned the same |
| 19:56.56 | MrParity | [TK]D-Fender: hmmm... thanks, but if i use h i can not decide which umber it is, right? |
| 19:56.59 | robl^ | [TK]D-Fender: just wanted to see if you had any experience with the AA50, and if so had any feedback on current state, performance, features (or lack of), etc. Considering one for a new small install and trying to solicit some feedback before ordering |
| 19:57.24 | blitzrage | MrParity: unless you set the values into a channel variable ahead of time |
| 19:57.41 | [TK]D-Fender | robl^, no experience. Its a puny embedded system and not a solution I would ever target |
| 19:58.09 | MrParity | blitzrage: ah, okay. i'll try that. |
| 19:58.47 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
| 20:00.30 | robl^ | [TK]D-Fender: *nod* yeah.. the person I am considering the AA50 for was wanting a small Nortel BCM system (~15 phones). He's not too keen on "computer based phones system" and wants appliance-ish. figured it might be a good sell |
| 20:01.10 | blitzrage | robl^: or AstLinux on a Soekris is similar idea |
| 20:01.15 | [TK]D-Fender | robl^, And what does that shmuck think the BCM is? Its a friggen P1 + **WINDOWS** |
| 20:01.21 | robeph | heh |
| 20:01.34 | [TK]D-Fender | Or was that a 486? |
| 20:01.47 | robeph | just build an asterisk machine and put it inside of a non pc looking chassis, he'll never know |
| 20:01.50 | blitzrage | 486 I think |
| 20:01.55 | blitzrage | running winnt |
| 20:01.59 | [TK]D-Fender | BCM is widely regarded as a flamining piece of shit by most of the telelcom industry |
| 20:02.01 | robl^ | [TK]D-Fender: it's actually Linux based.. but a low end PC |
| 20:02.12 | robeph | nortel switched to nix |
| 20:02.15 | [TK]D-Fender | WINNT <--- |
| 20:02.16 | blitzrage | robl^: it is now -- it wasn't until about 1-2 years ago |
| 20:02.23 | robeph | which is odd, didn't they also sign a telephony bit with m$? |
| 20:02.34 | [TK]D-Fender | Apple Computers "Crash Different" (tm) |
| 20:02.36 | *** join/#asterisk billybongo (n=rich@82-33-82-73.cable.ubr03.trow.blueyonder.co.uk) |
| 20:02.56 | *** join/#asterisk techie (n=techie@adsl-76-214-9-1.dsl.lsan03.sbcglobal.net) |
| 20:02.57 | robl^ | right the older BCMs were windows.. but the BCM50 and newer BCM200 and 400 are Linux now |
| 20:03.08 | robeph | robl^: how is that not "pc based" system though heh |
| 20:03.17 | [TK]D-Fender | robl^, * really undercuts BCM no matter what... |
| 20:04.20 | robl^ | robeph: I *KNOW* its pc-ish.. but the person I am trying to do an install for can't grasp the concept. he sees it as just something that hangs on the wall and magically works |
| 20:04.51 | robl^ | [TK]D-Fender: yup! I am trying every tactic to sway him to something Asterisk based. |
| 20:05.27 | robeph | robl^: yeh, I know you know, i just was saying the guy sounds a bit silly. |
| 20:05.39 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 20:05.49 | robeph | robl^: he sounds like the kind of guy that if you make a powerpoint presentation big enough on the wall with a projector he'll listen |
| 20:06.51 | robl^ | robeph: you gotta understand.. if he was more tech savvy-- he would likley want to build an asterisk system himself. this guy is just an office administrator. he can send emails and play with word and excel |
| 20:06.58 | blitzrage | if not -- then run -- he's more work than he's worth |
| 20:07.02 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 20:07.18 | *** part/#asterisk billybongo (n=rich@82-33-82-73.cable.ubr03.trow.blueyonder.co.uk) |
| 20:08.53 | robl^ | blitzrage: I would like to run, I think. not sure if I can, as I would end up having christmas dinner with him. this person is a relative. |
| 20:09.16 | blitzrage | ugh |
| 20:09.56 | robeph | robl^: yeh I was kidding, I know the type, my old job had an it guy that was ill suited for the position and didn't like the idea of new technologies...guess it scared him since he was working off a ccna from like 10 yrs ago |
| 20:11.19 | Siya | eek , notes in asterisk cvs: remove the uses of the deprecated STANDARD_LOCAL_USER |
| 20:11.50 | Siya | how do I adapt to this when app_rxfax.c and app_txfax.c both refer to this definition |
| 20:12.08 | robl^ | robeph: Yup... I was thinking if I could get him hooked with the AA50, I could sell him on an "upgrade" later ;-) |
| 20:12.13 | robeph | hahahah |
| 20:12.20 | Kobaz | allrightey |
| 20:12.43 | Kobaz | so i got the ooh323 channel driver going, netmeeting can make a call, and an iax extension will ring |
| 20:13.10 | Kobaz | but once the iax picks up, netmeeting still is waiting for a connection and eventually times out |
| 20:13.24 | Kobaz | i'm essentially in the same situation this guy is in: http://www.mail-archive.com/ooh323c-devel@lists.sourceforge.net/msg00349.html |
| 20:13.34 | Kobaz | anyone else use h323? |
| 20:15.25 | Kobaz | [Nov 25 15:15:10] WARNING[12458]: channel.c:2634 ast_indicate_data: Unable to handle indication 3 for 'OOH323/Mark H323-9a9b' |
| 20:32.39 | *** join/#asterisk j0 (n=dan@S01060016b6b541d2.va.shawcable.net) |
| 20:33.12 | *** join/#asterisk saftsack (n=oliver@p54A7E0E0.dip.t-dialin.net) |
| 20:33.22 | saftsack | hi is somebody here who compiled t38modem with sip support |
| 20:33.55 | j0 | outbound CID isn't working on my SIP trunks (IAX is fine)... when looking at SIP debug, it has From, Contact, set to *@ip.address |
| 20:34.26 | j0 | ah.. i should probably be in the trixbox forum.. :) |
| 20:34.31 | *** join/#asterisk niekie (i=niek@bergnetworks.com) |
| 20:36.14 | Kobaz | okay |
| 20:36.17 | Kobaz | robeph: hmm |
| 20:36.22 | Kobaz | robeph: no more indication error |
| 20:36.38 | Kobaz | robeph: but the endpoints still cant communicate :( |
| 20:38.08 | Kobaz | http://www.pastebin.ca/797179 |
| 20:40.36 | Grnd-Wire | So how do I troubleshoot "NOT REACHABLE" messages with IAX trunks? I am at my wits end here. Trying to link 1.4.14 (first time to use 1.4) with a previously installed 1.2 server. |
| 20:40.51 | saftsack | hi building t38modem with sip hangs at this point for me .... g++ -DUSE_OPAL -D_REENTRANT -fno-exceptions -Wall -DNDEBUG -I/usr/local/share/pwlib//include -DPTRACING -I../opal_v2_4_0//include -DUSE_LEGACY_PTY -Os -felide-constructors -Wreorder -c opal/sipep.cxx -o obj_linux_x86_opal_r/sipep.o |
| 20:41.01 | fujin_ | Grnd-Wire: not configured correctly |
| 20:41.03 | Grnd-Wire | I've got IAX debugging turned on.. I see a whole lot of PINGs, PONGs, POKEs, and ACKs on both sides.. |
| 20:41.04 | fujin_ | check the network configuration |
| 20:41.08 | saftsack | does somebody have any time to help me? |
| 20:41.19 | fujin_ | Grnd-Wire: are they across a lan, or routed across the intertrons? |
| 20:41.25 | Grnd-Wire | fujin_: Of course it's not configured correctly.. |
| 20:42.16 | Grnd-Wire | fujin_: Yeah, intertrons.. One side doesn't have a firewall at all, the other side has 4569 forwarded correctly.. I actually have one half of the connection up, it is qualifying and showing OK when you do a show iax2 peers. |
| 20:42.50 | fujin_ | Grnd-Wire: one side is sitting on a public IP, with no firewall whatsoever? and one is NATTED, with 4569 forwarded? |
| 20:42.58 | fujin_ | can you nmap <> 4569 |
| 20:43.02 | fujin_ | from either one to the other? |
| 20:43.15 | fujin_ | (nmap -vv -sS -p 4569 ) |
| 20:43.15 | *** join/#asterisk asdx (n=diego@adsl-150-178.click.com.py) |
| 20:44.03 | Grnd-Wire | fujin_: Let me check.. |
| 20:45.45 | Grnd-Wire | fujin_: hmm.. Aren't we working with UDP here? |
| 20:46.01 | fujin_ | right |
| 20:46.01 | fujin_ | yeah |
| 20:46.03 | fujin_ | sU |
| 20:46.06 | fujin_ | =fail |
| 20:46.17 | fujin_ | you should be able to see the port with Netcat |
| 20:46.38 | fujin_ | nc -u -v <hostname> <port> |
| 20:48.08 | *** join/#asterisk techie (n=techie@adsl-76-240-177-78.dsl.lsan03.sbcglobal.net) |
| 20:48.11 | Grnd-Wire | Yeah, so when I execute that - I should be getting garbage on the screen, right? Like protocol negotiation stuff? |
| 20:53.15 | fujin_ | Grnd-Wire: you should see it talk to it |
| 20:53.19 | fujin_ | or at least connect to the port |
| 20:53.34 | fujin_ | Grnd-Wire: is it listening on the right interface? |
| 20:53.44 | fujin_ | netstat -l|grep 4569 |
| 20:54.25 | Grnd-Wire | fujin_: I have voipstreet running on the "unreachable" host already - so I know we're in a good place to make this work, and alot of the stupid stuff shouldn't be the cause. |
| 20:54.44 | fujin_ | I'm not familiar with voipstreet |
| 20:54.51 | fujin_ | It talks to IAX on the unreachable? |
| 20:55.17 | fujin_ | What's the IP of the unreachable one |
| 20:56.01 | Grnd-Wire | fujin_: Yeah, they are an ITSp |
| 20:57.53 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 20:59.20 | Grnd-Wire | You know.. I'm starting to wonder if my provider is filtering 4569.. ugh.. That would explain this behaviour.. |
| 20:59.59 | Grnd-Wire | I can execute the nc command one way and it works properly - I execute it the other, and it fails from the host.. From another host, it works just fine.. (so I know it's not my firewall, or my configuration) |
| 21:01.08 | *** join/#asterisk GoldFingaZ (n=wt5@bas13-ottawa23-1177808000.dsl.bell.ca) |
| 21:01.59 | fujin_ | Grnd-Wire: I've not heard of it.. that is quite odd. |
| 21:02.14 | fujin_ | would you like me to attempt a connection to it from here, in NZ? |
| 21:02.21 | fujin_ | (just an nmap -sU |
| 21:03.20 | *** join/#asterisk etfonhomey (n=chatzill@74-131-136-195.dhcp.insightbb.com) |
| 21:03.24 | Grnd-Wire | fujin_: No thank you.. You've actually been very helpful, now I know I can use netcat to diagnose UDP connectivity issues.. I've always been stuck unless it was TCP, and I could telnet to it. :P |
| 21:03.24 | Nugget | telnet is eeeeeeevil! |
| 21:03.42 | fujin_ | heh. I *generally* always nmap -sS or nmap -sU |
| 21:03.59 | fujin_ | netcat is pretty much the equivalent for UDP as telnet is for TCP, in terms of checking if a port is responding anyways |
| 21:04.06 | Siya | Is anyone actually using asterfax on asterisk-1.4? |
| 21:04.28 | Grnd-Wire | PORT STATE SERVICE |
| 21:04.28 | Grnd-Wire | 4569/udp open|filtered unknown |
| 21:04.32 | Siya | I'm running into issues due to rxfax and txfax using old definitions |
| 21:04.51 | Grnd-Wire | fujin_: So is that output good, or bad? |
| 21:05.24 | Siya | http://forums.asteriskit.com.au/index.php/topic,283.0.html for anyone willing to take a look at the issue |
| 21:05.33 | Siya | gnight all |
| 21:10.42 | *** join/#asterisk Strom_M (n=strom@ip70-170-60-8.lv.lv.cox.net) |
| 21:15.32 | Grnd-Wire | fujin_: Did you see where I pasted the output of nmap? |
| 21:16.29 | fujin_ | yes |
| 21:16.29 | fujin_ | that'd imply to me that it's open, but being filtered on some level |
| 21:16.29 | fujin_ | if not by a stateful router through a NAT |
| 21:16.29 | fujin_ | then by a firewall on the device |
| 21:16.30 | fujin_ | if not that, then upstream (which may be hard to pinpoint) |
| 21:17.17 | Grnd-Wire | fujin_: yeah.. ok - It is being filtered by my firewall, but since I do pass IAX2 traffic - that is certainly not the issue. |
| 21:17.45 | fujin_ | are you passing *all* IAX2 traffic, i.e.; not just state established,related and new? |
| 21:18.23 | Grnd-Wire | fujin_: I am able to setup and accept calls from Voipstreet, so I am confident my configuration is sound. |
| 21:18.42 | fujin_ | hrm |
| 21:18.44 | fujin_ | yeah, guess so |
| 21:19.20 | Grnd-Wire | fujin_: Someone just told me about insecure=invite,port |
| 21:19.31 | fujin_ | Heh |
| 21:19.34 | fujin_ | Have you not got those already? |
| 21:19.37 | Grnd-Wire | fujin_: I can't find anything information on voipinfo about the insecure keyword, other than that it exists.. |
| 21:19.42 | fujin_ | you generally *have* to do it, over the intertrons |
| 21:19.46 | fujin_ | that's because voipinfo is shit |
| 21:19.57 | fujin_ | vi /usr/src/asterisk-*/configs/iax.conf.sample |
| 21:20.05 | Grnd-Wire | fujin_: Well, I had insecure=very .. but that's cause all I know is 1.2 at this point |
| 21:20.13 | *** join/#asterisk karme (n=user@dslb-088-067-044-006.pools.arcor-ip.net) |
| 21:20.53 | Grnd-Wire | umm - that file doesn't even reference "insecure" |
| 21:22.04 | fujin_ | It doesn't? |
| 21:22.16 | Grnd-Wire | nope! |
| 21:22.27 | Grnd-Wire | grep for it |
| 21:22.52 | Grnd-Wire | grep "insecure" iax.conf.sample |
| 21:23.08 | *** part/#asterisk karme (n=user@dslb-088-067-044-006.pools.arcor-ip.net) |
| 21:23.24 | fujin_ | hm |
| 21:23.29 | fujin_ | I think insecure is SIP-specific. |
| 21:23.52 | Grnd-Wire | Yeah, because it has to the with the IP-headers as opposed to the INVITE packets |
| 21:24.29 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 21:24.33 | fujin_ | sorry, this is getting a little out of my comfort zone |
| 21:24.39 | fujin_ | I've never really bothered with IAX. |
| 21:24.48 | fujin_ | Have always had enough bandwidth, and sane configurations to warrant using SIP |
| 21:24.59 | Grnd-Wire | heh.. It's out of my comfort zone too.. |
| 21:25.44 | *** join/#asterisk |omni| (n=rob@c-67-185-144-206.hsd1.wa.comcast.net) |
| 21:26.30 | Grnd-Wire | fujin_: Thanks for at least talking it through with me.. |
| 21:29.32 | *** join/#asterisk Defraz (n=tim@24-116-152-177.cpe.cableone.net) |
| 21:37.58 | Grnd-Wire | fujin_: I have decided that the remote asterisk host isn't even TRYING to connect to me on 4569/UDP - I'm not seeing it on my firewall (accept and log).. I do howevery see the NMAP come through on the logs. |
| 21:45.52 | *** join/#asterisk techie (n=techie@adsl-76-214-31-151.dsl.lsan03.sbcglobal.net) |
| 21:56.27 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 22:03.04 | *** join/#asterisk CVirus (n=GoD@62.135.96.14) |
| 22:04.31 | *** join/#asterisk The_Charlie (n=J_Cutler@131.178.46.136) |
| 22:13.25 | *** join/#asterisk techie (n=techie@adsl-76-214-21-220.dsl.lsan03.sbcglobal.net) |
| 22:15.09 | *** part/#asterisk briantumor (n=echelon@ool-44c7f686.dyn.optonline.net) |
| 22:20.47 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 22:29.20 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
| 22:30.25 | fujin_ | Grnd-Wire: that's odd indeed |
| 22:33.47 | loompek | indeed |
| 22:36.42 | *** join/#asterisk ZakMcRofl (n=unknown@dslb-084-057-045-091.pools.arcor-ip.net) |
| 22:37.00 | ZakMcRofl | hey all, major noob with questions here |
| 22:38.29 | ZakMcRofl | i have a fritzbox (voip router on which asterisk can be installed) and if want calls from different internal S0-MSN's to use different outgoing settings |
| 22:39.08 | Freman | my polycoms keep breeding |
| 22:39.14 | Freman | I left on friday with 5 on my desk |
| 22:39.18 | Freman | I came in on monday and there's 7 |
| 22:39.34 | Freman | problem is, they appear to have inbread cos the new arrivales are retarded |
| 22:40.18 | *** join/#asterisk craigk (n=ckowald@58.174.122.198) |
| 22:40.22 | ZakMcRofl | calls from internal S0 share a context right now. should i edit capi.conf (to have more contexts depending on MSN) or should i filter which MSN the call came from in extensions.conf |
| 22:41.08 | Freman | the revision F phones work fantastically, the revision 4 phones won't load the bootrom, they won't register the handset (all calls on speaker phones) |
| 22:41.45 | ZakMcRofl | so if anyone knows how to set this up or a link to a tutorial, please let me know |
| 22:41.54 | *** join/#asterisk ccaron (n=cedric@85-218-16-115.dclient.lsne.ch) |
| 22:45.12 | ZakMcRofl | so there's 231 people in here and nobody answers? is it 1) because my question is too lame 2) because everybody idles or 3) because my question is too hard (doubt it) |
| 22:50.02 | [TK]D-Fender | ZakMcRofl, Its sunday following thanksgiving in the USA. |
| 22:50.10 | [TK]D-Fender | ZakMcRofl, So clearly #3 :p |
| 22:50.54 | [TK]D-Fender | ZakMcRofl, Since you already know the 2 approaches for this, pick whichever you like! |
| 22:51.19 | ZakMcRofl | the problem is i dont know how to implement them and if they would work |
| 22:51.57 | fujin_ | you're doing it wrong |
| 22:52.01 | ZakMcRofl | i would prefer an option where i can say "calls coming from MSN 123-> contenxt = msn |
| 22:52.07 | ZakMcRofl | msn123 i mean |
| 22:52.10 | [TK]D-Fender | ZakMcRofl, pastebin your capi.conf and your extensions.conf |
| 22:52.12 | [TK]D-Fender | ~pb |
| 22:52.12 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 22:52.13 | [TK]D-Fender | ^^^^^^^^^^^^^^^^ |
| 22:52.29 | ZakMcRofl | ok hold on. might be embarrassing, total noob here ;) |
| 22:53.08 | ZakMcRofl | http://pastebin.ca/797348 |
| 22:53.20 | ZakMcRofl | ISDN3 (internal S0). comments NOT mine |
| 22:53.35 | ZakMcRofl | posting extensions after cleanup in a few |
| 22:56.03 | JT | ISDN3, eh? |
| 22:56.11 | ZakMcRofl | http://pastebin.ca/797353 |
| 22:57.10 | ZakMcRofl | yep i have a ISDN box with 3 (internal) MSNs on ISDN3 |
| 22:57.37 | ZakMcRofl | and i'd like them to set a different outgoing id or use a different SIP account |
| 22:58.15 | ZakMcRofl | also if someone knows a cleaner solution for this, let me know: |
| 22:58.15 | ZakMcRofl | exten => sipid0,1,Set(Var_TO=${SIP_HEADER(TO)}) |
| 22:58.15 | ZakMcRofl | exten => sipid0,2,GotoIf($["${Var_TO}" = "<sip:004989123454430@sipgate.de>"]?geli,sipid0,1:3) |
| 22:59.12 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 22:59.27 | ZakMcRofl | (i'm distingushing by the <TO>-Field in the header to find out which number the called dialed. but its unrelated to the ISDN3/ internal S0 problem |
| 22:59.30 | JT | ZakMcRofl: are you on drugs? |
| 22:59.35 | JT | no such thing as an ISDN3 |
| 22:59.43 | ZakMcRofl | what do you mean? |
| 22:59.51 | JT | i mean it does not exist |
| 22:59.56 | JT | unless it happened overnight |
| 23:00.06 | ZakMcRofl | its just a label, isn't it? |
| 23:00.17 | JT | eh? |
| 23:00.20 | ZakMcRofl | the capi.conf is mostly premade by a distribution of asterisk for fritzbox |
| 23:00.29 | ZakMcRofl | [ISDN3] ; fritzbox 7050 internal S0 |
| 23:00.44 | JT | ah, i thought you meant you had an ISDN3 service |
| 23:01.03 | ZakMcRofl | nope i dont have any ISDN service, just internal ISDN S0 bus |
| 23:01.13 | ZakMcRofl | but i recon its not that popular in the US |
| 23:01.21 | JT | ISDN2 then |
| 23:01.35 | JT | S0 implies ISDN2 |
| 23:01.59 | ZakMcRofl | ok ISDN2 then. but the question is how to i get a different "outgoing context" for each MSN on that S0 bus |
| 23:06.40 | ZakMcRofl | [TK]D-Fender any clues? |
| 23:06.51 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@216-110-94-252.static.twtelecom.net) |
| 23:07.08 | JT | ISDN is great |
| 23:07.19 | JT | i prefer the PRI variant though |
| 23:07.37 | fujin_ | indeed |
| 23:07.59 | JT | it is still ISDN though |
| 23:09.54 | endre | are there anyone using .ael intead of the good old extensions.conf? any experiences? |
| 23:10.00 | endre | instead* |
| 23:11.07 | ZakMcRofl | do you know if there's a german asterisk channel somewhere? they might be more familiar with ISDN handling |
| 23:12.28 | fujin_ | endre: yes, all of my callcentre is AEL |
| 23:12.46 | fujin_ | dynamic queue members |
| 23:12.48 | fujin_ | all kinds of magical stuff |
| 23:12.49 | craigk | does anybody know if i can use mp3 directly for music on hold ... or do i have to convert to wav and/or ulaw ? |
| 23:12.58 | fujin_ | craigk: it's smarter to convert to $CODEC |
| 23:13.07 | fujin_ | to avoid having to transcode to play it down the line |
| 23:13.07 | *** join/#asterisk file (n=file@asterisk/developer-and-muffin-lover/file) |
| 23:13.07 | *** mode/#asterisk [+o file] by ChanServ |
| 23:13.18 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 23:13.18 | fujin_ | but yes, you can use native mp3 |
| 23:13.22 | fujin_ | with asterisk-addons |
| 23:13.29 | craigk | by $CODEC ... do you mean all the codecs that i support ? |
| 23:13.38 | fujin_ | your primary outbound codec |
| 23:13.46 | craigk | kk - thank you :) |
| 23:14.45 | endre | fujin_: so it's quite usable ;) |
| 23:17.42 | *** join/#asterisk ManxPower (n=manxpowe@70.sub-70-220-221.myvzw.com) |
| 23:18.07 | JT | endre: sure, but keep in mind it's no more efficient than standard extensions.conf under the hood |
| 23:18.21 | endre | yeah i see |
| 23:18.40 | fujin_ | endre: indeed |
| 23:18.45 | fujin_ | no more efficient |
| 23:18.46 | fujin_ | saner, though :) |
| 23:18.51 | JT | .ael is generated into .conf style |
| 23:18.53 | fujin_ | for one that is familiar with c/cpp/java etc. |
| 23:19.13 | endre | fujin_: yeah, it's closer to my mind this way |
| 23:19.32 | fujin_ | As it is mine. |
| 23:19.43 | fujin_ | I was able to achieve much more complex dialplans |
| 23:19.48 | fujin_ | with AEL |
| 23:20.37 | *** part/#asterisk MrParity (n=MrParity@dslb-088-076-195-117.pools.arcor-ip.net) |
| 23:24.05 | hmmhesays | oh some days |
| 23:24.32 | hmmhesays | isn't ael just turned into a regular extensions.conf dialplan ? |
| 23:24.40 | [TK]D-Fender | yup |
| 23:25.00 | hmmhesays | then it would stand to reason you could create no more or less complex dialplans using either |
| 23:25.47 | fujin_ | hmmhesays: that's incorrect |
| 23:25.52 | [TK]D-Fender | hmmhesays, AEL can't be more efficient because there are no doubt places you could do better than it gets parsed to |
| 23:26.08 | craigk | my music on hold is really bad... keeps cutting in and out. i am using the default .wav fiels that come with asterisk - any suggestions/ideas ? |
| 23:26.17 | hmmhesays | I see |
| 23:26.18 | fujin_ | craigk: check duplex? :P |
| 23:27.09 | craigk | fujin_: thanks for the tip ... now i have to work out where to check it :) |
| 23:27.53 | fujin_ | `mii-tool` |
| 23:27.56 | fujin_ | `ethtool ethX` |
| 23:29.28 | craigk | mii-tool -> eth0: negotiated 100baseTx-FD, link ok |
| 23:29.39 | *** part/#asterisk asdx (n=diego@adsl-150-178.click.com.py) |
| 23:29.44 | craigk | ethtool eth0 -> No data available |
| 23:29.59 | fujin_ | looks fine |
| 23:30.07 | fujin_ | It's odd that the wav files are crackling. |
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| 23:30.22 | *** mode/#asterisk [+o file] by ChanServ |
| 23:30.22 | JT | craigk: do you have a zap timing source? |
| 23:30.30 | *** join/#asterisk asdx (n=diego@adsl-150-178.click.com.py) |
| 23:30.58 | craigk | JT: ah - probably not. I have a digum card installed but have removed it ... and did not explicitly put ztdummy there |
| 23:31.16 | JT | hmm |
| 23:31.19 | craigk | sorry - i _had_ a digium card installed |
| 23:31.20 | JT | may be an issue |
| 23:32.12 | craigk | kk - thanks for the tip ... right now i have to run off to the dentist. I will look at that when i get back from the drilling :/ |
| 23:44.39 | Grnd-Wire | ok, I am back from lunch - and my IAX trunks still don't work.. :P |
| 23:44.51 | mvanbaak | gheh |
| 23:45.05 | mvanbaak | you thought going out for lunch would automagically fix it ? |
| 23:45.32 | Grnd-Wire | HAHA.. yeah.. Didn't you haxx0r my machines while I was gone, and fix the problems? |
| 23:46.06 | mvanbaak | ehm, no |
| 23:50.13 | ZakMcRofl | i'm off. thanks for trying to help me - i guess |
| 23:50.17 | ZakMcRofl | cu |