| 00:04.54 | *** join/#asterisk russ (~russ@206.29.188.187) |
| 00:08.04 | *** join/#asterisk russ (~russ@206.29.188.235) |
| 00:09.59 | *** join/#asterisk sezuan (bouncer@irc.scheff32.de) |
| 00:12.21 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
| 00:26.04 | *** join/#asterisk gabdrach (~Adium@190.177.72.40) |
| 00:26.58 | gabdrach | hello everybody, i need some help. sip show channels show hundreds of lines like this: 127.0.1.1 (None) 2714721306 00101/00001 unkn No Rx: REGISTER |
| 00:29.07 | gabdrach | im running asterisk on solaris: asterisk-1.2.7.1-solvoip-143 |
| 00:34.50 | carrar | Whats in sip.conf? |
| 00:34.59 | *** join/#asterisk sezuan (bouncer@irc.scheff32.de) |
| 00:37.20 | carrar | should delete trixbox |
| 00:46.19 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 00:47.46 | gabdrach | in sip.conf i have type=friend entry |
| 00:48.11 | gabdrach | also have this on messages log: Sep 22 00:31:42 NOTICE[18267] chan_sip.c: Registration from '"gabriel" <sip:gabriel@8.17.172.134>' failed for '147.83.20.40' - Username/auth name mismatch |
| 00:48.29 | gabdrach | last message repeats and repetas |
| 00:48.55 | gabdrach | its not a trixbox, is just asterisk |
| 00:54.47 | *** join/#asterisk mac-mini (~mac-mini@unaffiliated/macmini/x-648924) |
| 00:57.30 | carrar | so block that IP |
| 00:58.28 | carrar | make sure your SIP passwords are strong |
| 00:59.23 | *** join/#asterisk boodu (~antoine@host-115-126-167-240.adsl.nautile.nc) |
| 00:59.25 | golikwid|mac | like the same as your extension |
| 00:59.32 | boodu | hello |
| 00:59.42 | golikwid|mac | you like if the extension is 200 go ahead and make that the password too |
| 00:59.47 | golikwid|mac | ;) |
| 01:00.56 | golikwid|mac | hey i have a weird hudlike-server question |
| 01:01.14 | golikwid|mac | no matter who someone calls it says they are calling the same extension |
| 01:01.19 | golikwid|mac | in this case the kitchen... |
| 01:01.22 | golikwid|mac | kinda weird |
| 01:03.25 | golikwid|mac | you know im in the asterisk room... |
| 01:03.31 | golikwid|mac | not the trixbox room lol |
| 01:03.32 | golikwid|mac | hm |
| 01:03.40 | golikwid|mac | i really should have noticed that |
| 01:03.46 | golikwid|mac | god...im such a noob |
| 01:03.51 | b11d` | aye |
| 01:08.27 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 01:25.52 | *** join/#asterisk boodu (~antoine@host-115-126-167-240.adsl.nautile.nc) |
| 01:26.40 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 01:28.50 | *** join/#asterisk bongfrog (~quassel@2001:470:f1bc:1:213:2ff:fe67:39c5) |
| 01:34.57 | *** join/#asterisk coppice (~chatzilla@m121-202-20-176.smartone-vodafone.com) |
| 01:44.35 | *** join/#asterisk GlobeTrotterz (chatzilla@pool-96-232-156-186.nycmny.fios.verizon.net) |
| 01:56.59 | *** join/#asterisk cj (~cjac@router0.colliertech.org) |
| 01:57.01 | cj | hey all |
| 01:57.17 | *** join/#asterisk brunner (~chris@35-137.175-24.bham.res.rr.com) |
| 01:57.23 | brunner | does anyone here have a Gizmo5 account? |
| 01:57.32 | cj | I just received certification from the state utilities commission as a CLEC |
| 01:58.04 | brunner | cj, congrats! What state? |
| 01:58.21 | cj | My primary goal is to get access to the copper loop for DSL purposes, but it would also be nice to provide PSTN voice services, too ;) |
| 01:58.24 | cj | brunner: WA |
| 01:59.08 | cj | what do I need to know in order to give customers dial tones and route calls? I'd probably start out with local-only service. Specifically, nothing off of the island. |
| 01:59.28 | cj | http://en.wikipedia.org/wiki/Orcas_Island |
| 01:59.57 | *** part/#asterisk gabdrach (~Adium@190.177.72.40) |
| 02:00.47 | WIMPy | Compare IADs. |
| 02:03.28 | *** join/#asterisk CharlieBoisseau (~Adium@204.153.192.4) |
| 02:05.22 | underdog | cj: congrats....and prepare for A LOT of headaches working with the LECs |
| 02:07.22 | cj | underdog: I am prepared. I've spent the last few weeks reviewing 480-120 WAC and friends |
| 02:07.43 | underdog | cj: technically you can be a CLEC and not own any equipment....you are just the middle-man "resaling" telco services provided by the LEC |
| 02:08.21 | cj | *shrug* I like avoiding middle-men and keeping the MTU high ;) |
| 02:08.46 | jarrod | cj how many co's in the area you want to service? |
| 02:08.47 | underdog | had to deal with SWBell working for a CLEC...their customer service treated our customers badly...because they knew they weren't "their" customers |
| 02:10.04 | *** part/#asterisk CharlieBoisseau (~Adium@204.153.192.4) |
| 02:13.47 | cj | jarrod: there's only one on the island ;) |
| 02:15.00 | cj | jarrod: when I get off the island, I'll want to service Bellingham, Everett and Seattle. |
| 02:15.07 | cj | but that's probably a couple years down the road |
| 02:15.58 | *** join/#asterisk OrNix (~ornix@l49-0-149.cn.ru) |
| 02:20.49 | Dougy | weird |
| 02:20.58 | Dougy | my outgoing phone forwarding randomly stopped workign |
| 02:20.59 | Dougy | lame |
| 02:21.17 | underdog | random feature |
| 02:21.42 | Dougy | <PROTECTED> |
| 02:21.45 | Dougy | it sends the call properly |
| 02:21.48 | Dougy | but it doesnt connect |
| 02:21.49 | Dougy | hrmm |
| 02:23.38 | Dougy | i suspect a firewall |
| 02:23.39 | Dougy | tsk tsk |
| 02:30.47 | ChannelZ | I suspect squirrels |
| 02:31.50 | underdog | squirrel season starts in october |
| 02:31.56 | underdog | heh |
| 02:33.14 | ChannelZ | stockpiles some ammo |
| 02:35.50 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 02:38.15 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 02:51.44 | *** join/#asterisk voxter (~voxter@76.77.73.130) |
| 02:52.15 | voxter | Hello fellow * geeks. I'm hoping you can point me in the most favorable implementation of call limiting on sip peers in asterisk 1.4. |
| 02:52.49 | voxter | I have been using call-limit, but it has its own gotchas... Mainly you need to take into account call waiting scenarios or 3 way calling scenarios |
| 02:53.46 | voxter | Also, If im not mistaken, there are other effects where it may consider a call active during an OPTIONS or NOTICE packet? |
| 03:02.30 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 03:05.05 | b11d` | welp.. shoulda known.. |
| 03:05.12 | b11d` | Asterisk w/ libpri & dahdi works great in linux ;) |
| 03:05.16 | b11d` | my card works flawlessly |
| 03:05.29 | b11d` | sigh.. i may just convert away from FreeBSD :( |
| 03:06.08 | drmessano | It's about time.. Nobody uses BSD anymore |
| 03:06.16 | drmessano | Except that guy in the basement with all the cats |
| 03:06.26 | underdog | jarrod: ^ |
| 03:06.33 | underdog | oh wait...he has dogs |
| 03:06.37 | underdog | heh |
| 03:06.50 | b11d` | damnit.. i love FreeBSD though |
| 03:06.52 | b11d` | i LOVE it |
| 03:07.05 | drmessano | Who do you love more.. people or your cats |
| 03:07.09 | b11d` | but.. gotta use what works. |
| 03:07.11 | drmessano | DECISION TIME |
| 03:07.18 | b11d` | depends on the people :) |
| 03:07.52 | underdog | use debian...it progresses as fast as BSD |
| 03:08.02 | drmessano | Think of it this way |
| 03:08.12 | b11d` | i am using Debian |
| 03:08.12 | b11d` | lol |
| 03:08.20 | drmessano | BSD has jails. There are no women in jail. Coincidence? |
| 03:08.32 | *** join/#asterisk MrJackson (Mr@173-86-54-137.dr01.wlbr.pa.frontiernet.net) |
| 03:08.39 | b11d` | oh.. dont be so quick to think so.. there are girls in jail :) |
| 03:08.54 | b11d` | you just need to use a little imagination |
| 03:09.00 | b11d` | plus, female guards.. |
| 03:09.02 | b11d` | didnt you watch Oz? |
| 03:09.13 | drmessano | No, I was busy learning FreeBSD |
| 03:09.28 | drmessano | and collecting cats |
| 03:09.32 | b11d` | :( |
| 03:09.37 | b11d` | :P |
| 03:09.38 | drmessano | I know, right |
| 03:09.45 | drmessano | :`( |
| 03:09.59 | b11d` | i really wish this had worked out for me in FreeBSD.. but i cant burn any more time.. gotta have this system in place. |
| 03:11.17 | Dougy | night fellas |
| 03:14.25 | drmessano | I hear ya |
| 03:14.54 | carrar | gotta have iti n 30 mins or less? |
| 03:15.02 | carrar | I hear ya |
| 03:15.03 | carrar | too |
| 03:15.06 | drmessano | All jokes aside, I know it's either not an easy thing, or doesn't work at all.. I'm not much versed on FreeBSD, but I know it comes up from time to time and thats the sticker |
| 03:15.23 | drmessano | Dahdi that is |
| 03:15.45 | drmessano | Asterisk seems to run great on it, but Zaptel, and now DAHDI, seemed to be a problem |
| 03:16.11 | drmessano | May be something with oskernel64.dll or some shit |
| 03:16.13 | drmessano | Dunno |
| 03:17.22 | drmessano | freebsd.exe has segfaulted. Invalid function in dahdi.dll? Dunno.. All GREEK TO ME |
| 03:17.59 | boodu | I have a problem with the option m(musicClass) of Dial. It's work fine on internal but don't pass over my misdn BRI Card |
| 03:18.44 | boodu | how fix that please ? |
| 03:21.05 | carrar | You need the freebsd.dll |
| 03:21.12 | carrar | in order for freebsd.exe to work |
| 03:21.32 | boodu | may be few options in misdn.conf but what |
| 03:21.36 | carrar | Assuming your calling it from freebsd.bat |
| 03:21.40 | drmessano | carrar, I thought that was included in the .UNIX Framework 3.5 SP2.1 SR-5 |
| 03:22.05 | carrar | It's a add on patch |
| 03:22.12 | *** join/#asterisk radic (~radic@dslb-094-216-254-101.pools.arcor-ip.net) |
| 03:22.29 | drmessano | You know what I love about this channel being logged and then indexed by google |
| 03:22.43 | drmessano | "Windows XP Service Pack 4" |
| 03:23.03 | drmessano | That'll be around for a while |
| 03:23.03 | underdog | dll's in bsd are a pain to get working |
| 03:23.26 | underdog | gcc freebsd.dll -o freebsd.so |
| 03:23.29 | carrar | yeah no ld.so.conf |
| 03:23.38 | russellb | o.O |
| 03:23.48 | carrar | well it does kinda sorta |
| 03:23.52 | drmessano | There he is |
| 03:24.12 | drmessano | russellb is FreeBSD guru.. he loves it so much, his toilet runs on it |
| 03:24.21 | underdog | you have to compile it on linux and copy it over to bsd |
| 03:24.27 | carrar | makes for great toilet UPTIME |
| 03:24.32 | underdog | it's the compiler trifecta |
| 03:24.43 | carrar | having 5 9's for a toilet is important |
| 03:25.10 | drmessano | carrar: You don't want to be that .00001th flush, you know |
| 03:25.18 | carrar | I KNOW |
| 03:25.28 | carrar | You get a overrun |
| 03:25.32 | carrar | not a good thing |
| 03:25.42 | underdog | buffer overflow vuln |
| 03:25.44 | drmessano | Yank the handle........ *crickets* |
| 03:25.53 | drmessano | "NOO, PLS FLUSH" |
| 03:25.58 | drmessano | *crickets* |
| 03:26.01 | drmessano | reboot! |
| 03:26.25 | carrar | being able to process large amounts of data is critical |
| 03:26.43 | carrar | in a TIMELY fashion |
| 03:27.16 | underdog | is waiting for the 32bit, 64bit joke |
| 03:29.15 | drmessano | Windows is a 64-bit operating system, emulating a 32-bit operating system, that won't run 16-bit code, that's as slow as an 8-bit Nintendo, with a GUI inspired by 4-bit graphics, written by a bunch of 2-bit con men? |
| 03:29.34 | drmessano | Sorry, it needed a refresh |
| 03:29.52 | underdog | heh |
| 03:30.50 | carrar | Thats right!!! <meta name="keywords" content="poop,windows,32bit"> |
| 03:31.07 | drmessano | Could be worse.. BSD is a 64-bit operating system run by 32 people. I got nothing else. |
| 03:33.21 | carrar | helping good with searching |
| 03:33.24 | carrar | err google |
| 03:39.33 | drmessano | "Asterisk 1.10" |
| 03:39.40 | drmessano | On that note, night |
| 03:59.52 | *** join/#asterisk WindBack (~quassel@200-122-74-15.cab.prima.net.ar) |
| 04:00.57 | WindBack | Somebody can tell me how I can make asterisk register in a ericsson pbx trough h323? |
| 04:01.18 | WindBack | is there any special configuration in h323.conf |
| 04:01.20 | WindBack | ? |
| 04:04.44 | *** join/#asterisk p3nguin_ (~xwQ5kwYl6@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 04:04.46 | *** join/#asterisk pinoyskull (~pinoyskul@122.55.80.205) |
| 04:10.25 | ChannelZ | OK what would be causing files on my filesystem to make their creation dates look like they've changed (but only by small amounts, like +/- 15 seconds) even though I'm positive the files have not been touched? |
| 04:32.45 | *** join/#asterisk shamelessn00b (~chatzilla@58-65-172-114.nayatel.pk) |
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| 05:49.28 | *** join/#asterisk schmidts (~schmidts@lmlo.sil.at) |
| 05:49.30 | schmidts | good mornging |
| 06:18.50 | *** join/#asterisk deonv (~adium@pixfirewall.itn.com.na) |
| 06:22.22 | *** join/#asterisk timahvo1 (~rogue@41.191.224.178) |
| 06:26.34 | shamelessn00b | hello |
| 06:28.38 | *** join/#asterisk timahvo1 (~rogue@41.191.224.178) |
| 06:35.04 | *** join/#asterisk mpe (~mpe@109.70.54.7) |
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| 06:57.35 | ChannelZ | ohell |
| 06:57.53 | schmidts | o hell? is it that bad? |
| 06:58.14 | ChannelZ | it's the oppose of hello |
| 07:00.49 | schmidts | ;) |
| 07:02.13 | *** join/#asterisk c0rnoTa (~c0rnoTa@109.188.28.120) |
| 07:02.17 | *** part/#asterisk c0rnoTa (~c0rnoTa@109.188.28.120) |
| 07:03.46 | *** join/#asterisk waschtl (~waschtl@3ed8a58a.d.d9tcloud.de) |
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| 07:09.24 | *** join/#asterisk Russ (foobar@ip70-176-251-1.ph.ph.cox.net) |
| 07:10.42 | ChannelZ | ARGH |
| 07:12.00 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 07:14.23 | Russ | ChannelZ, relax |
| 07:14.33 | Russ | ChannelZ, maybe pretend they are rocks? |
| 07:17.33 | *** join/#asterisk sgimeno (~chatzilla@163.117.211.10) |
| 07:17.47 | ChannelZ | who? |
| 07:18.09 | Russ | I don't know, it make sense at 4am yesterday |
| 07:21.12 | boodu | bye |
| 07:21.22 | *** join/#asterisk wierdo (jimmy@77.78.3.197) |
| 07:23.19 | schmidts | maybe everything make sense on 4 am ;) |
| 07:28.32 | ChannelZ | I'm ARGHing on a wierd timestamp problem I can't find an explanation for |
| 07:33.23 | *** join/#asterisk inovaspa (~lorenzo@151.8.202.59) |
| 07:33.27 | schmidts | time is relativ :D |
| 07:33.58 | ChannelZ | it appears so |
| 07:42.37 | *** join/#asterisk Intel`` (~clarencec@94.200.7.26) |
| 07:42.47 | Intel`` | gusy how can i edit the voicemail filename? |
| 07:43.03 | Intel`` | also the recording filename? |
| 07:46.19 | ectospasm | Intel``: those are two different things |
| 07:46.37 | ectospasm | Intel``: why do you need to edit the voicemail filename? |
| 07:47.04 | Intel`` | ectospasm: sorry i looked into /var/spool/asterisk/monitor/voicemail/default |
| 07:47.14 | Intel`` | i can see its already been organized by folders |
| 07:47.27 | Intel`` | the only thing is the monitor folder |
| 07:47.47 | ectospasm | Intel``: that's set in /etc/asterisk/voicemail.conf |
| 07:48.00 | Intel`` | there are file that has: |
| 07:48.01 | Intel`` | #!/bin/bash |
| 07:48.03 | Intel`` | ''shopt -s extglob'' |
| 07:48.04 | Intel`` | #$m=`find /var/spool/asterisk/monitor -name *.gsm` |
| 07:48.06 | Intel`` | #$l=`find /var/spool/asterisk/monitor -name *.gsm` |
| 07:48.07 | Intel`` | #if ($m=0 && $l=0) |
| 07:48.09 | Intel`` | #echo nothing to do; |
| 07:48.10 | Intel`` | #else |
| 07:48.12 | Intel`` | for f in @(IN|OUT)+([[:digit:]])-*; do n=${f#@(IN|OUT)} n=${n%-*}; mkdir -p "$n" && mv "$f" "$n"; done |
| 07:48.13 | Intel`` | #fi |
| 07:48.15 | Intel`` | ##mkdir "$n" && |
| 07:48.16 | Intel`` | sooorryy |
| 07:48.20 | ectospasm | don't flood |
| 07:48.34 | Intel`` | sorry i pasted the wrong text |
| 07:48.43 | ectospasm | use pastebin |
| 07:48.46 | Intel`` | some are like this: OUT3162-20100922-105152-1285138312.14835.gsm |
| 07:48.48 | *** join/#asterisk waschtl (~waschtl@3ed8a58a.d.d9tcloud.de) |
| 07:49.06 | Intel`` | but some are this: 20100914-062857-1284431321.4493.gsm |
| 07:49.47 | Intel`` | i have a script that organize the files by folder but my reference is the OUT<extension> |
| 07:50.00 | Intel`` | but the second filename doenst have this |
| 07:52.04 | ectospasm | are you recording any files with Monitor/MixMonitor? |
| 07:55.36 | Intel`` | yes outbound calls |
| 07:55.53 | Intel`` | actually im using asterisknow with freepbx |
| 07:56.16 | Intel`` | and we set outbound calls to record |
| 07:58.26 | ectospasm | those are your Monitor files |
| 07:58.35 | ectospasm | Monitor records two files, IN and OUT |
| 07:58.46 | ectospasm | ...not sure, but your IN may not be labeled IN |
| 07:59.08 | ectospasm | (or they may be labeled rx/tx) |
| 07:59.19 | ectospasm | it depends on how you call Monitor in your dialplan |
| 08:02.26 | Intel`` | i have also IN<Extension> |
| 08:02.33 | Intel`` | but weird i get this files |
| 08:03.39 | *** join/#asterisk timahvo1 (~rogue@41.191.224.178) |
| 08:04.41 | Intel`` | ectospasm: sorry it seems these are IN files |
| 08:04.47 | Intel`` | i was looking at my other asterisk server |
| 08:05.27 | hrhrhr | gooooooooood mooorrrnnniiiiinnnnnn chaps |
| 08:06.56 | Intel`` | morning hrhrhr |
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| 08:45.08 | metiu_ | is it possible to use asterisk as a public address system? I'd need it to be "serverless" (no single point of failure) and possibly with multicast audio. I'm cooking an in-house software, but I'd rather use proven solutions |
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| 09:02.51 | schmidts | metiu_ what do you want to do? i didnt understand what you mean with public address system |
| 09:03.17 | Intel`` | you mean something like "Pager"? |
| 09:04.13 | *** join/#asterisk Justman (~just@78.108.73.46) |
| 09:07.32 | metiu_ | I have a system where I could need to speak from a headset to a series of loudspeakers, or from a headset to another headset |
| 09:08.43 | metiu_ | the sound should go to the speakers in multicast to save bandwidth |
| 09:09.00 | metiu_ | the system is modular, and the network must be robust, so no single server |
| 09:09.14 | metiu_ | (which would have been way easier) |
| 09:25.34 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
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| 09:32.10 | hrhrhr | anyone got nagios/asterisk integration on 1.6 up and running? |
| 09:36.00 | EmleyMoor | Is there any reason against running a Jabber server on the same hardware as Asterisk? Also, will exporting spare disk space over NFS and/or samba affect performance? |
| 09:36.53 | *** part/#asterisk eLBati (~elbati@93.37.64.171) |
| 09:39.43 | schmidts | emleymoor jabber shouldnt be that impact on a system and nfs could cause some special issues, specially when you record voicemails ( see asterisk -t option for this) |
| 09:39.56 | schmidts | if there is enough bandwith it shouldnt be a problem |
| 09:40.31 | EmleyMoor | schmidts: Thanks - am seriously thinking of merging my Jabber server with my new Asterisk box |
| 09:42.26 | schmidts | i really dont know why this could be a problem, depends how many jabber user and asterisk peers you have |
| 09:43.12 | EmleyMoor | ID can do without exporting spare space, but there's going to be loads of spare space doing nothing |
| 09:44.20 | Russ | EmleyMoor, my concern would be security |
| 09:44.29 | EmleyMoor | Jabber users: 2 + Asterisk itself |
| 09:44.51 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 09:45.13 | Russ | course, 1 machine has half as many hardware failures as 2 machine |
| 09:46.49 | EmleyMoor | I'm hoping I don't have to patch asterisk and DAHDI myself on the new install - at least one of the patches I was using has gone into the main... |
| 09:46.55 | Russ | 'ìê·ëª¨ ìì¹´ìì ë°´ë - 모르ë ì¼ë¤' |
| 09:47.33 | Russ | EmleyMoor, its always nice to refresh your install skills for when the box takes a nosedive |
| 09:48.08 | EmleyMoor | I am switching to amd64 this time |
| 09:49.13 | ectospasm | Russ: yes, but one machine *potentially* costs less than two |
| 09:50.08 | *** join/#asterisk Carmageddon (~Bauer@109.65.184.134) |
| 09:51.50 | Russ | sometimes I like having a hot-ready install of asterisk on a machine that I use for some other function, so if the asterisk box dies, it can temporarily take over |
| 09:53.07 | EmleyMoor | Russ: If my asterisk setup was mission critical, I'd do the same |
| 09:54.17 | EmleyMoor | is probably ordering his AEX card today |
| 09:55.35 | Russ | mine is only mission critical because my I'll be sleeping on the couch if the phones don't work |
| 09:56.32 | EmleyMoor | If ours go down we just route them to mobiles from the ITSP end - did that in the planned power cut last spring |
| 09:58.24 | ectospasm | some yahoos say,"So? We'll just be without phones for *half a day (or more)*" |
| 09:58.45 | ectospasm | ...and then he says his job may be in jeopardy (-; |
| 10:01.43 | *** join/#asterisk mpe (~mpe@109.70.54.7) |
| 10:18.02 | shamelessn00b | anyone knows what filters does Denoise employ to reduce background noise, and how much hardware resources does it consume |
| 10:18.48 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
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| 11:43.11 | *** join/#asterisk asilva (~structz@gandalf.ai.unesp.br) |
| 11:43.31 | asilva | Anyone could advise what to do when this message occur > [Sep 22 08:37:25] WARNING[2508] channel.c: Exceptionally long voice queue length queuing to IAX2/contadundi-747 |
| 11:43.31 | asilva | [Sep 22 08:37:26] NOTICE[21532] chan_iax2.c: I should never be called! |
| 11:44.12 | asilva | ast ver 1.4.36 on debian 5 lenny 32bit |
| 11:48.04 | *** join/#asterisk leif[mobile] (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 11:48.04 | *** mode/#asterisk [+o leif[mobile]] by ChanServ |
| 12:00.58 | *** join/#asterisk haroon1 (~chatzilla@58-65-172-114.nayatel.pk) |
| 12:01.10 | haroon1 | hi everybody |
| 12:02.16 | fauxalliance | hello haroon1 |
| 12:02.29 | drmessano | <PROTECTED> |
| 12:04.11 | haroon1 | i am facing a strange behavior on asterisk 1.6 |
| 12:04.22 | haroon1 | it restarts after sometime |
| 12:05.31 | haroon1 | kern log show this error |
| 12:05.32 | haroon1 | Sep 22 15:30:10 a1 kernel: [12352014.721002] asterisk[15035]: segfault at 0 ip 7feb5405df62 sp 45254a88 error 4 in libc-2.7.so[7feb53fe1000+14a000] |
| 12:05.54 | haroon1 | asterisk log does not show anything |
| 12:06.26 | haroon1 | is there anything i can do about it, any help is appreciated |
| 12:07.00 | fauxalliance | haroon1, https://issues.asterisk.org/view.php?id=11486 |
| 12:07.32 | fauxalliance | perhaps you should google [asterisk libc segfault] haroon1 |
| 12:08.05 | fauxalliance | http://webcache.googleusercontent.com/search?q=cache:MdLTYKEawd0J:https://issues.asterisk.org/view.php%3Fid%3D16365+asterisk+segfault+libc&cd=2&hl=en&ct=clnk&client=iceweasel-a |
| 12:08.13 | haroon1 | fauxlliance: alright |
| 12:08.28 | fauxalliance | haroon1, indeed |
| 12:14.37 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 12:15.23 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 12:15.39 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 12:16.23 | asilva | Does anyone indicate some call route discovery for asterisk(something like DUNDi)? |
| 12:17.10 | Naikrovek | man this sucks. a 1.58kg tin of mixed nuts and already someone has eaten almost all of the cashews |
| 12:17.15 | Naikrovek | grumbles |
| 12:17.33 | Naikrovek | I WAS GOING TO EAT ALMOST ALL THE CASHEWS! |
| 12:17.55 | Naikrovek | guess i'll have to go buy a jar of cashews |
| 12:18.48 | Naikrovek | wonder how many more sentences I can put the word cashews into? |
| 12:19.52 | *** join/#asterisk Nwab (~Benwa@unaffiliated/benwa) |
| 12:20.33 | Jasnejac | Naikrovek: depends on how many sheep you wish to buy I guess |
| 12:21.32 | [TK]D-Fender | Naikrovek: cashew? |
| 12:21.36 | [TK]D-Fender | Naikrovek: Gesundheit! |
| 12:23.38 | *** join/#asterisk Shadow_aok (~Shadok@tok69-2-82-224-121-185.fbx.proxad.net) |
| 12:23.40 | Shadow_aok | hello |
| 12:23.54 | Naikrovek | thanks d-fender |
| 12:23.59 | Naikrovek | caught a sniffle |
| 12:24.09 | Shadow_aok | Is there a way, in an extension, to target all queues but one ? |
| 12:24.18 | Naikrovek | BUT THIS IS UNRELATED TO THE LACK OF CASHEWS IN THIS GIANT CAN OF NUTS |
| 12:24.29 | Naikrovek | Shadow_aok: yes of course |
| 12:24.39 | Naikrovek | just dial them all but one |
| 12:24.43 | Shadow_aok | How do you do that ? |
| 12:24.58 | Naikrovek | Dial(SIP/queue1&SIP/queue2&SIP/queue3...) |
| 12:25.02 | drmessano | Did they leave the Macadamia nuts? |
| 12:25.03 | Shadow_aok | I tried to detect answering machines and it works well, by using that |
| 12:25.15 | Shadow_aok | [ext-queues-custom] |
| 12:25.15 | Shadow_aok | exten => _X.,1,Answer |
| 12:25.16 | Shadow_aok | exten => _X.,n,Set(cola=${EXTEN}) |
| 12:25.16 | Shadow_aok | exten => _X.,n,BackgroundDetect(silence/5, 1000, 50) |
| 12:25.16 | Shadow_aok | exten => _X.,n,Hangup |
| 12:25.16 | Shadow_aok | exten => talk,1,Goto(ext-queues,${cola},1) |
| 12:25.16 | Shadow_aok | exten => talk,n,Hangup |
| 12:25.18 | Shadow_aok | but |
| 12:25.19 | drmessano | ACK |
| 12:25.28 | drmessano | DONT PASTE IN HERE OR WE KILL YOU |
| 12:25.32 | Naikrovek | lol |
| 12:25.40 | Shadow_aok | forgot pastebin sorry |
| 12:25.45 | Shadow_aok | well |
| 12:25.53 | fauxalliance | drmessano, perhaps we could tattoo it somewhere |
| 12:26.05 | drmessano | On his corpse, yes |
| 12:26.08 | Russ | what the hell was that comic called? |
| 12:26.08 | Shadow_aok | i have a queue for incoming call (number 800) |
| 12:26.10 | Shadow_aok | mmh |
| 12:26.17 | Shadow_aok | i vote for the motd |
| 12:26.27 | Naikrovek | Shadow_aok: continue with your question |
| 12:26.35 | Naikrovek | we're punchy in the mornings just ignore it and continue |
| 12:26.36 | Shadow_aok | so i'm looking to target all queues except for the incoming one |
| 12:26.36 | Naikrovek | :) |
| 12:26.40 | Shadow_aok | :) |
| 12:26.43 | fauxalliance | Shadow_aok, common sense is not so common I guess... |
| 12:26.45 | Naikrovek | are the queues static |
| 12:26.47 | drmessano | Didn't he just answer it? |
| 12:26.57 | Shadow_aok | well, i'm not used to asterisk, i'm not the one managing it usually |
| 12:27.01 | Naikrovek | hey wait i did just answer it |
| 12:27.12 | Shadow_aok | so sorry, but i'm not sure i understood it |
| 12:27.13 | drmessano | Yes, you did.. and he pasted anyway |
| 12:27.20 | Naikrovek | okay |
| 12:27.22 | Naikrovek | well |
| 12:27.35 | Naikrovek | you have to add an extension to the dialplan that will ring all queues except the incoming queue |
| 12:27.39 | Naikrovek | so pick an extension |
| 12:27.44 | Naikrovek | call it 888 |
| 12:27.52 | Naikrovek | in the dialplan you'll need to add something like this |
| 12:27.57 | fauxalliance | (which is zap barge on some systems) |
| 12:28.02 | Naikrovek | ah true |
| 12:28.10 | Naikrovek | pick a free extension number |
| 12:28.16 | Naikrovek | 888 was bad example |
| 12:28.19 | Naikrovek | pick one that's free |
| 12:28.20 | drmessano | 6969 |
| 12:28.23 | drmessano | 269 |
| 12:28.27 | drmessano | 477 |
| 12:28.33 | Naikrovek | ^^^ |
| 12:28.45 | drmessano | or my fav, 11 |
| 12:28.50 | Naikrovek | then, you'll need to add a line to the dialplan so that extension rings all the queues |
| 12:28.52 | fauxalliance | 7222222.... local cabie... ad reads... 'even drunk girls can remember two numbers' |
| 12:28.55 | drmessano | She'll never know what hit her |
| 12:28.58 | Naikrovek | fauxalliance: lol |
| 12:29.06 | drmessano | hahah |
| 12:29.18 | Naikrovek | so you'll need to add a line to the dialplan that looks like this |
| 12:29.30 | Shadow_aok | ok so i add an extension using the Dial command and targetting my queues except the incoming one |
| 12:29.46 | Naikrovek | exten => 477,1,Dial(SIP/queue1&SIP/queue2&SIP/queue3) and so on |
| 12:29.47 | Naikrovek | then |
| 12:29.50 | Naikrovek | when you dial 477 |
| 12:29.53 | Shadow_aok | and in my avoid-answering-machines routine, i only target this new extension |
| 12:30.04 | Naikrovek | call will go to all queues specified |
| 12:30.16 | Naikrovek | yep |
| 12:30.43 | Naikrovek | replace queue1 queue2 queue3 with your actual queue extension numbers |
| 12:30.51 | Naikrovek | whatever one would dial to go to those queueus |
| 12:30.55 | Naikrovek | lol |
| 12:30.56 | Naikrovek | queueues |
| 12:31.04 | Naikrovek | i know how to spell it i just don't know when to stop |
| 12:31.14 | Shadow_aok | ok |
| 12:31.21 | drmessano | fauxalliance: http://www.facebook.com/group.php?gid=6191254679&v=info |
| 12:31.23 | Shadow_aok | one more question |
| 12:31.27 | drmessano | ^^ Jiffy Cabs |
| 12:31.28 | Shadow_aok | I used [ext-queues-custom] |
| 12:31.35 | Shadow_aok | can i use [ext-queues-custom-2] and so one ? |
| 12:31.43 | Naikrovek | i don't know contexts very well |
| 12:31.49 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 12:31.53 | Naikrovek | i have yet to set up a vanilla asterisk phone system :) |
| 12:31.58 | fauxalliance | drmessano, "On de way" |
| 12:32.17 | Naikrovek | "ON TEH WAYYYYYY" rofl |
| 12:32.23 | Naikrovek | their typo, not mine |
| 12:32.44 | asilva | <PROTECTED> |
| 12:32.45 | asilva | 24955 root 20 0 43672 18m 7544 S 221 0.9 9:23.04 asterisk |
| 12:32.47 | asilva | what o do ? |
| 12:32.49 | fauxalliance | thays the way their dispatcher sounds... |
| 12:32.51 | asilva | what to* |
| 12:32.52 | asilva | oh sorry |
| 12:32.55 | Shadow_aok | well, i think a long i include it in extensions_additional, this should work |
| 12:33.05 | Naikrovek | Shadow_aok: i think so too |
| 12:33.06 | fauxalliance | asilva, stop pasting. get more cores |
| 12:33.13 | Naikrovek | lol more cores |
| 12:33.25 | Naikrovek | asilva: how many calls going on right now |
| 12:33.29 | asilva | fauxalliance, alreaady have 4 |
| 12:33.36 | Shadow_aok | so anyway, how can i change a queue number ? |
| 12:33.37 | fauxalliance | asilva, how many is MORE |
| 12:33.43 | drmessano | asilva: Were you NOT PAYING ATTENTION when we BLASTED that other dude about PASTING? |
| 12:33.59 | fauxalliance | threatened bodily harm even. |
| 12:34.01 | drmessano | ~cluebat |
| 12:34.01 | infobot | *WHACK* *WHACK* *WHACK* |
| 12:34.25 | [TK]D-Fender | drmessano: It was 2 lines... |
| 12:34.28 | [TK]D-Fender | drmessano: chill |
| 12:34.44 | Naikrovek | [TK]D-Fender: did you paste that response? if so... |
| 12:34.56 | fauxalliance | ...we're punchy in the mornings just ignore it and continue... |
| 12:35.00 | Russ | HOORAY FOR MR NUTTY!! |
| 12:35.02 | Naikrovek | ^^^^ |
| 12:35.20 | drmessano | [TK]D-Fender, 2 lines is the gateway to a 25MB log file |
| 12:35.23 | [TK]D-Fender | Naikrovek: It's been a long time since I last send someone skidding outta here ;) |
| 12:35.33 | Naikrovek | lol |
| 12:35.35 | Naikrovek | skidding |
| 12:35.40 | Naikrovek | jesus i'm in a good mood today |
| 12:35.49 | Naikrovek | let's see how long after someone else comes in to the office that that's ruined |
| 12:35.53 | fauxalliance | all the cashews? |
| 12:36.00 | Naikrovek | CASHEWS! |
| 12:36.21 | Naikrovek | if you want all the cashews, buy a jar of cashews. this is probably what I should do |
| 12:36.30 | [TK]D-Fender | drmessano: Oh yeah? And you're still breathing. And you know who else breathed? HITLER. There, everyone suspected, but I said it :p |
| 12:36.41 | Naikrovek | HAHA okay Mr. Beck |
| 12:36.43 | bougyman | conversation ender. |
| 12:36.46 | [TK]D-Fender | channels some more Glen Beck |
| 12:36.55 | Naikrovek | god that man is insane |
| 12:37.07 | bougyman | or a money-making genius. |
| 12:37.10 | Russ | IT EVEN COMES WITH SIDE-MOUNTED COOLING VENTS!! |
| 12:37.14 | [TK]D-Fender | Naikrovek: The term is "bat-shit crazy" IIRC |
| 12:37.15 | Naikrovek | [TK]D-Fender: throw some tears in and declare love for Canada |
| 12:37.29 | [TK]D-Fender | just loves his country... |
| 12:37.44 | [TK]D-Fender | WHERE'S MY FUCKING EMMY?! |
| 12:37.48 | Naikrovek | hands [TK]D-Fender a tissue to wipe the patriotism dripping from his eyes. |
| 12:38.18 | fauxalliance | all right [TK]D-Fender, you are starting to sound like Susan Lucci |
| 12:38.33 | fauxalliance | "that emmy is mine" jonesing.... |
| 12:38.34 | [TK]D-Fender | fauxalliance: tHAT WAS sTEPHEN cOLBERT ACTUALLY |
| 12:38.39 | [TK]D-Fender | dangnammit |
| 12:38.47 | Naikrovek | lol |
| 12:39.28 | fauxalliance | http://fliiby.com/file/147891/3dy3jswciz.html it was alan t actually |
| 12:40.29 | fauxalliance | http://www.youtube.com/watch?v=0PT9a5EaDEk rather |
| 12:55.54 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-tqxpfmtaazzwqlfv) |
| 12:56.15 | *** join/#asterisk GlobeTrotterz (chatzilla@pool-96-232-156-186.nycmny.fios.verizon.net) |
| 13:02.04 | *** join/#asterisk oej_ (~olle@static-213-115-251-100.sme.bredbandsbolaget.se) |
| 13:04.12 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 13:08.26 | Nwab | hi, where can i find a simple tuorial just for the bases (sip.conf, extensions.conf, voicemail.conf) ? |
| 13:10.17 | *** join/#asterisk suprstar (~suprstar@216.54.131.253) |
| 13:10.33 | [TK]D-Fender | ~bok |
| 13:10.37 | [TK]D-Fender | ~book |
| 13:10.37 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 13:10.41 | [TK]D-Fender | Nwab: ^^^^^ |
| 13:12.03 | Katty | drags in |
| 13:13.58 | Nwab | [TK]D-Fender, noooo, i said a SIMPLE one :) not a 5000 pages tutorial ... |
| 13:14.46 | beek | hugs Katty, waves to 'Fender |
| 13:15.17 | [TK]D-Fender | Nwab: the book is a fast read if you skim past the parts you don't need |
| 13:16.11 | Katty | hugs beek |
| 13:16.14 | *** join/#asterisk gradgrind (~MichelRP@2001:470:c10d::feed) |
| 13:23.33 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 13:28.17 | Naikrovek | wow reddit is a bunch of pyros: http://www.reddit.com/r/pics/comments/dhas7/anyone_else_loved_these_as_a_child/ |
| 13:32.26 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 13:32.31 | *** part/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 13:33.31 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 13:33.54 | *** part/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 13:35.22 | *** join/#asterisk [Outcast] (~anonymous@64.202.62.5) |
| 13:40.28 | lirakis | is away: rebooting |
| 13:42.11 | *** join/#asterisk UQlev (~Yuriy@212.50.99.8) |
| 13:42.48 | *** join/#asterisk jonmasters (~jcm@dallas.jonmasters.org) |
| 13:43.10 | lirakis | is back (gone 00:00:07) |
| 13:43.39 | drmessano | You rebooted? |
| 13:43.42 | drmessano | Did you pee too? |
| 13:48.51 | *** join/#asterisk [cannibalera] (~cannibale@201-35-231-156.fnsce703.dsl.brasiltelecom.net.br) |
| 13:49.35 | *** join/#asterisk krash812 (~roni@190.196.71.206) |
| 13:50.26 | krash812 | hi everybody .. here i go again .. can somebody giveme a help with ivr ? i got this |
| 13:50.27 | krash812 | exten => s,1,Set(TIMEOUT(digit)=3) |
| 13:50.27 | krash812 | exten => s,2,Set(TIMEOUT(response)=9) |
| 13:50.27 | krash812 | exten => s,3,BackGround(hosting) |
| 13:50.27 | krash812 | exten => s,4,WaitExten(2) |
| 13:50.27 | krash812 | exten => 1,1,Dial(SIP/1300,20,r) |
| 13:50.34 | Naikrovek | pasting... |
| 13:50.35 | Naikrovek | is a |
| 13:50.38 | Naikrovek | NO NO |
| 13:50.39 | Naikrovek | ~pb |
| 13:50.40 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood 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://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 13:51.01 | krash812 | sorry |
| 13:51.32 | krash812 | i just need to know how to make , if nobody answer the extensions go to another record .. |
| 13:51.33 | *** join/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 13:51.43 | krash812 | something like "agents are busy now" |
| 13:51.53 | *** join/#asterisk frigidzephyr (~rnewton@nat/digium/x-yodsdvqsugdjnigb) |
| 13:52.39 | [TK]D-Fender | krash812: exten => 1,2,Playback(soundfilethatsayswhatyouwant) |
| 13:52.50 | krash812 | thanks |
| 13:54.19 | ruben23 | hi guys, why do asterisk carshed...i get asterisk 3-4 crashes a day..how do prevent this.. |
| 13:54.38 | fauxalliance | ruben23, logs... |
| 13:54.53 | fauxalliance | show us the verbose logging output... via pastebin... please and thanks... |
| 13:55.37 | pabelanger | ruben23: doc/backtrace.txt |
| 13:55.53 | *** join/#asterisk crowbar7 (~JKLOBUCAR@12.2.84.2) |
| 13:56.03 | ruben23 | fauxalliance: when i get the asterisk crashed.>? what particular section on the logs..? |
| 13:56.32 | ruben23 | pabelanger: where is the docs..? |
| 13:56.47 | *** join/#asterisk zerohalo (~zerohalo@173-13-92-17-NewEngland.hfc.comcastbusiness.net) |
| 13:57.03 | pabelanger | ruben23: In the source directory of the tarball |
| 13:57.07 | *** join/#asterisk jonmasters (~jcm@dallas.jonmasters.org) |
| 13:58.45 | ruben23 | pabelanger: what will i see in that text..? |
| 13:59.10 | dwayne | o.o |
| 13:59.32 | WIMPy | Lots of letters. |
| 14:00.23 | pabelanger | ruben23: How to generate a backtrace from a coredump. That will then tell you why Asterisk keeps crashing |
| 14:01.35 | *** join/#asterisk odenkos (~odenkos@ip-212-081-019-170.static.nextra.sk) |
| 14:03.55 | *** join/#asterisk pif (~ldm@zenon.apartia.fr) |
| 14:04.40 | ruben23 | <PROTECTED> |
| 14:04.45 | ruben23 | :-( |
| 14:04.49 | fauxalliance | JFGI |
| 14:05.24 | fauxalliance | http://www.voip-info.org/wiki/view/Asterisk+Documentation+1.6.0+backtrace.txt |
| 14:06.49 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip68-99-199-10.ph.ph.cox.net) |
| 14:08.29 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 14:10.04 | *** join/#asterisk n3hxs (~HAMming@static-151-196-93-200.balt.east.verizon.net) |
| 14:10.44 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:10.44 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:14.12 | pabelanger | ruben23: read doc/backtrace.txt |
| 14:14.13 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:14.28 | pabelanger | That is the purpose, to explain how to do it |
| 14:14.42 | leifmadsen | :) |
| 14:16.01 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 14:17.56 | *** join/#asterisk Trixboxer (~Trixboxer@office.supportdepartment.net) |
| 14:26.29 | *** join/#asterisk davido1 (~david@p54B0861E.dip0.t-ipconnect.de) |
| 14:26.32 | *** join/#asterisk metiu_ (~chatzilla@85-18-228-185.ip.fastwebnet.it) |
| 14:26.55 | metiu_ | thank you for advice on using Page() to page all phones |
| 14:27.09 | metiu_ | even if the multicast RTP is still in beta I see |
| 14:27.34 | metiu_ | is there a way of setting up a "peer to peer registrar" to have no single point of failure? DUNDI? |
| 14:27.56 | davido1 | Hello room. I have Asterisk 1.6 and I use a Patton SmartNode to connect to my ISDN line. When I call any number and the called party answers really fast, I miss the first 1 or 2 seconds of what he said.. .Any ideas why? |
| 14:27.56 | metiu_ | I will have around 10 sites with one phone and one page speaker |
| 14:28.44 | *** join/#asterisk BANSAL (~bansal@117.199.115.173) |
| 14:31.19 | pabelanger | network delay |
| 14:33.03 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 14:33.45 | davido1 | pabelanger: network delay? |
| 14:35.44 | pabelanger | yes, delay. Far end answers, but for some reason, there is network delay on you receiving a notification the call was answered. So, RTP is established before the call control notifies you |
| 14:36.03 | pabelanger | hence the missing 1 - 2 seconds of audio |
| 14:36.57 | *** join/#asterisk francispereira (~francis@124.153.66.246) |
| 14:38.17 | francispereira | I am taking Now for a spin and I keep getting WARNING[9653] pbx.c: Channel 'SIP/6000-00000012' sent into invalid extension '9999999999' in context 'DLPN_outgoing_dial_plan', but no invalid handler when i try and make a outgoing call |
| 14:38.46 | davido1 | pabelanger: Yes, that makes sense... But do you think that can happen if they are both in the same LAN? And if so, should I configure something on Asterisk to prevent this behaviour? |
| 14:39.05 | davido1 | pabelanger: Besides, it happens with every call I make... Not only once in a while |
| 14:39.21 | *** join/#asterisk myster (~myster@207.148.172.210) |
| 14:39.29 | francispereira | i have I have defined a trunk , an outgoing calling rule , a dial plan and an user |
| 14:40.39 | pabelanger | davido1: Depending on your dialplans, you can usually set Wait(.5) prior to Answer() |
| 14:41.13 | davido1 | pabelanger: Can I try that on outgoing calls? |
| 14:41.37 | pabelanger | francispereira: The error tells you the problem. 9999999999@DLPN_outgoing_dial_plan does not exist |
| 14:41.39 | davido1 | pabelanger: I can't really Wait->Answer->Dial? |
| 14:42.13 | pabelanger | davido1: Correct, so no. If you don't have any control over the far end, there is little you can do |
| 14:42.16 | *** join/#asterisk pif (~ldm@zenon.apartia.fr) |
| 14:42.40 | pabelanger | davido1: What interface you dialing out? |
| 14:42.48 | pabelanger | IE: SIP, DAHDI |
| 14:42.49 | davido1 | pabelanger: SIP |
| 14:43.01 | pabelanger | davido1: an ITSP? |
| 14:43.11 | davido1 | pabelanger: my patton smartnode is registered as a SIP friend on Asterisk |
| 14:43.13 | drmessano | [10:27] <davido1> Hello room. I have Asterisk 1.6 and I use a Patton SmartNode to connect to my ISDN line. When I call any number and the called party answers really fast, I miss the first 1 or 2 seconds of what he said.. .Any ideas why? |
| 14:43.31 | pabelanger | Then call smartnode and tell them the problem |
| 14:43.43 | pabelanger | the delay is on their network |
| 14:43.54 | davido1 | pabelanger: Nono, smartnode is the model of the device |
| 14:44.08 | pabelanger | whatever, call your ISDN provider then |
| 14:44.45 | drmessano | davido1: I would be more inclinded to blame the Patton box. Maybe the manufacturer is aware of an issue with call delay? |
| 14:44.53 | pabelanger | Unless the delay is coming from your smartnode |
| 14:45.02 | drmessano | davido1: Not the first issue with a Patton box I have heard about lately |
| 14:45.30 | *** join/#asterisk trelane` (~trelane@funtoo/staff/trelane) |
| 14:45.34 | davido1 | drmessano: I know, they are very convenient devices, but sometimes a bit buggy |
| 14:45.42 | davido1 | pabelanger: That's what I think might be happening |
| 14:45.53 | drmessano | davido1, Yes, this is probably another case |
| 14:45.55 | davido1 | pabelanger: I wanted to know if someone else has this problem... :s |
| 14:46.30 | davido1 | drmessano: Ohwell... |
| 14:46.56 | pabelanger | So next time, state what you think the problem is so we can help confirm / deny it. But your right, the problem is outside of asterisk |
| 14:48.01 | drmessano | pabelanger: I think he wanted more than picking out the 3 nouns in the sentence and telling him to call each of them. |
| 14:48.58 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 14:49.49 | francispereira | pabelanger, i have an outgoing plan called outgoing_dial_plan |
| 14:50.33 | francispereira | maped to an outgoing rule calls_to_mobile |
| 14:50.37 | davido1 | pabelanger & drmessano: tanks for your time dudes... |
| 14:51.15 | francispereira | and in call_to_mobile i say pattern = _XXXXXXXXXX -> use trunk |
| 14:51.22 | pabelanger | francispereira: regardless, the warning you posted tells you your problem. 9999999999@DLPN_outgoing_dial_plan does not exist, so you need to create it |
| 14:51.30 | francispereira | and i have defined my trunk |
| 14:51.55 | francispereira | 999999999 is the number i dam trying to dial |
| 14:51.56 | pabelanger | francispereira: *CLI> dialplan show 9999999999@DLPN_outgoing_dial_plan |
| 14:53.14 | francispereira | pabelanger, how do i get to the CLI. I am logged in as root via ssh |
| 14:53.31 | pabelanger | ~book |
| 14:53.31 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 14:53.35 | pabelanger | francispereira: ^^ |
| 14:53.41 | pabelanger | $ asterisk -r |
| 14:55.44 | francispereira | here is the relavent config http://pastebin.com/b7ZtfcjH |
| 14:57.23 | *** part/#asterisk davido1 (~david@p54B0861E.dip0.t-ipconnect.de) |
| 14:57.30 | pabelanger | <pabelanger> francispereira: *CLI> dialplan show 9999999999@DLPN_outgoing_dial_plan |
| 14:57.45 | pabelanger | what output do you get |
| 14:57.59 | *** join/#asterisk Letoric (~Letoric@tpi-dfw-uc-f1-69-94-238-22.totalprocess.net) |
| 14:58.51 | francispereira | dialplan show 9999999999@DLPN_outgoing_dial_plan |
| 14:58.51 | francispereira | [ Included context 'CallingRule_calls_to_mobile' created by 'pbx_config' ] |
| 14:58.52 | francispereira | <PROTECTED> |
| 14:59.20 | *** part/#asterisk jarrod (~jarrod@69.31.128.212) |
| 15:01.06 | francispereira | -= 1 extension (1 priority) in 1 context. =- |
| 15:01.52 | pabelanger | So exists, but maybe possible you don't have access to it from your outgoing context. |
| 15:01.58 | [TK]D-Fender | francispereira: Show us the complete call failure and dilaplan |
| 15:02.02 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 15:02.29 | *** join/#asterisk cusco (~trilili@213.63.137.210) |
| 15:02.30 | cusco | hi |
| 15:02.42 | leifmadsen | ${,${EXTEN:0})} <-- crazy |
| 15:02.45 | Nwab | hi, i'm using twinkle and sip-communicator, i cannot hear anything from the caller. But he hears me and himself with a 4-5 seconds lag. What happen ? any idea ? |
| 15:02.55 | Nwab | on a lan |
| 15:03.11 | [TK]D-Fender | Nwab: Both of you are on the same lan? |
| 15:03.12 | francispereira | [TK]D-Fender, how do i pull out the complete call failuare ? |
| 15:03.18 | drmessano | An AT&T cell tower walks into a bar. Bartender says, "Why the long face?" Tower says, "I've got ni...and th...b...*click*" |
| 15:03.19 | Nwab | [TK]D-Fender, yes |
| 15:03.19 | [TK]D-Fender | francispereira: * CLI <- |
| 15:03.46 | [TK]D-Fender | Nwab: Try different clients, and check your FIREWALLS |
| 15:03.50 | pabelanger | ~collectdebug |
| 15:03.51 | infobot | methinks collectdebug is a method of collecting logs allowing others help troubleshoot an issue. Refer to doc/HOWTO_collect_debug_information.txt or http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt |
| 15:03.53 | pabelanger | francispereira: ^ |
| 15:04.27 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 15:04.56 | Nwab | [TK]D-Fender, which client (for example) ? my firewalls are all set like this ACCEPT/ACCEPT/ACCEPT |
| 15:05.20 | *** join/#asterisk sgimeno (~chatzilla@163.117.211.10) |
| 15:05.20 | [TK]D-Fender | Nwab: ANY. Just check to something else and test |
| 15:05.25 | Nwab | [TK]D-Fender, do i miss some codecs ? |
| 15:05.34 | Nwab | [TK]D-Fender, ok i try this |
| 15:05.37 | [TK]D-Fender | Nwab: No, otherwise the call would have been refused |
| 15:07.17 | *** join/#asterisk nighty^ (~nighty@x122091.ppp.asahi-net.or.jp) |
| 15:07.50 | Nwab | [TK]D-Fender, but the we use the same client |
| 15:08.16 | [TK]D-Fender | [11:02]<Nwab>hi, i'm using twinkle and sip-communicator <=--- doesn't sound the SAME to me |
| 15:09.34 | cusco | how can I find out since when a pri span is down ? |
| 15:09.52 | Nwab | [TK]D-Fender, we both use twinkle for the moment ... |
| 15:10.27 | pabelanger | cusco: should be listed in the logs, if you had it enabled in logger.conf |
| 15:10.47 | cusco | pabelanger: in full? |
| 15:11.24 | pabelanger | where ever you redirected warning or debug too |
| 15:12.00 | francispereira | [TK]D-Fender, here is the debug output |
| 15:12.01 | francispereira | http://pastebin.com/qbyQLusm |
| 15:12.59 | cusco | well.. I have constant messages like "Primary D-Channel on span 7 down" |
| 15:13.07 | cusco | but its not span 7 I am worried about, its span 3 |
| 15:13.11 | cusco | and it doesn't come up... |
| 15:13.49 | [TK]D-Fender | francispereira: All junk. disable sip & core debug and provide basic CLI @ verbose 10 |
| 15:14.02 | cusco | actually it is, I looked up further back.. ok thanks |
| 15:14.25 | [TK]D-Fender | Nwab: and check all firewalls on all systems |
| 15:14.46 | pabelanger | [Sep 22 20:40:51] VERBOSE[11322] pbx.c: -- Executing [9890960855@DLPN_outgoing_dial_plan:1] Macro("SIP/6000-00000016", "trunkdial-failover-0.3,SIP/trunk_1/,,trunk_1,") in new stack |
| 15:14.48 | pabelanger | is your problem |
| 15:15.40 | Nwab | [TK]D-Fender, they are all on ACCEPT/ACCEPT/ACCEPT |
| 15:16.01 | Nwab | [TK]D-Fender, i think i got a problem with my sound card ... :/ |
| 15:16.10 | francispereira | pabelanger, what is wrong with it ? |
| 15:16.15 | Nwab | [TK]D-Fender, i try to fix it |
| 15:16.40 | pabelanger | francispereira: *CLI> core show application Dial |
| 15:16.46 | *** join/#asterisk psilikon (~joel@cerberus.vicimarketing.com) |
| 15:17.14 | pabelanger | you should have Dial(SIP/provider/<actual number you want to dial> |
| 15:17.52 | [TK]D-Fender | pabelanger: .... You don't know do you... |
| 15:18.06 | [TK]D-Fender | francispereira: GUI's are not supported here. #asterisk-gui <-------------- |
| 15:18.43 | [TK]D-Fender | francispereira: And that one isn't even being maintained. If you're lucky you catch one of the handful of people who use it an may be able to help you there. |
| 15:18.45 | pabelanger | drop AsteriskNOW and FreePBX, just install asterisk on your box and create your dialplans youself. Otherwise, you'll never understand how it works |
| 15:18.57 | francispereira | done |
| 15:19.28 | francispereira | thanks for that tip |
| 15:19.37 | francispereira | starting again from square 1 |
| 15:21.48 | [TK]D-Fender | pabelanger: He isn't USING FreePBX |
| 15:21.51 | [TK]D-Fender | ... |
| 15:21.51 | *** join/#asterisk ccomp5950 (~ccomp5950@24.204.47.5) |
| 15:22.02 | [TK]D-Fender | looks around for his rusty-nail upgraded ClueBat (tm) |
| 15:23.07 | francispereira | I was using Asterisk-GUI |
| 15:24.39 | pabelanger | s/FreePBX/Asterisk-GUI/ |
| 15:31.21 | *** join/#asterisk Nwab (~Benwa@unaffiliated/benwa) |
| 15:31.56 | Nwab | [TK]D-Fender, Arf, the probleme came from my soundcard, sorry :/ Works great now !! |
| 15:37.56 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 15:38.44 | p3nguin | What kind of device would be needed to connect a "normal" phone to ISDN? |
| 15:39.00 | *** join/#asterisk Tim_Toady (~moi@77.49.122.124.dsl.dyn.forthnet.gr) |
| 15:41.38 | metiu_ | is it possible to use DUNDi to share contacts among 10 servers? |
| 15:46.16 | [TK]D-Fender | metiu_: "contacts"? Pardon? |
| 15:48.17 | pabelanger | p3nguin: a gateway? |
| 15:50.04 | leifmadsen | define contacts |
| 15:51.21 | *** join/#asterisk muffinz (~thomas@ppp-58-8-240-86.revip2.asianet.co.th) |
| 15:52.04 | p3nguin | pabelanger: Do you know of any inexpensive ISDN gateways? |
| 15:52.31 | leifmadsen | Patton |
| 15:52.37 | leifmadsen | those are the only cheap ones I'm aware of |
| 15:52.52 | pabelanger | how cheap is cheap? |
| 15:53.30 | muffinz | Hello room |
| 15:54.29 | p3nguin | Cheap, but still retaining a fair amount of quality, I hope. |
| 15:56.00 | drmessano | DUNDi is pretty good for sharing dialplans across many, many boxes and making beer money for leifmadsen |
| 15:56.03 | metiu_ | [TK]D-Fender: I need a no single point of failure net, so I am planning of having one asterisk server on each machine, with two softphones connected to each server, but I'd need all the asterisk servers to cooperate as a single network |
| 15:56.29 | leifmadsen | that is more complicated than stated |
| 15:56.34 | metiu_ | and they will have dynamic IP addresses |
| 15:56.37 | leifmadsen | DUNDi solves only part of that equation |
| 15:56.49 | pabelanger | p3nguin: Always setup an asterisk box with a digium card, I would consider that cheap |
| 15:56.57 | muffinz | I'm trying to ringall extensions but limit the call duration depending on which extention picks up. I use this command to ringall and limit the call time 'Dial(sip/123&sip/234&sip/345, 30, S(n)) ' what I'm really looking for is something like this if possible : 'Dial(sip/123,S(n)&sip/234,S(n)&sip/345,S(n), 30) ' , is this possible? |
| 15:57.43 | Letoric | muffinz - what I did for that, which isn't exactly what you are asking, but close, is had 3 ring groups |
| 15:58.10 | Letoric | it would ring phones 1/2/3 for 30 seconds, 1/2/3/4/5 for 30, and then 1/2/3/4/5/6/7 for 10 before rolling to an oncall person |
| 15:58.14 | muffinz | Letoric, please do tell, any info would be a help |
| 15:58.17 | Letoric | don't know if that helps you or not |
| 15:58.25 | Letoric | it was 3 separate dial lines |
| 15:59.08 | Letoric | ie, Dial(Sip/1&Sip/2&Sip/3,30,m) first line, then next step was that plus the other 2 extensions added in |
| 15:59.22 | muffinz | tried that but really need to ring all extensions in one go only |
| 15:59.42 | p3nguin | You won't be able to tell the difference. |
| 15:59.43 | Letoric | wish I could be more help, I'm really new to the product still |
| 16:00.23 | muffinz | Letoric, thanks though |
| 16:00.51 | Letoric | muffinz: If you use my concept, I think you can achieve what you are going for, just ring all phones in the firs tline for x seconds |
| 16:00.57 | drmessano | 2 Dial lines would work fine |
| 16:00.58 | Letoric | then in the second line, the remaining phones for x seconds |
| 16:02.17 | muffinz | yes, it would be very close to the real thing |
| 16:02.27 | drmessano | No close about it |
| 16:02.28 | bougyman | i'd think you can make a cheaper one with open source and an isdn card than you could buy a gateway for, p3nguin |
| 16:02.38 | *** join/#asterisk drudge` (tacos@unaffiliated/drudge/x-837452) |
| 16:02.40 | drudge` | now |
| 16:02.46 | drudge` | er.. |
| 16:02.46 | pabelanger | muffinz: setup a macro, executed after the extension answers, then check which extension answer, then set timeout for that extension |
| 16:02.59 | drmessano | Can you really detect the time between the last ring of the first dial line and the first ring of the second, no |
| 16:03.03 | metiu_ | leifmadsen: I have an in-house solution which addresses the problem without SIP and everything, but I'd rather use some proven solution |
| 16:03.06 | p3nguin | The Patton SmartNode SN4552/2BIS can be purchased for about $300. |
| 16:04.12 | *** join/#asterisk [cannibalera] (~cannibale@201-41-238-109.fnsce703.dsl.brasiltelecom.net.br) |
| 16:04.27 | muffinz | pabelanger: that's what I was thinking but how do I set timeout for that extension after it's answered? |
| 16:04.56 | pabelanger | muffinz: *CLI> core show function TIMEOUT |
| 16:05.35 | muffinz | pabelanger: if this works, I could kiss you!!! :D |
| 16:05.47 | bmoraca_work | p3nguin: how does that behave with faxes? the SPA8000 is a bit cheaper but sucks for faxing |
| 16:05.56 | pabelanger | muffinz: beer > kiss |
| 16:06.12 | muffinz | exactly :D |
| 16:06.22 | *** join/#asterisk Kalidarn (~unknown@unaffiliated/kalidarn) |
| 16:06.34 | *** part/#asterisk [cannibalera] (~cannibale@201-41-238-109.fnsce703.dsl.brasiltelecom.net.br) |
| 16:06.36 | Nwab | is it better to use a real time kernel for an asterisk server ? |
| 16:06.43 | Kalidarn | this probably isn't the best place to ask but has anyone got a cisco 7960? |
| 16:07.00 | bmoraca_work | Kalidarn: yes. |
| 16:07.24 | Kalidarn | the XML directoryURL thing i've got perfectly working for my 7912 phones, but as soon as i go into the directories on my 7960 i can't see it in there |
| 16:07.30 | Kalidarn | just missed calls, recieved calls and placed calls |
| 16:07.42 | Kalidarn | both phones are getting it when i go in there i see requests on my apache logs |
| 16:07.55 | Kalidarn | im just not sure why the 7960 refuses to show it in the Directory menu. |
| 16:08.05 | pabelanger | Nwab: no, depends on what you are doing. I usually use the stock Ubuntu-server kernel |
| 16:08.12 | Kalidarn | and ive done about 4 hours of research |
| 16:08.20 | Kalidarn | im using SCCP |
| 16:08.46 | Kalidarn | (so there's no sip configuration, i did see people mentioning that) |
| 16:08.52 | bmoraca_work | Kalidarn: only thing I could suggest is that your XML file is incorrect |
| 16:08.53 | Nwab | pabelanger, nothing special, i was asking this because of the latency ... |
| 16:09.11 | Nwab | (sorry for my bad english, weird syntax, i do know) |
| 16:09.12 | Kalidarn | for my 7912 i didn't have to do anything but set the url and have a properly formatted xml file in my directory |
| 16:09.25 | Kalidarn | bmoraca_work: yeah i thought about that have you used this feature on a 7960? |
| 16:09.36 | metiu_ | would it help to use some sip proxy that handles the registration? or could I use multicast to register simultaneously on all the servers which are listening? they will all be on the same network, so no route issues |
| 16:09.41 | bmoraca_work | Kalidarn: yes, and 7940s, but only in SIP firmware |
| 16:09.49 | pabelanger | Nwab: latency in what? |
| 16:09.50 | Kalidarn | ah |
| 16:10.06 | Kalidarn | (it shows the directory url correctly under the network settings) |
| 16:10.29 | Kalidarn | on my 7912 it just appeared as another option on the menu |
| 16:10.37 | Kalidarn | (which then had all the numbers in it) |
| 16:10.45 | *** join/#asterisk eugeneoden (~goden@rrcs-97-77-255-86.sw.biz.rr.com) |
| 16:10.52 | *** join/#asterisk [sr] (~Unknowned@pa8-84-91-197-8.netvisao.pt) |
| 16:11.04 | muffinz | pabelanger: I'm still struggling with the concept of injecting a timeout in a call that's in progress, I've googled for something like this 2 days |
| 16:11.06 | drmessano | metiu_, you could use two boxes, dundi, and DNS srv records to set priority.. |
| 16:12.19 | Nwab | pabelanger, well, with a real time kernel you ve'got less latency, no ? |
| 16:12.20 | pabelanger | muffinz: in progress, don't think you can. After the phone answered, yes you can |
| 16:13.08 | metiu_ | drmessano: my network could be partitioned at any time and I'd need to be able from any headset to page all the reachable clients, that's why I need a server at each headset |
| 16:13.30 | pabelanger | Nwab: That's a generic question. What latency are you experiencing. A realtime kernel will not help with network latency |
| 16:13.42 | p3nguin | bmoraca_work: The SmartNode supports fax, but I couldn't say how well it works. |
| 16:14.09 | pabelanger | muffinz: What do you want to set the timeout on, the ringing of the extension, or when to hangup the phone after answered? |
| 16:14.14 | *** join/#asterisk thansen (~thansen@c-98-202-28-239.hsd1.ut.comcast.net) |
| 16:14.14 | drmessano | metiu_, how is an asterisk box going to be more reachable than a client if the network conditions change? |
| 16:14.24 | Kalidarn | bmoraca_work: http://pastebin.com/mHS97eCS that was my php script and this is how it comes out when requested http://pastebin.com/t1yanguu |
| 16:14.34 | muffinz | pabelanger: when to hangup after answer |
| 16:15.33 | *** join/#asterisk ybit (~quassel@unaffiliated/ybit) |
| 16:15.35 | Nwab | pabelanger, but if i've got a problem of latency (not yet), i do know it's from network and not kernel. Am i right ? |
| 16:15.41 | Kalidarn | err rather http://pastebin.com/YwwyDntp there's no gap |
| 16:17.03 | pabelanger | Nwab: no, you would have to figure out what is causing it. IE: lack of CPU resources may cause latency. If you are getting audio delay, it could be a result of lack of bandwidth on your network |
| 16:17.19 | *** join/#asterisk ybit (~quassel@unaffiliated/ybit) |
| 16:17.30 | Nwab | pabelanger, IE ?? |
| 16:17.47 | pabelanger | but in general, a kernel (specifically) is not the likely cause of delay |
| 16:17.58 | pabelanger | Nwab: example |
| 16:17.59 | Naikrovek | Kalidarn: your $rows array is a bit wonky. you set it 5 times to the same thing. |
| 16:18.03 | Nwab | but it could be, no ? |
| 16:18.47 | pabelanger | Yes, it could be. But I could also get hit by a car walking to the corner store today |
| 16:19.02 | metiu_ | drmessano: the system is made of 10 eth switches in daisy chain, each box will be connected to a switch and will have a headset attached to it, I need to be able to page all reachable headsets any time, even if the network is partitioned (i.e. one switch fails) |
| 16:19.20 | pabelanger | My point, in my experience, a realtime kernel is not required for asterisk |
| 16:19.28 | Kalidarn | Naikrovek: yeah i obfuscated out the real numbers and names |
| 16:19.40 | Kalidarn | those values are substituted for real values |
| 16:19.44 | Kalidarn | that are different each row |
| 16:19.45 | p3nguin | nwab: IE is Internet Explorer |
| 16:20.05 | p3nguin | pabelanger: ^^ |
| 16:20.05 | Naikrovek | Kalidarn: k |
| 16:20.05 | p3nguin | e.g. means for example. |
| 16:20.24 | Nwab | p3nguin, :) |
| 16:20.30 | drmessano | metiu_, 2 Asterisk boxes gives you TWO points of failure.. all 10 would connect to ONE box at a time.. if one of the switches with a client goes down, they're gone.. so be it. If one of the switches with an Asterisk box goes down, the available clients jump to box 2. You dont need 10 asterisk boxes |
| 16:20.45 | Naikrovek | i.e. is "id est" which means "that is" |
| 16:20.59 | p3nguin | and that is an ALL-INCLUSIVE list. |
| 16:21.05 | p3nguin | not just a single example. |
| 16:21.21 | carrar | But 10 boxes can't hurt! |
| 16:21.23 | Naikrovek | e.g. is "exempli gratia" meaning "for example" roughly |
| 16:21.33 | Naikrovek | studied latin in high school |
| 16:21.37 | *** join/#asterisk espiceland (~erin@nat/digium/x-ueoclifmathsoatu) |
| 16:21.39 | Kalidarn | i get the idea it doesn't change anyway depending on model of phone based on http://www.voip-info.org/wiki/view/Asterisk+Cisco+79XX+XML+Services |
| 16:21.39 | p3nguin | is glad at least one person on IRC knows Latin. |
| 16:21.51 | Naikrovek | i only know a little |
| 16:22.10 | p3nguin | More than most demonstrate. |
| 16:22.14 | Kalidarn | latin is even taught in school anymore? |
| 16:22.18 | p3nguin | no |
| 16:22.22 | underdog | it's dead |
| 16:22.30 | Kalidarn | the only people i know who learn it in school are 40+ years old |
| 16:22.35 | Naikrovek | it's been dead for a while |
| 16:22.42 | underdog | yes...that's the joke |
| 16:22.49 | Naikrovek | some schools still teach it but good luck finding students who want to learn it |
| 16:23.08 | Naikrovek | i'm 35 and my latin II class was the last year it was taught in my district |
| 16:23.27 | drmessano | metiu_, You don't need a star topology between 10 Asterisk boxes to be able to page 10 available clients... 2 boxes gives you TWO points of failure.. if you're paranoid, use 3 boxes, DNS SRV records, and DUNDI.. 10 is insane |
| 16:24.37 | muffinz | pabelanger: using AMI I can send an AbsoluteTimeout to an active call, I think |
| 16:25.06 | metiu_ | drmessano: your analysis is correct, it's just that the system needs to be "safety critical" (will have to alert people) and the best we can offer is that we would be able to reach anyone in reach, right now we are using simple rtp multicast and spread daemon to coordinate |
| 16:25.08 | drmessano | metiu_, if you use 10 boxes, you don't need any sort of high level logic (like DUNDI) to locate available clients.. just blast your Page to the other 9 boxes.. If one is down, it won't go there |
| 16:25.52 | muffinz | pabelanger: thank you for your time |
| 16:26.00 | [TK]D-Fender | drmessano: 2x SER, 2 x *, DNSSRV |
| 16:26.17 | metiu_ | drmessano: that looks interesting: what if the boxes have dynamic IP addresses? I'd guess it's a DNS problem |
| 16:26.41 | drmessano | metiu_, Why wouldn't you assign a static to the 10 Asterisk boxes? |
| 16:27.13 | carrar | multicast paging! |
| 16:27.20 | [TK]D-Fender | \o/ --- yay, moving targets! |
| 16:27.32 | metiu_ | drmessano: not my choice, I don't own the switches, I'll have DHCP |
| 16:27.33 | drmessano | carrar, 10 * boxes = multiblast paging |
| 16:27.48 | carrar | 10 isn't enough |
| 16:27.53 | drmessano | True |
| 16:28.01 | drmessano | What if one of the Asterisk boxes crashes |
| 16:28.19 | drmessano | You really need 20.. 2 at each location, phone registered to both |
| 16:28.33 | carrar | At the min!! |
| 16:28.43 | drmessano | Wait |
| 16:29.00 | drmessano | What if you do 10 asterisk boxes, and 10 line phones.. have each phone reg to all 10 boxes? |
| 16:29.02 | [TK]D-Fender | No. clearly 2 server's per phone, and a backup phone for each phone as well |
| 16:29.07 | metiu_ | Studied Latin, too (and are under 40), but I'm in Italy... |
| 16:29.19 | [TK]D-Fender | WARNING: res_clusterfuck.so is already loaded! |
| 16:29.41 | drmessano | [TK]D-Fender, 20 x 20-line phones, 20 boxes, each phone registers to all 20 boxes |
| 16:29.47 | pabelanger | drmessano: I would be more worried about power source for redundancy |
| 16:30.02 | drmessano | pabelanger, Good point.. generators |
| 16:30.07 | drmessano | Gonna need 2 each per phone |
| 16:30.08 | [TK]D-Fender | pabelanger: Get the fuck back on your wheel before you cause a brown-out! |
| 16:30.12 | [TK]D-Fender | MUSH! |
| 16:30.14 | [TK]D-Fender | :p |
| 16:30.21 | drmessano | 2 Generators and 2 PDUs for each phone |
| 16:30.24 | underdog | don't forget DR site just in case you lose your primaries at once |
| 16:31.04 | [TK]D-Fender | http://www.youtube.com/watch?v=y81bW4jE9wI <--------- |
| 16:31.39 | drmessano | 3 PDU's and 2 Generators.. Each PDU connects to one source of 120v, one side of the PDU goes to a generator, the other goes to another PDU that plugs into the second generator/120v combo |
| 16:32.11 | pabelanger | [TK]D-Fender: I actually heard a story from Mark about a village in south america the uses bicycle power to power the village asterisk box. |
| 16:32.27 | drmessano | pabelanger, I heard it was AT&T |
| 16:32.51 | drmessano | pabelanger, No, getting my stories mixed up.. AT&T uses homeless people on bicycles to power their cell sites |
| 16:32.59 | underdog | and by south america you mean the southern part of the states |
| 16:33.11 | underdog | anything below kansas |
| 16:33.16 | drmessano | "Hobo fade" is when one kicks the bucket and needs to be replaced |
| 16:34.24 | drmessano | Hobo fade is a big problem in the winter time, since AT&T takes away their jackets because it would only weigh them down |
| 16:34.24 | metiu_ | drmessano: thank you for your time |
| 16:34.52 | Naikrovek | anyone here live in india |
| 16:35.02 | Naikrovek | and knows india telecom law regarding voip |
| 16:35.54 | drmessano | I guess I shouldn't bust on AT&T in here too much.. I started busting on Skype and 2 months later we had chan_skype. I bet chan_evil or chan_att is right around the corner |
| 16:37.01 | Naikrovek | [TK]D-Fender: i lol'd at that video |
| 16:39.39 | Nwab | how to calcule |
| 16:39.54 | Naikrovek | calculate what |
| 16:40.07 | Nwab | how to calculate the bandwith of one or more call |
| 16:40.29 | [TK]D-Fender | Nwab: cALCULATE 1. mULTIPLY. tHE eND |
| 16:40.42 | Naikrovek | caps lock again? |
| 16:40.42 | underdog | http://www.google.com/search?q=voip+bandwidth+calculator&ie=utf-8&oe=utf-8&aq=t&rls=org.mozilla:en-US:official&client=firefox-a |
| 16:40.50 | underdog | ^ google search |
| 16:41.04 | pabelanger | underdog: please use lmgtfy.com |
| 16:41.06 | pabelanger | :) |
| 16:43.56 | underdog | so there's a defacto channel search site? |
| 16:44.25 | underdog | goes out to bring back dogpile.com |
| 16:45.07 | Nwab | underdog, thanks i did not know this kind of tool |
| 16:45.28 | underdog | huh? what tool |
| 16:45.58 | underdog | the bandwidth calculators? |
| 16:46.53 | carrar | defacto! |
| 16:47.21 | carrar | underdog, whatcha lookin fur? |
| 16:47.24 | bougyman | does asterisk have codec2 in any of the production releases? |
| 16:47.49 | *** part/#asterisk gradgrind (~MichelRP@2001:470:c10d::feed) |
| 16:47.56 | Naikrovek | codec2? |
| 16:48.13 | [TK]D-Fender | bougyman: No |
| 16:48.20 | bougyman | Naikrovek: http://news.slashdot.org/story/10/09/21/0428259/Codec2-mdash-an-Open-Source-Low-Bandwidth-Voice-Codec |
| 16:48.27 | [TK]D-Fender | bougyman: Its brand new and nobody cares about it |
| 16:48.33 | Naikrovek | it's a super-shitty name |
| 16:48.36 | bougyman | [TK]D-Fender: well, i care. |
| 16:48.52 | Naikrovek | ilbc is a low bandwidth open source codec i believe |
| 16:48.53 | bougyman | to see if I can lessen the bw on branch office vici (asterisk) servers. |
| 16:48.59 | [TK]D-Fender | bougyman: Like I said "nobody" cares about it :p |
| 16:49.06 | bougyman | anything but ulaw has ended up with horrible quality. |
| 16:49.19 | carrar | 729? |
| 16:49.19 | bougyman | which means i'm using a lot of bw I don't need to. |
| 16:49.25 | *** part/#asterisk sekil (~sekil@80.93.247.26) |
| 16:49.26 | [TK]D-Fender | bougyman: G.729 is fine. I find GSm fine as well.. |
| 16:49.27 | Naikrovek | 729 his 90% of the quality and 1/8th of the bandwidth |
| 16:49.30 | citywok | gsm works great for me |
| 16:49.53 | citywok | if your upstream provider doesnt support it like one of mine (qwest), you have to go g729 for that. |
| 16:50.01 | bougyman | 729 isn't open source. |
| 16:50.08 | carrar | doesn't need to be! |
| 16:50.08 | bougyman | gsm sounds horrible |
| 16:50.25 | citywok | bougyman: i run a call center on it for 2 years now without any trouble. it sounds just as good as our old PRI's. |
| 16:50.33 | bougyman | codec2 tests sound identical to ulaw |
| 16:50.56 | Naikrovek | it's open source. implement it or pay someone to do it |
| 16:51.00 | Naikrovek | then you can use it |
| 16:51.03 | carrar | yeah |
| 16:51.04 | carrar | HURRY |
| 16:51.15 | carrar | I can help with my sound card and solidering iron |
| 16:51.23 | citywok | or pay for the $10 g729 licenses. lol |
| 16:51.36 | WIMPy | BTW: Am I right in assuming that GSM only means the original codec? |
| 16:51.39 | bougyman | citywok: the bandwidth for ulaw is cheaper than that. |
| 16:51.43 | bougyman | makes no financial sense. |
| 16:51.50 | bougyman | notwithstanding my preference for open source. |
| 16:52.07 | carrar | 711 is always better if you can afford the bw |
| 16:52.09 | [TK]D-Fender | WIMPy: GSM610 |
| 16:52.12 | citywok | bougyman: i highly doubt that. 10 calls saturates a T1. g729 will get you 50. |
| 16:52.16 | carrar | problem solved |
| 16:52.17 | bougyman | 10 calls? |
| 16:52.21 | bougyman | we get 30 out of a T1 |
| 16:52.27 | Naikrovek | on g711? |
| 16:52.31 | bougyman | yes. |
| 16:52.31 | citywok | using ulaw? |
| 16:52.32 | carrar | hahah |
| 16:52.33 | citywok | lmao |
| 16:52.42 | Naikrovek | 18 maybe |
| 16:52.46 | citywok | ulaw IIRC uses 160kbit of bandwidth per call |
| 16:52.47 | carrar | TK |
| 16:52.52 | carrar | need add that to the bot |
| 16:52.53 | citywok | 1500/160 < 30 |
| 16:53.02 | bougyman | compress that nearly *2 |
| 16:53.21 | bougyman | we have an adtran that does the compression between us and the provider for that. |
| 16:53.33 | carrar | ouch |
| 16:53.34 | citywok | lol... a router can't compress voice traffic |
| 16:53.37 | bougyman | 18's about right, uncompressed. |
| 16:53.41 | carrar | a router CAN compress |
| 16:53.42 | bougyman | citywok: i beg to differ. |
| 16:53.45 | bougyman | we have two which do. |
| 16:53.52 | citywok | do you control both endpoints? |
| 16:53.55 | WIMPy | So that makes a yes. Would be nice to get a little update. |
| 16:53.58 | citywok | and if you use GSM, does your router still compress? |
| 16:54.00 | bougyman | no, the provider provided the compressor. |
| 16:54.08 | bougyman | and it's strictly ulaw |
| 16:54.09 | [TK]D-Fender | bougyman: Since G.711 = 64kbps, 64 x 1920kbps. T1 = 1544kbps. Please explain this "compression" of yours... Also given this would seem to assume it not being carried by IP since UDP overhead would kill you |
| 16:54.13 | citywok | compression on top of compression = bad |
| 16:54.23 | [TK]D-Fender | bougyman: Please explain what you are doing that allows for this./ |
| 16:54.32 | bougyman | [TK]D-Fender: it's not my solution, it's airesprings. |
| 16:54.33 | [TK]D-Fender | citywok: Lossy compression, yes |
| 16:54.39 | Naikrovek | ulaw = 64bit, overhead = ~8kbit/s |
| 16:54.46 | [TK]D-Fender | bougyman: And what are you connecting using? |
| 16:54.48 | bougyman | i was doubtful at first but it's matching or exceeding the call quality of pure PRI. |
| 16:55.05 | bougyman | [TK]D-Fender: we connect PRI to the adtran, it does ual voip over 1 T1 to produce two PRIs. |
| 16:55.07 | Naikrovek | 1544 / (64+8) = 21.4 |
| 16:55.11 | carrar | I compress my ZIP files 10 times |
| 16:55.15 | bougyman | er ulaw voip. |
| 16:55.16 | [TK]D-Fender | [12:54]<Naikrovek>ulaw = 64bit, overhead = ~8kbit/s <- actually, its 20kbps for UDP overhead |
| 16:55.25 | Naikrovek | stands corrected |
| 16:55.26 | citywok | carrar: i zip then rar and ace to get the best. |
| 16:56.02 | [TK]D-Fender | [12:54]<bougyman>i was doubtful at first but it's matching or exceeding the call quality of pure PRI. <- impossible. You can't make the PSTN better than it is |
| 16:56.20 | carrar | how much latency does this compression add? |
| 16:56.21 | [TK]D-Fender | bougyman: PSTN runs at G.711 |
| 16:56.27 | bougyman | carrar: none. |
| 16:56.29 | bougyman | [TK]D-Fender: i know. |
| 16:56.41 | WIMPy | Or G.722 |
| 16:56.44 | [TK]D-Fender | bougyman: And the thought that Codec2 can rival G.711 for quality is pretty much ludicrous |
| 16:56.45 | bougyman | the technical details are separate from the live agent feedback. |
| 16:56.54 | bougyman | it could just be psychosematic, but that's what they report. |
| 16:56.58 | WIMPy | Even if I haven't seen that anywhere. |
| 16:57.13 | [TK]D-Fender | bougyman: Oh so yuo're spout ing the ravings of some nut to us as "fact"? |
| 16:57.18 | citywok | if you replaced the phones and everything it could sound better to them if what you had before sucked. |
| 16:57.19 | [TK]D-Fender | bougyman: EGood to know :) |
| 16:57.33 | bougyman | some nut, or a call center full of agent who've been doing this 5+ years on average? |
| 16:57.35 | citywok | hardly emprical evidence |
| 16:57.35 | carrar | heh |
| 16:57.35 | carrar | you're funny |
| 16:57.35 | carrar | Are you sure you are not doing PPP header compression? |
| 16:57.39 | bougyman | yes, i'll take their feedback. |
| 16:57.46 | WIMPy | Saying so.. It would be nice if dahdi supported G.722. |
| 16:57.55 | citywok | i dont take feedback from people that make $12/hr without a grain of salt. |
| 16:58.14 | carrar | I can do 50 g.722 calls on mydialup modem |
| 16:58.15 | p3nguin | What about $13/hr? |
| 16:58.15 | bougyman | heh, some of our agents make 20k/month. |
| 16:58.33 | Naikrovek | ! |
| 16:59.03 | *** join/#asterisk riscphree (~riscphree@h46.45.90.75.dynamic.ip.windstream.net) |
| 16:59.10 | citywok | selling drugs!? |
| 16:59.18 | bougyman | collecting money |
| 16:59.33 | carrar | is he in Hawaii |
| 16:59.35 | bougyman | a good commercial collector (business to business) averages 10k/mo |
| 16:59.39 | carrar | they call him DOG |
| 16:59.44 | citywok | lmao |
| 17:00.19 | citywok | well since you get 30+ calls on a T1 i'd just stick to what you are doing. |
| 17:00.36 | carrar | yeah |
| 17:00.36 | citywok | especially if it sounds better than the PSTN |
| 17:00.40 | carrar | get another T1 |
| 17:00.44 | carrar | I like those T1's |
| 17:00.57 | citywok | lol, yea. they only cost a couple hundred bucks in most areas. |
| 17:01.01 | bougyman | i like them at $0.0079/minute, for sure. |
| 17:01.05 | citywok | and if your agents all make 20k, then 200 is nothing. |
| 17:01.14 | carrar | If you bond those T1's together you can probably get 70+ calls on it |
| 17:01.25 | carrar | heh |
| 17:01.47 | citywok | lol. i dont even know how many calls i can run to my philippines office on 4mbit. we've never had any problems though! lol |
| 17:01.53 | bougyman | we have a few ds3's worth of these, i'm just looking at reducing traffic to branch offices so they don't need as much data, airespring doesn't offer that dual-T-in-one-T everywhere. |
| 17:02.16 | *** join/#asterisk ahowlader (~Adnan@119.30.39.49) |
| 17:02.19 | carrar | dual-T-in-one-T!! |
| 17:02.22 | citywok | you have 56+ T1s? |
| 17:02.26 | carrar | OMG thats awesome |
| 17:02.33 | citywok | wtf are you doing?!?!? |
| 17:02.41 | bougyman | citywok: no, we have 5 ds3's total. |
| 17:02.49 | bougyman | i think i already mentioned that, collecting money. |
| 17:02.55 | carrar | DOUBLE T1's ALL THE WAY |
| 17:02.57 | citywok | sounds like you guys need to learn what you are doing. lol. |
| 17:03.04 | citywok | how many agents do you have? |
| 17:03.07 | citywok | 5,000+? |
| 17:03.38 | *** join/#asterisk [cannibalera] (~cannibale@201-25-250-53.fnsce703.dsl.brasiltelecom.net.br) |
| 17:03.41 | carrar | each agent does two peoples wrok |
| 17:03.42 | carrar | work |
| 17:03.45 | bougyman | we have multiple offices with hundreds, but when using a predictive dialer it's common to dial 10/1 |
| 17:03.46 | *** join/#asterisk b0gatyr (~b0gatyr@host-208-88-126-198.biznesshosting.net) |
| 17:03.49 | citywok | Dual person in person? |
| 17:03.50 | bougyman | 10 outbound dials for every agent. |
| 17:03.51 | carrar | yes |
| 17:04.16 | carrar | 10-in-1? |
| 17:04.18 | citywok | Sounds like you need to rearchicect your office, i'm assuming you have 5 DS3s and some are point to points. |
| 17:04.36 | citywok | nothing saps money from you like private lines. MPLS FTW. |
| 17:04.39 | bougyman | somwhere between 9-1 and 10-1 is where the predictive dialers usually end up, yes. |
| 17:05.01 | carrar | 9 calls at 1 time? |
| 17:05.03 | bougyman | citywok: no, no p2p, we have vpns for the branch offices. |
| 17:05.18 | bougyman | carrar: no, it makes 9 dials expecting one to complete and transfer to an agent. |
| 17:05.18 | citywok | ewwww i'm sorry |
| 17:05.25 | carrar | ah |
| 17:05.34 | bougyman | the other 9 generally get left messages or are bad numbers/RNA/whatever. |
| 17:05.39 | carrar | what if more then 3 answer? |
| 17:05.46 | citywok | you get put on hold :P |
| 17:05.52 | bougyman | carrar: they get spun in a queue until an agent is free. |
| 17:06.02 | carrar | spamming is so wonderfull |
| 17:06.03 | citywok | carrar: they are working law of averages. you can do it with hundreds of agents. |
| 17:06.19 | citywok | carrar: they also have clients that dont care about the laws they are breaking. :) |
| 17:06.49 | carrar | bougyman, why not get a second T1 then? |
| 17:06.53 | citywok | unfortunately we don't have the luxury of being able to break laws. our clients wont let us. |
| 17:07.02 | b0gatyr | Hi everyone. I would like for someone to clarify something for me.. if I have a t1 line (Data & Voice) and would like to use it as my main link to the PSTN using asterisk and also as a backup internet link..where would I plug the T1 cable to ?? Router or Asterisk box? |
| 17:07.08 | *** join/#asterisk timahvo1 (~rogue@41.223.57.73) |
| 17:07.25 | [TK]D-Fender | [13:03]<citywok>Dual person in person? <-- oh this is becoming one of THOSE kinds of movies... |
| 17:07.32 | carrar | b0gatyr, Asterisk can split that out |
| 17:07.35 | citywok | [TK]D-Fender: oh man, that's bad. lol. |
| 17:07.45 | bougyman | carrar: T1s at the branch offices are not cheap, they're usually well above normal, cause our branch offices are in places like Tyler, TX. |
| 17:08.19 | [TK]D-Fender | b0gatyr: Depends what your current router spits out and how you want to implement things |
| 17:08.23 | citywok | oh, then why dont you pay for G729 since clearly bandwidth > licensing |
| 17:08.47 | bougyman | carrar: it's not spam, these people have relationships with the companies we are contracted by or partners with. |
| 17:08.50 | [TK]D-Fender | b0gatyr: Your currect T1 could literally be VoIP to your telco for all we know and data for tehr emained. Or it could be TDM channelized data, etc |
| 17:08.55 | citywok | if your phones support g729 and your upstream provider does you dont even need very many licenses. |
| 17:09.03 | bougyman | the bulk of our revenue comes as 'first-party' which basically means someone hired us to do their billing. |
| 17:09.57 | b0gatyr | I don't have a T1 yet |
| 17:10.08 | bougyman | no one likes collectors, but no one likes to go unpaid, either. |
| 17:10.23 | bougyman | we work for the latter. |
| 17:10.40 | [TK]D-Fender | b0gatyr: Ok, so that "if" was in fact "future tense"? |
| 17:10.57 | [TK]D-Fender | b0gatyr: What do you WANT to do? |
| 17:10.59 | b0gatyr | yes, sorry. |
| 17:12.14 | drmessano | 56 T1s? wow |
| 17:12.37 | carrar | thats not enough |
| 17:12.39 | citywok | drmessano: he acutally has 5 DS3's. IIRC DS3's have 28 channels? |
| 17:12.40 | Naikrovek | "wow" doesn't begin to describe my reaction to that |
| 17:13.04 | carrar | DS3's have 28 T1 channels |
| 17:13.19 | Naikrovek | so, 2 DS3s then |
| 17:13.26 | carrar | or 672 DS0 channels |
| 17:13.38 | b0gatyr | [TK]D-Fender: I would like to have a set up where the T1 also serves as a backup line to the internet.. I was just wondering where to plug things |
| 17:13.55 | b0gatyr | T1 >> Router or T1 >> Asterisk |
| 17:14.16 | *** join/#asterisk bent_screwdriver (~socain00@74.255.249.66) |
| 17:14.23 | carrar | b0gatyr, why not do all data and do SIP over the data T1 |
| 17:14.28 | drmessano | That's 3360 channels |
| 17:14.31 | carrar | with QoS! |
| 17:14.45 | citywok | carrar: he does with dual t1 in one t1 compression. remember? lololol |
| 17:14.56 | bougyman | we only have 2 of those. |
| 17:15.00 | *** join/#asterisk bent_screwdriver (~socain00@74.255.249.66) |
| 17:15.01 | bougyman | i was testing before buying more. |
| 17:15.07 | drmessano | Holy Jesus Easter and Thanksgiving.. that's a lot of lines |
| 17:15.13 | bougyman | plus my current DS3s don't go out of contract until december. |
| 17:15.42 | b0gatyr | carrar: what equipment do I need to do QoS? |
| 17:15.50 | carrar | 5 DS3's of voice? |
| 17:15.54 | carrar | router would be a great start |
| 17:15.55 | Naikrovek | my neighbor has a 100mbit fiber connection and he pays very little for it |
| 17:16.08 | Naikrovek | he uses a lot of it for voice |
| 17:16.10 | citywok | Naikrovek: i have a 100mbit cable connection and pay very little for it :) |
| 17:16.16 | carrar | and simple 2621 will do qos |
| 17:16.17 | Naikrovek | my employer's neighbor, i should say |
| 17:16.19 | carrar | cisco |
| 17:16.22 | b0gatyr | 2811 ? |
| 17:16.31 | citywok | yup. |
| 17:16.35 | carrar | did I say 2811 |
| 17:17.02 | b0gatyr | no, just saying.. we have 2811s |
| 17:17.04 | citywok | we use 3640, 2811, 2621, 7204's here |
| 17:17.10 | citywok | all doing MPLS + QOS |
| 17:17.24 | carrar | 2811 > 2621 so sure |
| 17:17.52 | drmessano | You only need 220 Mb/s for 3360 channels of G711 |
| 17:18.16 | Katty | omnomnoms smoked sausage |
| 17:18.38 | citywok | drmessano: he has a 10:1 ratio so he's got to make 33,360 calls to fill 3,360 agents! |
| 17:19.22 | drmessano | citywok: I am guessing around 300 agents @ 10:1 to max out the 5 DS3s |
| 17:19.35 | drmessano | Still.. craziness |
| 17:19.44 | citywok | drmessano: sort of. 10:1 only while the agent is idle. once the agent is in a call you dont need to dial the othe 10 lines. |
| 17:19.57 | citywok | so it's actually only 10:1 * number of idle agents |
| 17:20.14 | b0gatyr | so if I go for a T1 (Voice/Data) it would go connected to the asterisk box and then if I would like to use this line as an internet link the asterisk box would be the gateway for all hosts on the network? |
| 17:20.25 | citywok | with that assumption you get far more agents per T1 |
| 17:20.29 | drmessano | citywok, really doesn't matter ,, just doing the math on available channels of 5 DS3s |
| 17:20.29 | carrar | yes |
| 17:20.47 | Naikrovek | b0gatyr: the asterisk box would be just like any other box on your internal side of the router |
| 17:20.56 | citywok | drmessano: yup, i know heh. i'd guess he could run 2000 agents on 5 DS3s of bandwidth. |
| 17:20.57 | Naikrovek | and it would be a data T1 |
| 17:21.00 | Katty | hi Naikrovek |
| 17:21.09 | Naikrovek | Hi Katty :) |
| 17:21.23 | *** join/#asterisk Natureshadow (~Nik@p5B028C78.dip0.t-ipconnect.de) |
| 17:21.25 | citywok | my 100 agents use under 2mbit most of the time. |
| 17:21.26 | Naikrovek | b0gatyr: and the router would be doing QoS, giving voice priority over other things |
| 17:21.48 | Natureshadow | good evening out there ;) |
| 17:22.06 | drmessano | citywok: Put them all on G.729 and cram them into a Comcast connection |
| 17:22.35 | citywok | heh, we use g729/gsm depending on the upstream provider. no comcast or fios available in my neighborhood. just verizon copper. |
| 17:22.37 | b0gatyr | Naikrovek: but the default route on the router would be that of the asterisk box? |
| 17:22.49 | Natureshadow | I have been runnign Asterisk + FreePBX for about 2 hours with no problems, then rebooted, and Asterisk won't come up again. Even the highest debug level doesn't output anyhting useful, it just breaks away some point during statup with no error message ... |
| 17:22.52 | drmessano | 26 Mb/s for 3360 G.729 channels |
| 17:22.57 | Naikrovek | b0gatyr: no. do not use the asterisk box as a router |
| 17:22.58 | *** join/#asterisk arielb27 (~chatzilla@63.214.236.169) |
| 17:23.08 | Naikrovek | b0gatyr: the asterisk box cannot terminate a T1 anyway |
| 17:23.14 | bougyman | why not? |
| 17:23.21 | b0gatyr | Naikrovek: So how to I route all internet traffic from hosts on my network through that T1? |
| 17:23.22 | drmessano | I could cut his bill by about $3000 a month |
| 17:23.31 | drmessano | I just need SSH access and a credit card number |
| 17:23.48 | b0gatyr | do* |
| 17:23.49 | citywok | lmao. yea no kidding. |
| 17:23.55 | citywok | but he said badnwidth is cheaper than g729 |
| 17:24.00 | citywok | he must not see the bills! |
| 17:24.00 | drmessano | Oh, and my $5000 consulting fee |
| 17:24.06 | citywok | our DS3 cost us $5,000/mo |
| 17:24.21 | [TK]D-Fender | [13:23]<Naikrovek>b0gatyr: the asterisk box cannot terminate a T1 anyway <- actually, it can |
| 17:24.22 | Naikrovek | b0gatyr: all machines in office, including servers --> firewall --> router --> internet |
| 17:24.37 | Naikrovek | with hardware yes, but he wants to put internet over the T1 as well |
| 17:24.44 | [TK]D-Fender | b0gatyr: First confirm what kind of voice services your telco will offer you over that T1 |
| 17:24.51 | drmessano | What's the ROI on $20000 worth of G.729 licenses spread across a monthly savings of $3000 to $5000 |
| 17:24.54 | [TK]D-Fender | b0gatyr: then fill us in |
| 17:25.20 | drmessano | citywok: In 4 months he could pay for the G.729 licenses and remove 3359 points of failure |
| 17:25.32 | Naikrovek | why you wouldn't use a dedicated router and firewall for that is beyond me. iptables is nice but how much CPU do you want to waste on routing |
| 17:26.25 | Naikrovek | i dunno |
| 17:26.38 | Naikrovek | i believe in using hardware designed for firewalls, not iptables |
| 17:26.40 | Naikrovek | same for routing |
| 17:26.45 | Naikrovek | maybe it's the old school in me |
| 17:27.30 | Naikrovek | my Cisco ASA cost like $499 and it is a freaking champ |
| 17:27.32 | drmessano | Naikrovek, Too bad most of those toasters run *nix anyway... though I agree a toaster is a better idea that maintaining a PC/Server... let the manufacturer decide if the HD or CPU fan is bad.. |
| 17:27.46 | b0gatyr | [TK]D-Fender: As per the quote I received, says: ISDN PRI (B+D Channels) Qty 24 , Bandwitdh: 1.5Mb |
| 17:27.56 | drmessano | Last thing I want is another "server" to admin |
| 17:28.08 | drmessano | Give me a red toaster firewall/router/NAT/thingo any day |
| 17:28.22 | Naikrovek | ys |
| 17:28.23 | Naikrovek | yes |
| 17:29.01 | [TK]D-Fender | b0gatyr: that is a pure PRI T1. No ata. |
| 17:29.04 | [TK]D-Fender | data |
| 17:29.29 | drmessano | That's the ONLY thing I know I am stuck with on my Asterisk install.. I am taking on the maint of the hardware too.. So if we lost a HD, fan, stick of RAM, etc.. it's yet another server I am adminung |
| 17:29.34 | drmessano | admining* |
| 17:29.56 | drmessano | Easier to just tell people "it's broken, smells funny.. FIX IT" |
| 17:30.09 | citywok | drmessano: especially if you are installing appliance sized pbx's for small offices. |
| 17:30.38 | *** join/#asterisk timahvo1 (~rogue@41.223.57.75) |
| 17:31.06 | drmessano | "Push the handle.. did bread pop out?" "No" "Ok, toaster's busted" |
| 17:33.56 | citywok | what does everybody use when installing a pbx for a small office (25 people) that they aren't there to physicall deal with the device? Appliance like the AA50? Something else? |
| 17:34.38 | carrar | put it in a datacenter remotely |
| 17:34.40 | b0gatyr | [TK]D-Fender: Ok, I'll ask them to re-quote me for a data/voice.. how is the relevant to the question though? I just simply would like to know how to achieve a set up where I get voice for my asterisk box and data (internet for all my hosts) routed through this T1. |
| 17:34.59 | citywok | why would you do that? then they have to pay for something else. lol. |
| 17:35.00 | [TK]D-Fender | b0gatyr: the T1 you described ONLY carries VOICE |
| 17:35.16 | Naikrovek | drmessano: virtualize that phone server. modern hardware will handle it just fine |
| 17:35.17 | carrar | at least their phone systems works when their power is out |
| 17:35.20 | [TK]D-Fender | b0gatyr: When IT only carries voice it DOESN'T carry data therefor there is nothing you can do about that fact |
| 17:35.21 | Naikrovek | then you don't have hardware to worry about |
| 17:35.30 | Naikrovek | unless you have telco hardware |
| 17:35.34 | Naikrovek | which everyone in here seems to have |
| 17:35.35 | [TK]D-Fender | b0gatyr: So you need it to do something different |
| 17:35.48 | citywok | i suppose if you use canreinvite and let the phones in the office talk to each other directly you can get away with it. |
| 17:36.06 | drmessano | Naikrovek: Virtualizing it won't help me.. it's still another box to admin |
| 17:36.19 | Naikrovek | true |
| 17:36.21 | citywok | drmessano: but it does make the hardware problem somebody elses insetad of yours. |
| 17:36.36 | Naikrovek | won't have to worry about hard drive or memory |
| 17:36.37 | drmessano | How so? |
| 17:36.43 | Naikrovek | but will have to worry about OS |
| 17:36.57 | citywok | yea but it's set it & forget it. once you have it build you never touch it again. |
| 17:37.02 | b0gatyr | [TK]D-Fender: sure, but this was only a quote. I'll get a data/voice shortly.. sorry , but I'm not following you, is the set up I asked for possible or not? |
| 17:37.09 | drmessano | Are we talking about virtualizing it, or getting it HOSTED? |
| 17:37.10 | citywok | no updates needed, if it aint broke dont fix it. |
| 17:37.24 | Naikrovek | virtualizing it in house |
| 17:37.28 | [TK]D-Fender | citywok: Go pay for a hosted service then and they will take care of everything for you |
| 17:37.31 | drmessano | Ok, again |
| 17:37.34 | Naikrovek | if you need to back it up, just back up the virtual hard drive |
| 17:37.41 | Naikrovek | yeah it's another OS to manage, i agree |
| 17:37.54 | citywok | [TK]D-Fender: yea not a bad idea. that way if it goes down they wont call me and i wont have to drive 50 miles to fix it :P |
| 17:38.02 | Naikrovek | but there's no hardware dedicated to you, you share it all with other people and someone else yet will have to worry about upgrades and maintenance |
| 17:38.07 | drmessano | If I DON'T HAVE hardware to run it on, and I need to put another box in for it.. it's still MY problem.. Clouds don't magically appear |
| 17:38.18 | Naikrovek | aah |
| 17:38.22 | carrar | yes they do |
| 17:38.23 | Naikrovek | i missed the "no existing hardware" part |
| 17:38.35 | [TK]D-Fender | drmessano: Dunno... I don't ask for clouds and my summer just sorta magically sucked... |
| 17:38.35 | citywok | yea i'm replacing a pbx for a 25 seat non profit |
| 17:38.36 | carrar | add water |
| 17:38.37 | Naikrovek | so you still have to get SOMETHING |
| 17:38.46 | citywok | starting with nothing at all :) |
| 17:39.00 | citywok | and i'm not a huge fan of the AA50 i used for the last PBX replacement. |
| 17:39.08 | [TK]D-Fender | AA50 = dead end |
| 17:39.19 | leifmadsen | AstLinux on a Soekris? |
| 17:39.34 | drmessano | I have servers, but nothing I am going to dedicate to virtualization.. Shit's like 4 or 5 years old |
| 17:39.35 | citywok | They've been helping me out now with the bug that caused it to blow up every couple days. it only took 9 months and me complaining about it in here to get this far though. |
| 17:39.44 | [TK]D-Fender | leifmadsen: I'd sooner do mini-itx.... |
| 17:39.53 | leifmadsen | well whatever your hardware of choice is |
| 17:39.55 | drmessano | Atom is the bomb |
| 17:39.56 | [TK]D-Fender | leifmadsen: Real processor, et |
| 17:39.59 | leifmadsen | I don't really care |
| 17:40.05 | citywok | 25 people dont need a real processor lol |
| 17:40.09 | [TK]D-Fender | Atom is quite viable.. all depends on the platform |
| 17:40.15 | leifmadsen | *facepalm* |
| 17:40.26 | [TK]D-Fender | citywok: Soekris is a 500mhz P3 :p |
| 17:40.28 | *** join/#asterisk dan__t (~dant@vpn.withparity.net) |
| 17:40.33 | dan__t | 'morning. |
| 17:40.35 | [TK]D-Fender | citywok: I think you could do a LITTLE better :p |
| 17:40.38 | [TK]D-Fender | sorry... P5 |
| 17:40.50 | citywok | What's in the AA50? lol |
| 17:40.53 | [TK]D-Fender | err.. 586 |
| 17:40.55 | [TK]D-Fender | blarg |
| 17:41.05 | [TK]D-Fender | citywok: Blackfin embedded |
| 17:41.39 | dan__t | Anyone using any sort of video support with asterisk, like h.264 or MPEG4 or anything like that? I haven't started on this yet, just wanted to get some feedback and see how well it works. I'm trying to come up with a solution to do video conferencing for the big guys at our company, and for the sake of a new project and doing something new, I wanted to do video with Asterisk, rather than something like Jabber |
| 17:41.40 | citywok | granted the aa50 i have only runs like 15 people. lol. |
| 17:41.42 | drmessano | My office install is going to be insanely over spec'd.. Only because I have a spare server that's been fired up once, and it won't cost me anything to use it. 50 extensions, dual 2.4 GHZ quad core CPUs, 6 GB RAM, 1TB RAID 5 |
| 17:41.52 | drmessano | lol |
| 17:42.02 | leifmadsen | ahhh, much better |
| 17:42.17 | citywok | drmessano: i'm in somewhat the same boat. Dual 3.46 Xeons, 8gb of ram, dual 136gb 15k scsi. lol. |
| 17:42.17 | drmessano | a $1000 box for Asterisk would cost me $1000.. this is "free" |
| 17:42.41 | citywok | i had a pair of them, they run my asterisk setup lol. no cost! :) |
| 17:42.44 | [TK]D-Fender | dan__t: * doesn't "video conference". 2 devices can pass video however |
| 17:42.50 | [TK]D-Fender | dan__t: but * doesn't "mix" |
| 17:43.07 | *** join/#asterisk ukine (~ukine@14-145.97-97.tampabay.res.rr.com) |
| 17:43.37 | dan__t | Oh, ok. |
| 17:43.44 | dan__t | Yeah I don't even know how it would work, honestly. |
| 17:44.00 | drmessano | Yeah, my day job can be like pulling teeth sometimes.. $1000 was all the difference on a new PBX. Not like I need the extra server.. I have 25 users on 2 file servers of the same spec that I am only going to consolidate later |
| 17:44.38 | drmessano | I could put 25 users on a GX620 running Ubuntu |
| 17:44.40 | citywok | lol. somebody spent wayyy to much money there. |
| 17:44.47 | drmessano | I did :) |
| 17:45.02 | citywok | i have 50 users on a P3 1.13ghz as a file server. Compaq DL320 |
| 17:45.27 | citywok | it's 7,000 miles away from me and still going strong. no sense in screwing it up lol. |
| 17:45.55 | drmessano | Actually, we have 3 locations right now.. and I had planned to put a server at each.. But one building now has 5 users, so they are fine syncing across the WAN.. deployed the other 2, but we're consolidating in a few months, and I won't need both servers in production |
| 17:46.20 | dan__t | Alright, thanks for the help, [TK]D-Fender |
| 17:46.26 | drmessano | So I may actually get another box to throw Ubuntu on for a "misc projects" box |
| 17:47.07 | drmessano | Know any good Torrent clients with a WEB UI? |
| 17:47.16 | drmessano | I KID I KID, LOGGER BOT |
| 17:47.35 | drmessano | <no-archive> damnit |
| 17:47.40 | [TK]D-Fender | drmessano: Actually I was looking for jsut that for my server.. |
| 17:47.50 | [TK]D-Fender | drmessano: Ubuntu relase next month... |
| 17:47.52 | drmessano | I'd actually like that for home |
| 17:48.11 | dan__t | [TK]D-Fender, turns out actually they just want 1:1 video right now, not so much conferencing |
| 17:48.27 | dan__t | telepresence of the conference rooms. From what I can tell, Asterisk can help me out. |
| 17:48.30 | dan__t | I'm going to start hacking on it. |
| 17:48.31 | drmessano | You know, for downloading Linux ISO's.. Not movies, games, pr0n, or music |
| 17:48.35 | Naikrovek | dan__t: that's possible now with asterisk, |
| 17:48.53 | Naikrovek | dan__t: but it won't do .. video conferencing. 8 people in a room, you won't see all other participants |
| 17:48.54 | dan__t | Yep. |
| 17:48.59 | dan__t | Yep. |
| 17:49.29 | Naikrovek | seems like it could be easy to do though, ffmpeg is open source, vlc is open source, surely they have the chutzpa (sp?) to do something like that between them |
| 17:49.59 | dan__t | See this is all new to me, I don't even know what I'd use for a client. |
| 17:49.59 | drmessano | Naikrovek: I am sure it already exists.. Need to JFGI I guess |
| 17:50.12 | Naikrovek | in the year 3000 someone will get clever and use GPUs for that kind of heavy lifting. |
| 17:50.18 | Naikrovek | but this is only 2010 |
| 17:50.52 | Katty | or so you think. |
| 17:51.53 | dan__t | How do you figure VLC working with this, Naikrovek? |
| 17:52.14 | carrar | dan, http://www.gnugk.org/video-conferencing.html |
| 17:52.17 | Naikrovek | well it has all that video processing stuff in it |
| 17:52.36 | Naikrovek | surely it could be made to combine video signals in a "video wall" type format then reencode it and send it to phones |
| 17:53.23 | Naikrovek | i imagine it would be a few calls to some already-written .so somewhere |
| 17:53.41 | WIMPy | "make it so!" |
| 17:53.51 | Katty | engages |
| 17:54.02 | Naikrovek | sadly I know only Java and C#. Not C/C++ or anything else |
| 17:54.26 | Naikrovek | if Asterisk were written in Java, I'd have had it written a year ago. |
| 17:54.33 | Katty | Naikrovek: you mean you know D flat. |
| 17:54.41 | Naikrovek | lol |
| 17:54.42 | Naikrovek | sure |
| 17:54.51 | drmessano | I sing in J |
| 17:55.16 | Katty | do they also serve food? |
| 17:55.26 | Katty | or just drinks |
| 17:55.35 | dan__t | I think I'd want/need more of a SIP client. |
| 17:55.50 | dan__t | Sure, I might be avle to get VLC to act as a client for the call or whatnot... |
| 17:58.16 | *** join/#asterisk lvlolvlo (~lvlolvlo@unaffiliated/lvlolvlo) |
| 17:58.26 | Naikrovek | ... |
| 17:58.38 | Naikrovek | dan__t: ffmpeg or vlc would do the muxing on the SERVER |
| 17:58.53 | Naikrovek | you'd still use your hard/soft video phone |
| 17:59.25 | Naikrovek | the same way app_meetme does muxing of the audio in meetme rooms then sends everyone the proper audio |
| 18:00.02 | Naikrovek | the server handles the mixing and reencoding and rebroadcast to conf participants |
| 18:00.12 | drmessano | You could also use the bridging framework in 1.6.2.x |
| 18:00.22 | drmessano | (and above) |
| 18:00.44 | drmessano | That would probably be more the awesome |
| 18:01.04 | Naikrovek | i would be surprised if you couldn't make a few api calls to some already-written video processing library and do what needed to be done |
| 18:01.07 | *** join/#asterisk Mhaddog_ (~Mhaddog_@adsl-074-186-011-199.sip.mia.bellsouth.net) |
| 18:01.20 | dan__t | yep. |
| 18:01.23 | Naikrovek | people always think "oh asterisk can't do that, i'll use something else" rather than "how hard would it be to make asterisk do this" |
| 18:01.28 | Naikrovek | answer is probably "not very damn hard" |
| 18:01.38 | Kobaz | asterisk can do anything |
| 18:01.42 | Kobaz | if you want to write the code for it |
| 18:01.52 | dan__t | I understand. I've not actually produced much with Asterisk, but hacked on it enough to be able to establish little proofs of concept here and there. |
| 18:02.04 | Naikrovek | my bicycle can mine coal, if you put coal mining equipment on it.... ................ |
| 18:02.17 | dan__t | exactly |
| 18:02.18 | dan__t | haha |
| 18:02.25 | Naikrovek | i need to read about this bridging framework |
| 18:08.34 | Naikrovek | i'm asking in #ffmpeg about this combining signals thing |
| 18:09.43 | *** join/#asterisk speedy (~speedy@89.203.106.75) |
| 18:12.47 | Naikrovek | that's a dead channel so far heh. 160 people in there and no movement at all |
| 18:13.01 | dan__t | viva la porno |
| 18:13.08 | Naikrovek | lol! |
| 18:13.15 | Naikrovek | maybe the're all pooping |
| 18:13.18 | dan__t | 98% of the people in there are "webmasters" that transcode video |
| 18:13.19 | *** part/#asterisk andresmujica (~andresmuj@ubuntu/member/andresmujica) |
| 18:13.23 | dan__t | seriously. i used to be one of them. |
| 18:13.30 | dan__t | Bad business to be in heh |
| 18:13.34 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 18:14.09 | *** join/#asterisk andresmujica (~andresmuj@ubuntu/member/andresmujica) |
| 18:14.11 | Naikrovek | seems like a money making business to me |
| 18:14.17 | Naikrovek | there are so many of them |
| 18:14.36 | *** part/#asterisk andresmujica (~andresmuj@ubuntu/member/andresmujica) |
| 18:14.36 | Naikrovek | maybe i'll ask in ffmpeg-devel |
| 18:15.43 | dan__t | It is saturated. |
| 18:16.00 | dan__t | I've worked for several companies in the past that have catered to that business. |
| 18:16.08 | Naikrovek | crazy |
| 18:16.20 | dan__t | Used to be lots of money in it, that's why I started hosting it. |
| 18:16.23 | Naikrovek | why do they all hang out and not say anything |
| 18:16.43 | dan__t | because ffmpeg glows with "cool", and they want to bask in it. |
| 18:16.54 | Naikrovek | cripes |
| 18:17.54 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 18:19.55 | *** join/#asterisk kristianpaul (~kristianp@unaffiliated/kristianpaul) |
| 18:20.55 | *** join/#asterisk timahvo1 (~rogue@41.223.57.74) |
| 18:20.59 | kristianpaul | Hello, how i can restrict ie a extension A just can calll to B and no more.. |
| 18:21.19 | dan__t | Well. Aside from the video thing, I want a neat project with Asterisk. |
| 18:22.13 | *** join/#asterisk oej (~olle@2001:470:1f15:d79:225:ff:fe44:74ec) |
| 18:22.59 | Corydon76-home | kristianpaul: stick it in its own context |
| 18:25.07 | Naikrovek | dan__t: ah avisynth can do it out of the box with two inputs "stack"ing into 1 output |
| 18:25.25 | Naikrovek | wonder if that is available on linux, the api i mean |
| 18:27.15 | Naikrovek | ah |
| 18:27.20 | Naikrovek | mencoder does it too, apparently |
| 18:27.23 | Naikrovek | still looking for that code |
| 18:27.31 | Naikrovek | it's open source and available on linux |
| 18:29.23 | *** part/#asterisk Natureshadow (~Nik@p5B028C78.dip0.t-ipconnect.de) |
| 18:31.54 | dan__t | I feel like S. I'm going home. |
| 18:32.02 | Naikrovek | later |
| 18:33.22 | kristianpaul | Corydon76-home: but an extension can have multiples contexts? |
| 18:33.58 | Corydon76-home | kristianpaul: no, an extension has one context |
| 18:34.11 | Corydon76-home | but contexts may be included in other contexts |
| 18:34.30 | kristianpaul | ok |
| 18:34.37 | kristianpaul | interesting :) |
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| 18:52.18 | Naikrovek | if someone who knew C and a bit about video could write a filter for ffmpeg, asterisk could use libavcodec to stack multiple video calls together into a vid_meetme or something |
| 18:53.24 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 18:53.45 | Naikrovek | the existing vid filters for ffmpeg are pretty light, the biggest is only 500 lines of code, including comments and whitespace |
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| 18:59.25 | bmoraca_work | man i am out of touch with PHP |
| 18:59.34 | Naikrovek | heh |
| 18:59.41 | bmoraca_work | but, woo! figured out how to do pagination on the external directory of a Cisco 7940 phone! |
| 18:59.41 | Naikrovek | hard to be IN touch with PHP |
| 18:59.54 | bmoraca_work | should have said "out of practice" i suppose |
| 18:59.59 | Naikrovek | php is like a crazy author. lots and lots of stuff, none of it consistent, but the words do actually form sentences |
| 19:00.00 | bmoraca_work | been doing too much coldfusion |
| 19:00.59 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 19:01.00 | bmoraca_work | obscure reference on google books lead me down the path...it uses the HTTP refresh header to find the next page of the directory...ick |
| 19:01.01 | Naikrovek | php has crap like "sort_array" then also "arraySort" and "array_sort" which all do slightly different things |
| 19:01.29 | Naikrovek | i'm going by memory, that may not be an accurate example |
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| 19:06.22 | *** join/#asterisk t_dot_zilla (~chatzilla@firewall-a.buf.ny.i-evolve.net) |
| 19:07.08 | t_dot_zilla | we are having a problem where our TNT is sending BYE after 60 seconds during voicemail messages |
| 19:09.04 | t_dot_zilla | why would a UA send multiple INVITES |
| 19:10.59 | [TK]D-Fender | t_it wants you to feel very welcomed. |
| 19:12.48 | t_dot_zilla | TK-ur funny...do you get ur kicks off being a sarcastic dick on here |
| 19:14.52 | bougyman | t_dot_zilla: because it's not getting an ACK would be one reason. |
| 19:15.25 | [TK]D-Fender | t_dot_zilla: you should have sprung for the snese of humour optins when signing in ;) |
| 19:15.30 | [TK]D-Fender | option* |
| 19:15.45 | Benwa | what's the s in : 'exten => s,1,Wait(1)' ? |
| 19:16.55 | SuPrSluG | start |
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| 19:17.42 | [TK]D-Fender | ~stdextens |
| 19:17.42 | infobot | [~stdextens] The "s" Standard Extension : Where a call goes to when * does not know the destination of the call. Ex : Calls coming in on FXO ports (no DID), or from an ITSP that doesn't specify or where it was not set in the REGISTER line, or FXS port goes off-hook and "immediate=yes" in zapata.conf. "s" is also used to make IVRs & macros. |
| 19:17.54 | [TK]D-Fender | benGo read up on "Asterisk Standard Extensions" |
| 19:18.18 | [TK]D-Fender | Benwa: Also it could be ANYTHING technically, jsut that it is specifically referenced by certain things. |
| 19:26.16 | *** join/#asterisk jhirley (~jhirley@c-75-74-13-194.hsd1.fl.comcast.net) |
| 19:27.00 | jhirley | anyone out there using voip.ms having issues today ? |
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| 19:30.34 | p3nguin | What kind of issues? |
| 19:32.45 | jhirley | as of about 1300 all my resistration attempts are timing out. |
| 19:32.57 | leifmadsen | resistration is futile |
| 19:34.31 | thehar | *force* |
| 19:36.30 | jhirley | registration is also futile. |
| 19:36.44 | jhirley | what do you mean by "force" ? |
| 19:37.00 | thehar | s/is/are/ |
| 19:38.06 | t_dot_zilla | should asterisk ever receive sip:<null> request? |
| 19:38.18 | t_dot_zilla | sip:<null>@i.p.a.d.ress. |
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| 19:39.51 | [TK]D-Fender | t_dot_zilla: depends. What kind of request, and what is sending it? |
| 19:46.33 | t_dot_zilla | UA is sending ACK |
| 19:46.54 | t_dot_zilla | w/ request-line <null>@asterisk.ip |
| 19:54.30 | Katty | heeeeeeeeeellllllllllllllooooooooooo nurse |
| 19:55.48 | leifmadsen | Katty++ |
| 19:56.15 | Katty | leifmadsen++ |
| 19:58.23 | underdog | t_dot_zilla: the big question is "what has changed"? if it was working before then someone has changed something somewhere |
| 19:58.35 | *** join/#asterisk defsdoor (~andy@plingit.gotadsl.co.uk) |
| 19:58.55 | leifmadsen | I am upgrading my asterisk system to 1.8! |
| 19:58.57 | t_dot_zilla | underdog: we think the problem has always been there |
| 19:59.31 | underdog | "think"? |
| 19:59.33 | t_dot_zilla | someone just brought it to our attention |
| 20:00.00 | t_dot_zilla | basically our TNT is hanging up on voicemails after excatly 60 seconds, it only happens w/ this specific TNT |
| 20:00.11 | leifmadsen | sounds like a possible bug in the TNT then |
| 20:00.15 | t_dot_zilla | yes |
| 20:00.24 | underdog | TNT is explosive |
| 20:00.28 | t_dot_zilla | the packet capture i have tells me that as well |
| 20:00.42 | leifmadsen | asterisk probably shouldn't get sip:<null>@a.b.c.d, which would indicate a problem with the far end |
| 20:00.55 | leifmadsen | file an issue with the manufacturer then |
| 20:01.09 | t_dot_zilla | right, it looks like TNT and asterisk arent completing handshake |
| 20:01.35 | t_dot_zilla | the strange thing is, it only happens during a voicemail, during normal calls they do not end after 60seconds |
| 20:08.07 | *** part/#asterisk pwnguin (~jldugger@ubuntu/member/jldugger) |
| 20:25.28 | t_dot_zilla | should the address in the request-line ever change during handshake? |
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| 20:48.49 | leifmadsen | System uptime: 5 weeks, 2 days, 9 hours, 11 minutes, 23 seconds, 600 calls processed. And now upgrading to 1.8! |
| 20:50.34 | leifmadsen | well that was easy :) |
| 20:57.08 | *** join/#asterisk myster (~myster@207.148.172.210) |
| 20:57.31 | lirakis | is away: catching a train |
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| 21:08.21 | dwayne | lirakis, the best way to catch a train is to stand on the tracks and let it come to you |
| 21:10.50 | WIMPy | Maybe it's more like the other way round? |
| 21:19.51 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:23.58 | Katty | fender bender. |
| 21:24.06 | [TK]D-Fender | z0mg |
| 21:27.21 | jhirley | I would rather go on a "bender" .. bottoms up.. |
| 21:27.26 | dwayne | my wife is by far TheWorstDriverEver(TM) |
| 21:28.13 | jhirley | in this channel people are going to as if that is a tapi 32 or 64 bit driver. |
| 21:28.31 | jhirley | (TM) Dot Com even. |
| 21:29.31 | dwayne | I just spent 10 minutes giving directions and for some reason she kept driving in circles while asking which way to go and ignoring the responses |
| 21:30.50 | WIMPy | My wannabe-boss does so with his business strategies. |
| 21:31.09 | jhirley | thats not a driving thing, I would say it is a woman thing. (Don't hate me Katty ). My wife , Mother and Sister all do the same thing. Oh my sister with argue with you about not know what I am talking about even when she is the one calling because she is lost. |
| 21:31.09 | dwayne | WIMPy, everything is the highest priority? |
| 21:31.38 | WIMPy | Not so much, more like a new Plan every day. |
| 21:31.54 | dwayne | oooh .... even better |
| 21:32.00 | WIMPy | And constantly ignoring the prerequisites. |
| 21:32.46 | WIMPy | Liek teh new office that got internet today, so ppl can work now. |
| 21:32.47 | *** join/#asterisk deonv (~adium@pixfirewall.itn.com.na) |
| 21:33.00 | WIMPy | Except that there isn't a single PC yet. |
| 21:33.21 | [TK]D-Fender | jhirley: She wasn't really asking for directions, she just wanted to know you support her and for you to feel superficially useful to her :) |
| 21:33.22 | WIMPy | But he's already spent several days looking for the right ones. |
| 21:36.01 | [sr] | hi WIMPy |
| 21:36.48 | WIMPy | Hi [sr] |
| 21:36.53 | [sr] | :) |
| 21:38.00 | underdog | empowered the wife and bought her a phone that she can looup her own maps/directions with |
| 21:39.27 | jhirley | Fender: I am the wrong person to call for support. My average call durration is between , 5 to 15 seconds. While my sister can talk for an hour, I once dropped my cellphone and it feel behind the seat. I had to wait till I stopped to get it. The call was still active, 1st surprise, she was still talking, 2nd surprise. |
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| 21:49.28 | Beltechs | Hello Im using *1.6 and I have an extension that is not receiving calls and when voicemail picks up it picks up as if the person was on the phone |
| 21:49.53 | Beltechs | sip set debug 6471 http://pastebin.com/RvMa1tMX |
| 21:49.55 | Katty | i'm sorry, i don't understand. could you be a little more vague? |
| 21:50.27 | Beltechs | i could but it might go over your head again |
| 21:50.46 | Katty | k'then |
| 21:51.37 | beardy | Hey Katty |
| 21:53.26 | Kobaz | i can be vague |
| 21:54.17 | Beltechs | what part dont u understand |
| 21:54.19 | Beltechs | ? |
| 21:54.59 | Beltechs | <PROTECTED> |
| 21:56.54 | Katty | hi beardy (= |
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| 21:57.01 | Katty | hi p3nguin |
| 21:57.08 | p3nguin | Hello again. |
| 21:59.08 | [TK]D-Fender | Beltechs: You could try telling us what you're calling, where it is and where to start looking in that 1000 lines of PB you dumped for us |
| 21:59.56 | [TK]D-Fender | Beltechs: and you seem to want to enable SIP debug but never did |
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| 22:07.08 | bmoraca_work | why does the dbget AMI command have to be so obtuse? why can't I just have the answer in the first response message? |
| 22:14.01 | Kobaz | heh |
| 22:14.18 | Kobaz | yeah some of that stuff is kind of annoying |
| 22:15.56 | WIMPy | What difference does it make? There could come something inbetween anyway. |
| 22:16.44 | Kobaz | why have the extra needless verbosity |
| 22:17.03 | Kobaz | send command... yes the command was successful, but we wont give you the result... here is the result |
| 22:17.25 | WIMPy | "internals"? |
| 22:17.30 | Kobaz | just send the result with the success message |
| 22:18.43 | Kobaz | it's not totally needed |
| 22:18.44 | Kobaz | here's the code |
| 22:18.45 | Kobaz | astman_send_ack(s, m, "Result will follow"); astman_append(s, "Event: DBGetResponse\r\n" |
| 22:18.59 | Kobaz | just comment out the "Result will follow" |
| 22:19.15 | Kobaz | it will break clients that expect that message though |
| 22:20.00 | bmoraca_work | to delete an astdb key, would i issue a DBPut with the Val parameter blank? |
| 22:20.12 | bmoraca_work | (over AMI, that is) |
| 22:20.19 | Kobaz | or you could use the DBDel command |
| 22:20.36 | bmoraca_work | AMI doesn't have a DBDel command, from what I can see |
| 22:20.49 | Kobaz | yeah it does, in 1.6.0 anyway |
| 22:20.58 | bmoraca_work | i'm not in 1.6 |
| 22:21.12 | Kobaz | you can backport it from 1.6 |
| 22:21.45 | bmoraca_work | i'm sure I can, but I'm not really wanting to customize a customer's Asterisk box |
| 22:22.14 | exothermc | anyone here use adhearsion? |
| 22:22.27 | Kobaz | never too late to start a local branch |
| 22:23.09 | *** join/#asterisk ghoti (~paul@38.117.126.254) |
| 22:23.14 | bmoraca_work | worse comes to worse, I can always issue a system call "asterisk -rx 'database del ...'" |
| 22:23.28 | Kobaz | predial |
| 22:23.29 | Kobaz | er |
| 22:23.39 | ghoti | Anyone know if the work to port DAHDI to FreeBSD is on schedule to be completed this month, and if it'll be possible to add it to an existing 8-STABLE system once it's done? |
| 22:23.41 | leifmadsen | Uhh... does AMI have a Command instruction? |
| 22:23.54 | leifmadsen | ghoti: uh... it's been available for quite some time |
| 22:24.04 | bmoraca_work | leifmadsen: yes, but it's mostly worthless |
| 22:24.18 | leifmadsen | ghoti: http://downloads.asterisk.org/pub/telephony/dahdi-freebsd/ |
| 22:24.56 | exothermc | if I use the connect method from this class http://rubydoc.info/github/adhearsion/adhearsion/master/Adhearsion/VoIP/Asterisk/Manager/ManagerInterface how do I check if the connection is still valid? |
| 22:25.04 | ghoti | leifmadsen: but that's incomplete. The FreeBSD foundation committed funds to have a guy finish it this summer. September 2010 was the target completion date. |
| 22:25.26 | leifmadsen | ghoti: I'd check with that guy then |
| 22:25.33 | leifmadsen | ghoti: I don't understand what is incomplete |
| 22:26.21 | leifmadsen | ghoti: I don't know how we would know anything about that |
| 22:26.27 | ghoti | leifmadsen: http://www.freebsdnews.net/2010/07/19/freebsd-dahdi-driver-project-announcement/ fyi.. |
| 22:26.41 | ghoti | There might be someone paying attention to DAHDI development in this channel. |
| 22:26.59 | ghoti | seems just as likely as someone paying attention to FreeBSD development in #freebsd, no? :) |
| 22:27.37 | leifmadsen | ghoti: ya, but if a particular person is developing it and you're expecting him to answer, I'd just ask him. The dahdi-freebsd stuff is unsupported by the usual suspects |
| 22:28.17 | ghoti | that is indeed a pity. it would be nice for projects to be a little more OS agnostic. but okay, I'll look elsewhere for this info. |
| 22:28.20 | leifmadsen | ghoti: the only person who has been working on that and committing to it is Max Khon |
| 22:28.45 | leifmadsen | ghoti: there are only so many development resources to go around |
| 22:29.09 | leifmadsen | ghoti: and I have seen a release created as late as September 4, 2010 |
| 22:29.10 | ghoti | ain't that the truth. as I said, I thought someone closer to that line of dev might be in the channel. |
| 22:29.22 | leifmadsen | ghoti: nope, the only person working on that is Max |
| 22:29.40 | leifmadsen | I didn't even know there was a "deadline" for anything |
| 22:30.47 | ghoti | I think it was more of a "target" than a deadline. I'll ask on a freebsd mailing list, or see if I can figure it out myself from the SVN repo. |
| 22:31.05 | ghoti | (figure out what the patches will apply to, that is.) |
| 22:33.58 | leifmadsen | ghoti: ya because the only person working on that branch is Max since it is an unsupported branch by the DAHDI team |
| 22:34.11 | leifmadsen | ghoti: (and there is only 1 or 2 developers on that team afaik) |
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| 23:12.37 | bmoraca_work | woo...finished my Cisco XML services for freepbx...call forwarding, call waiting, and dnd are setable from the phone using menus! woo! |
| 23:12.50 | bmoraca_work | also, paged local directory! |
| 23:43.51 | eugeneoden | anyone have any idea what the intention is behind sip_pvt->redircodecs? it appears to be largely unused in 1.6.2.13 |
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