| 00:05.32 | *** part/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 00:08.14 | *** join/#asterisk apardo (n=apardo@211.64.220.87.dynamic.jazztel.es) |
| 00:17.41 | *** join/#asterisk Teln1100A (i=hello123@bas2-toronto12-1168022952.dsl.bell.ca) |
| 00:22.36 | *** join/#asterisk xezz (n=phob@trust-it.gr) |
| 00:24.30 | xezz | hello, im trying to call Uk, i have a trunk g0 and outbound routes pattern set to 8|0044. but i get the all circuits are buzy now message, any idea ? |
| 00:29.58 | *** part/#asterisk renier (n=renier@24.139.155.193) |
| 00:37.14 | *** join/#asterisk Qwell_ (n=north@pdpc/sponsor/digium/Qwell) |
| 00:37.14 | *** mode/#asterisk [+o Qwell_] by ChanServ |
| 00:45.57 | mistermocha | xezz: what kind of outbound trunk are you using? |
| 00:45.57 | mistermocha | analog? t1? voip? |
| 00:46.03 | xezz | analog |
| 00:46.44 | mistermocha | when you watch your call trace from the CLI, do you see "Dialing Zap/g0/0044xxxxxxxx |
| 00:46.49 | mistermocha | replace x's with your number |
| 00:47.03 | wishes | finally i have my custom messag working :D , on a side note wengophone = shit |
| 00:47.21 | mistermocha | heh... with a name like wengophone |
| 00:47.28 | Mercestes | Do you have a group 0? |
| 00:48.31 | xezz | its here man |
| 00:48.33 | xezz | http://pastebin.ca/723763 |
| 00:48.37 | mistermocha | l |
| 00:48.38 | mistermocha | k |
| 00:49.26 | mistermocha | prepend a w in your dialplan |
| 00:49.35 | mistermocha | 8|w0044 |
| 00:49.53 | mistermocha | maybe even a few w's |
| 00:50.12 | xezz | why is that man ? |
| 00:50.15 | xezz | why w ? |
| 00:50.36 | mistermocha | w is a half-second wait |
| 00:51.04 | mistermocha | when switching analog lines, it can take a moment for a softswitch to connect and drop the voltage |
| 00:51.45 | mistermocha | if * starts sending digits before the voltage drops (gets a dial tone), then it won't send |
| 00:51.55 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 00:52.26 | mistermocha | Channel 0/1, span 1 got hangup request |
| 00:52.32 | mistermocha | that says a bit of it right there |
| 00:53.14 | xezz | well |
| 00:53.19 | xezz | im trying to insert this w |
| 00:53.30 | xezz | but it says wrong pattern |
| 00:53.42 | xezz | it doesnt seem to understand the pattern |
| 00:53.44 | xezz | with w |
| 00:55.02 | mistermocha | hmm... |
| 00:55.43 | xezz | im able to call cell phones land lines perfectly with |. |
| 00:55.53 | mistermocha | what does your dialplan context look like? |
| 00:56.12 | xezz | but when im trying to make an international call i make an pattern like this 8|0044 |
| 00:57.59 | *** join/#asterisk ManxPower (n=manxpowe@64.246.207.186) |
| 00:58.46 | mistermocha | put your context into pastebin |
| 00:59.07 | *** join/#asterisk PepOSX (n=pepOSX@190.72.149.163) |
| 00:59.45 | *** join/#asterisk mltlnx (n=mltlnx@74.73.54.147) |
| 01:02.04 | Freman | hey... is AEL2 included in 1.4? |
| 01:03.41 | Qwell_ | Freman: yes |
| 01:03.49 | Freman | cool thanks |
| 01:03.57 | xezz | no |
| 01:04.09 | xezz | it doesnt work |
| 01:08.35 | ManxPower | Freman: 1.4 is the first release with AEL2. |
| 01:11.01 | Freman | I must have a borked install at home then, cos some of the AEL2 stuff no good on it, but AEL works fine |
| 01:12.27 | codefreeze | Freman: give me samples; I've been doing AEL bugs the past week. |
| 01:12.53 | Freman | I havn't looked at it for ages codefreeze. |
| 01:13.17 | Freman | I've just convinced work to go with asterisk so I've been updating the internal wiki with information |
| 01:13.35 | codefreeze | I'd guess it'd be natural that AEL2 wouldn't go so well on 1.2; AEL2 has some new stuff AEL (1) didn't have. |
| 01:15.26 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 01:15.50 | syzygyBSD | when upgrading from 1.2.X to 1.4.11, is there any reason why asterisk would hang on playing silence? |
| 01:16.04 | Freman | my home machine is 1.4.x |
| 01:16.23 | Freman | Asterisk 1.4.2 |
| 01:21.57 | ManxPower | syzygyBSD: nothing in the upgrade.txt to explain the problem |
| 01:22.57 | ManxPower | Freman: did you try the most common solution to 1.4.x problems? |
| 01:23.45 | syzygyBSD | hmm, old silence files didn't work, had to upgrade them |
| 01:24.06 | syzygyBSD | makes me wonder what other files are broken |
| 01:24.48 | Freman | ManxPower, nah - it didn't bother me that much, 99% of my applciationw as done with perl on AGI. |
| 01:25.08 | ManxPower | Freman: the most common solution to 1.4.x problems is "upgrade to the latest 1.4.x". |
| 01:25.31 | ManxPower | 1.4.x was pretty rough at the beginning |
| 01:26.16 | Freman | that it was |
| 01:26.41 | Freman | my install is at the 'it works well enough, cbf doing more to it for now' |
| 01:27.21 | ManxPower | Reports of early 1.4 users seemed to indicate it would blow up if you looked at it wrong. Much like a Fainting Goat http://en.wikipedia.org/wiki/Fainting_goat |
| 01:28.15 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 01:30.09 | Freman | heh |
| 01:30.38 | Freman | it's on the "to upgrade" list of things... but there's a few dozen things in front, if it breaks in the mean time it'll happen faster |
| 01:32.54 | *** join/#asterisk fujin (n=aj@unaffiliated/fujin) |
| 01:35.29 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 01:37.03 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 01:38.46 | Freman | wow... just filled up the console with |
| 01:38.47 | Freman | http://yro.slashdot.org/yro/07/10/02/1830211.shtml <- rofl, patenting the checkbox... good on yah... |
| 01:38.50 | Freman | err |
| 01:38.52 | Freman | bad c/p |
| 01:38.56 | Freman | [Oct 3 11:38:26] WARNING[22504]: chan_sip.c:3625 sip_write: Asked to transmit frame type 256, while native formats is 0x4 (ulaw)(4) read/write = 0x4 (ulaw)(4)/0x4 (ulaw)(4) |
| 01:40.25 | *** join/#asterisk hunginday (n=hungtd@210.245.57.162) |
| 01:40.27 | [TK]D-Fender | Freman, 256 = G.729. Good odds you don't have any licenses |
| 01:40.48 | Freman | yeh, but I globally disallowed G.729 |
| 01:42.02 | [TK]D-Fender | Freman, pastebin it all.... |
| 01:42.06 | [TK]D-Fender | (less passwords) |
| 01:43.26 | *** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 01:44.03 | hunginday | Help me, I added a new module in asterisk-source\res folder. In this module I call some funtions in chan_sip.c. Asterisk is compiled successfully but when running it show error of missing funtion symbol, so that the funtions in chan_sip could not be called from my new module. So what i have to change Makefile or something to make it work??? |
| 01:44.25 | Freman | nah it's ok, I've been playing with Asterisk-GUI, I must have unset it while testing something |
| 01:44.26 | Freman | fixed now |
| 01:45.10 | Freman | just so happens that provider prefers G.729 |
| 01:46.46 | [TK]D-Fender | ....no comment |
| 01:48.12 | Freman | hehe (c: |
| 01:48.36 | *** join/#asterisk syzygyBSD_ (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 01:48.49 | Freman | I've been tinkering with asterisk for quite some time now, it's just when it throws me new and strange errors I get the "WTF's" |
| 01:49.10 | Freman | (Started in the early 1.2's) |
| 01:49.50 | Freman | we'll get some G.729 lics before we swing into production, atm I'm fighting to prove it viable so they don't go and install some POS proprietary system in the new offices |
| 01:52.12 | [TK]D-Fender | i suppose you have to start somewhere... don't cheap out on the hardware though.... |
| 01:52.18 | [TK]D-Fender | anyways... heading out for a bit... |
| 01:54.27 | Freman | I've talked bosses into getting PolyCom SoundPoint IP 430 phones, there's an 8 port digium card coming with 4 lines on it... and eventually the asterisk server will become a dual core system with 2 gigs of ram |
| 01:55.21 | flenders | Freman: which digium card? |
| 01:55.52 | flenders | get one with hardware echo canceller |
| 01:56.04 | flenders | TDM2400P with EC module |
| 01:56.11 | Freman | I couldn't convince them to go that far I'm afraid |
| 01:56.17 | flenders | or go for sangoma a200 with EC |
| 01:56.28 | Freman | however they will pay for the $15 software echo canceller |
| 01:56.30 | flenders | Freman: from personal experience, get the hardware EC |
| 01:56.34 | Freman | I know |
| 01:56.39 | Freman | I got the old 4port digium at home |
| 01:56.53 | flenders | I have 6 POTS lines on TDM400s and it is a BITCH to get rid of echo |
| 01:57.07 | Freman | I've managed to clean it up alot tho with software and fxotune |
| 01:57.27 | flenders | yeah, with single calls, probably always to the same numbers while testing |
| 01:57.30 | Qwell_ | Freman: which card do you have? |
| 01:57.37 | flenders | wait until you have the thing in production |
| 01:57.41 | Freman | hopefully this will be a "I told you so" situation and they'll just go get a sangoma (they're cheaper from memory) |
| 01:58.14 | Freman | TDM800 |
| 01:58.15 | Qwell_ | oh, tdm800p.. yea |
| 01:58.17 | Qwell_ | ~hpec |
| 01:58.17 | jbot | hpec is probably Digium's High Performance Echo Cancellation software - http://www.digium.com/en/products/software/hpec.php - Free for Digium cards under warranty; US$10 per channel otherwise. |
| 01:58.35 | Freman | cool |
| 01:58.37 | *** join/#asterisk AJaymn (n=Me@71-82-218-158.dhcp.mdsn.wi.charter.com) |
| 01:58.42 | Freman | didn't even know that existed |
| 01:59.06 | AJaymn | Anyone used the Linksys WIP300 wifi phone? |
| 01:59.40 | Freman | how's that sangoma card work with drivers? |
| 02:00.13 | flenders | installing a a101 was a lot easier than I thought |
| 02:00.31 | flenders | so my guess is that the a200 is not hard either |
| 02:00.41 | Qwell_ | meh, closed source drivers, and a patch to zaptel |
| 02:01.18 | *** join/#asterisk mltlnx (n=mltlnx@cpe-74-66-78-152.nyc.res.rr.com) |
| 02:01.43 | flenders | Qwell_: you work for digium, I wouldn't expect you to say sangomas are better |
| 02:01.47 | flenders | :D |
| 02:01.58 | Nugget | <homer> stupid flenders! </homer> |
| 02:02.14 | flenders | :D |
| 02:02.45 | Freman | Qwell: How's that hpec stuff work when we purchase from a different supplier? |
| 02:03.05 | Qwell_ | $10 per channel |
| 02:03.14 | Qwell_ | it works if it uses zaptel, but...meh |
| 02:03.43 | flenders | Qwell_: can you use HPEC on digium's PRI cards? |
| 02:04.10 | Freman | cos we only just purchased this card |
| 02:05.05 | Freman | IE: we just purchased the TDM800 through an aussie reseller, is it still going to cost us $10/chan or is it covered with the under warrenty statement? |
| 02:05.12 | Qwell_ | flenders: You can, but it's not really recommended |
| 02:05.26 | Qwell_ | Freman: You're covered |
| 02:05.43 | Qwell_ | flenders: just because it's so many channels.. echo can can be quite CPU intensive |
| 02:06.02 | Freman | heh, it says that right there on the page |
| 02:06.16 | *** join/#asterisk mordaunt (n=mordaunt@unaffiliated/mordaunt) |
| 02:06.52 | Freman | 8 chans - 3 ghz, 4 chans - 2.5, might get a lic for my single chan at home (tdm400) |
| 02:10.15 | flenders | Qwell_: so, with HPEC is it the same as having an EC on the card? |
| 02:10.33 | Qwell_ | well, no, a hwec isn't going to hit your cpu |
| 02:10.41 | flenders | apart from that |
| 02:10.52 | flenders | I mean, sound quality wise |
| 02:11.23 | Qwell_ | yeah, it's the same software on the new hwecs |
| 02:12.24 | Freman | HPEC on a single channel on my duron 1gig should be ok yeh? (c: |
| 02:12.48 | Qwell_ | duron? heh |
| 02:12.55 | Qwell_ | should be fine |
| 02:15.03 | flenders | well, I guess it makes sense to use HPEC on small installs, like 4 FXOs |
| 02:15.20 | flenders | the new server (even a dell SC440) will be the same price, anyway. |
| 02:15.47 | flenders | but the card with no hwEC is half the price with 4 FXO modules |
| 02:17.04 | Freman | yeh |
| 02:18.05 | flenders | Qwell_: noob question, do you need (or could have) EC on FXS channels? |
| 02:18.31 | Freman | Another noob question: How long is the warrenty period on TDM400's? (c: |
| 02:19.49 | Qwell_ | Freman: 2 years |
| 02:20.00 | Qwell_ | flenders: umm... |
| 02:20.07 | Qwell_ | I don't know, actually |
| 02:20.40 | Nugget | Wwhhaatt ddooeess eecchhoo ccaanncceellaattiioonn ddoo?? |
| 02:20.51 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 02:20.54 | flenders | cause, in theory, you wouldn't have echo on FXS (internal) channels, as they won't be too far from the server, does taht make sense? |
| 02:21.03 | Freman | hmmm LumenVox looks interesting... cept I run a Gentoo house |
| 02:25.24 | lisandropm | Hello! |
| 02:25.38 | lisandropm | I am getting this in the CLI: |
| 02:25.39 | lisandropm | [Oct 2 22:47:49] WARNING[10473]: chan_zap.c:11117 process_zap: Ignoring |
| 02:25.39 | lisandropm | switchtype |
| 02:25.50 | lisandropm | More info in: http://pastebin.ca/723846 |
| 02:25.58 | lisandropm | ¿any ideas that may help me? |
| 02:26.11 | Freman | Qwell_, is it permissable to point out a minor english glitch on the asterisk website? |
| 02:26.48 | lisandropm | ah, one more question: ¿does anyone knows how can I know if a Siemnes DIUS2 board is using css and hdb3? |
| 02:29.48 | ManxPower | lisandropm: all that means is that you cannot change the switch type on a reload and so it ignored any changes to that setting |
| 02:30.20 | lisandropm | ManxPower: greta, so that does not shows a real problem |
| 02:30.24 | lisandropm | *great |
| 02:30.26 | *** join/#asterisk ming_zym (n=ming_zym@124.254.53.2) |
| 02:30.49 | lisandropm | ok, then I do not why I do have that yellow alarm :-/ |
| 02:31.43 | ManxPower | lisandropm: that message has nothing to do with you yellow alarm |
| 02:31.59 | lisandropm | no, I will have to re-formule the question |
| 02:32.04 | ManxPower | yellow alarm is usually an issue at the telco end. |
| 02:32.23 | ManxPower | I would recommend you contact the telco and see what they say. |
| 02:32.39 | lisandropm | ManxPower: the telco would be a Hicom 300 (not E nor H) with a DIUS2 board |
| 02:32.48 | ManxPower | Cable problems *usually" generate a RED alarm |
| 02:33.13 | lisandropm | I checked the cables form the DIUS2 to the asterisk |
| 02:33.13 | ManxPower | lisandropm: never heard of it. |
| 02:33.30 | lisandropm | yes, that's one of my biggest problems :-) |
| 02:33.57 | lisandropm | I can't get _any_ information about the DIUS nor the Hicom that would explain me what I think I need |
| 02:34.00 | ManxPower | I suspect it is a DIUS2 config issue. |
| 02:34.06 | lisandropm | great |
| 02:35.52 | ManxPower | A yellow alarm basically means that the far end is not receiving any signal from you and that far end device is sending a yellow alarm, which you are receiving. |
| 02:36.22 | ManxPower | lisandropm: What type of cable are you using. I suspect you will need a T-1/E-1 crossover cable. |
| 02:36.27 | *** join/#asterisk Qwell_ (n=north@pdpc/sponsor/digium/Qwell) |
| 02:36.27 | *** mode/#asterisk [+o Qwell_] by ChanServ |
| 02:36.48 | ManxPower | If you are using an ethernet crossover cable, I can see how it might cause a yellow alarm, but I would have to look at the pinouts to be sure. |
| 02:37.10 | lisandropm | ManxPower: the dius2 has two 3-layer coaxial cables. Those connect to a ballon that transforms them into a RJ45 |
| 02:37.12 | Freman | Qwell, Did you get my PM? |
| 02:37.28 | Qwell_ | yeah |
| 02:37.31 | lisandropm | ManxPower: I checked the ballon with a signal generator and an oscilloscope |
| 02:37.32 | ManxPower | "router2" would be asterisk in your situation: http://www.juniper.net/techpubs/software/nog/nog-interfaces/html/t1-alarms16.html |
| 02:37.44 | Freman | cool (c: |
| 02:38.07 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 02:39.14 | lisandropm | ManxPower: more than that, I checked that each coaxial turns out in one of the two pairs available at an E1 line |
| 02:39.22 | ManxPower | lisandropm: Try either a standard ethernet cable between the ballan and Asterisk or try a crossover T-1 cable. See http://www.voip-info.org/wiki/view/crossover+T1+cable |
| 02:39.50 | ManxPower | for RJ-45 ports, T-1 and E-1 would be wired the same. |
| 02:40.17 | lisandropm | ManxPower: already did that. You only need to switch the coaxials in the baloon |
| 02:40.32 | lisandropm | That's one of the things I checked with the oscilloscope |
| 02:40.34 | AJaymn | has anyone created a "watchdog" or "heartbeat" monitor for asterisk? If it stops responding it could generate an email to the admin? |
| 02:40.45 | ManxPower | lisandropm: then it must be a config issue. |
| 02:41.15 | lisandropm | ManxPower: you mean from the HiCom? |
| 02:42.37 | *** join/#asterisk minkus (n=minkus@pool-71-182-38-220.clrkwv.east.verizon.net) |
| 02:50.46 | *** join/#asterisk andrew` (n=andrew@69-12-140-101.dsl.dynamic.sonic.net) |
| 02:52.05 | ManxPower | yes |
| 02:52.23 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 02:52.56 | lisandropm | ManxPower: than you _very_ much for your help and time :-) |
| 02:54.03 | *** join/#asterisk Nic-I (n=monster@bb220-255-41-253.singnet.com.sg) |
| 02:56.07 | Nic-I | hi .. is there a chatroom to ask question on digium cards? |
| 02:56.25 | *** join/#asterisk IgorG (n=FeedomPa@195.162.32.126) |
| 02:56.42 | fujin | lol |
| 02:56.43 | fujin | I doubt it |
| 02:56.49 | fujin | here is probably your best bet |
| 02:56.51 | fujin | or digium support |
| 02:57.12 | Nic-I | ohh ok |
| 02:57.18 | Nic-I | thanks |
| 02:57.46 | Nic-I | i am having a SIOCSIFFLAGS.. digium conflict with the built in eth0 |
| 02:57.53 | Nic-I | TDM PCI Master abort |
| 02:58.15 | fujin | get a better motherboard |
| 02:59.28 | lisandropm | Nic-I: have you tried disabling the on-board ethernet? |
| 03:01.39 | Nic-I | lisandropm trying it now |
| 03:02.19 | lisandropm | Nic-I: if it works, it may be cheaper to buy aanother ethernet card than a motherboard ;-) |
| 03:02.42 | lisandropm | goodbye!! |
| 03:03.35 | Nic-I | lisandropm i need 2 ethernet cards.. this is box is a router and there is only 3 PCI slots |
| 03:04.21 | *** join/#asterisk mltlnx (n=mltlnx@74.73.54.147) |
| 03:06.23 | flenders | Nic-I: get a dual ethernet card |
| 03:06.28 | flenders | intels are good |
| 03:06.51 | flenders | you can get dual ethernet cards on ebay very cheap these days |
| 03:07.27 | *** join/#asterisk IgorG (n=FeedomPa@195.162.32.126) |
| 03:12.27 | *** join/#asterisk mltlnx (n=mltlnx@74.73.54.147) |
| 03:14.17 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 03:14.17 | *** mode/#asterisk [+o russellb] by ChanServ |
| 03:16.40 | Nic-I | flenders thanks but i already has this board and trying to get it work with the TDM |
| 03:17.37 | flenders | try moving the tdm to a different slot |
| 03:17.47 | Nic-I | what is the package in linux which auto detect hardware? kadzu? |
| 03:21.58 | *** join/#asterisk _pepo_ (n=Pepo@190.10.187.20) |
| 03:22.17 | _pepo_ | Hi friends |
| 03:22.27 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 03:28.36 | *** join/#asterisk tengulre (n=tengulre@222.90.140.114) |
| 03:28.39 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 03:28.45 | tengulre | hi,all |
| 03:29.05 | tengulre | why not have asterisk-1.2 version on asterisk.org? |
| 03:30.05 | Qwell_ | tengulre: because it isn |
| 03:30.07 | Qwell_ | t supported |
| 03:30.27 | tengulre | Qwell_: can not supported what? |
| 03:30.44 | *** join/#asterisk ZX81 (n=matt@202.49.106.158) |
| 03:31.08 | ZX81 | hi all - anyone know what the cause of "my_zt_write: Write returned -1" filling the console followed by a machine crash? |
| 03:37.18 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:39.01 | ZX81 | http://pastebin.ca/723900 |
| 03:40.19 | ZX81 | I know the message is normally associated with missing interrupts causing and/or the jb for zap being empty/too small but why did the machine crash? |
| 03:41.02 | fujin | faulty hardwrae? |
| 03:41.34 | ZX81 | hmmm maybe |
| 03:41.48 | ZX81 | am taking another unit up there tomorrow |
| 03:42.12 | ZX81 | but its a 4 hour drive and I've been up three times so far! |
| 03:42.23 | ZX81 | just want to make sure I've covered everything :) |
| 03:42.57 | ZX81 | have a new server, a new switch, a new ups, new phones, my test butt, krone tools etc |
| 03:45.54 | *** join/#asterisk tim0123 (n=cash247@ppp-70-247-126-150.dsl.rcsntx.swbell.net) |
| 03:47.44 | _pepo_ | Hi friends |
| 03:47.50 | _pepo_ | I am working in a TELCO, we have a trouble with our very old Alcatel Voicemail system (and now we dont have support and worst this system was forgotten for Alcatel) |
| 03:48.30 | _pepo_ | I've used Asterisk for just small jobs, but I've proposed use it and tomorrow begins with the tests :) ... so |
| 03:49.17 | _pepo_ | we have a lot of users, how do I have to configure my server Asterisk to works like voicemail system if some PSTN call is turned in SIP beacause it goes through a Softswitch? |
| 03:54.57 | *** join/#asterisk bmg505 (n=leon@196.209.179.116) |
| 04:01.46 | _pepo_ | :( |
| 04:10.16 | MooingLemur | I just had someone sit in a conference bridge for more than 24 hours (probably left the phone off the hook). The conference is recorded. The monitor file reached 2 gigs. asterisk got repeated SIGXFSZ and it responded by rotating the logs 15000 times. :P |
| 04:16.03 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 04:28.40 | Snake-eyes | lol |
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| 04:51.07 | hunginday | Help me, I added a new module in asterisk-source\res folder. In this module I call some funtions in chan_sip.c. Asterisk is compiled successfully but when running it show error of missing funtion symbol, so that the funtions in chan_sip could not be called from my new module. So what i have to change Makefile or something to make it work??? |
| 04:51.32 | Strom_C | ......huh? |
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| 05:09.58 | aod2 | Hi. Anyone awake out there? |
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| 05:11.38 | *** join/#asterisk WilliamK (n=noc@static-71-170-144-3.dllstx.fios.verizon.net) |
| 05:11.38 | aod2 | I will paypal $100 USD to anyone who can help me with an SCCP problem in Trixbox. |
| 05:12.00 | aod2 | It should be fairly simple. |
| 05:12.13 | WilliamK | 100 is like a cheap date :) |
| 05:12.36 | WilliamK | sorry - I just had to say it :) |
| 05:13.07 | *** join/#asterisk aiksa[LV] (n=aiks@mx.fiveplus.lv) |
| 05:13.43 | aod2 | Well, for this problem it is more than adequate, I think. |
| 05:14.18 | WilliamK | is thinking covered under the medical health plan? |
| 05:14.52 | aod2 | I cannot get a Cisco 7970 to use the SEP(MAC).xml.conf file |
| 05:15.03 | aod2 | no matter what I change in this file, it ignores it |
| 05:15.34 | jacq | hye you guys know if cluster of TLS/sRTP capable Asterisk's, with SER balancer has been done (or documented)? |
| 05:17.28 | aod2 | $100 USD to anyone who can make this Trixbox give these 7970 phones the correct time, even after daylight saving time. And as a bonus, I'd like the services button to do something. The SEP file tells it where to go for the services, but it ignores it. |
| 05:19.15 | aod2 | Noone, huh? |
| 05:20.50 | WilliamK | aod2, buying them beer or something might go farther |
| 05:20.52 | WilliamK | :) |
| 05:21.08 | aod2 | $100 should buy you a bit of beer, I'd think. |
| 05:21.40 | WilliamK | depends on the kind and where you live |
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| 05:23.31 | aod2 | I suppose you are right, but I have $100 sitting in my Paypal account and I need to get this working. I think $100 is more than enough for something like this. It must be some stupid issue I am overlooking. |
| 05:27.08 | aod2 | Wonderful. Yet another almost nice opensource project that I will have to ditch for proprietary bullshit because noone supports it. :( |
| 05:28.11 | WilliamK | nah, you're just asking at the wrong time of night |
| 05:29.06 | aod2 | My problem is that I have a Cisco 7970 phone and I'm using SCCP with Trixbox and it works great. The only thing it doesn't do is accept it's SEP(MAC).xml.conf file. Whatever I put in there, it ignores it. |
| 05:29.23 | aod2 | I cannot get the time right on the phone, and I cannot get the services button to go anywhere. |
| 05:29.44 | aod2 | Again, I'll pay $100 USD via Paypal if someone can make this work. |
| 05:31.34 | bjweeks | I'd make a mailing list post the with bounty, I'm sure somebody will bite |
| 05:31.39 | WilliamK | oh and just a note, if I had trixbox and a 7970 I'd probably try and figure it out, but I lack those 2 resources |
| 05:31.51 | outtolunc | i seem to remember hearing some of those phones wanting xml as XML |
| 05:32.01 | jql | I see |
| 05:32.18 | aod2 | Yeah, I was watching the tftp logs and I know every file it wants. |
| 05:32.47 | outtolunc | then what error did it give when it didn't 'take' the SIPxxxxx.xml.cnf |
| 05:33.15 | aod2 | It actually loads an SEPXXXX.xml.conf file, because it is SCCP. |
| 05:34.11 | jql | I regret not having a 7970. I don't regret not using sccp |
| 05:34.15 | aod2 | Everything in SEPXXXXX.xml.conf file is ignored, however. |
| 05:34.20 | jql | but the cisco phones really are nice |
| 05:34.32 | aod2 | That's the problem. :( |
| 05:34.57 | jql | does it download your conf? |
| 05:35.00 | outtolunc | you keep saying .conf where for my stuff it is .cnf |
| 05:35.29 | jql | yes, cnf |
| 05:35.33 | aod2 | It seems to download it, but it doesn't seem to do anything with it. |
| 05:35.46 | aod2 | Actually, it expects SEPXXXXXX.xml.conf |
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| 05:35.55 | outtolunc | do you have a SCCPDefault.cnf also |
| 05:36.26 | aod2 | It expects XMLDefault.cnf.xml |
| 05:36.26 | WilliamK | http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+7970+SIP if that helps at all for the file formatting |
| 05:36.42 | aod2 | Yeah, that is for SIP though, not SCCP. |
| 05:36.43 | bjweeks | Did they change the naming in a firmware update? |
| 05:37.26 | outtolunc | notes: the sip ones wants the XMLDefault.conf.xml also |
| 05:37.30 | WilliamK | bjweeks, all the notes that I see show that formatting |
| 05:37.36 | aod2 | I think so. I'm running 8.2.2SR4 |
| 05:37.42 | WilliamK | aod2, might wanna use etherreal and see what it's asking for |
| 05:37.57 | bjweeks | Wireshark now ;) |
| 05:38.08 | WilliamK | they changed that too?!?! |
| 05:38.25 | bjweeks | The main author left his company that owned the name |
| 05:38.34 | bjweeks | http://www.wireshark.org/ |
| 05:41.04 | WilliamK | ah |
| 05:45.21 | aod2 | It gets all the files it needs from the tftp server, it just ignores any settings in the SEP(mac).xml.conf file though. |
| 05:47.24 | outtolunc | sounds to me like you have an invalid conf setting |
| 05:48.32 | aod2 | I need help with finding that invalid setting then. |
| 05:49.16 | outtolunc | well then you have to post it somewhere, as the batteries in my magic wand have expired |
| 05:49.45 | bjweeks | They don't run on batteries silly |
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| 05:51.01 | WilliamK | aod2; pastebin.ca |
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| 05:52.07 | WilliamK | outtolunc, sorry that think just barely moves with fresh batteries anyway |
| 05:52.09 | WilliamK | :) |
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| 05:55.14 | WilliamK | welcome home Corydon76 :) |
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| 06:22.22 | The_Ball | When one has multiple outgoing lines zap and sip and want to use them as a outgoing trunk, is this called trunking or grouping? |
| 06:30.08 | litage|w | how do you configure a voicemail entry in voicemail.conf to send an email to multiple email addresses? |
| 06:33.50 | The_Ball | litage|w, i have tried, i don't think it is supported |
| 06:38.08 | litage|w | The_Ball: on the voip-info.org wiki, it's suggested to put a unix username rather than an email address, and then put the desired multiple email addresses in /etc/mail/aliases for the specified unix username |
| 06:38.57 | The_Ball | that will depend on the mail setup on your host. if you run for example ssmtp which delivers directly to server, no local relay, it will not work |
| 06:39.17 | litage|w | The_Ball: ah true |
| 06:39.27 | litage|w | The_Ball: would it work if you use nullmailer? |
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| 07:23.38 | ussrback | helloo |
| 07:23.44 | ussrback | I have question |
| 07:23.52 | ussrback | I have configures MusicOnHold |
| 07:24.09 | ussrback | when im conectng to asterisk throught the Sip phone |
| 07:24.15 | ussrback | its working perfectly |
| 07:24.36 | ussrback | but when im trying to connect it throught cisco AS 5350 |
| 07:24.46 | ussrback | i cant hear anything |
| 07:25.02 | ussrback | i use g711 codects in cisco dial-peer |
| 07:25.05 | ussrback | with asterisk |
| 07:25.13 | ussrback | what is a problem? |
| 07:25.21 | ussrback | and how can i fix? |
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| 07:27.46 | tuzhila | hi all |
| 07:27.55 | tuzhila | i've got a problem |
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| 07:33.10 | tuzhila | i have linksys spa3102 fxo-adapter, i want to make calls from voip to pstn. i did: i registered linksys on asterisk for sip id 70444, then registered my softphone for sip id 70111. in linksys i have whrote: (xxxxxx<:@gw0>). also http://pastebin.ca/724010 ---my sip.conf and extensions.conf. So, this stuff is not work |
| 07:33.16 | tuzhila | please, help me |
| 07:33.28 | Dandre | hello, |
| 07:33.54 | Dandre | Ares those two extensions fragment equivalent and valid: http://pastebin.org/3980 |
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| 07:40.26 | Chris-NB | hi |
| 07:41.43 | Chris-NB | if i have a database entrie like: /something/nr = 1 where ${nr} is dynamic is it possible to check in dialplan if there is such an entry: /something/?? |
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| 07:46.56 | nexilus | Is it possible to, from the AMI, place a call between two parties so that the phone rings on both sides? |
| 07:47.37 | nexilus | i.e i want to automatically call for example SIP/999 and ZAP/129381928319283, so that the phone rings on both ends and they can start talking once they pickup the phone |
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| 08:00.19 | Dandre | Are those two extensions fragments equivalent and valid: http://pastebin.org/3980 ? |
| 08:10.35 | nexilus | Dandre: equivalent in a manner, yes, valid, kindof, but the prior example is prefered for simplicity |
| 08:10.44 | nexilus | and comprehensiveness |
| 08:11.12 | nexilus | or first example rather |
| 08:13.21 | Dandre | Yes I know that the first is more comprehensive. But I have asked this question because I intend to have my extensions.conf generated by some tool and if the order as any importance so that I had to take this into account in the design. |
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| 08:37.43 | nexilus | Dandre: i would suggest you make the tool take the order into account for comprehensive manual debugging if needed |
| 08:37.59 | Dandre | ok |
| 08:38.14 | nexilus | And if you're indeed making the tool yourself, consider using comprehensive comments in the extensions.conf file aswell to clarify further |
| 08:38.55 | nexilus | like: ; Office phones ; Last update 2007-10-22 18:00 ; Some other phones ; Test phones |
| 08:38.55 | nexilus | etc |
| 08:39.11 | nexilus | ive found such comments are simplifying debugging and manual configuration alot "when needed" |
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| 08:41.33 | nexilus | And actually, i would also suggest you use some form of backup system that stores the old extensions.conf somewhere with a time date stamp so that you can easily revert to the old config just incase you mess up an update |
| 08:42.16 | nexilus | i could ramble about this all day :P |
| 08:44.18 | nexilus | I myself have taken a slightly different route to auto generation.. i use include statements to include files from directories which include the phonedata |
| 08:44.31 | nexilus | so i have directories containing small files about extensions |
| 08:45.04 | nexilus | that way i only need to generate a new file in my directory which will automatically be loaded without the need of changing extensions.conf and risk it to get b0rked |
| 08:45.56 | nexilus | (same with sip.conf too btw) |
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| 08:53.02 | Dandre | ok thanks for the advise |
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| 09:06.34 | Dandre | nexilus: can the #include statement be used to include all file in a particuliar directory? |
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| 09:19.46 | luke-jr | ⁂ |
| 09:22.05 | Strom_M | i'm sorry sir...multiple asterisks all at the same time does not save the universe |
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| 09:42.08 | agx | Using G.711 a/u and ISDN/Analog line if i connect a G3 FAX will it work in G3 or it will slowdown to 9600 baud? Also if i connect a Modem/POS 56K will it work at 56K or it will slow down to 9600 ? |
| 09:42.55 | Maliuta | 56K requires a digital switch at one end of the line |
| 09:43.38 | *** join/#asterisk tuzhila (i=tuzhila@84.47.128.99) |
| 09:43.40 | Strom_M | moreso than that, it requires digital entrance facilities to the modem you're connecting to |
| 09:43.45 | agx | Maliuta, Modem or G3 ---> Linksys ATA --- g711 --- asterisk --- analog or ISDN ---> Telco |
| 09:44.03 | agx | this is the scenario |
| 09:44.09 | tuzhila | please, help me, who works with linksys spa 3102 |
| 09:44.11 | Strom_M | agx: that's a recipe for failure |
| 09:44.23 | agx | Strom_M, so this will not work? |
| 09:44.31 | Strom_M | fax over voice over IP |
| 09:44.33 | tuzhila | i cant configure voip to pstn gateway |
| 09:44.34 | Strom_M | think about it |
| 09:44.39 | Strom_M | it's designed for voice, not fax |
| 09:44.45 | Strom_M | look into a T.38 gateway |
| 09:45.10 | agx | Strom_M, ok, what about Modem/POS? same problem as faxes? |
| 09:45.35 | tuzhila | please, help me, who works with linksys spa 3102 |
| 09:45.38 | tuzhila | ? |
| 09:47.38 | Strom_M | agx: it's still data calls rather than voice |
| 09:47.50 | Strom_M | tuzhila: stop being annoying and just ask a question please |
| 09:48.30 | tuzhila | Strom_M: how i can to create voip to pstn gateway with spa3102 |
| 09:48.31 | tuzhila | ? |
| 09:49.18 | Strom_M | um |
| 09:49.23 | Strom_M | perhaps read the instruction manual? |
| 09:49.27 | tuzhila | yes |
| 09:49.43 | tuzhila | i already created pstn to voip gateway |
| 09:50.23 | tuzhila | but i can't to do reverse stuff |
| 09:50.58 | tuzhila | this is my sip.conf and extensions.conf, and sip debug |
| 09:52.59 | tuzhila | wait |
| 09:53.05 | tuzhila | this is it: |
| 09:53.19 | tuzhila | http://pastebin.ca/724093 |
| 09:54.20 | tuzhila | what do you think? |
| 09:54.23 | tuzhila | Strom_M: ? |
| 09:54.34 | Strom_M | # |
| 09:54.34 | Strom_M | <--- SIP read from 192.168.5.24:5060 ---> |
| 09:54.34 | Strom_M | # |
| 09:54.34 | Strom_M | SIP/2.0 404 Not Found |
| 09:54.35 | Strom_M | read the error message :) |
| 09:54.46 | tuzhila | i see, but why? |
| 09:54.54 | tuzhila | where is the problem? |
| 09:55.17 | tuzhila | 192.168.5.24 - is linksys |
| 09:55.41 | thewiizle | its not found |
| 09:55.44 | thewiizle | the number you are calling |
| 09:55.49 | thewiizle | check your in the right context |
| 09:55.56 | tuzhila | in linksys? |
| 09:56.00 | tuzhila | or in asterisk? |
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| 09:56.40 | tuzhila | thewiizle: ? |
| 09:58.24 | luke-jr | Strom_M: as far as fax goes, I have had very few problems so long as I use ulaw ⁂ |
| 09:58.52 | luke-jr | now modems… never got them to work, but I'm not sure that was a VoIP issue ☺ ⁂ |
| 10:00.27 | Strom_M | luke-jr: the cute characters which terminate your sentences are beyond irritating :) |
| 10:00.37 | luke-jr | ☺ |
| 10:01.08 | Strom_M | stop that |
| 10:01.11 | Strom_M | seriously |
| 10:01.29 | luke-jr | :/ |
| 10:02.06 | luke-jr | what's your problem, seriously? -.- |
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| 10:04.01 | Strom_M | my problem is that on my display (and i'm sure on many others' displays as well) the characters are too small to be of any real use |
| 10:04.12 | Strom_M | the emoticon, for example, is just a dot |
| 10:04.26 | luke-jr | well, that's the fault of your fonts |
| 10:04.37 | luke-jr | that character predates your OS |
| 10:05.05 | Strom_M | yes, this I know |
| 10:05.31 | Strom_M | however, expecting people to change their typefaces just so they can read your silly nonstandard emotes is a little arrogant, don't you think? :) |
| 10:06.11 | luke-jr | I don't think any of my standard Unicode symbols are necessary to understanding what I say most of the time ;) |
| 10:06.26 | luke-jr | tbh, they're totally invisible to me |
| 10:06.40 | aiksa[LV] | i suppose :) :( and :P is enough for emotional gamma of an IT specialist |
| 10:06.54 | luke-jr | aiksa[LV]: that's what I type to get ☺ ☹ |
| 10:07.54 | aiksa[LV] | your irc client is doing replace magic? |
| 10:08.59 | luke-jr | yea |
| 10:09.03 | aiksa[LV] | then trash it or look for ways to disbale that. Thos chatting from console will find that annoying. |
| 10:09.14 | luke-jr | I intentionally enabled it ;) |
| 10:09.25 | luke-jr | any modern console should work fine with Unicode |
| 10:10.53 | aiksa[LV] | disregarding wheather console should do that or not. Emotion icons IMHO are ugly |
| 10:10.56 | tuzhila | linksys 3102 |
| 10:11.02 | tuzhila | i cant to create voip->pstn gw |
| 10:11.07 | aiksa[LV] | :) is clean and simple |
| 10:11.09 | tuzhila | from asterisk i received this message: SIP/2.0 404 Not Found |
| 10:11.16 | tuzhila | when i dialing 421818@linksys |
| 10:11.23 | tuzhila | in linksys i have this dialplan |
| 10:11.24 | luke-jr | aiksa[LV]: then find a nicer font ☺ |
| 10:11.32 | tuzhila | (xxxxxxS0<:@gw0>) |
| 10:11.40 | tuzhila | that is my problem |
| 10:11.46 | tuzhila | please, help me |
| 10:11.52 | luke-jr | a font could always draw it as : ) |
| 10:12.25 | aiksa[LV] | tuzhila: how have you defined that linksys in sip.conf ? |
| 10:12.44 | tuzhila | http://pastebin.ca/724093 |
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| 10:13.44 | tuzhila | aiksa[LV]: what do you think? |
| 10:13.52 | aiksa[LV] | a second |
| 10:13.55 | tuzhila | ok |
| 10:14.20 | aiksa[LV] | you have missmatching in [linksys] and username |
| 10:14.38 | aiksa[LV] | try giving username linksys or define sip entry as [222] |
| 10:14.54 | tuzhila | aiksa[LV]: ok, now... |
| 10:16.12 | tuzhila | aiksa[LV]: no, the same problem |
| 10:16.22 | tuzhila | it's not help |
| 10:17.07 | tuzhila | <--- SIP read from 192.168.5.24:5060 ---> |
| 10:17.07 | tuzhila | SIP/2.0 404 Not Found |
| 10:38.33 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 10:43.35 | aiksa[LV] | back |
| 10:44.06 | aiksa[LV] | ok, now you have both sip entry and username the same |
| 10:44.20 | tuzhila | yes |
| 10:44.31 | aiksa[LV] | and are referencing router with that name. |
| 10:44.49 | tuzhila | yes, linksys is 111 |
| 10:45.18 | aiksa[LV] | try calling to SIP/username:password@ipdaddress_of_the_linksys/${EXTEN} |
| 10:45.31 | aiksa[LV] | i guess that was the syntax |
| 10:48.24 | aiksa[LV] | btw I dont see from your extensions.conf where you are trying to access that linksys device. |
| 10:50.18 | tuzhila | ok, trying.. |
| 10:51.05 | *** join/#asterisk saftsack (n=saftsack@p54A76361.dip.t-dialin.net) |
| 10:51.28 | *** join/#asterisk vader-- (n=me@c-71-226-197-0.hsd1.nj.comcast.net) |
| 10:52.03 | tuzhila | aiksa[LV]: this way? exten => 421818,1,Dial(SIP/${EXTEN}@sipuser:password@192.168.5.24,20) |
| 10:52.04 | tuzhila | ? |
| 10:54.01 | tuzhila | no, the same result |
| 11:00.15 | *** join/#asterisk Bladerunner05 (n=feelme@81-174-56-54.static.ngi.it) |
| 11:00.43 | Bladerunner05 | If I run asterisk with a non privileged user it return error wriiting pid file |
| 11:01.53 | tuzhila | aiksa[LV]: ? |
| 11:02.04 | harryr | Bladerunner05: remove the pid file it was trying to write to and make sure the directory is writable by the user you're running asterisk as |
| 11:04.38 | *** join/#asterisk ai-a (n=jake2@cpc5-oxfd1-0-0-cust523.oxfd.cable.ntl.com) |
| 11:06.11 | ai-a | We've got an Asterisk PBX installed on our lan. We're getting "TOO LAGGED! (3014ms / 2000ms)" on the ext.'s within the network. We've got a switched network have a database on it, but the traffic isnt high.. any reason for getting these issues ? |
| 11:14.06 | *** join/#asterisk CaRb0n^ (n=Omer@202.133.69.117) |
| 11:15.56 | *** join/#asterisk coppice (n=chatzill@30.168.17.210.dyn.pacific.net.hk) |
| 11:18.36 | CaRb0n^ | all my IAX trunks are down |
| 11:18.45 | CaRb0n^ | after my Dynamic IP changed |
| 11:18.51 | CaRb0n^ | SIp trunks are working fine |
| 11:19.18 | thewiizle | heh |
| 11:19.20 | thewiizle | shoutcast MOH |
| 11:19.22 | thewiizle | quality! |
| 11:19.34 | thewiizle | ai-a sounds like a shit NAT router |
| 11:19.41 | aiksa[LV] | tuzhila: sorry i was away from comp. |
| 11:20.12 | aiksa[LV] | tuzhila: just a second |
| 11:21.34 | aiksa[LV] | i guess thats IAX specific then |
| 11:21.48 | tuzhila | aiksa[LV]: no, its sip |
| 11:24.07 | CaRb0n^ | hmm |
| 11:25.16 | tuzhila | aiksa[LV]: any ideas? |
| 11:27.45 | *** join/#asterisk luke-jr (n=luke-jr@2002:440d:6de2:0:20e:a6ff:fec4:4e5d) |
| 11:28.06 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.52) |
| 11:28.25 | *** join/#asterisk bjweeks (n=bjweeks@unaffiliated/bjweeks) |
| 11:30.19 | ai-a | thewiizle: the router is a linux box. |
| 11:30.38 | thewiizle | is the pbx on the LAN |
| 11:30.38 | thewiizle | ? |
| 11:31.50 | ai-a | yes |
| 11:32.03 | thewiizle | ah |
| 11:32.14 | thewiizle | Are the phones on the LAN also |
| 11:32.18 | ai-a | its all on the same lan, |
| 11:32.20 | thewiizle | hmmm |
| 11:32.24 | thewiizle | ok that is weird |
| 11:32.36 | thewiizle | your not using some bullshit switch are you? |
| 11:32.39 | thewiizle | or a hub |
| 11:33.23 | ai-a | switches are HP Procurve J4899A. each desk has a Netgear FS105 switch when required. |
| 11:33.30 | ai-a | no hubs are used. |
| 11:33.43 | ai-a | networks fine.. pings.. db access. Normal office is working. |
| 11:33.45 | thewiizle | using packet scheduling? |
| 11:33.49 | thewiizle | QoS etc |
| 11:33.50 | Strom_M | ai-a: why are you qualifying on-lan phones in the first place |
| 11:33.51 | ai-a | nope. |
| 11:33.58 | thewiizle | hmm |
| 11:34.13 | ai-a | Strom_M: trying to increase the fax on ata device reliability. |
| 11:34.20 | Strom_M | uh |
| 11:34.22 | thewiizle | odd |
| 11:34.33 | thewiizle | not sure tbh |
| 11:34.44 | Strom_M | that's much like tryig to make a car more fuel efficient by performing a ballet dance |
| 11:34.57 | ai-a | why / |
| 11:35.09 | Strom_M | what problem will qualify solve, exactly |
| 11:35.36 | *** join/#asterisk saftsack (n=saftsack@p54A77CFA.dip.t-dialin.net) |
| 11:36.02 | ai-a | we get about 2sec lag on these phones. A) this shouldnt happen anyway on a network.. B) its making the fax machines fail. Just asking in here to see if anyone knows of a way to track this down on a network. |
| 11:36.48 | Strom_M | well apart from the latency problems, you should be using t.38, not fax over voice over ip |
| 11:36.57 | Strom_M | but regardless |
| 11:37.06 | ai-a | t.38 requires new fax machines ? |
| 11:37.08 | Strom_M | describe your network |
| 11:37.09 | Strom_M | no |
| 11:37.30 | ai-a | the ata devices say they support t.38.. does asterisk support that ? |
| 11:37.54 | bjweeks | yeah, passthrough should just let it go to the ata |
| 11:38.14 | bjweeks | new to 1.4 IIRC |
| 11:38.18 | Strom_M | yes |
| 11:38.21 | ai-a | we're using 1.2 |
| 11:38.26 | Strom_M | upgrade time |
| 11:38.30 | ai-a | impossible. |
| 11:38.58 | Strom_M | turning cheese into a Howard Jones album is impossible |
| 11:39.04 | Strom_M | upgrading is not. |
| 11:39.09 | ai-a | network is 60+ pcs on about 5 switches. we've added snom phones for everyone to replace existing phone system. and added ata boxes for dec phones and 6 fax machines. |
| 11:39.25 | bjweeks | you have 60+ asterisk servers? |
| 11:39.36 | ai-a | pcs != pbx |
| 11:39.45 | bjweeks | then what is the problem? |
| 11:39.57 | bjweeks | oh, nevermind |
| 11:40.02 | ai-a | [12:05:21] <ai-a> We've got an Asterisk PBX installed on our lan. We're getting "TOO LAGGED! (3014ms / 2000ms)" on the ext.'s within the network. We've got a switched network have a database on it, but the traffic isnt high.. any reason for getting these issues ? |
| 11:40.35 | *** join/#asterisk tsurko (n=tsurko@78.90.100.214) |
| 11:40.39 | CaRb0n^ | check your pbx network card |
| 11:40.53 | Strom_M | ai-a: describe the entire network between the phone and the pbx |
| 11:40.57 | ai-a | The asterisk pbx is build by a 3rd party company. they have considered 1.4 not being a viable update at the moment. until they are sure its stable we are still using the one they initially installed. |
| 11:40.59 | CaRb0n^ | and systems cars on the entire network |
| 11:41.11 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 11:41.27 | Strom_M | ai-a: i want whatever that third party company is smoking |
| 11:41.32 | ai-a | Strom_M: (fax/phone) - [ata] - network - 1 or 2 switches -> [pbx] |
| 11:41.33 | Strom_M | clearly it's great shit |
| 11:41.39 | bjweeks | well, then you can't do reliable fax |
| 11:41.50 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 11:42.02 | ai-a | CaRb0n^: right, that will take forever. |
| 11:42.05 | agx | ai-a, use a separate server for just fax handling? |
| 11:42.22 | ai-a | agx: we're not. |
| 11:42.25 | agx | ai-a, i plug directly voip account onto the ATA |
| 11:42.28 | ai-a | if that was a question. |
| 11:42.40 | ai-a | agx: please repeat that in english. |
| 11:43.02 | Strom_M | ai-a: forget qualify; what's the latency when you just ping the ata? |
| 11:43.16 | agx | ai-a, sorry i'm not able to repeat in english |
| 11:43.51 | ai-a | 0.7ms to 1.2ms latency. |
| 11:43.57 | ai-a | agx: thats okay. |
| 11:44.18 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 11:44.24 | ai-a | i gather you said something like you enter your account details directly into the ata device. |
| 11:44.30 | Strom_M | ai-a: well then latency is not the problem |
| 11:44.44 | ai-a | the system automatlic picks up the settings from the server when dhcp directs it. |
| 11:44.53 | ai-a | Strom_M: thats what i said. |
| 11:44.59 | Strom_M | ai-a: the problem is that fax over voice over IP is just a brain-dead idea, and you should be using T.38 instead |
| 11:45.22 | Strom_M | alteratively...bypass the PBX entirely |
| 11:45.24 | *** join/#asterisk famicon (i=redz@c51447ddc.cable.wanadoo.nl) |
| 11:45.31 | ai-a | [Boss] <---> me <----> [Company] - boss wont pay until fax works, i say fax doesnt work,, company say its our network issue. |
| 11:45.43 | ai-a | Strom_M: i know that... you want my bosses phone number ? |
| 11:46.10 | Strom_M | don't spaz out on me :) |
| 11:46.16 | ai-a | lol |
| 11:46.33 | Strom_M | also, 1.4 is stable. |
| 11:46.42 | Strom_M | i run it here and ive not had problems. |
| 11:46.50 | Strom_M | digium uses it on their production systems |
| 11:47.11 | Strom_M | 1.2 is in "for the love of god won't you upgrade to 1.4 already before we completely abandon this version" mode |
| 11:48.28 | aiksa[LV] | tuzhila: sorry, gotta run. |
| 11:48.45 | thewiizle | ai-a why not get Fax2Email |
| 11:48.48 | thewiizle | fuck the whole shit off |
| 11:49.14 | tuzhila | aiksa[LV]: what do you think about my problem? |
| 11:49.37 | ai-a | thewiizle: the system as a softfax that works well, however we have people needing to sign documents. |
| 11:49.54 | thewiizle | before they are sent out? |
| 11:50.02 | ai-a | yep |
| 11:50.10 | thewiizle | adobe creater :) |
| 11:50.10 | ai-a | i said give them a scanner ;) |
| 11:50.19 | ai-a | lol.. get it to auto-sign HAHA. |
| 11:50.23 | thewiizle | uh huh |
| 11:50.29 | ai-a | i have a an Asterisk SVN-branch-1.4-r77571 on another box.. can i use that for the fax'es and then forward the data to the asterisk 1.2 box for calling outside ? |
| 11:50.30 | thewiizle | just load all the signitures in |
| 11:50.47 | *** part/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 11:50.49 | thewiizle | T.38 is ~ |
| 11:51.29 | Strom_M | no support for t.38 passthrough in 1.2 |
| 11:52.47 | ai-a | We are using ASterisk 1.2.24-BRIstuffed-0.3.0-PRE-1y-j |
| 11:53.35 | *** join/#asterisk tsurko (n=tsurko@78.90.100.214) |
| 11:55.05 | ThoMe | hello |
| 11:55.16 | ThoMe | how i can disabled auto-recording in a queue? |
| 11:55.36 | ThoMe | i have tried: monitor-type = "" |
| 11:55.43 | ThoMe | :/ |
| 11:55.57 | coppice | T.38 is a lightweight jet fighter |
| 11:56.43 | ThoMe | hmm |
| 11:57.22 | coppice | the talon |
| 11:59.22 | Spida | coppice: I wouldn't call it a "fighter", though |
| 12:00.08 | thewiizle | T.38 is a nice idea |
| 12:00.10 | thewiizle | :) |
| 12:02.16 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 12:02.56 | coppice | why? it kills people, though its mostly a trainer |
| 12:03.08 | coppice | T.38 is a terrible idea. T.37 is a nice idea |
| 12:05.36 | Spida | coppice: everything kill people, if you throw enough of it at them. |
| 12:05.46 | *** join/#asterisk DImGR_lap (n=DimGR_la@193.92.98.181) |
| 12:06.42 | coppice | Spida: I don't think the US domestic model of the T.38 was armed. the export ones were, though, and the F5 was a more heavily armed derivative |
| 12:09.34 | JT | ai-a: sounds like your network is defective |
| 12:12.13 | *** join/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 12:12.16 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:22.25 | *** join/#asterisk stmaher (n=stmaher@87.198.5.178) |
| 12:22.29 | stmaher | Hi guys. . |
| 12:23.24 | stmaher | I have a strange problem.. I make a sip call into asterisk and then it is being routed to a third party IMS.. If there is silence (about 14 seconds the call is dropped and rung again.. any ideas? |
| 12:24.20 | [TK]D-Fender | stmaher: We suggest you pastebin the CLI output of the entire call at verbose 10 & SIP DEBUG enabled |
| 12:24.28 | [TK]D-Fender | ~pb |
| 12:24.28 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 12:24.30 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^ |
| 12:25.20 | stmaher | thanks fender.. Ill try and figure out how to up the logs.. |
| 12:25.25 | stmaher | brb |
| 12:26.09 | stmaher | would this be the full file log in /var/log/asterisk/ ? |
| 12:26.14 | *** join/#asterisk clandmeter (n=Carlo@81.175.82.2) |
| 12:26.30 | [TK]D-Fender | stmaher: No, I just asked for *CLI OUTPUT* |
| 12:32.10 | *** join/#asterisk grEvenX (n=even@1ult2p8.ip.hipercom.no) |
| 12:39.04 | DImGR_lap | hi |
| 12:39.13 | DImGR_lap | anyone has the web-gui working good? |
| 12:40.11 | *** join/#asterisk guillote_GNU (n=bancaria@host236.190-30-115.telecom.net.ar) |
| 12:42.56 | *** join/#asterisk jetlagmk2 (i=jetlag@70.17.59.195) |
| 12:43.51 | [TK]D-Fender | DImGR_lap: Not the place to ask really... |
| 12:46.24 | thewiizle | hmm has IVR config changed from 1.2 to 1.4 |
| 12:46.25 | thewiizle | ? |
| 12:46.48 | *** join/#asterisk penguinFunk (n=penguin@unaffiliated/penguinfunk) |
| 12:48.32 | [TK]D-Fender | thewiizle: 1.0 deprecated stuff is gone, everything that was 1.2 standard is identical. |
| 12:50.26 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 12:51.11 | thewiizle | sweet |
| 12:51.23 | thewiizle | fancy taking a look at a ivr setup quick |
| 12:51.30 | thewiizle | seems standard but fails |
| 12:52.02 | thewiizle | fails when trying to set DigitTimeout,5 |
| 12:52.08 | *** join/#asterisk marexz (n=marexz@marexz.mil.lv) |
| 12:52.10 | [TK]D-Fender | thewiizle: that app is 1.0 and is GONE |
| 12:52.39 | thewiizle | thats 1.0! |
| 12:52.41 | thewiizle | my go |
| 12:52.41 | thewiizle | d |
| 12:52.44 | [TK]D-Fender | thewiizle: Go read up on 1.2 FUNCTION s |
| 12:52.57 | thewiizle | i just got that voip-info |
| 12:52.58 | thewiizle | bastards |
| 12:54.26 | [TK]D-Fender | thewiizle: Then again it'd be quickly apparent if you did "show applications" and "show functions"....... |
| 12:55.04 | [TK]D-Fender | thewiizle: The deprecated stuff si clearly gone ad rather obviously named replacements are clearly listed... |
| 12:57.52 | thewiizle | yeh fixed that now |
| 12:58.21 | aiksa[LV] | back to keyboard |
| 12:58.22 | *** join/#asterisk lbow (n=lbow@dsl-146-5-201.telkomadsl.co.za) |
| 12:58.50 | aiksa[LV] | [TK]D-Fender: so far no answer from digium support reagrding that PRI problem. What are usual response times for them? |
| 12:59.18 | [TK]D-Fender | aiksa[LV]: No clue... only needed them once 2 years ago... |
| 12:59.19 | aiksa[LV] | or should I better add this to bugtrack? |
| 12:59.45 | [TK]D-Fender | aiksa[LV]: Its always best to approach these problems from all angles. Work them all until you get what you want. |
| 12:59.57 | aiksa[LV] | [TK]D-Fender: my last contact with them was some 3 years ago or so. When trying to intsall * BE on test machine |
| 13:00.03 | *** join/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 13:04.05 | thewiizle | [TK]D-Fender what does sip prune realtime do? |
| 13:05.16 | penguinFunk | has anyone else noticed that anything < G711 (alaw or ulaw) is completely hopeless? |
| 13:05.33 | penguinFunk | we bought G729 codecs and the sound quality is pathetic |
| 13:05.39 | penguinFunk | :( what a waste of money |
| 13:05.58 | coppice | G.729 shouldn't sound too bad |
| 13:05.59 | cpm | it also sux, but the price is right |
| 13:06.15 | penguinFunk | coppice: very quiet and unclear is how i would describe it |
| 13:06.20 | coppice | speex should also sound OK. |
| 13:06.22 | penguinFunk | awful |
| 13:06.27 | coppice | your setup is broken |
| 13:06.31 | penguinFunk | nope |
| 13:06.36 | penguinFunk | alaw sounds perfect! |
| 13:06.46 | penguinFunk | just a bandwidth hungry monster thats all |
| 13:06.49 | coppice | yes. G.729 is neither quiet nor unclear |
| 13:06.58 | penguinFunk | pff |
| 13:07.08 | coppice | it just doesn't sound as good as alaw |
| 13:07.09 | penguinFunk | i beg to differ. interesting you have different views |
| 13:07.40 | [TK]D-Fender | penguinFunk: No... its just YOU. In the battle of you vs the rest of the world... bet on the world. |
| 13:07.48 | coppice | whatever you think of the quality. it is absolutely certainly definitely the same volume, unless something is broken |
| 13:08.05 | [TK]D-Fender | penguinFunk: Look at the sum of your solution. |
| 13:08.14 | penguinFunk | everything works perfect with alaw, but using G729 seems to be a bit variable. sometimes better than others (depending on handsets used i suppose) but it is so difficult to have a full conversation using G729 |
| 13:08.28 | cpm | coppice, yes, I like speex, and use it where ever I am able. |
| 13:08.52 | [TK]D-Fender | penguinFunk: I did testing with G.729 on my Polycom's and found it OK. |
| 13:08.55 | penguinFunk | okay forget volume |
| 13:09.02 | penguinFunk | we are talking quality |
| 13:09.35 | penguinFunk | [TK]D-Fender: and you could understand every word every person says using G729? |
| 13:09.37 | coppice | before you were talking volume as well, how did that suddenly improve? |
| 13:09.52 | coppice | you should have no problem understanding G.729 |
| 13:10.06 | coppice | its superior to 90% of cellphone calls |
| 13:10.11 | penguinFunk | hmmm |
| 13:10.27 | [TK]D-Fender | penguinFunk: Yes. The only difference was a tiny harmonic "warbling" if I concentrated or there was low music playing (music does not encode so well over g729) |
| 13:10.34 | *** join/#asterisk southtel (n=southtel@68-114-23-151.dhcp.gwnt.ga.charter.com) |
| 13:10.52 | [TK]D-Fender | penguinFunk: conversation itself was fine |
| 13:10.56 | coppice | if there is music it sounds bloody awful, but a voice should sound fine |
| 13:11.04 | penguinFunk | ok |
| 13:11.16 | [TK]D-Fender | penguinFunk: Remember this is almost the primary codec used by ITSP's etc... |
| 13:11.18 | ai-a | Whats wrong with using g711 ? |
| 13:11.37 | penguinFunk | G711 uses too much bandwidth |
| 13:11.56 | penguinFunk | perfect quality though |
| 13:12.00 | ai-a | well, less bandwidth usage almost always means less quality. |
| 13:12.05 | penguinFunk | of course |
| 13:12.09 | coppice | G.711 is far from perfect quality |
| 13:12.15 | penguinFunk | but i want to meet half way between G711 and G729 |
| 13:12.18 | *** join/#asterisk zerohalo (n=zeroHalo@pool-72-70-79-233.bstnma.east.verizon.net) |
| 13:12.19 | penguinFunk | is there a codec that does this |
| 13:12.24 | ai-a | how much bandwidth does 1 g711 conversation take ? |
| 13:12.50 | ai-a | penguinFunk: the g713 seems half way ;) |
| 13:12.51 | Strom_M | 64kbps |
| 13:12.58 | creativx | ee |
| 13:13.01 | ai-a | geez, my maths suck. |
| 13:13.04 | penguinFunk | g729 sounds like shit, g711 too much bandwidth |
| 13:13.19 | Strom_M | you people are too damn picky |
| 13:13.22 | ai-a | penguinFunk: stop gassing so much. |
| 13:13.27 | coppice | your setup is broken if it sounds that bad |
| 13:13.30 | penguinFunk | i need one that will use a bit more b/w than g729 and has a bit more quality too |
| 13:13.39 | Strom_M | how about g726 |
| 13:13.46 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:13.57 | penguinFunk | coppice: then how do you explain the fact that using the SAME setup but using Alaw. sounds lovely |
| 13:14.03 | ai-a | penguinFunk: if there was a pefect codec we wouldnt need to use the others. |
| 13:14.05 | penguinFunk | g726? |
| 13:14.07 | penguinFunk | is it free |
| 13:14.12 | penguinFunk | lol |
| 13:14.12 | Strom_M | yes |
| 13:14.15 | coppice | its not the same setup. you changed codecs |
| 13:14.29 | ai-a | mp3 codec.... |
| 13:14.32 | penguinFunk | so you have just admitted there is a problem with the codec then coppice |
| 13:14.50 | penguinFunk | 729 is shit |
| 13:15.01 | coppice | are you this much of a dumbass in real life, or it is just a part you play on IRC? |
| 13:15.09 | penguinFunk | ? |
| 13:15.24 | penguinFunk | <coppice> its not the same setup. you changed codecs |
| 13:15.26 | penguinFunk | exactly |
| 13:15.37 | penguinFunk | you must be the stupid one |
| 13:16.14 | penguinFunk | the only thing changing is the codec. codec1 = lovely, codec2 = shit. therefore codec2 is shit, and then you call me a dumbass |
| 13:16.19 | *** part/#asterisk southtel (n=southtel@68-114-23-151.dhcp.gwnt.ga.charter.com) |
| 13:16.20 | bjweeks | http://www.javierzanetti.net/Handoyo/Pictures/070319%20Special%20Olympics.jpg |
| 13:16.21 | penguinFunk | that is pure logic |
| 13:16.27 | penguinFunk | you fuckin tit |
| 13:16.30 | coppice | broken logic |
| 13:16.37 | penguinFunk | please explain |
| 13:16.57 | coppice | to an idiot? you should I. bye |
| 13:17.11 | bjweeks | haha, I love this channel |
| 13:17.12 | deeperror | bjweeks: indeed |
| 13:17.22 | *** join/#asterisk nettie (n=nettie@ns.coolgadgets.it) |
| 13:18.13 | [TK]D-Fender | penguinFunk: We're back to the "just you" side of things... look at your codec revisions and the exact hardware being used. Details might help.... |
| 13:18.27 | [TK]D-Fender | penguinFunk: And please try to keep it civil.... |
| 13:18.33 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.157.180) |
| 13:18.46 | bjweeks | in before "he started it" |
| 13:18.58 | [TK]D-Fender | *sigh* |
| 13:18.58 | nettie | Hi guys, I'm running a bristuffed asterisk 1.2.17 + Junghanns DuoBRI card, on the console I keep seeing == Primary D-Channel on span 2 down messages -- Is it supposed to be normal? Thanx in advance |
| 13:20.06 | ai-a | nettie: you have your isdn line plugged in ? |
| 13:20.15 | nettie | ai-a of course I do |
| 13:20.26 | ai-a | then do a pri debug.. |
| 13:20.44 | ai-a | pastebin your ext / zaptel / zapata / debug / cli output.... |
| 13:20.59 | ai-a | is any alarms happening on the isdn card? |
| 13:21.01 | ai-a | what card is it ? |
| 13:21.08 | ai-a | oh i missed that bit ;) |
| 13:21.11 | nettie | Junghanns DUOBRI |
| 13:21.11 | penguinFunk | coppice: and from what evidence are you suggesting that i am an idiot? you cant even explain yourself |
| 13:21.15 | thewiizle | [TK]D-Fender quick ones for you, rapid fire stylee |
| 13:21.24 | thewiizle | sip prune realtime |
| 13:21.26 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:21.27 | thewiizle | and auth sent |
| 13:21.34 | penguinFunk | you got no answer, so the nobs way out is to say that. you are pathetic |
| 13:21.57 | ai-a | penguinFunk: yer, hes pathetic.. convo over.. your codec still doesnt work. |
| 13:22.18 | thewiizle | penguinFunk, your rich calling someone trying to help you out pathetic |
| 13:22.28 | thewiizle | Just ask yourself why your here |
| 13:22.29 | thewiizle | ;) |
| 13:22.30 | ai-a | if you think name calling fixes your codec problems then continue. |
| 13:23.12 | bjweeks | if name calling fixed things this channel would be empty ;) |
| 13:23.21 | thewiizle | heh |
| 13:23.34 | ai-a | penguinFunk: logic states nobody will create a 'pay for' codec that sounds too quite and unclear.. so it wouldnt be the codec's issue. |
| 13:23.57 | penguinFunk | thewiizle: he was not helping me at all. i have a lot of respect for people on irc giving free help. I have done it myself in other channels so no need to say any of that. |
| 13:24.07 | penguinFunk | he was simply giving me abuse for no reason |
| 13:24.10 | penguinFunk | not helping |
| 13:24.20 | penguinFunk | so i can call him what i like, just like he calls me what he likes |
| 13:24.21 | ai-a | penguinFunk: there is an /ignore command. |
| 13:24.32 | bjweeks | I still don't think it should cost anything, every codec is covered under some BS patent but we don't pay for them |
| 13:24.35 | thewiizle | penguinFunk Rise above |
| 13:24.53 | ai-a | thewiizle ;) |
| 13:25.02 | thewiizle | bjweeks run that theory by the people that develop it :) |
| 13:25.04 | penguinFunk | true |
| 13:26.01 | bjweeks | thewiizle: the people that hold the patents? I'm sure I would get some shit about how their patents kick every other patent's ass |
| 13:28.13 | creativx | I will patent your ass. |
| 13:28.57 | [TK]D-Fender | penguinFunk: Perhaps you can try to help our understanding and tell us exactly what hardware & software is involved..... |
| 13:28.59 | bjweeks | Microsoft already owns it |
| 13:29.23 | [TK]D-Fender | penguinFunk: Because they do put out new releases of even the G.729 codec for quality improvements now and then... |
| 13:29.29 | penguinFunk | really? |
| 13:29.38 | penguinFunk | that's interesting |
| 13:29.45 | penguinFunk | well we have 2 asterisk systems |
| 13:29.52 | penguinFunk | a bunch of sip users either end |
| 13:29.56 | penguinFunk | an IAX2 trunk between them |
| 13:30.02 | penguinFunk | snom300's mostly |
| 13:30.15 | penguinFunk | though some people in the other office have grandshit's |
| 13:30.22 | [TK]D-Fender | penguinFunk: IAX2 ALONE has audio issues regardless of codec in my experience.... not sure why... broken and stuttery |
| 13:30.30 | penguinFunk | hmm |
| 13:30.59 | [TK]D-Fender | penguinFunk: Snom is second rate audio right from the get-go with G711, and I'll leave a giant "no comment" hanging over grandstream.... |
| 13:31.18 | penguinFunk | when talking between the two asterisk boxes using alaw, quality is perfect. i briefly tried g729 and i could barely hear what my colleague was saying |
| 13:31.37 | [TK]D-Fender | penguinFunk: You are using almost EVERYTHING I would advise against. Congratulations, lemme gt you a door prize ;) |
| 13:31.46 | penguinFunk | pff |
| 13:32.06 | penguinFunk | i am dead against the grandstreams myself |
| 13:32.09 | bjweeks | Well, so am I but I don't have problems ;) |
| 13:32.09 | penguinFunk | what else is bad then? |
| 13:32.13 | [TK]D-Fender | penguinFunk: In this call with your colleage, what hardware, and protocols? |
| 13:32.17 | ai-a | We're here to please. |
| 13:32.38 | penguinFunk | sip, iax2, g729 |
| 13:32.49 | penguinFunk | snom300 -> snom300 |
| 13:33.03 | [TK]D-Fender | penguinFunk: Polycom > all. Linksys is "ok", and is Aastra, but are clearly 2nd place. Cisco beats them both on audio quality, but their SIP stack sorta sucks and costs too much |
| 13:33.32 | penguinFunk | polycom is really that much better than snom? |
| 13:33.35 | [TK]D-Fender | penguinFunk: Snom's bottom of the line... I really wouldn't get my hopes up.... they've been known for flakey firmware... |
| 13:33.44 | bjweeks | [TK]D-Fender: cordless phones connected to a 5 year old D-link ATA, trunk is IAX2 over the internet. can't beat that for suck :P |
| 13:33.45 | [TK]D-Fender | penguinFunk: Polycom kills jsut about everything out there. |
| 13:34.05 | penguinFunk | after a year of using grandstream, we are well impressed with the snoms |
| 13:34.08 | *** join/#asterisk luke-jr (n=luke-jr@2002:440d:6de2:0:20e:a6ff:fec4:4e5d) |
| 13:34.11 | penguinFunk | for outside calls |
| 13:34.13 | penguinFunk | alaw calls |
| 13:34.15 | penguinFunk | etc |
| 13:34.17 | penguinFunk | well chuffed |
| 13:34.42 | penguinFunk | so i guess the answer is use polycom is you want g729 to sound good |
| 13:34.42 | [TK]D-Fender | penguinFunk: Quick way to understand that last comment of yours... "shit looks really good... when compared to CRAP" <- |
| 13:34.48 | penguinFunk | if* |
| 13:34.53 | penguinFunk | lol |
| 13:34.57 | bjweeks | are granstreans THAT bad? :( |
| 13:35.05 | [TK]D-Fender | ~gs |
| 13:35.05 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 13:35.05 | *** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-aedda85069469ecb) |
| 13:35.10 | [TK]D-Fender | ~grandstream |
| 13:35.10 | jbot | methinks grandstream is the Yugo of VoIP hardware. Run. Run away now. |
| 13:35.13 | bjweeks | haha, point taken |
| 13:35.16 | [TK]D-Fender | bjweeks: YES |
| 13:35.40 | bjweeks | I guess I will stick with Linksys when I get some IP phones |
| 13:35.47 | penguinFunk | the only poor thing i have noticed about any of our setup is g729 calls |
| 13:35.53 | penguinFunk | we are more than happy with everything else |
| 13:35.57 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:35.57 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:36.03 | [TK]D-Fender | bjweeks / penguinFunk : where are you each located? |
| 13:36.07 | penguinFunk | UK |
| 13:36.13 | bjweeks | Arizona |
| 13:36.24 | [TK]D-Fender | penguinFunk: Go try a linksys then. They are very affordable in the UK |
| 13:36.35 | penguinFunk | i will keep that in mind thanks |
| 13:37.05 | [TK]D-Fender | bjweeks: For you three is NO reason to use anything except Polycom. North American pricing is on par with Every other even remotely acceptable choice. And thats for getting a BETTER product |
| 13:37.36 | [TK]D-Fender | penguinFunk: I haven't tried G729 on them SPECIFICALLY, but I owned an SPA-941. It is "acceptable". |
| 13:37.41 | [TK]D-Fender | there* |
| 13:38.00 | coppice | the linksys phones look really bad, though |
| 13:38.03 | [TK]D-Fender | bjweeks: www.telephonydepot.com |
| 13:38.19 | bjweeks | oh, hey I didn't know Polycoms were that cheap :/ I always wrote them off as too expensive |
| 13:38.25 | thewiizle | heh snoms |
| 13:38.34 | thewiizle | we have a load of polycoms here |
| 13:38.35 | thewiizle | 301 |
| 13:38.41 | thewiizle | 50 each :) |
| 13:38.43 | *** join/#asterisk FireMac (n=firemac@CPE000d88ae88b9-CM0011ae8bb0ee.cpe.net.cable.rogers.com) |
| 13:38.51 | [TK]D-Fender | coppice: against the SPA's : tinny speakerphone, SLIGHTLY tinny handset, base is too light and poor rubber feet means it'll shift under handset cord tension, poor use of their LCD. |
| 13:39.17 | coppice | painted like a 3 year old did it, and it wears off quickly |
| 13:39.28 | [TK]D-Fender | bjweeks: $< $90 for a PoE + Speakerphone. |
| 13:39.39 | [TK]D-Fender | coppice: Never heard that one... will add the the commentary. |
| 13:39.45 | penguinFunk | bye bye coppice |
| 13:39.51 | bjweeks | 87$! dang, I might skip getting a new ATA and just get polycoms |
| 13:40.18 | [TK]D-Fender | bjweeks: Don't forget to factor in the cost of either power bricks @ 17$ or PoE switch. |
| 13:40.28 | thewiizle | Polycom :( |
| 13:40.52 | bjweeks | anything is a step up from my dlink ata :(( |
| 13:40.58 | Qwell | thewiizle: If you don't like polycom - you're doing something wrong |
| 13:41.35 | thewiizle | Qwell, i hate polycom |
| 13:41.37 | thewiizle | HATE |
| 13:41.41 | Qwell | then you're doing something wrong |
| 13:41.47 | thewiizle | no, nothing being done wrong |
| 13:41.49 | thewiizle | just hate them |
| 13:42.22 | [TK]D-Fender | I had a top-of-the-line Aastra 57i CT as my desk phone here at the office. I took that as an "upgrade?) to my Polycom IP 600. Lets jsut say I'd have preferred my home bedsides's IP **301** to it hands down. |
| 13:42.51 | thewiizle | i have one of those aastra phones |
| 13:42.52 | lirakis | Qwell: i hate polycoms too... they are excellent quality.. but thier interfaces are not logical.. they are not user friendly in any way |
| 13:42.54 | coppice | penguinFunk: what a polite fellow you are, to abuse people in private messages :-) |
| 13:42.55 | thewiizle | with teh wireless handset |
| 13:43.21 | bjweeks | drama time again? |
| 13:43.27 | bjweeks | should I get Dr.Phil? |
| 13:43.34 | thewiizle | Get Jerry in here |
| 13:43.38 | [TK]D-Fender | lirakis: Configuration I agree is not friendly (web gui blows and should be REMOVED :p) but the USER interface is the best I've even seen. |
| 13:44.48 | lirakis | [TK]D-Fender: i have one of the "HD" 550's .. it looks like unknown callers a lot because it runs off the screen... the interface menu's are stacked deep.. and i just dont like soft buttons. |
| 13:45.24 | [TK]D-Fender | lirakis: What option do you actually have to go into menus for as a user? |
| 13:45.39 | lirakis | [TK]D-Fender: missed calls etc. |
| 13:45.57 | [TK]D-Fender | lirakis: thats *1* button to see your missed calls... what are you talking about? |
| 13:46.20 | [TK]D-Fender | lirakis: 1 button for : missed, placed,received, and speed-dials... |
| 13:46.33 | Bladerunner05 | I run asterisk -u asterisk -G asterisk but it return error using /var/run/asterisk.ctl permission denied. May I specify to write pid file in another location ? |
| 13:46.41 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-126-31-209.bflony.east.verizon.net) |
| 13:46.55 | lirakis | [TK]D-Fender: hmm.. maybe im confused.. (i dont have the phone on this desk) |
| 13:47.15 | [TK]D-Fender | lirakis: clearly |
| 13:47.30 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 13:48.05 | lirakis | [TK]D-Fender: i thought i had to do menu-> directory->placed calls | missed calls |
| 13:48.23 | [TK]D-Fender | lirakis: You thought wrong :) |
| 13:48.49 | [TK]D-Fender | lirakis: Step 1 : Down Arrow. Step 2. NONE! |
| 13:49.19 | bjweeks | 3. ... |
| 13:49.23 | bjweeks | 4. PROFIT! |
| 13:50.43 | [TK]D-Fender | lirakis: placed calls : Right Arrow. Latest call is listed first. Pressing Right Arrow again dials it. so who needs a dedicated redial button? Right-right. End of story. |
| 13:51.06 | *** join/#asterisk pardove (n=chatzill@80.191.113.132) |
| 13:51.21 | [TK]D-Fender | lirakis: recall the last person who called you? Left>right. |
| 13:51.57 | Bladerunner05 | I run asterisk -u asterisk -G asterisk but it return error using /var/run/asterisk.ctl permission denied. May I specify to write pid file in another location ? |
| 13:51.57 | [TK]D-Fender | lirakis: Haven't seena phone out there that can come even close to competing with Polycom on call handling. |
| 13:52.12 | Bladerunner05 | any ideas? |
| 13:52.21 | [TK]D-Fender | Bladerunner05: You're skipping like my old record player... don't make me smack you :p |
| 13:52.30 | coppice | I think [TK]D-Fender is a paid lobbyist for Polycom :-) |
| 13:52.42 | [TK]D-Fender | coppice: No... but I SHOULD be :) |
| 13:52.59 | [TK]D-Fender | coppice: As it stands I lobby for those WORTHY of it. |
| 13:53.10 | [TK]D-Fender | bjweeks: lirakis is already here ;) |
| 13:53.14 | pardove | is there any way to set the '#' key as dial key on fxs ports? |
| 13:53.28 | [TK]D-Fender | lirakis: But you're reforming now, right? |
| 13:53.38 | pardove | is there any way to set the '#' key as dial key on fxs ports? like most ATAs |
| 13:53.38 | coppice | I used to hate Polycom's conferencing phones, but I've never used (never even seen) their IP ones |
| 13:53.44 | [TK]D-Fender | pardove: what does "dial key" mean? |
| 13:54.41 | keith4_ | what, your phone doesn't have a dial key? :-P |
| 13:54.44 | tzafrir | Bladerunner05, asterisk -U asterisk |
| 13:54.45 | pardove | [TK]D-Fender: as send key. to speed up the dialing. |
| 13:55.13 | bjweeks | this must not be a US thing... |
| 13:55.14 | tzafrir | and - set varrundir in asterisk.conf |
| 13:55.22 | [TK]D-Fender | pardove: You use variable length dialplan patterns? |
| 13:55.25 | pardove | keith4_: now when using fxs ATAs you can hit # at the end of your dial number. |
| 13:55.33 | pardove | [TK]D-Fender: yes |
| 13:56.20 | [TK]D-Fender | pardove: For Zaptel FXS the only way is to use "immediate=yes", and dump them into an IVR and start processing. |
| 13:56.31 | [TK]D-Fender | pardove: less than convenient. |
| 13:56.36 | keith4_ | like the check mark button on a snom? |
| 13:56.41 | [TK]D-Fender | pardove: but it would technically work. |
| 13:57.00 | *** join/#asterisk edwin_quijada (n=m@200.88.116.25) |
| 13:57.06 | pardove | [TK]D-Fender: is there any better way? |
| 13:57.07 | [TK]D-Fender | keith4_: We're talking about Zaptel FXS and an analog phone here... |
| 13:57.15 | [TK]D-Fender | pardove: not on Zaptel FXS. |
| 13:57.23 | [TK]D-Fender | pardove: ATA's support that however. |
| 13:57.31 | [TK]D-Fender | pardove: Decent ones anyways. |
| 13:58.16 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 13:58.54 | pardove | where in the code is responsible for this? i think hacking the code makes it possible. because the # key is not used as dial pattern any where |
| 13:59.00 | anonymouz666 | ${ANI} is only configured if I use the setcallerid(clid|a) ? Otherwise I can't just use the ${ANI}? |
| 13:59.13 | keith4_ | [TK]D-Fender: don't yell at me, but I don't understand what he wants to do |
| 13:59.40 | anonymouz666 | unless i set ${ANI} manually |
| 13:59.41 | keith4_ | setting immediate=yes dumps you right into a context without a dial tone, doesn't it? |
| 13:59.50 | [TK]D-Fender | keith4_: I wasn't yelling at you. He was quite clear in his request though... |
| 13:59.52 | thewiizle | so |
| 13:59.59 | thewiizle | about this sip prune realtime that i cant find anything on google about |
| 14:00.10 | keith4_ | [TK]D-Fender: it was a pre-emptive "don't yell at me" ;-) |
| 14:00.14 | Bladerunner05 | <tzafrir> : I do but permission denied in /var/run |
| 14:00.22 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 14:00.22 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:00.30 | tzafrir | right, bacause asterisk can't write there |
| 14:00.44 | [TK]D-Fender | keith4_: Want to blame me for WW3 while you're at it? That hasn't happend yet either, but I'm clearly to blame! |
| 14:00.52 | tzafrir | so set the varrundir to /var/run/asterisk , create such a directory, and permit asterisk to write there |
| 14:00.55 | keith4_ | lol |
| 14:00.58 | syzygyBSD | [TK]D-Fender: if he doesn't I will |
| 14:01.03 | pardove | keith4_: if you have variable length dialplan, on zaptel fxs ports when you dial a number you should wait for some seconds to have your dialed number sent to * |
| 14:01.11 | [TK]D-Fender | syzygyBSD: feelin' the love... |
| 14:01.25 | syzygyBSD | it always starts cuz of love... |
| 14:01.37 | [TK]D-Fender | pardove: This isn't something I would consider worthwhile to try to hack in.... |
| 14:01.49 | keith4_ | pardove: you have to wait, because it's not sure you're done entering digits? |
| 14:02.05 | keith4_ | seems like you could maybe re-arrange the dialplan so there aren't such ambiguities, no? |
| 14:02.06 | [TK]D-Fender | syzygyBSD: Yup... GWB would *love* to have Iran glow in the dark.... |
| 14:02.25 | syzygyBSD | lol.. you really think he is running the country? |
| 14:02.35 | pardove | keith4_: so the # key in all ATAs means that i'm done with dialing and send it to the * |
| 14:02.56 | keith4_ | gotcha |
| 14:02.58 | pardove | keith4_: i want this behavior on zaptel fxs ports |
| 14:03.03 | keith4_ | so it *is* like the checkmark on snoms |
| 14:03.08 | keith4_ | ...sort of |
| 14:03.11 | [TK]D-Fender | keith4_: issue is he has already confirmed that he has variable length pattern matches. this precludes any kind of prediction of length for speedier processing. |
| 14:03.12 | pardove | yeah |
| 14:04.04 | [TK]D-Fender | pardove: I'd say that in all practicality, this is not something to pursue with Zaptel. |
| 14:04.26 | [TK]D-Fender | pardove: I have always recommended against Zaptel FXS for normal phone usage. |
| 14:04.38 | pardove | no, it's something in chan_zap i think |
| 14:06.13 | *** join/#asterisk mindCrime__ (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 14:10.03 | stmaher | hi guys.. how do i remove call limit counter from a sip trunk? |
| 14:10.22 | *** join/#asterisk ussrback (n=MAX@81.95.160.147) |
| 14:10.26 | ussrback | Hi all |
| 14:10.38 | [TK]D-Fender | stmaher: Tried reloading sip, or chan_sip? |
| 14:11.53 | ussrback | there is some module called confcall, but its released for Asterisk 1.2 |
| 14:11.58 | tzanger | whirred from the amd driver course |
| 14:12.10 | ussrback | is it possible to use it for 1.4? |
| 14:12.44 | stmaher | [TK: The problem im having is it repeats calls when the IMS side has terminated the call .. |
| 14:13.00 | ussrback | hi [TK]D-Fender |
| 14:13.07 | [TK]D-Fender | stmaher: What is this "it" that is repeating calls? |
| 14:13.18 | stmaher | sorry.. |
| 14:13.24 | stmaher | I should explain better.. |
| 14:13.30 | [TK]D-Fender | stmaher: indeed. |
| 14:13.44 | stmaher | I have a setup of sipphone -> asterisk -> IMS (VoiceXML interpreter) |
| 14:13.54 | ussrback | http://www.freeswitch.org/node/75 <-- Is it possible to install this application for Asterisk 1.4 ? |
| 14:14.13 | stmaher | when I make a SIP call to the asterisk box.. it routes it to the IMS |
| 14:14.44 | stmaher | The problem is when the call finishes with the IMS asterisk keeps the sip phone session and makes another call to the IMS |
| 14:14.53 | stmaher | ROFL |
| 14:15.19 | *** join/#asterisk tsurko (n=tsurko@213.91.216.130) |
| 14:16.32 | [TK]D-Fender | stmaher: Remember that pastebin I asked you for... TWO HOURS AGO? Now would be a good time.... |
| 14:16.41 | *** join/#asterisk clyrrad (n=darryl@CPE000802212b48-CM0011aea484a4.cpe.net.cable.rogers.com) |
| 14:16.50 | stmaher | LOL.. |
| 14:16.53 | stmaher | One sec :-) |
| 14:17.02 | creativx | morning [TK]D-Fender |
| 14:17.04 | [TK]D-Fender | ussrback: I believe Anthm recoded it for 1.4 Ask him whenever he shows up here. |
| 14:17.05 | stmaher | ~pb |
| 14:17.05 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 14:17.17 | penguinFunk | coppice: gone quiet? |
| 14:17.36 | [TK]D-Fender | penguinFunk: I think you burnt him out. |
| 14:17.36 | thewiizle | penguinFunk, bored? |
| 14:17.52 | thewiizle | [TK]D-Fender |
| 14:17.54 | thewiizle | busy? |
| 14:17.54 | penguinFunk | in more ways than one it seems |
| 14:18.04 | penguinFunk | thewiizle: not any more :] |
| 14:18.09 | thewiizle | got any IVR exmaples i could leech over? |
| 14:18.21 | syzygyBSD | I need an upgraded server tested before I switch over to it if you are really bored... |
| 14:18.30 | ReDNeQ | my customer is complaining about not being able to make outbound calls. The calls are going out randomly over zap |
| 14:18.46 | *** join/#asterisk Brixius (n=Brixius@PDN-VBA.OnvoyInc.fw.onvoy.net) |
| 14:18.47 | ReDNeQ | any idea in what direction to look at |
| 14:18.58 | syzygyBSD | ReDNeQ: does he have enough zap lines? are they in use when the calls fail? |
| 14:18.59 | [TK]D-Fender | thewiizle: not much to say for IVR's that hasn't been put in print all over the place... |
| 14:19.19 | [TK]D-Fender | ReDNeQ: Try looking in the direction of a PASTEBIN |
| 14:19.20 | ReDNeQ | syzygyBSD: they have 4 zap channels |
| 14:19.21 | [TK]D-Fender | ~pb |
| 14:19.21 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 14:19.23 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^ |
| 14:19.41 | stmaher | [TK]D-Fender http://pastebin.com/d34075bcc thank you :l-) |
| 14:19.58 | ReDNeQ | [TK]D-Fender, i cant see the failure.. So ok, ill try this with someone elses esys |
| 14:20.37 | [TK]D-Fender | stmaher: I asked for CLI output, not debug log file output... |
| 14:20.52 | [TK]D-Fender | ReDNeQ: Cool... we can't see the failure either ;) |
| 14:21.03 | stmaher | Opps.. sorry.. I thought the CLI outputted to the log file.. |
| 14:21.32 | stmaher | the output seems to be the same for the CLI as the log file has |
| 14:21.46 | thewiizle | cant find anything on the web |
| 14:21.52 | thewiizle | nothing that works anyhow |
| 14:21.57 | ReDNeQ | yeh yeh i got yah |
| 14:21.58 | clyrrad | Can anyone recommend a good book for Asterisk 1.4? |
| 14:22.01 | [TK]D-Fender | thewiizle: Somes in THE BOOK, and tons on the web.... |
| 14:22.05 | Qwell | ~book |
| 14:22.06 | jbot | Asterisk: The Future of Telephony 2nd Edition --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 14:22.06 | [TK]D-Fender | clyrrad: the BOOK |
| 14:22.14 | thewiizle | im lookin |
| 14:22.16 | Qwell | ooo, there's a pdf now? |
| 14:22.26 | [TK]D-Fender | Qwell: Has been for semeral weeks... |
| 14:22.29 | Qwell | oh |
| 14:22.41 | Corydon76-dig | [TK]D-Fender: uh, one week |
| 14:22.51 | ussrback | I have some .c file for module. How can i build new module? |
| 14:22.53 | clyrrad | [TK]D-Fender: thanks checking into that |
| 14:22.59 | [TK]D-Fender | Corydon76-dig: No.. plural. I never said "publicly annonced". |
| 14:23.06 | ReDNeQ | [TK]D-Fender, the biggest problem for me is the random times |
| 14:23.23 | Corydon76-dig | I still prefer my dead tree edition |
| 14:23.24 | ReDNeQ | they have not maxed out their channels, but if they did what would the error say? |
| 14:23.31 | [TK]D-Fender | Corydon76-dig: I've had it for nearly 3, and it was not particularly hidden on O'Reilly's either... |
| 14:23.52 | clyrrad | haha nice! I loved ATFOT first edition - nice to see there is a new one out - thanks guys, will be sure to buy this one |
| 14:23.58 | [TK]D-Fender | Corydon76-dig: Just that we avoided making it "public knowledge" at jsmith's request |
| 14:24.26 | Corydon76-dig | I'd still prefer that people show support for the authors by buying a dead tree copy |
| 14:24.49 | [TK]D-Fender | ussrback: I told you, anthm made a 1.4 version of this. ask him when he's here |
| 14:25.15 | [TK]D-Fender | Corydon76-dig: And why do you think thats the first link & reference in the jbot announcement ;) |
| 14:25.54 | [TK]D-Fender | Corydon76-dig: And a lot of people like to buy the dead-tree version, and more power to them for doing so. |
| 14:26.23 | clyrrad | I actually prefer a hard copy instead of a PDF, can sit at the couch and read in comfort :D |
| 14:26.32 | ussrback | please, help me when im trying to install addons i got errors -> http://sial.org/pbot/27862. |
| 14:27.30 | thewiizle | tk gunna have a nice post for you in a sec |
| 14:27.50 | thewiizle | http://pastebin.ca/724316 |
| 14:27.56 | thewiizle | does that look like it should work? |
| 14:28.41 | ReDNeQ | hey [TK]D-Fender : http://www.pastebin.ca/724318 |
| 14:29.10 | [TK]D-Fender | thewiizle: Well you have no error handling in there, you can dial 6000 to go in circles..... |
| 14:29.25 | [TK]D-Fender | thewiizle: You should avoide running IVR's on anything except "s" |
| 14:29.41 | [TK]D-Fender | thewiizle: but "legal choices" should work. |
| 14:29.44 | thewiizle | well its meant to be a quick IVR that will connect me to different hold music |
| 14:29.54 | thewiizle | it connects my call |
| 14:29.59 | [TK]D-Fender | thewiizle: Dial something illegal = get hung up on. |
| 14:30.04 | ussrback | please, help me when im trying to install addons i got errors -> http://sial.org/pbot/27862. |
| 14:30.09 | thewiizle | plays (after-the-tone) but when i key press nothing happens.. |
| 14:30.13 | thewiizle | can i monitor for DTMF on the CLI ? |
| 14:31.26 | *** join/#asterisk coppice (n=chatzill@153.201.17.210.dyn.pacific.net.hk) |
| 14:31.37 | [TK]D-Fender | ReDNeQ: what kind of zap channel is that? |
| 14:31.50 | [TK]D-Fender | ReDNeQ: Looks like it dialed, was answered, and then hung up.... |
| 14:31.53 | ReDNeQ | [TK]D-Fender, its a tdm800 |
| 14:32.10 | ReDNeQ | [TK]D-Fender, THATS EXACTLY MY POINT! ;) |
| 14:32.15 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 14:32.24 | [TK]D-Fender | ReDNeQ: pastebin your zapata.conf |
| 14:32.28 | ReDNeQ | they say sometimes it takes up to 10-15 secs before they get answer |
| 14:32.33 | ReDNeQ | then it just hangs up |
| 14:32.35 | ReDNeQ | ok here it comes |
| 14:33.01 | [TK]D-Fender | horrors* |
| 14:33.09 | [TK]D-Fender | ReDNeQ: ALL of it... |
| 14:33.50 | ReDNeQ | ; Zapata telephony interface |
| 14:33.51 | ReDNeQ | ; |
| 14:33.51 | ReDNeQ | ; Configuration file |
| 14:33.51 | ReDNeQ | [trunkgroups] |
| 14:33.51 | ReDNeQ | [channels] |
| 14:33.53 | ReDNeQ | language=en |
| 14:33.55 | ReDNeQ | context=from-zaptel |
| 14:33.57 | ReDNeQ | signalling=fxs_ks |
| 14:33.58 | creativx | rofl. |
| 14:33.58 | [TK]D-Fender | ReDNeQ: PASTEBIN! |
| 14:33.59 | ReDNeQ | rxwink=300 ; Atlas seems to use long (250ms) winks |
| 14:33.59 | clyrrad | oh no |
| 14:34.01 | ReDNeQ | DOE |
| 14:34.03 | ReDNeQ | sorry |
| 14:34.05 | ReDNeQ | wrong paste |
| 14:34.07 | ReDNeQ | SORRY SORRY |
| 14:34.08 | De_Mon | this is #asterisk not #pastebin! |
| 14:34.09 | ReDNeQ | i have multi windows open |
| 14:34.12 | thewiizle | lol |
| 14:34.15 | bjweeks | epic |
| 14:34.50 | [TK]D-Fender | ReDNeQ: Congratuations, Odds are you have 1 IRC channel window and 10 other possible windows that WOULDN'T have spammed us, and you beat the 10-to-1 odds! |
| 14:35.02 | ReDNeQ | http://www.pastebin.ca/724324 |
| 14:35.07 | ReDNeQ | Yep Im good like that |
| 14:35.10 | ReDNeQ | ;l? |
| 14:35.37 | [TK]D-Fender | ReDNeQ: and I said ALL OF IT. Where's the INCLUDED file? |
| 14:35.52 | ReDNeQ | all ok i will create one big pastebin |
| 14:36.00 | ReDNeQ | i was putting it in another |
| 14:37.21 | creativx | well i'll be damned |
| 14:37.41 | creativx | [TK]D-Fender: it seems i fixed my problem =) patched app_Queue to support call-limit for members.. and voilahhhhh |
| 14:38.07 | [TK]D-Fender | creativx: Suppose thats one approach... |
| 14:41.02 | thewiizle | tk man sorry to keep hassling u, ive changed this IVR to use, s,1, etc etc |
| 14:41.19 | thewiizle | is there any way to check and see if asterisk is receiving the DTMF key press ? |
| 14:41.35 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 14:41.43 | thewiizle | because its reaching WaitExten,15 and then doing nothing but hanging up as ive told it to |
| 14:42.37 | mvanbaak | heya all |
| 14:42.50 | creativx | [TK]D-Fender: indeed.. i'm not up to upgrading from 1.2 yet |
| 14:42.51 | mvanbaak | in asterisk 1.2, can you alter the sip payload size ? |
| 14:43.04 | mvanbaak | I thought that it was 20ms and not ajustable |
| 14:43.06 | mvanbaak | right ? |
| 14:43.13 | *** join/#asterisk Aughey (n=jha@64.219.54.125) |
| 14:43.22 | thewiizle | depends on the codec doesnt it? |
| 14:43.30 | [TK]D-Fender | mvanbaak: that'd be RTP, not SIP. codecs.conf <- |
| 14:43.35 | ReDNeQ | http://www.pastebin.ca/724335 |
| 14:43.41 | ReDNeQ | there you go that is all of it |
| 14:43.44 | ReDNeQ | all of them |
| 14:43.47 | ReDNeQ | indexed |
| 14:44.00 | thewiizle | woohoo |
| 14:44.02 | thewiizle | it was DTMF |
| 14:44.07 | thewiizle | changed to inband all is good |
| 14:44.42 | ReDNeQ | i have 5 lines plugged into this 8 port card |
| 14:44.57 | mvanbaak | [TK]D-Fender: I'm using ulaw and alaw and gsm |
| 14:45.06 | [TK]D-Fender | ReDNeQ: looks fine. Test your ports with a line splitter & analog phone. |
| 14:45.06 | mvanbaak | I dont see anything I can set in codecs.conf |
| 14:45.32 | ReDNeQ | [TK]D-Fender,: when i use plain jane phone im able to make calls out just fine |
| 14:45.53 | ReDNeQ | how do i send diagnostics down card? |
| 14:45.59 | ReDNeQ | may card ports are bad? |
| 14:46.02 | [TK]D-Fender | mvanbaak: What ver of *? |
| 14:46.09 | mvanbaak | [TK]D-Fender: 1.2.23 |
| 14:46.10 | [TK]D-Fender | ReDNeQ: Possibly |
| 14:46.17 | [TK]D-Fender | mvanbaak: Read up -> http://www.voip-info.org/wiki/view/Asterisk+codecs |
| 14:46.24 | [TK]D-Fender | mvanbaak: Bad news for you... |
| 14:46.38 | mvanbaak | I'm fired ? |
| 14:46.40 | coppice | ReDNeQ: if they are simple phone lines, why are you using _ks instead of _ls? |
| 14:46.49 | [TK]D-Fender | "Asterisk 1.2 and earlier only supports 20ms packetization in RTP-based protocols like SIP and MGCP" |
| 14:46.58 | mvanbaak | meh |
| 14:47.06 | [TK]D-Fender | mvanbaak: upgrade time |
| 14:47.10 | mvanbaak | not bad news for me, bad news for NEC-Philips |
| 14:48.26 | *** join/#asterisk Trionnis (n=blah@65-117-172-195.dia.static.qwest.net) |
| 14:48.56 | thewiizle | oh this is too cool |
| 14:49.05 | thewiizle | ive just built an Asterisk PBX Radio station |
| 14:49.21 | Trionnis | anyone here familiar with the limiting settings of Dial()? I need a bit of direction since the wiki isn't all that clear |
| 14:49.26 | drako | [Oct 3 16:48:56] NOTICE[11562]: chan_iax2.c:2848 __auto_congest: Auto-congesting call due to slow response |
| 14:49.33 | drako | need to get rid of this |
| 14:49.42 | drako | the qualify is set to NO |
| 14:49.45 | thewiizle | Dial(TECH/EXTEN),timeout,do_next |
| 14:49.46 | thewiizle | :) |
| 14:49.46 | drako | keep getting this... |
| 14:50.01 | Trionnis | little more complex than that |
| 14:50.02 | Trionnis | :) |
| 14:50.18 | Trionnis | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Dial |
| 14:50.24 | Trionnis | have a look at the "L" parameter |
| 14:50.26 | *** join/#asterisk saftsack (n=saftsack@84.167.121.47) |
| 14:50.32 | Trionnis | that's the stuff I need a bit of pointer on :) |
| 14:50.34 | [TK]D-Fender | Trionnis: dated... CARBON DATED even... "show application dial" |
| 14:50.39 | thewiizle | limit call to x |
| 14:51.20 | Trionnis | Andrew, that's the same thing in the wiki... what I don't get is how to set the variables |
| 14:51.55 | Trionnis | are those just supposed to be global variables? e.g. Set()? |
| 14:52.42 | Aughey | I need some help. I have a A200 card that has quit on me. in dmesg I get "wanpipe1: No FXO/FXS modules are found!" (among other error messages and ztcfg reports "ZT_CHANCONFIG failed on channel 1: No such device or address" All of this has been working for over a year until this morning. No changes. |
| 14:53.18 | [TK]D-Fender | Trionnis: Exactly |
| 14:53.23 | Trionnis | ok |
| 14:53.24 | Trionnis | thanks |
| 14:54.37 | tzafrir | Aughey, what changed this morning? |
| 14:54.45 | Aughey | absolutely nothing |
| 14:54.58 | mvanbaak | Aughey: I think it's time to slapin the sparepart |
| 14:55.02 | tzafrir | except the date, that is? |
| 14:55.11 | Aughey | it'd be nice if I had a spare part |
| 14:55.17 | coppice | and the smoking PCI slot |
| 14:56.08 | Aughey | you know, guys, that's not what I wanted to hear. I want to hear, ahh, just run "ztjfoiew" and it'll all be fixed |
| 14:56.14 | Trionnis | oh... you can't let the magic smoke out... that's what makes it work! |
| 14:56.23 | coppice | Aughey: I guess you've tried obvious things like reseating the card? |
| 14:56.52 | Aughey | the computer has been reset. I don't think we've removed power. I don't have physical access to it at the moment |
| 14:56.56 | [TK]D-Fender | Electronic devices depend on their smoke... if it gets released, things stop working! |
| 14:57.02 | Trionnis | absolutely |
| 14:57.22 | coppice | the dumb thing with the A200 (other than using the wrong connector) is the cards push upwards. I always wonder about the effects of gravity and thermal cycling on anything like that |
| 14:59.09 | coppice | though the fun one is when the edge connector completely wears away with vibration :-) |
| 15:00.17 | *** join/#asterisk jingles (n=dfbarth@39.183.dowl.anc.borealisbroadband.net) |
| 15:02.30 | stmaher | [TK]D-Fender: I will have to come back to that problem.. But maybe you could help me with tranfer problems.. I keep getting a 603 declined by the asterisk box when it tries to transfer the call.. |
| 15:02.55 | [TK]D-Fender | stmaher: .... pastebin.... |
| 15:03.09 | stmaher | comeing right up |
| 15:03.21 | [TK]D-Fender | coppice: You should never leave your server too close to your personal... "toys" ;) |
| 15:04.31 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 15:05.15 | coppice | the "personal toy" I had wear through edge connectors was an airliner. real airborne equipment doesn't sue direct edge connectors, but we had test systems flying for a few months that wore through. strange things is they can wear while making perect contact, so you only know about it when they finally go right through |
| 15:05.39 | stmaher | [TK]D-Fender: http://pastebin.com/d2fd502f2 thank you so much for looking into this! |
| 15:05.49 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 15:06.30 | [TK]D-Fender | stmaher: Found no matching peer or user for '10.0.0.151:64908' <- already not good... unauthed calls |
| 15:06.51 | stmaher | we are just doing some interop tests.. so if its unauthed its ok |
| 15:07.03 | *** join/#asterisk nfi|ermes (n=ErMeS@217.220.121.62) |
| 15:07.52 | [TK]D-Fender | stmaher: To: "Stephen"<sip:stephen@127.0.0.1>;tag=48182750 <- like talking to yourself? |
| 15:08.09 | creativx | aah god damn this is nice.. no more multiple calls to queue members. |
| 15:08.34 | [TK]D-Fender | stmaher: Transfer to 1961 in default <- care to share the dialplan associated with this? |
| 15:08.44 | Trionnis | My dad always says... "It's ok to talk to yourself... It's even ok to answer yourself. If you have to ask yourself to repeat something, seek professional help." |
| 15:08.52 | [TK]D-Fender | creativx: Easy fix in pure dialplan for that you know... |
| 15:09.35 | *** join/#asterisk dlynes (n=dlynes@d154-20-9-152.bchsia.telus.net) |
| 15:09.39 | coppice | "Why do you think you are God?" |
| 15:09.41 | coppice | "I found that when I prayed I was talking to myself." |
| 15:09.44 | thewiizle | lol Trionnis |
| 15:09.47 | thewiizle | i like that |
| 15:10.16 | creativx | [TK]D-Fender: I dont know.. and i couldnt figure out how.. :) |
| 15:10.29 | creativx | since I dont use queue agents but dyn members |
| 15:10.40 | [TK]D-Fender | creativx: You should change your nick then... "LackOfCreativityX" |
| 15:10.46 | [TK]D-Fender | :p |
| 15:10.59 | Trionnis | uh... here goes the creativity rant |
| 15:11.32 | [TK]D-Fender | Gonna get me a shotgu and kill all the whiteys I seeeeeeeee!!!! |
| 15:11.33 | stmaher | [TK]D-Fender: thank you http://pastebin.com/d38d6ba72 |
| 15:11.54 | creativx | [TK]D-Fender: hehe, I usually live up to my nick |
| 15:12.10 | creativx | not saying I didnt _try_ to figure out how to do it with dialplan magic |
| 15:12.18 | coppice | names are not chosen to illuminate, but to divert attention |
| 15:12.19 | [TK]D-Fender | stmaher: Does that look like [defaul] to YOU? |
| 15:12.35 | creativx | indeed coppice ;) |
| 15:12.52 | [TK]D-Fender | stmaher: And words cannot express how much I *DON'T* support FreePBX in here.... |
| 15:13.13 | stmaher | Opps.. :-( my bad |
| 15:13.56 | stmaher | Im not using the Freepbx part.. |
| 15:13.56 | [TK]D-Fender | stmaher: You've collected at least 3 "bads" this morning... you're nearly ready to claim your door-prize on the way out ;) |
| 15:14.19 | [TK]D-Fender | stmaher: And my faith in that comment would wane if it HAD any leeway with me to begin with... |
| 15:14.41 | stmaher | My apologies.. |
| 15:15.03 | stmaher | Thank you for your time.. |
| 15:16.03 | [TK]D-Fender | stmaher: But I will accept your actually showing me the context that was being called for there... |
| 15:16.12 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 15:17.27 | *** part/#asterisk Brixius (n=Brixius@PDN-VBA.OnvoyInc.fw.onvoy.net) |
| 15:17.42 | stmaher | [TK]D-Fender: so there is no hope of asking you for further advice? |
| 15:18.15 | [TK]D-Fender | stmaher: I have jsut asked you to provide the actual context that was being called by that redirect attempt.... |
| 15:18.32 | stmaher | Oh right sorry.. |
| 15:18.43 | stmaher | Can you please rephrase that? |
| 15:19.59 | [TK]D-Fender | stmaher: ...... |
| 15:20.03 | stmaher | or what is it exactly you want to see .. |
| 15:20.12 | stmaher | sorry.. Im not too firmiliar with the terminology |
| 15:20.15 | [TK]D-Fender | [TK]D-Fender>stmaher: Does that look like [default] to YOU? |
| 15:20.19 | Trionnis | you don't support FreePBX... but you'll support TrixBox, right? |
| 15:20.39 | [TK]D-Fender | [TK]D-Fender>stmaher: Transfer to 1961 in default <- care to share the dialplan associated with this? |
| 15:20.40 | syzygyBSD | meh, read the topic |
| 15:20.56 | [TK]D-Fender | stmaher: And you gave me THIS : [ Context 'from-sip-external' created by 'pbx_config' ] <----------- |
| 15:21.36 | stmaher | Ahhhhh.. I think i might know whats going on.. |
| 15:21.47 | stmaher | I dont think i have a default entry that covers this problem |
| 15:21.54 | [TK]D-Fender | stmaher: Yes, failure to read + unauthed calls. |
| 15:22.51 | stmaher | its just three lines.. |
| 15:23.13 | stmaher | playback vmgoodbye 2 macro(hangup) and include ext-local |
| 15:23.48 | [TK]D-Fender | stmaher: And you think that actually says ANYTHING to me the way your wrote that? |
| 15:24.27 | stmaher | I didnt write that.. it was there by default :-) |
| 15:24.30 | *** join/#asterisk aninoSAdilim (n=a@58.69.243.203) |
| 15:24.39 | stmaher | Kidding.. |
| 15:24.44 | stmaher | want me to pastebin it? |
| 15:25.04 | [TK]D-Fender | stmaher: funny.. with out "exten =>" and a whole bunch of CRUCIAL stuff what you just showed me is useless |
| 15:25.17 | *** join/#asterisk jingles (n=dfbarth@39.183.dowl.anc.borealisbroadband.net) |
| 15:25.50 | stmaher | LOL.. Im just learning Asterisk.. it was a miracle i got it to talk basic sip with a sip phone |
| 15:26.11 | [TK]D-Fender | stmaher: Using FreePBX doesn't qualify as learning *. |
| 15:26.37 | stmaher | too true.. but we needed it this interop done quickly and one of the guys suggested it in the office.. |
| 15:26.49 | stmaher | If i had the time.. I would sit down and do this from scratch.. |
| 15:26.58 | [TK]D-Fender | stmaher: Go shoot them now. If there are any survivors, shoot AGAIN. |
| 15:27.03 | stmaher | ROFL.. |
| 15:27.10 | [TK]D-Fender | stmaher: ....this is where you should consider : |
| 15:27.11 | stmaher | Trust me.. You think I like the guy who suggested this? |
| 15:27.12 | [TK]D-Fender | ~hafc |
| 15:27.13 | jbot | extra, extra, read all about it, hafc is hire a freaking consultant. Look, if you're having difficulty understanding what you're doing and need a solution soon, you will be far better off hiring a competent consultant than continuing to pull your hair out. |
| 15:27.21 | [TK]D-Fender | stmaher: |
| 15:27.25 | stmaher | : I would shoot |
| 15:27.43 | stmaher | is there a shortcut to writing your name before text in irc? |
| 15:28.14 | [TK]D-Fender | stmaher: "Who's the more foolish: The fool, or the fool who follows him?" - Obi-Wan Kenobi |
| 15:28.40 | stmaher | Old men dont use viagra cause their impotent its because old women are so very ugly - Jimmy Carr |
| 15:28.45 | stmaher | :-) |
| 15:28.48 | [TK]D-Fender | stmaher: depends on your client. usually most autocomplete on <tab> |
| 15:28.59 | stmaher | [TK]D-Fender, nice one :-) |
| 15:29.05 | stmaher | tab works :-) |
| 15:29.10 | coppice | [TK]D-Fender: if the fool is rich, the one following might be very smart :-) |
| 15:29.34 | stmaher | coppice, rich? you must be joking.. im fresh out of college.. |
| 15:29.38 | stmaher | ie broke for my next 3 lifetimes |
| 15:30.10 | [TK]D-Fender | stmaher: Careful... China is regulating reincarnation... |
| 15:31.12 | stmaher | Anyway.. |
| 15:31.25 | stmaher | Can you please help me with the concept then.. ? |
| 15:31.40 | stmaher | If i do a transfer.. I have to have an entry in the extensions.conf? |
| 15:32.04 | stmaher | [TK]D-Fender, sorry keep forgetting to use tab.. |
| 15:32.26 | coppice | [TK]D-Fender: I haven't heard of anyone around here receiving government orders to remain dead after dying |
| 15:32.37 | stmaher | [TK]D-Fender, is there a way to get default to accept any sip connection and allow transfer? |
| 15:33.14 | coppice | insecurity might be, though |
| 15:33.20 | [TK]D-Fender | stmaher: First your calls are coming in UNAUTHED. Second I've asked you to PASTEBIN [default] **3 times now** |
| 15:33.36 | stmaher | Coming right up sir.. |
| 15:33.51 | syzygyBSD | [TK]D-Fender: it is a freepbx issue, have them support it... |
| 15:34.13 | ai-a | We're here to serve and please. |
| 15:34.32 | stmaher | syzygyBSD, SHHHHHHHHHH.. besides.. after all we have been through in sure fender now likes :-) (kidding) |
| 15:34.33 | [TK]D-Fender | syzygyBSD: It is, and liekly the ultimate response, but I figured I'd bring all the incriminating evidence out into the open before I begin executing people.... |
| 15:36.09 | syzygyBSD | I hope you drew blood before sheathing that |
| 15:36.37 | stmaher | [TK]D-Fender, http://pastebin.com/m7fc6ed12 |
| 15:36.45 | [TK]D-Fender | syzygyBSD: That's what chiburi is for.... |
| 15:37.02 | [TK]D-Fender | stmaher: and 'ext-local' now.... |
| 15:37.07 | syzygyBSD | not like this guy though, http://scienceblogs.com/neurotopia/2007/10/martial_idiocy.php |
| 15:37.15 | stmaher | cant seem to find an entry like that |
| 15:37.44 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 15:37.55 | syzygyBSD | stmaher: why not just pastebin your entire config? |
| 15:38.01 | syzygyBSD | it would save a lot of time |
| 15:38.27 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:39.34 | stmaher | syzygyBSD, My entire config http://pastebin.com/d65065eb2 |
| 15:39.42 | stmaher | [TK]D-Fender, My entire config.. http://pastebin.com/d65065eb2 |
| 15:40.05 | [TK]D-Fender | stmaher: Welcome to "dead end" |
| 15:40.12 | stmaher | thought that.. |
| 15:40.18 | stmaher | so there is no ext-local.. |
| 15:40.24 | stmaher | Im lernding :-) |
| 15:40.52 | *** join/#asterisk grEvenX (n=even@pc105-222.ktv.no) |
| 15:40.58 | [TK]D-Fender | stmaher: your calls are landing in a useless context. You should not be pioneering this sort of project if it has any kind of close dead-line attached to it. I strongly suggest you get a consultant. |
| 15:41.17 | [TK]D-Fender | stmaher: And FreePBX will have you fighting every step of the way. |
| 15:41.18 | drako | [Oct 3 16:48:56] NOTICE[11562]: chan_iax2.c:2848 __auto_congest: Auto-congesting call due to slow response <- Any idea how to make iax try longer? |
| 15:41.21 | *** join/#asterisk Dan0maN_Work (n=dschuh@64.149.174.136) |
| 15:42.10 | stmaher | [TK]D-Fender, I just have these two to complete.. |
| 15:42.27 | stmaher | [TK]D-Fender, Transfers and that weird other problem from earlier |
| 15:43.04 | pardove | [TK]D-Fender: i done it! now hitting # sends the dial string on zaptel fxs |
| 15:43.19 | [TK]D-Fender | pardove: How did you go about it? |
| 15:43.33 | pardove | just a little hack on channel.c |
| 15:43.50 | *** join/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) |
| 15:44.22 | stmaher | [TK]D-Fender, do i have to change the default to accept an incomming call to do a transfer? |
| 15:44.46 | [TK]D-Fender | stmaher: your calls are coming in UNAUTHED. thats the problem. |
| 15:44.54 | pardove | [TK]D-Fender: i'm going to optimize the code ;-0 |
| 15:45.12 | [TK]D-Fender | pardove: And its isolated to chan_zap on dial only? |
| 15:45.48 | flujan | Hi guys, I need to implement this behavior on asterisk: if the queue agent is on a call do not try to place a second call to it. When I set the call-limit to 1 it works flawless... I had have the incoming calls delivered in less than one second. |
| 15:45.53 | stmaher | [TK]D-Fender, ok.. is there a way to make anything authed? |
| 15:45.59 | pardove | no it also works on disa which i also had this problem on it |
| 15:46.08 | stmaher | [TK]D-Fender, or do i have to make an entry somewhere? |
| 15:46.14 | flujan | Now, I have a queue of two level. People from the first level, answers the call and tries to solve the problem. If then do not know how to solve the caller problem, they put the call on hold and call the second level for help. The issue is that if I set the call-limit to 1 I lost this behavior. |
| 15:46.38 | *** join/#asterisk tsurko (n=tsurko@213.91.216.130) |
| 15:46.42 | [TK]D-Fender | pardove: Test if it intereferes with IVR's and being able to use # as a CHAR <- |
| 15:46.44 | stmaher | [TK]D-Fender, If you have any plans to come to Ireland IOU several drinks at this stage!! |
| 15:47.16 | tristanbob | http://www.microsoft.com/responsepoint/default.mspx |
| 15:47.27 | flujan | Does asterisk 1.4.12 and the changes in app_queue.c solve this issue? Or how can I avoid this issue? I was checking the app_queue.c code and see some places where they change a bit of code... At least in the try_calling function. If it is not support in asterisk now, I can use some help to develop a patch and address this "issue" of mine. |
| 15:47.32 | [TK]D-Fender | stmaher: My advise is ditch FreePBX and have a proper system built to meet your needs |
| 15:47.38 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 15:47.39 | pardove | [TK]D-Fender: ? |
| 15:48.32 | [TK]D-Fender | pardove: go dial into an IVR in your dialplan and see that your stripping of "#" doesn't interfere |
| 15:49.34 | coppice | tristanbob: MS must be having real problems pushing response point. did you see the weak hardware page? |
| 15:49.51 | tristanbob | coppice: no - where is it? |
| 15:50.09 | coppice | click on hardware at the left side |
| 15:50.45 | stmaher | [TK]D-Fender, I cant.. 90% of the testing has been done on this platform |
| 15:50.47 | drako | http://pastie.caboo.se/103344 <- somethign to do in here to avoid the error? |
| 15:50.49 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 15:50.55 | pardove | [TK]D-Fender: OK |
| 15:51.01 | drako | is very random when it works and stopworking the iax trunk |
| 15:51.11 | drako | and is only 250ms right now |
| 15:52.12 | [TK]D-Fender | stmaher: "Can't" is a word abused by those of little imagination and less sense of what should be done. |
| 15:52.39 | stmaher | Your right.. |
| 15:52.39 | [TK]D-Fender | stmaher: Either way, this is way outside the scope of this channel . |
| 15:52.41 | [TK]D-Fender | #freepbx |
| 15:53.02 | [TK]D-Fender | ~freepbx |
| 15:53.03 | jbot | somebody said freepbx was unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 15:53.04 | stmaher | [TK]D-Fender, ok.. ill give it a go.. |
| 15:53.17 | [TK]D-Fender | stmaher: Best of luck with whatever you try to do about this... |
| 15:53.33 | pardove | [TK]D-Fender: it works wonderful. even on IVRs. now zaptel fxs ports really act like a ATA port |
| 15:53.35 | Trionnis | ~trixbox |
| 15:53.35 | jbot | extra, extra, read all about it, trixbox is a full linux distro that includes FreePBX, and other 3rd party add-ons. It is all this extra stuff which makes trixbox seriously painful to support and hence you will find little help here for it. Try asking in #trixbox or on their forums & WIKI at http://www.trixbox.org |
| 15:53.37 | [TK]D-Fender | syzygyBSD: Just got to the end of that video.... |
| 15:53.58 | Trionnis | ah... I would have been a bit more... unfriendly... in my description :) |
| 15:54.06 | [TK]D-Fender | pardove: see if you can have that based on a zapata.conf var... that's patch-worth |
| 15:54.07 | syzygyBSD | lol, ya |
| 15:54.09 | [TK]D-Fender | pardove: see if you can have that based on a zapata.conf var... that's patch-worthy |
| 15:54.27 | [TK]D-Fender | syzygyBSD: You can deny my blade all you want... you'll be just as dead :) |
| 15:54.54 | *** join/#asterisk AlienPenguin (n=Miranda@213.188.207.153) |
| 15:54.59 | [TK]D-Fender | syzygyBSD: And I've only cut myself with my blade once. Trashed a shirt and the only other student there that day didn't even know it :) |
| 15:55.15 | coppice | you've been watching too many bad HK movies |
| 15:55.18 | Trionnis | haha |
| 15:55.22 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 15:55.43 | AlienPenguin | hi ppl, i am using asterisk-1.4.4 and i cannot hear the musicOnHold either using MusicOnHold or Queue() application |
| 15:55.56 | [TK]D-Fender | coppice: Nope... Akira Kurosawa films more like :) |
| 15:55.56 | AlienPenguin | calls are managed ok |
| 15:56.09 | AlienPenguin | i tried switching from wav and mp3 |
| 15:56.17 | syzygyBSD | AlienPenguin: ever had moh working? any errors in your log? |
| 15:56.25 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 15:56.26 | ai-a | AlienPenguin: dont use mp3. show cli log / errors |
| 15:56.31 | *** join/#asterisk xezz (n=asdasd@athedsl-218955.home.otenet.gr) |
| 15:56.49 | *** join/#asterisk bkruse_home (n=bkruse@69.73.127.92) |
| 15:56.56 | AlienPenguin | syzygyBSD: never tried moh before |
| 15:57.02 | AlienPenguin | ai-a no errors on cli |
| 15:57.06 | stmaher | [TK]D-Fender, Thank you for your support.. even tho I know it was a nightmare :-) |
| 15:57.15 | AlienPenguin | it sys it is playing and the queue members are correctly dialed |
| 15:57.23 | ai-a | AlienPenguin: can you pastebin website your cli output of the call. |
| 15:57.31 | coppice | [TK]D-Fender: those are similar - remade even worse by Hollywood :-) |
| 15:57.31 | ai-a | dont explain,, show. |
| 15:57.36 | AlienPenguin | ai-a sure |
| 15:57.43 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 15:57.45 | syzygyBSD | also pastebin your moh.conf |
| 15:58.03 | [TK]D-Fender | coppice: I'm talking 1950's originals here.... |
| 15:58.15 | [TK]D-Fender | stmaher: I've had worse.... |
| 15:58.33 | stmaher | Yeah me too.. kinda goes by the name of my ex girlfriend :-) |
| 16:01.20 | xezz | hello, i have a siemens hipath 3700 call center, can i connect it to asterisk via isdn card ? |
| 16:02.34 | AlienPenguin | http://pastebin.com/m2f0feaeb |
| 16:02.44 | pardove | setvar just works on sip.conf? |
| 16:02.59 | AlienPenguin | pasted: [moh|queues|extensions].conf too |
| 16:03.14 | coppice | [Tk[D-Fender: 黒澤明旳影好, or don't you deal with the true original? :-) |
| 16:04.30 | AlienPenguin | btw the "Stopped music on hold" message appeared when i put down the phone after some seconds. |
| 16:07.17 | flujan | guys, which hardphone that supports g729 do you recommend? |
| 16:07.28 | ai-a | AlienPenguin: have files in /var/lib/asterisk/moh/techQ ? and are they readable by asterisk ? |
| 16:07.54 | AlienPenguin | ai-a yes indeed |
| 16:08.00 | coppice | since most hard phones support G.729, you aren't really limited very much |
| 16:08.18 | ai-a | AlienPenguin: what file format are the wav files ? tried gsm files instead ? |
| 16:08.29 | AlienPenguin | ai-a: however i have the very same behaviour if i use the default moh files |
| 16:08.34 | *** join/#asterisk elixer (i=elixer@65.207.74.18) |
| 16:08.35 | AlienPenguin | ai-a: nottried the gms |
| 16:08.41 | AlienPenguin | gsm |
| 16:09.00 | AlienPenguin | i'll try right now |
| 16:09.04 | flujan | coppice: does polycom ip 330 ad 320 support it? |
| 16:09.15 | ai-a | use sox to convert them,. |
| 16:09.26 | AlienPenguin | i already do that :) |
| 16:09.28 | AlienPenguin | however |
| 16:09.46 | AlienPenguin | if i use Playback(mymohfile) it works |
| 16:09.53 | coppice | yep. very few hard phones don't support it. the free soft phones don't support it, because it costs. most other things do |
| 16:10.01 | tristanbob | looks like the main benefit is the IVR |
| 16:10.01 | tristanbob | "Reach anyone in the company directory or your Microsoft Outlook address book by simply saying their name." |
| 16:10.04 | ai-a | i always say use the same alaw/ulaw your isdn calls are using. |
| 16:10.30 | flujan | coppice: thanks for the help. :D |
| 16:11.01 | *** join/#asterisk mltlnx (n=mltlnx@cpe-74-73-178-39.nyc.res.rr.com) |
| 16:13.00 | AlienPenguin | ai-anope, gsm files are just the same |
| 16:13.23 | AlienPenguin | i dont see any rtp packets so qasterisk is not streaming anything on the network |
| 16:14.09 | *** join/#asterisk Kandinsky (n=Kandinsk@perla2.tm.ew.ro) |
| 16:15.26 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 16:15.40 | Kandinsky | hi. anybody using BRI ISDN (with HFC-S chipset on a PCI cad) on Asterisk? |
| 16:15.42 | Katty | I have a horrible, horrible TERRIBLE problem. |
| 16:15.51 | Katty | it's bugging me so much, i'm about about to splode For Real(tm)_ |
| 16:16.00 | Katty | i need help :< |
| 16:16.01 | Katty | badly! |
| 16:16.40 | Katty | i decided i wanted to make turkey breast for dinner. and i think baby carrots..but i don't know if baby carrots really go with turkey breast.. or what a nice healthy way to make turkey breast is...or... what my carb should be. |
| 16:16.43 | coppice | if you want to be helped badly, you've come to the right place |
| 16:16.44 | Katty | i'm about to insane. |
| 16:17.29 | Katty | there is my problem. |
| 16:17.38 | Katty | i have nothing to pastebin, sorry. |
| 16:17.56 | ai-a | AlienPenguin: no idea ;) |
| 16:18.03 | lirakis | wtf is huggles? .. everything gets 'cutesey' when Katty enters ;P |
| 16:18.23 | coppice | she did say breast |
| 16:18.24 | Katty | _ShrikE: everything? what carb? |
| 16:18.34 | *** join/#asterisk Defraz (n=t0tal@208-44-169-243.dia.static.qwest.net) |
| 16:18.37 | Katty | _ShrikE: new potatoes? |
| 16:18.43 | Katty | coppice: yes, yes i did. BREAST. |
| 16:18.51 | _ShrikE | exactly! |
| 16:18.55 | Katty | coppice: boneless, skinless BREAST |
| 16:18.57 | ai-a | Whoooo... whos breasts ? |
| 16:18.58 | coppice | rice, turkey and carrots - make turkey congee |
| 16:19.06 | Katty | ooh rice. |
| 16:19.08 | ai-a | skinless breasts sound sick. |
| 16:19.23 | Katty | coppice: tell me more about this Turkey Congee |
| 16:19.33 | outtolunc | reminds me bodyworks II is in town <G> |
| 16:19.35 | coppice | ai-a: you must be playing too roughly |
| 16:20.01 | Katty | lirakis: a huggle is like a hug. except usually involve much picking up and twirling about. |
| 16:20.25 | Katty | lirakis: and i can't help the 'cutesey' - it just happens. |
| 16:20.33 | coppice | sounds too energetic. i'll go for a plain snuggle |
| 16:20.37 | Katty | lirakis: it's a common female trait. |
| 16:21.34 | coppice | congee: a Hindi word adopted into English. |
| 16:21.38 | Kandinsky | anyone: how to use a PCI HFC-S BRI card on Asterisk? |
| 16:22.22 | *** join/#asterisk fuzzbawl (n=akennedy@blackhole.cyberlinkint.com) |
| 16:22.50 | *** join/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 16:23.13 | fuzzbawl | hey all, got a question about checking Zap channels |
| 16:23.54 | fuzzbawl | say my telco is acting retarded and I need to make a test call on each outbound Zap channel, individually, to test for issues. how do I do that? =] |
| 16:24.03 | twisted | omg katty |
| 16:24.20 | twisted | why you gotta come in here and talk about food? |
| 16:24.32 | Trionnis | mmm |
| 16:24.34 | Trionnis | foooood |
| 16:24.46 | tzafrir | Kandinsky, zapbri/zaphfc from bristuff or misdn |
| 16:24.56 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 16:25.28 | Katty | twisted: sorry, deary. |
| 16:25.30 | Kandinsky | tzafrir..installed asterisk bristuff on a kubuntu |
| 16:25.31 | twisted | heh. |
| 16:25.39 | Kandinsky | but how do i see the isdn card? |
| 16:25.49 | Katty | rice, baby carrots, and turkey breast. |
| 16:25.53 | WilliamK | in an ofc or in a cubicle? |
| 16:26.06 | Katty | now what do i do to the turkey breast? |
| 16:26.09 | twisted | we dont' have offices here |
| 16:26.12 | Katty | just saute in tons and TONS of garlic |
| 16:26.14 | twisted | not even the bosses have offices |
| 16:26.18 | Katty | really? |
| 16:26.19 | WilliamK | wow |
| 16:26.21 | Katty | i have an office, with a door. |
| 16:26.23 | Katty | which locks |
| 16:26.27 | tzafrir | Kandinsky, you have sample configs in the zaphfc directory. or use genzaptelconf |
| 16:26.29 | twisted | good for you :) |
| 16:26.37 | Katty | twisted: you can come hang out in my office if ya want ;) |
| 16:26.47 | twisted | can we lock the door? :P |
| 16:26.51 | Katty | no :P |
| 16:26.54 | fuzzbawl | ha |
| 16:26.54 | twisted | awwwww |
| 16:27.07 | Kandinsky | zaphfc directory..where? |
| 16:27.11 | tzafrir | Kandinsky, see http://updates.xorcom.com/astribank/bristuff/INSTALL.html |
| 16:27.30 | tzafrir | Kandinsky, you asterisk and astrisk-bristuff deb ? |
| 16:27.51 | Kandinsky | yes |
| 16:28.09 | Kandinsky | and i don't seem to find any zaphfc stuff |
| 16:28.33 | tzafrir | genzaptelconf should generate you a working config . Though the sample config files I mentioned should be under /usr/share/doc/zaptel/examples |
| 16:28.46 | [TK]D-Fender | coppice: isn't congee a kind of chinese porridge? |
| 16:29.26 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 16:29.32 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 16:29.41 | *** join/#asterisk thansen|laptop (n=thansen@19.243.sfcn.org) |
| 16:30.09 | coppice | yes, but the word is actually Hindi. We had to teach a chef in Rajastan how to make it once :-\ |
| 16:33.36 | [TK]D-Fender | coppice: Will commit that little nugget to memory... |
| 16:35.17 | coppice | in cantonese, congee is juk - 粥 |
| 16:36.25 | [TK]D-Fender | coppice: My friend speaks mandarin and used the word as-is. Was it absorbed as such directly, or simply misappropriated? |
| 16:37.30 | coppice | I think its something like juk in mandarin too. I can't remember, congee is always used as the english word |
| 16:39.10 | [TK]D-Fender | coppice: If I ever see the guy again, I'll ask him.... |
| 16:39.36 | [TK]D-Fender | (friend was a bit of a stretch for our frequency of contact) |
| 16:39.44 | *** join/#asterisk jksM (n=jks@87.57.88.86) |
| 16:39.55 | stmaher | [TK]D-Fender, hhehehe.. now on vanilla asterisk :-) |
| 16:40.41 | coppice | [TK]D-Fender: well, I'm sure you can easily remember how the chinese is written |
| 16:40.44 | [TK]D-Fender | stmaher: Congratulations. |
| 16:40.58 | [TK]D-Fender | coppice: LOL... no dice :) |
| 16:40.58 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 16:41.00 | stmaher | [TK]D-Fender, thank you.. now do i get my lollypop? |
| 16:41.07 | *** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 16:41.08 | jksM | can anyone recommend books on setting up VoIP on a service provider scale? (or similar, i.e. more in-depth information on for example multi-server setups, than for example I can find in the O'reilly asterisk book) |
| 16:41.31 | coppice | [TK]D-Fender: 你冇用 |
| 16:41.39 | Trionnis | jksM: you should have went to Astricon |
| 16:41.50 | jksM | Trionnis, oh :-| |
| 16:42.02 | [TK]D-Fender | coppice: You don't say! |
| 16:42.03 | Trionnis | there were 3 presentations on it |
| 16:42.05 | jksM | Trionnis, hmm, it wasn't taped and put online or something? - I'll go check it out |
| 16:42.13 | Trionnis | they're supposed to be uploaded to astricon.net |
| 16:42.23 | Trionnis | no idea when anyone is going to get around to doing that though |
| 16:43.22 | jksM | Astricon is just a bit far away for me... they have to do it in Scandinavia next year ;-) |
| 16:43.24 | coppice | [TK]D-Fender: insults don't look so bad, when you can't read them :-) |
| 16:43.26 | coppice | 你 = you |
| 16:43.27 | coppice | 冇 = not have |
| 16:43.29 | coppice | 用 = use |
| 16:43.53 | Trionnis | I could handle that |
| 16:44.06 | *** part/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 16:44.34 | jksM | Trionnis, can you recommend any books? - I've found a tonnes on amazon... but I'm afraid that they're not very good |
| 16:44.56 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 16:45.07 | Trionnis | nothing very comprehensive on large scale * |
| 16:45.19 | Trionnis | there seems to be a few different camps for how to go about it |
| 16:45.29 | jksM | anything on a medium scale? :-) (I'm looking to setup something for about 500 users) |
| 16:45.30 | [TK]D-Fender | coppice: http://jitcrunch.cafepress.com/jitcrunch.aspx?bG9hZD1ibGFuayxibGFuazoxMDZfRi5qcGd8bG9hZD1MMCxodHRwOi8vaW1hZ2VzLmNhZmVwcmVzcy5jb20vaW1hZ2UvODYxNzA2NV80MDB4NDAwLmpwZ3x8c2NhbGU9TDAsMTQ0LDE0NCxXaGl0ZXxjb21wb3NlPWJsYW5rLEwwLEFkZCwxNzMsMTE4fGNwPXJlc3VsdCxibGFua3xzY2FsZT1yZXN1bHQsMCw0ODAsV2hpdGV8Y29tcHJlc3Npb249OTV8 |
| 16:45.41 | Trionnis | blah, spam |
| 16:46.07 | Trionnis | nothing I've seen, sorry :( |
| 16:46.18 | coppice | [TK]D-Fender: I hope someone tattoos you with "gullible white boy" |
| 16:46.19 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 16:46.27 | [TK]D-Fender | coppice: Loved that one too :) |
| 16:46.48 | jksM | Trionnis, okay, I wonder that all the knowledable people did... perhaps it's matter of trial and error or something ;-) |
| 16:46.53 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 16:46.59 | jksM | knowledgeable* |
| 16:47.00 | Trionnis | pretty much |
| 16:47.02 | coppice | ah. but you couldn't read it, so you have to assume it didn't say something nastier :-) |
| 16:47.31 | Trionnis | the guy from callfire.com had a pretty good presentation about it |
| 16:47.34 | [TK]D-Fender | coppice: Just because I can't read it doesn't mean I'm incapable of having it translated. I am resourceful you know... |
| 16:47.39 | Trionnis | lot of interesting concepts |
| 16:47.42 | stmaher | [TK]D-Fender, Ok.. what needs to be done to allow call transfer? http://pastebin.com/d760c2071 <- my sip.conf and extensions.conf |
| 16:48.12 | Trionnis | I have his email address around here somewhere... you could email him and ask for a copy of the presentation I'd guess |
| 16:48.13 | [TK]D-Fender | stmaher: You are so significantly far from having a clue I can't really help you right now... |
| 16:48.14 | coppice | [TK]D-Fender: someone else was resourceful like that. they asked me to translate. that's where I saw it :-) |
| 16:48.17 | Trionnis | hold on |
| 16:48.38 | [TK]D-Fender | coppice: Yeah, POSYHUMOUSLY doesn't help them very much :p |
| 16:48.39 | stmaher | [TK]D-Fender, ok thanks |
| 16:48.48 | jksM | Trionnis, I'm perhaps overdoing this... I have setup so far a front-end to handle the incoming trunks, a server for handling IVR and the actual calls, a server for handling CDR and blocked calls and a server for voicemail + webinterface |
| 16:48.53 | [TK]D-Fender | stmaher: No minced words there.... |
| 16:49.05 | [TK]D-Fender | stmaher: you can't jsut take 5 lines and call it a config.... |
| 16:49.28 | [TK]D-Fender | stmaher: And clearly you either have a lot more in there you're not showing or its now completely broken. |
| 16:49.29 | jksM | Trionnis, but where I'm clueless is in stuff like combining SER with Asterisk, etc. |
| 16:49.54 | stmaher | I have a debian box i just apt-got it.. |
| 16:50.19 | Trionnis | SER is it's own beast |
| 16:50.22 | stmaher | [TK]D-Fender, its a default installation .. I can paste the complete extensions.conf and sip.conf file |
| 16:50.26 | *** join/#asterisk ZaVoid (n=zavoid@host-24-225-239-34.patmedia.net) |
| 16:50.41 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 16:50.57 | [TK]D-Fender | stmaher: You should seriously have a consultant get you set up to start and learn from there... |
| 16:51.20 | Trionnis | but.. but.. but.. he's using freepbx.. isn't that enough? ;) |
| 16:51.56 | ZaVoid | hey guys |
| 16:52.19 | ZaVoid | is there a way to show registered sip devices to my asterisk box? sip show registry: Show SIP registration status (when Asterisk registers as a client to a SIP Proxy) <-- is the reverse of what i'm looking for |
| 16:53.14 | ZaVoid | and sip show peers isn't it either |
| 16:53.14 | tzafrir | stmaher, apt-got what exatly? Asterisk? FreePBX? |
| 16:53.29 | *** part/#asterisk Dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 16:53.33 | stmaher | tzafrir, Asterisk |
| 16:53.46 | tzafrir | the Etch package? |
| 16:54.01 | stmaher | yes |
| 16:54.02 | Kandinsky | anybody who uses ISDN with HFC-S chipset PCI cards on Asterisk? |
| 16:54.46 | tzafrir | Kandinsky, modprobe zaphfc |
| 16:55.05 | Kandinsky | not found |
| 16:55.33 | Kandinsky | i ran genzaptelconf -sdMv |
| 16:55.39 | Kandinsky | and i got 1 error |
| 16:56.01 | Kandinsky | unable to open device /dev/zap/ctl |
| 16:56.04 | tzafrir | have you run m-a a-i zaptel |
| 16:56.31 | Kandinsky | i am running it now |
| 16:56.47 | Kandinsky | it failed |
| 16:56.56 | Kandinsky | and i got a menu |
| 16:56.56 | tzafrir | uname -r |
| 16:57.06 | Kandinsky | view continue or stop |
| 16:57.15 | *** part/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) |
| 16:57.21 | *** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr) |
| 16:57.24 | tzafrir | "view" |
| 16:57.25 | Kandinsky | what should i choose? |
| 16:57.27 | Kandinsky | ok |
| 16:57.31 | tzafrir | hmm... stop |
| 16:57.42 | tzafrir | I prefer the text interface |
| 16:57.58 | Kandinsky | ? |
| 16:57.59 | tzafrir | m-a -t -i a-i zaptel |
| 16:58.02 | Kandinsky | ok |
| 16:58.03 | Kandinsky | wait |
| 16:58.42 | Kandinsky | failed |
| 16:59.06 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 16:59.11 | *** join/#asterisk blinkbot2k (n=me@67.106.175.130.ptr.us.xo.net) |
| 17:00.00 | tzafrir | What error? |
| 17:00.09 | tzafrir | Can you pasebin the last lines? |
| 17:00.17 | Kandinsky | wait a minnute |
| 17:00.34 | styelz | Kandinsky: apt-get install module-assistant |
| 17:03.41 | tzafrir | zaptel-source depends on m-a |
| 17:03.54 | *** join/#asterisk Kandinsky (n=cristi@perla2.tm.ew.ro) |
| 17:04.00 | tzafrir | zaptel-source depends on m-a |
| 17:04.01 | Kandinsky | ok back |
| 17:04.09 | tzafrir | (unless Ubuntu managed to mess something) |
| 17:04.19 | Kandinsky | so |
| 17:04.26 | Kandinsky | from the start |
| 17:04.33 | Kandinsky | i have a kubuntu 7.04 |
| 17:04.42 | Kandinsky | installed asterisk from the repo |
| 17:04.52 | Kandinsky | asterisk bristuff |
| 17:05.28 | Kandinsky | asterisk-app-dtmftotext |
| 17:05.34 | Kandinsky | -app-fax |
| 17:05.42 | Kandinsky | -app-misdn-v110 |
| 17:05.48 | Kandinsky | -chan-capi |
| 17:05.57 | Kandinsky | -chan-misdn |
| 17:06.06 | Kandinsky | a |
| 17:06.07 | Kandinsky | sorry |
| 17:06.10 | Kandinsky | :P |
| 17:06.25 | *** join/#asterisk tripps (n=ss@66.60.235.100) |
| 17:06.26 | Kandinsky | chan-capi and chan-capi-misdn not installed |
| 17:06.39 | Kandinsky | do i need them? |
| 17:06.46 | *** part/#asterisk tripps (n=ss@66.60.235.100) |
| 17:06.51 | tzafrir | no |
| 17:06.56 | Kandinsky | ok |
| 17:06.59 | Kandinsky | -config |
| 17:07.01 | *** join/#asterisk tripps (n=ss@66.60.235.100) |
| 17:07.01 | Kandinsky | -dev |
| 17:07.03 | *** part/#asterisk tripps (n=ss@66.60.235.100) |
| 17:07.05 | Kandinsky | -doc |
| 17:07.09 | Kandinsky | -sounds-extra |
| 17:07.12 | *** join/#asterisk tripps (n=ss@66.60.235.100) |
| 17:07.14 | Kandinsky | -sounds-main |
| 17:07.16 | tzafrir | -dev: no. -config: asterisk depends on it |
| 17:07.18 | Kandinsky | -web-vmail |
| 17:07.37 | tzafrir | -doc - can help . -sounds-extra - nice . -web-vmail - you don't need |
| 17:07.52 | Kandinsky | then added zaptel |
| 17:08.06 | Kandinsky | and zaptel source |
| 17:08.07 | styelz | Kad: i find it easier to just download asterisk/addons and zaptel from the digium ftp and compile form source |
| 17:08.08 | tzafrir | but you got a specific error |
| 17:08.12 | tzafrir | can I see it? |
| 17:08.21 | Kandinsky | ok |
| 17:08.34 | Kandinsky | /usr/src/modules/zaptel/vzaphfc/vzaphfc_main.c:1684: warning: passing argument 2 of ‘request_irq’ from incompatible pointer type |
| 17:08.38 | Kandinsky | make[5]: *** [/usr/src/modules/zaptel/vzaphfc/vzaphfc_main.o] Error 1 |
| 17:08.42 | Kandinsky | make[4]: *** [/usr/src/modules/zaptel/vzaphfc] Error 2 |
| 17:08.46 | Kandinsky | make[3]: *** [_module_/usr/src/modules/zaptel] Error 2 |
| 17:08.50 | Kandinsky | make[3]: Leaving directory `/usr/src/linux-headers-2.6.20-16-generic' |
| 17:08.50 | Kandinsky | make[2]: *** [linux26] Error 2 |
| 17:08.50 | Kandinsky | make[2]: Leaving directory `/usr/src/modules/zaptel' |
| 17:08.50 | Kandinsky | make[1]: *** [binary-modules] Error 2 |
| 17:08.50 | Kandinsky | make[1]: Leaving directory `/usr/src/modules/zaptel' |
| 17:08.50 | Kandinsky | make: *** [kdist_build] Error 2 |
| 17:09.08 | *** join/#asterisk xidarian (n=santor@pool-71-243-251-222.tampfl.fios.verizon.net) |
| 17:09.35 | Kandinsky | do u see it? |
| 17:09.45 | Kandinsky | or was it to much text |
| 17:11.02 | Kandinsky | tzafrir: ? |
| 17:11.18 | Kandinsky | stylez do u use ISDN ? |
| 17:11.30 | Kandinsky | styelz sorry |
| 17:11.37 | styelz | no |
| 17:11.40 | Kandinsky | k |
| 17:11.55 | styelz | just fxo/fxs |
| 17:12.23 | styelz | pstn |
| 17:12.40 | Kandinsky | tzafrir: u still there? |
| 17:15.33 | styelz | Kandinsky: http://downloads.digium.com/pub/zaptel/zaptel-1.4-current.tar.gz |
| 17:15.41 | styelz | see if that compiles |
| 17:15.48 | Kandinsky | but i have asterisk 1.2.16 |
| 17:15.51 | styelz | or 1.2 .. |
| 17:15.56 | styelz | Kandinsky: http://downloads.digium.com/pub/zaptel/zaptel-1.2-current.tar.gz |
| 17:16.54 | styelz | just extract and cd into the folder and do ./configure && make menuselect && make install |
| 17:17.20 | *** join/#asterisk luisjose (n=ljd@nelug/coreteam/luisjose) |
| 17:17.37 | Kandinsky | yeah..but i need the bristuff version |
| 17:17.43 | Kandinsky | i think |
| 17:17.46 | *** join/#asterisk ljd (n=ljd@nelug/coreteam/luisjose) |
| 17:17.58 | Kandinsky | everybody said it would be the best for my isdn |
| 17:18.15 | styelz | thats asterisk dependant though.. i thought |
| 17:18.36 | Kandinsky | yes..it a patch for astersik |
| 17:18.40 | Kandinsky | it's >P |
| 17:19.23 | styelz | what hardware do you have |
| 17:19.40 | Kandinsky | a server and 2 isdn pci cards |
| 17:19.43 | Kandinsky | bri |
| 17:19.49 | Kandinsky | with hfc-s chipset |
| 17:19.58 | Kandinsky | and 2 nt from the isdn provider |
| 17:20.11 | Kandinsky | and voip phones |
| 17:20.56 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 17:21.01 | Kandinsky | the voip seems to work...got the 2 phones up and running |
| 17:21.12 | styelz | so you have custom driver for zaptel ? |
| 17:21.12 | Kandinsky | talked from one to the other |
| 17:21.25 | Kandinsky | what do you mean by that |
| 17:21.35 | Kandinsky | i got no cds with the pci cards |
| 17:21.41 | Kandinsky | if that's what u mean |
| 17:23.04 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 17:23.10 | *** join/#asterisk Chris-NB (n=chris@home.fuerstaller.com) |
| 17:23.54 | *** join/#asterisk rpm (n=russell@75.155.167.90) |
| 17:25.16 | styelz | hmm |
| 17:25.32 | styelz | ok i see in the source for 1.4 it says. |
| 17:25.36 | styelz | xpp/ChangeLog: * genzaptelconf will detect vzaphfc. |
| 17:26.20 | Kandinsky | the problem is...that I don't think the zaptel package contains bristuff drivers |
| 17:26.32 | Kandinsky | not sure 100% |
| 17:27.23 | Kandinsky | http://www.linuxdays.lu/downloads/linuxdays-2006/plonelocalfolderng.2006-03-07.9017369414/plfng_view |
| 17:27.24 | styelz | asterisk-bristuff - Open Source Private Branch Exchange (PBX) - BRIstuff-enabled version |
| 17:27.46 | Kandinsky | <PROTECTED> |
| 17:28.00 | Kandinsky | page 21 |
| 17:28.01 | *** join/#asterisk krondorl (n=chatzill@acid.auricnet.ca) |
| 17:28.41 | Kandinsky | that's what i should do...but it isn't very detailed |
| 17:30.31 | *** join/#asterisk krondorl (n=chatzill@acid.auricnet.ca) |
| 17:30.37 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:30.37 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 17:31.17 | Kandinsky | but there they did use capi |
| 17:31.21 | *** join/#asterisk thansen|laptop (n=thansen@19.243.sfcn.org) |
| 17:31.25 | krondorl | Is there a channel for the (FOP) Front Operator Panel?? |
| 17:31.34 | deeperror | does asterisk play well with dual/quad core systems? |
| 17:31.48 | Kandinsky | i have it on a dual core and it works |
| 17:31.57 | krondorl | Ours is on a dual. |
| 17:32.07 | deeperror | does it use them both? |
| 17:32.48 | deeperror | maybe a better question. is anyone familiar or know of anything online that would compare systems, results, load etc? |
| 17:33.04 | krondorl | couldn't tell ya.. |
| 17:33.17 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 17:33.26 | De_Mon | why sis there a genzaptelconf, zapconf AND an ztcfg? |
| 17:34.00 | De_Mon | s~an~a~ |
| 17:35.32 | blitzrage | ztcfg loads the configuration into memory I believe |
| 17:35.55 | blitzrage | like ztcfg -vv will tell you if things are good or not |
| 17:36.40 | styelz | there are "man" pages for each |
| 17:38.16 | Kandinsky | anybody using isdn with PCI hfc-s chipset cards? |
| 17:39.29 | styelz | echo echo echo... |
| 17:39.33 | *** join/#asterisk TedNJ38 (n=HungLad@ool-4573adc7.dyn.optonline.net) |
| 17:39.35 | TedNJ38 | How can I adjust the time in which my box logs events? Everything in the log file seems to be a few hours off. |
| 17:39.57 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 17:40.32 | styelz | TedNJ38: set the correct time and timezone on your system |
| 17:41.05 | *** join/#asterisk dimgr (n=dimgr@athedsl-119981.home.otenet.gr) |
| 17:44.02 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 17:44.04 | tru_`z24 | exten => s,n,Answer ... .what does the "s" stand for in that dialplan line? |
| 17:44.14 | anonymouz666 | start? |
| 17:44.21 | [TK]D-Fender | ~stdextens |
| 17:44.22 | jbot | from memory, stdextens is "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), a call coming in 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. ... |
| 17:44.36 | [TK]D-Fender | tru_`z24: time to read... THE BOOK |
| 17:44.39 | [TK]D-Fender | ~book |
| 17:44.39 | jbot | Asterisk: The Future of Telephony 2nd Edition --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 17:44.43 | tru_`z24 | :-) |
| 17:44.53 | tru_`z24 | thank you |
| 17:45.14 | [TK]D-Fender | ~botsnack |
| 17:45.14 | jbot | :), [TK]D-Fender |
| 17:45.21 | Kandinsky | anybody using isdn with PCI hfc-s chipset cards? |
| 17:47.16 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 17:47.43 | *** join/#asterisk jingles (n=dfbarth@39.183.dowl.anc.borealisbroadband.net) |
| 17:48.10 | Spida | Kandinsky: barely. |
| 17:48.22 | Kandinsky | :( |
| 17:49.08 | Spida | used barely. what I wanted DID work (I had a jabber bot that notified me about the caller-id when the phone rang) |
| 17:49.28 | Spida | these cards should work, and can be used in NT-mode, afaik |
| 17:49.39 | Kandinsky | i'm interested on how to get the asterisk to see and use the isdn cards |
| 17:49.59 | Kandinsky | asterisk-bristuffed |
| 17:52.13 | Katty | hi! |
| 17:52.18 | Katty | i bought turkey breast. |
| 17:52.20 | Katty | and carrots. |
| 17:52.21 | Katty | and brown rice. |
| 17:53.13 | Kandinsky | hurey :P |
| 17:53.58 | _ShrikE | Katty: you going with the congee? |
| 17:54.09 | *** join/#asterisk Mrchicken (n=dorphals@200.71.58.39) |
| 17:54.37 | Katty | _ShrikE: no,i don't think so. |
| 17:54.48 | Katty | _ShrikE: i think i'll just make something light and healthy (= |
| 17:54.54 | Mrchicken | Hello, I need to get speech synthesis into my *, which package would you recommend me? |
| 17:55.16 | Mrchicken | Flite or Festival? |
| 17:55.17 | _ShrikE | Katy: Well, I live in New Orleans so I dont know anything about that. |
| 17:55.21 | _ShrikE | :) |
| 17:55.38 | Qwell | _ShrikE: hmm... I'm gonna be down that way this weekend, or early in the week. Anywhere I *must* go? |
| 17:55.55 | Qwell | or anything I must see? anything like that :D |
| 17:55.57 | Katty | _ShrikE: that's okay (= |
| 17:56.37 | _ShrikE | Redfish Grill anytime.. Palace Cafe for lunch (get the bananas foster).. If you wanna drop a few more bucks than usual you cant beat commanders palace. |
| 17:57.12 | _ShrikE | Take a tour of the 9th ward and see the devastation. |
| 17:58.02 | _ShrikE | and the D-Day museum is very impressive |
| 17:58.09 | Qwell | _ShrikE: thanks |
| 17:58.27 | Kandinsky | anybody using isdn with PCI hfc-s chipset cards? |
| 18:01.56 | *** join/#asterisk akx^ (n=fddsfs@adsl-69-209-162-188.dsl.sfldmi.ameritech.net) |
| 18:02.57 | akx^ | i'm having some major issues with voicemail could anyone please try to help me out i would really appreciate it |
| 18:05.05 | Corydon76-dig | ~ask |
| 18:05.06 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 18:05.51 | mvanbaak | ~question |
| 18:05.51 | jbot | extra, extra, read all about it, question is If you have a question and want people to give useful answers, make sure you have read this first: http://www.catb.org/~esr/faqs/smart-questions.html |
| 18:06.12 | wishes | heh man that gts quotes all the time :D |
| 18:06.18 | Mrchicken | Hello, I need to get speech synthesis into my *, which package would you recommend me? |
| 18:06.24 | wishes | wonder how many hits he gets on url |
| 18:06.36 | wishes | Mrchicken: Festival works easily enough - but i wouldnt try it on 1.2 |
| 18:06.55 | Corydon76-dig | akx^: that was directed to you. Don't ask if somebody can help, just ask the questions |
| 18:07.00 | wishes | nor would i try and older version of festival |
| 18:07.12 | Kandinsky | anybody using isdn with PCI hfc-s chipset cards? |
| 18:07.36 | coppice | you are |
| 18:07.58 | _x86_ | ha |
| 18:08.35 | Kandinsky | funny |
| 18:10.02 | akx^ | sorry my internet keeps going up and down |
| 18:10.12 | _x86_ | get better internets! |
| 18:10.33 | Katty | mmm, internets. |
| 18:10.37 | akx^ | i have to stay on the phone network that's why it sucks |
| 18:11.11 | Katty | :< |
| 18:11.43 | _x86_ | i got teh internetz version 9.0 |
| 18:12.10 | akx^ | my voicemail used to work perfect and in time the voicemail for extentions kept going out one by one right now only one of them wrks it's really weird they're all set up the same way |
| 18:13.12 | akx^ | i try to login i put in the password and once i put it in stays without doing anything for a couple seconds and the call gets hanged up |
| 18:15.09 | wishes | whats the console output ? |
| 18:15.58 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 18:16.04 | *** join/#asterisk Boones (i=Boones@port-212-202-42-40.dynamic.qsc.de) |
| 18:16.05 | J4k3 | I got INX |
| 18:16.07 | J4k3 | InterNet X |
| 18:16.35 | J4k3 | I PAY $129 PER UPGRADE AND I CAN'T DO ANYTHING WITH IT BUT ITS BETTER THAN YERZ |
| 18:16.56 | J4k3 | (oops, didn't mean to offend any mac weenies) |
| 18:17.56 | akx^ | Executing VoiceMailMain("SIP/704-a566", "704") in new stack |
| 18:17.56 | akx^ | <PROTECTED> |
| 18:18.23 | akx^ | and it stays like that the call doesn't even get hung up anymore |
| 18:18.31 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
| 18:18.31 | *** mode/#asterisk [+o angler] by ChanServ |
| 18:19.09 | *** join/#asterisk saftsack (n=saftsack@217.224.114.212) |
| 18:20.24 | wishes | set debug higher |
| 18:20.26 | wishes | verbose more |
| 18:22.17 | akx^ | my debug is 9999999 and that's all im getting |
| 18:23.23 | akx^ | tried from a different extension same thing |
| 18:24.21 | akx^ | this is how i have them set up |
| 18:24.22 | akx^ | 704 => 1111,****** *****,******@*****.com,,attach=yes|saycid=yes|envelope=yes|delete=no |
| 18:25.26 | [TK]D-Fender | akx^: Sounds like DTMF simply isn't getting in. What is your SIP device? |
| 18:25.43 | akx^ | Polycom 430s |
| 18:26.18 | wishes | mm mines a little more simplistic 815 => 1234,Liz Quilty,liz@zeald.com |
| 18:26.36 | [TK]D-Fender | akx^: Make sure in sip.conf you use "dtmfmode=rfc2833" for each phone's entry |
| 18:26.52 | *** join/#asterisk GreggB (n=GreggB@66.206.86.107) |
| 18:28.01 | *** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
| 18:28.44 | *** join/#asterisk russellb (i=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 18:28.44 | *** mode/#asterisk [+o russellb] by ChanServ |
| 18:29.53 | akx^ | just checked they're all good |
| 18:30.45 | [TK]D-Fender | akx^: Where is the phone relative to your server? |
| 18:31.00 | akx^ | [704] |
| 18:31.00 | akx^ | username=704 |
| 18:31.00 | akx^ | type=friend |
| 18:31.00 | akx^ | secret=***** |
| 18:31.00 | akx^ | record_out=Adhoc |
| 18:31.01 | akx^ | record_in=Adhoc |
| 18:31.03 | akx^ | qualify=no |
| 18:31.05 | akx^ | port=5060 |
| 18:31.07 | akx^ | nat=never |
| 18:31.09 | akx^ | mailbox=704@device |
| 18:31.10 | Katty | >.< |
| 18:31.11 | akx^ | host=dynamic |
| 18:31.13 | akx^ | dtmfmode=rfc2833 |
| 18:31.15 | akx^ | context=from-internal |
| 18:31.17 | akx^ | canreinvite=no |
| 18:31.19 | akx^ | callerid=***** ***** <704> |
| 18:31.21 | akx^ | here's how i have the extensions set up |
| 18:31.23 | akx^ | what do you mean by that? |
| 18:31.29 | Katty | akx^: FYI, pastebin.ca is fantastic (= |
| 18:31.29 | [TK]D-Fender | PASTEBIN |
| 18:31.31 | [TK]D-Fender | ~pb |
| 18:31.32 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 18:31.41 | Katty | akx^: it keeps my head from sloding (= |
| 18:32.04 | [TK]D-Fender | akx^: Where is the phone relative to your server networking-wise? |
| 18:32.17 | akx^ | on the same network |
| 18:32.20 | FuriousGeorge | im noticing after a few weeks, asterisk starts getting hung channels if i dont restart it |
| 18:32.24 | [TK]D-Fender | akx^: And as a warning FreePBX is *NOT* supported here. |
| 18:32.46 | akx^ | not using freepbx |
| 18:32.51 | FuriousGeorge | i got a call today, all their trunk lines were busy (pots / sangoma a201), but no one was on the phone |
| 18:32.52 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 18:33.04 | [TK]D-Fender | akx^: if you're typing in the box # and hitting "#" and * isn't seeing it, it should likely be a networking issue somewhere. * setup looks fine. |
| 18:33.17 | FuriousGeorge | i logged in and sure enough all the zap channels were in use, i had to soft hangup all of them |
| 18:33.24 | [TK]D-Fender | akx^: I recodgnize their configs anywhere..... been seeing it for years now... |
| 18:33.43 | *** join/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 18:34.10 | FuriousGeorge | i cant seem to find anything on the bug reports about it. i cant leave the system in that state when it happens to seek help for debugging, and this doesnt result in a crash or deadlock so there is no core dump |
| 18:34.45 | akx^ | i didn't set up this box the person that worked here before me did it's running version 1.2.9 |
| 18:35.50 | [TK]D-Fender | akx^: Do "set debug 10 |
| 18:36.00 | [TK]D-Fender | akx^: and check the call as it goes through. |
| 18:38.20 | akx^ | http://pastebin.com/d655944e0 |
| 18:38.24 | akx^ | same exact thing |
| 18:39.50 | [TK]D-Fender | akx^: enable SIP DEBUG and pastebin the entire call from beginning to end. |
| 18:45.47 | *** join/#asterisk tim0123 (n=cash247@ppp-70-128-139-44.dsl.rcsntx.swbell.net) |
| 18:46.14 | akx^ | if it would be a dtmf problem how could i try to fix that? |
| 18:46.49 | tim0123 | I have a question about realtime static |
| 18:47.30 | *** join/#asterisk monux (n=fsantiag@216.106.170.59) |
| 18:47.41 | monux | hi!:d |
| 18:47.51 | monux | <PROTECTED> |
| 18:48.00 | tim0123 | Anyone know about realtime static |
| 18:48.05 | monux | <PROTECTED> |
| 18:48.15 | [TK]D-Fender | akx^: I jsut told you what I needed to see. Please provide the pastebin. |
| 18:48.17 | monux | involving yellow alarms |
| 18:49.10 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 18:50.19 | tim0123 | [TK]D-Fender I had another question about realtime static |
| 18:50.41 | [TK]D-Fender | tim0123: just ask, and don't limit yourself to asking only me. |
| 18:50.54 | [TK]D-Fender | tim0123: So, go on... |
| 18:51.03 | tim0123 | k |
| 18:52.38 | tim0123 | Can you use the mysql driver for realtime static ,I curious if the only difference between realtime and realtime static is the table layout |
| 18:53.27 | [TK]D-Fender | tim0123: ok, really couldn't say.. |
| 18:54.31 | akx^ | how do you make putty let you see more history |
| 18:56.00 | tim0123 | well im using realtime with mysql ,im just wondering if i change the table layout will it go back to static |
| 18:56.22 | monux | does any one know why yellow alarms happen |
| 18:56.52 | monux | they happen randomly and takes exactly 5 seconds to clear |
| 18:56.58 | monux | <PROTECTED> |
| 18:57.00 | monux | ? |
| 18:58.11 | [TK]D-Fender | akx^: in your connection setting, scroll-back <- |
| 18:58.16 | fuzzbawl | http://www.cisco.com/warp/public/116/T1_alarms.html |
| 18:58.19 | akx^ | http://pastebin.com/d1217881f |
| 18:58.21 | [TK]D-Fender | akx^: I typically set to about 2000 likst |
| 18:58.22 | [TK]D-Fender | lines |
| 18:58.58 | fuzzbawl | monux, the tests are cisco specific, but it at least describes the various alarms |
| 18:59.55 | *** join/#asterisk _pepo_ (n=Pepo@190.10.187.20) |
| 19:00.46 | monux | fuzzbal: ok but, here is my set up , i have an asterisk server with a digium card, and they are hooked up to the telco |
| 19:01.01 | monux | fuzzbawl: via fiber optics |
| 19:01.41 | [TK]D-Fender | akx^: Hrm, you hear the prompts, right? |
| 19:01.41 | fuzzbawl | you still have a T1 interface at some point, correct? |
| 19:01.45 | monux | <PROTECTED> |
| 19:02.13 | fuzzbawl | monux: I would assume your telco is the time source? |
| 19:02.22 | monux | fuzzbawl : yes |
| 19:02.22 | [TK]D-Fender | monux: pastebin your zaptel.conf & zapata.conf. |
| 19:02.24 | [TK]D-Fender | ~pb |
| 19:02.25 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 19:02.27 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 19:02.58 | akx^ | i hear enter the password after that i don't hear anything |
| 19:03.07 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 19:03.16 | monux | oks |
| 19:03.24 | monux | <PROTECTED> |
| 19:03.34 | [TK]D-Fender | akx^: :/ |
| 19:04.16 | akx^ | did u take a look at the sip output? |
| 19:04.55 | monux | copying them |
| 19:05.05 | monux | <PROTECTED> |
| 19:05.16 | [TK]D-Fender | akx^: Yeah, I see nothing out of the ordinary... |
| 19:06.29 | akx^ | its really weird because it was working before for some extensions and one by one kept getting mest up without any changes beeing made in the configuration of the box |
| 19:08.43 | fetcher | has anyone tried these cheap SIP phones from China? http://www.5111soft.com/5111softNEW/en/PH802.html |
| 19:09.02 | fetcher | boss wants to order some, and they sound OK on paper (even supporting IAX2). What kind of problems might we expect? |
| 19:09.29 | monux | http://pastebin.com/d477785ae that is my zapata.conf |
| 19:10.20 | *** join/#asterisk etfonhomey (n=chatzill@12.169.248.226) |
| 19:10.48 | monux | http://pastebin.com/d34f9194a that is my zaptel |
| 19:11.07 | [TK]D-Fender | fetcher: yup, look like cheap shit.... |
| 19:11.16 | monux | fuzzbawl, D-fender tell me whta you make of them |
| 19:11.57 | *** join/#asterisk HCEvan (i=edude@pool-72-77-226-91.tampfl.fios.verizon.net) |
| 19:13.13 | monux | fetcher: Grandstream you say? |
| 19:13.15 | [TK]D-Fender | monux: switchtype=national <- think that should be euroisdn <- |
| 19:13.43 | [TK]D-Fender | monux: national = NI1 |
| 19:13.57 | [TK]D-Fender | or was that NI2? Either way not for EU |
| 19:14.26 | monux | D-Fender : i live in latin america , Guatemala exactly , so that was the one my boss used at the telco |
| 19:14.40 | monux | <PROTECTED> |
| 19:15.22 | *** join/#asterisk trippss (n=ss@66.60.235.100) |
| 19:15.42 | [TK]D-Fender | monux: Hrm... not sure of what the norms are there.... One would think that your telco would use a complete set of matching standards from one region... |
| 19:17.13 | trippss | is there any detriment (performance, etc.) to having qualify set to yes on SIP peers? |
| 19:17.27 | monux | D-fender: telco's here don't tendo to give out this info... |
| 19:17.29 | [TK]D-Fender | tripps : generally no. |
| 19:17.42 | monux | D-Fender: also i use vici-dial |
| 19:18.14 | *** join/#asterisk klictel (n=klictel@189.31.64.100) |
| 19:19.42 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 19:20.16 | *** join/#asterisk karlh626 (n=kharris@IP-66.249.227.5.indigital.net) |
| 19:21.54 | karlh626 | While at astricon there was mention of a location where we could download the presentations that were given at the various talks. Is this available yet. |
| 19:23.27 | GreggB | Anyone using a cheap or pay-as-you-go PSTN to SIP/IAX provider they're happy with? Such as one you can port an existing number to and maintain services for just a couple $$/month plus ~$0.01 to $0.03 per-minute (US) rates. |
| 19:25.46 | [TK]D-Fender | GreggB: ... |
| 19:25.49 | [TK]D-Fender | ~itsp |
| 19:25.50 | jbot | An ITSP (Internet Telephony Service Provider.) is a "VoIP Phone Company". Example : Vonage, Broadvoice, Teliax, VoicePulse, etc. "All ITSPs suck. Some suck less than others. Teliax seems to suck less than most.." (tm) (c) 2007 ManxPower |
| 19:25.54 | karlh626 | GreggB The company I work for has a MetaSwitch which provides a SIP gateway. I can ask about rates if you want to give me an email address |
| 19:26.07 | lirakis | fetcher: .. how cheap?? i mean.. it would have to be REALLY cheap for me to even think about buying a single phone. I mean Grandstream is cheap.. but they do work ( and get the job done despite what others say) ... but a no name.. random chinese manufacturer... ehhhh... |
| 19:26.27 | *** join/#asterisk akx^ (n=fddsfs@adsl-69-209-162-188.dsl.sfldmi.ameritech.net) |
| 19:26.35 | lirakis | fetcher: by .. really cheap.. i mean .. like $10-20 handset depending on features.. lol |
| 19:27.41 | *** join/#asterisk tristezo2k (n=seba@200.117.247.43) |
| 19:27.44 | karlh626 | Is anyone on that attended Astricon? |
| 19:28.08 | tristezo2k | helo * :D |
| 19:29.21 | tristezo2k | I am using ast 1.4.11 and it works ok. |
| 19:29.21 | tristezo2k | now, I am seeing |
| 19:29.28 | tristezo2k | some errors like chan_sip.c:1938 retrans_pkt: Maximum retries exceeded on transmission |
| 19:29.34 | tristezo2k | and it seems that the call is dropped. |
| 19:30.00 | trippss | [TK]D-Fender: i'm configuring this mediant box now that our PRI has been installed. I have it registered to *. Now I'm trying to figure out the best way to go: 1)register mediant with * and leave phones registered to * and set up * to route outbound calls to mediant and similarly config mediant to hand off phones to *, or 2) register phones directly with mediant and let mediant do the handing off. thoughts? |
| 19:30.06 | tristezo2k | I can not find what that error means, nor if it is fatal for the call.. |
| 19:31.05 | [TK]D-Fender | tripps : *'s role is to control what you dial. The Mediant is NOT a soft-switch |
| 19:32.33 | trippss | [TK]D-Fender: right. so the way i've got it now is the way to go then . . . |
| 19:33.54 | *** join/#asterisk guillote_GNU (n=bancaria@host225.190-30-159.telecom.net.ar) |
| 19:34.08 | lirakis | trippss: the mediant, as [TK] said, is not a soft switch... it is a media gateway. You should use a sip proxy, or SBC to route traffic to the Mediant for conversion and send the converted on to another SBC .. it isnt designed to handle a crapload of endpoints. |
| 19:35.23 | deeperror | Any clues why at random times in the middle of calls dtmf would start? In messages I see....chan_zap.c: Started VLDTMF digit '8' |
| 19:41.40 | watchy | let cuddle |
| 19:43.52 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 19:45.19 | *** join/#asterisk russellb (i=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 19:45.20 | *** mode/#asterisk [+o russellb] by ChanServ |
| 19:45.40 | Aughey | If anyone was around from this morning, the problem with the Sangoma card was a blown FXO module. Found the bad one and pulled it and the rest works fine. |
| 19:46.41 | monux | bbiam |
| 19:46.58 | [TK]D-Fender | Aughey: See if you can RMA it |
| 19:47.03 | *** join/#asterisk Alric (n=nbowyer@masq.hyperusa.com) |
| 19:48.17 | watchy | hey mr tk |
| 19:48.28 | watchy | hows i fix this |
| 19:48.28 | watchy | -- Incoming call: Got SIP response 500 "Internal Server Error" back from 192.168.0.41 |
| 19:48.40 | watchy | my polys always do that |
| 19:49.06 | [TK]D-Fender | watchy: Its just spam, ignore it |
| 19:49.14 | watchy | well i knew that |
| 19:49.18 | watchy | its just annoying :( |
| 19:49.27 | Spida | spit? |
| 19:49.30 | [TK]D-Fender | watchy: reboot your phones & restart * and it will disappear for a while. |
| 19:49.36 | watchy | yea |
| 19:50.35 | lirakis | jeeze.. whats wrong with people |
| 19:50.40 | [TK]D-Fender | "AIDS dies of Herpies, news at 11" |
| 19:51.20 | lirakis | if no where else in this world... i thought IRC would be a safe haven where people could just "get along" ... |
| 19:51.25 | lirakis | ;P |
| 19:52.19 | Alric | Found an interesting situation today with MixMonitor and Queue using the monitor-join option. If I take a call recorded by MixMonitor directly and perform "sox recordedFile.wav -g -r 8000 -c1 recordedFileNew.wav" on it, I get a file that will play normally. If I perform the same command on a call recorded by Queue with monitor-join, I get a file that is either 256 or 320 bytes large. This seems to be happening 100% of the time. |
| 19:52.48 | watchy | i want a freakin icey |
| 19:54.56 | Alric | Any ideas what is causing that? Queue seems to use the same functions MixMonitor does. |
| 19:55.29 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 19:57.28 | *** part/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 19:59.11 | monux | D-fender: suppose the switch is set to another type or it has been updated or whatever... right? zaptel and asterisk would give you some sort of warning? right? |
| 20:04.49 | *** join/#asterisk fujin (n=aj@unaffiliated/fujin) |
| 20:10.26 | rpm | does asterisk support sip aliases? so i can make multiple aliases to a device/peer? |
| 20:11.03 | rpm | i guess that'd be dialplan specific exten => hello@mydomain.com,1,Dial(SIP/exten) |
| 20:11.58 | fujin | sip aliases |
| 20:12.05 | fujin | now why would you want to do that? |
| 20:12.45 | rpm | because i like url dialing and don't like telling people to dial sip:myexten@mydomain.com, i like doing like sip:russell@mydomain.com |
| 20:14.35 | russellb | russell is the coolest name ever |
| 20:14.37 | fujin | I'm not sure I'm seeing the problem. |
| 20:14.57 | rpm | russellb, you got that right :) |
| 20:15.10 | [TK]D-Fender | rpm: thats just an exten, and no you don't get a domain in there. |
| 20:15.48 | [TK]D-Fender | rpm: exten => dumbass,1,Dial(DIP/exten) <------ |
| 20:15.53 | [TK]D-Fender | :p |
| 20:16.06 | rpm | yeah i figured. |
| 20:16.22 | rpm | back to work for me. ttyl. |
| 20:16.33 | *** join/#asterisk akx^ (n=fddsfs@adsl-69-209-162-188.dsl.sfldmi.ameritech.net) |
| 20:16.55 | krondorl | Is there a channel for the (FOP) Front Operator Panel?? |
| 20:19.41 | krondorl | Ok, is there anyone that might be able to help me with the FOP? |
| 20:30.10 | fetcher | krondorl: What trouble are you having? |
| 20:31.45 | krondorl | fetcher: I have the FOP on a web server looking at the * box on a different server and I know I have the configs correct, but the op is not getting any info from the looks of it.. No firewalls.. |
| 20:32.34 | fetcher | krondorl: all the icons are blinking? |
| 20:32.44 | krondorl | fetcher: nope. |
| 20:32.53 | krondorl | fetcher: solid green. |
| 20:33.49 | krondorl | fetcher: we are running the 1.4 * and the latest fop. |
| 20:34.27 | J4k3 | prolly in a jail, to keep my own sanity. |
| 20:34.42 | fetcher | krondorl: did you add an entry for FOP in your /etc/asterisk/manager.conf ? I think * has to be restarted after changing that file, also |
| 20:35.00 | Corydon76-dig | fetcher: it does not |
| 20:35.10 | krondorl | fetcher: I did, and I did a manager reload. |
| 20:35.23 | Corydon76-dig | manager.conf is re-read on each new connection |
| 20:35.33 | fetcher | Corydon76-dig: ah, good to know |
| 20:35.47 | Katty | oh. |
| 20:35.48 | Katty | hello. |
| 20:35.55 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 20:36.03 | Corydon76-dig | Hello, Katty |
| 20:36.34 | krondorl | fetcher: the manager show connected but nothing there, but if i do a manager show users I see the managers.conf info. |
| 20:36.44 | fetcher | I think all green icons means the Flash applet is at least communicating with your op_server.pl |
| 20:37.41 | krondorl | fetcher: :) yup. I have run this on the same machine before but this time we want the fop on a complete different system.. |
| 20:38.29 | fetcher | krondorl: in manager.conf, you may need to change a line under [general] from "bindaddr = 127.0.0.1" to include the IP you want it to listen on (or 0.0.0.0 for all) |
| 20:39.02 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 20:39.03 | fetcher | krondorl: 127.0.0.1 accepts manager connections only from clients local to the Asterisk machine itself |
| 20:40.27 | krondorl | fetcher: Isn't the bindaddr the address of the machine * is running on and the permit=222.222.222.222/.... the one it's listening for? |
| 20:41.21 | trippss | mmmm trying to get mediant to work - keeps telling me [ERROR] #0:TrunkGroup::AllocateEndPoint - can't find endpoint for number 8005551212 . . . . but the manual says endpoints are only for FXS and i'm using T1 digital interface . . . any ideas? |
| 20:42.11 | fetcher | krondorl: yes. listen= can be the local eth0 address. |
| 20:43.23 | *** join/#asterisk t3rror (n=t3rror@gateway.sscgp.com) |
| 20:43.41 | lirakis | bye everyone |
| 20:43.41 | fetcher | krondorl: which also needs to be specified in op_server.cfg (manager_host=ip.of.asterisk.box) |
| 20:43.54 | *** join/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 20:44.02 | *** part/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 20:44.40 | krondorl | fetcher: and it is. both are to port 5038 also. |
| 20:45.28 | fetcher | krondorl: can you telnet to port 5038 (on * host) from the web server? |
| 20:45.29 | Nugget | telnet is eeeeeeevil! |
| 20:45.31 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 20:46.52 | krondorl | fetcher: Hmmm, I get Name or service not known. IP:5038 |
| 20:47.55 | *** join/#asterisk ZackZ (n=zzumbaug@rrcs-24-123-106-250.central.biz.rr.com) |
| 20:48.18 | ZackZ | hello |
| 20:48.59 | krondorl | Hi Zack. |
| 20:49.08 | ZackZ | i have a T1 PRI with 100 phone numbers, is there any way to force Asterisk to dial out on a specific number? It always dials out using the first number in the pool |
| 20:49.14 | ZackZ | I have a Digium TE120P |
| 20:49.29 | ZackZ | Asterisk 1.2 |
| 20:50.58 | outtolunc | you don't 'dial out' on a number, you dial out on a channel |
| 20:51.03 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 20:51.15 | Strom_C | ZackZ: set your callerid number before you dial |
| 20:51.21 | ZackZ | yes but is there a way to make it "pick" a number to show the callerID of the person you are calling? |
| 20:51.32 | *** part/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 20:51.34 | Strom_C | ZackZ: set your callerid number before you dial |
| 20:51.40 | ZackZ | meaning, we have 555-1000 through 555-1099 |
| 20:51.54 | ZackZ | it always shows 555-1000 when calling out |
| 20:51.58 | Strom_C | clearly, I'm IRCing into /dev/null/ |
| 20:52.09 | outtolunc | as strom is telling you, set the callerid(num) prior to Dial |
| 20:52.32 | watchy | ive been poisioned |
| 20:53.51 | watchy | i wish i was britney spears |
| 20:54.38 | ZackZ | so would it be Set(CALLERID(555-4525))? |
| 20:55.41 | *** join/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) |
| 20:55.46 | Strom_C | ZackZ: no |
| 20:55.58 | Strom_C | Set(CALLERID(num)=3115552368) |
| 20:56.27 | fujin | some providers make it so you can only set the outgoing callerid to one of the numbers pointed at the PRI circuit |
| 20:56.28 | fujin | ;[ |
| 20:56.29 | ZackZ | ok |
| 20:56.30 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 20:56.57 | ZackZ | i did have Set(CALLERID(all) = Name 5553435353) but that wasnt working |
| 20:57.06 | fujin | heh |
| 20:57.12 | fujin | I usually set name/num seperately |
| 20:57.17 | fujin | just so you know it's working |
| 20:57.18 | ZackZ | alright |
| 20:57.24 | ZackZ | ill give it a try, thakns guys |
| 20:58.21 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 20:59.57 | ZackZ | all of our numbers should be pointed at the PRI circuit |
| 21:01.49 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 21:03.24 | *** join/#asterisk BockBilbo (n=BockBilb@eu85-84-62-227.clientes.euskaltel.es) |
| 21:04.17 | *** join/#asterisk zydrunas (n=Miranda@216.49.228-ip-184.ckt.net) |
| 21:04.31 | BockBilbo | hello! |
| 21:04.53 | BockBilbo | ive just configured my asterisk server to let my users call externa sip server's users |
| 21:05.31 | BockBilbo | it works fine, but when the other servers user receive my call, instead of showing my domain it shows my IP |
| 21:05.54 | BockBilbo | is there a way to fix this so my domain is showed there? Maybe using a key value on the database? |
| 21:06.13 | aiksa[LV] | Strom_C: didnt you know that freenode actually dumps everythin into /dev/null and any response we see is only a sign of being self deluded crackpot? |
| 21:06.27 | Strom_C | O RLY |
| 21:08.03 | krondorl | fetcher: I give up for today.. something is blocking it somewhere... time to go home... |
| 21:08.11 | aiksa[LV] | nevermind, I am just tired and talking rubish |
| 21:08.41 | aiksa[LV] | have been talking to dev/null IRL for last 12 hrs. |
| 21:16.17 | ZackZ | ah, /dev/null, hillarious |
| 21:16.18 | ZackZ | not really |
| 21:16.23 | *** join/#asterisk WilliamK (n=noc@static-71-170-144-3.dllstx.fios.verizon.net) |
| 21:16.49 | *** part/#asterisk ZackZ (n=zzumbaug@rrcs-24-123-106-250.central.biz.rr.com) |
| 21:18.46 | *** join/#asterisk tsurko (n=tsurko@78.90.100.214) |
| 21:18.53 | edwin_quijada | we have a probelm compiling zaptel 1.20 |
| 21:19.18 | edwin_quijada | where can I paste thsi error? |
| 21:19.21 | edwin_quijada | paste |
| 21:19.49 | elixer | ~pb |
| 21:19.50 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 21:22.15 | edwin_quijada | http://pastebin.com/m6cea0b16 |
| 21:22.38 | edwin_quijada | this is the error that I get when I compiling zaptel http://pastebin.com/m6cea0b16 |
| 21:22.42 | *** part/#asterisk karlh626 (n=kharris@IP-66.249.227.5.indigital.net) |
| 21:23.09 | aiksa[LV] | edwin_quijada: cant you get that in english? |
| 21:23.50 | fujin | lol |
| 21:23.52 | fujin | that'd be useful :) |
| 21:24.15 | tristezo2k | edwin_quijada: it seems you are having two different libc! |
| 21:24.24 | tristezo2k | or the linker is linking against the wrong one/ |
| 21:24.41 | tristezo2k | what does configure tells you? |
| 21:24.54 | edwin_quijada | i dont run configure |
| 21:25.25 | edwin_quijada | how can i see the libc |
| 21:25.28 | edwin_quijada | version |
| 21:25.29 | aiksa[LV] | i am not sure of zaptel. but running ./configure before the make is usually a good idea :) |
| 21:25.51 | tristezo2k | try something like ldconfig -v|grep libc |
| 21:26.21 | tristezo2k | see there is a nonshared version of libc |
| 21:26.31 | edwin_quijada | aiksa[LV]: zaptel doesnt have configure |
| 21:26.36 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 21:26.36 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 21:27.12 | *** join/#asterisk duckz (n=duckz@81-31-157-49.vm.dnshosting.it) |
| 21:27.15 | tristezo2k | Wich linux distro are you using? |
| 21:27.44 | edwin_quijada | debian 3.1 |
| 21:28.03 | aiksa[LV] | edwin_quijada: okay; been awhile since i last compiled it |
| 21:28.25 | tristezo2k | I should tell you to use etch 4.0.. |
| 21:28.27 | tristezo2k | but anyway, |
| 21:28.40 | tristezo2k | it seems you have libc static installed |
| 21:28.51 | edwin_quijada | thsi is the output from command |
| 21:28.52 | edwin_quijada | http://pastebin.com/m48b8c440 |
| 21:30.06 | edwin_quijada | tristezo2k: how can I get another |
| 21:31.14 | anonymouz666 | disconnect a call is the same as Hangup ? |
| 21:31.39 | tristezo2k | mmmm I don´t really know. |
| 21:33.19 | tristezo2k | can you afford to reinstall? |
| 21:37.40 | edwin_quijada | yes |
| 21:37.46 | edwin_quijada | but reinstall what? |
| 21:37.52 | aiksa[LV] | btw. zaptel has active versions of 1.2 and 1.4 so does * |
| 21:38.03 | aiksa[LV] | does they cross mix? |
| 21:38.20 | edwin_quijada | aiksa[LV]: yes |
| 21:38.25 | aiksa[LV] | I mean 1.2 ast with 1.4 zaptel and vice versa |
| 21:38.28 | edwin_quijada | this is the last version for 1.2 |
| 21:38.39 | aiksa[LV] | edwin_quijada: i know |
| 21:38.41 | edwin_quijada | cant I |
| 21:39.02 | edwin_quijada | can i use zaptel 1.4 with aster 1.2 |
| 21:39.03 | edwin_quijada | ? |
| 21:39.15 | aiksa[LV] | that was my question :) |
| 21:40.26 | anonymouz666 | I wouldn't |
| 21:43.46 | *** join/#asterisk frest (n=stromber@loke.csbnet.se) |
| 21:44.58 | frest | is there a special channel for asking asterisk-related questions? |
| 21:45.10 | *** part/#asterisk Alric (n=nbowyer@masq.hyperusa.com) |
| 21:45.40 | zerohalo | other than this one? |
| 21:46.04 | frest | great :) |
| 21:46.31 | Strom_C | no,i'm sorry sir, this is the channel for asking questions related to the asterisk key on your numeric keypad |
| 21:46.46 | Strom_C | how hard to press it, when to use it, what to give it for Christmas |
| 21:46.50 | frest | :) |
| 21:47.17 | zerohalo | I've found the best way to ask * related questions is to just ask.... You'll get flames, whining, and a lot of nothing, but eventually, someone may answer. |
| 21:47.32 | frest | Ive just installed asterisk, and when I connect I just hear the first split second of the test message |
| 21:47.51 | aiksa[LV] | and the the asterisk crashes to the halt? |
| 21:47.51 | frest | then it is quiet, but no hangup |
| 21:48.01 | frest | no |
| 21:48.28 | aiksa[LV] | set verbose 4 and paste cli output in pastebin |
| 21:48.41 | zerohalo | That's a little vague. Connecting using what? What are you seeing on the CLI when you test call? |
| 21:48.46 | aiksa[LV] | btw did they remove callbackagentlogin in 1.4 |
| 21:49.11 | zerohalo | aiksa: No. It's being deprecated, but it's still there and usable. |
| 21:50.00 | hmmhesays | INVITE sip:6783094161;rn=17702009999;npdi=yes@216.253.240.99;user=phone SIP/2.0 |
| 21:50.00 | aiksa[LV] | zerohalo: the point of it being depricitaed is? That users should try to achieve that with dsimple ialplan comand? |
| 21:50.07 | hmmhesays | you guys ever see an invite like that |
| 21:50.11 | hmmhesays | what is the rn=? |
| 21:52.09 | zerohalo | aiksa: not sure of the why... I have a bit of dialplan I'm putting off working on which uses it. |
| 21:52.26 | file | hmmhesays: routing number |
| 21:53.02 | hmmhesays | thats what I thought, it doesn't make sense to me why you would use something like that |
| 21:53.10 | hmmhesays | instead of routing based on the to: field |
| 21:54.06 | file | it's for number portability stuff... sending the info along |
| 21:55.02 | aiksa[LV] | zerohalo: sad. I have a large AMI connected server doing a lot of stuff with those commands. when i am imagining the amount of redo for that (brrr - scary) |
| 21:55.05 | hmmhesays | can you explain that a little more in depth? |
| 21:55.08 | hmmhesays | or should I google |
| 21:55.15 | *** part/#asterisk tristezo2k (n=seba@200.117.247.43) |
| 21:55.15 | Katty | hmmhesays: herro (= |
| 21:55.33 | *** join/#asterisk LeddyHM (n=NONE@70.242.16.97) |
| 21:55.36 | hmmhesays | hey Katty |
| 21:55.36 | file | hmmhesays: Google can tell you more than I can |
| 21:56.10 | frest | the CLI just says "Playing 'demo-congrats' (language 'en')", but I only hear the first half second or so |
| 21:56.18 | aiksa[LV] | file: dont underestimate your abilities |
| 21:56.43 | frest | then there is no sound, but the call isnt ended |
| 21:56.44 | aiksa[LV] | frest: no warnings, errors , anything? |
| 21:57.13 | frest | well, I start asterisk using "asterisk -c", and it displays a few warnings |
| 21:57.23 | aiksa[LV] | what are you using to dial in into that number? softphone? |
| 21:57.40 | aiksa[LV] | frest: i mean during and upon termination of the call |
| 21:58.29 | frest | no, no warnings |
| 21:58.35 | frest | aiksa[LV]: yes, a softphone |
| 21:58.43 | frest | SIP Communicator for Mac OS X |
| 21:58.53 | frest | do you recommend something else? |
| 21:58.57 | aiksa[LV] | i have seen this ages ago with ast BE |
| 21:59.13 | aiksa[LV] | cant remember the cause right now |
| 21:59.32 | aiksa[LV] | frest: sorry, i am win32 and linux user |
| 21:59.43 | aiksa[LV] | cant recomend on softphones for macos |
| 22:00.35 | frest | ok. seems there isnt much to choose from on mac os x |
| 22:02.43 | *** part/#asterisk rudholm (i=rudholmm@nat/yahoo/x-5c3b210caae55255) |
| 22:03.49 | aiksa[LV] | anyone with rather deep understanding of zaptel and PRI here? |
| 22:04.05 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:05.32 | aiksa[LV] | nevertheless I'll give this a shot: |
| 22:05.40 | aiksa[LV] | <PROTECTED> |
| 22:05.51 | aiksa[LV] | i have E1 line |
| 22:06.07 | aiksa[LV] | incomming calls have worked perfectly for about 3 years |
| 22:06.58 | aiksa[LV] | now hen trying to make and outgoing calls , on 10-40% of those calls the receiving party gets a loud noise on their telephones |
| 22:07.23 | aiksa[LV] | and asterisk spits out 'unable to set linear on ...' |
| 22:08.03 | hmmhesays | so it would be unusual to see disconnected numbers in the rn= field |
| 22:09.20 | aiksa[LV] | more details: http://pastebin.com/m2a315c7b |
| 22:12.17 | aiksa[LV] | what seems strange to me is the naming of those channels |
| 22:12.41 | *** join/#asterisk TUplink_ (n=mythtv@c-24-126-34-203.hsd1.wv.comcast.net) |
| 22:12.53 | aiksa[LV] | while a normal outgoing call (no noise) would have something like Zap/3-1 |
| 22:12.58 | TUplink_ | is there a way to download AMP to run on a standalone install of asterisk? |
| 22:13.10 | aiksa[LV] | the failing calls would have Zap/1:3-1 |
| 22:13.17 | outtolunc | aiksa, few issues i noticed were you are using callerid but not sending any<G>, then you have the it 'moving' the active channel from 15 to 3 (first pastebin), then you have Facility (len=20, codeset=0) [ 0x91, 0xa1, 0x0f, 0x02, 0x02, 'qm', 0x02, 0x01, 0x0f, '0', 0x06, 0x02, 0x01, 0x01, 0x0a, 0x01, 0x01 ] notice the 'qm' and '0' |
| 22:14.03 | aiksa[LV] | okaym who is moving that channel? |
| 22:14.09 | aiksa[LV] | remote party? |
| 22:14.11 | outtolunc | # |
| 22:14.11 | outtolunc | <PROTECTED> |
| 22:14.11 | outtolunc | # |
| 22:14.11 | outtolunc | Oct 2 17:06:32 WARNING[26197]: chan_zap.c:4950 zt_write: Unable to set linear mode on channel 3 |
| 22:14.25 | aiksa[LV] | i have seen that |
| 22:14.39 | aiksa[LV] | and googled my ass off trying to find some clues |
| 22:14.46 | outtolunc | looks like asterisk (chan_zap/libpri/zaptel) or whatever else you got in there |
| 22:15.22 | outtolunc | what version you using? |
| 22:15.44 | [TK]D-Fender | TUplink_, AMP = FreePBX now. goto http://www.freepbx.org/ to find out about, but understand we do NOT support it here. |
| 22:15.45 | aiksa[LV] | <PROTECTED> |
| 22:15.45 | aiksa[LV] | <PROTECTED> |
| 22:15.45 | aiksa[LV] | <PROTECTED> |
| 22:15.59 | aiksa[LV] | sorry about that |
| 22:16.03 | TUplink_ | i jsut want the webportal |
| 22:16.13 | TUplink_ | just want to see what it is all about |
| 22:16.14 | aiksa[LV] | the system has Asterisk 1.2.20 installed with zaptel 1.2.18 & libpri 1.2.5. The kernel version is 2.6.21.5. |
| 22:16.23 | [TK]D-Fender | TUplink_, that IS all that FreePBX is. |
| 22:16.44 | TUplink_ | oh... i thought freepbx was asterisk and all |
| 22:16.46 | outtolunc | did you apply that last user-user mod? to q931.c? |
| 22:16.48 | [TK]D-Fender | TUplink_, and there is no "just" about it. Once you install it, it owns you. |
| 22:17.03 | TUplink_ | ok... forget that then |
| 22:17.14 | aiksa[LV] | outtolunc: no, its installed as is |
| 22:17.17 | TUplink_ | maybe ill install it on another box |
| 22:17.21 | aiksa[LV] | no patches applied |
| 22:17.23 | [TK]D-Fender | TUplink_, TRIXBOX is a distro that includes *, FreePBX, and some other bits, and it is EQUALLY unsupported here. |
| 22:17.43 | TUplink_ | any of you all used it,,,, is it a live distro? |
| 22:17.55 | [TK]D-Fender | TUplink_, No, it formats & installs. |
| 22:17.59 | TUplink_ | like do you have to install it |
| 22:18.02 | TUplink_ | damn |
| 22:18.35 | aiksa[LV] | outtolunc: 'last user-user mod? to q931.c' - its a patch? |
| 22:18.56 | outtolunc | aiksa[LV], i would try applying the user-user mod, and/or either set usecallerid to no, and or send it and see if it changes |
| 22:19.15 | outtolunc | yeah the user-user and another ie were switched |
| 22:19.39 | *** part/#asterisk TUplink_ (n=mythtv@c-24-126-34-203.hsd1.wv.comcast.net) |
| 22:19.41 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 22:19.44 | aiksa[LV] | I should grab them from bugtrack / digium? - where to find it? |
| 22:19.51 | outtolunc | just a sec |
| 22:22.05 | *** join/#asterisk remmo (n=junk@203.32.47.250) |
| 22:22.06 | outtolunc | http://dynx.net/ASTERISK/diff-patches/useruser.notes |
| 22:22.08 | aiksa[LV] | my dialplan has Set(CALLERID(num)=26200000) before dialing through that interface. Will I have to to diable that? |
| 22:22.21 | outtolunc | that is for libpri/q931.c |
| 22:22.49 | aiksa[LV] | many thanks. |
| 22:23.08 | aiksa[LV] | I will try that tomorrow when there are persons in the office who can call me |
| 22:23.24 | outtolunc | k |
| 22:24.00 | aiksa[LV] | outtolunc: another strange thing was - that it has irregular nature |
| 22:24.18 | aiksa[LV] | under seemingly smae conditions one call would work and the next would fail |
| 22:24.35 | aiksa[LV] | made from same inner extension |
| 22:24.47 | outtolunc | are you sure you aren't overloading the facilities in the path |
| 22:25.25 | aiksa[LV] | overloading - like restarting the channels? |
| 22:26.04 | outtolunc | overloading as in 'overrunning' the switches in the call path with too many calls (aren't you the one that mentioned vici-dial?) |
| 22:26.26 | aiksa[LV] | outtolunc: no its not me |
| 22:26.31 | outtolunc | sorry |
| 22:26.39 | aiksa[LV] | must be somenone else here |
| 22:26.51 | aiksa[LV] | the physical path is rather simple |
| 22:27.13 | aiksa[LV] | [telco1] --- (leased line from another telco) --- my pbx |
| 22:27.24 | outtolunc | so even single calls on otherwise idle system are 'irratic'? |
| 22:28.07 | aiksa[LV] | yes. |
| 22:28.13 | outtolunc | what changed |
| 22:28.41 | aiksa[LV] | and there might be situation where active incoming call is completly fine while at the same time outbound fails |
| 22:29.08 | aiksa[LV] | nothing: that line was never used for outgoing calls for 3 years |
| 22:29.14 | outtolunc | ah |
| 22:29.22 | outtolunc | but now you are starting to use outbound |
| 22:29.29 | aiksa[LV] | oh yes :) |
| 22:29.35 | aiksa[LV] | and here the goodness starts |
| 22:29.58 | *** join/#asterisk anthm (n=anthm@mb60736d0.tmodns.net) |
| 22:29.58 | *** mode/#asterisk [+o anthm] by ChanServ |
| 22:30.15 | aiksa[LV] | could it be related to me having dchan on the first timeframe? |
| 22:31.21 | aiksa[LV] | like some hardcoded things very deep inside zap which are trying to use 1st timefarme as a carrier for voice? |
| 22:32.14 | outtolunc | the only issues i seen, were the ones i mentioned |
| 22:32.21 | outtolunc | i'd focus on them first |
| 22:32.55 | outtolunc | so make sure you are sending callerid (set in zapata.conf a generic one for your server/site), or disable it |
| 22:33.19 | outtolunc | then make sure you aren't attempting any channel to channel tranfers |
| 22:33.39 | outtolunc | (because iirc, those do not work in 1.2) |
| 22:34.01 | aiksa[LV] | outtolunc: how could I accomplish those transfers? |
| 22:34.10 | outtolunc | and make sure you test fixing the setup_ies |
| 22:34.35 | aiksa[LV] | perhaps i am doing it and not being aware of it |
| 22:34.48 | hmmhesays | ok number portability is making my head hurt |
| 22:35.22 | outtolunc | meaning, for the 'test' do not pickup one channel in a 'group' as outbound and have it dial a inbound DID in the same group <G> |
| 22:36.21 | *** join/#asterisk frocos11292 (n=ask@firewall.vipvoz.com) |
| 22:36.29 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 22:36.33 | outtolunc | dial someone elsewhere just to get a clean pri debug |
| 22:36.37 | aiksa[LV] | outtolunc: many thanks for the help nevertheless - been asking about this prob. for last few days snd this is first answering giving at least some clues |
| 22:36.46 | *** join/#asterisk astraeis (n=sbma44@dsl092-173-003.wdc2.dsl.speakeasy.net) |
| 22:36.48 | frocos11292 | anyone can help me with callerid on supervised call transfers?? |
| 22:36.54 | outtolunc | no prob |
| 22:37.41 | frocos11292 | asterisk sends the transferer callerid instead of the original... |
| 22:38.02 | aiksa[LV] | guess - most of the problems discussed here arent that hard to solve |
| 22:38.03 | outtolunc | add o to dial string iirc |
| 22:38.10 | outtolunc | er /o |
| 22:38.43 | aiksa[LV] | sorry for stupid question - /o means what? and what iirc stands for? |
| 22:38.47 | frocos11292 | outtolunc-> that works for blind transfers |
| 22:38.49 | [TK]D-Fender | frocos11292, thats not a problem, that what is EXPECTED for behavior |
| 22:39.11 | [TK]D-Fender | frocos11292, If you want the original CID passed you must do a blind transfer. |
| 22:39.12 | frocos11292 | TK]D-Fender-> any possible workaround? |
| 22:39.19 | [TK]D-Fender | frocos11292, BLIND <------- |
| 22:39.43 | astraeis | Hey all. Got an AGI question. I'm trying to run a Tetris game on the monitors in our lobby with an Asterisk interface. It works pretty well, but the AGI that catches keypresses and sends them to the game doesn't exit cleanly when the user hangs up. As a result a socket and a lockfile don't get deleted, and I have to manually reset the system after each call. When hangup occurs the script is in a pretty simple loop consisting |
| 22:39.50 | astraeis | any perl AGI folks with ideas? |
| 22:40.09 | frocos11292 | TK]D-Fender-> client doesn't want blind wants supervised |
| 22:40.24 | [TK]D-Fender | frocos11292, tell them "too bad". This is the way it is. |
| 22:40.39 | frocos11292 | [TK]D-Fender-> lol, i wish |
| 22:40.50 | Corydon76-dig | astraeis: do you have a signal handler for $SIG{HUP} ? |
| 22:41.05 | [TK]D-Fender | frocos11292, fine then lie to them and tell them it'll take you more programming that they can budget for. |
| 22:41.16 | astraeis | no I don't -- didn't realize that the AGI package made that available |
| 22:41.27 | Corydon76-dig | It doesn't. Perl does. |
| 22:41.31 | astraeis | ah |
| 22:41.31 | hmmhesays | I need to find a dumbed down guide on how local number portability works |
| 22:42.08 | astraeis | aha... sorry about that. this project has demanded more Perl skills than I first thought it would. Let me try dropping that in. |
| 22:42.11 | frocos11292 | [TK]D-Fender, hum... anyway this is a normal feature in tradicional pbx.. like alcatel or siemens, and it makes sense... |
| 22:42.30 | WilliamK | hmmhesays: from the consumer perspective or service provider perspective? |
| 22:42.30 | [TK]D-Fender | frocos11292, Doesn't match most that I've seen. |
| 22:43.06 | [TK]D-Fender | frocos11292, the point of an attended transfer is so they can see that YOU are calling them so maybe they don't ignore the caller you want to pass off. If they saw the CID themselves they might otherwise ignore it |
| 22:43.11 | frocos11292 | [TK]D-Fender-> i receive a transfered call from the outside, maybe i want to call back that person, but my phone keeps the extension that passed the call in the logs instead of the original |
| 22:43.54 | [TK]D-Fender | frocos11292, why are you bothering with "attended" in this case? |
| 22:44.10 | frocos11292 | [TK]D-Fender->i see ur point, but in here this is a normal expectable feature |
| 22:44.36 | [TK]D-Fender | frocos11292, there are going to be differences between any set of systems, THAT is to be expected. |
| 22:44.54 | hmmhesays | WilliamK: service provider perspective, this rfc is confusing as sh1t |
| 22:45.11 | frocos11292 | [TK]D-Fender-> ok.. suppose it would be nice to have this feature, any ideas how could we do it? |
| 22:45.22 | *** join/#asterisk saint_ (n=saint@c-69-242-118-124.hsd1.nj.comcast.net) |
| 22:45.23 | tristanbob | http://useopensource.blogspot.com/2007/10/digium-is-doing-things-right.html |
| 22:45.30 | [TK]D-Fender | frocos11292, serious reprogramming of chan_sip at a minimum. |
| 22:45.49 | frocos11292 | [TK]D-Fender-> that's what i thought |
| 22:45.53 | astraeis | Corydon76: you're my hero. looks like that did it. |
| 22:46.12 | saint_ | hi all.. anyone ever configured a sip trunk with asterisk, going to an Alcatel PBX ? |
| 22:49.31 | [TK]D-Fender | saint_, have YOU tried? |
| 22:49.32 | *** join/#asterisk drwelby (n=mpfister@mail.enplan.com) |
| 22:49.58 | *** join/#asterisk jetlagmk2 (i=jetlag@pool-70-17-59-195.pskn.east.verizon.net) |
| 22:50.04 | saint_ | [TK]D-Fender, yes.. and i have some issues with the voicemail .. |
| 22:50.20 | [TK]D-Fender | saint_, which sides? And what "issues"? |
| 22:50.44 | *** join/#asterisk pfn (n=pfnguyen@hanhuy.com) |
| 22:50.53 | saint_ | when a sip user from the asterisk calls the user on the alcatel, instead of gonig to the personal alcatel voicemail, it goes to the main voicemail |
| 22:52.44 | [TK]D-Fender | saint_, also on the alcatel? |
| 22:52.48 | *** part/#asterisk frocos11292 (n=ask@firewall.vipvoz.com) |
| 22:52.59 | saint_ | [TK]D-Fender, no, everything works fine from alcatel to alcatel |
| 22:53.14 | saint_ | it s just when my asterisk sip user dials the alcatel through the siup trunk |
| 22:53.23 | [TK]D-Fender | saint_, I mean this "main" voicemail that it lands on. Thats on the ALcotel as well? Just not the CALLEE's VM? |
| 22:53.36 | saint_ | i can see the re-invite message for the voicemail , with the extention number, but it does not go to the destination user's voicemail |
| 22:54.06 | [TK]D-Fender | saint_, pastebin it up with SIP debug enabled. |
| 22:54.07 | [TK]D-Fender | ~pb |
| 22:54.08 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 22:54.09 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^ |
| 22:54.16 | saint_ | it goes to the main VM on the alcatel "Welcome to the VM, if you have a voicemail on this system, press xxx" |
| 22:54.40 | saint_ | [TK]D-Fender, you have an alcatel ? |
| 22:55.04 | [TK]D-Fender | saint_, No but may have some insight from what your PB will show |
| 22:56.09 | saint_ | let me play with it some more before i pastebin. if you don t have an alcatel, it s going to be hard to troubleshot. i m sure there is an issue on the alcatel somewhere. i ll try with another sip client on the asterisk .. we ll see |
| 22:56.17 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 22:57.01 | [hC] | Is it in any way possible on a polycom phone, when doing a blind xfer using the softkeys, to have asterisk show the original caller id to the transfer recipient? |
| 22:57.06 | [hC] | fender, you might know this out of anyone.. |
| 22:57.14 | [hC] | I dont think its possible without using asterisk's transfer function itself.. |
| 22:57.22 | [TK]D-Fender | saint_, jsut a note that it should cause a redirect from 1 exten on the Alcatel to another just for VM. the call should be maintained end-to-end on the INSIDE. no reason for a re-invite that I can see. |
| 22:57.40 | [TK]D-Fender | saint_, unless there are multiple SIP servers implemented in its archetecture. |
| 22:57.54 | *** part/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 22:58.08 | [TK]D-Fender | [hC], A blind transfer DOES this already |
| 22:58.56 | [TK]D-Fender | [hC], so 123 calls YOU. YOU see 123. you then [Transfer] [Blind] 456 [Send]. 456 sees 123 calling, not YOU. |
| 22:59.22 | [hC] | [TK]D-Fender: Well then, classic case of trusting what someone came to me with as true before checking myself |
| 22:59.25 | [hC] | I'll take my whippings. |
| 22:59.26 | [TK]D-Fender | </capitalization_abuse> |
| 23:00.08 | [TK]D-Fender | [hC], The customer is always right? Hell no! You do this for a LIVING! |
| 23:00.27 | [TK]D-Fender | [hC], You'd swear they break it for FUN! |
| 23:01.14 | [hC] | Hahaha. |
| 23:01.22 | [hC] | I will send trolls with flaming arrows........ NOW |
| 23:02.06 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 23:03.53 | *** join/#asterisk crispier (n=crispy@pool-72-64-106-201.dllstx.fios.verizon.net) |
| 23:07.26 | blitzrage | but those colourful haired creatures are so cute! |
| 23:07.47 | blitzrage | I bet they'd make a great molitov cocktail with that built in wick though :) |
| 23:08.27 | *** part/#asterisk zerohalo (n=zeroHalo@pool-72-70-79-233.bstnma.east.verizon.net) |
| 23:09.02 | *** join/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 23:10.37 | crispier | hello, can someone help me provision an iaxy? |
| 23:27.32 | *** join/#asterisk syle (n=blag@unaffiliated/syle) |
| 23:27.44 | *** join/#asterisk craigk (n=ckowald@58.174.122.198) |
| 23:30.27 | riddlebox | hrmm I wonder why when I dial out of my zap channel, when the called party answers, there is one more ring in my ear? |
| 23:30.37 | *** join/#asterisk TimothyP (n=timothy@116.252-243-81.adsl-static.isp.belgacom.be) |
| 23:31.44 | TimothyP | Hello, I have a working 1.2 installation with b410p card connected to the telco and some SIP clients. I would like to upgrade to 1.4. The system is ubuntu server. Is it safe to update, what should I look out for ? Do first? |
| 23:35.07 | *** part/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 23:35.15 | *** join/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 23:35.46 | *** part/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 23:35.46 | luke-jr | TimothyP: why do you want to upgrade? |
| 23:35.55 | TimothyP | I want to get AsteriskGUI working |
| 23:35.59 | TimothyP | and it said you need 1.4 for that? |
| 23:36.08 | *** join/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 23:36.26 | TimothyP | hoping it will easy the management |
| 23:36.43 | TimothyP | We implemented asterisk some time ago, but the company we work for isn't to impressed :( |
| 23:37.06 | TimothyP | most of the time IAX uplinks to FWD aren't working and they complain there is no managebility except for when we edit the config files manually |
| 23:37.30 | *** join/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 23:37.46 | *** part/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 23:38.34 | TimothyP | what do you think? |
| 23:38.59 | *** join/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 23:41.05 | TimothyP | luke-jr ? |
| 23:42.53 | *** join/#asterisk coppice (n=chatzill@153.201.17.210.dyn.pacific.net.hk) |
| 23:43.48 | tzanger | good evening coppice |
| 23:43.59 | coppice | good morning |
| 23:44.52 | [TK]D-Fender | good grief |
| 23:45.26 | chemikk | good morning (1:45 AM) :) |
| 23:46.01 | coppice | that's so "old world" |
| 23:47.33 | *** join/#asterisk blinky42 (n=me@c-71-230-47-244.hsd1.pa.comcast.net) |
| 23:54.27 | blitzrage | good afternoon! |
| 23:59.11 | *** part/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
