| 00:00.32 | mmlj4 | ManxPower: my client is liking the idea of letting * do voicemail for them |
| 00:00.41 | Moonwick | it definitely seems like it's caused by a feedback loop, because if I mute the phone the fading goes away |
| 00:00.48 | mmlj4 | the manager was asking me about IP phones, in fact |
| 00:00.55 | *** join/#asterisk _SMP_ (~SMP@pandora.burned.net) |
| 00:01.04 | mmlj4 | that surprised me, in fact |
| 00:01.17 | Moonwick | heh, I suppose that's a good enough fix. I just use speakerphone for waiting on hold. |
| 00:03.08 | drumkilla | you know what is annoying? waiting 5 minutes for the stupid web interface on the polycom to respond after a reboot |
| 00:03.15 | *** join/#asterisk REdOG (~REdOG@REdOG.user.gentoo) |
| 00:03.25 | file | yay drumkilla |
| 00:03.45 | REdOG | im getting an error trying to load asterisk; Loading module chan_zap.so failed! |
| 00:03.53 | *** join/#asterisk adker (~adker@67-51-234-116.dsl1.glv.ny.frontiernet.net) |
| 00:03.53 | REdOG | what provides chan_zap.so ? |
| 00:07.42 | Nethab | moonwick, that's the compressor in the speakerphone |
| 00:08.46 | Nethab | it's what allows you to hear very quiet background noises when your not talking, and very loud voices right next to the microphone |
| 00:11.01 | Nethab | are the polycoms bad? |
| 00:13.47 | *** part/#asterisk REdOG (~REdOG@REdOG.user.gentoo) |
| 00:17.22 | blitzrage | Nethab: anything but :) |
| 00:17.45 | Nethab | anything but grandstreams you mean |
| 00:18.11 | Moc | Nethab, polycom are very VERY good |
| 00:18.12 | drumkilla | i wonder how there new phone is ... |
| 00:18.21 | drumkilla | Moc: want to set mine up for me? :) |
| 00:18.30 | Nethab | good i just got two of them |
| 00:18.42 | Moc | drumkilla, having problem setting it up ? |
| 00:18.54 | drumkilla | yeah, I just have no idea what I'm doing |
| 00:19.00 | Moc | lol |
| 00:19.05 | Moc | IP 500 ? |
| 00:19.07 | drumkilla | so, that's pretty much the problem. :) |
| 00:19.08 | drumkilla | ip300 |
| 00:19.20 | drumkilla | they sent me the files for the boot server |
| 00:19.26 | drumkilla | just not sure what to put where |
| 00:19.43 | *** join/#asterisk los415 (~los415@c-24-126-63-233.hsd1.ca.comcast.net) |
| 00:19.48 | Moc | let me package you a folder for the phone |
| 00:19.54 | drumkilla | ok :D |
| 00:19.56 | drumkilla | you're awesome |
| 00:20.27 | Nethab | then why does everyone get Cisco phones, just cause there on TV shows? |
| 00:20.29 | phantasis | when doing iax to iax on a lan all I hear is a pulse type noise...what gives? |
| 00:20.43 | Moc | Nethab, these day, people get less and less cisco.. |
| 00:20.55 | Nethab | i hear nothing but upgrade problems for Cisco |
| 00:21.02 | Moc | Nethab, they just look 'good' but they arnt very practical |
| 00:21.14 | drumkilla | yeah, this 7960 is looking pretty sweet on my desk |
| 00:21.28 | Moc | but I still prefere my IP 500 and 600 |
| 00:21.31 | Nethab | but the best looking cisco 7970 isn't SIP |
| 00:21.38 | drumkilla | no kidding |
| 00:21.42 | Shido | yet |
| 00:21.43 | drumkilla | hopefully it will be soon ... |
| 00:21.48 | los415 | cisco's are a good looking phone |
| 00:21.56 | los415 | but they suck with nat |
| 00:21.56 | Nethab | well the 7971 came out and it wasn't sip either |
| 00:21.58 | Moc | Cisco will flush their skinny protocol in favor of SIP |
| 00:22.03 | syle | damn checking out gmail |
| 00:22.08 | syle | 2 gigs of space |
| 00:22.26 | syle | i got a new spam box :) |
| 00:22.28 | los415 | u can upgrade the os to have sip |
| 00:22.28 | Nethab | yeah i was online apr 1st as the capacity was increasing before my etes |
| 00:22.36 | los415 | but it still has some horrible echo with nat |
| 00:22.54 | drumkilla | Nethab: that was just some javascript that used your system clock :) |
| 00:23.08 | drumkilla | you could change your system clock to change how much space it said you had :) |
| 00:23.12 | Nethab | yeah but that was the capacity increasing |
| 00:23.26 | Nethab | it just wasn't directly synced to your browser |
| 00:23.27 | niZon | I wanna get a 7960... |
| 00:23.36 | Moc | niZon get a polycom instead |
| 00:23.56 | niZon | how much for a used one with the same features as a cisco (including xml)? |
| 00:23.59 | gambolputty | I want the Snom 360 :) |
| 00:24.06 | drumkilla | that the new one? |
| 00:24.15 | Nethab | i don't see the new one on the polycom site |
| 00:24.18 | Nethab | only 300 500 600 |
| 00:25.15 | Nethab | and polycom isn't a bastard about including the AC adapter |
| 00:26.24 | *** join/#asterisk bjohnson (~bjohnson@66.11.165.59) |
| 00:27.44 | file | yay blanket |
| 00:28.27 | Nethab | you found my blankie? |
| 00:28.34 | drumkilla | he stole mine |
| 00:29.16 | file | that better? |
| 00:29.53 | niZon | Nethab: 48v adaptors aren't too hard to find |
| 00:30.21 | Nethab | but polycom includes it by default while all the cisco phones i see don't have one |
| 00:30.49 | Nethab | cause cisco wants you to buy a PoE switch |
| 00:31.29 | los415 | do the polycom's do poe? |
| 00:32.19 | zip | check out the netgear fs108 |
| 00:32.25 | zip | google, it is nice, and cheap :) |
| 00:32.54 | Nethab | there's a cat5 cable with a ac recepticle on it |
| 00:32.56 | Nugget | I have a bunch of netgear fs and gs switches here at home, they're great |
| 00:33.03 | Nethab | that has the chip for PoE |
| 00:39.33 | Nethab | is it particularly hard to switch from MGCP to SIP on a polycom IP500 |
| 00:44.28 | *** join/#asterisk meppl (mephisto@pD9E68F34.dip.t-dialin.net) |
| 00:45.14 | *** join/#asterisk peter222 (peter222@dsl-202-173-142-98.sa.westnet.com.au) |
| 00:45.20 | peter222 | hi everyone |
| 00:45.31 | peter222 | what is latest rev of TDM400 ? |
| 00:46.47 | blitzrage | H I think |
| 00:47.12 | Qwell | I'd look at mine if my cabling wasn't such a mess... |
| 00:48.16 | peter222 | with rev H it fixes all issues and i don't have to rmmod + insmod ? |
| 00:49.01 | Nethab | H isn't a magic bullet |
| 00:51.59 | tzanger | actually I have heard that rev I is out soon, but no major change over H |
| 00:52.20 | Nethab | There's no I in team, but there is a me |
| 00:52.24 | tzanger | ha |
| 00:52.36 | tzanger | I wish digium had a changelog for their cards :-) |
| 00:52.40 | Qwell | What gets modified, the TDM, or the modules? |
| 00:53.01 | tzanger | both |
| 00:53.22 | tzanger | I have a rev F and H card and rev C FXO. FXS don't seem to have a mod on them |
| 00:53.26 | Nethab | nugget! |
| 00:55.43 | Nethab | is there a bug or feature request to unify all the manager portions of the channel modules |
| 00:56.21 | kb1_kanobe | 'chan_zap.c:4393 my_zt_write: Write returned -1 (Resource temporarily unavailable) on channel 23' - indication of an interrupt servicing problem? Card is a T110p |
| 01:01.39 | *** join/#asterisk meppl (mephisto@pD9E68A93.dip.t-dialin.net) |
| 01:04.20 | *** join/#asterisk Defraz (~t0tal@sonicwall.dcdi.net) |
| 01:04.39 | *** join/#asterisk jaxxan (~jaxxan@202.70.125.109) |
| 01:04.44 | jaxxan | hey ya'll |
| 01:04.45 | peter222 | what problems are people having with H ? |
| 01:05.03 | jaxxan | anyone ever have a problem with a 7960 rebooting itself in no certain timeframe or circumstances ? |
| 01:05.10 | jaxxan | with firmware 7.1 |
| 01:05.26 | Qwell | jaxxan: bad power? don't know if that would cause it |
| 01:05.34 | Shido | bad power would do that |
| 01:05.39 | nestAr | jaxxan: i had a IP300 do that |
| 01:05.43 | jaxxan | it's connected to a 3750 POE-48 Switchstack |
| 01:05.44 | nestAr | was bad power |
| 01:05.52 | peter222 | jaxxan: what cube are you using ? |
| 01:05.55 | Shido | bad power connector to the phone |
| 01:06.00 | Shido | bad powerbrick |
| 01:06.15 | jaxxan | i have a poe switch, no power brick |
| 01:06.47 | Qwell | jaxxan: bad poe connection |
| 01:06.48 | peter222 | can u try a spare brick ? |
| 01:07.07 | peter222 | how often ? daily reboot ? weekly ? |
| 01:07.10 | Shido | $15? |
| 01:07.17 | Shido | no wait i think they are $25 |
| 01:07.21 | Shido | lemme look |
| 01:07.24 | Qwell | Shido: in that range |
| 01:07.25 | jaxxan | no |
| 01:07.31 | jaxxan | just today |
| 01:07.47 | jaxxan | phones prolly been running for approximately... oh... since december |
| 01:07.48 | Nugget | jaxxan: one of the used 7960s I bought off ebay locks up after a few days of uptime. I just scripted it to reboot every night and it's been problem-free ever since |
| 01:07.50 | Qwell | jaxxan: does another 7960 work on the same ports, etc? |
| 01:08.06 | jaxxan | i have 44 phones on that switch |
| 01:08.13 | Qwell | same port then.. |
| 01:08.24 | Shido | 27.00 |
| 01:08.31 | Shido | 27 bucks for a new powerbrick ( cisco ) |
| 01:09.00 | jaxxan | what's the latest firmware for the 7960 ? |
| 01:09.04 | jaxxan | i'm running 7.1 right now |
| 01:09.19 | *** join/#asterisk file (~file@mctn1-3974.nb.aliant.net) |
| 01:09.21 | Nethab | 7.4 |
| 01:09.29 | jaxxan | ok |
| 01:09.32 | jaxxan | i'll just upgrade firmware |
| 01:10.15 | jaxxan | maybe it's just a software bug that has already been fixed or something |
| 01:10.35 | jaxxan | i'm kinda just thinking the phone encountered an internal error and rebooted? |
| 01:10.38 | *** join/#asterisk chris_d (~chris@66.88.142.66.ptr.us.xo.net) |
| 01:10.44 | Qwell | jaxxan: it only happened once? |
| 01:11.00 | jaxxan | qwell, yeah |
| 01:11.03 | chris_d | Any Broadvoice users who can help me with my configuration? |
| 01:11.08 | Qwell | You made it sound like it was happening repeatedly |
| 01:11.10 | Nethab | i use broadvoice |
| 01:11.14 | jaxxan | no |
| 01:11.38 | chris_d | Nethab: What do I have to do if I'm behind a firewall? |
| 01:11.46 | Qwell | the rule is, if it happens once, you say "hmm, that was odd. oh well" |
| 01:12.01 | jaxxan | maybe it happened once or twice on other phones since december. i vaguely remember something about it from someone else. just wanted to see if anyone else experienced it |
| 01:12.10 | Nethab | usually nothing, when you register to them, it picks up your ip information |
| 01:12.43 | chris_d | Nethab: OK. Is the current setup information on the Broadvoice page correct? |
| 01:13.01 | chris_d | Nethab: Because I followed those directions and things aren't quite working. |
| 01:13.03 | Nethab | i don't use it, |
| 01:13.13 | Nethab | i use a simplified version |
| 01:13.20 | chris_d | Nethab: For example, neither inbound nor outbound calling is working. |
| 01:13.21 | jaxxan | i'm still looking for acd type of metrics |
| 01:13.22 | jaxxan | for asterisk |
| 01:13.27 | chris_d | Nethab: Simplified is good. :-) |
| 01:13.35 | Nugget | plenty of ADD and OCD types in here. |
| 01:13.44 | Nugget | not quite the same thing |
| 01:14.07 | jaxxan | management just wants to see daily call stats and stuff |
| 01:14.18 | jaxxan | i'd kill for some kind of reporting to present to them weekly |
| 01:14.39 | jaxxan | i'm definitely not a database guy |
| 01:15.38 | jaxxan | btw, offtopic, anyone running a 15" powerbook that would like a really kewl wallpaper of jessica alba ? |
| 01:15.38 | *** join/#asterisk xheliox (~jeff@user-0c6se11.cable.mindspring.com) |
| 01:16.48 | Nugget | no thanks, I'm an adult. |
| 01:17.18 | jaxxan | meh? |
| 01:17.52 | syle | an adult? |
| 01:18.01 | Nugget | well, for one thing, I have underwear that's older than her. and secondly, I outgrew putting hot chixxx on my desktop at about age 15. |
| 01:18.22 | syle | so whats that make you 18? |
| 01:18.32 | Nugget | we went over this already, syle. |
| 01:18.48 | syle | age? |
| 01:18.51 | Nugget | yes |
| 01:19.10 | syle | 18 it is |
| 01:19.23 | *** join/#asterisk menger (~menger@static-88.243.240.220.dsl.comindico.com.au) |
| 01:19.30 | Nugget | the other night you explained to me that you were right in an argument because you were obviously older than me. |
| 01:19.35 | Nugget | you weren't right, and you aren't older than me. |
| 01:19.41 | xheliox | Greetings everyone. I just ordered an FXO a few days ago, got it installed today, configured asterisk and achieved my goal with more than enough time to spare. It's really an amazing product, I'm sure I've not even scratched the surface yet. :) |
| 01:19.42 | Nugget | we can have that argument again if you want. |
| 01:20.12 | syle | well i question you again because i can;t even remember what the hell i did anymore when i was 15 |
| 01:20.45 | Nugget | I guess you just did less interesting things than I did at 15. |
| 01:21.14 | syle | not sure, wasn;t really any fun till high school if i remember right |
| 01:21.32 | syle | drivers license, getting laid at parties all the time etc |
| 01:21.35 | syle | those were the days |
| 01:22.18 | syle | actually i think i got laid more in university |
| 01:22.37 | syle | hell i lost count who knows |
| 01:22.40 | syle | was fun though |
| 01:22.48 | Nugget | heh |
| 01:23.12 | remmo | foo |
| 01:23.25 | Nugget | bar |
| 01:23.28 | syle | i still remember getting stoned with school buddies and hacking each other with the new slackware 3.0 lol |
| 01:23.38 | *** part/#asterisk CoderCR (~creyna@ip68-8-131-103.sd.sd.cox.net) |
| 01:23.59 | Nugget | From: syle |
| 01:24.02 | Nugget | To : syle |
| 01:24.05 | Nugget | Subj: drnuk |
| 01:24.15 | Nugget | I sent myself more than my fair share of those |
| 01:24.52 | syle | your not really stoned on a unix system till your sitting there doing multiple ls commands then wonder what the hell your doing |
| 01:26.55 | syle | actually guy i went to school with went to math class stoned every single day, he actually got better grades doing that lol |
| 01:27.05 | Nugget | state-dependent learning. |
| 01:27.15 | Nugget | you have to take the tests in the same condition you learned the material |
| 01:27.19 | *** join/#asterisk FuriousGeorge (~brian@ool-43516ebb.dyn.optonline.net) |
| 01:27.22 | syle | he codes c on kernels and cross compilers or some shit now, still stoned as usual |
| 01:27.25 | jaxxan | i went to 1st period history class stoned and slept through it for the entire school year. aced every exam |
| 01:27.31 | *** part/#asterisk xheliox (~jeff@user-0c6se11.cable.mindspring.com) |
| 01:27.40 | jaxxan | those were the days |
| 01:27.49 | mmlj4 | our high-school physics teacher was stoned every day |
| 01:27.59 | Qwell | mmlj4: its a requirement |
| 01:28.07 | mmlj4 | he used to have to grip the lab table to keep from falling over |
| 01:28.09 | tzanger | our high school accounting teacher was proof of why you didn't do cocaine |
| 01:28.17 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
| 01:28.17 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 01:28.49 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 01:29.17 | syle | well if you think how pathetic a teachers life really is, low wage, take rest of their life to pay off their student loans, you;d probably resort to some drug to hehe |
| 01:29.49 | tzanger | ha |
| 01:30.30 | syle | i bet in the back of all their minds everyday they think, fuck why didn't i get a PHD |
| 01:30.57 | Nethab | had a biology teacher drunk every day i was at that school, electricuted himself and kept teaching class afterward |
| 01:31.18 | Nethab | didn't skip a beat, well maybe only one or two |
| 01:32.01 | syle | well thats definately a different high lol |
| 01:32.12 | Qwell | or low |
| 01:32.15 | syle | lets see how many volts this body can take today haha |
| 01:32.24 | jaxxan | <PROTECTED> |
| 01:32.41 | jaxxan | <PROTECTED> |
| 01:32.47 | jaxxan | from cisco |
| 01:32.57 | Shido | my phone crashed once |
| 01:33.03 | jaxxan | that could be it |
| 01:33.03 | Shido | but only once |
| 01:33.13 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 01:33.14 | *** join/#asterisk furiousgeorge (~brian@ool-43516ebb.dyn.optonline.net) |
| 01:33.27 | syle | how often does it update, can;t you control that? |
| 01:33.37 | Nethab | do you use an ntp time server |
| 01:33.43 | jaxxan | yes |
| 01:33.44 | Shido | I do |
| 01:33.48 | Shido | 7-4 |
| 01:33.48 | Shido | tho |
| 01:33.52 | Shido | I dont see my date and time |
| 01:33.56 | Qwell | 7-4? |
| 01:34.06 | Nethab | firmware 7.4 i presu,e |
| 01:34.09 | Qwell | oh |
| 01:34.09 | jaxxan | i changed ntp time server and dont see the time anymore |
| 01:34.45 | syle | have you tested time server locally? |
| 01:34.45 | jaxxan | say's it can intermittently crash during tftp requests |
| 01:34.52 | jaxxan | yes |
| 01:35.07 | jaxxan | time.apple.com |
| 01:35.14 | jaxxan | use it every day on my mac |
| 01:35.23 | syle | i don;t know about crash, usually after they tftp update they reboot the IOS |
| 01:35.38 | jaxxan | our internal ntp server on one of our novell platforms freaking dies like once a year, so i figured i'd just move to something more reliable |
| 01:35.43 | *** join/#asterisk tld (~tld@80.203.70.227) |
| 01:35.54 | jaxxan | now the time doesn't show up at all |
| 01:35.58 | jaxxan | it's weird |
| 01:36.05 | syle | just use ntp server on linux |
| 01:36.10 | syle | never had a problem |
| 01:37.27 | syle | novell , damn there is a word i hoped i;d never hear again |
| 01:37.40 | Qwell | syle: hmm, thanks |
| 01:37.54 | Qwell | syle: Somehow you just reminded me of something I was thinking earlier. Something entirely unrelated to novell :p |
| 01:38.15 | bkw_ | jaxxan, a mac useer |
| 01:38.16 | bkw_ | w00t |
| 01:38.40 | jaxxan | you gotta powermac bkw_ ? |
| 01:38.40 | Qwell | bkw_: I'd get one, if I weren't so damn poor :p |
| 01:38.50 | jaxxan | erm.. powerbook (= |
| 01:38.52 | syle | why are you poor? |
| 01:39.03 | Qwell | syle: IT sucks |
| 01:39.10 | furiousgeorge | i think im having udev problems. i got the modules loaded, and i have zaptel.conf set up correctly yet uhave nothing in /dev/zap/ and so naturally ztcfg complains |
| 01:39.21 | furiousgeorge | i have a tdm400p |
| 01:39.22 | jaxxan | stop spending your money on weed nugget |
| 01:39.22 | Qwell | furiousgeorge: README.udev |
| 01:39.25 | jaxxan | erm... qwell lol |
| 01:39.51 | furiousgeorge | Qwell: now that i see you type that, i remember having this problem last time, and you telling me to go do that |
| 01:39.52 | Qwell | weed nugget...interesting typo |
| 01:40.00 | Qwell | furiousgeorge: yeah... |
| 01:40.12 | furiousgeorge | i guess i gotta stop pending my money on weed nugget |
| 01:40.17 | Qwell | furiousgeorge: :P |
| 01:40.19 | syle | well depends on what your doing with IT i guess, unix admin is ok, but i;d probably kill myself if i had to deal with people on the phone for tech support |
| 01:40.23 | jaxxan | lol |
| 01:40.32 | Qwell | syle: Windows programmer. nuff said |
| 01:40.53 | furiousgeorge | syle: i dont deal with people on the phone, forget i said that |
| 01:40.57 | Nugget | heh |
| 01:41.34 | bkw_ | jaxxan, I have an ibook and an imac |
| 01:41.58 | *** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
| 01:42.18 | jaxxan | check that out |
| 01:42.20 | syle | i suck at customer support, i just don;t have the patience, i remember once i worked at a video store, that last 3 days after i told the boss;s friend he couldn;t have a membership and to fuckoff |
| 01:42.27 | iCEBrkr | ok.. I've went over my iax.conf and extensions.conf ( and all this worked once before ) but why the heck would I be getting rejection messages from inbound voicepulse calls? |
| 01:43.29 | Voicelynx | iCEBrKr: Some other were complaining about VoicePulse inbound earlier today |
| 01:43.33 | Silik0n | imacs suck |
| 01:43.38 | *** join/#asterisk drraid (~drraid@c-24-22-3-41.hsd1.or.comcast.net) |
| 01:43.46 | jaxxan | the new imacs are pretty slick actually |
| 01:43.59 | Qwell | LI looks like U. Coincidence? I think not |
| 01:43.59 | iCEBrkr | Voicelynx: I'm wondering if they prepended a 1 to your number.. |
| 01:44.02 | iCEBrkr | I'm gonna try |
| 01:44.19 | drraid | Is it possible to impliment asterisk with 2 modems on a computer, to do basic dial-pin authentication and call forwarding? |
| 01:44.20 | Silik0n | jaxxan not when you gotta G5 |
| 01:44.20 | Qwell | re: slick vs suck |
| 01:44.31 | jaxxan | i used to do isp tech support and got fired after i told a user they were to stupid to use the internet |
| 01:44.35 | syle | no shit voicelynx? , hmm aren;t they reselling level3.com's network ? |
| 01:44.39 | Silik0n | drraid yes thats possible |
| 01:45.04 | drraid | Silik0n, without other equipment ( i mean obviously phone lines.. but no other pbx type stuff) |
| 01:45.16 | Voicelynx | style: Don't think so |
| 01:45.19 | Qwell | drraid: something like some FXOs, to connect to the phonelines |
| 01:45.21 | Silik0n | drraid you need the correct "modems" but yes its possible |
| 01:45.29 | drraid | Please pardon my lack of knowledge on the subject, i havent played with telephony much |
| 01:45.34 | Qwell | ~fxo |
| 01:45.37 | jbot | foreign exchange office - type of port you need to connect a POTS (Plain Old Telephone Service) line from your telco to a pbx http://www.digium.com/index.php?menu=fxsvfxo |
| 01:45.37 | Silik0n | you need to gind a FXO board cloan |
| 01:45.37 | Voicelynx | Style: They would have more area codes if they were |
| 01:45.52 | syle | maybe they are going direct with qwest/global |
| 01:46.01 | drraid | Silik0n, do you know any good documentation on those type of modems? or what type they would be? |
| 01:46.02 | syle | level3.com resells their shit |
| 01:46.09 | iCEBrkr | blargh |
| 01:46.15 | Qwell | drraid: x100p, but the clones aren't great... |
| 01:46.20 | Voicelynx | livevoip use Level 3 DIDs |
| 01:46.22 | Silik0n | Voicelynx: not nessecarily true... L3 charges you a setup fee per rate center in NA to set up DIDs in them... |
| 01:46.28 | drraid | Qwell: clones as in generics? |
| 01:46.37 | Qwell | drraid: no, its still an x100p |
| 01:46.40 | Silik0n | drraid: voip-info.org and search for FXO or x101p |
| 01:46.41 | Qwell | its just...a clone |
| 01:46.51 | Silik0n | drraid or look for x100p or x101p on ebay |
| 01:47.05 | drraid | Fairly pricy equipment? |
| 01:47.18 | Silik0n | the clones usually suck for hangup detection or you can just get a tdm400 from digium |
| 01:47.27 | drraid | Oh not at all |
| 01:47.28 | Voicelynx | Everything is negotiable with Level 3 as long as you give them lots of $$$$s / mionth ;-) |
| 01:47.29 | Silik0n | drraid: depends on what you get |
| 01:47.29 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 01:47.56 | Silik0n | the clone cards are sub $20 each the TDM400 is ~100 to 300 depending on its configuration |
| 01:48.04 | Silik0n | Voicelynx I know |
| 01:48.08 | Qwell | the TDM is well worth it though |
| 01:48.31 | drraid | I really appreciate the help. Ill read up more on it. Thanks again |
| 01:48.32 | iCEBrkr | I read that you typically get a 'reject' message when you don't have an extension setup or a way to handle the incoming call. |
| 01:48.42 | Qwell | tdm400p |
| 01:48.44 | Qwell | ^^ more googlable |
| 01:48.45 | iCEBrkr | I'm almost thinking it has something to do with Voicepulse.. cuz I haven't really changed anything. |
| 01:48.55 | iCEBrkr | haha googlable. |
| 01:48.57 | drraid | tdm400p? |
| 01:49.01 | Qwell | iCEBrkr: What would you say? |
| 01:49.04 | Silik0n | ~tdm400p |
| 01:49.06 | Qwell | drraid: Thats Digium hardware |
| 01:49.10 | drraid | Oh |
| 01:49.13 | iCEBrkr | Qwell: The same thing :{ |
| 01:49.15 | iCEBrkr | err :P |
| 01:49.18 | Qwell | iCEBrkr: Thought so. ;] |
| 01:49.18 | Silik0n | just go to www.digium.com and look at it |
| 01:49.39 | Qwell | iCEBrkr: I tend to make up words alot... |
| 01:49.41 | furiousgeorge | Qwell: after i have set up rules and perms, i gotta unload and reload driver or something? |
| 01:49.54 | Qwell | furiousgeorge: probably, dunno |
| 01:51.02 | drraid | you guys are awsome. thx again |
| 01:51.04 | *** part/#asterisk drraid (~drraid@c-24-22-3-41.hsd1.or.comcast.net) |
| 01:51.56 | iCEBrkr | chrome*CLI> iax2 debug |
| 01:51.56 | iCEBrkr | IAX2 Debugging Enabled |
| 01:51.56 | iCEBrkr | Apr 27 21:51:43 NOTICE[22225]: chan_iax2.c:5405 socket_read: Rejected connect attempt from 66.234.228.170 |
| 01:52.02 | iCEBrkr | yay. that's informative :| |
| 01:52.07 | furiousgeorge | Qwell: according to voip info i actually gotta reboot |
| 01:52.12 | *** part/#asterisk furiousgeorge (~brian@ool-43516ebb.dyn.optonline.net) |
| 01:52.28 | Qwell | or just run the udev scripts... |
| 01:54.02 | *** join/#asterisk sandnigg0r (~niggerplz@66-55-197-254.gwi.net) |
| 01:55.45 | iCEBrkr | <PROTECTED> |
| 01:55.49 | iCEBrkr | Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REJECT |
| 01:55.49 | iCEBrkr | there we go.. |
| 01:55.52 | iCEBrkr | <PROTECTED> |
| 01:55.54 | iCEBrkr | <PROTECTED> |
| 01:55.55 | iCEBrkr | more info, but I don't quite understand it. |
| 01:57.12 | kb1_kanobe | 'chan_zap.c:4393 my_zt_write: Write returned -1 (Resource temporarily unavailable) on channel 23' - indication of an interrupt servicing problem? Card is a T110p |
| 01:58.22 | kb1_kanobe | iCEBrkr: 66.234.228.170 tried to dial into chrome over IAX and it was not permitted by crome (ie. see iax.conf on both sides - perhaps they need to be 'friends'?) |
| 01:59.03 | iCEBrkr | I'll check again. |
| 01:59.15 | iCEBrkr | But I had this working months ago.. I just noticed it wasn't working last night |
| 01:59.33 | iCEBrkr | [vpconnect-in] |
| 01:59.33 | iCEBrkr | type=friend |
| 02:00.02 | syle | what are DID rates |
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| 02:00.39 | syle | is that like what your charged for every initial call |
| 02:00.48 | Qwell | syle: per minute, usually |
| 02:01.10 | FuriousGeorge | Qwell: i think im in big trouble, i took care of udev, but i think i have a much bigger problem. seg fault |
| 02:01.23 | syle | eels |
| 02:01.27 | Qwell | FuriousGeorge: Did you just upgrade/downgrad to/from cvs? |
| 02:01.30 | syle | ohhh |
| 02:01.38 | syle | thats termination to TDM line |
| 02:01.49 | FuriousGeorge | Qwell: well from portage. |
| 02:02.00 | syle | 1.1 cents a minute |
| 02:02.01 | syle | eeks |
| 02:02.06 | FuriousGeorge | Qwell: maybe it makes the wrong modules by default and i gotta make linux26? |
| 02:02.06 | syle | how the hell do you make money |
| 02:02.13 | Qwell | syle: at 1.1c, you don't |
| 02:02.16 | FuriousGeorge | that doesnt make sense |
| 02:02.18 | Qwell | FuriousGeorge: no |
| 02:02.25 | Qwell | FuriousGeorge: tip: Don't use packages for * |
| 02:02.56 | FuriousGeorge | sigh, worked on this same box before i reinstalled |
| 02:03.34 | FuriousGeorge | Qwell: so i should not try to remake the same modules using make linux26, i should get the source from cvs? |
| 02:03.51 | Qwell | FuriousGeorge: or get stable |
| 02:03.59 | Qwell | but, remove all the old modules first |
| 02:04.08 | syle | omg all these sites are doing 1.1 cents a minute |
| 02:04.54 | Voicelynx | Does anybody know if there is a simple way of pushing all extensions to (say) lower case so that case sensitivity on inbound SIP calls is not an issue? If the problem is solved already I don't want to reinvent the wheel! |
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| 02:05.40 | FuriousGeorge | Qwell: i have version 1.07 from asterisk, u still think its a non-stable release? |
| 02:05.54 | Qwell | FuriousGeorge: no, packages just suck |
| 02:06.10 | FuriousGeorge | packages=ebuilds? |
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| 02:09.49 | key2 | hi |
| 02:09.51 | key2 | is anyone here? |
| 02:10.01 | opus_ | hui |
| 02:10.08 | kb1_kanobe | not a good question to start with.... |
| 02:10.19 | key2 | :) |
| 02:10.22 | key2 | sorry |
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| 02:10.39 | key2 | I just had a question, how can I call from a sip phone with my FXO card ? |
| 02:10.56 | key2 | like VoIP > telephone line |
| 02:11.40 | Qwell | key2: by setting up a proper dialplan |
| 02:11.41 | Qwell | ~docs |
| 02:11.42 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 02:12.29 | tainted- | weee |
| 02:12.29 | Sedorox | hola crap |
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| 02:12.51 | key2 | Qwell: what you mean ? |
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| 02:13.38 | key2 | it's basically gonna be a gateway between IP and telephone line ? |
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| 02:20.53 | niZon | wee |
| 02:20.54 | iq | lol |
| 02:20.54 | syle | asterisk support H323? |
| 02:21.21 | Sedorox | and I wonder why freenode sucks..... |
| 02:21.21 | iq | FASTEN YOUR SEAT BELTS |
| 02:21.38 | iq | Sedorox, it does not |
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| 02:22.25 | hassler | HI !!! Just getting started with asterisk. Using the TDM400P with FXO / FXS. I *think* I have everything set up (nothing customized), but I'm not getting ANYTHING on the phone line. |
| 02:22.54 | hassler | trying to figure out debugging... any suggestions? |
| 02:23.19 | iq | hassler, try asterisk -vvvvvvvvvvvvvvvvvvvvvvvvvvvvvgc |
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| 02:24.45 | hassler | iq: yea, did that. Not seeing anything. |
| 02:24.51 | hassler | (other than the startup junk) |
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| 02:25.40 | hassler | should "show channels" show me the card? |
| 02:26.14 | kb1_kanobe | hassler: try 'zap show channels' |
| 02:26.29 | kb1_kanobe | regular 'show channels' only shows the active ones |
| 02:26.42 | hassler | says no such command. Huh? |
| 02:27.06 | kb1_kanobe | try 'help' |
| 02:27.46 | hassler | yea, help shows no "zap" commands. OK, now I'm really confused. |
| 02:28.17 | kb1_kanobe | 'show modules' - is chan_zap.so listed? |
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| 02:28.50 | hassler | no, its not. I'm missing something apparently? |
| 02:29.51 | kb1_kanobe | is it present in /usr/lib/asterisk/modules? |
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| 02:32.39 | hassler | no, chan_zap.so is NOT there...... searching for it now. probably stupid mistake |
| 02:32.50 | tzanger | hassler: install zapata BEFORE compiling asterisk |
| 02:33.22 | kb1_kanobe | tzanger: back again already so soon? |
| 02:33.37 | tzanger | dishes are done, just looking at kde-look.org so I have a new theme for tomorrow |
| 02:33.46 | kb1_kanobe | lol |
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| 02:38.18 | hassler | re-emerging zaptel now (running gentoo installation) |
| 02:38.34 | JerJer | tzanger: will you come over and be by dish bitch? |
| 02:38.41 | JerJer | :D |
| 02:39.40 | tzanger | JerJer: depends, how good are your blowjobs |
| 02:39.50 | Nugget | heh |
| 02:40.00 | Nugget | someone drive over to cupertino and make them ship my mac. |
| 02:40.25 | mishehu | nah |
| 02:40.39 | mishehu | mac users always tell people to buy a mac when something goes wrong or doesn't work |
| 02:40.55 | mishehu | "my toaster burnt my toast" --> "get a mac!" |
| 02:40.59 | mishehu | or in this case: |
| 02:41.02 | Nugget | it's funny because it's true. |
| 02:41.10 | Qwell | "my mac doesn't work" --> "get a mac!" |
| 02:41.16 | pjz | linux people tell people to install linux in the same circumstances :) |
| 02:41.19 | Qwell | Don't deny it. :p |
| 02:41.21 | mishehu | "they aren't shipping my computer from cupertino." --> "get a mac!" |
| 02:41.40 | mishehu | pjz: I don't. |
| 02:41.47 | mishehu | linux is not the solution to all problems. |
| 02:41.49 | pjz | "I spent my last $ on a mac so now I can't eat." --> "get another mac!" |
| 02:42.02 | bjohnson | "I'm hungry" --> "Get a (Big) Mac" |
| 02:42.07 | Nugget | at least when mac people proselytize it's not followed up with "I can totally come over and help you set it up. shouldn't take more than a few days" like with the linux guys. :) |
| 02:42.29 | bjohnson | Nugget: they're just hoping to get lucky with you |
| 02:42.33 | mishehu | Nugget: I don't do that either. I do a "here's a knoppix cd..." |
| 02:42.42 | mishehu | unless it's a hot girl |
| 02:42.46 | Nugget | "...call me when you can't get the modem to work" :) |
| 02:42.46 | mishehu | then I offer to move in. |
| 02:43.20 | Nugget | although I think we all make the "hot girl" exception. well, except for bkw. |
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| 02:44.16 | mishehu | somebody must have already moved in with katty |
| 02:44.22 | mishehu | you snooze, you lose |
| 02:44.54 | hassler | <= guess you need "zaptel" use flag on asterisk.... |
| 02:45.13 | bjohnson | well .. she said she wanted to learn linux and asterisk from scratch .. and then wanted people to walk her through every step |
| 02:45.48 | *** join/#asterisk PTG1234 (PTG123@ip68-106-24-139.ph.ph.cox.net) |
| 02:46.13 | mishehu | bjohnson: maybe she was looking for one of us to move in with her... |
| 02:46.20 | mishehu | and not just the other way around |
| 02:46.40 | mishehu | like "I want to bag me a geek, maybe I should go on that asterisk channel I heard about..." |
| 02:47.10 | bjohnson | I doubt it .. she was pretty stand offish |
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| 02:47.52 | tengulre | hi,all |
| 02:48.04 | shido6 | FUCK WINDOWS!!!!! |
| 02:48.17 | bjohnson | what an exit |
| 02:48.46 | hassler | <= runs a linux laptop :-) |
| 02:49.29 | syle | <- freebsd |
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| 02:49.42 | Nugget | linux is poo |
| 02:49.44 | Nugget | :) |
| 02:50.12 | bjohnson | hassler <= geek |
| 02:50.27 | bjohnson | syle <= elitist geek |
| 02:50.47 | bjohnson | Nugget <= jealous |
| 02:50.50 | Nugget | on a latop, os x is better. on a server, freebsd is better. for networking openbsd is better. for a desktop, xp is better. why would I want linux? |
| 02:50.50 | hassler | <== definitely not a geek. corporate VP. Been doing this since 83.... |
| 02:51.07 | fugitivo | Nugget: you' |
| 02:51.09 | iCEBrkr | NOTICE: Wed Apr 27 10:00AM EST -- We are aware of an issue that is currently affecting incoming calls and outgoing audio for some users. We are working with our provider for incoming numbers in that area to resolve this issue as quickly as possible. |
| 02:51.11 | fugitivo | Nugget: you're nuts |
| 02:51.32 | iCEBrkr | Hrrrm, Voicepulse mentions audio on inbound calls.. I wonder if this has something to do with inbound calls in general as well?? |
| 02:51.36 | Nugget | my desktop os is just a bootloader for a web browser, ssh, and games. |
| 02:51.40 | fugitivo | Nugget: linux is good for laptop, desktop and server |
| 02:51.50 | iCEBrkr | Nugget: haha right on! |
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| 02:52.07 | onlyI | fugitivo space station also |
| 02:52.14 | Nugget | fugitivo: why settle for good? I opt for "better" |
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| 02:53.11 | fugitivo | Nugget: start to explain why an os is better than another and maybe I agree with you :) |
| 02:53.39 | drbrown | I don't know how anyone can tollerate an xp desktop |
| 02:53.48 | fugitivo | drbrown: me neither |
| 02:53.57 | iCEBrkr | XP is what 95 should have been :P |
| 02:53.58 | drbrown | freebsd is pretty good |
| 02:54.08 | drbrown | exactly |
| 02:54.37 | iCEBrkr | The thing I hate about XP's interface is the Duplo-Icons.. |
| 02:54.37 | Nugget | XP isn't too bad. I'm not a fan of explorer, but the kernel is pretty solid and I'm jealous of ntfs when I'm on crufty old unix UGO systems |
| 02:54.41 | fugitivo | windows is not in my list of "useable" os, i don't feel comfortable with it |
| 02:54.42 | iCEBrkr | It's like for a 3yr old or something |
| 02:55.08 | Nugget | explorer is clunky and difficult to configure, though. that's obvious to anyone who has used windows for more than a week |
| 02:55.15 | iCEBrkr | fugitivo: it takes some customizing to make it tolorable. |
| 02:55.20 | tessier | Nugget: Apparently not. |
| 02:55.26 | Nugget | ok, fair point. :) |
| 02:55.28 | tessier | Nugget: As most people seem quite happy with it. |
| 02:55.35 | fugitivo | iCEBrkr: i prefer to customize my linux |
| 02:55.38 | iCEBrkr | fugitivo: but even so, Windows is just a front-end for launching games :) |
| 02:55.51 | iCEBrkr | Wintendo(tm) |
| 02:55.59 | Nugget | I just don't expect much from an XP box. It needs to be able to launch putty and halflife. |
| 02:56.10 | iCEBrkr | Nugget: haha that's great |
| 02:56.21 | drbrown | I expect to fix an xp box |
| 02:56.21 | iCEBrkr | You forgot about firefox. |
| 02:56.29 | fugitivo | iCEBrkr: nwn runs on linux, i don't need windows, thanks god |
| 02:56.31 | Nugget | yeah, firefox too. |
| 02:56.46 | iCEBrkr | fugitivo: yea, how's the performance? |
| 02:56.53 | drbrown | ie is a sign of the apocolyse |
| 02:56.55 | *** join/#asterisk overflow (~overflow@200.84.174.58) |
| 02:56.58 | iCEBrkr | fugitivo: I used to be pretty hardcore into NWN scripting.. |
| 02:56.59 | Nugget | I'm migrating to a mac desktop so I can get one of those ginormous monitors, though. |
| 02:57.04 | fugitivo | iCEBrkr: great with nvidia, a bit slow with ati |
| 02:57.05 | Nugget | I'll miss the gaming, though. |
| 02:57.23 | at561 | what ginormous monitor? |
| 02:57.28 | Nugget | the apple cinema display |
| 02:57.42 | iCEBrkr | Apple will be the first witha 360' display. |
| 02:57.42 | at561 | is that like a projector tv on top of your desk? |
| 02:58.02 | Nugget | the only wintel cards that can push one are workstation cards that cost a grand and suck for gaming, so it's not an attractive solution for an xp box |
| 02:58.05 | *** join/#asterisk TheEmperor (~user@203.114.48.47) |
| 02:58.15 | overflow | Hi. asterisk couldn't use my modem. It's a Motorola Voicesurf external .. any Idea? what can I do? |
| 02:58.25 | *** join/#asterisk FuriousGeorge (~brian@ool-43516ebb.dyn.optonline.net) |
| 02:58.29 | drbrown | buy digium |
| 02:58.31 | Nugget | although I understand that the upcoming x850 will be able to do it, which is cool |
| 02:58.40 | iCEBrkr | overflow: pitch the modem and find something with an intel chipset on it. ;) |
| 02:59.00 | iCEBrkr | Or yea.. Buy hardware from digium. |
| 02:59.10 | drbrown | buy the $100 digium card |
| 02:59.11 | fugitivo | overflow: hmmm, you can buy one that works on ebay for $10 |
| 02:59.25 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 02:59.31 | FuriousGeorge | Qwell: good news, i utarred source from the web and compiled it up and it worked, disfortunately, it didnt autmatically assign an irq, since i didnt build in acpi, but that is a good thing, i want to control that |
| 02:59.31 | overflow | Im poor and I don't live in Usa :( |
| 02:59.39 | drbrown | digium's support is very good |
| 02:59.44 | fugitivo | overflow: or Digium, if you want more quality |
| 02:59.56 | fugitivo | overflow: where do you live? |
| 02:59.59 | iCEBrkr | I got a cheap-o intel voice modem for $10. Works fine for what I do. |
| 03:00.00 | overflow | fugitivo, Venezuela |
| 03:00.10 | iCEBrkr | Then again, I have Asterisk running as a glorified answering machine. |
| 03:00.32 | at561 | yeah i got one for $5 |
| 03:00.37 | fugitivo | overflow: check ebay, is the best solution for low budgets |
| 03:00.41 | at561 | but then i cancelled my phone line :( |
| 03:00.48 | iCEBrkr | At work, I bought one of those TP100 cards... |
| 03:00.51 | at561 | pricewatch.com |
| 03:01.05 | iCEBrkr | Just to do some R&D |
| 03:01.11 | iCEBrkr | *Cough* |
| 03:01.15 | drbrown | I have tdm400p at work, and the tp100 at home |
| 03:01.15 | iCEBrkr | Yeah, that's the ticket. |
| 03:01.23 | overflow | fugitivo, could I build my on asterisk modem driver ? |
| 03:01.36 | iCEBrkr | overflow: if you were sick and twisted. |
| 03:01.53 | fugitivo | overflow: if you know how to do it, you can build your own OS |
| 03:02.07 | overflow | iCEBrkr, why exactly asterisk need a modem driver? |
| 03:02.23 | overflow | even if it's an external modem? |
| 03:02.33 | fugitivo | overflow: i think it's going to be cheaper and easier buying a $10 card from ebay |
| 03:02.33 | iCEBrkr | overflow: well, I'm going to assume you're trying to get Asterisk on POTS? |
| 03:02.51 | overflow | iCEBrkr, POTS? |
| 03:03.04 | iCEBrkr | errr you're telephone line. |
| 03:03.05 | iCEBrkr | err |
| 03:03.06 | iCEBrkr | your |
| 03:03.15 | TheEmperor | does anyone know where I can get the line to connect a PRI to the Digium card? |
| 03:03.16 | overflow | iCEBrkr, correct |
| 03:03.31 | iCEBrkr | sip show |
| 03:03.32 | iCEBrkr | err |
| 03:04.02 | at561 | i need to break into my school's phone closet and forward an extension to home :/ |
| 03:04.08 | JerJer | TheEmperor: the line? |
| 03:04.19 | overflow | iCEBrkr, yes, a SIP |
| 03:04.21 | JerJer | like use a regular cat-5 cable, yo |
| 03:04.30 | FuriousGeorge | does anyone know how to specify whar irq a module is loaded on? |
| 03:04.47 | JerJer | FuriousGeorge: that's a bios thing |
| 03:04.55 | at561 | loonix module? usually modprobe blah irq=999 |
| 03:04.55 | iCEBrkr | I'm going to assume this 'No authority Found' error is a VoicePulse issue. |
| 03:06.10 | FuriousGeorge | JerJer: i thought there was some way to control that from the command line, im not using ACPI -->(automtically assign interrupts) because when i do it makes the tdm share an irq |
| 03:06.17 | overflow | iCEBrkr, why driver can I specify ? (instead autodetect) |
| 03:06.21 | FuriousGeorge | at561: gonna try |
| 03:06.53 | iCEBrkr | overflow: Just save yourself the trouble and spend the $10 on eBay for an Intel based voice modem. |
| 03:07.19 | *** join/#asterisk wolfson (~hehe@bcp-68-187-183-234.man.nc.charter.com) |
| 03:07.58 | overflow | iCEBrkr, my laptop has an integrated Intel based modem |
| 03:09.17 | iCEBrkr | JerJer: damnit.. make nufone's website like connect.voicepulse.com |
| 03:09.52 | hassler | YEA, got it working! :-) Thanks folks! |
| 03:10.57 | fugitivo | overflow: check for x100p on www.ebay.com |
| 03:11.53 | FuriousGeorge | at561: i looked in help and man and tried modprobe irq= and i got this in dmseg |
| 03:12.04 | FuriousGeorge | Apr 27 23:07:02 be wcfxs: Unknown parameter `irq |
| 03:12.08 | TheEmperor | JerJer:yes the line |
| 03:14.16 | at561 | why is there a backtick |
| 03:14.40 | at561 | are you sure you want to specify an irq anyway? |
| 03:14.51 | at561 | i've never used wcfxs so i'm in alien territory |
| 03:15.12 | *** join/#asterisk bah (048830696@ACAC44F7.ipt.aol.com) |
| 03:16.44 | *** join/#asterisk rjburkh (~rjburkh@dialup-4.233.125.77.Dial1.LosAngeles1.Level3.net) |
| 03:22.16 | Luke-Jr | Is it possible to make a SIP call to a Vonage customer? |
| 03:22.44 | *** join/#asterisk techie (gus@asterisk.horizonte.us) |
| 03:23.22 | overflow | fugitivo, iCEBrkr , You'r gonna see, I gonna make this work without buying any new modem |
| 03:23.44 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 03:24.17 | FuriousGeorge | i know there is some way to manually assign irqs, but i cant find it with modprobe. not using acpi |
| 03:24.39 | Nugget | assigning irqs is done in your bios. |
| 03:24.45 | Nugget | it will either be easy or not possible |
| 03:26.24 | FuriousGeorge | gonna try in the bios |
| 03:27.58 | *** join/#asterisk Rick_Hunter (~rhunter@05-052.008.popsite.net) |
| 03:28.25 | bajanman | can anyone tell me if you have to get alsa working to listen to voicemail? |
| 03:33.53 | Nugget | no. |
| 03:34.22 | Nugget | I'd wager that the vast majority of asterisk servers don't even have a sound card installed |
| 03:35.17 | kb1_kanobe | 'chan_zap.c:4393 my_zt_write: Write returned -1 (Resource temporarily unavailable) on channel 23' - Apparently this is EAGAIN, ie. the write would cause a block on a non-blocking descriptor. Any ides why? It's very intermittant |
| 03:36.22 | file | having a sound card can actually cause problems *G* |
| 03:36.27 | Nugget | heh, true |
| 03:37.02 | *** join/#asterisk samueltc (~samuel@levinux.UQAR.UQUEBEC.CA) |
| 03:37.06 | samueltc | hi |
| 03:37.15 | Nugget | Moo |
| 03:37.33 | samueltc | i'm trying to add a zonedata to zaptel |
| 03:37.48 | *** join/#asterisk rtomson-home (~rtomson@ip70-181-140-181.sd.sd.cox.net) |
| 03:38.21 | samueltc | but when I load wcfxs I got a Invalid 'next' pointer and ioctl(ZT_LOADZONE) failed: Invalid argument |
| 03:38.42 | *** join/#asterisk anthm (~anthm@CPE-69-76-83-52.wi.res.rr.com) |
| 03:38.42 | *** mode/#asterisk [+o anthm] by ChanServ |
| 03:42.18 | iCEBrkr | does iaxtel still work? |
| 03:42.44 | at561 | furiousgeorge, if your card is isa for some godawful reason then you can use isapnp to play with irqs. but you can usually do the same in a safer way with your bios |
| 03:46.07 | *** part/#asterisk hassler (~hassler@cpe-65-31-36-179.woh.res.rr.com) |
| 03:52.53 | niZon | anyone have a handytone 488? |
| 03:54.47 | iCEBrkr | ?? |
| 03:56.35 | *** join/#asterisk chaoscon (~ph33r@chaoscon.user) |
| 03:57.31 | niZon | a handytone 488... |
| 03:58.09 | Carp1 | an ATA lol |
| 03:58.42 | *** part/#asterisk Hogie (daniel@alpha.dfwservers.net) |
| 03:58.45 | niZon | http://www.grandstream.com/y-ht488.htm |
| 04:01.21 | Carp1 | anyone have a GXP-2000? |
| 04:01.29 | niZon | they look sweet |
| 04:02.27 | Carp1 | Yeah, I want to know if they're worth it |
| 04:03.19 | niZon | hm |
| 04:03.35 | iCEBrkr | I know what it is.. Ask your question :P |
| 04:12.31 | *** join/#asterisk heison (~heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 04:13.22 | *** join/#asterisk cc (~cc@byte.fedora) |
| 04:14.04 | *** join/#asterisk mud (~mud@CPE000f660c6997-CM0011ae8fd786.cpe.net.cable.rogers.com) |
| 04:16.10 | niZon | iCEBrkr: you have a 488? |
| 04:16.26 | niZon | sveasoft's QoS seems to work well.. |
| 04:17.01 | iCEBrkr | no, but I have a BT100 and a Sipura2k |
| 04:17.03 | niZon | uploading a large file to someone and a voip call sounds flawless |
| 04:17.05 | heison | does anyone know what this means?? |
| 04:17.09 | iCEBrkr | matter of fact I have 3 Sipura2k's |
| 04:17.11 | heison | ZT_CHANCONFIG failed on channel 3: No such device or address (6) |
| 04:17.28 | niZon | i want an SPA-3k |
| 04:17.35 | heison | happens when i run ztcfg -vv |
| 04:17.37 | niZon | or a ht 488 |
| 04:17.40 | iCEBrkr | I thought the 3k was overkill :) |
| 04:17.56 | niZon | well i want something with an fxo interface |
| 04:18.18 | iCEBrkr | I already have that on my asterisk server |
| 04:18.22 | niZon | since i'm running * in a vmware machine right now digium hardware is out of the question |
| 04:18.31 | iCEBrkr | doh |
| 04:18.34 | iCEBrkr | !!! |
| 04:18.43 | niZon | i'll buy another machine one of these days |
| 04:19.00 | *** join/#asterisk overflow (~overflow@200.84.174.58) |
| 04:19.14 | overflow | its working .. (well close to) :P |
| 04:19.28 | niZon | and what would that be? |
| 04:20.05 | overflow | hey, What can I do to make asterisk to recognize when the called party hang? Im working with SIP and a voice modem. |
| 04:20.41 | Qwell | voice modem? x100p? |
| 04:21.03 | overflow | Qwell, US Robotic & Motorola ... |
| 04:23.21 | *** join/#asterisk predictive (~jeff@adsl-4-122-143.cae.bellsouth.net) |
| 04:23.27 | predictive | greetings |
| 04:23.36 | *** join/#asterisk montoya (montoya@200.195.89.166) |
| 04:23.37 | niZon | do you come in peace? |
| 04:23.42 | predictive | the voip.org wiki page for the manager api states "Only one action may be outstanding at a time." |
| 04:23.45 | predictive | is this true? |
| 04:23.51 | predictive | yes, I am pretty peaceful |
| 04:24.21 | niZon | hm try two actions and see what happens! |
| 04:24.26 | predictive | I'm writing an asynchronous event handler that uses the manager api to receive events and dispatch actions |
| 04:24.51 | predictive | this sentence makes it sound like the api can't handle async interaction: i.e. I have to block on every action and wait for a response |
| 04:25.09 | predictive | or am I misinterpreting that sentence |
| 04:25.49 | iCEBrkr | You have to 'filter' responses. |
| 04:26.07 | predictive | I'm not sure what you mean by that |
| 04:26.12 | iCEBrkr | It is asynchronous |
| 04:26.18 | predictive | can I send action a, and then send action b while waiting for a's response? |
| 04:26.24 | iCEBrkr | Sure |
| 04:26.27 | predictive | ok |
| 04:26.28 | kb1_kanobe | 'chan_zap.c:4393 my_zt_write: Write returned -1 (Resource temporarily unavailable) on channel 23' - Apparently this is EAGAIN, ie. the write would cause a block on a non-blocking descriptor. Any ides why? It's very intermittant |
| 04:26.51 | predictive | perhaps someone should reword that, I was pretty worried by this point that I had wasted my time |
| 04:32.08 | *** join/#asterisk heison (~heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 04:32.15 | heison | ~seen JerJer |
| 04:32.24 | jbot | jerjer is currently on #asterisk (2h 9m 5s). Has said a total of 5 messages. Is idling for 1h 27m 37s |
| 04:36.24 | blitzrage | anyone know how to crack a password on an AdTran 750? |
| 04:36.32 | drumkilla | haha |
| 04:36.45 | drumkilla | why exactly do you need to do that |
| 04:37.12 | blitzrage | Jim got an AdTran off of eBay and didn't get a password for it |
| 04:37.30 | drumkilla | ohhhh |
| 04:37.49 | blitzrage | through the serial port I think is what he's trying to do |
| 04:38.40 | shido6 | blitzrage, stdby |
| 04:38.50 | blitzrage | shido6: cool! |
| 04:39.17 | *** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net) |
| 04:43.14 | predictive | iCEBrkr: trying this through telnet, I don't see a reliable way to match responses to actions |
| 04:43.31 | iCEBrkr | You can key off the 'channel id' |
| 04:43.45 | iCEBrkr | SIP/1234-b01a2 |
| 04:43.58 | iCEBrkr | It's a pain in the ass, but it's doable |
| 04:44.01 | predictive | yes, I see that, I am not sure it will cover all the possiblites |
| 04:44.12 | predictive | you'd think it'd be easier to do dynamic dialplans |
| 04:44.41 | iCEBrkr | Not exactly sure what you're trying to do, but I did a proof of concept for a call manager. |
| 04:44.50 | iCEBrkr | It wasn't easy, but I got it to work |
| 04:45.07 | predictive | it's pretty simple, I take dtmf entries, use them to look up data somewhere like a db |
| 04:45.15 | predictive | then return a response based on the data |
| 04:45.39 | iCEBrkr | I did something like that, but not via the Manager port. |
| 04:45.46 | iCEBrkr | I did it via AGI |
| 04:46.02 | predictive | yeah but I need persistence |
| 04:46.05 | iCEBrkr | Of course the DB had the translated DTMF in a column and I did a SELECT off that |
| 04:46.34 | predictive | I really didnt want to dig into the c |
| 04:46.48 | iCEBrkr | Dunno if you ever used MovieFone.. But you 'spell' the name of the movie you wanna see and it'll look it up. |
| 04:47.12 | predictive | yeah, I'm familiar with that kind of interface |
| 04:47.13 | predictive | useful |
| 04:47.22 | predictive | this is simpler even |
| 04:47.42 | predictive | I just want to play arbitrary wav files based on what I find in my data source |
| 04:48.16 | iCEBrkr | ah |
| 04:48.26 | predictive | I dont see a good way to do that in the manager api |
| 04:48.36 | iCEBrkr | I agree |
| 04:48.39 | predictive | you could fake it by setting variables that exist in the dialplan |
| 04:48.47 | predictive | but since I desire asynchronous operation |
| 04:48.49 | predictive | that's unreliable |
| 04:49.19 | predictive | that seems wierd, I would have thought this was a common desire |
| 04:49.20 | iCEBrkr | Thu Apr 28 00:49:19 EDT 2005 |
| 04:49.23 | iCEBrkr | welp.. |
| 04:49.24 | iCEBrkr | Zzzzz time. |
| 04:49.28 | rtomson-home | Hello all. Anyone using a TE410 with a Supermicro 1U server? |
| 04:49.48 | iCEBrkr | predictive: it's a pain in the butt no matter how you cut it. :) |
| 04:51.05 | predictive | time to dig into the source I guess |
| 04:51.43 | shido6 | who needed the adtran hack? |
| 04:52.06 | shido6 | http://www.governmentsecurity.org/articles/DefaultLoginsandPasswordsforNetworkedDevices.php |
| 04:53.42 | *** join/#asterisk psycodad (~obiwan@2001:4060:4419:b1:0:0:0:2) |
| 04:57.08 | *** join/#asterisk R-Guy (~ron-mirc@24.87.100.41) |
| 04:59.50 | *** join/#asterisk flot (CCCP@213.152.157.68) |
| 04:59.56 | blitzrage | shido6: I did :) |
| 05:01.09 | blitzrage | shido6: thanks |
| 05:01.39 | shido6 | i havent done anything yet blitzrage |
| 05:01.51 | shido6 | do you have physical access to the adtran? |
| 05:01.51 | blitzrage | lol |
| 05:01.53 | blitzrage | yes |
| 05:01.57 | shido6 | sweet |
| 05:01.58 | blitzrage | well, my friend does |
| 05:02.06 | blitzrage | its he who is trying to crack it :) |
| 05:02.09 | shido6 | does the friend know who set the pword? |
| 05:02.17 | blitzrage | nope - obtained off eBay |
| 05:02.20 | shido6 | good lord |
| 05:02.30 | shido6 | and the previous owner has no idea? |
| 05:02.40 | blitzrage | lets assume no access to the previous owner :) |
| 05:03.03 | *** join/#asterisk outtolunc (~me@adsl-69-110-56-125.dsl.pltn13.pacbell.net) |
| 05:03.07 | shido6 | great |
| 05:03.36 | blitzrage | fun eh? :) |
| 05:06.21 | outtolunc | s/you/out/g; |
| 05:06.43 | newl | heh |
| 05:06.45 | outtolunc | the thrill is gone <G |
| 05:07.57 | *** join/#asterisk libpcp (libpcp@210.16.20.5) |
| 05:07.58 | shido6 | I take it USER and PASSWORD dont work blitzrage ? |
| 05:08.12 | shido6 | those are default |
| 05:08.19 | shido6 | can u reset it to default? |
| 05:08.52 | libpcp | hi all |
| 05:09.05 | libpcp | what is the rcf standard of asterisk in sip? |
| 05:14.29 | *** join/#asterisk irv999 (~irv999@ool-457249eb.dyn.optonline.net) |
| 05:15.04 | irv999 | hey guys. how is everyone?! :-D |
| 05:15.12 | *** join/#asterisk Trickyphillips (~Trickyphi@adsl-68-124-57-143.dsl.irvnca.pacbell.net) |
| 05:16.35 | outtolunc | i'm watching bad kunfu flicks <G> |
| 05:16.58 | outtolunc | how are you? |
| 05:17.22 | irv999 | not bad |
| 05:17.28 | irv999 | I am trying to find another asterisk programmer |
| 05:17.30 | outtolunc | good |
| 05:17.35 | outtolunc | for? |
| 05:17.59 | heison | shido6: can u check something for me? |
| 05:18.09 | heison | i'm experiencing one-way audio with Nufone on IAX |
| 05:18.26 | heison | i suspect trunking mismatch |
| 05:18.34 | outtolunc | how new of cvs? (one way audio) |
| 05:18.47 | heison | <PROTECTED> |
| 05:18.49 | irv999 | I have a new asterisk system 35 stations 4 voicemail boxes 6 direct analog extensions.. |
| 05:19.11 | JerJer | heison: are you registered to switch-1 ? |
| 05:19.21 | JerJer | how about trying trunk=no |
| 05:19.27 | outtolunc | try takiing 'trunk=yes' out and cvs up'n |
| 05:19.33 | heison | 209.91.145.154:4569 heison 69.196.96.46:4569 60 Registered |
| 05:19.33 | heison | 198.22.67.70:4569 heison 69.196.96.46:4569 60 Registered |
| 05:19.40 | JerJer | that's not switch-1 |
| 05:19.44 | JerJer | or are you sending outboudn to switch-2? |
| 05:19.50 | outtolunc | omg, sorry, didn't know th master was in |
| 05:19.51 | rtomson-home | Anyone using spandsp with a PRI / T100P and getting clear faxes? |
| 05:20.03 | JerJer | rtomson-home: hell yes we are |
| 05:20.23 | rtomson-home | O.k. what am I doing wrong? |
| 05:20.24 | heison | JerJer: i'm registered to switch-2 |
| 05:20.54 | rtomson-home | I have P4 3.2 GHZ, 2GB Ram and SUSE 9.2, latest CVS of Asterisk. |
| 05:21.29 | rtomson-home | I get lines on the pages from both HP and Brother fax machines, about every 6-8 inches? |
| 05:21.36 | heison | JerJer: i'm sending calls to switch-2, with trunk=no |
| 05:21.56 | heison | Dial("SIP/cisco1-3255", "IAX2/heison@switch-2/14164178893") |
| 05:22.16 | heison | Call accepted by 198.22.67.70 (format g729) |
| 05:23.40 | heison | has anyone here experienced a bad TDM400 card? modprobe wcfxs is not going anywhere, and taking up 99% CPU... |
| 05:24.10 | JerJer | try ulaw or gsm |
| 05:25.15 | heison | Call accepted by 198.22.67.70 (format gsm) |
| 05:25.23 | heison | still no sound from PSTN |
| 05:25.42 | heison | JerJer: should i try ulaw? |
| 05:25.46 | outtolunc | or try switch-1 without trunk=anything |
| 05:25.58 | outtolunc | just comment that shit out |
| 05:26.22 | outtolunc | always try ulaw before complaining.. |
| 05:26.26 | heison | outtolunc: i tried that too |
| 05:26.37 | rtomson-home | JerJer, What versions of software / hardware are you using? |
| 05:27.16 | outtolunc | then paste it |
| 05:27.20 | heison | Call accepted by 198.22.67.70 (format ulaw) |
| 05:27.22 | heison | nothing... |
| 05:27.39 | outtolunc | using what as a client? |
| 05:27.55 | heison | 7960 - * - nufone - PSTN |
| 05:28.02 | outtolunc | the client |
| 05:28.04 | outtolunc | ? |
| 05:28.09 | heison | what client? |
| 05:28.15 | outtolunc | the cisco? |
| 05:28.20 | heison | yes, 7960 |
| 05:28.24 | outtolunc | and cvs date of? |
| 05:28.31 | heison | CVS-D2005.03.09.05.00.00-04/18/05-17:25:45 |
| 05:28.43 | outtolunc | that is you prob |
| 05:28.54 | heison | why? |
| 05:29.14 | outtolunc | it wasn't until 4-ish that the slinear issue was resolved |
| 05:29.22 | outtolunc | cvs up |
| 05:29.48 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:29.51 | heison | slinear? am i using slinear? |
| 05:30.33 | outtolunc | regardless, if you can't cvs up, why can you expect current help |
| 05:30.57 | outtolunc | you aren't using stable ... so HELLO |
| 05:30.59 | heison | what makes u think i can't cvs up |
| 05:31.08 | outtolunc | because you haven't |
| 05:31.22 | outtolunc | please prove me wrong |
| 05:31.22 | heison | but what does slinear have to do this? |
| 05:31.46 | outtolunc | just don't come pouting with a month old build |
| 05:31.56 | heison | lol |
| 05:32.27 | outtolunc | whats stopping you? |
| 05:32.31 | heison | lol |
| 05:32.36 | outtolunc | WELL? |
| 05:33.02 | outtolunc | comeon |
| 05:33.05 | *** join/#asterisk afrosheen (~afro@c-67-166-172-141.hsd1.tx.comcast.net) |
| 05:33.24 | outtolunc | i'm toasted, so i frankly don't give a rats ass |
| 05:33.51 | outtolunc | but i'm also fairly good with asterisk and therefore you "MIGHT" want to listen |
| 05:34.20 | outtolunc | if not, kiss off, and be done with it |
| 05:34.33 | outtolunc | i just don't care |
| 05:34.49 | afrosheen | well said old bean |
| 05:35.09 | outtolunc | if you want, and need HELP, AND are willing to listen, then i'll help |
| 05:35.22 | afrosheen | dude who's the phantom you're going off on |
| 05:35.22 | blitzrage | help no one! |
| 05:35.27 | afrosheen | it's just you and I here |
| 05:35.43 | outtolunc | afro, there are some here that just can't comprehend my age |
| 05:35.50 | blitzrage | outtolunc has lost it :) |
| 05:35.57 | afrosheen | dude go hit the pillows |
| 05:36.01 | blitzrage | outtolunc: I can't count that high :) |
| 05:36.21 | outtolunc | no, i've just lost the willingness to help whinnies |
| 05:37.20 | blitzrage | outtolunc: I hear yah |
| 05:37.37 | blitzrage | outtolunc: I stopped helping in here because I lost the interest in being someones personal google |
| 05:37.44 | afrosheen | so does everyone have asterisk working well these days? |
| 05:38.07 | blitzrage | works great here |
| 05:38.14 | outtolunc | after 25+ years in this industry.. i just can't take the whining |
| 05:38.15 | rtomson-home | Works wonderfully, except for those pesky dropped lines in my faxes. |
| 05:38.18 | blitzrage | heck, I even have incoming H.323 working :) |
| 05:38.22 | outtolunc | sorry |
| 05:38.44 | blitzrage | outtolunc: you need to worry less about other people or you'll go insane like me |
| 05:38.48 | blitzrage | :) |
| 05:39.07 | afrosheen | I figured most people had * working well now, because each time I visit this channel now it's kinda quiet |
| 05:39.23 | rtomson-home | Anyone care to share there combination, of hardware / versions that have clear faxes? |
| 05:39.28 | outtolunc | naw, i snatch people by the throat still (obviously in person) so that is not an issue |
| 05:40.13 | outtolunc | there are alot of us that have asterisk hummm'in |
| 05:40.23 | outtolunc | ask a f'n question |
| 05:40.23 | blitzrage | rtomson-home: run it at 9600 baud with spandsp |
| 05:41.31 | rtomson-home | blitzrage, O.k. Stupid question, where can I set that. |
| 05:41.33 | *** join/#asterisk oej (~oej@40.186.204.213.sol.worldonline.se) |
| 05:41.34 | blitzrage | Missouri is MI or MO ? |
| 05:41.42 | blitzrage | oej: !!! |
| 05:41.56 | outtolunc | for any of you new guys, i've been dev'n on asterisk for a couple years now.. so please don't take the nastiness as normal |
| 05:41.58 | oej | Blitzrage: Good morning. Missouri is mo |
| 05:42.03 | blitzrage | oej: do you have the address for IPSando world headquarters? |
| 05:42.06 | blitzrage | oej: thanks |
| 05:42.15 | blitzrage | MI is Michigan then I presume :) |
| 05:42.24 | blitzrage | oej: I'm thinking of going to bed soon ... :) |
| 05:42.54 | afrosheen | hey has the meetme delay bug been fixed yet? |
| 05:43.07 | blitzrage | outtolunc: thats true - I've never seen you so hostile :) |
| 05:43.16 | outtolunc | what delay |
| 05:43.42 | afrosheen | the 500ms or so delay that gets introduced with each added caller, there's a bug on the list, just seems like the devs got stalled on a solution |
| 05:43.45 | oej | blitzrage: No, I've lost the new address somewhere |
| 05:43.50 | outtolunc | i've been is shit for a couple weeks and i'm just plain tired of 'stupid shit' |
| 05:43.59 | outtolunc | er is=in |
| 05:44.00 | oej | blitzrage: The old one still works though |
| 05:44.58 | outtolunc | hehe |
| 05:45.13 | blitzrage | oej: ok cool - just trying out some new invoicing software |
| 05:45.22 | afrosheen | nobody knoooows my sor-rooowwww |
| 05:45.52 | outtolunc | sadly, i'm just toasted enough to care'a wee bit less' than normal |
| 05:46.07 | blitzrage | I'm not toasted at all.. just tired :) |
| 05:46.08 | afrosheen | I just want to get that delay fixed asap so my boss doesn't have to play 'coach' on conferences |
| 05:46.15 | outtolunc | i'm tired also |
| 05:46.23 | newl | Don't sweat the small shit, because it's all small shit. |
| 05:46.23 | afrosheen | he's always saying 'ok you go!' and it's retarded |
| 05:47.09 | outtolunc | newl, i'll ask you in 25 years what you consider 'small shit' |
| 05:47.28 | outtolunc | respond how you feel at that time |
| 05:47.49 | newl | 25 years ago, it was still small shit. :) |
| 05:48.08 | outtolunc | been there done that? |
| 05:48.13 | afrosheen | now it's 25 year old small shit? so it's dry and crusty. |
| 05:48.14 | outtolunc | if so, sorry |
| 05:48.14 | *** join/#asterisk Corydon76-home (gold@pcp08665860pcs.500ash01.tn.comcast.net) |
| 05:48.54 | outtolunc | hey cory |
| 05:48.55 | *** join/#asterisk akl- (~akl@akl-.user) |
| 05:49.27 | outtolunc | just us discontent elder assbytes hangin <G> |
| 05:50.14 | outtolunc | and noone has a f'n thing to say now |
| 05:51.00 | outtolunc | if any of you have anything even remotely 'dev' oriented i'd love to hear it |
| 05:51.18 | afrosheen | the devs are sleeping in the corner |
| 05:51.22 | afrosheen | all huddled together |
| 05:51.28 | outtolunc | obviously not |
| 05:51.40 | outtolunc | or didn't you get that part |
| 05:51.42 | afrosheen | well all of them except for you |
| 05:51.49 | PTG1234 | !seen file |
| 05:51.52 | PTG1234 | ~seen file |
| 05:51.58 | jbot | file is currently on #asterisk (4h 42m 39s) #elinux (4h 42m 39s) #tacobeam (4h 42m 39s). Has said a total of 3 messages. Is idling for 2h 11m 9s |
| 05:52.11 | afrosheen | tacobeam? |
| 05:52.30 | newl | I theeenk I need a beeger bawx. |
| 05:53.45 | outtolunc | file can't help you, he knows me better than most |
| 05:56.13 | opus_ | has anyone made a thinclient phone that works with asterisk? |
| 05:56.41 | afrosheen | thinclient phone? |
| 05:57.40 | Corydon76-home | outtolunc: #4098 ? |
| 05:58.42 | opus_ | yeah, like java |
| 05:58.44 | opus_ | or something |
| 05:58.59 | opus_ | like, skype but no download |
| 05:59.00 | oej | e |
| 05:59.10 | Corydon76-home | Yeah, they're called Zaptel devices |
| 05:59.13 | *** join/#asterisk chaoscon (~ph33r@chaoscon.user) |
| 05:59.22 | afrosheen | if it's java you'll be downloading the jar to run in your browser or wherever |
| 05:59.22 | Corydon76-home | They're about as thin a client as you can get |
| 05:59.51 | opus_ | if you link to talk:username net meeting comes up in IE |
| 06:00.08 | Corydon76-home | Or maybe you didn't really want a thin client. Maybe you wanted a smart client |
| 06:00.15 | outtolunc | checking |
| 06:00.38 | afrosheen | do yourself a favor and code a client for blackberries or something like that |
| 06:01.07 | opus_ | yeah, that would be cool to use a smartphone as a usb phone |
| 06:01.21 | *** join/#asterisk jdims (~sleepless@24-119-121-6.cpe.cableone.net) |
| 06:01.29 | afrosheen | usb for what, just use it as a phone but running sip over your data link |
| 06:02.10 | opus_ | have you used a usb phone before |
| 06:02.11 | *** join/#asterisk McUnixJr (~mc@McUnixJr.gold.supporter.pdpc) |
| 06:03.56 | kb1_kanobe | What happens if I have one side of an iax peer with trunking enabled and the other side without? |
| 06:04.09 | kb1_kanobe | s/peer/connection |
| 06:04.17 | afrosheen | that's a camouflaged question :) |
| 06:04.29 | outtolunc | trunking is trunking |
| 06:04.50 | outtolunc | if it can be, it is.. if NOT, obviously it isn't |
| 06:04.55 | kb1_kanobe | outtolunc: the enabled side of the channel will be trunked and the disabled side won't? |
| 06:05.06 | kb1_kanobe | or will the enabled side fall back? |
| 06:05.08 | outtolunc | so, it can ONLY trunk as far as it CAN |
| 06:05.29 | outtolunc | understand? |
| 06:05.53 | outtolunc | it attempts to trunk (iax2) regardless |
| 06:05.54 | kb1_kanobe | yes. ie. a-(trunk)-b-(no trunk)-c |
| 06:06.00 | outtolunc | if it can it does |
| 06:06.52 | outtolunc | meaning, if you are running iax2, REMOVE any trunk=yes in your iax.conf.. it's worthless, probably hurts |
| 06:07.34 | McUnixJr | question - if i have 3 VoIP providers (Vonage, Packet8 and Sunrocket) what cards should i buy to tie these into a asterisk system? |
| 06:08.02 | kb1_kanobe | outtolunc: errr... Actually I have to specifically disable it for now - it's breaking newjb. My question was just theoretical. |
| 06:09.57 | outtolunc | if trunk=yes is causing one-way audio, you need to cvs-update because it's really the slinear issue |
| 06:10.01 | *** join/#asterisk FuriousGeorge (~brian@ool-43516ebb.dyn.optonline.net) |
| 06:10.24 | kb1_kanobe | Hmmm... I'm seeing a miniframe timestamp problem. |
| 06:10.30 | outtolunc | i just got done 'priv'n a guy to fix with the same damn issue |
| 06:10.36 | FuriousGeorge | anyone home? i built acpi into my kernel to get my zaptel modules to load, and it worked, but video and usb are dead now |
| 06:10.53 | blitzrage | acpi to get zaptel to work? thats a new one... |
| 06:11.02 | kb1_kanobe | I'm running yesterdays head and the audio drops out after the timestamp exceeds 70000. |
| 06:11.19 | FuriousGeorge | blitzrage: well maybe the fact that enabling acpi allowed me to enable pluig and play support in make menuconfig |
| 06:11.28 | kb1_kanobe | outtolunc: if I turn off the jb it's fine all the way through |
| 06:11.34 | outtolunc | kanobe, that sounds like a switch/router issue not asterisk |
| 06:12.02 | outtolunc | jb? |
| 06:12.11 | outtolunc | sorry i'm not versed |
| 06:12.11 | kb1_kanobe | jitterbuffer |
| 06:12.19 | outtolunc | oh my god |
| 06:12.30 | outtolunc | you are running a mod |
| 06:12.35 | kb1_kanobe | ? |
| 06:12.37 | outtolunc | 'd version aren't you |
| 06:12.49 | outtolunc | with steve's stuff |
| 06:12.58 | *** join/#asterisk Nethab (~chatzilla@adsl-67-113-141-170.dsl.sntc01.pacbell.net) |
| 06:13.03 | kb1_kanobe | head w/patches from #3965 and #3961, per stevek. |
| 06:13.26 | outtolunc | cvs up with the freaking patches... test THEN come back |
| 06:13.32 | outtolunc | er without |
| 06:13.52 | kb1_kanobe | I only put in the patches tonight, it's been borken all day. |
| 06:14.09 | outtolunc | add one patch, test, then the other, i feel you'll find the issue |
| 06:14.28 | outtolunc | just start from a stable base |
| 06:14.29 | blitzrage | FuriousGeorge: I still don't see the relation :) |
| 06:14.30 | kb1_kanobe | Trust me, head - newjb problem. head+either or both patches, same problem. |
| 06:14.32 | outtolunc | please |
| 06:14.46 | outtolunc | i've heard it all before |
| 06:14.52 | outtolunc | really |
| 06:14.54 | kb1_kanobe | I trust you. :-) |
| 06:15.01 | outtolunc | you should |
| 06:15.14 | outtolunc | i'm the only on here with nothing at stake |
| 06:15.34 | outtolunc | well there might be others |
| 06:15.48 | *** join/#asterisk SilentBob420 (silentbob@mrburns.4twenty.us) |
| 06:15.50 | outtolunc | but few with as many of years at TS as i |
| 06:16.01 | outtolunc | afk nature |
| 06:16.06 | kb1_kanobe | TS? |
| 06:17.23 | outtolunc | tech support |
| 06:17.35 | kb1_kanobe | Ah yes. I assumed 'This Shit' :-) |
| 06:18.04 | outtolunc | people don't call me for simple shit, they call me when they others can't find the issue |
| 06:18.46 | kb1_kanobe | do you consult on asterisk code or implementation? |
| 06:19.17 | outtolunc | i haven't to date, though i have done few mod's |
| 06:19.54 | outtolunc | when i said, i have nothing at stake here, i meant it |
| 06:20.02 | kb1_kanobe | cool. |
| 06:20.20 | outtolunc | i do mod's, for any OS i find |
| 06:20.40 | outtolunc | it's that simple |
| 06:20.49 | kb1_kanobe | mod's? |
| 06:21.07 | afrosheen | you know, those 4 track digital audio files from the amiga days |
| 06:21.09 | outtolunc | sometimes it takes awhile for those mod's to get IN (grin)... |
| 06:21.25 | predictive | afrosheen: hah |
| 06:21.36 | McUnixJr | can someone point me where to look to find cards to integrate 3 VoiP vendors into one asterisk system? |
| 06:21.38 | outtolunc | remembers 9track data still what a pain |
| 06:21.40 | McUnixJr | please |
| 06:21.41 | kb1_kanobe | oh god, I haven't played with mod files in years.... last time was tracker on an Archimedes I think. |
| 06:22.17 | afrosheen | stormtracker |
| 06:22.37 | outtolunc | mcJr, cards are cards.. it's the protocol you let them in on that makes a difference |
| 06:22.44 | afrosheen | McUnixJr: incorporate vendors? as in..how |
| 06:23.17 | McUnixJr | i have 3 VoIP providers (Packet8, Voange and Sunrocket) want to have those lines come into asterisk system |
| 06:23.34 | McUnixJr | would i use analog cards? |
| 06:23.43 | afrosheen | packet8 uses some bullshit proprietary hardware or codecs right? |
| 06:23.52 | outtolunc | kanobe, we could spend hours talking about 'the old days'... sad as there were |
| 06:24.02 | afrosheen | vonage is the same |
| 06:24.03 | McUnixJr | all three have boxes |
| 06:24.09 | afrosheen | dunno about sunrocket |
| 06:24.27 | kb1_kanobe | outtolunc: indeed. lots of good material back there. |
| 06:24.38 | afrosheen | I suppose you could pull off some hack like plugging tdm cards into the ports on those boxes but hell..why not just get a real trunk provider |
| 06:24.40 | shido6 | brb |
| 06:24.46 | outtolunc | it was the 'path' not the destination |
| 06:25.22 | McUnixJr | afrosheen, already have these - thought it would be nice to utilize them |
| 06:25.28 | outtolunc | rarely do you find people that can code in as many languages |
| 06:25.29 | opus_ | mcunixjr - do you have a bunch of linksys boxes |
| 06:25.55 | McUnixJr | one linksys box, one funky no name box from Sunrocket and a Packet8 box |
| 06:26.05 | McUnixJr | the linksys is from vonage |
| 06:26.14 | opus_ | they are generally called FXS ports.. |
| 06:26.25 | afrosheen | in theory, if each box will give you a dialtone, you could run them into a tdm card |
| 06:26.25 | *** join/#asterisk Nethab (~chatzilla@adsl-67-113-141-170.dsl.sntc01.pacbell.net) |
| 06:26.43 | opus_ | unless you know how to burn your own firmware, or you know how to do a mitm attack, they're basically useless |
| 06:26.48 | afrosheen | it's wacky and somewhat wasteful but I imagine it'd work |
| 06:27.36 | outtolunc | e |
| 06:27.39 | outtolunc | e said |
| 06:27.42 | McUnixJr | packet8 we got as it has free european calls for a flat fee per month |
| 06:29.10 | opus_ | they are hard coded to call the mothership. |
| 06:29.34 | afrosheen | so does anyone think running some cables from a tdm card to the magic boxes will work |
| 06:29.40 | outtolunc | yes, the vonage toys, are set to phone home |
| 06:30.11 | outtolunc | if any of you read the archives this WOULD NOT be a new thing |
| 06:30.18 | McUnixJr | so get a TDM card - that would also integrate my POTS lines as well? |
| 06:30.28 | opus_ | why? |
| 06:30.51 | outtolunc | opus, the why is called 'contracts' |
| 06:31.09 | outtolunc | and shit like market share |
| 06:31.21 | outtolunc | deal with the devil |
| 06:31.28 | outtolunc | understand? |
| 06:31.31 | opus_ | you can hack them.. :) |
| 06:31.42 | outtolunc | you can hack some of them |
| 06:32.06 | afrosheen | McUnixJr, primarily that's what tdm cards are for, pstn or pots lines |
| 06:32.11 | outtolunc | others (the latest ones) as i understand it are not 'hackable' |
| 06:32.26 | afrosheen | McUnixJr, they come with fxo or fxs ports installed on them depending on what you need them to do |
| 06:33.02 | *** join/#asterisk nkryptr (nkryptr@katana.ninjas.org) |
| 06:33.06 | McUnixJr | so if i wanted to integrat 3VoIP (basically then equal to POTS) and 1 POTS - that means 4 FXO ports? |
| 06:33.23 | outtolunc | time may tell, but as i understood, the ones from the last 6+ months aren't 'asterisk friendly' |
| 06:33.26 | afrosheen | I believe so |
| 06:33.41 | *** join/#asterisk Nethab (~chatzilla@adsl-67-113-141-170.dsl.sntc01.pacbell.net) |
| 06:33.51 | outtolunc | because of issues put forth by the mnfr |
| 06:33.57 | afrosheen | would be a tdm400x |
| 06:34.03 | Nethab | can you hear me now |
| 06:34.13 | outtolunc | eh? |
| 06:34.23 | outtolunc | more to the left |
| 06:34.26 | Nethab | is that a no? |
| 06:34.27 | outtolunc | no more |
| 06:34.30 | outtolunc | more |
| 06:34.32 | *** join/#asterisk bjohnson (~bjohnson@66.11.165.59) |
| 06:34.33 | outtolunc | more |
| 06:34.49 | Nethab | yes? no? |
| 06:34.58 | outtolunc | no can't hear you |
| 06:35.05 | afrosheen | move closer to that open window |
| 06:35.06 | Nethab | ok finally, stupid chatzilla |
| 06:35.37 | outtolunc | yeah, it's the 'client' |
| 06:35.44 | afrosheen | McUnixJr, since asterisk uses fxs signalling for fxo, I'm permanently confused when it comes to the distinction, but I'm pretty sure you want a card with 4 fxo ports |
| 06:35.46 | McUnixJr | afrosheen, the tdm400x - would digium be the sole provider? (yes i am so new at this) |
| 06:36.08 | outtolunc | i wish i had $.05 for every freakin time i heard that |
| 06:36.12 | knight_ | hey |
| 06:36.24 | afrosheen | yeah digium is the only reliable source for this stuff, when they have it stocked |
| 06:36.28 | opus_ | fxs supplys ring voltage, fxo can answer a ring voltate but can't generate one. |
| 06:36.44 | predictive | I just bought one, it only took 3 days to arrive |
| 06:36.54 | McUnixJr | predictive, how much? |
| 06:37.03 | afrosheen | then they're getting faster, we wanted a replacement last month and they said 2 weeks at least |
| 06:37.03 | predictive | McUnixJr: it's on their website |
| 06:37.09 | outtolunc | if any of you read the ML, there ARE other options |
| 06:37.09 | afrosheen | www.digium.com |
| 06:37.44 | outtolunc | side note: |
| 06:37.45 | afrosheen | predictive: then he needs a 4 port fxo, wants to run asterisk behind some proprietary voip hardware from vonage et al |
| 06:37.45 | predictive | McUnixJr: digium's cards aren't the only ones that work, but they are the source behind asterisk |
| 06:37.55 | predictive | afrosheen: yea quad fxo |
| 06:38.15 | McUnixJr | so TDM04B ? |
| 06:38.18 | opus_ | they're just $5 dollar fax cards |
| 06:38.29 | afrosheen | well digium's hardware is a safe bet, if you buy shit from someone else, nobody will ever help you |
| 06:38.32 | outtolunc | if YOU were digium, and an explosion occured, in regards to your product.. might you also run out? |
| 06:39.05 | outtolunc | digium does NOT sell winmodems anymore.. so why not update your crap list |
| 06:39.13 | predictive | TDM04B |
| 06:39.15 | afrosheen | yeah |
| 06:39.18 | predictive | oh you just said that |
| 06:39.22 | predictive | sorry |
| 06:39.28 | afrosheen | that's the bundle with 4 fxo |
| 06:39.45 | afrosheen | McUnixJr, and since you're so new to this stuff, you might want to start with asterisk@home also |
| 06:39.55 | outtolunc | that is NOT a winmodem nor should be dealt with as such |
| 06:40.11 | afrosheen | http://asteriskathome.sourceforge.net/ |
| 06:40.33 | opus_ | basically it is |
| 06:40.41 | outtolunc | haha |
| 06:40.53 | afrosheen | opus_, yeah the tigerjet hardware is definitely nothing new :) |
| 06:40.57 | predictive | McUnixJr: also check out http://www.voip-info.org/wiki-Asterisk |
| 06:41.10 | outtolunc | i want to respond... please respond |
| 06:41.17 | afrosheen | McUnixJr, yep, good source of info there but overwhelming for noobs |
| 06:42.17 | outtolunc | opus, what hardware do you have? |
| 06:42.26 | *** join/#asterisk rjburkh (~rjburkh@dialup-4.231.169.69.Dial1.LosAngeles1.Level3.net) |
| 06:42.29 | *** join/#asterisk shido6 (~shido@d57-87-253.home.cgocable.net) |
| 06:42.35 | Nethab | it wasn't overwhelming for me when I started, but I compile my own kernels just to stick it to the Man |
| 06:42.52 | predictive | it's not too bad |
| 06:42.57 | predictive | dialplans could be a little easier |
| 06:42.58 | opus_ | outtolunc - i don't use any analog crap, an old pentium III and a few polycom ip phones. |
| 06:43.12 | Nethab | if you're familiar with non VoIP phone systems it's not bad |
| 06:43.33 | outtolunc | oh so channel banks (being analog) are crap |
| 06:43.46 | afrosheen | Nethab, my problem originally is, I jumped into all this with almost zero phone system knowledge other than beige boxing ;) |
| 06:44.01 | afrosheen | but that was 8 months ago and I know a little more now |
| 06:44.11 | opus_ | to expensive to invest in a technology that it days are numbered. |
| 06:44.16 | *** join/#asterisk gres (~serg@81.222.48.242) |
| 06:44.24 | outtolunc | so whan i have a few 4 port t1/pri cards and some channel banks..thats just crap also? eh? |
| 06:44.38 | Nethab | my first jump was setting up a T1 line with a sangoma card about 5 years ago, had nothing to do with asterisk though |
| 06:44.40 | predictive | considering the size of the 'legacy' system, it's a large number of days |
| 06:45.02 | McUnixJr | afrosheen, i followed these instructions http://www.asterisk.org/index.php?menu=download and installed without a hitch (pretty exciting with no hitch at all) onto a RH EL 3 system tonite - i did it from CVS stable |
| 06:45.38 | McUnixJr | i decided not to take the easy route - thought it would be difficult - turned out to work like a charm - except no card :) |
| 06:46.13 | predictive | McUnixJr: if you have a quad fxo card then you can just plug your boxes in, define the channels, and figure out what your dialplan should look like |
| 06:46.16 | predictive | it's only hard the first time |
| 06:46.18 | afrosheen | McUnixJr, but stuff like *@home gives you tons of functionality out of the box with little hassle, alot of people like it |
| 06:46.38 | predictive | does anyone have one of the TE410Ps? |
| 06:46.53 | predictive | the data sheet doesn't say if it does ESF |
| 06:47.00 | McUnixJr | afrosheen, cant i build that functionality into what i have instaleld already? isnt the packages available seperately? |
| 06:47.04 | outtolunc | i so |
| 06:47.07 | outtolunc | er do |
| 06:47.14 | predictive | does it? |
| 06:47.17 | afrosheen | McUnixJr, yeah but for some of that stuff it's a big hassle |
| 06:47.23 | kb1_kanobe | predictive: it doesn't - asterisk does the framing. |
| 06:47.33 | kb1_kanobe | so, yes - they do ESF. |
| 06:47.34 | afrosheen | McUnixJr, if you ever install AMP you'll probably love it |
| 06:47.36 | predictive | hm |
| 06:47.36 | outtolunc | sf/esf |
| 06:47.56 | kb1_kanobe | see zaptel.conf and zapata.conf |
| 06:48.37 | afrosheen | ha, my local blockbuster has a movie on Zapata |
| 06:48.56 | wildgoose | anyone know how to change the voicemail prompts? I want to turn off the "at the end you can hangup or press * to re-record", or whatever it says |
| 06:49.08 | McUnixJr | afrosheen, AMP is?? where?? |
| 06:49.28 | McUnixJr | never mind |
| 06:49.30 | McUnixJr | found it |
| 06:49.31 | McUnixJr | duh |
| 06:49.32 | afrosheen | McUnixJr, amp.voxbox.ca |
| 06:49.35 | afrosheen | hehe |
| 06:51.33 | kb1_kanobe | wildgoose: load the appropriate .gsm file into an audio editor and cut the end off. |
| 06:52.18 | afrosheen | kb1_kanobe, is that whole thing one single gsm or is it scripted? |
| 06:53.00 | kb1_kanobe | dunno. I think it's one - I had a patch a while back that would disable the instructions if there was a customised greeting for the mbox. Dunno what happened to it. |
| 06:53.09 | predictive | kb1_kanobe: I see the file but it's not obvious to me which one is D5 RBS |
| 06:53.43 | kb1_kanobe | predictive: robbed bit? |
| 06:53.48 | predictive | kb1_kanobe: yes |
| 06:53.55 | kb1_kanobe | D5 = Feature Group D? |
| 06:54.11 | predictive | d5 is esf |
| 06:54.14 | predictive | sorry |
| 06:54.26 | predictive | I see sf_featd |
| 06:54.56 | kb1_kanobe | set the framing in zaptel.conf for the span |
| 06:55.09 | predictive | oh hadn't poked there yet |
| 06:55.16 | kb1_kanobe | then, if using RBS, set the appropriate signalling for each channel of the span in zapata.conf |
| 06:55.28 | kb1_kanobe | Your life would be easier with ISDN. |
| 06:55.35 | predictive | hehe tell me about it |
| 06:56.14 | predictive | I see I looked at the wrong file first, thanks |
| 06:56.58 | predictive | I learned this in the military, their equipment calls it 'D5 Framing', not 'Extended Superframe' |
| 06:57.27 | kb1_kanobe | I've heard it now you mention it - but isn't D5 slightly different? |
| 06:57.28 | afrosheen | the amp install is not fun but once you get it right it'll work forever..upgrades are easy also |
| 06:57.38 | outtolunc | though i've seen quite a few esf/b8zs/wink lines being ordered |
| 06:57.46 | predictive | kb1_kanobe: not to my knowledge, but what do I know |
| 06:58.16 | *** join/#asterisk teamjet (~teamjet@lfc.tor.istop.com) |
| 06:58.33 | kb1_kanobe | google agrees with you: ESF, aka D5, aka FE |
| 06:58.57 | outtolunc | <- used to be in the military also, actually i was one of the guys that helped them realise what info tech could do |
| 06:59.12 | predictive | outtolunc: you ought to talk to them again then |
| 06:59.31 | McUnixJr | outtolunc - what branch? |
| 06:59.38 | outtolunc | they didn't listen then, nothing has changed |
| 06:59.43 | outtolunc | marines |
| 07:00.06 | teamjet | closest i got to a gun is a first person shooter |
| 07:00.11 | *** join/#asterisk Gh0sty (~ghosty@ip-81-11-210-49.dsl.scarlet.be) |
| 07:00.20 | Gh0sty | hello |
| 07:00.29 | predictive | I did comms in the navy |
| 07:00.38 | predictive | then comms for office of naval intelligence |
| 07:00.39 | teamjet | Gh0sty: belgium? |
| 07:00.42 | Gh0sty | yep |
| 07:00.42 | predictive | now comms for me! |
| 07:00.43 | predictive | yay |
| 07:01.13 | predictive | MIO stuff is fun |
| 07:01.23 | teamjet | you guys are old |
| 07:01.28 | predictive | yes |
| 07:01.34 | zoa | goood morning |
| 07:01.36 | Gh0sty | got a problem, just setup a demo yesterday in vmware (i know don't shoot me, shoot my boss) but when i dial the demo i get terrible lag and packet loss it seems |
| 07:02.01 | Gh0sty | offcourse this is the same pc as the asterisk is running :s |
| 07:02.01 | outtolunc | pred: i was 1 of the 2 guys that first got battalion level ismo going |
| 07:02.11 | outtolunc | in the marines |
| 07:02.12 | kb1_kanobe | Gh0sty: don't run it in vmware. |
| 07:02.13 | Gh0sty | is there a way to take another codec perhaps ? |
| 07:02.16 | zoa | teamjet and ghosty: nederlandstalig ? |
| 07:02.20 | teamjet | um, what's the difference between JTAPI/TAPI and Asterisk? |
| 07:02.22 | Gh0sty | ja |
| 07:02.23 | predictive | yeah vmware is probably not a great idea |
| 07:02.33 | *** join/#asterisk odie_flocon_ (~chatzilla@S01060011953d7c4c.cg.shawcable.net) |
| 07:02.34 | Gh0sty | i guessed that :p |
| 07:02.37 | odie_flocon_ | hmmm. |
| 07:02.41 | kb1_kanobe | use knoppix or something |
| 07:02.56 | predictive | teamjet: asterisk is a software pbx, tapi is an api to talk to them mostly |
| 07:03.07 | Gh0sty | no i'm just going to tell him to buy hardware ffs! |
| 07:03.09 | predictive | though ill defined and not well supported |
| 07:03.16 | kb1_kanobe | lol |
| 07:03.21 | predictive | i wish ip phones were cheap |
| 07:03.39 | teamjet | predictive: if i wrote clients that back to jtapi/tapi, they can talk to asterisk? |
| 07:03.41 | kb1_kanobe | use softphones, just make sure * is running on real hardware. |
| 07:04.13 | Gh0sty | yeah, i'm running softphones, but * is running vm, even with high priority set it still lags :p |
| 07:04.14 | predictive | teamjet: I haven't seen a tapi interface but coding a bridge between fastagi and it shouldnt be too hard |
| 07:04.25 | Gh0sty | interesting however :) |
| 07:04.31 | zoa | asterisk in vm is no good for a demo |
| 07:04.50 | Gh0sty | yeah, well it was my boss his idea :p |
| 07:05.06 | libpcp | anyone is using as5300 with asterisk? |
| 07:05.07 | outtolunc | zoa, stop using celerons for demos <G> |
| 07:05.23 | teamjet | predictive: sorry i am a little lost, i'll go read up on asterisk more so i can understand the difference better |
| 07:05.24 | Gh0sty | i wonder what problem might be however |
| 07:05.33 | zoa | VMware introduced lag ? |
| 07:05.40 | zoa | emulated network layer ? |
| 07:05.43 | *** join/#asterisk yxa (~void@203.118.40.42) |
| 07:05.49 | predictive | teamjet: you use a tapi library to write programs to talk to a PBX |
| 07:05.53 | Gh0sty | i've been told * doesn't take up that much resources, and if priority is set to high and the pc is idling ... |
| 07:05.56 | predictive | teamjet: asterisk *is* the PBX |
| 07:05.59 | zoa | true |
| 07:06.00 | zoa | but |
| 07:06.11 | zoa | VM is not really realtime |
| 07:06.12 | predictive | teamjet: but you don't need a TAPI lib to talk to asterisk |
| 07:06.21 | predictive | teamjet: there are other ways already to do that |
| 07:06.21 | zoa | i mean you could do a demo on an old p3 or so |
| 07:06.29 | outtolunc | here we go.. wait till i get another drink for the good stuff eh <G> |
| 07:06.30 | Gh0sty | yeah, i know |
| 07:06.31 | teamjet | asterisk has something like tapi? |
| 07:06.39 | Gh0sty | just beat my boss :p |
| 07:06.57 | Gh0sty | anyways this is only school job, in a month i'm outta here :) |
| 07:06.58 | Makenshi | teamjet, there is a tapi driver for * |
| 07:06.59 | predictive | teamjet: asterisk has agi which is sort of like cgi, and the manager interface |
| 07:07.00 | libpcp | maybe anyone can give me a complete specs on as5300 with sip complaint |
| 07:07.05 | Nethab | my asterisk runs on a p3 450 |
| 07:07.15 | Gh0sty | should a celeron 300 do ? |
| 07:07.16 | teamjet | predictive: thanks for answering all these noob questions |
| 07:07.22 | Gh0sty | got something like that at home :p |
| 07:07.30 | odie_flocon_ | my * runs on a P4 3ghz. |
| 07:07.35 | Makenshi | teamjet, http://www.voip-info.org/wiki-Asterisk+TAPI |
| 07:07.36 | Gh0sty | perhaps i even got P2 300 to put on the board :) |
| 07:07.46 | zoa | ghosty: get us to give the demo to your boss :P |
| 07:07.58 | predictive | teamjet: is there a reason you want to use TAPI specifically? |
| 07:08.00 | Gh0sty | :) |
| 07:08.46 | *** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it) |
| 07:08.53 | teamjet | predictive: if i have some free time i thought i'd learn how to learn hwo to write a pbx or telephony app |
| 07:09.03 | *** join/#asterisk Cronus-secut (~none@d5152D609.access.telenet.be) |
| 07:09.10 | predictive | I ask because if you just want to 'write an asterisk app' |
| 07:09.17 | predictive | that's not the easiest way to do it |
| 07:09.27 | teamjet | predictive: *nod* |
| 07:09.36 | Gh0sty | man i'd love a job in this stuff, seems asterisk really beats all the other stuff out of the market :) |
| 07:09.39 | Nethab | i wrote my own manager interface as a firefox extension |
| 07:09.39 | *** join/#asterisk RestLessGemini (~umairbari@202.142.189.86) |
| 07:09.51 | Makenshi | Gh0sty, hah, i dont think so :> |
| 07:09.56 | Gh0sty | at least if you can do it yourself (seen pricing on commercial asterisk, too expensive) |
| 07:09.59 | Makenshi | Gh0sty, maybe everything in the free market :> |
| 07:10.30 | predictive | teamjet: first learn asterisk, then read up on AGI and fastAGI |
| 07:10.48 | odie_flocon_ | hey. |
| 07:10.53 | predictive | that's the path of least resistance and you can use any language you happen to know |
| 07:10.53 | Cronus-secut | i've searched on sourceforge, frechmeat and google for a PHP interface to manage asterisk, but I don't seem to find much |
| 07:11.04 | odie_flocon_ | ever seen the price on a Mitel system? |
| 07:11.10 | Cronus-secut | do i have to start from scratch or are there started projects? |
| 07:11.19 | Makenshi | Cronus-secut, http://www.voip-info.org/wiki-Asterisk+GUI |
| 07:11.20 | teamjet | predictive: asterisk sounds fun |
| 07:11.24 | Makenshi | big list there |
| 07:11.25 | predictive | yeah |
| 07:11.29 | predictive | mitel hahaha |
| 07:11.30 | Gh0sty | Makenshi: well, if i see the situation here: existing isdn pbx, they don't wanna switch over to something fancy (and pricy) and want some remote offices connected and stuff ... |
| 07:11.36 | Gh0sty | then it counts :p |
| 07:11.36 | teamjet | how much to get started? |
| 07:11.39 | predictive | the SMDR apps alone are thousands |
| 07:11.52 | Cronus-secut | thx Makenshi, how stupid not to lok on voip-info :s |
| 07:12.01 | predictive | teamjet: money? find a box and buy a dev kit card for $200 |
| 07:12.06 | predictive | that will be enough for a long time |
| 07:12.13 | teamjet | oh |
| 07:12.16 | teamjet | USD$200? |
| 07:12.21 | predictive | unless you dont care about analog phones |
| 07:12.24 | odie_flocon_ | if you go the expensive way. |
| 07:12.27 | predictive | in which case a box and asterisk |
| 07:12.43 | odie_flocon_ | the install I"m doing is costing over 3k for the hardware. |
| 07:12.44 | teamjet | oh you mean like PC->NET<-PC calls? |
| 07:12.49 | predictive | right |
| 07:12.57 | Gh0sty | btw i was wondering: which line in sip.conf is actually the password for the softphone? |
| 07:12.59 | Makenshi | look on ebay for a rhino |
| 07:13.12 | odie_flocon_ | secret= |
| 07:13.16 | Gh0sty | i set em all the same, and filled up everything the same in the phone :p |
| 07:13.19 | teamjet | predictive: i got lost trying to find a suitable card, what model is currently for sell now days? |
| 07:13.26 | Gh0sty | hm, i don't even have a secret :) |
| 07:13.28 | predictive | look on digium.com |
| 07:13.42 | odie_flocon_ | I have to install mine tommorow. |
| 07:13.43 | teamjet | predictive: i believe they have many cards there |
| 07:13.49 | predictive | their card that has one module that talks to the phone system and one card for a telephone set is $200 |
| 07:13.55 | predictive | it's called teh 'dev kit pci' |
| 07:14.45 | Gh0sty | teamjet: what card are you looking for? |
| 07:14.59 | Gh0sty | i've made some research on cards available in be :) |
| 07:15.07 | zoa | ghosty |
| 07:15.11 | zoa | they are all available |
| 07:15.15 | zoa | we sell all of em |
| 07:15.28 | Gh0sty | oh and what is your company? :) |
| 07:15.35 | zoa | everything you can put into asterisk servers anyway |
| 07:15.45 | teamjet | predictive: i see it |
| 07:15.55 | Gh0sty | well, i meanth i've been looking around for prices and available types |
| 07:16.00 | teamjet | predictive: thx |
| 07:16.34 | teamjet | predictive: do you think a card like this has jtapi/tapi drivers even if you decide to develop without asterisk? |
| 07:16.48 | predictive | teamjet: tapi doesn't have anything to do with the card |
| 07:16.55 | predictive | the card just lets you talk to the phone system |
| 07:17.05 | predictive | tapi is one way to talk to asterisk |
| 07:17.30 | teamjet | lol wish i got a telecom degree, this is confusing |
| 07:17.39 | *** join/#asterisk oej (~oej@40.186.204.213.sol.worldonline.se) |
| 07:17.47 | predictive | you could develop for some time happily with nothing more than a machine, asterisk, and softphones |
| 07:17.55 | Nethab | no it's not |
| 07:18.10 | predictive | teamjet: first learn about asterisk |
| 07:18.14 | predictive | then worry about tapi later |
| 07:18.14 | Nethab | the worms crawl in the worms crawl out, the worms play penuccle in your snout |
| 07:18.20 | Nethab | it's not that complicated |
| 07:19.06 | odie_flocon_ | teamjet is stuck in a WINDOWS world. |
| 07:19.22 | odie_flocon_ | where if it isn't TAPI it doesn't work. |
| 07:19.39 | teamjet | must be |
| 07:19.54 | odie_flocon_ | * just works. |
| 07:19.57 | odie_flocon_ | and it works great. |
| 07:20.01 | teamjet | i am just unfamiliar with telephony in general, but pidgin hole me if you like |
| 07:20.26 | teamjet | i am platform agnostic, given the chance |
| 07:20.31 | peter222 | does anyone know if both the AVM Fritz V1 and V2 work in * ? |
| 07:20.38 | predictive | teamjet: telephony is largely arcane because people liked it that way, but it's not complicated |
| 07:20.55 | odie_flocon_ | it only get's ugly when you add windows :D |
| 07:21.16 | teamjet | i thougth i'd mention jtapi works with sun/linux world |
| 07:21.20 | predictive | how else could they charge $50,000 for the equivalent of a log parser |
| 07:21.29 | Makenshi | its a nice gravy train |
| 07:21.34 | odie_flocon_ | hahaha |
| 07:22.07 | odie_flocon_ | we used to charge that for a PRI - SS7 Conversion box. |
| 07:22.17 | McUnixJr | http://www.flickr.com/photos/pmtorrone/10572600/ |
| 07:23.08 | predictive | and somtimes they are arcane because they want to protect thier ricebowls |
| 07:23.10 | teamjet | what's the difference between a dialogic card and a asterisk card? |
| 07:23.22 | predictive | like everyone in teh world but ATT talks normal stuff on the serial port |
| 07:23.42 | Makenshi | and in the uk we have this bt dpnss crap |
| 07:23.51 | predictive | but they have to use some godawful protocol that like only procomm plus talks |
| 07:24.31 | McUnixJr | what provider should i look into getting for my new asterisk system? |
| 07:24.41 | predictive | iono |
| 07:24.45 | teamjet | i think telcom has over-obscure their tools in some ways |
| 07:24.58 | predictive | oh yeah |
| 07:25.17 | predictive | it allows grumpy old men to retain their jobs |
| 07:25.26 | teamjet | huh |
| 07:25.35 | predictive | 'the phone guy' |
| 07:25.40 | teamjet | or to allow the industry to implode once all these old men retires |
| 07:26.50 | teamjet | so what's a telephony card? i hear about dialogic cards a lot |
| 07:28.48 | *** part/#asterisk teamjet (~teamjet@lfc.tor.istop.com) |
| 07:28.52 | *** join/#asterisk teamjet (~teamjet@lfc.tor.istop.com) |
| 07:31.10 | Nethab | is that why a Fluke is so dang expensive |
| 07:31.35 | *** join/#asterisk usam (~usam@203.156.41.62) |
| 07:32.12 | McUnixJr | so should i use Broadvoice for inbound (at $6 per month) and then use my SunRocket (prepaid for one year of $199 unlimited calling in USA) for outbound calls ??? |
| 07:32.17 | usam | hello, nufone ppl here? Is your service down or something wrong with my confing? |
| 07:33.46 | Nethab | i use broadvoice for incoming, use up my 100 minutes of broadvoice outbound then automatically roll over to another provider |
| 07:34.05 | McUnixJr | nethab what provider do you roll over to? |
| 07:34.24 | Nethab | i have a prepaid account like you |
| 07:34.40 | Nethab | voipjet, but some people don't like them |
| 07:35.18 | Nethab | I also have a prepaid net2phone account that came free with my linksys, so i can roll over to that too |
| 07:35.36 | Nethab | all in asterisk |
| 07:37.09 | McUnixJr | very cool - thats what i am planning on doing (like that, sort of...) |
| 07:38.14 | McUnixJr | i am merging the POTS and 3 VoIP providers, then i want my new SIP phones (yet to be purchased) when they dial local, have it go out on normal POTS, for out of area calls go out on SunRocket, and out of country to eruope go out on Packet8 |
| 07:40.50 | Nethab | that should work easy, i have it keeping track of how many minutes i use and dial out a different provider after one runs out |
| 07:41.01 | *** join/#asterisk rkioko (~rkioko@196.200.25.253) |
| 07:41.32 | rkioko | hi |
| 07:41.37 | Nethab | hello |
| 07:41.49 | McUnixJr | can SIP phones have unique rings for the different incoming numbers? |
| 07:41.57 | rkioko | i have asterisk on public IP address |
| 07:42.12 | Nethab | depends on the soft phone, it's an internal function |
| 07:42.17 | Nethab | or hard phone |
| 07:42.18 | rkioko | and my SIP clients on private addressing - no NAT |
| 07:42.26 | rkioko | the asterisk box is multihomed |
| 07:42.41 | rkioko | calls between the SIP clients are OK |
| 07:42.49 | McUnixJr | would the SIP know where the call was inbound to? (meaning what phone number it was destined for?) |
| 07:42.52 | Nethab | is asterisk bridging two subnets |
| 07:43.01 | rkioko | yes |
| 07:43.11 | rkioko | but when i call out, i get one way audio |
| 07:43.15 | Nethab | yes the incoming sip url is sent to the phone |
| 07:43.33 | Nethab | rkioko, that's a common nat issue |
| 07:43.51 | rkioko | as soon as the call is picked, asterisk trues to bridge the private and public destination |
| 07:43.51 | Nethab | is asterisk also your firewall |
| 07:44.00 | rkioko | no firewalling |
| 07:44.02 | rkioko | no NAT |
| 07:44.18 | Nethab | do you have canreinvite = no for both sip clients |
| 07:44.26 | rkioko | yes |
| 07:44.48 | Nethab | and it's your sip phones able to hear the call not the outside person correct? |
| 07:45.03 | rkioko | shouldn't asterisk behave as a media proxy? |
| 07:45.14 | Nethab | yes it should |
| 07:45.23 | rkioko | the outside person hears but the sip on provate doesn't |
| 07:45.34 | Nethab | oh really, that's opposite of normal |
| 07:45.39 | *** join/#asterisk wildcard0 (~generic@S0106006097e16040.vc.shawcable.net) |
| 07:45.44 | rkioko | hehe |
| 07:46.17 | Nethab | then the outside person is trying to send to the internal private IP of the sip phone |
| 07:46.30 | rkioko | he shouldn't |
| 07:46.32 | Nethab | but since it's not public and inaccessible it can't |
| 07:46.38 | wildcard0 | hey. im building an agi that a .call file attaches to. it runs the agi before the phone answers the call. is there any way i can change this? using deadagi maybe? or detect a connection? |
| 07:47.08 | rkioko | because the call should come through asterisk and flow to the client |
| 07:47.12 | Nethab | does your outside peer entry have canreinvite = no |
| 07:47.20 | usam | Can someone comfirms me if nufone is still working? I always get circuit-busy :( |
| 07:47.34 | rkioko | it's out of my control, i don't know about that |
| 07:48.18 | Nethab | the other thing to check is if the incoming RTP ports are open to your asterisk server from the outside |
| 07:48.52 | Nethab | sip transmits control data on 5060/udp and voice data on 10000/udp - 20000/udp |
| 07:49.04 | Nethab | both can be controlled in rtp.conf |
| 07:49.46 | *** join/#asterisk covet (michael@146.145.209.138) |
| 07:49.49 | rkioko | they are |
| 07:50.29 | rkioko | the problem is that asterisk attempts the native bridge btwn the 2 devices |
| 07:50.41 | rkioko | and i is on private |
| 07:50.45 | *** part/#asterisk teamjet (~teamjet@lfc.tor.istop.com) |
| 07:51.02 | rkioko | how can i stop asterisk from - attempting native bridge |
| 07:51.19 | rkioko | btwn the 2 sip devices |
| 07:51.52 | rkioko | and have the media stream from end user - asterisk - client and not end user - client direct |
| 07:53.16 | Makenshi | rkioko, you need to configure your sip clients to use an rtp proxy |
| 07:54.05 | Nethab | isn't asterisk able to proxy rtp |
| 07:54.17 | Nethab | it can translate from gsm to g711 |
| 07:56.12 | Makenshi | Nethab, yes it is |
| 07:56.42 | Nethab | so other than canreinvite = yes what else needs to be set in sipl.conf |
| 07:56.49 | Nethab | er no |
| 07:58.46 | psycodad | anbyody successfully compiled asterisk on OpenBSD ? |
| 08:00.34 | facek_ | how to make transfer (blind transfer, normally by #) using manager ? |
| 08:01.07 | Luke-Jr | Is it possible to make a SIP call to a Vonage customer? |
| 08:01.13 | McUnixJr | is there a complete RPM of AMP anywhere? |
| 08:01.37 | Nethab | Action: Redirect |
| 08:02.02 | facek_ | Nethab thanks |
| 08:02.04 | Nethab | it redirects a channel to a new dialplan extension |
| 08:02.18 | facek_ | Nethab but is a normaln blind transfer? |
| 08:02.55 | Nethab | yes, it is automatically sent to that dialplan entry, no questions asked and if it's Dial then it's completely blind |
| 08:03.10 | rkioko | isn't asterisk supposed to proxy rtp as well? |
| 08:03.34 | Nethab | yes |
| 08:05.24 | facek_ | Nethab thanks |
| 08:05.31 | Nethab | np |
| 08:09.39 | Nethab | I search the mailing list, and it says to put canreinvite = no in all your peer and friend entries |
| 08:10.06 | Nethab | it's a common private IP to outside public IP problem |
| 08:10.13 | rkioko | canreinvite=no |
| 08:10.22 | Nethab | yes in all the peers and friends |
| 08:10.27 | kb1_kanobe | asterisk on 2.6.x - is this a good idea? |
| 08:10.33 | Nethab | yes, I run it |
| 08:10.55 | Nethab | but i don't use PCI cards for FXS or FXO ports |
| 08:10.57 | kb1_kanobe | how much load? |
| 08:11.25 | Nethab | cpu load? |
| 08:11.33 | kb1_kanobe | channels/codecs etc. |
| 08:12.11 | Nethab | i don't have more than 5 to 10 calls at any given time |
| 08:12.16 | kb1_kanobe | ok. |
| 08:12.32 | Nethab | and 90% aren't translated |
| 08:13.05 | Nethab | i have a p3 450 and never get about 0.3 load average |
| 08:14.14 | Nethab | although i had to get rid of mpg123 cause it sucked the system down |
| 08:14.15 | wildcard0 | so...im building an agi that a .call file attaches to. it runs the agi before the phone answers the call. is there any way i can change this? using deadagi maybe? or detect a connection? i've tried changing it to an extension instead of calling the agi application directly and it does the same thing |
| 08:14.17 | peter222 | <PROTECTED> |
| 08:14.17 | wildcard0 | any ideas? |
| 08:15.11 | Nethab | you want the agi to run only after the call? |
| 08:15.11 | *** join/#asterisk amir_ (~amir@195.226.9.186) |
| 08:15.21 | Trickyphillips | How can I play the "hold" music while there is more than one person in a teleconference? |
| 08:15.23 | wildcard0 | after the call completes |
| 08:15.36 | wildcard0 | basically, after the phone is answered |
| 08:15.44 | wildcard0 | not after it's hung up |
| 08:16.02 | Nethab | i think there's a fork command |
| 08:16.15 | Nethab | to dial and continue in the dialplan |
| 08:16.27 | Nethab | at the same time |
| 08:16.35 | wildcard0 | hmm. im not sure how that would help |
| 08:16.43 | wildcard0 | maybe im not explaining the problem correctly |
| 08:17.12 | Nethab | is the call triggering the agi or the agi triggering the call |
| 08:17.35 | wildcard0 | it's a .call file. as per http://www.voip-info.org/wiki-Asterisk+auto-dial+out |
| 08:17.43 | wildcard0 | i put it in the outgoing directory |
| 08:17.57 | wildcard0 | /var/spool/asterisk/outgoing |
| 08:19.10 | wildcard0 | basically the "Channel" variable points to my cell phone right now. the "Extension" variable points to an extension that runs an agi (i've also tried running the "Applications: AGI" directly...does the same thing) |
| 08:19.28 | Nethab | in the extension section to connect to after the call connects you put the agi call there |
| 08:19.48 | Nethab | "If the call answers, connect it here" |
| 08:19.53 | *** join/#asterisk TheEmperor (~user@203.114.48.47) |
| 08:20.01 | wildcard0 | yes, i thought that was how it was supposed to work |
| 08:20.19 | wildcard0 | but...it's not doing that. it's running the AGI before the channel answers |
| 08:20.40 | Nethab | ah, the channel is connected though, they just haven't picked up the phone |
| 08:20.55 | Nethab | so asterisk runs the agi anyway |
| 08:21.20 | wildcard0 | yes |
| 08:21.49 | wildcard0 | the console says it has answered...but it hasn't. maybe im unclear as to what asterisk considers 'answered' |
| 08:22.07 | Nethab | waittime |
| 08:22.19 | Nethab | the example farther down uses waittime |
| 08:22.47 | wildcard0 | isn't that how long it rings the phone before giving up? |
| 08:22.47 | wildcard0 | i have it set to 30 currently |
| 08:23.47 | Nethab | have the extension called wait(2) or something? |
| 08:23.54 | wildcard0 | i've tried it both on an iax channel and a zaptel card |
| 08:24.02 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
| 08:24.16 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
| 08:24.20 | wildcard0 | i could do that, but it's not a great solution. it assume that it takes (X) seconds for someone to answer |
| 08:24.32 | Nethab | or have the extension check the channel status for answered |
| 08:24.44 | Nethab | you'd have to use variables |
| 08:25.14 | Nethab | does the AGI have access to channel variables |
| 08:25.24 | Nethab | you could check it from in the agi |
| 08:25.49 | Nethab | wait for State Up and wait while State Ring or Ringing |
| 08:25.56 | wildcard0 | i don't know. does agi usually have access to channel variables? |
| 08:26.03 | Nethab | it should |
| 08:26.28 | wildcard0 | ah, channel status |
| 08:26.35 | wildcard0 | thanks. this will give me something else to try |
| 08:27.19 | Nethab | there's a get variable command |
| 08:27.26 | Nethab | it gets channel variables |
| 08:27.27 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
| 08:27.35 | wildcard0 | http://www.voip-info.org/wiki-channel+status |
| 08:27.40 | wildcard0 | that's what im looking at now |
| 08:28.52 | *** join/#asterisk cc (~cc@byte.fedora) |
| 08:28.57 | wildcard0 | just have to find the channel name |
| 08:29.15 | Nethab | if no channel name is fiven |
| 08:29.20 | Nethab | it uses the current channel |
| 08:29.24 | wildcard0 | ah yes |
| 08:29.52 | wildcard0 | hehe |
| 08:29.52 | *** join/#asterisk Delvar (~irc@83.146.53.34) |
| 08:29.57 | Nethab | you'll want to check for up |
| 08:55.49 | wildcard0 | hmm. it seems asterisk thinks the channel is up |
| 08:56.00 | wildcard0 | im going to try the latest version. maybe something got fixed |
| 09:00.39 | *** join/#asterisk nrc (~username@zeus.eurotux.com) |
| 09:00.50 | nrc | hi all |
| 09:04.23 | *** join/#asterisk takezo (~bob@191-49.244.81.adsl.skynet.be) |
| 09:04.27 | *** part/#asterisk takezo (~bob@191-49.244.81.adsl.skynet.be) |
| 09:06.48 | *** join/#asterisk zeedo (~zeedo@www.bsrf.org.uk) |
| 09:11.59 | *** join/#asterisk RoyK (~roy@80.239.107.80) |
| 09:15.31 | *** join/#asterisk meppl (mephisto@pD9E68A93.dip.t-dialin.net) |
| 09:18.02 | facek_ | hm |
| 09:18.11 | *** join/#asterisk lehel (~lehel@82.79.20.17) |
| 09:18.27 | lehel | good day |
| 09:22.04 | facek_ | little |
| 09:22.10 | *** join/#asterisk Blackvel (~blackvel@dsl-213-023-034-170.arcor-ip.net) |
| 09:22.37 | lehel | how can i verify if my asterisk is installed? with zaptel.. ?? |
| 09:22.39 | Blackvel | is there any Manager API proxy written in Java to pool authentication and commands/events? |
| 09:23.08 | lehel | and it's working |
| 09:23.40 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
| 09:23.48 | kb1_kanobe | g'day all. |
| 09:24.54 | kb1_kanobe | I've got four virtually identical * hosts, each running one t100p card. One of them is generating a fair amount of 'chan_zap.c:4395 my_zt_write: Write returned -1 (Resource temporarily unavailable) on channel 23' - should I suspect the t100p card as faulty? I've tried swapping cables, also changing to a 2.4.29/lowlat/preempt based kernel. |
| 09:25.23 | *** join/#asterisk basta (~kqj@62-101-126-233.fastres.net) |
| 09:26.19 | lehel | ask help |
| 09:26.26 | basta | help ! * just stopped working, if i try to restart it it says "chan_capi.c:2636 load_module: CAPI not installed!" |
| 09:26.37 | zoa | capi is not loaded basta |
| 09:27.11 | *** join/#asterisk emitrax (~dido@ingnatdyn149.unime.it) |
| 09:27.29 | basta | yes, but it worked since 20' ago, and I have not touched anything ! |
| 09:28.32 | basta | lsmod | grep capi -> shows isdn, kernelcapi etc... |
| 09:28.32 | emitrax | did anybody try to place a call with x-lite to a cisco 7940 phone with SIP firmware ? |
| 09:28.41 | zoa | yes |
| 09:28.44 | zoa | it works |
| 09:29.47 | Blackvel | try to reload |
| 09:29.49 | Blackvel | dmesg |
| 09:29.49 | facek_ | how to make a transfer - lbind transfer - like using #, but using manager api? |
| 09:29.52 | Blackvel | maybe something crashed |
| 09:32.28 | basta | yes, isdn restart and it works ... |
| 09:32.28 | basta | maybe 200 days without a restart is too much even for linux sometimes ;P |
| 09:32.32 | *** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
| 09:35.23 | *** join/#asterisk pif (ldm@zenon.apartia.fr) |
| 09:36.42 | *** join/#asterisk darby_t (~tom@host-ip226-209.crowley.pl) |
| 09:40.26 | lehel | how do i know what kind of card is installed in my comp? |
| 09:40.40 | *** join/#asterisk n4y (~frodo7@host-ip226-209.crowley.pl) |
| 09:41.00 | lehel | i didn't find anything in the /var/log/messages |
| 09:41.28 | *** join/#asterisk fenlander (~neils@82.152.81.57) |
| 09:41.39 | lehel | ?? |
| 09:43.57 | *** join/#asterisk Manipura (~chatzilla@dsl-ep-209-115-250-i114-cgy.nucleus.com) |
| 09:44.29 | *** join/#asterisk christo (~chris@office.enovi.com) |
| 09:44.31 | christo | hi guys |
| 09:45.06 | lehel | somebody.. i need asterisk information |
| 09:45.23 | inspired | lehel: lspci ? |
| 09:45.38 | zoa | lehel |
| 09:45.39 | zoa | lspci |
| 09:46.08 | fenlander | Does anyone know any details about Asterisk Business Edition from Digium? |
| 09:46.15 | kajtzu | call digium? :) |
| 09:46.22 | fenlander | :) |
| 09:46.24 | lehel | don't understood "ispci"? |
| 09:46.32 | inspired | LSPCI |
| 09:46.36 | inspired | in lower case |
| 09:46.38 | lehel | i have to configure asterisk |
| 09:46.49 | inspired | you type lspci in your console |
| 09:46.52 | inspired | linux console |
| 09:46.54 | lehel | zaptel |
| 09:46.55 | lehel | k |
| 09:47.04 | inspired | then you'll se info about your card |
| 09:47.28 | inspired | see* |
| 09:53.38 | lehel | and now i download the Zaptel software package from the Asterisk CVS |
| 09:54.11 | lehel | correct? |
| 09:56.08 | *** join/#asterisk fantomax1 (~fanto@81.208.114.250) |
| 09:57.04 | tzafrir | lehel, what distro do you use? |
| 09:57.27 | fantomax1 | hi all |
| 09:57.48 | fantomax1 | i got an error that say : Warning:unable to write frame |
| 09:57.54 | fantomax1 | what does it mean ? |
| 09:58.03 | fantomax1 | how can I solve it ? |
| 09:58.05 | fantomax1 | thanks |
| 09:58.05 | *** join/#asterisk isamar (~isamar@p8131-ipadfx21sasajima.aichi.ocn.ne.jp) |
| 09:58.08 | isamar | hi folks |
| 09:58.11 | Mavvie | it's a warning. |
| 09:58.15 | kb1_kanobe | fantomax1 What is the complete error? |
| 09:58.16 | isamar | anybody using sangoma here |
| 09:58.17 | Mavvie | most likely because the socket was closed. |
| 09:58.18 | isamar | ? |
| 09:58.27 | fantomax1 | wait |
| 09:58.44 | fantomax1 | WARNING[20524]: Failed to write frame |
| 09:58.45 | isamar | having problems setting up zaptel.conf and zapata.conf for it.... |
| 09:58.50 | fantomax1 | this is teh error |
| 09:58.57 | lehel | tzafrir: "distro" ? |
| 09:58.57 | Mavvie | error |
| 09:59.04 | Mavvie | warning! |
| 09:59.05 | fantomax1 | warning |
| 09:59.06 | tzafrir | lehel, linux distribution |
| 09:59.07 | fantomax1 | yes |
| 09:59.13 | lehel | Debian |
| 09:59.17 | fantomax1 | do I have to worry about it ? |
| 09:59.17 | lehel | why? |
| 09:59.23 | tzafrir | Stable? |
| 09:59.33 | lehel | yes.. i think |
| 09:59.45 | isamar | n1 using Sangoma boards with * ? |
| 09:59.47 | lehel | how can i found out if it is stable or not? |
| 10:00.20 | tzafrir | woody/Stable/3.0? or Sarge/Testing/3.1, or Sid/Unstable? |
| 10:00.52 | kb1_kanobe | isamar: cat /etc/debian_version |
| 10:02.22 | isamar | no debian here.. |
| 10:02.25 | isamar | slackware |
| 10:02.40 | isamar | drivers installed.. everything compiled.. |
| 10:03.06 | isamar | I just need to know what to put in zaptel.conf and zapata.conf for Sagoma A101 card... |
| 10:03.07 | isamar | just it |
| 10:03.09 | lehel | tzafrir: 3.1 Sarge |
| 10:03.31 | tzafrir | lehel, as it happens there are some nice * packages for Sarge. |
| 10:03.51 | tzafrir | apt-get install module-assitant zaptel-source |
| 10:04.55 | lehel | module-assistant is already the newest version. |
| 10:04.55 | lehel | zaptel-source is already the newest version. |
| 10:04.55 | lehel | 0 upgraded, 0 newly installed, 0 to remove and 0 not upgraded. |
| 10:05.31 | kb1_kanobe | isamar: sorry! Should look at the screen when I'm typing... |
| 10:05.43 | tzafrir | now, as a user: |
| 10:06.01 | tzafrir | m-a -u . -t build zaptel |
| 10:06.29 | *** join/#asterisk nextime (~nextime@213-140-6-96.fastres.net) |
| 10:06.36 | lehel | "m-a -u . -t biuld zaptel" <correct ? |
| 10:06.46 | tzafrir | Actually, if you're prepared to build the package as root, it will even install the source packages, and will install the generated deb |
| 10:06.51 | tzafrir | m-a -u . -t build zaptel |
| 10:07.02 | tzafrir | b*ui*ld |
| 10:07.11 | lehel | sorry, i'm root |
| 10:07.26 | lehel | is this a problem? |
| 10:07.37 | tzafrir | not much |
| 10:08.50 | lehel | Bad luck, the kernel headers for the target kernel version could not be |
| 10:08.50 | lehel | found and you did not specify other valid kernel headers to use. |
| 10:08.50 | lehel | it's not working |
| 10:09.08 | lehel | apt-get install kernel-headers-2.4.27-2-686-smp ? |
| 10:10.14 | lehel | tzafrir: what do u think? |
| 10:11.15 | tzafrir | is that the kernel you use? |
| 10:11.18 | tzafrir | uname -r |
| 10:11.50 | lehel | 2.4.27-2-686-smp |
| 10:11.54 | lehel | yeah |
| 10:12.08 | lehel | The following NEW packages will be installed: |
| 10:12.08 | lehel | <PROTECTED> |
| 10:12.09 | lehel | 0 upgraded, 1 newly installed, 0 to remove and 0 not upgraded. |
| 10:12.24 | tzafrir | lehel, if you're root, you can try let module-assistant do everything: |
| 10:12.28 | lehel | now i try again build zaptel |
| 10:12.31 | tzafrir | m-a a-i zaptel |
| 10:13.01 | Manipura | ~seen cybermage |
| 10:13.02 | jbot | cybermage <~CyberMage@user-24-236-84-79.knology.net> was last seen on IRC in channel #asterisk, 382d 17h 42m 21s ago, saying: 'biot: Thanks'. |
| 10:13.06 | lehel | biulding.. step 1 |
| 10:13.12 | Manipura | ~seen cybrmage |
| 10:13.13 | jbot | i haven't seen 'cybrmage', Manipura |
| 10:13.41 | lehel | i should login as root? |
| 10:13.53 | lehel | Setting up zaptel-modules-2.4.27-2-686-smp (1.0.7-3+2.4.27-8) ... |
| 10:14.00 | lehel | it's ok |
| 10:14.04 | tzafrir | lehel, generally the idea is to do the least things as root. |
| 10:14.35 | lehel | ok.. how ami continuing now? |
| 10:14.51 | tzafrir | becausr with root it is easier to inflict damage on your system. The prime example is rm -rf / (don't try this at /home) |
| 10:15.17 | lehel | i need to re-compile Asterisk ?? |
| 10:15.34 | tzafrir | lehel, I figure it has actually installed you zaptel modules. |
| 10:15.52 | lehel | "cvs checkout zaptel"..? |
| 10:15.54 | tzafrir | what version of asterisk you use? you can basically simply 'apt-get install asterisk' |
| 10:16.33 | lehel | asterisk is already the newest version. |
| 10:16.51 | tzafrir | one thing: run "groups asterisk" |
| 10:16.56 | lehel | how can i find out the verison? |
| 10:17.02 | tzafrir | does it give "dialout" as well? |
| 10:17.30 | lehel | asterisk : asterisk dialout audio |
| 10:17.53 | tzafrir | goody. so it won't have problem writing the zaptel devices |
| 10:18.21 | lehel | i'm happy.. next step? |
| 10:18.47 | tzafrir | now, for slightly easier configuration: http://tzafrir.org.il/genzaptelconf <-- wget it |
| 10:19.06 | tzafrir | A little config script of mine |
| 10:19.22 | tzafrir | still not in the debian package. |
| 10:19.55 | inspired | no ncurses? :P |
| 10:20.01 | tzafrir | to simulate the hardware detection at startup: |
| 10:20.12 | tzafrir | /etc/init.d/discover restart |
| 10:20.12 | lehel | wget |
| 10:20.24 | tzafrir | (very noisy, but should pick up TDM cards) |
| 10:20.50 | tzafrir | wget http://tzafrir.org.il/genzaptelconf |
| 10:21.07 | lehel | wget http://tzafrir.org.il/genzaptelconf |
| 10:21.13 | lehel | k |
| 10:21.40 | key2 | hey |
| 10:21.45 | key2 | had a question |
| 10:21.57 | key2 | the asterisk .conf file, what script language is it ? |
| 10:22.00 | key2 | specific ? |
| 10:22.00 | lehel | genzaptelconf' saved [14393/14393] |
| 10:22.23 | predictive | key2 just asked a very deep question, heh |
| 10:22.34 | key2 | predictive: sorry |
| 10:22.36 | key2 | dunno :) |
| 10:22.43 | predictive | it's pretty much an ini file |
| 10:22.49 | lehel | tzafrir: how can i stimulate the hardware detection ? |
| 10:22.50 | key2 | ok |
| 10:22.58 | tzafrir | key2, no special script. Asterisk config files are generally sections of key=value (but a bit more complicated) |
| 10:22.59 | key2 | predictive: where are the spec ? |
| 10:23.07 | tzafrir | /etc/init.d/discover restart |
| 10:23.21 | key2 | k |
| 10:23.27 | tzafrir | key2, what do you need it for? |
| 10:23.51 | tzafrir | Generally the parser generates a list of values. |
| 10:24.07 | key2 | tzafrir: I need to set a SIP gateway for Voip --> Telephone line using my FXO card, is it easy |
| 10:24.07 | key2 | ? |
| 10:24.18 | predictive | I say it's deep because it's an eternal discussion in programming circles about how all configuration files are little languages |
| 10:24.23 | Blackvel | how many manager api concurrent connections can you have open? 60? 120? |
| 10:24.28 | predictive | and wouldn't that really be better served by using a real language |
| 10:24.33 | predictive | you get going in circles talking about it |
| 10:25.18 | key2 | tzafrir: is it hard to do ? So when I call from a SIP softphone using IP, it uses the FXO card to place the call |
| 10:25.39 | lehel | tzafrir: Detecting hardware: i810-tco sk98lin piix ide-scsi i810_audio usb-uhci ehci-hcd hisax |
| 10:25.39 | lehel | i810-tco disabled in configuration. |
| 10:25.47 | tzafrir | key2, why do you want to re-implement the config parser? |
| 10:26.04 | key2 | tzafrir: is there somethine already doing that ? |
| 10:26.04 | three55ml | key2: It's not hard at all |
| 10:26.16 | three55ml | key2: You'll find a lot of examples of that at http://www.voip-info.org |
| 10:26.25 | tzafrir | lehel, does 'modinfo zaptel' give any output? (don't paste it here, please) |
| 10:26.38 | key2 | three55ml: what shouls I look for |
| 10:26.49 | key2 | any specific words ? |
| 10:26.51 | three55ml | key2: There's a section full of example configs |
| 10:26.55 | three55ml | key2: Let me find a link |
| 10:27.14 | lehel | yes |
| 10:27.21 | key2 | thx |
| 10:27.29 | predictive | so I vote for python |
| 10:27.45 | tzafrir | predictive, then go and implement it |
| 10:28.04 | lehel | what could be the problem tzafrir ? |
| 10:28.06 | tzafrir | predictive, on a second thought, why not just add your code to DeStar? |
| 10:28.15 | predictive | what's DeStar? |
| 10:28.22 | three55ml | key2: http://www.voip-info.org/wiki-Asterisk+Configuration+Examples |
| 10:28.24 | key2 | predictive: why not perl |
| 10:28.30 | predictive | oh, I just like python |
| 10:28.36 | key2 | predictive: at least it looks more like something |
| 10:29.03 | predictive | everyone has favorites - I know maybe 12 languages well enough to write in, they all have nice things about them |
| 10:29.06 | fenlander | Can anyone recommend an FXO that is approved for the UK market? |
| 10:29.10 | tzafrir | any programming language as a configuration is a nightmare for a parser |
| 10:29.17 | lehel | tzafrir: u could help me out pls? |
| 10:29.22 | key2 | three55ml: thanks but which one |
| 10:29.28 | samueltc | hi |
| 10:29.29 | tzafrir | lehel, try: depmod -a |
| 10:29.41 | predictive | tzafrir: I think part of the point was that the parser already exists |
| 10:29.48 | lehel | than? |
| 10:29.48 | key2 | fenlander: Wildcard x100P |
| 10:29.51 | samueltc | i'm trying to add a zonedata (tunisia) to zaptel |
| 10:29.53 | three55ml | key2: You'll have to meld some of them together |
| 10:29.54 | samueltc | but when I load wcfxs I got a Invalid 'next' pointer and ioctl(ZT_LOADZONE) failed: Invalid argument |
| 10:30.05 | three55ml | key2: You'll setup the X100P in zapata.conf, then the rest in extensions.conf |
| 10:30.16 | fenlander | key2: they don't make them anymore - only clones |
| 10:30.25 | lehel | tzafrir: than? |
| 10:30.31 | key2 | fenlander: then clone! |
| 10:30.38 | lehel | then? |
| 10:31.06 | tzafrir | hmm, oops. modinfo zaptel did produce output ealier, right? |
| 10:31.08 | key2 | lehel: it means :then, get yourself the clone... |
| 10:31.29 | tzafrir | lehel, is there /proc/zaptel ? |
| 10:31.37 | tzafrir | ls /proc/zaptel |
| 10:31.57 | lehel | cd: /proc/zaptel: No such file or directory |
| 10:32.16 | tzafrir | predictive, http://www.voip-info.org/wiki-Asterisk+GUI+DeStar |
| 10:32.23 | fenlander | key2: just seeing if there was an "approved" route to an FXO - there are many mixed reports of clones |
| 10:32.50 | tzafrir | let's get genzaptelconf to do everything |
| 10:32.52 | key2 | fenlander: go to www.ebay.co.uk |
| 10:32.54 | *** join/#asterisk c_k (~ck@82-43-178-166.cable.ubr06.newm.blueyonder.co.uk) |
| 10:32.55 | key2 | and find one |
| 10:33.11 | lehel | ok tzafrir |
| 10:33.18 | c_k | Hi all, just a quick question is there any documentation on the installation and inital setup of asterisk at all? |
| 10:33.37 | three55ml | All of the clones are identical to the original, as far as I know. It's a very common modem chipset that's used. |
| 10:33.39 | key2 | well, if it uses the ambiant chipset, it should be ok |
| 10:33.40 | tzafrir | bash genzaptelconf -vsd |
| 10:34.57 | predictive | tzafrir: interesting |
| 10:35.08 | lehel | tzafrir: Asterisk not yet configured. Edit /etc/default/asterisk first. |
| 10:35.36 | tzafrir | lehel, edit that file. It is plain English |
| 10:35.56 | c_k | yeah what do you have to edit though I am getting the same error |
| 10:35.59 | Blackvel | how many manager api concurrent connections can you have open? 60? 120? |
| 10:36.00 | c_k | hense asking for docs |
| 10:36.39 | tzafrir | Blackvel, why would there be a limit? |
| 10:37.13 | Blackvel | according to docs it is quite unstable, deadlocks, etc. |
| 10:37.35 | three55ml | tzafrir - Mine is prettier :) Not opensource though. |
| 10:37.42 | three55ml | tzafrir - http://www.premierpbx.com/misc/server_appliance_1.jpg |
| 10:37.48 | predictive | heh I like the choice of words about amp: "deficiently organized." |
| 10:37.57 | predictive | that's an interesting turn of phrase |
| 10:38.45 | lehel | tzafrir: errors ;( |
| 10:38.48 | tzafrir | three55ml, it is not free software. Which means someone is able to pull the plug from a software I ppotentially use. Not for me. |
| 10:38.59 | c_k | lehel: you get that fixed yet at all |
| 10:39.12 | tzafrir | lehel, the errors are from modprobing a number of modules. |
| 10:39.15 | three55ml | Blackvel: It will vary heavily on what you're trying to do. What will the connections be doing? |
| 10:39.27 | lehel | 0 channels configured |
| 10:39.33 | tzafrir | however: asterisk -rx 'zap show channels' |
| 10:39.33 | lehel | genzaptelconf: line 556: ast-cmd: command not found |
| 10:40.01 | tzafrir | ignore that. It is used to report status in the end. |
| 10:40.07 | Blackvel | three55ml: for AGI channels calling redirect/hangup, for outgoing part originate, play file, etc. |
| 10:40.15 | tzafrir | it is a wrapper around asterisk -rx |
| 10:40.27 | three55ml | Blackvel: Are you talking about AGIs? I thought you said manager connections? |
| 10:40.49 | three55ml | Blackvel: Or controlling AGI channels through the manager? |
| 10:41.01 | lehel | tzafrir: asterisk: option requires an argument -- x |
| 10:41.30 | c_k | is there a web based config tool that can be used for asterisk at all? |
| 10:41.34 | c_k | or at least some docs |
| 10:41.47 | tzafrir | asterisk -rx 'zap show channels' |
| 10:42.10 | three55ml | c_k: The most widely used one is AMP, but you either love it or hate it. Search for it at http://www.voip-info.org |
| 10:42.12 | three55ml | ~AMP |
| 10:42.13 | jbot | i guess amp is an Audio MPEG Player. [non-free], or http://amp.coalescentsystems.ca/ |
| 10:42.17 | lehel | tzafrir: Unable to connect to remote asterisk |
| 10:42.27 | three55ml | There's a link to the site |
| 10:42.29 | tzafrir | c_k, I think DeStar is now getting usable |
| 10:42.47 | c_k | is that a web based app then |
| 10:42.53 | three55ml | c_k: Yes |
| 10:43.10 | tzafrir | lehel, asterisk -U asterisk -vvvvc |
| 10:43.21 | c_k | nice thanks |
| 10:43.25 | tzafrir | to see what's the problem |
| 10:43.37 | *** join/#asterisk cmk (~cmk_@p54A3D3EE.dip.t-dialin.net) |
| 10:44.31 | tzafrir | BTW: genzaptelconf writes its zapata.conf stuff to /etc/asterisk/zapata-channels.conf |
| 10:45.19 | lehel | tzafrir: pasted in query the problem |
| 10:45.21 | tzafrir | add in the end of your zapata.conf: #include "zapata-channels.conf" |
| 10:45.51 | tzafrir | Logger Warning: Unable to open log file '/var/log/asterisk/messages': Permission denied |
| 10:46.12 | tzafrir | This is the result of originally running Asterisk as root |
| 10:46.44 | tzafrir | chown asterisk /var/log/asterisk/* |
| 10:47.28 | lehel | i should "unroot"? .. i'm "su" ing |
| 10:47.28 | lehel | sorry, where is the zapata.conf file? |
| 10:47.39 | tzafrir | add in the end of your zapata.conf: #include "zapata-channels.conf" |
| 10:48.16 | tzafrir | hmmm, does /etc/default/asterisk mentions running as root? |
| 10:49.03 | tzafrir | lehel, no. Asterisk should be executed as root. This is needed for a number of things. Asterisk setuid-s on its own to the user asterisk |
| 10:49.42 | lehel | i'm not logged as root |
| 10:49.53 | lehel | i should login as root? |
| 10:49.59 | *** join/#asterisk gres (~gres@81.222.48.242) |
| 10:50.12 | *** join/#asterisk knight_ (~knight@blackhole.phunc.com) |
| 10:50.17 | lehel | sorry, where is the zapata.conf file? |
| 10:51.35 | *** join/#asterisk knight_ (~knight@blackhole.phunc.com) |
| 10:51.55 | lehel | tzafrir: u hate me, isn't?;) |
| 10:52.10 | tzafrir | /etc/asterisk/zapata.conf |
| 10:52.13 | lehel | i'm a kind of beginner using debian.. |
| 10:52.18 | knight_ | heh |
| 10:52.18 | lehel | k |
| 10:52.29 | ManxPower | http://www.theregister.co.uk/2005/04/28/exploding_german_toads/ |
| 10:52.39 | tzafrir | lehel, actually, I was about to recommend you on our custom debian-based distro (from which genzaptelconf is taken) |
| 10:52.48 | tzafrir | http://www.xorcom.com/ |
| 10:53.09 | c_k | mmm getting internal server error when running destar any ideas anyone , cant find anything in the logs either |
| 10:53.26 | *** join/#asterisk NirS (~NirS@192.118.53.180) |
| 10:53.28 | NirS | hello all |
| 10:53.33 | tzafrir | c_k, in channel #destar, please |
| 10:53.33 | NirS | anybody home this fine morning ? |
| 10:53.40 | NirS | hey tzafrir |
| 10:53.40 | tzafrir | hi NirS |
| 10:53.42 | NirS | wassup ? |
| 10:53.43 | predictive | greetings |
| 10:53.59 | NirS | I added a remark on the forum to what you wrote :-) |
| 10:54.29 | lehel | tzafrir: added |
| 10:55.09 | tzafrir | I saw it |
| 10:55.10 | *** join/#asterisk olivier_ (~olivier_@obs92-4-82-239-116-113.fbx.proxad.net) |
| 10:55.34 | lehel | tzafrir: now what? |
| 10:55.58 | tzafrir | can you try re-running asterisk? |
| 10:56.34 | lehel | Unable to connect to remote asterisk |
| 10:56.59 | lehel | how can i re-run? |
| 10:57.15 | tzafrir | just like before |
| 10:57.33 | tzafrir | do you get the same error? |
| 10:57.59 | NirS | tzafrir, I just checked out a new CVS of asterisk, and I think I saw something wrong, but zapata.conf now has entries for SS7 ? |
| 10:57.59 | lehel | the same.. "...Permission denied" |
| 10:58.02 | NirS | did I see correctly ? |
| 10:59.31 | *** part/#asterisk knight_ (~knight@blackhole.phunc.com) |
| 10:59.40 | *** join/#asterisk knight_ (~knight@blackhole.phunc.com) |
| 11:00.33 | knight_ | heh |
| 11:05.02 | predictive | tzafrir: after reading, I think the difference is you're working to make it easy to configure for the novice |
| 11:05.20 | predictive | I was talking more about doing something like adding macros to teh config to make it more useful |
| 11:05.31 | predictive | two different problems |
| 11:07.33 | lehel | i still don't know how can i run asterisk.... |
| 11:08.10 | jakepdev | type "asterisk" |
| 11:08.23 | lehel | nothing happens |
| 11:08.32 | jakepdev | type asterisk -r |
| 11:08.44 | lehel | Unable to connect to remote asterisk |
| 11:09.02 | lehel | now what? |
| 11:09.39 | jakepdev | "asterisk" should run it |
| 11:09.50 | jakepdev | "asterisk - r" should connect to the running instance |
| 11:09.50 | lehel | i don't know.. maybe the card... don't know what to do;( |
| 11:09.51 | ManxPower | If asterisk -r does not work then either Asterisk is not running or you do not have permission to access /var/run/asterisk.ctl |
| 11:10.12 | ManxPower | lehel: try asterisk -cvvv |
| 11:10.12 | lehel | it is necesarry to be logged as root? |
| 11:10.15 | tzafrir | lehel, asterisk -gvvvv |
| 11:10.18 | ManxPower | lehel: YES! |
| 11:10.25 | lehel | ........... |
| 11:10.35 | ManxPower | You must be root to run/use asterisk unless you took the time to make everything run as non-root |
| 11:11.52 | lehel | the last 4 lines: |
| 11:11.54 | lehel | Apr 28 14:11:34 NOTICE[6709]: chan_capi.c:2635 load_module: CAPI not installed! |
| 11:11.54 | lehel | Apr 28 14:11:34 WARNING[6709]: loader.c:345 ast_load_resource: chan_capi.so: load_module failed, returning -1 |
| 11:11.55 | ManxPower | What is "Rapid"? |
| 11:11.55 | lehel | Apr 28 14:11:34 WARNING[6709]: chan_capi.c:2811 unload_module: Unable to unregister from CAPI! |
| 11:11.55 | lehel | <PROTECTED> |
| 11:11.56 | lehel | Apr 28 14:11:34 WARNING[6709]: loader.c:440 load_modules: Loading module chan_capi.so failed! |
| 11:12.04 | tzafrir | http://www.xorcom.com/ |
| 11:12.07 | RoyK | chan_crappy |
| 11:12.09 | ManxPower | lehel: NOW you are seeing the real error. |
| 11:12.19 | RoyK | ~pastebin |
| 11:12.20 | jbot | [pastebin] a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 11:12.25 | lehel | ok.. i have to install CAPI ? |
| 11:12.40 | ManxPower | lehel: Do you have ISDN BRI? |
| 11:12.43 | RoyK | lehel: if you want to use a CAPI device, well, it might help...... |
| 11:12.47 | tzafrir | lehel, what hardware do you have? |
| 11:13.52 | lehel | i don't know.. what is the command to find out? |
| 11:14.03 | ManxPower | "Rapid: Asterisk 1.0.2 on Debian" |
| 11:14.09 | jakepdev | Rapid looks like it might be a good alternative to @home |
| 11:14.26 | ManxPower | lehel: If you do not know what hardware and lines you have, you are FAR beyond any of our help |
| 11:14.41 | lehel | 0000:02:04.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface |
| 11:14.45 | lehel | this is it? |
| 11:14.58 | ManxPower | lehel: It might be. What card did you PURCHASE? |
| 11:15.21 | lehel | zaptel is an answer? |
| 11:15.38 | ManxPower | lehel: What card did you PURCHASE? I'm NOT asking a 3rd time. |
| 11:16.47 | tzafrir | lehel, this is a TDM card |
| 11:16.54 | tzafrir | this is not an ISDN card |
| 11:17.17 | jakepdev | would be great to know the model number |
| 11:17.29 | lehel | TDM 400 |
| 11:17.32 | ManxPower | jakepdev: he's not going to get very far unless he knows the model number |
| 11:17.34 | tzafrir | Who do we need to bug to fix that text? |
| 11:17.55 | ManxPower | lehel: Then you need to install zaptel and load the wcfxs (1.0.x) or wctdm (CVS-HEAD) driver. |
| 11:18.07 | *** join/#asterisk Skarmeth (~Skarmeth@201009023158.user.veloxzone.com.br) |
| 11:18.08 | ManxPower | tzafrir: the maintainer of the lspci program. |
| 11:18.59 | tzafrir | zaptel is already installed |
| 11:19.09 | ManxPower | lehel: you didn't read any docs, did you? |
| 11:19.15 | jakepdev | hehe |
| 11:19.29 | tzafrir | lehel, take a look at /proc/zaptel |
| 11:19.39 | lehel | Wildcard TDM400P |
| 11:19.52 | ManxPower | lehel: and do a "lsmod" to make sure you actually have the zaptel still running. |
| 11:20.31 | tzafrir | ManxPower, the module is loaded. Otherwise there would be no /proc entry |
| 11:20.34 | christo | does this look alright for getting an outside line when hitting '9' ? |
| 11:20.34 | christo | exten => _9XXXXX,1,Dial(Zap/g1/${EXTEN}:1) |
| 11:20.58 | jakepdev | if your outside line needs 5 digits |
| 11:21.13 | christo | jakepdev aah |
| 11:21.15 | tzafrir | lehel, does asterisk load? Did you sort out the chan_capi thing? |
| 11:21.15 | christo | bad expression |
| 11:21.17 | ManxPower | and if you have correctly put channels in group 1 |
| 11:21.39 | ManxPower | tzafrir: he just showed us that astersk does not load. |
| 11:21.55 | ManxPower | It's bombing out on chan_capi, but I don't even know why chan_capi is even being built. |
| 11:22.00 | *** join/#asterisk Druken (~druken@CPE00119539b9cc-CM000e5cde4ca2.cpe.net.cable.rogers.com) |
| 11:22.07 | lehel | i think i haven't isntalled any CAPI |
| 11:22.28 | *** part/#asterisk ManxPower (~eric@dsl-209-16-67-160.datasync.com) |
| 11:22.32 | jakepdev | modules.conf? |
| 11:22.39 | *** join/#asterisk Blissex (~Blissex@82-69-39-138.dsl.in-addr.zen.co.uk) |
| 11:23.26 | lehel | tzafrir: Span 1: WCTDM/0 "Wildcard TDM400P REV E/F Board 1" |
| 11:24.15 | lehel | 1, 2, 3, 4 WCTD FXOKS |
| 11:24.26 | tzafrir | So there's no reason to load chan_capi.so |
| 11:24.31 | lehel | ok |
| 11:24.41 | tzafrir | in modules.conf: unload=>chan_capi.so |
| 11:28.16 | *** join/#asterisk Jakes (~jakes@209.212.122.90) |
| 11:28.22 | christo | right |
| 11:28.40 | christo | I've update this to: exten => _9.,1,Dial(Zap/g1/${EXTEN}:1) for outside lines |
| 11:28.43 | christo | still no cigar :S |
| 11:29.18 | lehel | tzafrir: DONE |
| 11:29.36 | jakepdev | christo - how many digits are needed? |
| 11:29.56 | jakepdev | christo - i.e. in the US it's 10 or 7 |
| 11:30.10 | tzafrir | lehel, so, can you start asterisk now? |
| 11:30.11 | jakepdev | christo - i(or 11) |
| 11:30.17 | christo | jakepdev - it depends on how many are dialled - somebody might call '0870123123' or '003353358922' |
| 11:30.57 | lehel | tzafrir: no :((( : Unable to connect to remote asterisk |
| 11:31.04 | jakepdev | christo can you just try a specific number for now, instead of .? |
| 11:31.11 | christo | yeah |
| 11:31.22 | tzafrir | so, what's the problem now? |
| 11:31.32 | tzafrir | again, asterisk -vvvvc |
| 11:32.03 | christo | jakepdev - so to call 02072342342 it would be exten => _9XXXXXXXXXXX,1,Dial(Zap/g1/${EXTEN}:1) |
| 11:32.25 | christo | still no cigar |
| 11:32.28 | jakepdev | that looks right |
| 11:32.31 | christo | nothing in the console either |
| 11:32.33 | christo | hmmm |
| 11:32.38 | christo | bummer and now I have to go to a meeting |
| 11:32.47 | christo | back in 1 hr :( |
| 11:32.52 | jakepdev | christo - ok - try later to pastebin your output |
| 11:32.55 | Druken | christo: make sure they belong to the correct context |
| 11:33.22 | lehel | tzafrir: there are some NOTICEs and WARNINGs... |
| 11:33.45 | tzafrir | Did it crash on anything? |
| 11:34.10 | lehel | no crash |
| 11:34.23 | jakepdev | you got the asterisk CLI prompt then? |
| 11:35.03 | *** join/#asterisk Corydon76-home (five@pcp08665860pcs.500ash01.tn.comcast.net) |
| 11:35.09 | lehel | tzafrir: i have a mention |
| 11:36.02 | lehel | i think when i'm booting linux there should appear something Wildcard. pbx.. and stuff |
| 11:36.23 | lehel | but i looked after in the "messages" file.. no such things |
| 11:36.46 | lehel | pls don't be mad at me |
| 11:37.22 | lehel | tzafrir: what do u think? |
| 11:38.33 | nrc | lehel: it doesn't have to appear nothing during boot or dmes unless u're already autoloading zaptel modules to kernel |
| 11:38.42 | nrc | s/dmes/dmesg/ |
| 11:39.00 | *** join/#asterisk tzanger (~tzanger@mixdown.ca) |
| 11:39.15 | lehel | then what is the problem? |
| 11:39.38 | tzafrir | when you run asterisk from the command line it won't start? |
| 11:39.39 | *** part/#asterisk samueltc (~samuel@levinux.UQAR.UQUEBEC.CA) |
| 11:40.08 | nrc | lehel: do u want to use capi or zaptel ? |
| 11:40.27 | nrc | if u have a TDM u need zaptel |
| 11:40.29 | tzafrir | nrc, he has no capi |
| 11:40.33 | tzafrir | he has zaptel |
| 11:40.38 | tzafrir | zaptel module is loaded |
| 11:40.38 | lehel | nothing happening tzafrir.... nrc: zaptel! |
| 11:40.51 | *** join/#asterisk darkskiez (~darkskiez@usergc137.dsl.pipex.com) |
| 11:40.59 | tzafrir | asterisk -rx 'show version' |
| 11:41.17 | *** join/#asterisk Maxim (~max@max.m-telecom.ru) |
| 11:41.33 | lehel | "Unable to connect to remote asterisk" ;( |
| 11:42.04 | jakepdev | i'd say the WARNINGS might have some clue as to what is wrong |
| 11:42.22 | lehel | i don;t know.. maybe |
| 11:43.16 | jakepdev | lehel - that means it's still loading capi |
| 11:44.31 | nrc | lehel: do lsmod | grep zaptel |
| 11:44.31 | jakepdev | lehel - (or at least trying to load capi) |
| 11:44.31 | lehel | nrc: zaptel 176896 0 [wcusb wcfxs] |
| 11:44.45 | jakepdev | lehel - can you pastebin your modules.conf? |
| 11:44.46 | tzafrir | lehel, so maybe there is a problem with the socket? |
| 11:44.51 | tzafrir | ps aux |grep asterisk |
| 11:45.03 | jakepdev | tzafrir - it's still loading capi |
| 11:45.19 | tzafrir | jakepdev, how come it loads fine when starting manually? |
| 11:45.20 | lehel | tzafrir: root 6879 0.0 0.1 1824 576 pts/4 R+ 14:45 0:00 grep asterisk |
| 11:45.21 | jakepdev | and bombing on that |
| 11:45.31 | jakepdev | tzafrir - doesn't load fine |
| 11:45.40 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 11:45.42 | jakepdev | he just msg'd me with the output |
| 11:46.01 | jakepdev | still tries to load CAPI and bombs |
| 11:46.07 | tzafrir | in modules.conf: unload=>chan_capi.so |
| 11:46.08 | *** join/#asterisk ClayReiche123 (fwuser@mail.accxx.com) |
| 11:46.20 | jakepdev | it's loading it then unloading it |
| 11:46.23 | lehel | yes it still loading... but i modified in modules.conf: unload... |
| 11:46.31 | jakepdev | i say take it out altogether |
| 11:46.48 | ClayReiche123 | Can anyone recommend a good wholesale provider of long distance and nationwide DIDs? |
| 11:46.54 | tzafrir | jakepdev, how? remove the load all? |
| 11:46.57 | nrc | lehel: did u check dmesg after modprobe zaptel ? |
| 11:47.06 | lehel | no |
| 11:47.09 | Druken | ClayReiche123: of which nation? |
| 11:47.20 | ClayReiche123 | Sorry. United States. |
| 11:47.43 | ClayReiche123 | ...stupid American... :) |
| 11:47.51 | jakepdev | lehel - do you have a linr that says load=>capi or something like that? |
| 11:47.55 | lehel | Wildcard USB FXS Interface driver registered |
| 11:47.55 | lehel | Registered tone zone 0 (United States / North America) |
| 11:48.12 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com) |
| 11:48.33 | nrc | lehel: and dmesg after modprobe wcfxs and wcfxo ? |
| 11:48.33 | jakepdev | in modules.conf |
| 11:48.59 | lehel | where is that modules.conf.. ?? i'm getting craazy |
| 11:49.40 | lehel | modprobe: Invalid line 122 in /etc/modules.conf |
| 11:49.40 | lehel | <PROTECTED> |
| 11:50.01 | ClayReiche123 | Druken: Any suggestions? |
| 11:50.36 | jakepdev | lehel - if you would just pastebin your modules.conf - it might help us |
| 11:50.48 | lehel | immediatly |
| 11:51.49 | *** join/#asterisk WilliamK (~wkeller@c-24-0-130-177.hsd1.tx.comcast.net) |
| 11:52.07 | *** join/#asterisk orchard (~support@80.176.100.211) |
| 11:52.43 | lehel | cannot... i don't know why |
| 11:52.53 | lehel | can i send the file? |
| 11:52.58 | jakepdev | what can't you do? |
| 11:53.05 | jakepdev | can't access pastebin.ca? |
| 11:53.32 | lehel | "pastebin.ca" ? |
| 11:53.35 | jakepdev | yes |
| 11:53.38 | jakepdev | ~pastebin |
| 11:53.39 | jbot | pastebin is probably a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 11:54.12 | lehel | i can't copy the text from the .conf file |
| 11:54.48 | jakepdev | can you copy it to your editor and paste from there? |
| 11:55.51 | orchard | First hello everyone - does anyone know how to set up CID for UK BT using a TDM11B (TDM400P)? I'm using stable version 1.0 of asterisk. From what I have read it should already be in that version of the CVS. |
| 11:56.14 | lehel | there are 122 lines! |
| 11:56.17 | lehel | where to paste? |
| 11:57.18 | jakepdev | orchard - check zaptel.conf and make sure usecallerid is on |
| 11:57.21 | *** part/#asterisk Maxim (~max@max.m-telecom.ru) |
| 11:57.34 | *** join/#asterisk ckruetze (HydraIRC@cpc3-cmbg7-5-0-cust100.cmbg.cable.ntl.com) |
| 11:57.45 | jakepdev | lehel - paste into http://pastebin.ca |
| 11:57.46 | lehel | jakepdev: Your paste has been accepted an added to the database. You will be redirected to it momentarily. The URL for it is: |
| 11:57.46 | lehel | http://pastebin.ca/10495 |
| 11:57.47 | lehel | You may use that URL for referencing your submission from now on. |
| 11:57.59 | lehel | DONE |
| 11:58.12 | lehel | look tzafrir u too pls |
| 11:58.54 | orchard | jakepdev - usecallerid=yes |
| 12:00.06 | tzafrir | lehel, you've edited the wrong file |
| 12:00.12 | *** join/#asterisk deno (~deno@194.94.81.60) |
| 12:00.22 | *** join/#asterisk vihai (~orlandi@nabla.orlandi.com) |
| 12:00.36 | tzafrir | you shouldn't have edited /etc/modutils/zaptel or anything in that directory. no /etc/modules.conf |
| 12:00.47 | tzafrir | the file is /etc/asterisk/modules.conf |
| 12:00.57 | *** join/#asterisk riksta (~rick@212.85.228.176) |
| 12:01.06 | lehel | so... i'm going to erase that line from.. |
| 12:01.20 | lehel | and i'm going to /etc/asterisk/modules.conf |
| 12:02.20 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 12:02.30 | jakepdev | tzafrir - tnx - i was a bit confused as to why that looked so different |
| 12:03.21 | lehel | DONE... and sorry |
| 12:03.27 | tzafrir | now I really must be going... |
| 12:03.36 | vihai | Hello, I'm writing a channel driver for * and I have to generate standard indication tones. Is there a way to use the res_indication infrastructure before the channel is connected? Is there a way to start_pbx without connecting the channel to another party? |
| 12:03.53 | jakepdev | vihai - try on asterisk-dev |
| 12:03.54 | lehel | dev01:/etc/asterisk# asterisk -r |
| 12:03.54 | lehel | Unable to connect to remote asterisk |
| 12:04.05 | jakepdev | wait lehel - not yet :) |
| 12:04.16 | lehel | ok;p |
| 12:04.17 | jakepdev | pastebin your modules.conf |
| 12:04.25 | jakepdev | the one in the proper folder |
| 12:04.32 | vihai | jakepdev: Thanks, I didn't know it existed :) |
| 12:04.55 | orchard | jakepdev - did you really mean zaptel.conf or zapata.conf? zapata.conf has usecallerid=yes zaptel.conf does not. |
| 12:05.07 | *** part/#asterisk vihai (~orlandi@nabla.orlandi.com) |
| 12:05.09 | jakepdev | orchard - yes |
| 12:05.10 | jakepdev | :) |
| 12:05.15 | fenlander | orchard: can I ask where you got a TDM11B in the UK? |
| 12:07.59 | orchard | fenlander - I am just given stuff to make work - from what I can see it was supplied by TelAppliant for development use only - no support. |
| 12:08.14 | *** join/#asterisk Ciber (~Ciber@70.84.122.194) |
| 12:08.40 | Ciber | anyone here using asterisk with broadvoice? |
| 12:09.31 | orchard | jakepdev - do I need a line in my zaptel.conf that reads usecallerid=on or usecallerid=yes? |
| 12:09.49 | jakepdev | usecallerid=yes |
| 12:10.39 | *** join/#asterisk BuffPuff (~Buff@benja.demon.nl) |
| 12:10.39 | deno | Hi everbody! Can anybody tell me how I can transcode the RTP Payload during a VoIP Session? |
| 12:10.48 | BuffPuff | good afternoon all |
| 12:11.52 | jakepdev | BuffPuff - depends where you are |
| 12:12.04 | deno | ?? |
| 12:12.15 | *** join/#asterisk MattB2 (~m@pcp01068561pcs.andrsn01.tn.comcast.net) |
| 12:12.16 | Ciber | think i'm gonna shoot myself soon :P |
| 12:12.27 | fenlander | orchard: thanks - do you have any problems with hangup detection? |
| 12:12.27 | Ciber | been trying to get this working with broadvoice all night |
| 12:12.39 | jakepdev | Ciber - join #suicide |
| 12:12.40 | MattB2 | hangup detection! argh! |
| 12:12.43 | orchard | jakepdev - still nothing. Just check with sipure 3000 - it does get the CID. |
| 12:13.19 | Ciber | heh |
| 12:13.31 | *** join/#asterisk ManxPower (~eric@dsl-209-16-67-160.datasync.com) |
| 12:13.43 | BuffPuff | jakepdev - hmmm, damn global village ;-) |
| 12:13.54 | fenlander | MattB2: I take it that you have problems? |
| 12:14.01 | MattB2 | fenlander: yup |
| 12:14.13 | MattB2 | my lines have disconnect supervision (tested with backlit phone) |
| 12:14.13 | jakepdev | orchard - you have full debug on? |
| 12:14.15 | BuffPuff | can anyone give some advice on oh.323 issues? |
| 12:14.31 | MattB2 | with debuggin on X100P I can see what's going on, and it's not what I expect |
| 12:14.32 | jakepdev | ciber - what is the problem connecting to broadvoice? |
| 12:14.33 | MattB2 | BATTERY when no call |
| 12:14.37 | *** join/#asterisk durex (~ironman@weber.anpa.org.br) |
| 12:14.38 | MattB2 | as soon as call is made I get NO BATTERY |
| 12:14.41 | Ciber | well i have no idea |
| 12:14.50 | MattB2 | then when I forcefully hangup (ie not when other end hangs up) I get BATTERY again |
| 12:14.50 | jakepdev | ciber - do you get an error |
| 12:14.52 | Ciber | i tried the config settings at broadvoice |
| 12:15.06 | MattB2 | there's no hint that the X100P is detecting the drop in battery at the proper end of call |
| 12:15.07 | jakepdev | ciber - what is the error? |
| 12:15.08 | Ciber | and when i call the number it just tells me the subscriber is busy or something |
| 12:15.24 | jakepdev | ciber - you'll need the error from the asterisk CLI |
| 12:15.28 | Ciber | no error that i know of since i'm on a mac |
| 12:15.32 | fenlander | hmm |
| 12:15.33 | Ciber | how do i get the cli? :P |
| 12:15.39 | jakepdev | ciber - using asterisk? |
| 12:15.51 | Ciber | well i'm using this asterisk launcher on my mac |
| 12:16.05 | newl | erm..there shold always be exchange battery on the line with a PSTN service, on hook or off. |
| 12:16.09 | orchard | fenlander - a bit hit and miss - caed arrived this morning so not a lot of testing yet. When you do the install use the opermode e.g modprobe wcfxs opermode=UK |
| 12:16.11 | jakepdev | can you drop to a command prompt? |
| 12:16.15 | Ciber | yes |
| 12:16.24 | lehel | jakepdev: i really don't have any time today |
| 12:16.24 | jakepdev | ciber - type asterisk -rvvvvc |
| 12:16.27 | deno | Im able to chan the Paylode type in the header but i dont know how to transcode the payload |
| 12:16.29 | BuffPuff | they have a mac astersik? |
| 12:16.32 | Ciber | yes |
| 12:16.34 | newl | unless you're off hook for an extended time at which your exchange may lock the device out. :) |
| 12:16.41 | lehel | we talk tommorrow! |
| 12:16.44 | jakepdev | ok |
| 12:16.45 | Skarmeth | anyone here using Digium's S100I — "IAXy" for POTS? |
| 12:16.51 | lehel | take care |
| 12:16.56 | orchard | jakepdev - will do just a mo. |
| 12:16.56 | jakepdev | you too |
| 12:17.01 | kajtzu | hmmm is there anyway to rewrite the ANI on an inbound call? |
| 12:17.04 | Skarmeth | it's ethernet/wan port are 10/100Mbps? |
| 12:17.06 | lehel | bye |
| 12:17.26 | Ciber | Cibers-Computer:~ Ciber$ asterisk -rvvvvc |
| 12:17.26 | Ciber | Unable to connect to remote asterisk |
| 12:17.29 | BuffPuff | I have a small 323 problem, can anyone help me debug it please? |
| 12:17.45 | jakepdev | ciber - try "asterisk -vvvvc" |
| 12:17.46 | durex | Ciber: sudo asterisk -rvvvvc |
| 12:18.04 | jakepdev | sudo - like karate? |
| 12:18.06 | deno | :-( |
| 12:18.27 | Ciber | k i'm on it now with durex's command heh |
| 12:18.30 | kajtzu | I would like to prepend a 0 on inbound calls so the users can immediately call back instead of editing the number separately |
| 12:18.36 | Ciber | Apr 28 08:18:21 NOTICE[476]: chan_sip.c:4035 sip_reg_timeout: -- Registration for '16462050612@sip.broadvoice.com@sip.broadvoice.com' timed out, trying again |
| 12:18.38 | ManxPower | Ciber: you need to be root to access asterisk |
| 12:18.40 | Ciber | Apr 28 08:18:27 WARNING[476]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call 7af443e33a952b251743a82f5ad6b197@192.168.1.100 for seqno 106 (Critical Request) |
| 12:18.43 | Ciber | Cibers-Computer*CLI> |
| 12:20.16 | ManxPower | ~mailingist |
| 12:20.18 | jakepdev | kajtzu - you can use SetVar(PREPENDED=0${CALLERIDNUM}) |
| 12:20.22 | ManxPower | ~mailinglist |
| 12:20.23 | jbot | rumour has it, mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
| 12:20.50 | ManxPower | why not just SetCIDNum(9${CALLERIDNUM}) |
| 12:21.01 | newl | or SetCIDNum("0${CALLERIDNUM}") |
| 12:21.06 | ManxPower | setvar isn't going to actually do anythong useful. |
| 12:21.10 | ManxPower | newl: don't use quotes! |
| 12:21.19 | newl | Why? It's a string. |
| 12:21.21 | jakepdev | it depends on what the user is trying to do |
| 12:21.31 | ManxPower | SetCIDNum(0${CALLERIDNUM}) |
| 12:21.39 | vaewyn | newl: no... calleridnum is numeric |
| 12:21.50 | jakepdev | ManxPower - in this case - they want to use the number - not reset the callerid |
| 12:21.52 | ManxPower | newl: because asterisk doesn't require strings to be quoted and it can actually cause problems. |
| 12:22.21 | newl | erm..you may think it's numeric. It's a string. Leading zeros aren't preserved on most operating systems when using numeric datatypes. :) |
| 12:22.51 | *** join/#asterisk cc (~cc@byte.fedora) |
| 12:22.55 | jakepdev | the CALLERIDNUM should be a string |
| 12:22.57 | ManxPower | jakepdev: Sorry. His question is SO incredibly BASIC that I read it wrong. |
| 12:22.59 | newl | ManxPower: fair enough. Kind hard to get rid of older habits. |
| 12:23.26 | vaewyn | no... CALLERID is a string... CALLERIDNUM is a numeric and CALLERIDNAME is string |
| 12:23.33 | ManxPower | newl: some ip phones refuse calls with bad callerid. many providers will refuse to accept bad callerid |
| 12:23.52 | jakepdev | CALLERIDNUM is a numeric - are you sure? |
| 12:24.06 | ManxPower | Asterish thinks everything is a string, even numbers. |
| 12:24.07 | vaewyn | try and put a character besides a number in it |
| 12:24.12 | ManxPower | At least from the dialplan |
| 12:24.32 | ManxPower | vaewyn: recent CVS=HEAD will accept ANYTHING in calleridnum I think. |
| 12:24.35 | vaewyn | *nods* but it does do type checking on the set values... |
| 12:24.37 | jakepdev | a true test wouldbe a leading 0 |
| 12:24.53 | ManxPower | Dial(Zap/g1/0${EXTEN}) |
| 12:25.15 | vaewyn | the data type is a string in the background... but the only data you can put in it is numeric... hence... it isn't a generic string |
| 12:25.48 | kajtzu | jakepdev: I'll try and see if I can just rewrite the outgoing number instead through cisco's dialplan.xml. it would be neater. |
| 12:25.51 | ManxPower | newl: in your exmaple asterisk would include the quotes as part of the callerid number |
| 12:25.58 | newl | okay |
| 12:26.05 | ManxPower | and we all know neither callerid name nor callerid number normally have quotes in them. |
| 12:26.10 | newl | in yours, asterisk happily accepts the leading zero :) |
| 12:26.13 | heison | is anyone here familiar with this error: ZT_CHANCONFIG failed on channel 3: Invalid argument (22)? |
| 12:26.28 | Zeeek | it's true that the examples all use quotes but it's also true that quotes screw up some phones and it always works without quotes, so... |
| 12:26.39 | ManxPower | the examples should not use quotes |
| 12:26.43 | Zeeek | they do |
| 12:26.52 | ManxPower | I guess I should go get fast food breakfast before the idiots start calling. |
| 12:27.05 | jakepdev | idiots = customers? |
| 12:27.14 | ManxPower | jakepdev: of course. |
| 12:27.19 | durex | ManxPower lol |
| 12:27.32 | ManxPower | jakepdev: or more specifically "users at customer's sites" |
| 12:27.38 | jakepdev | i hear that :) |
| 12:27.38 | BuffPuff | Can anyone understand this error message?: |
| 12:27.44 | BuffPuff | chan_sip.c:1829 sip_write: Asked to transmit frame type 1, while native formats is 2 (read/wri |
| 12:27.56 | Ahrimanes | anyone here running * and ser together? |
| 12:27.56 | BuffPuff | what is frame type 1? |
| 12:27.58 | ManxPower | BuffPuff: "show codecs" will tell you the format numbers |
| 12:28.04 | deno | how can I transcode the RTP Payload during a VoIP Session? |
| 12:28.07 | orchard | back in a mo |
| 12:28.09 | BuffPuff | thx |
| 12:28.10 | *** part/#asterisk orchard (~support@80.176.100.211) |
| 12:28.11 | tzafrir | jakepdev, nah, not all the idiots are your customers |
| 12:28.24 | ManxPower | BuffPuff: you prolly have a bandsidth= line or an allow=all line in sip.conf |
| 12:28.30 | Zeeek | abyone ever seen this? : chan_zap.c:1521 zt_call: cidspill already exists?? |
| 12:28.40 | jakepdev | tzafrir - my customers are fine |
| 12:28.47 | Zeeek | cidspill? |
| 12:28.55 | jakepdev | luckily |
| 12:29.02 | Zeeek | I don't want to get any of that on me if it spills |
| 12:29.08 | Zeeek | but what is it? |
| 12:29.56 | jakepdev | if never seen cid spill - just use a cup holder |
| 12:29.56 | ManxPower | I don't mind customers. It's people I hate. |
| 12:29.56 | BuffPuff | ManxPower : i got allow=all, bad? |
| 12:29.56 | ManxPower | BuffPuff: Yes. |
| 12:29.56 | ManxPower | It allows things like G723.1, which Asterisk can't transcode. |
| 12:30.09 | ManxPower | disallow=all and allow= lines for ONLY the codecs you need to use. |
| 12:30.26 | BuffPuff | hmmm |
| 12:30.42 | BuffPuff | i have a provider that supports only 729 and 723 |
| 12:31.00 | BuffPuff | cant it bridge it? |
| 12:31.02 | jakepdev | disallow=all allow=g729 allow=g723 |
| 12:31.03 | ManxPower | BuffPuff: then you need to purchase G729 licenses from Digium. |
| 12:31.12 | jakepdev | or g723 |
| 12:31.15 | jakepdev | oh |
| 12:31.16 | jakepdev | sorry |
| 12:31.19 | jakepdev | no g723 |
| 12:31.39 | BuffPuff | hmm, u have a clue about the costs? |
| 12:31.40 | jakepdev | BuffPuff - how many simultanious calls? |
| 12:31.40 | ManxPower | jakepdev: As I'm sure you know if a call is in G729 or G723 format and you don't have a license then you cannot use voicemail, playback, background, t/T in Dial, and lots of other stuff. |
| 12:31.46 | ManxPower | BuffPuff: $10/channel |
| 12:31.51 | jakepdev | right Mnx - just remebered |
| 12:31.56 | *** join/#asterisk rlg (~umairbari@202.142.189.86) |
| 12:32.06 | BuffPuff | i c |
| 12:32.39 | jakepdev | that's cheap unless you gots several PRIs |
| 12:32.58 | BuffPuff | exactly |
| 12:33.08 | ManxPower | You CAN work around the issue once you understand SIP, codecs, transcoding, etc. But why waste all those months when you can just not go out drinking on Friday night and spend the money on G729 licenses? |
| 12:33.10 | jakepdev | how many PRIs? |
| 12:33.36 | tainted- | quick question |
| 12:33.43 | tainted- | how do i turn off notice messages |
| 12:33.52 | ManxPower | tainted-: logger.conf |
| 12:33.54 | tainted- | i have this in logger.conf |
| 12:33.55 | tainted- | console => warning,error |
| 12:33.56 | BuffPuff | 3-4 |
| 12:34.00 | tainted- | but still get notice msgs |
| 12:34.13 | BuffPuff | meny endstations also |
| 12:34.20 | tainted- | or should i be editing messages => |
| 12:34.22 | jakepdev | tainted - in logger? |
| 12:34.25 | jakepdev | or in CLI? |
| 12:34.28 | ManxPower | or dump your girlfriend and use the extra money to buy VoIP hardware, and channel the sexual frustration into learning Asterisk? |
| 12:34.48 | ManxPower | tainted-: where do you not want the NOTICE messages? Console or logs? |
| 12:34.48 | tainted- | jakepdev in CLI |
| 12:34.49 | newl | haha |
| 12:34.54 | tainted- | console |
| 12:34.58 | ManxPower | tainted-: don't start asterisk -rvvv, start asterisk -r |
| 12:35.05 | jakepdev | tainted - use less verbose |
| 12:35.09 | tainted- | i dun want NOTICE but i want WARNING |
| 12:35.13 | Ciber | does anyone have a good ip for broadvoice? |
| 12:35.24 | Ciber | Apr 28 08:35:11 NOTICE[538]: chan_sip.c:4035 sip_reg_timeout: -- Registration for '16462050612@sip.broadvoice.com@sip.broadvoice.com' timed out, trying again |
| 12:35.27 | tainted- | Ciber good ip for broadvoice? |
| 12:35.28 | Ciber | <PROTECTED> |
| 12:35.29 | BuffPuff | Manx: she is at work anyway, do u have some info on how to get around the bridging issue? |
| 12:35.36 | Ciber | getting that for their lax proxy |
| 12:35.41 | ManxPower | Ciber: that means they didn't like the number you are trying to dial |
| 12:35.48 | tainted- | Ciber try nyc proxy |
| 12:35.54 | Ciber | i tried nyc proxy |
| 12:35.57 | tainted- | ManxPower no, his registration is not going through |
| 12:36.03 | ManxPower | Ciber: registration is only for INCOMING calls. |
| 12:36.03 | Ciber | get timeouts with the nyc one |
| 12:36.20 | *** join/#asterisk FaithX (~Faithful@202-6-145-116.ip.adam.com.au) |
| 12:36.26 | Ciber | get 404 not found with the miami one |
| 12:36.30 | tainted- | Ciber could be your conf |
| 12:37.10 | *** part/#asterisk ClayReiche123 (fwuser@mail.accxx.com) |
| 12:37.12 | tainted- | is there a table stating what verbosity says what? |
| 12:37.34 | tainted- | prolly on voip-info huh |
| 12:37.45 | tainted- | huh manxy |
| 12:37.52 | tainted- | guide me damnit |
| 12:38.35 | jakepdev | do you really want that guidance remeber "<ManxPower> or dump your girlfriend" |
| 12:38.56 | tainted- | lol |
| 12:40.00 | tainted- | i like "channel sexual frustration into learning asterisk".. <-- that's a gem |
| 12:40.12 | jakepdev | yep - definately a keeper |
| 12:40.12 | BuffPuff | ManxPower: is g711 also problematic |
| 12:40.14 | ManxPower | 8-) |
| 12:40.25 | BuffPuff | ? |
| 12:40.26 | ManxPower | BuffPuff: no, only G279 and G273 |
| 12:40.30 | BuffPuff | k thx |
| 12:40.38 | RoyK | tainted-: likke this? http://karlsbakk.net/mp3fun/asterisk-installation.wav |
| 12:40.40 | ManxPower | those are the only patented codecs asterisk says it supports |
| 12:40.55 | RoyK | ilbc |
| 12:41.28 | BuffPuff | ManxPower: so it should be allow=711u ? |
| 12:41.57 | Ciber | lol nice wav |
| 12:41.59 | ManxPower | BuffPuff: "show codecs" will tell you what the format names are |
| 12:42.00 | tainted- | RoyK love ya bitch |
| 12:42.12 | BuffPuff | and i would like to hear if there is a way to have asterisk work as a bridge |
| 12:42.13 | RoyK | :) |
| 12:42.14 | BuffPuff | thx |
| 12:42.28 | tainted- | WHAT |
| 12:42.33 | tainted- | love ya bitch is taken out! |
| 12:42.44 | RoyK | BuffPuff: asterisk usually bridges |
| 12:42.46 | RoyK | tainted-: ? |
| 12:42.54 | tainted- | she doesn't say love ya bitch at the end |
| 12:43.05 | Ciber | damn stupid mac |
| 12:43.11 | tainted- | original = "mmkay - love ya bitch" |
| 12:43.24 | Ciber | changed the hosts file but now it's not updating grrr |
| 12:43.25 | ManxPower | I've never used Asterisk to build a bridge over anything. |
| 12:43.53 | Nugget | If BuffPuff weighs less than a duck, he's made of wood, and therefore a witch. |
| 12:43.59 | tainted- | not even a bridge over troubled waters? |
| 12:44.44 | ManxPower | Nope! |
| 12:44.46 | jakepdev | just make sure you're using stable |
| 12:45.56 | ManxPower | ARGH! |
| 12:46.25 | ManxPower | My cat just spend the time I was getting clothes on to roll around on and paw the docs I printed and set on the floor. |
| 12:46.48 | tainted- | gooood cat |
| 12:46.53 | jakepdev | the cat obvoiusly wasn't trained on what documentation was |
| 12:46.58 | tainted- | eeeeexcellllllllllent |
| 12:47.09 | christo | RoyK that wav is hilarious |
| 12:47.17 | tainted- | cat man doc |
| 12:47.26 | *** join/#asterisk asteriskdz (~djazcall@82.101.142.52) |
| 12:47.34 | asteriskdz | hello all |
| 12:47.43 | jakepdev | hello dz |
| 12:47.51 | asteriskdz | it's the first time |
| 12:47.57 | asteriskdz | that i 'm connected here |
| 12:48.10 | jakepdev | congrats |
| 12:48.11 | Zeeek | was it good? |
| 12:48.11 | asteriskdz | i'm happy to speak with you |
| 12:48.13 | tainted- | RoyK i hate u |
| 12:48.30 | asteriskdz | ok so i'm welkom?? |
| 12:48.31 | tainted- | RoyK i just accidentally killed my ultra custom playlist to hear that wav file |
| 12:48.48 | Zeeek | you're velcro anytime |
| 12:48.55 | jakepdev | thank you now your welcome |
| 12:48.56 | asteriskdz | pls any can help me to install my TDM22 |
| 12:49.03 | RoyK | tainted-: rotfl |
| 12:49.04 | RoyK | :D |
| 12:49.05 | Zeeek | of course |
| 12:49.32 | *** join/#asterisk Ciber (~Ciber@70.84.122.194) |
| 12:49.37 | asteriskdz | i'm installing asterisk into |
| 12:49.43 | asteriskdz | mandrak 9.2 is this ok?? |
| 12:50.06 | durex | asteriskdz no problem |
| 12:50.09 | Ciber | well back to this |
| 12:50.10 | Ciber | Apr 28 08:49:52 NOTICE[251]: chan_sip.c:4035 sip_reg_timeout: -- Registration for '16462050612@sip.broadvoice.com@sip.broadvoice.com' timed out, trying again |
| 12:50.14 | Ciber | Apr 28 08:49:58 WARNING[251]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call 7ed4d86647cb51977559b553548f9855@sip.broadvoice.com for seqno 110 (Critical Request) |
| 12:50.49 | Zeeek | Ciber have they changed ip adrreses lately? |
| 12:51.00 | asteriskdz | i can't install the zaptel modul |
| 12:51.03 | Ciber | i just pinged it and added the ip to my config |
| 12:51.16 | Zeeek | DNS cache prob? |
| 12:51.17 | Ciber | http://www.broadvoice.com/support_install_asterisk.html i followed their instructions |
| 12:51.25 | *** join/#asterisk Fraeggl (~Fraeggl@rkom.r-kom.de) |
| 12:51.36 | Zeeek | there are chapters of stuff on bv on the mailing list |
| 12:51.42 | Zeeek | volumes! |
| 12:51.46 | Zeeek | tomes! |
| 12:51.52 | Zeeek | library shelves |
| 12:51.57 | Ciber | ... |
| 12:52.04 | tainted- | could be weseals |
| 12:52.09 | tainted- | weasels too |
| 12:52.10 | Zeeek | amazon has a new category "books about broadvoice and asterisk" |
| 12:52.27 | tainted- | Zeeek really |
| 12:52.37 | Zeeek | and O'Reilly has "Broadvoice Annoyances" |
| 12:52.38 | tainted- | u forgot library of congress |
| 12:52.51 | Zeeek | yea you can telnet in |
| 12:52.53 | tainted- | BV Hacks |
| 12:54.17 | asteriskdz | what is the step to install a pbx ?? with asterisk |
| 12:54.25 | Zeeek | two of the big recurring subjects are BV and BT |
| 12:54.27 | asteriskdz | installing kernel , asterisk |
| 12:54.29 | tainted- | asteriskdz YES! |
| 12:54.43 | tainted- | 1) install kernel |
| 12:54.46 | asteriskdz | ? |
| 12:54.48 | tainted- | 2) install asterisk |
| 12:54.51 | tainted- | 3 ?? |
| 12:54.53 | asteriskdz | yes |
| 12:54.54 | Zeeek | play |
| 12:54.55 | tainted- | 4) PROFIT!! |
| 12:55.16 | jakepdev | 5) answer support calls |
| 12:55.20 | asteriskdz | 3?? |
| 12:55.21 | Zeeek | 3 - screw around configuring until you finally decide to dump @hole |
| 12:55.32 | Zeeek | 3a then configure |
| 12:55.41 | tainted- | 6) realize BV sucks and drop BV |
| 12:55.46 | Zeeek | heh |
| 12:55.57 | asteriskdz | what ?? |
| 12:55.58 | christo | If a user presses '9' followed by '02072342342' I want asterisk to pass that call to Zap/g1 as follows: exten => _9XXXXXXXXXXX,1,Dial(Zap/g1/${EXTEN}:1) I have this in the [default] section in extensions.conf, but it's not working at all. What else should I be looking out for? |
| 12:56.24 | tainted- | asteriskdz are u paying attention!!! |
| 12:56.30 | asteriskdz | yes |
| 12:56.38 | tainted- | what is #2 |
| 12:56.40 | asteriskdz | i'm waiting 3) |
| 12:56.46 | tainted- | 3) ?? |
| 12:56.59 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
| 12:57.04 | tainted- | u need unicode support on your IRC client to see 3) ?? |
| 12:57.08 | asteriskdz | i think tha linux kernel, asterisk , zaptel and other module |
| 12:57.13 | *** join/#asterisk Faithful (~Faithful@202-6-145-116.ip.adam.com.au) [NETSPLIT VICTIM] |
| 12:57.16 | Zeeek | asteriskdz here is a doc that goes from OS install to asterisk. It shows RedHat but you skip that part |
| 12:57.16 | Zeeek | http://www.automated.it/guidetoasterisk.htm |
| 12:57.16 | asteriskdz | but i don't know what doing after!!!? |
| 12:57.16 | Zeeek | http://www.automated.it/guidetoasterisk.htm |
| 12:57.22 | Zeeek | asteriskdz here is a doc that goes from OS install to asterisk. It shows RedHat but you skip that part |
| 12:57.26 | *** join/#asterisk zeedo (~zeedo@www.bsrf.org.uk) [NETSPLIT VICTIM] |
| 12:57.32 | tainted- | Zeeek good thing u repeated that twice |
| 12:57.34 | Zeeek | look at that guide, it will help you I think |
| 12:57.39 | *** join/#asterisk ScaredyCat (~ScaredyCa@i169173.upc-i.chello.nl) |
| 12:57.43 | tainted- | my lazy eye nearly missed it |
| 12:57.43 | *** join/#asterisk twisted[work] (~twisted@twisted-professional-pdpc.developer.and.friend.of.asterisk) [NETSPLIT VICTIM] |
| 12:57.43 | *** mode/#asterisk [+o twisted[work]] by ChanServ |
| 12:57.49 | Zeeek | my keybounce software has a bug |
| 12:57.54 | *** join/#asterisk A-Tuin|work (~A-Tuin@nat.office.legend.net.uk) [NETSPLIT VICTIM] |
| 12:58.06 | asteriskdz | i will chek |
| 12:58.07 | Zeeek | it waits for one full URL then repeates it |
| 12:58.14 | tainted- | Introduction. 2 |
| 12:58.14 | tainted- | Who is your daddy and what does he do? |
| 12:58.21 | Zeeek | that's the best part |
| 12:58.25 | tainted- | this guide is obviously authoritative |
| 12:58.33 | Zeeek | it is very good for beginners |
| 12:58.46 | *** join/#asterisk guyee (~izomtriko@nextra.nudli.equitas.hu) |
| 12:59.04 | tainted- | "Anyway, enough about me, you didn’t even read that bit anyway did you…" |
| 12:59.04 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) [NETSPLIT VICTIM] |
| 12:59.11 | asteriskdz | ok ok |
| 12:59.11 | tainted- | i wish more man pages were written this way |
| 12:59.17 | Zeeek | apparently you did :) |
| 12:59.20 | asteriskdz | thnak you i will chek |
| 12:59.34 | Zeeek | asteriskdz here are a few good books |
| 12:59.37 | Zeeek | Starter tutorial: |
| 12:59.37 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
| 12:59.37 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html |
| 12:59.37 | Zeeek | http://www.automated.it/guidetoasterisk.htm |
| 12:59.37 | Zeeek | THE reference of the moment: |
| 12:59.37 | Zeeek | http://www.asteriskdocs.org |
| 13:00.01 | tainted- | shameless plug at the end there Zeeek |
| 13:00.15 | Zeeek | yes, I own all those and get a big commission |
| 13:00.23 | tainted- | lol |
| 13:00.27 | tainted- | big as in 0? |
| 13:00.33 | tainted- | or big as in z-e-r-o |
| 13:00.35 | Zeeek | actually, I wronte one paragraph in asteriskdocs.org that stayed in :) |
| 13:00.36 | zoa | hehe |
| 13:00.46 | Zeeek | I am famous |
| 13:01.02 | Zeeek | look at my /proc/interrupts chart |
| 13:01.15 | tainted- | fantastic |
| 13:01.20 | Zeeek | I wrote the fucking book! |
| 13:01.26 | tainted- | keep the dream alive |
| 13:01.30 | Zeeek | even tho I don't know what half that shit means |
| 13:01.46 | asteriskdz | [djazcall@Actos djazcall]$ modprobe |
| 13:01.47 | asteriskdz | bash: modprobe: command not found |
| 13:01.49 | Zeeek | but I knowed a IRQ when I sawed one |
| 13:02.03 | tainted- | such modesty |
| 13:02.04 | asteriskdz | i think that my linux kernel don't know the command modeprob |
| 13:02.17 | Ciber | now getting these two Apr 28 09:01:52 NOTICE[343]: chan_sip.c:4035 sip_reg_timeout: -- Registration for '6462050612@sip.broadvoice.com' timed out, trying again |
| 13:02.18 | ManxPower | Yay! Food! Yay! |
| 13:02.19 | tainted- | speaking in layman for us unwashed masses |
| 13:02.19 | Ciber | Apr 28 09:01:52 NOTICE[343]: chan_sip.c:6819 handle_response: Failed to authenticate on REGISTER to '<sip:6462050612@sip.broadvoice.com>;tag=as2d6578c8' |
| 13:02.25 | *** join/#asterisk bowman (~bowman@snert3.tal.de) |
| 13:02.27 | tainted- | we're not worthy~ |
| 13:02.38 | bowman | heya. asterisk's internal "EnumLookup" seems to be disfunctional. any hints? |
| 13:02.39 | tainted- | asteriskdz u sure it's linux? |
| 13:02.50 | *** join/#asterisk MattH (~matth@noc-wireless.chilitech.net) [NETSPLIT VICTIM] |
| 13:03.02 | MattH | Hi.. if you destroy a zap channel with zap channel destory x.. how do you recreate it? |
| 13:03.04 | Zeeek | he said Mandrake |
| 13:03.13 | bowman | MattH: restart the server ;) |
| 13:03.18 | Zeeek | MattB2 only *DOG can do that |
| 13:03.46 | MattH | hrmmm I restarted... but it still says it can't use it |
| 13:03.54 | MattH | Apr 28 09:02:50 ERROR[1816]: Unable to open channel 1: No such device or address |
| 13:04.13 | tainted- | lol |
| 13:04.39 | MattH | I even tried rebooting the whole thing |
| 13:05.07 | tainted- | try rebooting just the RAM part |
| 13:05.20 | MattH | ? |
| 13:05.22 | tainted- | or the IRQ part |
| 13:05.27 | tainted- | Zeeek has a guide |
| 13:05.29 | MattH | umm... I'm not aware that you can do that |
| 13:05.38 | Zeeek | haha |
| 13:05.42 | MattH | I also tried shutting down the system, uunplugging the PRI.. waiting.. and then starting up again |
| 13:05.59 | *** join/#asterisk newl (~newlook@203-59-112-161.dyn.iinet.net.au) [NETSPLIT VICTIM] |
| 13:05.59 | *** mode/#asterisk [+n] by irc.freenode.net |
| 13:06.01 | *** mode/#asterisk [-n] by kram |
| 13:06.01 | *** mode/#asterisk [+n] by ChanServ |
| 13:06.09 | tainted- | MattH the scientific community watches you with envy |
| 13:06.12 | MattH | lol |
| 13:06.32 | ManxPower | If it can't open a channel then there's a problem with the device or it's config. |
| 13:06.32 | MattH | what exactly DOES 'zap channel destroy' do (I was kinda playing and learning).... like could it be possible my phone company still thinks the channel is down? |
| 13:06.37 | johnnyb | Yeah! zttest is consistently giving 100% on my new asterisk box! |
| 13:06.54 | johnnyb | To get it to work, I had to: |
| 13:06.54 | tainted- | i'm visualizing a Gary Larson - Far Side comic.. anyone else? |
| 13:06.55 | ManxPower | MattH it kills a zap channel and it won't come back until you restart Asterisk |
| 13:07.03 | johnnyb | <PROTECTED> |
| 13:07.17 | ManxPower | johnnyb: what SPECIFIC motherboard? |
| 13:07.29 | ManxPower | I WANT MODEL AND PART NUMBERS! |
| 13:07.33 | johnnyb | <PROTECTED> |
| 13:07.35 | *** join/#asterisk rkioko (~kiokorobe@196.200.26.42) |
| 13:07.55 | johnnyb | ManxPower: It's an old vaio -- I'll try to find the part numbers for you. |
| 13:08.02 | johnnyb | <PROTECTED> |
| 13:08.06 | ManxPower | oh, nevermind then. |
| 13:08.08 | johnnyb | <PROTECTED> |
| 13:08.14 | ManxPower | vaio = purple barbie computer |
| 13:08.22 | tainted- | why, is it that hard to get 100%? |
| 13:08.30 | johnnyb | It was for us. |
| 13:08.48 | *** join/#asterisk UPMeduardo (~UPMeduard@tauro2.dit.upm.es) |
| 13:08.48 | tainted- | vaio = beautiful woman -- thin, pretty -- but useless |
| 13:08.51 | ManxPower | tainted-: lack of buffering on the zaptel card, non-realtime OS, crappy motherboard chipset. |
| 13:08.55 | johnnyb | Over a 10-minute period, we were getting consistent drops down to 99.2% |
| 13:09.22 | johnnyb | Generally it would stay at 99.987, but it would dip down far too frequently. |
| 13:09.22 | tainted- | except for bang bang bang |
| 13:09.36 | tainted- | bang bang |
| 13:09.38 | johnnyb | This continued with the new box until we got everything locked down. |
| 13:10.04 | johnnyb | I was pissed that on the cheap Dell servers you can't fix the interrupts better. |
| 13:10.19 | johnnyb | Why on earth does USB have to share an interrupt with EVERY SINGLE PCI SLOT!?!?!?!? |
| 13:10.32 | Makenshi | just disable usb |
| 13:10.35 | tainted- | johnnyb they have to save money somehow |
| 13:12.30 | johnnyb | Makenshi: I did, but was pissed that I had to. |
| 13:12.41 | *** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
| 13:12.41 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 13:12.45 | asteriskdz | what about TDM22B |
| 13:12.53 | asteriskdz | i w'd liek to connect it |
| 13:12.57 | asteriskdz | with pstn |
| 13:13.10 | asteriskdz | but it has 8pin like rj45 port |
| 13:13.31 | Makenshi | i dont see why rack servers need usb ports anyway |
| 13:13.59 | *** join/#asterisk ramtha (~tk@gw.01063telecom.de) |
| 13:14.04 | ramtha | hi |
| 13:14.20 | Nugget | usb is safer than ps2 for keyboard swapping. if you're too cheap to buy a real rm kvm, usb is a decent ghetto solution |
| 13:15.19 | asteriskdz | i'd like to connect it the TDM22 with PSTN line (analog so how can i do that !! because i find 4 port with 8PIN into my TDM22card |
| 13:15.45 | ramtha | how can i solve this: i have a quad span card with 4 e1 connected. each e1 is one group. in mein extension i have first g1, than g2, then g3 last g4 and than hangup. if g1 is full, the calls goes to g2 and so on. |
| 13:16.16 | ramtha | the problem, if i call a gsm phone and the person at the other side is buxy, asterisk trys a second call over g2 |
| 13:16.25 | ramtha | the the gsm phone rings again. |
| 13:16.30 | ManxPower | asteriskdz: as you would have found out if you were on the asterisk-users mailing list, or if you had spent 10 mins searching the asterisk-users mailing list archives on Google, you would know that you can plug an RJ-11 into the RJ-45 ports on the TDM cards just fine. Be careful you plug them straight as they can twist and jam if you are not careful. |
| 13:16.50 | ramtha | how can i do loadbalancing on the e1 without this result? |
| 13:17.10 | ManxPower | ramtha: that's because you have a badly designed dialplan. |
| 13:17.33 | ManxPower | ramtha: You should check the value of ${DIALSTATUS} and determine what to do before trying to dialout again. |
| 13:18.02 | ManxPower | "show application dial" and /path/to/src/asterisk/docs/README.variables |
| 13:18.14 | *** join/#asterisk meppl (mephisto@pD9E68A93.dip.t-dialin.net) |
| 13:20.57 | ManxPower | ~mailinglist |
| 13:20.58 | jbot | rumour has it, mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
| 13:21.00 | ManxPower | ~docs |
| 13:21.01 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 13:21.12 | *** join/#asterisk Aze` (~aze@host229-162.pool80105.interbusiness.it) |
| 13:21.13 | *** join/#asterisk Jas_Williams (~jason@host217-44-216-142.range217-44.btcentralplus.com) |
| 13:23.23 | Aze` | SIP A <-> ROUTER <-> * <-> SIP B (SIP A can listen SIP B but SIP A cant listen SIP B) why (RTP and 5060 redirected on *)? |
| 13:24.53 | bjohnson | ramtha: you need the superdial macro |
| 13:25.05 | *** join/#asterisk iq (~iq@70-59-161-163.omah.qwest.net) |
| 13:25.12 | *** join/#asterisk Gand_DJ (~gandalf@ptr-207-54-104-24.ptr.terago.ca) |
| 13:25.34 | bjohnson | Aze`: you've just discovered why people avoid putting * behind a nat router |
| 13:25.43 | bjohnson | (and use SIP) |
| 13:26.44 | bjohnson | Aze`: the easiest fix is to replace the router with an * box |
| 13:27.05 | Makenshi | Nugget, the poweredge ps/2 ports emulate a keyboard so you don't even need a keyboard |
| 13:27.11 | ManxPower | Aze`: you can use tcpdump on the asterisk box to confirm RTP is indeed getting to the box |
| 13:27.16 | Makenshi | er a kvm |
| 13:28.02 | bjohnson | Makenshi: ssh can be used to connect from anywhere .. and don't even have a keyboard on the server |
| 13:29.07 | Makenshi | bjohnson, can't use it for changing bios though |
| 13:29.20 | *** join/#asterisk Signuts (~signuts@209.172.11.54) |
| 13:29.24 | Makenshi | bjohnson, the dell drac card gives you a remote console over the network through a dedicated rj45 port |
| 13:29.25 | tainted- | how often do u update bios |
| 13:29.37 | bjohnson | exactly |
| 13:29.42 | Makenshi | tainted-, whenever there's an update, or need to network boot, or cycle power.. |
| 13:29.45 | Makenshi | can be frequently |
| 13:29.55 | bjohnson | weird |
| 13:29.55 | echion | hi, i noticed a problem with asterisk... it seems it is loosing a big percent of the dtmf sent, anyone know about this problem? |
| 13:30.09 | Signuts | I'm having quite the task access variables in a macro, the macro is called using M() in Dial() , on asterisk 1.0.7, is my only solution to upgrade? |
| 13:30.13 | tzanger | echion: what codec, what channel driver |
| 13:30.14 | tainted- | echion press buttons harder |
| 13:30.20 | Makenshi | or say when you are installing a new server |
| 13:30.22 | ManxPower | echion: that's usually a dtmfmode mismatch problem |
| 13:30.23 | Makenshi | you want to net boot |
| 13:30.31 | SuPrSluG_ | bjohnson: can yu use superdial w/ dundi? |
| 13:30.33 | Makenshi | once its cabled in you can do a remote os installation |
| 13:30.38 | echion | using rfc2833, alaw |
| 13:30.44 | bjohnson | echion: I have dtmf problems with other peoples IVRs |
| 13:30.46 | echion | sip -> sip |
| 13:30.46 | tainted- | echion make sure u have inband |
| 13:30.49 | ManxPower | Signuts: no, your solution is to report it as a bug |
| 13:30.56 | tzanger | tainted-: uh, why? |
| 13:30.56 | tainted- | echion with u/alaw |
| 13:31.05 | tzanger | rfc2833 should work with any codec |
| 13:31.06 | ManxPower | echion: WHAT is using rfc2833? Asterisk or the phone |
| 13:31.07 | bjohnson | SuPrSluG_: I don't know .. isn't dundi just another lookup .. like e164 |
| 13:31.11 | rg1 | can anyone help me - i have an X100P installed on linux, and i just got a trunk line installed by SWB configured as loop-start - i dial the trunk line# and i get a busy - nothing in the "full" log |
| 13:31.18 | ManxPower | tainted-: you don't have to use inband with ulaw or alaw. |
| 13:31.27 | tainted- | i know u don't have to |
| 13:31.28 | tzanger | but everything in the SIP path shoul dbe set the same. asterisk, the phone, and the terminating provider |
| 13:31.41 | tainted- | i just have better luck with inband w/ u/alaw |
| 13:31.51 | echion | ManxPower, well the phone is talking to a softswitch which is using rfc2833, which is talking to asterisk which is using rfc2833 which talkes to a audiocodes which is using rfc2833 |
| 13:32.07 | ManxPower | echion: you should confirm that with sip debug |
| 13:32.12 | *** join/#asterisk [Outcast] (~knoppix@c-24-218-94-11.hsd1.ma.comcast.net) |
| 13:32.14 | echion | I see the RTP Events going into the asterisk, but not going out... |
| 13:32.38 | ManxPower | echion: you see DTMF events being received by Asterisk, but not sent by Asterisk? |
| 13:32.48 | [Outcast] | has anyone had problems with chanspy crashing asterisk? |
| 13:33.00 | tainted- | i think he means dtmf not being passed through asterisk |
| 13:33.02 | *** part/#asterisk Patrick^ (~patrickm@birch4.mountaincable.net) |
| 13:33.08 | echion | ManxPower, exactly |
| 13:33.15 | SuPrSluG_ | bjohnson: just don't want it to loop.i looked at it and it's nice. good job.so i'd just put dundi lookup 1st?then other ways to connect? |
| 13:33.18 | ManxPower | echion: What is your Dial line? |
| 13:33.25 | c_k | anyone know any good IP phone emulation apps at all |
| 13:33.31 | ManxPower | PASTE it. |
| 13:34.00 | echion | like: aaa.aaa.aaa.aaa > bbb.bbb.bbb.bbb but not bbb.bbb.bbb.bbb > ccc.ccc.ccc.ccc where bbb.bbb.bbb.bbb is the asterisk |
| 13:34.06 | Gand_DJ | If I make changes to the extensions.conf file, must I reload asterisk, or will it read the updates on the fly |
| 13:34.28 | ManxPower | Gand_DJ: asterisk will never update unless you do a reload |
| 13:34.29 | zno | Gand_DJ: reload |
| 13:34.34 | ManxPower | reload does not terminate existing calls |
| 13:34.41 | rg1 | x100P - can it handle loop-start with wink-start signaling? |
| 13:35.06 | [Outcast] | Gand_DJ: you can execute a "extensions reload" |
| 13:35.10 | Makenshi | oh the other neat thing you can do with drac is to remote mount a floppy or cd, that you can boot the server with |
| 13:35.14 | ManxPower | rg1: The X100P only supports loop start (the default line type in most of the world). I don't believe it supports any other protocol. |
| 13:35.32 | ManxPower | echion: I'm still waiting for your paste of the Dial command you are using. |
| 13:35.41 | rg1 | manxpower - you ever deal with phone-co "Trunck" lines? |
| 13:36.05 | Gand_DJ | Hrm... I'm trying to get Eyebeam to support parking.... but it says my network doesn't support call parking |
| 13:36.06 | ManxPower | rg1: "trunk" can mean MANY different things. |
| 13:36.15 | *** join/#asterisk Patrick^ (~patrickm@birch4.mountaincable.net) |
| 13:36.26 | ManxPower | rg1: The telco-talk for the line you want for the X100P is "1FB" if it's for business. |
| 13:36.48 | tainted- | what does 1FB stand for |
| 13:36.49 | ManxPower | I believe "1FB" us the telco USOC. (Universal Service Ordering Code) |
| 13:36.54 | ManxPower | 1 flatrate Business |
| 13:37.09 | rg1 | ManxPower: I ordered a "trunk" direct-inward to a station line DID |
| 13:37.10 | ManxPower | i.e. POTS |
| 13:37.18 | tainted- | is that basically one unlimited analog line |
| 13:37.25 | ManxPower | rg1: that doesn't work with the X100P and may not work with the TDM400P |
| 13:37.36 | znoG | mmm, on normal ring (0,0,0) caller ID works, on my distinctive ring (367,17,0) it doesn't. Any ideas why or is this something the telco has to enable? |
| 13:37.50 | ManxPower | tainted-: yup, but 1FB means nothing else, not wink, not DID, etc. |
| 13:38.14 | tainted- | so how does the DID get assigned to the 1FB |
| 13:38.26 | rg1 | MaxPower/tainted: Doesn't the x100p handle DID (where it recognizes the 4-digit #) |
| 13:38.32 | tainted- | is that done by the telco? |
| 13:38.48 | *** join/#asterisk Blackvel (~blackvel@dsl-082-082-056-074.arcor-ip.net) |
| 13:38.53 | ManxPower | rg1: I just said that it didn't. |
| 13:39.11 | rg1 | ManxPower: sorry, just saw that |
| 13:39.11 | vaewyn | x100p are stupid cards... be glad they usually get callerid :P |
| 13:39.21 | ManxPower | rg1: As I understand it DID is only supported by Digium T-1/E-1 cards. |
| 13:39.23 | tainted- | so how do DIDs get assigned to channels in a PRI line |
| 13:39.34 | ManxPower | tainted-: Huh? |
| 13:39.37 | *** join/#asterisk iq_ (~iq@70-59-161-163.omah.qwest.net) |
| 13:39.44 | [Outcast] | tainted-: that is handled on the telco side |
| 13:39.46 | ManxPower | PRI runs over T-1/E-1 |
| 13:40.06 | *** join/#asterisk mjmac (~mjmac@cpe-24-198-203-132.maine.res.rr.com) |
| 13:40.07 | asteriskdz | any one know how to configure R2CAS signaling with E1 |
| 13:40.08 | asteriskdz | ?? |
| 13:40.11 | bjohnson | SuPrSluG_: I don't use dundi |
| 13:40.15 | tainted- | i'm asking -- how does telco know to route DID to your PRI |
| 13:40.21 | bjohnson | SuPrSluG_: but I use e164 like this: |
| 13:40.34 | tainted- | is the DID bought beforehand and then basically assigned to the PRI? |
| 13:40.35 | ManxPower | tainted-: they configure it in their switch |
| 13:40.45 | ManxPower | tainted-: no. |
| 13:40.52 | tainted- | so DID is purchased separate from PRI |
| 13:40.58 | rg1 | MaxPower/tainted: I'm on with SWBell and these guys are as dumb as rocks - no one there knows anything about the products THEY sell |
| 13:41.03 | bjohnson | exten => s,4,EnumLookup(${tfnumber}) |
| 13:41.03 | bjohnson | exten => s,5,GotoIf($[${ENUM:0:3} = IAX]?6:7) ; SIP behind NAT not working |
| 13:41.03 | bjohnson | exten => s,6,Macro(superdial,${ENUM},,,,voip,${MAXVOIPCALLS},,,e164) |
| 13:41.16 | asteriskdz | any one know how to configure R2CAS signaling with E1 |
| 13:41.19 | Jas_Williams | asteriskdz Asterisk does not support R2Cas over E1 as far as I know only EuroISDN |
| 13:41.23 | ManxPower | tainted-: When you order the PRI you get DIDs with it. If you want more you can order them from the telco. |
| 13:41.26 | vaewyn | rg1: hahaha.. welcome to telcos around the world :P |
| 13:41.27 | echion | ManxPower, sure... coming... |
| 13:41.38 | asteriskdz | no |
| 13:41.44 | onlyI | rg1 telco staff ;) |
| 13:41.46 | ManxPower | Jas_Williams: chan_unicall supports R2 for some contries. |
| 13:41.47 | bjohnson | tainted-: if you're getting a new PRI .. usually you buy some DID at the same time |
| 13:41.49 | asteriskdz | som one has installed it with r2cas |
| 13:41.56 | onlyI | rg1 same here with Bell Canada |
| 13:42.02 | Blackvel | who knows manager API REDIRECT? Do I always have to specifiy the exten? What if I like to redirect an AGI client to an agent (e.g channel ZAP/100, SIP/100) |
| 13:42.02 | Blackvel | ? |
| 13:42.10 | asteriskdz | in roumani |
| 13:42.12 | asteriskdz | in roumania |
| 13:42.18 | echion | ManxPower, exten => _X.,5,Dial(SIP/${routing_target}) |
| 13:42.22 | tainted- | is it possible to get DID to your PRI from another locale (area code) |
| 13:42.36 | bjohnson | tainted-: they have defined areas |
| 13:42.37 | jakepdev | is there an IsNumeric cmd for asterisk? |
| 13:42.45 | vaewyn | tainted-: depends on your provider |
| 13:42.49 | jakepdev | don't want the * in the string |
| 13:42.50 | ManxPower | echion: I need to know the value of ${routing_target} |
| 13:42.56 | bjohnson | tainted-: based on their cabling system |
| 13:43.01 | ManxPower | echion: just paste the Dial line from the console when you make a call. |
| 13:43.12 | tainted- | so the big guys like XO and L3 have PRIs basically everywhere then |
| 13:43.18 | ManxPower | tainted-: that totally depends on your telco. |
| 13:43.24 | vaewyn | tainted-: yep... or peering agreements |
| 13:43.43 | ManxPower | echion: I have a lot to do and cannot help you for much longer. |
| 13:43.50 | echion | ManxPower, it is always the number@gateway |
| 13:43.50 | tainted- | peering agreements = trading minutes? |
| 13:44.12 | vaewyn | tainted-: usually trading endpoints... but... same idea |
| 13:44.20 | Gand_DJ | anyone using FWD for incoming? For some reason I'm still getting a busy signal when I call myself. |
| 13:44.20 | ManxPower | echion: if you dial by host or ip then anything except for (maybe) [general] is ignored in sip.conf |
| 13:44.55 | echion | dtmfmode is rfc2833 in general in sip.conf |
| 13:45.00 | ManxPower | echion: you can also do a "sip show channel blah" to see stuff, including dtmfmode, of an actuall call. |
| 13:45.01 | tainted- | so are toll frees basically the same thing as DIDs with a per/minute surchage then? |
| 13:45.04 | vaewyn | tainted-: usually agreements like ... 100 concurrent calls from our pool for 100 concurrent calls from yours |
| 13:45.11 | vaewyn | tainted-: yep |
| 13:45.15 | ManxPower | tainted-: sort of. |
| 13:45.16 | *** join/#asterisk Ciber (~Ciber@70.84.122.194) |
| 13:45.20 | tainted- | wow |
| 13:45.26 | Ciber | man |
| 13:45.27 | echion | ManxPower, and that's the only dtmfmode parameter in sip.conf |
| 13:45.29 | Ciber | Apr 28 09:44:27 NOTICE[418]: chan_sip.c:4035 sip_reg_timeout: -- Registration for '6462050612@sip.broadvoice.com' timed out, trying again |
| 13:45.31 | tainted- | this stuff used to seem so mysterious |
| 13:45.32 | Ciber | Apr 28 09:44:27 NOTICE[418]: chan_sip.c:6819 handle_response: Failed to authenticate on REGISTER to '<sip:6462050612@sip.broadvoice.com>;tag=as1ca826d7' |
| 13:45.35 | Ciber | i'm gonna blow this thing up |
| 13:45.36 | tainted- | and now it seems so clear |
| 13:45.50 | ManxPower | echion: some options in sip.conf (like qualifty=) don't work in [general] |
| 13:46.12 | ManxPower | Ciber: stop spamming us and complain to Broadvoice. |
| 13:46.15 | vaewyn | would rock if qualify did :P |
| 13:46.25 | Ciber | on hold now :P |
| 13:46.44 | tainted- | PRIs ride on T1/E1, uses TDM and is separated into voice / data signalling channels to provide DID |
| 13:46.48 | Blackvel | ciber: insecure=very? |
| 13:46.53 | Ciber | yup |
| 13:47.01 | tainted- | is that right? |
| 13:47.30 | Gand_DJ | hrm, how does one stop sip debug |
| 13:47.37 | vaewyn | sip no debug |
| 13:47.40 | Gand_DJ | :) |
| 13:47.50 | Gand_DJ | thx. |
| 13:47.57 | vaewyn | np |
| 13:48.00 | tainted- | can an ISP like speakeasy provide PRI over T1? or do they have to be a clec/ilec w/ proper ISDN equipment |
| 13:48.05 | Blackvel | http://www.voip-info.org/wiki-Asterisk+Manager+API+Action+Redirect do I always have to use an exten here? Can I make use of the more direct command like Dial(ZAP/g1/number)? |
| 13:48.12 | Gand_DJ | Trying to figure out why fwd incoming is not worknig |
| 13:48.26 | jakepdev | blackvel |
| 13:48.33 | Blackvel | hey ;) |
| 13:48.35 | ManxPower | If you don't have ISDN you can't talk ISDN |
| 13:48.36 | jakepdev | don't use dial through manager - it blocks |
| 13:48.58 | jakepdev | it's a bad idea |
| 13:49.07 | echion | ManxPower, DTMF Mode: rfc2833 |
| 13:49.24 | echion | ManxPower, from sip show channel |
| 13:49.32 | tainted- | so if i got a PRI line with one DID, i'd essentially have 23 lines on that DID? |
| 13:49.43 | vaewyn | yep |
| 13:49.55 | vaewyn | as long as the telco is bright :P |
| 13:50.00 | onlyI | 23 B + 1 D |
| 13:50.01 | [Outcast] | hehe |
| 13:50.16 | Gand_DJ | Expensive for 1 did only... Pri is like $750-$1000 CDN.. heh |
| 13:50.21 | Gand_DJ | monthly |
| 13:50.22 | Gand_DJ | :( |
| 13:50.32 | tainted- | that's not expensive |
| 13:50.41 | [Outcast] | vaewyn: is most cases the tel would make you buy those other 22 DID and charge you for roll-over |
| 13:50.42 | vaewyn | depends how many concurrent calls you need on that number |
| 13:50.44 | tainted- | 750 CDN is like 0.002 USD |
| 13:50.51 | [Outcast] | s/is/in |
| 13:50.52 | Gand_DJ | lol |
| 13:51.02 | onlyI | [Outcast] yes sir |
| 13:51.04 | ManxPower | echion: calls have two legs. what about the other leg of the call? |
| 13:51.17 | echion | ManxPower, that's the answer on both |
| 13:51.39 | vaewyn | [Outcast]: SBC doesn't... they just assume you want single DID pool... they do charge you more for single DIDs then they charge for them in bundle but... |
| 13:51.40 | tainted- | [Outcast] what does roll over mean in that context.. signalling a different channel? |
| 13:52.01 | ManxPower | echion: I have no more suggestions |
| 13:52.05 | onlyI | tainted- when busy rollover next B channel |
| 13:52.25 | tainted- | onlyI that can't be done in asterisk then? |
| 13:52.30 | tainted- | i guess not |
| 13:52.35 | [Outcast] | tainted-: if the first DID is busy the next one in the group will ring |
| 13:52.40 | *** join/#asterisk UK_Mister (~fred@host81-137-167-89.in-addr.btopenworld.com) |
| 13:52.44 | tzanger | uh |
| 13:52.45 | onlyI | tainted- hmmmm not that i know of |
| 13:52.48 | tainted- | since the telco would see a busy channel before it even reaches asterisk |
| 13:52.49 | tzanger | that's not quite how it works in PRI land |
| 13:52.51 | onlyI | hey tzanger |
| 13:53.13 | tainted- | ManxPower did u suggest inband? ;) |
| 13:53.21 | tzanger | any DID can "consume" any number of B channels in simultaneous calls |
| 13:53.33 | UK_Mister | hello all, can anyone recommend a good SIP handset to work with asterisk? |
| 13:53.46 | tainted- | tzanger yea but [Outcast]'s point is that telcos charge extra for that |
| 13:53.51 | tzanger | onlyI: hey -- I haven't forgotten about you. I am waitin gfor more stock now, I had someone come and buy my last two cards irrespective of their "age" |
| 13:54.09 | tzanger | I didn't want to sell you older cards (especially FXO modules) |
| 13:54.11 | *** join/#asterisk icexx (~jj@213.170.75.191) |
| 13:54.17 | tzanger | tainted-: they do?? |
| 13:54.22 | onlyI | tzanger it's ok my got one comming my way |
| 13:54.38 | tzanger | ok I kind of figured you got tired of waiting on my slow ass, it's usually not like this :-( |
| 13:54.38 | onlyI | tzanger i'll order some more |
| 13:54.45 | tainted- | <[Outcast]> vaewyn: is most cases the tel would make you buy those other 22 DID and charge you for roll-over |
| 13:54.52 | tzanger | uh |
| 13:54.59 | tzanger | DID != B chan |
| 13:55.09 | *** join/#asterisk nix000 (~dos000@66.11.191.156) |
| 13:55.12 | tzanger | DID is just a number that appears on a PRI |
| 13:55.18 | tzanger | you can have 100 DIDs on a single 23B+D PRI |
| 13:55.20 | onlyI | tzanger it's ok like i said next one is for you i'll let you know few days in advance |
| 13:55.22 | tzanger | you can only take 23 calls at a time |
| 13:55.32 | tainted- | no the scenario was purchasing one DID with one PRI line |
| 13:55.33 | Ahrimanes | anyone here have SER routing to asterisk working? |
| 13:55.34 | tzanger | onlyI: thanks, I appreciate it |
| 13:55.40 | *** join/#asterisk drbrown (~chatzilla@65.121.240.182) |
| 13:55.42 | Fraeggl | hello |
| 13:55.47 | drbrown | hi |
| 13:55.54 | tzanger | tainted-: in that case you still get 23 simultaneous calls to that # unless you can wrangle fewer Bchan |
| 13:55.55 | onlyI | tzanger your website is up ??? |
| 13:56.03 | tzanger | onlyI: no I have no website for this |
| 13:56.07 | tainted- | [Outcast] was saying that the telco would make u purchase 22 more |
| 13:56.09 | MattB2 | UK peeps - do I need to setup anything special to get X100P working in UK? |
| 13:56.10 | drbrown | Has anyone used a cf card instead of an Hd to run asterisk? |
| 13:56.12 | onlyI | tzanger bad boy |
| 13:56.12 | tzanger | onlyI: I have no intenetion of doing this on alarge scale |
| 13:56.17 | MattB2 | it's cutting off as soon as the remote party picks up |
| 13:56.30 | onlyI | tzanger roger sir |
| 13:56.47 | tainted- | and charge u extra for signalling different B channels on busy |
| 13:56.55 | tzanger | tainted-: what?? |
| 13:56.56 | tzanger | no |
| 13:57.03 | tzanger | there is no "busy" if there are Bchan avail |
| 13:57.16 | BuffPuff | oh, now i get it!!!!!!!!!!!! u americans got less channels per pri... :-) |
| 13:57.22 | Fraeggl | i have a running asterisk with hfc ISDN card and zaphfc driver on a ptmp line. my problem: i dont get the called MSN via any predefined variable in extensions.conf |
| 13:57.22 | tainted- | that's the impression i got |
| 13:57.28 | BuffPuff | im slow today |
| 13:57.33 | Gand_DJ | Heh.... it appears to properly use Park via #700, you have to use the T comment in conjunction of tr |
| 13:57.37 | tainted- | BuffPuff that's right.. 23 per line |
| 13:57.43 | Fraeggl | is there a way to route incomming calls based on MSN ? |
| 13:57.44 | tzanger | the switch will route a DID to your PRI if there are Bchans avail. There is no concept of "rollover" or "hunting" in a PRI unless you have multiple ones |
| 13:57.47 | [Outcast] | tainted-: I said in most cases. I got i feeling the sales guy would really like to sell as many dids as possible to bump up his commmission |
| 13:58.09 | icexx | anybody has problems asterisk+iconnecthere? |
| 13:58.12 | tzanger | BuffPuff: yes, we have 23B+D (T1 is 24 channels) -- you guys run PRI on E1 which is 32 channels, but you get 30B+D+something |
| 13:58.24 | tzanger | [Outcast]: Bell will sell you DIDs in blocks of 30 only |
| 13:58.30 | [Outcast] | k |
| 13:58.31 | tzanger | [Outcast]: Bell Canada that is |
| 13:58.40 | tzanger | if you need 31 DIDs, you buy 60 |
| 13:59.06 | *** join/#asterisk Blackvel (~blackvel@dsl-213-023-029-124.arcor-ip.net) |
| 13:59.12 | nix000 | is zaptel supported under kernel 2.6 ? |
| 13:59.16 | tzanger | nix000: of course |
| 13:59.39 | nix000 | tzanger, till latest ? |
| 13:59.45 | [Outcast] | tzanger: 30 did for only 23 channels that make sense...x( |
| 13:59.47 | christo | If a user presses '9' followed by '02072342342' I want asterisk to pass that call to Zap/g1 as follows: exten => _9XXXXXXXXXXX,1,Dial(Zap/g1/${EXTEN}:1) I have this in the [default] section in extensions.conf, but it's not working at all. What else should I be looking out for? |
| 13:59.48 | tzanger | <PROTECTED> |
| 13:59.56 | tzanger | [Outcast]: again, 23 channels != 23 phone numbers |
| 14:00.01 | tzanger | 23 channels = 23 simultaneous calls |
| 14:00.08 | tzanger | from up to 30 different numbers |
| 14:00.10 | tzanger | makes perfect sense |
| 14:00.21 | tzanger | like I said you can have 100 DIDs appear on a 23B+D PRI |
| 14:00.24 | nix000 | tzanger, just checkin in case i am banging my head on an unsupported system. |
| 14:00.28 | tzanger | you will only be able to accept 23 calls at once though |
| 14:00.40 | tainted- | tzanger how much for 30 DIDs in CDN? |
| 14:00.40 | tzanger | nix000: nope, I'm runnign 2.6.10 and 2.6.11 |
| 14:00.44 | [Outcast] | tzanger: not if you dealing with direct dial in larger setups |
| 14:00.47 | tzanger | tainted-: I think $30/mo |
| 14:00.49 | SuPrSluG_ | christo: should be {EXTEN:1} |
| 14:00.53 | ManxPower | _9XXXXXXXXXXX,1,Dial(Zap/g1/${EXTEN:1}) Notice the placement of the :1 |
| 14:00.54 | nix000 | tzanger, thanks |
| 14:00.59 | tzanger | [Outcast]: don't change the parameters of the discussion |
| 14:01.06 | tainted- | tzanger a buck each? that is cheap! |
| 14:01.11 | Fraeggl | does someone know how to get the msn in extensions.conf for an isdn ptmp line (zaphfc), predefined variables are empty, is there a trick ? |
| 14:01.14 | tzanger | in fact it's the same with larger systems |
| 14:01.30 | UK_Mister | can anyone recommend a good handset, i've used grandsteam ones before but want some feature keys etc |
| 14:01.30 | tzanger | I have 5 DS3s -- there's only about a dozen DIDs on all 5 though |
| 14:01.48 | guyee | Does NE1 know of a VoIP provider I can test H.323 with? My prov doesn't support SIP so I need it very badly... |
| 14:01.59 | tzanger | but you could just as easily have an entire NXX routed to them.. you'd only be able to accept as many calls as you had free channels though |
| 14:02.06 | icexx | guyee: there are none ;) |
| 14:02.07 | *** join/#asterisk sivana (~sivana@mixdown.ca) |
| 14:02.08 | tainted- | tzanger that is a lot of line |
| 14:02.10 | SuPrSluG_ | UK_Mister:many people like polycom ip 500. |
| 14:02.18 | tzanger | tainted-: dialup ISP |
| 14:02.19 | guyee | icexx: thx :/ |
| 14:02.28 | UK_Mister | i cant get a polycom anywhere in the UK :( |
| 14:02.29 | [Outcast] | tzanger: what is the purpose of those lines |
| 14:02.35 | UK_Mister | any others? |
| 14:02.40 | Blackvel | http://www.voip-info.org/wiki-Asterisk+Manager+API+Action+Redirect do I always have to use an exten here? Can I make use of the more direct command like Dial(ZAP/g1/number)? |
| 14:03.00 | tainted- | so PRI can go over DS3 as well then? |
| 14:03.13 | tzanger | [Outcast]: dialup ISP but I'm pushing calls through them shortly (doing inbound only atm for testing) |
| 14:03.16 | tzanger | tainted-: of course |
| 14:03.19 | icexx | peoples, anybody tried asterisk with iconnecthere and had error 480 2 seconds into the conversation? |
| 14:03.31 | tainted- | is PRI basically a signalling protocol? |
| 14:03.36 | tzanger | a DS3 = 28 DS1. If you're runnign PRI on those DS1s then you've got 28 PRIs in the DS3 |
| 14:03.50 | tainted- | OMG |
| 14:03.50 | Fraeggl | is there anyone working with an hfc isdn-Card (TE) on an german euroisdn ptmp (mehrgeraeteanschluss) line ? |
| 14:03.55 | tzanger | now you play games like NFAS to avoid duplicating so many D channels |
| 14:03.58 | tainted- | 28 * 23 * 5? |
| 14:04.10 | tzanger | it's more than that |
| 14:04.21 | [Outcast] | anyways gentlemen and ladies, i need to get back work have fun |
| 14:04.24 | tzanger | we use NFAS so we bind 7 PRIs' D channels into 1 |
| 14:04.24 | christo | SuPrSluG - these examples all use a dollar sign: http://www.voip-info.org/wiki-Asterisk+variable+hangupcause |
| 14:04.29 | christo | eg exten => 91NXXNXXXXXX,2,Macro(dial-result,${NUFONE}/${EXTEN:1}) |
| 14:04.42 | tainted- | is NFAS some kind of D channel compression protocol? |
| 14:04.42 | tzanger | but yeah for basic numbres it's 28*24*5 |
| 14:04.51 | tzanger | tainted-: not compressio |
| 14:04.59 | tzanger | remember a Dchan is very low bandwidth and you're allocating 64k to it |
| 14:05.04 | bkw_ | tainted-, no |
| 14:05.07 | tainted- | right |
| 14:05.17 | icexx | mr. tainted- ;) da experto, have you heard of problems with iconnecthere? |
| 14:05.18 | bkw_ | you can run a whole DS3 on two D channels |
| 14:05.19 | tzanger | so you can bind multiple Dchans to a single one and recoup that 64k timeslot for a voice channel on other PRis |
| 14:05.27 | tzanger | bkw_: I've never seen it that high |
| 14:05.30 | tainted- | icexx what is that? |
| 14:05.47 | icexx | asterisk + iconnecthere, getting 480 2 secs into conversations |
| 14:05.51 | bkw_ | you gain 28 more channels on a DS3 if you use NFAS |
| 14:05.54 | tzanger | bkw_: I've only seen binding 7 DS1s (i.e. a DS2) to a single Dchan |
| 14:05.56 | icexx | wtf may this be |
| 14:05.57 | christo | aaan I gettit |
| 14:06.10 | Gand_DJ | cool... Eyebeam allows you to do assisted call transfers |
| 14:06.11 | Gand_DJ | :) |
| 14:06.13 | SuPrSluG_ | christo: I know. I was just pointing to where the problem lies. |
| 14:06.15 | bkw_ | the first 8 T1's get a D channel.. the last 20 get a D channel. |
| 14:06.42 | Gand_DJ | not true assisted... but a form of it :) |
| 14:06.42 | tzanger | bkw_: whoa |
| 14:06.45 | tzanger | that's weird |
| 14:06.47 | icexx | 480 from iconnecthere (voice!!!!!!!, not via SIP) |
| 14:06.57 | tzanger | I dind't think you could put that many PRIs in a single dchan |
| 14:07.01 | bkw_ | yep |
| 14:07.02 | bkw_ | you sure can |
| 14:07.05 | tzanger | you must have some terrible call setup times at busy times though |
| 14:07.09 | bkw_ | nope |
| 14:07.22 | bkw_ | you don't realize that the d channel traffic is so small it don't m atter |
| 14:07.26 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 14:07.26 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:07.34 | tzanger | Dchan is 8kbps is it not? 160kbps into a 64kbps channel? |
| 14:07.45 | tzanger | bkw_: that's why I said "at busy times" :-) |
| 14:07.55 | tainted- | is NFAS implemented in hardware or software? |
| 14:07.59 | tzanger | tainted-: software |
| 14:08.00 | bkw_ | no its 64k |
| 14:08.02 | tzanger | asterisk supports it |
| 14:08.02 | bkw_ | on a T1 |
| 14:08.07 | bkw_ | and 16k on a BRI |
| 14:08.18 | bkw_ | its a full DS0 |
| 14:08.21 | bkw_ | unlike BRI |
| 14:08.52 | tzanger | bkw_: right I understand that but still |
| 14:08.56 | tzanger | 160kbps in a 64kbps channel |
| 14:08.57 | [Outcast] | bkw_: are you using asterink on all of those lines? |
| 14:08.59 | Ciber | alrighty got incoming working |
| 14:09.07 | Ciber | getting a busy signal on outgoing |
| 14:09.07 | Blackvel | My originate fails. Why? http://www.pastebin.com/277095 |
| 14:09.08 | tainted- | ciber what was the issue? |
| 14:09.09 | tzanger | if you had to do a call setup/teardown on a hundred channels at once you'd have delays |
| 14:09.14 | Ciber | wrong password |
| 14:09.21 | Ciber | they seem to have two passwords :P |
| 14:09.24 | bkw_ | [Outcast], no our's are not configured liek that right now.. i'm just saying I have seen/done configs like that in the past |
| 14:09.27 | tainted- | Ciber lol |
| 14:09.30 | bkw_ | this is a TNT btw |
| 14:09.30 | nix000 | what do you guys use as distro when running asterisk on servers ? |
| 14:09.30 | Ciber | one for the site and one for phone *rolls eyes* |
| 14:09.34 | tzanger | I realize that dchan traffic is very small and infrequent but as I said if you had to do it for a ton of channels at once you'd be hosed |
| 14:09.39 | tzanger | bkw_: yeah we use TNTs too |
| 14:09.42 | christo | SuPrSluG - fixed it with: exten => _9XXXXXXXXXXX,1,Dial(Zap/g1/${EXTEN:1},120) |
| 14:09.48 | christo | I was missing the 120 bit at the end |
| 14:09.50 | tzanger | I had no idea you could do 20DS1 to a single D |
| 14:09.51 | bkw_ | tzanger, no |
| 14:10.05 | christo | er.... |
| 14:10.14 | christo | so what's the '120' for at the end of exten => _9XXXXXXXXXXX,1,Dial(Zap/g1/${EXTEN:1},120) ? |
| 14:10.18 | bkw_ | tzanger, you sure can |
| 14:10.20 | tainted- | so w/o TNT hardware u wouldn't be able to do this kind of NFAS work? |
| 14:10.20 | christo | timeout I guess |
| 14:10.28 | bkw_ | tainted-, doubt it |
| 14:10.29 | bjohnson | well .. I just can't get suitable quality over internet for us to use voip that way |
| 14:10.33 | vaewyn | yeah... for modem pools we usually do 95B + 1D for call centers 47B+1D |
| 14:10.38 | SuPrSluG_ | christo: 120 is a long time to ring phones |
| 14:10.39 | tzanger | tainted-: I did NFAS with AS5248s... of course you could only do 2 PRI NFAS then :-) |
| 14:11.29 | SuPrSluG_ | christo: 120 seconds to ring phone |
| 14:11.31 | bkw_ | tangel, have you seen the size of the setup and tear down messages on a D channel? |
| 14:11.34 | Zeeek | christo show application dial |
| 14:11.39 | bkw_ | its NOTHING really |
| 14:11.44 | nix000 | is digium cards able to act as modems so ppl can dial into hem ? |
| 14:11.46 | Gand_DJ | Ok.... that's just fuq'd up.... I setup a DID entry in * for my FWD number, and now incoming works... but if I take the DID out.... I get a busy signal. |
| 14:11.59 | tainted- | bkw_ yea but he's talking during busy times.. |
| 14:12.03 | tzanger | nix000: if you have enough processor and a v92 softmodem channel driver, sure |
| 14:12.09 | bkw_ | still its not an issue |
| 14:12.09 | tainted- | bkw_ with the SETUP/TEARDOWN overhead |
| 14:12.19 | bkw_ | at all |
| 14:12.20 | tzanger | bkw_: I'm just paintint corner cases here :-) |
| 14:12.40 | *** part/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 14:12.43 | Blackvel | Who could explain me this: http://www.voip-info.org/wiki-Asterisk+Manager+API+Action+Originate? |
| 14:12.53 | bkw_ | Blackvel, can you not read it? |
| 14:13.00 | nix000 | tzanger, 1) where you get the soft modem 2)what kind of horsepower for the quad e1 |
| 14:13.00 | Blackvel | I can, I can not understand :( |
| 14:13.06 | MikeJ[Laptop] | hehe |
| 14:13.07 | tzanger | nix000: don't bother trying |
| 14:13.15 | tzanger | nix000: if you are asking those questions you won't be able to do it |
| 14:13.19 | Makenshi | what about t3? |
| 14:13.26 | bkw_ | nix000, 9600 is about all you can do legally |
| 14:13.32 | bkw_ | the rest of the modem stuff has patents on it |
| 14:13.34 | tzanger | bkw_: well without closed source |
| 14:13.49 | nix000 | bkw_, thats al i need for now btw ! |
| 14:14.08 | tainted- | tzanger the TNT is basically a media gateway right? |
| 14:14.12 | tzanger | tainted-: correct |
| 14:14.12 | bkw_ | nix000, spandsp can do it but you'll have to do most if not all the hardwork your self |
| 14:14.31 | vaewyn | reminds me... spandsp playing well with -head nowdays? |
| 14:14.32 | tzanger | yeah but a quad E1 of 9600 baud modems flying ... that's some beefy processor |
| 14:15.17 | bkw_ | vaewyn, sure ship me one please |
| 14:15.19 | vaewyn | I think the cooling bill on that would bankrupt me :} |
| 14:15.24 | MikeJ[Laptop] | vaewyn, I have spandsp running on recent head, but have not updated in a couple weeks or so |
| 14:15.25 | bkw_ | my address is on my whois records for bkw.org |
| 14:15.42 | tainted- | are there F/OSS implementations of media gateways? |
| 14:15.42 | MikeJ[Laptop] | hehe |
| 14:15.56 | nix000 | bkw_, all i want is for the modems to pass few parameters for authentication. i am very confortable if i can stablis an ip connection. |
| 14:15.56 | bkw_ | vaewyn, the thoughts of 8way make me feel funny inside |
| 14:16.02 | vaewyn | bkw_: hehehe... I would love to... but we only ordered one... and I gets it.... my precccciousss |
| 14:16.13 | tzanger | tainted-: asterisk? |
| 14:16.16 | bkw_ | vaewyn, well say it didn't get there.. have them ship another one |
| 14:16.19 | vaewyn | bkw_: heck yeah... like climbing the rope in gym class ;P |
| 14:16.26 | bkw_ | vaewyn, ya really |
| 14:16.29 | vaewyn | bkw_: hahahaha |
| 14:16.32 | tainted- | tzanger i mean on larger scale |
| 14:16.42 | tainted- | near same class as TNT perhaps? |
| 14:16.47 | UK_Mister | whats a good handset to use, i cant get Polycom IP500's in the UK atm |
| 14:17.04 | nix000 | bkw_, where is the hard work ? |
| 14:17.09 | tainted- | i think asterisk tops around 100 concurrent calls if not doing transcoding correct? |
| 14:17.16 | ManxPower | UK_Mister: Polycoms. 8-) Ciscos too, but they end up being significantly more expensive. |
| 14:17.20 | bkw_ | nix000, actually creating the code for the modem to do data |
| 14:17.31 | bkw_ | tainted-, NOT |
| 14:17.37 | vaewyn | Polycoms rock |
| 14:17.38 | tzanger | tainted-: well there is the DS3000 card now... but you'd need a cluster of boxes since there's no DSP |
| 14:17.39 | bkw_ | tainted-, you're brain has been warped by bad media |
| 14:17.43 | ManxPower | UK_Mister: Also SIPura SPA-841 if you accept their limitations. |
| 14:17.54 | tainted- | i mean one box |
| 14:17.57 | bkw_ | tzanger, wrong again.. all ulaw no need to transcode.. one box |
| 14:17.59 | tzanger | asterisk tops up at whatever the hardware starts to go at |
| 14:18.04 | vaewyn | tainted-: hehehe... keep going... you can run a DS3 through it easily :P |
| 14:18.05 | tzanger | bkw_: media gateway isn't all ulaw only |
| 14:18.08 | nix000 | bkw_, there is no implementation available even commercially ? i really have no time doing it myself. |
| 14:18.09 | Blackvel | What is this Manager API REDIRECT like? Can I use it without exten but use ZAP/G1/number as with DIAL? |
| 14:18.19 | tainted- | i rarely see ulaw coming through my box |
| 14:18.22 | bkw_ | tzanger, well in my eyes it is |
| 14:18.27 | *** join/#asterisk o_cee (~o_cee@h250n5c1o1095.bredband.skanova.com) |
| 14:18.27 | tainted- | its mostly 729/732 |
| 14:18.30 | bkw_ | I love ulaw |
| 14:18.30 | tainted- | 723 |
| 14:18.32 | UK_Mister | ManxPower, limitations? like what? |
| 14:18.42 | riksta | I love bkw_ |
| 14:18.48 | bkw_ | UK_Mister, 10mbit, 1 ethernet port |
| 14:18.51 | bkw_ | having to pay for two more lines |
| 14:18.54 | riksta | but...i'm only human :) |
| 14:18.54 | bkw_ | no backlight |
| 14:18.54 | tzanger | bkw_: media gateway does voice, data and probably video too... you ain't doing that on one box, especially if transcoding |
| 14:18.56 | *** join/#asterisk lunchbox08 (~geoff@64.128.43.66) |
| 14:19.01 | vaewyn | you make transcoding boxes... git rid of the g72x and make it ulaw :} |
| 14:19.07 | zoa | hey brian!!!! |
| 14:19.10 | tzanger | vaewyn: well that's what I was getting at |
| 14:19.16 | bkw_ | tzanger, true.. but I don't do video.. or data. |
| 14:19.20 | bkw_ | YET!! |
| 14:19.22 | bkw_ | zoa yes |
| 14:19.24 | ManxPower | UK_Mister: The SIPura SPA-841 does not support PoE, it does not have a second switch port to plug a PC into, it does not have a good speakerphone, the display is not backlit. However, it's better than any other < $100 phone. |
| 14:19.26 | zoa | hi hon! |
| 14:19.27 | bkw_ | I got something for YOU |
| 14:19.27 | tzanger | two head-end boxes with DS3000s in them (failover), feeding a cluster of transcode boxes behind them |
| 14:19.32 | vaewyn | zoa: PROTO! |
| 14:19.32 | zoa | AHA! |
| 14:19.32 | tainted- | vaewyn dedicated * boxes to do transcoding? |
| 14:19.39 | bkw_ | we need zoa level call testing on something |
| 14:19.40 | zoa | PROTO TO you TO |
| 14:19.46 | zoa | hehe |
| 14:19.47 | tzanger | no |
| 14:19.49 | tzanger | protozoa |
| 14:19.50 | zoa | what do you want ? |
| 14:19.51 | tzanger | that's what they were getting at |
| 14:19.51 | *** join/#asterisk Essobi (kstone@75.137.26.216.host.teledvance.com) |
| 14:20.00 | vaewyn | tainted-: yep... then your main 'switch' can handle thousands of calls if done right |
| 14:20.04 | zoa | maybe tonight stable sip jitter buffer pathes |
| 14:20.18 | ManxPower | tainted-: there are a fair number of boxes that take T-1/E-1 and spits out SIP. There are a small number that do DS-3. All of them are expensive. |
| 14:20.31 | tainted- | vaewyn but why bother transcoding if it is 729 end to end |
| 14:20.45 | lunchbox08 | anyone have any idea what kind of latency is added when turning on notransfer=yes inside IAX |
| 14:20.53 | Essobi | zoa How's that looking? |
| 14:20.55 | ManxPower | tainted-: the moment you hit the PSTN you have to transcode. |
| 14:20.56 | vaewyn | tainted-: if it is 729 end to end then you never transcode... and you can handle many many many calls |
| 14:21.02 | Gand_DJ | can someone who is able to call fwd #'s call me so I can see if music-on-hold works externally. It works between extensions.... and also see if you can dial the park #70 number when calling in from external sourse. |
| 14:21.09 | tzanger | tainted-: uh DS3 is ulaw/alaw |
| 14:21.21 | tainted- | i never hit analog |
| 14:21.24 | tainted- | it's all voip |
| 14:21.33 | tainted- | that's why i'm such a telephony newb |
| 14:21.44 | christo | Zeek - excellent thanks for that! |
| 14:21.45 | tzanger | tainted-: ahh... wlel technically PRI isn't analog :-) |
| 14:21.47 | tzanger | it's TDM |
| 14:21.54 | tainted- | well u know what i mean |
| 14:21.55 | tainted- | heh |
| 14:22.03 | tainted- | i never hit POTS/TDM |
| 14:22.14 | christo | Zeek - I was referring to the 'show application dial' I didn't realise that there were man pages available from the console |
| 14:22.19 | christo | ace! |
| 14:22.38 | tainted- | vaewyn okay so u put some * powerhouse boxes between PSTN and your VOIP boxes then |
| 14:22.49 | lunchbox08 | bkw question for you |
| 14:23.19 | vaewyn | tainted-: our setup is... phone -> transcoder * -> switch * -> T1 |
| 14:23.29 | Hymie | anyone try using a SMP system, and forcing all digium card interrupts to one CPU, so that other can be free of interrupt madness? |
| 14:23.33 | lunchbox08 | anyone have any idea what kind of latency is added when turning on notransfer=yes inside IAX |
| 14:23.36 | tainted- | ulaw/alaw -> * transcode box -> 729 -> voip box -> line |
| 14:23.49 | tainted- | i see! |
| 14:23.54 | tainted- | what was i thinking with SER |
| 14:24.11 | tainted- | there's no need! |
| 14:24.12 | tainted- | muahahahah |
| 14:24.15 | tainted- | evvvil |
| 14:24.19 | vaewyn | Then you can add cheap hardware for transcoding and only your main 'switch' needs to be beefy |
| 14:24.43 | tainted- | u guys are too cool |
| 14:24.45 | *** join/#asterisk Donuil (~fpatria@217.9.64.234) |
| 14:25.34 | Gand_DJ | How do you setup transcoding? My * box seems to forward whatever codec is being used from the originating source, and using that on the client. |
| 14:25.49 | tzanger | Gand_DJ: that's because that's the proper way to do it |
| 14:25.52 | tzanger | you only transcode if you have to |
| 14:26.03 | Donuil | I've just installed correctly asterisk but I've observed the lack of dial CLI command... is a problem of istallation?Can someone help me? |
| 14:26.18 | tzanger | in your case asterisk is set up to accept a set of codecs ... since both sides are accepting the current codec there's no need to transcode |
| 14:26.20 | jakepdev | why would you call dial in the CLI? |
| 14:26.23 | vaewyn | Gand_DJ: my 'switch' box only accepts ulaw... so everyone connecting to it transcodes |
| 14:27.31 | jakepdev | Donuil - you'd typically dial through your dialplan not the CLI |
| 14:28.25 | Gand_DJ | What's this? |
| 14:28.28 | Gand_DJ | Ignoring unknown information element 'Unknown IE' (38) of length 1 |
| 14:28.28 | Gand_DJ | Ignoring unknown information element 'Unknown IE' (39) of length 1 |
| 14:28.28 | Gand_DJ | Ignoring unknown information element 'Unknown IE' (40) of length 2 |
| 14:28.41 | Donuil | jakepdev yes I know, but I've already installed asterisk on another pc and there's the dial CLI command... why now it doesn't exist? |
| 14:28.53 | tzanger | Gand_DJ: your switch is sending you info asterisk doesn't know about. unless it's causing problems, ignore it |
| 14:28.56 | tzanger | just as asterisk did |
| 14:28.59 | *** join/#asterisk n4y (~frodo7@host-ip226-209.crowley.pl) |
| 14:29.09 | Nuxi | Anybody got any cool AGI games? I have rock-paper-scissors, thinking of a number, poker, and blackjack. |
| 14:29.09 | icexx | so......... does iconnecthere work with asterisk or no? :) |
| 14:29.09 | Gand_DJ | ok. being sent by fwd |
| 14:29.22 | jakepdev | Donuil - I don't see a dial cmd in the CLI |
| 14:29.26 | *** join/#asterisk kimc (~freenode@pcp09643046pcs.wbrmfd01.mi.comcast.net) |
| 14:29.36 | kimc | helloo |
| 14:29.56 | Blackvel | Why can I not dail in Manager API Action: Originate Channel: ZAP/g1/<number>? |
| 14:30.05 | christo | If I create a file called homecontext.inc and then use #include homecontext.inc in my dialplan, does asterisk then treat the contents of that file as the context 'homecontext' ? Do I still need to put [homecontext] at the top of it? |
| 14:30.18 | bkw_ | Blackvel, becuas it need more than that to orginate |
| 14:30.19 | bkw_ | duh |
| 14:30.22 | jakepdev | Donuil - http://lists.digium.com/pipermail/asterisk-users/2003-August/018881.html |
| 14:30.34 | Blackvel | bkw: yes, I give all the other params |
| 14:30.44 | lunchbox08 | anyone have any idea what kind of latency is added when turning on notransfer=yes inside IAX |
| 14:30.44 | tzanger | christo: no it inserts the entire contents of that file into the dilaplan at that point |
| 14:30.45 | MattB2 | anyone know why ast hangs up X100P as soon as remote party answers? |
| 14:30.47 | Blackvel | with Channel: SIP/100 it works |
| 14:30.49 | MattB2 | UK BT featurelines? |
| 14:30.53 | tzanger | lunchbox08: measure it yourslef |
| 14:30.56 | Blackvel | but with ZAP/g1/<number> not |
| 14:31.15 | bkw_ | do you have zap and group 1? |
| 14:31.17 | Blackvel | I always use ZAP/g1/exten in my DIAL commands |
| 14:31.26 | christo | tzanger - aaah I see, so I would have to do [homecontext] and then #include homecontext.inc |
| 14:31.53 | tzanger | no |
| 14:32.02 | tzanger | put [homecontext] IN homecontext.inc |
| 14:32.04 | tzanger | make it self-contained |
| 14:33.56 | *** join/#asterisk grolloj (~chatzilla@slim-eth0.horizonlive.net) |
| 14:34.20 | *** join/#asterisk vlan (~iq@70-59-161-163.omah.qwest.net) |
| 14:34.29 | Donuil | jakepdev what version of asterisk have you installed?mine is 1.0.7... and there's the lack of the dial CMD command... I want to know if it is an installation problem or not... someone can tell me about this? |
| 14:35.02 | jakepdev | Donuil - http://lists.digium.com/pipermail/asterisk-users/2003-August/018881.html |
| 14:35.22 | jakepdev | mine is 1.07 |
| 14:35.47 | kimc | Trying to dial a 700xxxxxxx number with: |
| 14:35.50 | kimc | exten => _9700XXXXXXX,1,Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN}@iaxtel) |
| 14:35.52 | jakepdev | the dial cmd seems to only appear if you have a sound card and the appropriate drivers have been loaded |
| 14:36.05 | jakepdev | kinc - alittle mod |
| 14:36.06 | kimc | But it sends the '9' to iaxtel |
| 14:36.09 | jakepdev | right |
| 14:36.13 | jakepdev | need to use :1 |
| 14:36.20 | jakepdev | ${EXTEN:1} |
| 14:36.24 | kimc | ahh |
| 14:37.00 | *** join/#asterisk Rick_Hunter (~rhunter@03-033.008.popsite.net) |
| 14:37.47 | kimc | Great :) thanks |
| 14:38.03 | jakepdev | np |
| 14:40.15 | Donuil | jakepdev I've installed all Alsa libraries... Do need to any specific drivers related to my soundcards, according to u? |
| 14:40.36 | ManxPower | Donuil: you also need the alsa header files |
| 14:40.56 | ManxPower | You also need ALSA setup, configured, and running and able to play sound files outside of asterisk. |
| 14:41.17 | ManxPower | then "make clean; make install" in the Asterisk source directory |
| 14:41.44 | tzanger | how do I enable 2bct in *? |
| 14:42.14 | ManxPower | tzanger: You need 5ESS as I undetand it. |
| 14:42.18 | ManxPower | not NI2 |
| 14:42.24 | tzanger | ManxPower: ahh |
| 14:42.34 | tzanger | so I'd have to reprovision my PRI |
| 14:42.35 | tzanger | no thanks |
| 14:42.36 | tzanger | :-) |
| 14:44.03 | Donuil | Manx Pwer I've downloaded alsa headers... to setup Alsa I need to recompile kernel or is there a specific command? |
| 14:44.17 | ManxPower | Donuil: I am not a Linux Support Person. |
| 14:44.27 | ManxPower | Go ask somewhere linux specific. |
| 14:47.21 | tzanger | http://public.paynacea.com/ |
| 14:47.26 | tzanger | that would be TRIVIAL to implement with asterisk |
| 14:47.49 | SuPrSluG_ | Donuil:alsaconf |
| 14:50.49 | *** join/#asterisk Moc____ (~mochouina@h66-201-214-109.gtconnect.net) |
| 14:51.51 | Ciber | grrr |
| 14:51.59 | Ciber | can't make outgoing calls |
| 14:52.03 | Ciber | anyone have any ideas? :P |
| 14:52.22 | durex | Ciber first of all... run asterisk in verbose mode and see what happening |
| 14:52.26 | Moonwick | broken dialing finger? |
| 14:52.40 | Ciber | i think it's in verbose :P |
| 14:52.41 | Moonwick | if you require a dialing wand, please mash the keypad now. |
| 14:52.52 | Ciber | i just get a busy signal heh |
| 14:53.01 | *** join/#asterisk watson (cancerea@203.170.71.63) |
| 14:53.09 | durex | ok, but why the busy signal? try to see the logfiles, verbose information.... |
| 14:53.25 | Ciber | you mean the debug thing right? |
| 14:53.41 | watson | hello room |
| 14:55.50 | Donuil | Just another question... I'm installing a h323 gatekeeper... I've a registration problem... after correctly installed pwlib and openh323 both astersik pc and gatekeeper pc I just lunch the make command in the directory /asterisk/channels/h323 as suggested by READEME file.. I've a problem with ast_h323.o file... have any suggestions? |
| 14:57.12 | *** join/#asterisk chris_d (~chris@66.88.142.66.ptr.us.xo.net) |
| 14:58.45 | *** join/#asterisk ontae (~ontae@chello213047229097.tirol.surfer.at) |
| 14:59.24 | watson | i m using format_mp3, but i m not able to play music on hold :-( |
| 14:59.32 | ontae | hi ... |
| 14:59.34 | zno | with the tdm400P, you only have a choice between fxo and fxs per port, so does that mean I can only have for example, 2 incoming and 2 outgoing lines? |
| 14:59.38 | MattB2 | watson: have you got mpg123 installed |
| 14:59.44 | watson | nops |
| 14:59.45 | nix000 | anyone tell me what is wrong with kernel 2.6 ? i keep getting Apr 28 10:58:48 WARNING[12323]: loader.c:440 load_modules: Loading module chan_zap.so failed! |
| 14:59.55 | ManxPower | zno: you can have a total of 4 ports per card. any combo of fxo or fxs modules |
| 15:00.01 | watson | i installed but i found on some wiki to remove it |
| 15:00.06 | *** join/#asterisk kiokorobert (~kiokorobe@196.200.26.42) |
| 15:00.06 | Gand_DJ | ne1 here using Broadvoice for voip, and have a toll-free # setup with them? |
| 15:00.07 | watson | before using format_mp3 |
| 15:00.07 | ManxPower | nix000: that usually means zaptel isn't loaded |
| 15:00.19 | MattB2 | watson: ah, sorry, not used format_mp3 |
| 15:00.33 | nix000 | Manipura, it is loaded and i see the logs via dmesg ! as well as lsmod |
| 15:00.45 | watson | same thing some body else told me yesterday not to use mpg123 |
| 15:00.47 | watson | :-) |
| 15:01.01 | jalsot | hi |
| 15:01.07 | zno | ManxPower: so if I have 4 fxo modules, I cannot make outgoing calls? |
| 15:01.15 | nix000 | ManxPower, it is loaded and i see the logs via dmesg ! as well as lsmod |
| 15:01.20 | watson | plus he said mpg123 will eat up system |
| 15:01.27 | zno | or just that I can't plug in regular phones with fxo modules |
| 15:01.50 | Gand_DJ | You can't plug in phones to fxo modules |
| 15:02.02 | watson | i m getting this |
| 15:02.02 | Gand_DJ | FXO = inbound & outbound calling |
| 15:02.04 | watson | res_musiconhold.c:370 moh1_exec: Unable to start music on hold |
| 15:02.07 | Gand_DJ | FXS = phone extensions |
| 15:02.13 | jalsot | does anybody use grandstream GXP-2000 here? it says, it supports 11 lines indicators, does astersisk support something like that? |
| 15:02.13 | zno | Gand_DJ: ah ok |
| 15:02.29 | SuPrSluG_ | nix00: did you also run ztcfg ? |
| 15:03.08 | ontae | is it possible to receive fax with asterisk when using asterisk as a voip client which registers at the sip proxy ? |
| 15:03.12 | nix000 | SuPrSluG_, yes it says line 147: Unable to open master device '/dev/zap/ctl' |
| 15:03.41 | *** join/#asterisk HeadachesAbound (HeadachesA@wsip-68-99-73-32.tu.ok.cox.net) |
| 15:03.56 | Ciber | where do the outgoing settings go? |
| 15:04.13 | SuPrSluG_ | nix00: sounds like a udev issue. you have to add to udev. there's a readme somewhere in zaptel dir |
| 15:04.51 | nix000 | SuPrSluG_, any idea why it says that when in dmesg i see Found a Wildcard FXO: Generic Clone |
| 15:05.51 | ManxPower | ~foxfxs |
| 15:05.52 | nix000 | SuPrSluG_, how do i know i have udev ! |
| 15:06.04 | ManxPower | nix000: are you running Asterisk as root? |
| 15:06.07 | vlan | exit |
| 15:06.13 | ManxPower | ~fxofxs |
| 15:06.14 | jbot | rumour has it, fxofxs is An FXO port expects to receive dialtone and receive ring voltage. An FXS port expects to provide dialtone and provide ring voltage. |
| 15:06.14 | nix000 | ManxPower, yes |
| 15:06.24 | SuPrSluG_ | nix00: usually 2.6.8 or later |
| 15:06.29 | ManxPower | zno: you can have a total of 4 ports per card. any combo of fxo or fxs modules |
| 15:06.39 | ManxPower | zno: why do you see this as an issue. |
| 15:07.04 | HeadachesAbound | Anyone here familiar with dsp.c? In particular, with fax detection? |
| 15:07.06 | nix000 | SuPrSluG_, i am on ubuntu warty and its got 2.6.8.1 |
| 15:07.08 | SuPrSluG_ | nix00: if u ls /dev u prolly won't see a zap dev |
| 15:07.15 | ManxPower | if you want more than 4 ports you should be using a T-1/E-1 card and a channel bank. |
| 15:07.18 | *** join/#asterisk user0815 (~bea@chello213047229097.tirol.surfer.at) |
| 15:07.32 | Donuil | Just another question... I'm installing a h323 gatekeeper... I've a registration problem... after correctly installed pwlib and openh323 both astersik pc and gatekeeper pc I just lunch the make command in the directory /asterisk/channels/h323 as suggested by READEME file.. I've a problem with ast_h323.o file... have any suggestions? |
| 15:07.50 | nix000 | SuPrSluG_, i see zap1, zapchannel, zaptimer tho |
| 15:07.57 | ManxPower | Donuil: did you do a make install too? |
| 15:08.10 | ManxPower | Donuil: and you never say what your problem with chan_h323.o is. |
| 15:08.46 | user0815 | help |
| 15:08.51 | SuPrSluG_ | nix00: exactly. but it wants to make a /dev/zap/channel etc for /dev/zap/ |
| 15:09.39 | SuPrSluG_ | nix00: /usr/src/zaptel/README.udev |
| 15:12.47 | ontae | o.k. once again: is it possible to receive fax with asterisk when using asterisk as a voip client which registers at the sip proxy ? |
| 15:13.14 | ontae | or gatekeeper? |
| 15:14.04 | HeadachesAbound | anybody here ever done answering machine detection using frequencies? our current dialer which we plan to replace in the next 30 days does it using frequencies and we would like to try doing this on asterisk as well. |
| 15:14.44 | nix000 | SuPrSluG_, do i just rmmod then moprobe or restart ? |
| 15:14.55 | sivana | what's the polycom ip 500 default pwd? |
| 15:15.02 | sivana | for set up |
| 15:15.12 | SuPrSluG_ | nix00: restart should be ok |
| 15:15.40 | SuPrSluG_ | nix00: taht should create the proper /dev/zap dir's |
| 15:16.26 | SuPrSluG_ | nix00: and then the channels should load ok |
| 15:16.48 | ManxPower | sivana: it's documented in the admin guide. |
| 15:16.49 | SuPrSluG_ | sivana:123=user 456=admin |
| 15:16.50 | Donuil | ManxPower I've errors with the make command.. make install is next step... have error on several libraries just like dict.h |
| 15:17.01 | sivana | thank you |
| 15:17.16 | sivana | ManxPower: the admin guide is online? All I got is some CD |
| 15:17.18 | ManxPower | Donuil: H323 with Asterisk is one of the hardest things to install. I don't use it. |
| 15:17.24 | ManxPower | sivana: yes. |
| 15:17.46 | SuPrSluG_ | sivana:yes.the cd is only a polycom commercial |
| 15:19.15 | Donuil | Is there someone with experience about h323 asterisk installation? |
| 15:21.58 | nix000 | SuPrSluG_, $%^$%^ it still does not create the /dev/zap device ! |
| 15:22.02 | sivana | what's a good NTP server I can use? |
| 15:22.26 | nix000 | SuPrSluG_, actuaaly its only the /dev/zap/ctl that is missing now |
| 15:23.02 | watson | i m getting <Gand_DJ> FXO = inbound & outbound calling |
| 15:23.03 | watson | <watson> res_musiconhold.c:370 moh1_exec: Unable to start music on hold |
| 15:23.03 | watson | <Gand_DJ> FXS = phone extensions |
| 15:23.21 | watson | i m getting this, res_musiconhold.c:370 moh1_exec: Unable to start music on hold when i user format_mp3 |
| 15:23.24 | SuPrSluG_ | nix00: ztcfg ? |
| 15:23.33 | watson | what might be the problem ? |
| 15:23.38 | *** join/#asterisk mhnoyes (~mhnoyes@user-2ivfmiv.dialup.mindspring.com) |
| 15:24.23 | nix000 | SuPrSluG_, ztcf is still looking for '/dev/zap/ctl but there is a /dev/zapctl |
| 15:24.35 | ontae | anyone here who can answer me a question regarding fax recipe ? |
| 15:25.34 | stoyan | i am wondering what is the username in sip.conf used for since I can authenticate with sip client only with the sip cahnnel name - the thingy in the square brackets |
| 15:25.48 | Donuil | I'm installing a h323 gatekeeper... I've a registration problem... after correctly installed pwlib and openh323 both astersik pc and gatekeeper pc I just lunch the make command in the directory /asterisk/channels/h323 as suggested by READEME file.. I've a problem with ast_h323.o file... have any suggestions? |
| 15:26.11 | Delvar | stoyan: its used for out bound calls |
| 15:26.19 | c_k | whats the best way to make a test call |
| 15:27.58 | bkw_ | with a phone |
| 15:28.00 | bkw_ | :P |
| 15:28.02 | stoyan | Delvar: sorry but I don't quite understand - you mean from asterisk to the sip channel or? |
| 15:28.12 | c_k | lol |
| 15:28.31 | SuPrSluG_ | nix00: my /etc/udev/rules.d/ looks like this http://pastebin.ca/10504 |
| 15:28.36 | c_k | can you test connectivity to a phone I meant without calling it |
| 15:28.43 | kiokorobert | hi |
| 15:28.49 | bkw_ | isn't their a readme.udev? |
| 15:29.03 | kiokorobert | anyone managed to change voice prompts from comedian mail to something else? |
| 15:29.54 | nix000 | SuPrSluG_, rebooting again ... |
| 15:30.26 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 15:30.49 | nix000 | bkw_, i did read it after SuPrSluG_ sugestion ... |
| 15:31.09 | ManxPower | kiokorobert: they are all just .gsm files on the hard drive. change them to say anything you want. |
| 15:32.08 | nix000 | SuPrSluG_, wooooooooooooo hooooooooooooo works ! |
| 15:32.25 | SuPrSluG_ | gtg |
| 15:33.22 | *** join/#asterisk bah (048830696@AC91AB89.ipt.aol.com) |
| 15:34.08 | sivana | with the Polycom, do I have to use FTP/TFTP? |
| 15:34.10 | kiokorobert | how? |
| 15:34.47 | ManxPower | sivana: you can do basic config over a web browser |
| 15:34.48 | *** part/#asterisk Donuil (~fpatria@217.9.64.234) |
| 15:35.17 | ManxPower | kiokorobert: record the files, use sox to convert them to GSM format, copy them to the asterisk sounds directory. |
| 15:35.40 | stoyan | are there cases when I would need the to have a username that doesn't match the sip channel in sip.conf? |
| 15:36.07 | ManxPower | stoyan: Not normally |
| 15:36.21 | stoyan | ManxPower: thanks |
| 15:36.21 | Nivex | They don't have to be .gsm. They can be .wav as long as they are 8Khz Mono |
| 15:36.29 | ManxPower | unless your device/provider uses different username/secret for incoming .vs. outgoing calls. |
| 15:36.50 | Nivex | depends on if you want to take the disk space hit for improved quality when using ulaw |
| 15:37.04 | ManxPower | [blah] is the RECEIVED username for incoming calls. username= is username sent on outgoing calls. |
| 15:37.29 | *** join/#asterisk heison (~heison@ns.somanetworks.com) |
| 15:38.22 | stoyan | ManxPower: I don't understand who uses the username on outgoing calls and for what :( |
| 15:39.16 | ManxPower | ~docs |
| 15:39.17 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 15:39.19 | ManxPower | ~mailinglist |
| 15:39.20 | jbot | i heard mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
| 15:39.28 | kiokorobert | ManxPower: I don't want to record the voice message myself and convert |
| 15:39.47 | ManxPower | kiokorobert: then you'll have problems replacing the existing ones. |
| 15:40.33 | kiokorobert | ManxPower: Where can I get the voice behind the other voice messages? |
| 15:40.46 | ManxPower | thevoice.digium.com |
| 15:40.53 | ManxPower | it's linked on the digium web site. |
| 15:41.03 | zno | Allison Smith |
| 15:41.22 | zno | sultriest accent in all of english speaking canada |
| 15:41.38 | ManxPower | zno: some of my users complained she sounded TOO sexy. |
| 15:41.44 | blitzrage | lol |
| 15:41.45 | ManxPower | of course, they were all women doing the complaining. |
| 15:42.02 | *** join/#asterisk Blackvel (~blackvel@dsl-082-083-171-152.arcor-ip.net) |
| 15:42.06 | zno | She must be making tons of money |
| 15:42.10 | zno | off of asterisk |
| 15:42.25 | *** join/#asterisk CSG (foobar@i-195-137-6-228.freedom2surf.net) |
| 15:42.59 | ManxPower | zno: read her bio sometime. She does a lot of other work in addition to asterisk. voiceovers, commercials, animation voices, other phone systems, etc. |
| 15:43.14 | zno | yeah, she's an actress |
| 15:43.32 | zno | a lot of those around in New York. Half the people I know here are aspiring actors |
| 15:43.37 | SuPrSluG_ | or sample selma simpson |
| 15:44.19 | ManxPower | zno: you know a lot of waitstaff? |
| 15:44.36 | zno | and bartenders |
| 15:45.46 | ontae | Help! |
| 15:45.59 | ontae | me! |
| 15:46.46 | SuPrSluG_ | ontae,w/ what |
| 15:48.48 | zno | last few words before being eaten by a shark |
| 15:48.54 | kiokorobert | ManxPower: Any other commercial solutions? |
| 15:49.52 | blitzrage | anyone know of a smartphone with touchscreen, BT and wireless? |
| 15:50.20 | *** join/#asterisk outtolunc (outtolunc@adsl-66-218-53-170.dslextreme.com) |
| 15:50.37 | Ciber | yay |
| 15:50.42 | Ciber | outgoing calls working now heh |
| 15:50.45 | blitzrage | the cynic returns! :) |
| 15:50.59 | vaewyn | that narrows it down |
| 15:51.00 | vaewyn | :P |
| 15:51.06 | Ciber | blitz, i have a treo 650 |
| 15:51.15 | blitzrage | Ciber: yah, but it doesn't have wifi, right? |
| 15:51.22 | Ciber | deal with it :P |
| 15:51.30 | Ciber | it's the best smartphone for now :P |
| 15:51.42 | Ciber | besides my cingular edge is fast enough heh |
| 15:51.51 | blitzrage | yah, but you pay for that ;) |
| 15:52.11 | Ciber | beats running around like an idiot trying people's routers :P |
| 15:52.25 | blitzrage | fair enough :) |
| 15:52.36 | blitzrage | but can you run VoIP over that sucker? :) |
| 15:52.52 | Ciber | 1 call sure |
| 15:52.57 | Ciber | but the latency will kill ya |
| 15:53.21 | blitzrage | they build it into the network so you can't run VoIP |
| 15:53.28 | blitzrage | thats my theory :) |
| 15:53.43 | Ciber | you can do it on verizon ev-do crap |
| 15:53.52 | Ciber | but it's like 80 a month :P |
| 15:54.35 | ontae | <SuPrSluG_>: I am new to asterisk. Want to set it up with my voip provider. Think this should be no problem as far as i know now. What do you know about receiving faxes with asterisk when using it with a voip provider? |
| 15:55.08 | Ciber | whatever you do, don't follow broadvoice's instructions :P |
| 15:55.11 | Ciber | waste of time ;) |
| 15:55.21 | SuPrSluG_ | ontae,look into NVdetect |
| 15:55.34 | blitzrage | oooo: http://www.nokia.com/nokia/0,,54106,00.html |
| 15:56.14 | Ciber | it would be useful if it had a real os :P |
| 15:56.17 | ontae | SuPrSluG_, NVdetect? |
| 15:56.34 | *** join/#asterisk rjburkh (~rjburkh@dialup-4.231.175.87.Dial1.LosAngeles1.Level3.net) |
| 15:56.39 | *** join/#asterisk ckruetze (~ckruetze@cpc3-cmbg7-5-0-cust100.cmbg.cable.ntl.com) |
| 15:57.52 | *** join/#asterisk porche (~a@dsl81-215-31900.adsl.ttnet.net.tr) |
| 15:57.53 | porche | hhi |
| 15:57.54 | porche | hi |
| 15:58.01 | SuPrSluG_ | ontae,look into NVfaxdetect |
| 15:58.09 | porche | got a question on recording, over iax channels |
| 15:58.34 | SuPrSluG_ | ontae,look into NVbackgrounddetect |
| 15:59.12 | ontae | SuPrSluG_, where do i find "NVfaxdetect"? What is "NVfaxdetect"? |
| 16:00.00 | SuPrSluG_ | ontae,from what i've read it can receive faxes over sip/iax . never tried it though |
| 16:00.40 | ontae | o.k. found something through google, thanks SuPrSluG_ |
| 16:00.45 | SuPrSluG_ | ~asterisk and faxes |
| 16:01.31 | Gand_DJ | I'm thinking of getting an Sipura 3000.... if you setup the fxo port to auto-pass the pstn caller to *, are you still able to call out through to the pstn or will the 3000 just pass the call back to * |
| 16:02.35 | *** join/#asterisk Siz1 (~Marco@osp.demon.nl) |
| 16:03.44 | *** part/#asterisk Siz1 (~Marco@osp.demon.nl) |
| 16:04.05 | *** join/#asterisk Fanguin (~Fanguin@p548F12CC.dip0.t-ipconnect.de) |
| 16:04.38 | Fanguin | Hello, does anybody use german sound files with the voicemail application? |
| 16:05.22 | zno | ich nicht |
| 16:05.45 | MattB2 | anyone used BT featurelines? |
| 16:08.01 | *** part/#asterisk kimc (~freenode@pcp09643046pcs.wbrmfd01.mi.comcast.net) |
| 16:11.34 | ontae | SuPrSluG_, One more question: Imagine one VoIP/SIP Number registerd with asterisk at the VoIP provider, so incoming calls to that number goes to asterisk. Is it now possible, when dialing an additional number to that offical phone number, to route the call to an extension? |
| 16:12.50 | *** join/#asterisk Syncros (~sysop@noc.routermonkey.net) |
| 16:17.56 | SuPrSluG_ | ontae,you tell * where to send incoming calls. wether through zap/sip/iax |
| 16:18.57 | *** join/#asterisk SuperN (SuperN@202satb3.codetel.net.do) |
| 16:19.12 | SuperN | hello everybody |
| 16:19.25 | SuperN | I would like to know I can use asterisk with suse 9.3? |
| 16:20.46 | ontae | SuPrSluG_, o.k. will find out |
| 16:21.14 | SuperN | I would like to know I can use asterisk with suse 9.3? |
| 16:22.28 | *** part/#asterisk ontae (~ontae@chello213047229097.tirol.surfer.at) |
| 16:23.21 | *** join/#asterisk flot (CCCP@213.152.157.68) |
| 16:23.25 | SuperN | I would like to know I can use asterisk with suse 9.3? |
| 16:24.26 | drumkilla | posting the same question over and over within a couple minutes isn't going to do anything but annoy people |
| 16:24.31 | drumkilla | yes, you can |
| 16:27.58 | pjz | anyone have phone recommendations? |
| 16:28.40 | johnnyb | Our grandstreams aren't bad, but I've heard you can get better for the same money. |
| 16:28.43 | SuperN | ok |
| 16:29.19 | *** join/#asterisk Lee__ (~lee@ool-44c26e7d.dyn.optonline.net) |
| 16:31.28 | Delvar | johnnyb: ' better for the same money' which phones? |
| 16:33.45 | pjz | <PROTECTED> |
| 16:33.50 | pjz | anyone know what might cause that? |
| 16:34.02 | pjz | clearly a reference to a 'default' context which doesn't exist |
| 16:34.09 | pjz | but where would that reference be? |
| 16:34.27 | ManxPower | pjz: sip.conf, iax.conf, zapata.conf, mgcp.conf, skinny.conf, h323.conf |
| 16:34.31 | ManxPower | Do you see a pattern there? |
| 16:35.15 | pjz | so an extension helper is the protocol helper? |
| 16:35.36 | *** join/#asterisk Bob_ (~icechat5@196.7.14.183) |
| 16:36.52 | mmlj4 | he's gone, but SuSE as of 9.2 packages * |
| 16:37.24 | Bob_ | Hi there....I have a question about iax2....it seems to send and receive on the same port (4569) - but if one is using a nat, and you have more than one client behind the nat connecting to someone on the outside, how are the packets routed to both clients if using the same port? Sorry, I am pretty new to this stuff |
| 16:37.52 | at561 | the nat gateway reassigns the port that the outside world sees |
| 16:38.01 | vaewyn | Bob_: NAT can handle that for clients... for servers you can only have 1 behind it |
| 16:38.04 | mmlj4 | Bob_: NAT is smart enough to figure out what client on the inside is doing what |
| 16:38.41 | mmlj4 | now if you have a server behind NAT, that's different... you have to forward the correct ports to the server |
| 16:39.07 | *** part/#asterisk vaewyn (freeman@mail.parrishmachine.com) |
| 16:39.08 | Bob_ | no - only the clients are behind the nat |
| 16:39.17 | mmlj4 | then it should just work |
| 16:39.37 | mmlj4 | have you had problems with it? |
| 16:39.50 | Bob_ | ja - I am assuming it will work, but just trying to understand how it works:-) |
| 16:39.57 | Bob_ | using sip at the moment, which is a nightmare |
| 16:40.03 | Bob_ | ....with nats that is |
| 16:40.11 | Bob_ | iax seems so much simpler |
| 16:40.20 | *** join/#asterisk vaewyn (freeman@mail.parrishmachine.com) |
| 16:40.31 | jakepdev | any way to trap out * on a read string? |
| 16:40.36 | ManxPower | Bob_: there is no special config required on the client side NAT router nor the SIP client |
| 16:40.49 | mmlj4 | SIP and NAT are enemies |
| 16:40.59 | ManxPower | on the asterisk side be sure to use nat=yes and qualifty=yes in the [sippeeroruserorfriend] section |
| 16:41.20 | ManxPower | or qualify=yes |
| 16:41.32 | *** join/#asterisk cmk (~cmk_@p54A3D3EE.dip.t-dialin.net) |
| 16:41.45 | Bob_ | ManPower - what is the ideal time for qualify, because you can also set the number of milliseconds |
| 16:42.12 | ManxPower | Bob_: it varies. "yes" means 2000 |
| 16:42.28 | Bob_ | ManxPower - and would I need canreinvite=no - what is the effect of this |
| 16:42.50 | ManxPower | all you need qualify=yes for is to keep traffic happening to the SIP device so the NAT router doesn't clost the NAT translations. You could accomplish the same thing by telling the SIP device to register every 60 seconds. |
| 16:42.53 | Bob_ | ManxPower - my setup is such that clients are dialling into a meetme conference - the clients never get called |
| 16:42.57 | ManxPower | Bob_: yes you need canreinvite=no |
| 16:43.25 | vaewyn | arggh... I need an openldap excorcist |
| 16:43.34 | *** join/#asterisk point (1000@213.27.44.55) |
| 16:43.37 | vaewyn | and evidently they don't have an IRC channel |
| 16:43.37 | Bob_ | ManxPower - is that because reinvites don't work if client is behind a nat? |
| 16:43.44 | johnnyb | Delvar: can't remember exactly -- I'd look at both snom and polycom, I think it was one of them. |
| 16:43.47 | ManxPower | Bob_: correct |
| 16:43.53 | vaewyn | polycom's rock |
| 16:44.12 | ManxPower | If you want reinvites to work THEN you must do port forwarding on the NAT router. That sounds simple. It's not. |
| 16:44.22 | Delvar | johnnyb: snom are a lot more expensive :) |
| 16:44.24 | vaewyn | snom's work well... but have horrible tactile and user interfaces |
| 16:44.29 | Gand_DJ | heh... I checked my Logwatch for *, and I'm showing alot of people trying to hack my machine. Wonder if it's someone from here since no-one knows my ip / domain & that I run * besides here |
| 16:44.33 | johnnyb | Delvar: it was probably the polycom, then |
| 16:44.49 | Delvar | johnnyb: ill take alook at them thanks |
| 16:45.08 | Bob_ | ManxPower - if I am dialling into meetme conferences then I wouldn't really need reinvites I would guess |
| 16:45.09 | ManxPower | Gand_DJ: what ports? |
| 16:45.27 | ManxPower | Bob_: you only really need reinvites if two phones are calling each other. |
| 16:45.32 | Gand_DJ | <PROTECTED> |
| 16:45.36 | Gand_DJ | Doesn't say ports |
| 16:45.48 | ManxPower | Gand_DJ: that's just automated bots trying to hack. |
| 16:45.52 | ManxPower | everyone gets them |
| 16:45.59 | Gand_DJ | most ports are blocked by my router |
| 16:46.01 | ManxPower | they scan large blocks of addresses. |
| 16:46.06 | Gand_DJ | I only have certain ports open |
| 16:46.11 | point | <PROTECTED> |
| 16:46.12 | Bob_ | ManxPower - It is difficult for me to test with every single type of NAT before going live - in your opinion, should nat=yes,qualify=yes and canreinvite=no work for most nats? |
| 16:46.28 | ManxPower | Bob_: yes. |
| 16:46.46 | Bob_ | ManxPower - thanks for the help - really appreciate it |
| 16:47.04 | ManxPower | Bob_: you can thank me by helping me to find a job in Europe. do a /whois ManxPower for more information. |
| 16:47.30 | Bob_ | ManxPower - where are you from? |
| 16:47.38 | zno | ManxPower: Don't you have any preferences of countries in Europe? |
| 16:47.38 | ManxPower | Bob_: near New Orleans |
| 16:47.58 | ManxPower | zno: I prefer Netherlands or Belgium, but all serious offers will be considered. |
| 16:48.22 | pjz | anyone know why all my internal phones get echo even when talking to another extension? |
| 16:48.26 | pjz | it's self-echo |
| 16:48.29 | zno | My personal favorites are the nordic countries,Holland, Germany |
| 16:48.31 | ManxPower | "I'll be happy to hire you, but I can only pay you two peanuts and a cow per day" is not serious offer, BTW. |
| 16:48.34 | zno | I hear Estonia is really nice too |
| 16:48.38 | Bob_ | ManxPower - sorry, I am in South Africa, so probably won't be able to help you |
| 16:48.59 | vaewyn | ManxPower: what if we made it 3 peanuts? |
| 16:49.02 | zno | ManxPower: wages are lower in Europe, you can't make as much money as you do in the states |
| 16:49.03 | Gand_DJ | ManxPower, heh.. I checked the IP vs URL... and it's mainly 1 person / bot that tried to hack me. there's a second IP but only 2 entries compared to the 30+ from this main ip (from taiwan) |
| 16:49.04 | vaewyn | :} |
| 16:49.13 | zno | unless it's Switzerland, or Norway |
| 16:49.27 | ManxPower | zno: I don't really expect to. |
| 16:49.27 | Gand_DJ | h220-210-68-71.seed.net.tw |
| 16:49.39 | Gand_DJ | and all61151025.allnet.ne.jp |
| 16:49.40 | Gand_DJ | lol |
| 16:49.58 | vaewyn | The asians are coming to take you away |
| 16:50.05 | Fanguin | ManxPower, I am astonished that it seems to be difficult to find a job in europe for such a wise guy like you are. |
| 16:50.05 | mmlj4 | yeah, eastern europe, ManxPower... go there, be back in 6 weeks |
| 16:50.31 | ManxPower | Fanguin: There's a lot of paperwork a company has to do to hire a non-EU citizen. |
| 16:50.35 | Fanguin | ManxPower, I read your message in the mailinglist some time ago ... |
| 16:50.37 | zno | Visa issues |
| 16:50.48 | zno | same problem for those trying to work in the US |
| 16:50.49 | Fanguin | ManxPower, ahh, ok |
| 16:51.17 | zno | unemployment is high in Europe, so it's really hard for politicians to allow foreigners to come in |
| 16:51.23 | ManxPower | I hope to meet with people interested in hireing me whem I'm in europe on holiday in a few weeks |
| 16:51.32 | zno | although they need them cos they don't have much skilled labor among the unemployed |
| 16:51.35 | Bob_ | ManxPower - one last question....I will not know ahead of time which users will be behind nats...if one has nat=yes and they happen to be on a public ip, then will this still work? |
| 16:52.03 | ManxPower | Bob_: for MOST SIP devices they will still work if you have nat=yes and they are on a public IP |
| 16:52.08 | mmlj4 | "on holiday"? you're already taking like a non-american |
| 16:52.10 | sivana | ManxPower: are you really familiar with Polycom IP 500? |
| 16:52.26 | ManxPower | mmlj4: only so non-americans will understand. |
| 16:52.26 | Bob_ | ManxPower - ok, thanks |
| 16:52.33 | mmlj4 | ah. |
| 16:52.38 | ManxPower | sivana: no. |
| 16:52.45 | zno | ManxPower: do you work for yourself now? |
| 16:52.47 | *** join/#asterisk billheckel (~billhecke@24.23.125.11) |
| 16:52.51 | ManxPower | zno: Correct. |
| 16:52.59 | billheckel | hola all |
| 16:53.05 | zno | ManxPower: is it hard to set up shop in Europe? |
| 16:53.09 | vaewyn | howdy ranger bill |
| 16:53.14 | billheckel | i am having a zhone brain fart |
| 16:53.21 | ManxPower | zno: I wasn't planning on doing independent consulting in europe. |
| 16:53.30 | ManxPower | The paperwork and funding to start a company in Europe is staggering. |
| 16:53.48 | zno | I noticed a while back someone in Berlin looking for an asterisk guy on the biz list |
| 16:53.59 | zno | Berlin is really cool |
| 16:54.03 | bsdfreak | <PROTECTED> |
| 16:54.10 | billheckel | i need the command to activate a T1 interface on my zplex-10. For some reason each time I power cycle it comes up oos |
| 16:54.19 | *** join/#asterisk miller7 (~none@adsl49-static-gw1.access.acn.gr) |
| 16:54.25 | billheckel | and it has been so long, I forgot the command and can't find my notes |
| 16:54.30 | ManxPower | Too bad the Dutch don't have a Right of Return 8-) |
| 16:54.44 | miller7 | anyone here using Ascend MAX routers? |
| 16:55.46 | watson | where i can get information about DID ? |
| 16:55.52 | watson | how to set in asterisk |
| 16:56.02 | miller7 | watson: DID = extension |
| 16:56.09 | watson | for inbound / outbound |
| 16:56.22 | miller7 | you just set an extension same as the DID. Outbound you just use setcallerid |
| 16:56.39 | watson | yes if i want some country number and the call will come to this number ! |
| 16:56.46 | pjz | anyone know why I can always hear myself on these voip phones? |
| 16:56.58 | watson | like ppl do |
| 16:57.17 | miller7 | watson: you are not making yourself clear |
| 16:57.26 | mmlj4 | ManxPower: what state were you born in? |
| 16:57.28 | watson | they use one number from providers and use it for incomming call |
| 16:57.31 | watson | any where |
| 16:57.46 | *** part/#asterisk vaewyn (freeman@mail.parrishmachine.com) |
| 16:57.51 | miller7 | watson: do you have service from any provider? |
| 16:57.54 | watson | sorry for my veyr limited knowledge |
| 16:57.56 | billheckel | DID on the T1 is set in the zapata.cfg for inbound |
| 16:58.08 | watson | no, what are the steps ? |
| 16:58.20 | shido6 | ZZzZ |
| 16:58.35 | mmlj4 | the french have a recent law that allowes anyone born in territory that was once a french colony to be granted french citizenship upon asking for it |
| 16:58.36 | miller7 | watson: you go to http://www.nufone.net/ and you buy a DID. You expect their information on how to set it up. |
| 16:58.41 | watson | i mean what to set on my asterisk |
| 16:58.43 | miller7 | It comes after 2 minutes in your email |
| 16:58.48 | watson | and what to ask to the provider |
| 16:58.52 | billheckel | the telco will send some digits to you as part of each call |
| 16:59.12 | mmlj4 | I, for instance, qualify, as I was born in south louisiana |
| 16:59.28 | billheckel | ask the provider to set your T1 for DID and depending on the telco and the size of your block you can get anywhere from 1 digit to the whole number sent in as DID digits |
| 16:59.38 | CoffeeIV | mmlj4: so does all of Texas. We should apply and move in. I bet they'd love that. |
| 17:00.00 | mmlj4 | huh? texas was a french colony? news to me |
| 17:00.09 | watson | do u know how much it cost ? |
| 17:00.18 | billheckel | depends on the carrier |
| 17:00.19 | watson | for 5-6 lines ? |
| 17:00.31 | watson | any idea ? |
| 17:00.40 | billheckel | usually no additioanl charge, it is a service with the T1 connection |
| 17:00.48 | billheckel | at least with bell atlantic |
| 17:00.52 | watson | approx ? |
| 17:01.19 | mmlj4 | ManxPower: how much is a T1 from I-55? do you remember? |
| 17:01.22 | billheckel | but we did HUGE call volume, $30,000-50,000 per month so we might have been getting a better deal |
| 17:01.31 | mmlj4 | argh, he's afk again |
| 17:02.17 | watson | basically i m trying to setup asterisk for company one in india and one in UK |
| 17:02.50 | shido6 | 2976190.47619047 minutes for 50k |
| 17:02.53 | watson | and i don't know the basics :-) but trying to learn |
| 17:02.54 | ManxPower | mmlj4: T-1 voice or T-1 data? |
| 17:02.59 | *** join/#asterisk Sedorox (~Sed@Neptune.client.wlmsprt.pa.sed6.net) |
| 17:03.05 | billheckel | my esp is US only, one of our esteemed european users will have better data |
| 17:03.06 | ManxPower | mmlj4: doesn't really matter, I don't recall the price for either |
| 17:03.06 | watson | only voice |
| 17:03.26 | mmlj4 | say 8 voice, the rest data... or what figures do you have? |
| 17:03.30 | mmlj4 | ah, well... |
| 17:04.10 | watson | yes how much for 8 voice ? |
| 17:04.19 | *** join/#asterisk los415 (~los415@c-24-126-63-233.hsd1.ca.comcast.net) |
| 17:04.26 | watson | its on voice channel or in minutes ? |
| 17:04.50 | ManxPower | mmlj4: call Gene at 504-200-4105 |
| 17:04.57 | ManxPower | to get a quote |
| 17:04.58 | mmlj4 | minutes? um, T1 is not billed in minutes, at least not here |
| 17:05.15 | mmlj4 | yeah, i'll probably do that, ManxPower |
| 17:06.06 | watson | at nufone "Due to system upgrades we cannot accept any new accounts at this time. |
| 17:06.06 | watson | Please be patient |
| 17:06.40 | shido6 | you can fire up a new account with paypal |
| 17:07.15 | watson | they will give me DID and T1 traffic ? |
| 17:07.23 | shido6 | yes |
| 17:07.29 | shido6 | just ask for it |
| 17:07.45 | watson | u ppl think i can get 8 lines traffic only ? |
| 17:07.54 | shido6 | you want 23 dedicated channels for yourself, specify that in your email to sales@nufone.net |
| 17:08.09 | watson | 23 ? or 8 ? |
| 17:08.21 | shido6 | T1 traffic is 23, 8 is less than T1 |
| 17:08.28 | shido6 | you can get 8 if you want it |
| 17:08.38 | mmlj4 | watson: do you also want data (Internet connectivity)? |
| 17:08.46 | watson | mean i can't get 8 ? |
| 17:08.52 | shido6 | yes you can get 8 |
| 17:09.02 | shido6 | 8 voice channels, 8 T1's |
| 17:09.07 | shido6 | its up to you you can get both |
| 17:09.14 | mmlj4 | a T1 has 23 usable "lines"... you use what you need for voice, and keep the rest for 'net traffic |
| 17:09.17 | watson | no i want the ppl in UK to call a number and the call will come to other office |
| 17:09.48 | shido6 | do you want to use voip at the "other office" ? |
| 17:09.48 | watson | like virtual office !! ? |
| 17:09.59 | mmlj4 | oh, this is in UK? it's called E1, not T1, and there are i think 29 usable channels |
| 17:10.08 | shido6 | 30 |
| 17:10.10 | porche | 30 |
| 17:10.12 | watson | yes 30 |
| 17:10.21 | mmlj4 | yes, 30, -1 D = 29 |
| 17:10.29 | porche | watson who will call, who? |
| 17:10.34 | shido6 | err |
| 17:10.35 | shido6 | no |
| 17:10.38 | mmlj4 | no? |
| 17:10.45 | porche | 30 straight lines |
| 17:10.48 | porche | in E1 |
| 17:10.49 | shido6 | 31, -1 |
| 17:10.52 | porche | true |
| 17:10.54 | mmlj4 | ah, ok |
| 17:10.56 | *** join/#asterisk rtomson-home (~rtomson@ip70-181-140-181.sd.sd.cox.net) |
| 17:10.59 | watson | any body who wants to call my office in UK |
| 17:11.03 | watson | uk number |
| 17:11.13 | porche | you want to give a uk number |
| 17:11.18 | watson | yes |
| 17:11.20 | porche | or us number to your customers? |
| 17:11.40 | shido6 | what prefix in the uk do you want? |
| 17:11.41 | watson | both will be same ? |
| 17:11.43 | porche | well, gossiptel.com |
| 17:11.45 | watson | i mean cost wise ? |
| 17:11.49 | porche | does provide some |
| 17:11.55 | mmlj4 | gossiptel? heh |
| 17:12.02 | porche | cost to your customers |
| 17:12.06 | porche | or to you? |
| 17:12.27 | watson | local call to user |
| 17:12.28 | porche | mml, i dont know any other, just wanted to test some calls to uk, it was the first one I met, it did go |
| 17:12.42 | billheckel | i found my zhone manual |
| 17:12.47 | billheckel | problem fixed |
| 17:12.54 | Fanguin | I have a problem with the [zonemessages] in voicemail.conf. when i write |
| 17:13.03 | Fanguin | [zonemessages] |
| 17:13.03 | Fanguin | de=Europe/Berlin|'vm-received' q 'digits/at' H 'digits/oclock' M |
| 17:13.08 | CoffeeIV | I registered for an IAXTel account on IAXTel.com and followed the instructions there for configuring * . However my * is behind a NAT, and I think that is why it isn't working. Is there a page on voip-info that tells me what ports to forward through my firewall ? I couldn't find it. |
| 17:13.21 | Fanguin | in voicemail.conf i still get the old format. |
| 17:13.36 | watson | what will user have to pay ? |
| 17:13.39 | Fanguin | "the old format" is 'vm-received' q 'digits/at' IMp |
| 17:14.04 | *** join/#asterisk Abbas (Abbas@203.81.194.242) |
| 17:14.05 | Fanguin | i have to change the source code to change the format to german language. |
| 17:14.08 | watson | this is my first try to any voip thing thats why :-( |
| 17:14.23 | Abbas | hi guys |
| 17:14.36 | porche | q: agi, record file, made me sick, it says -1 for unlimited timeout, but has got s=seconds for maximum silence |
| 17:14.43 | porche | well, it never stops recording |
| 17:14.46 | Abbas | we are working on 60 X 600 PBX which channel banks should we use? |
| 17:15.58 | MattB2 | anyone ever heard of MF4 signalling? |
| 17:16.54 | bkw_ | 36000 lines? |
| 17:17.01 | *** join/#asterisk Uther_P (~uther_p@66.180.120.83) |
| 17:18.00 | bkw_ | nm |
| 17:20.08 | *** part/#asterisk miller7 (~none@adsl49-static-gw1.access.acn.gr) |
| 17:20.08 | Delvar | nn all |
| 17:23.40 | MattB2 | any Zap gurus in here? |
| 17:23.58 | MattB2 | trying to persuade X100P to work with loopstart |
| 17:24.31 | MattB2 | the driver is detecting NO BATTERY then BATTERY when the remote party hangsup |
| 17:24.37 | MattB2 | but that's not getting into Ast |
| 17:24.50 | MattB2 | UK lines |
| 17:25.42 | MattB2 | opermode is set to CTR21 |
| 17:26.36 | Sedorox | fun fun |
| 17:26.37 | Sedorox | er |
| 17:27.27 | porche | bye all |
| 17:27.43 | ManxPower | MattB2: you didn't get any response when you asked on the mailing list or searched the mailinglist archives? |
| 17:28.25 | *** join/#asterisk Bile_One (~bile_one@pcp03281999pcs.gillst01.ar.comcast.net) |
| 17:28.31 | MattB2 | searched the mailing list, found nothing useful |
| 17:28.39 | MattB2 | admittedly haven't askefd on mailing like |
| 17:28.44 | *** join/#asterisk fugitivo (~ajf@201.255.99.226) |
| 17:28.45 | MattB2 | I feel a RTFM coming my way ;) |
| 17:29.09 | ManxPower | MattB2: I seem to recall that hangup detection simply doesn't work with the X100P in the UK, but I could be wrong. |
| 17:29.30 | ManxPower | For some REASON I thought you were in the USA. |
| 17:29.51 | *** part/#asterisk point (1000@213.27.44.55) |
| 17:29.54 | MattB2 | ManxPower: well I am kinda... I'm from UK, now live in USA but setting up asterisk in UK and USA at the same time! |
| 17:30.02 | fugitivo | MattB2: problems with x100p and hangup? |
| 17:30.07 | MattB2 | yeah |
| 17:30.11 | MattB2 | in two different countries as a matter of fact |
| 17:30.18 | fugitivo | i had problems too |
| 17:30.22 | *** join/#asterisk nrc (~username@zeus.eurotux.com) |
| 17:30.29 | fugitivo | a call to my provider solved the problem |
| 17:30.42 | MattB2 | UK: uses loopstart, driver shows that something happens on hangup but isn't actually hanging up or notifying asterisl |
| 17:30.50 | MattB2 | UK provider being useless as always |
| 17:31.26 | MattB2 | using fxs_ls signalling in zapata.conf and zaptel.conf works better than ks ;) |
| 17:31.30 | MattB2 | but still doesn't quite work |
| 17:31.35 | fugitivo | MattB2: did you set signalling=fxs_ks on your zapata.conf? |
| 17:31.45 | fugitivo | oh |
| 17:31.50 | Bile_One | Hey ManxPower you living good? |
| 17:31.54 | *** join/#asterisk PCadach (~paul@www.east.telecom.kz) |
| 17:31.58 | fugitivo | MattB2: busydetect=yes can help you |
| 17:32.02 | MattB2 | fugitivo: using _ls in both conf files |
| 17:32.32 | MattB2 | the lines we have are BT featurelines and BT usefully dno't send down a busy tone on disconnect |
| 17:32.39 | MattB2 | they only do that on normal lines |
| 17:32.44 | *** join/#asterisk Exstatica (Exstatica@jumping.on.the.bed.are.not.umpteenmonkeys.com) |
| 17:32.52 | MattB2 | BT don't want you to use a PBX with their lines... |
| 17:33.12 | fugitivo | well, then it's difficult to solve |
| 17:33.17 | MattB2 | yeah |
| 17:33.19 | MattB2 | I know! |
| 17:34.23 | *** join/#asterisk L|NUX (~linux@202.5.145.58) |
| 17:37.20 | Bile_One | Who do we talk to about getting an app included into the next release? |
| 17:37.43 | Corydon-w | You don't talk to anybody |
| 17:37.49 | Bob_ | ManxPower - I have kind of got clients working from behind a lan using nat=yes, but the problem is that the ports only stay open for a few seconds and then the audio is cut off |
| 17:37.53 | Corydon-w | You just make sure it gets into CVS |
| 17:38.07 | Corydon-w | by filing a bug report on bugs.digium.com |
| 17:38.18 | Bile_One | app = bug? |
| 17:38.24 | Bob_ | ManxPower - I have also specified qualify=yes |
| 17:38.36 | ManxPower | Bob_: you have to turn off VAD in the clients |
| 17:38.59 | *** join/#asterisk Blackvel (~blackvel@dsl-213-023-029-186.arcor-ip.net) |
| 17:39.50 | Bob_ | ManxPower - is that the only way around it, because turning off VAD wastes bandwidth |
| 17:41.27 | bkw_ | Asterisk Dev Conference April 28th at 1pm CDT IAX2/guest@switch-3.asterlink.com/996 |
| 17:41.55 | *** join/#asterisk Marlow (~martin@cerberus.bluetree.ie) |
| 17:43.43 | *** part/#asterisk mjmac (~mjmac@cpe-24-198-203-132.maine.res.rr.com) |
| 17:44.54 | bjohnson | Bob_: put a * server on the lan with the clients |
| 17:45.38 | Bob_ | bjohnson - not possible, because users will be dialling in form many different lans |
| 17:46.02 | Gand_DJ | setup * gateways and link them together |
| 17:46.06 | Gand_DJ | from each lan |
| 17:46.11 | bsdfreak | anyone know of a good SIP phone application for fbsd? |
| 17:46.22 | Micc | VPN (cough, cough) |
| 17:46.31 | bjohnson | Bob_: then you've created a situation that wastes the bandwidth you treasure |
| 17:48.01 | Bob_ | bjohnson - you know what, it doesn't seem as though you can turn off VAD with the clients I am using - maybe I should rather use and IAX2 based client |
| 17:49.40 | *** join/#asterisk jief- (~jief@modemcable196.182-80-70.mc.videotron.ca) |
| 17:49.42 | jief- | hello |
| 17:50.05 | *** join/#asterisk visik7 (~ciao@visik7.user) |
| 17:50.33 | visik7 | is there some hardware * compatible that allow me to create a video entryPhone ? |
| 17:50.39 | jief- | im trying to register my SIP phones with *, but whatever the config i try, it never does. i tried using static and dynamic IPs, with and without usernames/secrets. im kinda out of ideas now |
| 17:53.09 | *** join/#asterisk ManxPower (~eric@dsl-209-16-67-160.datasync.com) |
| 17:53.40 | tzafrir | anybody here using VoipGate? |
| 17:54.08 | tzanger | I FUCKING ROCK THE HOUSE |
| 17:54.10 | tzanger | http://pastebin.ca/10514 |
| 17:54.37 | *** join/#asterisk McUnixJr (~mcmer@McUnixJr.gold.supporter.pdpc) |
| 17:54.38 | *** part/#asterisk flot (CCCP@213.152.157.68) |
| 17:54.47 | bjohnson | Bob_: if you are only using softphones, then yes you should definitely use an iax client |
| 17:55.03 | bjohnson | visik7: entry phone? |
| 17:55.17 | bjohnson | visik7: you mean like an intercom phone? |
| 17:55.43 | Gand_DJ | anyone familiar with the digital receptionist in *? |
| 17:55.56 | bjohnson | jief-: try sip debug or tethereal and watch the packets |
| 17:56.09 | bjohnson | Gand_DJ: the IVR? |
| 17:56.14 | jief- | bjohnson: i tried sip debug. i get nothing. |
| 17:56.21 | Gand_DJ | yeah |
| 17:56.25 | visik7 | bjohnson the "phone" at the entrance of my house |
| 17:56.33 | bjohnson | jief-: the you likely have phone config issues or network routing problems |
| 17:57.03 | bjohnson | visik7: a few voip phones will auto-answer you calling their extension .. should be info on the wiki |
| 17:57.08 | jief- | bjohnson: can't be routing, * and the 2 phones are on the same switch. could be a phone issue, but im so new to this, i dont even know if all my config is ok |
| 17:57.12 | bjohnson | Gand_DJ: what about her? |
| 17:57.27 | *** part/#asterisk oej (~oej@40.186.204.213.sol.worldonline.se) |
| 17:57.32 | bjohnson | jief-: neither do I .. you haven't said what phones you have |
| 17:57.32 | Marlow | tzanger: stupido ... what sense is there in using the same guid all over the bloody place .. |
| 17:57.43 | jief- | bjohnson: we have Gnet SIP phones |
| 17:57.45 | Marlow | tzanger: that's plain insane .. |
| 17:57.50 | tzanger | Marlow: it works perfect if you don't have more tha none |
| 17:57.53 | tzanger | er more than one |
| 17:57.58 | Gand_DJ | Right now, when a prompt is played.. if no response is heard, the call is dropped. Is there a way that if there is no response when the prompt is done, to then goto another prompt? |
| 17:58.17 | visik7 | bjohnson mmm with video support ? |
| 17:58.22 | Marlow | tzanger: sure .. but that's not really feasable .. |
| 17:58.36 | tzanger | Marlow: I never said they made any sense |
| 17:58.38 | Marlow | tzanger: el cheapo tricks |
| 17:58.42 | tzanger | you know what's funny |
| 17:58.46 | Gand_DJ | I have a prompt setup to dial 1 for help, or 2 for voicemail... but if user presses nothing, call is dropped 3-5 seconds after prompt is done. |
| 17:58.50 | Marlow | tzanger : what ? |
| 17:58.52 | tzanger | I had this exact same issue with 4 GVC PnP modems about 6 years ago |
| 17:58.55 | jief- | on my phones, ive set my registrar server and proxy outbound to the * IP. is that ok? |
| 17:59.02 | Marlow | tzanger: eh :) |
| 17:59.06 | Blackvel | Could I explicitely drop Manager API "Event: Registry " posts? I am not interested in those events |
| 17:59.09 | tzanger | they all had the same serial # (1) -- I downloaded the PNP spec whcih said if you don't want to serialze them use -1 for the s/n |
| 17:59.11 | Blackvel | nor do I want to parse them |
| 17:59.12 | bjohnson | jief-: not familiar with them. basic troubleshooting is set them to host=dynamic in sip.conf and have them register to *. Also do set verbose 5 in * cli |
| 17:59.19 | tzanger | so I pulled the eeprom off programmed 4 unique #s and resoldered |
| 17:59.21 | bjohnson | visik7: no idea |
| 17:59.25 | tzanger | bingo, 4 workign PnP modems in one case |
| 17:59.27 | tzafrir | I asked if anybody's us*ing* voipgate, because I can't even get even a DNS resolution for them |
| 17:59.38 | jief- | bjohnson: do i set usernames/secrets? or just host=dynamic ? |
| 17:59.38 | bjohnson | Gand_DJ: you use backgroun? |
| 17:59.42 | bjohnson | Gand_DJ: you use background? |
| 17:59.47 | Gand_DJ | yeah |
| 17:59.57 | *** join/#asterisk zeedo (~zeedo@www.bsrf.org.uk) |
| 17:59.59 | Marlow | tzanger: still bullshit .. somebody who didn't read the specs |
| 18:00.05 | Gand_DJ | exten => s,7,Background(custom/aa_1) |
| 18:00.08 | bjohnson | jief-: you don't have to set usernames/passwords .. but most people want to separate their phones so they use different info |
| 18:00.10 | Marlow | tzanger: properly |
| 18:00.57 | bkw_ | OK WHERE IS EVERYONE |
| 18:01.00 | bkw_ | its 1pm CST |
| 18:01.09 | jief- | [501] |
| 18:01.09 | jief- | type=friend |
| 18:01.09 | jief- | host=dynamic |
| 18:01.10 | jief- | context=internal |
| 18:01.16 | jief- | so this would be enough in sip.conf? |
| 18:01.23 | Marlow | bkw_: everyone is going home .. it's 7pm GMT :) |
| 18:01.28 | evilrabbi | bkw_? |
| 18:01.36 | evilrabbi | i'm in my office |
| 18:01.38 | evilrabbi | ! |
| 18:01.53 | bjohnson | Gand_DJ: just make a s,8 extension |
| 18:03.03 | Gand_DJ | ok... should I have a wait command first? |
| 18:03.13 | Gand_DJ | or just have 3 background commands 1 after anothert? |
| 18:03.43 | bjohnson | jief-: if you don't get connection or errors on that .. the phones are not configured properly |
| 18:03.59 | bjohnson | Gand_DJ: try it both ways |
| 18:04.39 | Gand_DJ | What's the difference between DigitTimeout and ResponseTimeout |
| 18:04.46 | bjohnson | Gand_DJ: I forget |
| 18:04.57 | gambolputty | anyone tried to call iax.cc? |
| 18:04.57 | bjohnson | Gand_DJ: check wiki or show application |
| 18:05.00 | *** join/#asterisk bannerman (~bannerman@209.216.176.43) |
| 18:05.20 | bjohnson | Gand_DJ: one is time between digits being entered |
| 18:05.35 | Gand_DJ | ok |
| 18:06.16 | ManxPower | Gand_DJ: RTFAD |
| 18:06.24 | Gand_DJ | lol |
| 18:06.45 | jief- | bjohnson: when i do 'sip show peers', i see my 2 phones. but they are unmonitored. and i still get a busy signal after a while when i try to reach an extension |
| 18:07.03 | Bob_ | bjohnson - have you done any development with iax? |
| 18:08.23 | bjohnson | Bob_: development? |
| 18:08.40 | ManxPower | jief-: Do you see the IP addresses of the devices when you do a "sip show peers". If not then the devices are NOT registering. |
| 18:08.44 | bjohnson | jief-: add qualify=yes to the sip.conf sections |
| 18:08.56 | bjohnson | jief-: did you define extensions for them in extensions.conf? |
| 18:08.57 | Bob_ | bjohnson - sorry, think I forgot to mention that I have built a custom softphone |
| 18:09.13 | Bob_ | bjohnson - so would need to integrate iax instead of sip |
| 18:09.19 | jief- | that's for my extensions: exten => _5XX,1,Dial(SIP/${EXTEN}) |
| 18:09.21 | bjohnson | jief-: do you have a fxo or voip account? is it configured? can they dial out? |
| 18:09.27 | mmlj4 | ManxPower: no John? |
| 18:09.35 | jief- | i just wanna set my phones for internal uses for now |
| 18:10.13 | ManxPower | mmlj4: he's not due for a while yet. |
| 18:10.18 | mmlj4 | ah, ok |
| 18:10.26 | ManxPower | He's somewhere in Mississippi getting a new office on the network |
| 18:10.41 | mmlj4 | and you're in La.? |
| 18:10.50 | mmlj4 | or at home? |
| 18:10.59 | Marlow | can anyone give me a reseller for Draytek in the U.S. ? |
| 18:11.00 | ManxPower | at home. |
| 18:11.07 | mmlj4 | yeah, that new office, i heard about it |
| 18:11.11 | mmlj4 | small, right? |
| 18:11.21 | bjohnson | jief-: try making one exten for each phone |
| 18:11.21 | ManxPower | Some people at the Mandeville office have to be bitch smacked and I figured it would be more effective if I did it in person. |
| 18:11.35 | mmlj4 | heh, mind if i watch? |
| 18:11.39 | bjohnson | jief-: and drop the 'SIP/' from the dial command |
| 18:11.43 | jief- | bjohnson: ok. ill try that. i have to leave on a call for now. thanks for your help. ill come back when i have more time |
| 18:11.45 | ManxPower | I also have to figure out why the channel bank isn't working and need to install a hardware echocan device |
| 18:12.05 | mmlj4 | yeah... i'll bet it's the wrong amphenol type |
| 18:12.10 | ManxPower | mmlj4: me too |
| 18:12.22 | mmlj4 | one more person to slap? heh |
| 18:12.49 | shido6 | :) |
| 18:12.53 | mmlj4 | doing another overnighter? |
| 18:13.58 | *** join/#asterisk pawalls (~pawalls@pawalls.teamgleim.com) |
| 18:14.36 | pawalls | What's the best way sniff phone line traffic, and what hardware is needed? |
| 18:14.52 | HeadachesAbound | anyone interested in answering machine detection using frequencies? |
| 18:14.54 | pawalls | I basically want to see the protocol that's being used by a proprietary handset. |
| 18:15.01 | denon | anyone have a rough idea what Internet E1s cost in Mexico? |
| 18:15.05 | Bob_ | cheers guys - I am out of here |
| 18:15.10 | denon | mx city |
| 18:17.01 | *** join/#asterisk RoyK (~roy@ti211310a080-4502.bb.online.no) |
| 18:17.04 | ManxPower | mmlj4: no, i'll see the significant others this weekend too and come back sunday afternoon |
| 18:17.04 | *** join/#asterisk n4y (~tmalkut@fire2.orasoft.net.pl) |
| 18:17.18 | *** join/#asterisk alejandrosg (~chatzilla@201.134.223.2) |
| 18:17.21 | mmlj4 | ah, ok |
| 18:17.22 | denon | ManxPower: you the one that always has stuff in guatemala? |
| 18:17.45 | ManxPower | denon: No! |
| 18:17.50 | denon | haha sorry |
| 18:17.55 | denon | didnt remember who that was |
| 18:18.15 | bjohnson | HeadachesAbound: been some discussion on the mailing list |
| 18:18.44 | bjohnson | pawalls: tetehreal |
| 18:18.44 | alejandrosg | Hi, someone from nufone? |
| 18:18.50 | denon | this isnt the nufone support channel |
| 18:19.30 | *** join/#asterisk darby_t (~tom@dmn207.neoplus.adsl.tpnet.pl) |
| 18:19.46 | bjohnson | take that you yogourt |
| 18:20.01 | alejandrosg | Hi denon, shure, I not ask from support to nufone... |
| 18:20.03 | bjohnson | oh .. danone |
| 18:20.10 | denon | [17:43] <JerJer> this is not a nufone support channel |
| 18:20.19 | denon | im only quoting |
| 18:20.23 | *** join/#asterisk cmk (~cmk_@p54A3D93C.dip.t-dialin.net) |
| 18:20.45 | HeadachesAbound | Been digging thru the mailing list, but haven't seen it there. I have a dialer at present that uses frequency detection and would like to try and implement it in dsp.c but I haven't the foggiest clue where to begin. |
| 18:20.47 | three55ml | Anyone have experience with Snom phones? I have a user with a problem registering with one. |
| 18:22.03 | ManxPower | HeadachesAbound: the past person that asked for help creating a predictive dialer was found dead, in a ditch the next day. |
| 18:22.06 | ManxPower | Just letting you know. |
| 18:22.22 | bjohnson | they kicked his assterisk |
| 18:22.38 | *** join/#asterisk obsidian-studios (~obsidian-@c-66-177-188-197.hsd1.fl.comcast.net) |
| 18:23.14 | alejandrosg | denon: :-d |
| 18:23.16 | alejandrosg | :-d |
| 18:23.18 | bjohnson | HeadachesAbound: btw, program me as not interested |
| 18:23.23 | alejandrosg | ought... :-D |
| 18:23.42 | *** part/#asterisk watson (cancerea@203.170.71.63) |
| 18:23.49 | HeadachesAbound | thanks for the warning ManxPower. I'm not into telemarketing and most of the people we are calling deserve to be called. I also get quite sick anytime i think about someone using asterisk for telemarketing. |
| 18:24.03 | obsidian-studios | anyone know why when leaving a voicemail why it would get sped up? Seems to happens like in between sentences or when pausing for like a main menu. Not to mention the end of the message is getting truncated? |
| 18:24.22 | *** join/#asterisk TheEmperor (user@218.111.50.117) |
| 18:24.25 | ManxPower | The guy that did it said in his defense "But he wasn't on the doo not kill list!" |
| 18:25.01 | *** join/#asterisk R3DB0x (nobody@66.142.28.36) |
| 18:25.07 | *** join/#asterisk chaoscon (~ph33r@chaoscon.user) |
| 18:25.37 | *** join/#asterisk Hydr0p0nx (hidden-use@nat.wwisp.com) |
| 18:25.54 | Hydr0p0nx | anyone have any experience with sonicwall firewalls and VOIP ? |
| 18:26.00 | *** join/#asterisk Nuxi (~nuxi@lin-dsl-static-206-222-194-115.inetnebr.com) |
| 18:26.38 | bjohnson | ManxPower: but the killers didn't care because they were coming in from out of country |
| 18:26.44 | pawalls | bjohnson, If it's a digital signal, I'll need a null modem, eh? |
| 18:27.17 | bjohnson | pawalls: didn't you say they were voip phones? |
| 18:30.50 | *** part/#asterisk Marlow (~martin@cerberus.bluetree.ie) |
| 18:36.26 | Gand_DJ | ne1 uses one of those toll-free #'s that broadvoice offers? |
| 18:36.43 | Gand_DJ | I'm looking to add a toll-free #. |
| 18:37.10 | Gand_DJ | My local telco has one for $2.95 / month CDN... no free minutes.. 12 cents / min for canada, and 19 cents CDN from usa calls |
| 18:39.11 | bjohnson | Gand_DJ: switch to allstream |
| 18:39.28 | bjohnson | Gand_DJ: non-voip .. no monthly fee .. $0.04/min |
| 18:39.57 | bjohnson | Gand_DJ: same thing at selectcom I think |
| 18:40.02 | Gand_DJ | Guessing that is USD, which is still cheaper then 12 cents cdn |
| 18:40.14 | bjohnson | guessing wrong |
| 18:40.19 | bjohnson | both are Cdn companies |
| 18:40.34 | Gand_DJ | even better |
| 18:40.48 | bjohnson | not voip though .. so some limitations on features |
| 18:40.59 | Gand_DJ | Just need callerid |
| 18:41.16 | Gand_DJ | thinking of getting the sipura 3000 for fxo to my * ivr |
| 18:41.50 | Gand_DJ | one thing I haven't been able to find though, is if I setup fxo to forward pstn to * for answering.... can you still use * to call out via sp3000 to pstn #? |
| 18:42.45 | bjohnson | definitely |
| 18:43.01 | Gand_DJ | Ok. I heard the sp3000 would auto-forward your request back to * |
| 18:43.06 | TheEmperor | does anyone know how to create a user account that has a prepaid limit? i mean, you let them call and a voice tells them how much money they have left and if they don't top up they get disconnected? |
| 18:43.07 | bjohnson | fxo are not 1 way |
| 18:43.43 | bjohnson | TheEmperor: seach the wiki for prepaid or calling card systems |
| 18:44.08 | TheEmperor | bjohnson:which one do you recommend? areski, astcc? |
| 18:44.11 | bjohnson | TheEmperor: basically, you'll be custom programming at least a bit |
| 18:44.19 | bjohnson | TheEmperor: I haven't used them |
| 18:44.44 | TheEmperor | anyone want to do it for me? i'll pay :) |
| 18:44.45 | *** join/#asterisk loick (~loick@APuteaux-151-1-5-140.w82-120.abo.wanadoo.fr) |
| 18:45.04 | bjohnson | Gand_DJ: then your heard from someone who didn't understand what was going on |
| 18:45.17 | bjohnson | TheEmperor: try the dev mailing list for better results |
| 18:45.25 | Gand_DJ | bjohnson, you use the sp3000 right? |
| 18:45.28 | bjohnson | TheEmperor: or the biz mailing list |
| 18:45.28 | TheEmperor | bjohnson:ok thanks for that |
| 18:45.39 | bjohnson | Gand_DJ: Sipura SPA 3000 to be exact |
| 18:45.45 | bjohnson | TheEmperor: no charge |
| 18:45.52 | TheEmperor | :) |
| 18:46.25 | Gand_DJ | Do you have the 3000 auto-pass the call to *, so the fxo port doesn't actually answer the call, or does the fxo port first answer call, then forward to *? |
| 18:46.46 | bjohnson | I do it the most logical way (which of course is also the harder way) |
| 18:46.55 | bjohnson | the spa doesn't answer until * tells it to |
| 18:48.04 | Gand_DJ | Isn't that the method where you have the spa auto-forward the pstn call to * and then * will then either send it to your softphone to ring, ivr, or whever it's programmed to go? |
| 18:49.17 | Gand_DJ | It would auto-pass the CID from the pstn line... by using an * in the pass-cid field of the spa |
| 18:50.13 | Bile_One | has anyone looked at the Soyo g668 yet? |
| 18:50.27 | shido6 | areski needs more work |
| 18:51.56 | McUnixJr | is there an addon for a party line? |
| 18:52.23 | *** part/#asterisk Fanguin (~Fanguin@p548F12CC.dip0.t-ipconnect.de) |
| 18:54.40 | ManxPower | McUnixJr: It's called chan_booze.so |
| 18:54.47 | ManxPower | It's used with chan_easywomen.so |
| 18:54.53 | tzanger | oh man |
| 18:54.55 | tzanger | I need some of both |
| 18:54.59 | tzanger | where can I get the source to that |
| 18:55.16 | ManxPower | tzafrir: Sorry, I'm under NDA. |
| 18:55.17 | MattB2 | closely realted to chan_dollar-a-minute.so? |
| 18:55.29 | ManxPower | MattB2: not if it's a good party line |
| 18:56.24 | *** join/#asterisk darby_t (~tom@dmn207.neoplus.adsl.tpnet.pl) |
| 18:59.31 | obsidian-studios | when call comes in via a zap channel that I have going to a context, when I ring a sip phone, the caller id info is not there, but if a message is left it is? What do I need to add to my Dial(SIP/5550,30) to pass along caller id info ? |
| 19:00.46 | Uther_P | what kind of phone is it? |
| 19:00.51 | ManxPower | obsidian-studios: you don't. |
| 19:00.57 | ManxPower | Asterisk will pass callerid if it can |
| 19:01.25 | three55ml | Anyone use a Snom 360? |
| 19:01.27 | Uther_P | if it shows nothing for callerid, then the phone probably doesn't support it |
| 19:01.34 | obsidian-studios | ok I think it's the cisco router keeping that info from passing along, the sip peer is a fxs port on a cisco router, so analog phone |
| 19:01.53 | ManxPower | obsidian-studios: I guess it's time to look at www.cisco.com then, isn't it? |
| 19:02.02 | *** join/#asterisk fugitivo (~ajf@201.255.99.226) |
| 19:02.43 | obsidian-studios | yes but I was not sure if the problem would be there or *, so it seems like the Cisco end. It also seems to be speeding up conversations and reducing pause gaps and etc :) |
| 19:03.07 | ManxPower | obsidian-studios: turn off VAD on the Cisco |
| 19:03.11 | obsidian-studios | it is off |
| 19:03.19 | Uther_P | heh, it speeds up conversations? like, saying words you haven't said yet? :P |
| 19:03.32 | Uther_P | thats convinient |
| 19:03.34 | obsidian-studios | I have seen this before in other setups with the low end cisco units |
| 19:03.42 | ManxPower | Don't worry, grasshopper, you will soon stop dealing with Cisco for voice and start using Asterisk for your phones too. |
| 19:04.06 | Uther_P | these are not the routers you're looking for |
| 19:04.07 | obsidian-studios | not really I was recording a main menu and no matter how slow I talked it got sped up. I would leave like 2-3 second gaps between press this or that, and it would get reduced to like under a second :) |
| 19:04.21 | ManxPower | obsidian-studios: then vad isn't off |
| 19:04.59 | obsidian-studios | FYI, I use the Cisco's because they are there terminating internet access, and have fxs ports, not to mention in the past they have allowed me to do faxing and credit card processing via cc terminals over VOIP :) |
| 19:05.14 | Uther_P | wow, why the hell would it do that? is that a silence suppression? if it does that then it doesn't seem like its real time |
| 19:05.48 | Uther_P | to do fax you just need a high jitter buffer |
| 19:05.59 | obsidian-studios | <PROTECTED> |
| 19:05.59 | obsidian-studios | <PROTECTED> |
| 19:06.01 | obsidian-studios | it's there |
| 19:08.12 | Gand_DJ | I'm talking to allstream... they said it's 5.5cents / minute for calls from canada / usa |
| 19:08.19 | obsidian-studios | anyone care to comment why a el cheapo basic modem used as a fxo picks up lines faster than a tdm400 with 4 fxo modules |
| 19:08.22 | Gand_DJ | they are finding out about setup costs |
| 19:09.12 | McUnixJr | ManxPower, where can i find chan_booze ? |
| 19:10.16 | ManxPower | McUnixJr: Sorry, I'm under NDA. |
| 19:10.21 | kiokorobert | anyone successfully implemented fax over IP with * ? |
| 19:10.32 | McUnixJr | huh? |
| 19:10.36 | obsidian-studios | yes I have |
| 19:10.44 | kiokorobert | how? |
| 19:10.45 | obsidian-studios | with cisco routers via their fxs ports |
| 19:11.03 | Signuts | kiokorobert, one thing to turn off is echo cancellation and friends on the zap channel. |
| 19:11.25 | McUnixJr | ManxPower, err - okay, an NDA, but you told me something exists.... |
| 19:11.27 | kiokorobert | how about sip clients like the sipuras |
| 19:11.43 | obsidian-studios | possible but the ciscos have allot of stuff to preserve signal and etc |
| 19:11.46 | ManxPower | McUnixJr: I did not! |
| 19:11.50 | obsidian-studios | for faxes and etc |
| 19:11.50 | ManxPower | 8-) |
| 19:12.57 | Uther_P | obsidian-studios: the modem doesn't wait for caller-id information... |
| 19:12.57 | McUnixJr | man....ok ill ask again - i want a party line for SIP clients to call into my Asterisk box - how to implement cause im a total newb/noob |
| 19:12.58 | McUnixJr | :) |
| 19:13.14 | Uther_P | obsidian-studios: if you want it to pickup faster, turn off callerid for the zap channels |
| 19:13.17 | obsidian-studios | Uther_P: which one |
| 19:13.26 | *** join/#asterisk jetdotnet (~jetdotnet@adsl-64-219-216-41.dsl.hstntx.swbell.net) |
| 19:13.27 | Uther_P | which one hat? |
| 19:13.36 | obsidian-studios | Uther_P: I do not have caller id working at all with the tdm400 it worked right off the bat with my el cheapo modem ? |
| 19:13.56 | *** join/#asterisk RChadwell (~rob@rrcs-67-78-145-50.se.biz.rr.com) |
| 19:13.57 | obsidian-studios | complete going against the tdm400 being easier and better? |
| 19:14.29 | *** join/#asterisk kFuQ (~somedude@c-24-17-224-78.hsd1.wa.comcast.net) |
| 19:15.25 | obsidian-studios | I mean I will have to go see what I did for caller id to not work on the tdm400 like it does for me on my el cheapo modem? However my problem is the caller id info is not passed to analog phones, which I am pretty sure is the cisco routers config doing that. |
| 19:15.53 | *** join/#asterisk UBiQUiTY (~mike@68.160.103.76) |
| 19:15.57 | Gand_DJ | bjohnson, are you part of CFIB? |
| 19:16.00 | jakepdev | anyone know how to implement * to cancel a string when using cmd Read? |
| 19:16.24 | *** join/#asterisk Tili (~Tili@202-133-65-235-dialup.sat.net.pk) |
| 19:17.12 | RChadwell | Wanna hear something interesting? - We had Asterisk running on our LAN with much success - 1 hour ago we moved it from our LAN to our WAN (to connect other offices) and we immediately had a problem where we would lose audio unless we talked (or softly blew into the phone which made our office sound like it was full of snakes). Does anyone know of reasons why audio would screw up like that? |
| 19:17.32 | obsidian-studios | but I got bigger problems, I just converted a wav to mp3 via sox, and now it's super fast, so something is not right with my playback env :) |
| 19:17.58 | Uther_P | did you specify the wrong sampling rate? |
| 19:18.21 | Gand_DJ | RChadwell maybe from VAD? |
| 19:18.27 | Gand_DJ | not sure |
| 19:19.16 | RChadwell | I thought about that - I unchecked all silence suppression, but to no avail - very strange. I was getting in the car to go connect the other offices - would have been my last day working here. :) |
| 19:19.33 | obsidian-studios | no I am using the same sox commands I have for a ton of mp3 for *. I got a client who likes to change their menu and etc like at least once a month or so. Totally pain, I am trying to get them to at least stick with a standard main menu and only change certain sub menu's etc |
| 19:20.03 | ManxPower | ARGH! The facilities Astricon Europe is at has an ALL FLASH web site. |
| 19:20.16 | obsidian-studios | something has to be jacked with my env, I mean when I compiled and installed * and zapata, and zaptel, I never got a chan_zap.so ? |
| 19:20.24 | obsidian-studios | I used to love Flash, but no so much any more |
| 19:20.59 | shido6 | stop freaking out |
| 19:21.00 | *** join/#asterisk Slainte (~Slainte@66.55.113.158.ppp.northrock.bm) |
| 19:21.16 | shido6 | 318 ppl and at least one of us can help you RChadwell |
| 19:21.33 | Slainte | When I use the Festival cmd, it appears to lock up the channel, and a soft-hangup does not release it. |
| 19:21.55 | *** join/#asterisk oej (~oej@40.186.204.213.sol.worldonline.se) |
| 19:22.03 | obsidian-studios | fesitval is cool, but is hard to understand I had many problems with it and that sounds like one |
| 19:22.20 | Slainte | obsidian, did you get it working? |
| 19:22.24 | fugitivo | buy cepstral |
| 19:22.54 | obsidian-studios | yes it was cool |
| 19:22.59 | obsidian-studios | client did not like it |
| 19:23.41 | Slainte | do you recall how you worked around the issue I mentioned? |
| 19:23.43 | obsidian-studios | I did because the data that get's updated could be text based an update on site, pos systems, * all via updating the stuff in db |
| 19:24.23 | RChadwell | Shido6: Any Ideas? |
| 19:24.26 | obsidian-studios | would have to see your extension and etc to see what's going on. How do you get it to release? |
| 19:24.47 | *** join/#asterisk Cresl1n (~matt@216.207.245.23) |
| 19:25.03 | UBiQUiTY | does anybody know about the asterisk command line option "-p" for "pseudo-realtime thread" ? |
| 19:25.06 | *** join/#asterisk FarrisG (~jrush@h-68-164-19-170.dllatx37.covad.net) |
| 19:25.25 | Slainte | I tried to soft-hangup, but the only way was to Zap detroy, and then restart asterisk to get the channel back |
| 19:25.40 | UBiQUiTY | i am looking to run asterisk in real-time priority, is the -p option the correct way to do so? |
| 19:25.41 | FarrisG | can you change the codec/filetype used when sending voicemail through email? |
| 19:26.05 | obsidian-studios | paste bin your festival.conf and any relevant contexts |
| 19:26.14 | Slainte | k one sec |
| 19:26.14 | obsidian-studios | FarrisG: yes it's one of the first settings in voicemail.conf |
| 19:26.31 | obsidian-studios | format=g723sf|wav49|wav |
| 19:26.45 | obsidian-studios | but only the first is emailed, I prefer wav49 as they are the smallest file size, wav |
| 19:26.53 | obsidian-studios | wavs seem to be the largest |
| 19:27.17 | ManxPower | FarrisG: No. It will send whatever format is recorded. If you want something else you'll have to write a script that's called with externnotify= |
| 19:27.27 | FarrisG | obsidian-studios: That's what seems odd to me. Mine is set to "wav49|gsm|wav" but it's GSM-encoded wavs that are being emailed |
| 19:27.47 | obsidian-studios | get rid of the later ones only the first is used |
| 19:27.59 | obsidian-studios | also if you have all I believe the email will be kept locally on the machine in all formats |
| 19:28.31 | obsidian-studios | I do not keep copies on the machine, I just do format=wav49 and have it delete the message after emailing |
| 19:28.57 | *** join/#asterisk critch (critch@steven.basesys.com) |
| 19:29.57 | critch | any known issues with te410p and getting the D channel up on a PRI? |
| 19:30.34 | Slainte | critch, what OS? |
| 19:30.42 | critch | linux |
| 19:30.43 | Slainte | I know someone was having issues with Fedora C3 |
| 19:30.53 | critch | FC3 is a distro not an OS |
| 19:30.55 | Slainte | I mean what Distro |
| 19:30.57 | obsidian-studios | critch: really * on linux :) |
| 19:31.07 | critch | Debian |
| 19:31.21 | *** join/#asterisk syslod (~yurplsl@65.114.15.70) |
| 19:31.25 | FarrisG | Ok, I set it to format=wav49, but it's still sending GSM-encoded files |
| 19:31.27 | syslod | Hello. |
| 19:31.28 | Slainte | what is ztcfg telling you |
| 19:31.58 | Slainte | obsidian-studios: http://pastebin.ca/10522 |
| 19:32.00 | obsidian-studios | FarrisG: you restart |
| 19:32.09 | obsidian-studios | FarrisG: does * have the proper codecs loaded? |
| 19:32.10 | FarrisG | obsidian-studios: I reloaded. Is a full restart needed? |
| 19:32.16 | obsidian-studios | FarrisG: in your modules.conf ? |
| 19:32.50 | FarrisG | obsidian-studios: Maybe not. No codecs are mentioned in my modules.conf |
| 19:33.11 | obsidian-studios | I only load the ones I want |
| 19:33.17 | obsidian-studios | total pain to dial in |
| 19:33.36 | obsidian-studios | Slainte: have you tried festival without the usecache? |
| 19:33.42 | *** part/#asterisk Gand_DJ (~gandalf@ptr-207-54-104-24.ptr.terago.ca) |
| 19:34.09 | Slainte | no, and now am I nervous it is going to foobar the channel again (production machine) |
| 19:34.35 | Slainte | I will try in half an hour at EoB |
| 19:34.59 | obsidian-studios | fyi, I do hangups like exten => t,1,Hangup() |
| 19:34.59 | *** join/#asterisk Rick_Hunter (~rhunter@66.19.18.135) |
| 19:35.25 | obsidian-studios | or they way you have should work and release the channel |
| 19:35.48 | Slainte | Yeah I wonder if it is not hanging up properly. But the CLI is saying is is receiving the hangup, its saying it hung up, but a zap show channels show it in use |
| 19:36.09 | obsidian-studios | might be your zap settings then, festival and extensions look good |
| 19:36.24 | Slainte | lemme post my zapata.conf |
| 19:36.44 | durex | Slainte pastebin.ca |
| 19:37.31 | obsidian-studios | durex: should have gone for magnum :) |
| 19:37.59 | heison | anyone here from Paris? |
| 19:38.13 | obsidian-studios | Slainte: so you just want to call in via zap channel have it talk to you and hang up |
| 19:38.29 | durex | :-) |
| 19:38.29 | Nuxi | the one in France or Illinois? |
| 19:38.30 | Slainte | http://pastebin.ca/10524 |
| 19:38.41 | heison | Paris, France ;) |
| 19:38.47 | *** join/#asterisk mozrat (~mozrat@80.68.89.215) |
| 19:38.56 | Slainte | obsidian, jsut to start yes. It is a spare DID so I want to just test the reading, and playback |
| 19:39.35 | obsidian-studios | you do hear festival right? |
| 19:40.27 | Slainte | nooe, I dont even hear festival, but it is giving no errors on the CLI. I did have a connection error, but fixed the ACL, now it looks like it is playing but I do not hear it |
| 19:40.34 | obsidian-studios | do you see anything funky in ps xa? I remember I would see festival in some sort of [] not a seg fault but something close and at times I had to reboot the machine to get rid of the process |
| 19:40.53 | ManxPower | Does festival work OUTSIDE of asterisk |
| 19:40.54 | ManxPower | ? |
| 19:40.55 | Slainte | nope, it was clean. not zombied or anything |
| 19:40.57 | obsidian-studios | ok well you will need to hit up the log file and do a asterisk -vvvvvvr while making the call |
| 19:41.13 | Slainte | the -vvvvvvvvvvvr shows nothing suspsicious |
| 19:41.18 | obsidian-studios | ManxPower: my next questions :) should have been first, but I assumed it did |
| 19:41.30 | ManxPower | try stopping Asterisk, then starting it as "asterisk -cvvvddd" |
| 19:41.39 | obsidian-studios | it's not puking anything about the festival command I used to see allot of stuff there |
| 19:41.53 | Slainte | Will do that in 20 mintues (end of business then ) |
| 19:41.54 | shido6 | "/etc/asterisk/logger.conf" |
| 19:41.59 | ManxPower | maybe you have to issue and Answer() first? |
| 19:42.16 | Slainte | I do have an Answer first |
| 19:42.25 | obsidian-studios | yes |
| 19:42.33 | Blackvel | guys, where are the manager api gods here? |
| 19:42.35 | obsidian-studios | well if it's getting the zap channel directly |
| 19:42.49 | obsidian-studios | like if the zap channel is mapped to that context, if another context answers the zap channel then no |
| 19:42.56 | *** join/#asterisk Mavvie (edwin@edwin.adsl.barnet.com.au) |
| 19:44.46 | RChadwell | Gand_DJ: Yep, you were right! Several clients had VAD turned back on (they thought it would make them sound better). Cue - Napoleon Dynamite Clip "Ugh! Idiot!" :) |
| 19:46.29 | RChadwell | and thanks for the "Open Source" support for an open-source product. :) |
| 19:46.43 | *** part/#asterisk RChadwell (~rob@rrcs-67-78-145-50.se.biz.rr.com) |
| 19:53.21 | mozrat | Hey guys - does anyone have an example python agi script they could give me? I've written one but it doesn't seem to run or print to the asterisk console like I thought it should |
| 19:53.29 | *** join/#asterisk sean (~sean@iconoclast.caedmon.net) |
| 19:53.33 | bjohnson | any bash scripters here? I know how to use a for loop to make filenames like file1.txt, file2.txt .. how can I make it file001.txt through file012.txt? |
| 19:53.52 | sean | opinion needed: best "free" softphone? |
| 19:53.59 | sean | (windows -> SIP or IAX(2)) |
| 19:54.19 | mozrat | Free as in price or freedom? |
| 19:54.29 | mozrat | XLite is quite nice |
| 19:54.44 | sean | gratis -- libre if possible, but not necessary. |
| 19:55.11 | mozrat | linphone works but isn't too pretty :-) |
| 19:55.13 | shido6 | get a noise cancelling mic |
| 19:55.18 | sean | xlite isn't performing as well as I'd hoped, so I want to narrow down my clipping/cutting issues (eliminate the phone). |
| 19:55.35 | sean | hardware, I'm not worried about, yet.. (mic/headphones is what I'm using ATM) |
| 19:56.47 | sean | ... secondary opinion: anyone got a free (again, gratis) way to make asterisk hook into a VXML browser (I want my asterisk to render vxml). |
| 19:59.14 | shido6 | sean, stdby |
| 19:59.31 | ManxPower | sean: spend the money on a SIPura SPA-841 |
| 19:59.36 | pjz | bjohnson: man seq |
| 20:00.33 | pjz | is there any way to have asterisk turn up the volume on calls? |
| 20:00.48 | Bile_One | txgain |
| 20:00.49 | pjz | there's a general complaint that the calls are too soft |
| 20:00.59 | ManxPower | pjz: rxgain / txgain in /etc/asterisk/zapata.conf |
| 20:01.01 | pjz | Bile_One: does that work on internal extensions? |
| 20:01.05 | Bile_One | pjz turn up by adding to txgain |
| 20:01.49 | Bile_One | pjz Yes. but it is flat at 0.0 so be wise to add a bit at a time, like 1.0 gain |
| 20:01.50 | pjz | is tx toward or away from the zap line? |
| 20:02.02 | Bile_One | pjz away |
| 20:02.10 | Bile_One | rx is incoming |
| 20:02.41 | ManxPower | dB is not flat |
| 20:02.41 | pjz | so if the incoming voice on the lines is too soft, I need to adjust... ? |
| 20:02.48 | ManxPower | The numbers for rxgain and txgain are in dB |
| 20:02.48 | Bile_One | pjz you'll have to balnace volume and echo. Make sure to enable echo cancelling too |
| 20:02.49 | pjz | yeah, db is exponential |
| 20:03.12 | Bile_One | rxgain |
| 20:03.20 | pjz | so I'm still getting echo problems on purely internal extension |
| 20:03.26 | pjz | anyone have hints? |
| 20:03.41 | Bile_One | what are you using and what is your taffic like? |
| 20:03.44 | johnnyb | rxgain is for receive, txgain is for transmit. |
| 20:03.45 | *** join/#asterisk los415 (~los415@c-24-126-63-233.hsd1.ca.comcast.net) |
| 20:03.53 | johnnyb | Digium told me not to set either above 5.0 |
| 20:04.03 | johnnyb | I have mine set to 5.0 and 4.0, respectively. |
| 20:04.09 | *** join/#asterisk darby_t (~tom@dmn207.neoplus.adsl.tpnet.pl) |
| 20:04.12 | Bile_One | Haa haa haa, I tried 2.0 once and echo killed the connection' |
| 20:04.19 | obsidian-studios | if it's recordings, amplify them via sox -v where volume levels can be 1-9 |
| 20:04.53 | pjz | Bile_One: I've got a beefy astgerisk server, switched 100Mb net that I don't think is too loaded, and I get tech on both my ipDialog SipTone II and my uniden UIP200 |
| 20:05.03 | Bile_One | So pjz, what is the QoS like on your network? What phones are you using? |
| 20:05.13 | ScaredyCat | astgerisk ... mmmmm |
| 20:05.20 | ManxPower | pjz: it is IMPOSSIBLE to get echo when the audio path is ALL digital. |
| 20:05.33 | pjz | ManxPower: I'm getting it |
| 20:05.34 | ManxPower | pjz: So you are hearing something that sounds like echo, but is not. |
| 20:05.40 | pjz | ManxPower: okay |
| 20:05.47 | ScaredyCat | phased output |
| 20:05.49 | Bile_One | Could be dely, but I bet it is QoS |
| 20:05.55 | opus_ | its a ghost |
| 20:05.56 | pjz | ManxPower: I hear myself talk. delayed. Isn't that the definition of echo? |
| 20:05.58 | johnnyb | pjz: it could be delay or jitter |
| 20:06.10 | Bile_One | NO delay is not echo |
| 20:06.25 | johnnyb | pjz: no -- echo would sound cave-like |
| 20:06.28 | pjz | ah |
| 20:06.29 | pjz | okay |
| 20:06.29 | ManxPower | pjz: Echo is caused by 2-wire analog circuits. |
| 20:06.31 | pjz | so it's delay |
| 20:06.35 | *** join/#asterisk jwitte (~jwitte_su@firefly.alpha-lab.net) |
| 20:06.39 | Bile_One | delay can be a result of rtp and bandwidth |
| 20:06.43 | pjz | so where/how can I fix that? is that a phone thing or a sip.conf thing? |
| 20:06.48 | johnnyb | pjz: what kind of phones do you have? |
| 20:06.56 | ManxPower | pjz: Your problem is VERY unusual and so will be very hard to fix. |
| 20:07.06 | ManxPower | pjz: If you have a spare system available try using that. |
| 20:07.08 | pjz | johnnyb: Uniden UIP200 and an ipDialog SipTone II |
| 20:07.30 | johnnyb | pjz: do they have one or two ethernet ports? |
| 20:07.38 | ManxPower | evil motherboards can sometimes cause oddities that soundlike echo |
| 20:07.52 | pjz | johnnyb: two |
| 20:07.58 | Bile_One | ManxPower, I bet he has ppl doing heavy downloads even with 6 ppl using sip phones here, we can get problems without QoS for voice packets |
| 20:07.59 | johnnyb | are you using both of them? |
| 20:08.09 | ManxPower | Bile_One: That should not cause it. |
| 20:08.21 | ManxPower | it could cause many other problems, but not echo. |
| 20:08.22 | Bile_One | ManxPower, it does. I can testify |
| 20:08.29 | johnnyb | ManxPower: I'm guessing they are hubbed phones, not switched phones |
| 20:08.50 | johnnyb | Alternatively, he could be pre-feeding everything to a hub instead of directly to a switch. |
| 20:08.59 | pjz | everything goes to the same 100MB switch |
| 20:09.14 | pjz | I get sub-1ms pingtimes between phones and the pbx |
| 20:09.18 | johnnyb | pjz: are you using both ethernet ports on the phone? |
| 20:09.26 | Bile_One | pjz does your switch allow prioritizing packets? |
| 20:09.39 | ScaredyCat | does your switch run on cheese? |
| 20:09.42 | pjz | johnnyb: I've tried with and without the 'pc' net plugged in |
| 20:09.56 | pjz | Bile_One: nope, it's a dumb switch. |
| 20:10.13 | ManxPower | Echo is caused by analog 2-wire circuits. He doesn't have any of those anywhere. |
| 20:10.18 | pjz | Bile_One: but that's a good idea - I'll see if I can maybe move this stuff over to it's own switch. |
| 20:10.29 | pjz | ManxPower: yeah, we've established that what I've got is delay, not echo |
| 20:10.35 | *** join/#asterisk Anti_Mike (~antimike@48.129.8.67.cfl.res.rr.com) |
| 20:10.43 | johnnyb | pjz: what codec are you using? If going to a lower-bandwidth codec helps, then it's a network issue. If it hurts, then it's probably a machine issue |
| 20:10.46 | pjz | Bile_One: at least to test with. Then I can justify buying a QoS switch |
| 20:10.51 | pjz | johnnyb: ulaw/alaw |
| 20:10.58 | ManxPower | pjz: Any chance the mic on the phone is picking up the sound from the ear piece? |
| 20:11.04 | pjz | johnnyb: which is what the phones support |
| 20:11.12 | johnnyb | pjz: do they support anything else? |
| 20:11.14 | ManxPower | That's the most common form of "echo" where it's impossible to have echo. |
| 20:11.15 | pjz | ManxPower: mmm.. dunno. |
| 20:11.21 | pjz | johnnyb: what's g729? |
| 20:11.29 | Bile_One | pjz ManxPower has a good one there too. We used USB DSP mics and they were awful. |
| 20:11.34 | johnnyb | It's a codec, but I don't know if asterisk supports it |
| 20:11.51 | Micc | So what is this conference about? |
| 20:12.10 | johnnyb | Micc: we're helping pjz diagnose his delay problem. |
| 20:12.19 | Anti_Mike | Hello all |
| 20:12.24 | pjz | howdy :) |
| 20:12.36 | Micc | So am I muted? |
| 20:12.36 | syslod | Is there a virtual park app out there still? |
| 20:12.42 | Micc | how do I get unmuted? |
| 20:12.53 | Anti_Mike | this is my first time on IRC, and also my first time using Asterisk :( |
| 20:12.58 | Bile_One | Micc what are you talking about? |
| 20:13.13 | Micc | Rile_One, I'm on the conference. Its kinda cool. |
| 20:13.32 | Bile_One | Micc, what conference? |
| 20:13.34 | pjz | johnnyb: it looks liek the uniden only supports alaw/ulaw/g729 |
| 20:14.09 | pjz | I can increase the size of the jitter buffer |
| 20:14.15 | pjz | lemme try that |
| 20:14.38 | pjz | and I'll also try and test with the phones on a spare switch |
| 20:14.40 | pjz | thanks all |
| 20:15.16 | obsidian-studios | Anti_Mike: welcome to both, * is deep ;) take the blue pill |
| 20:15.22 | johnnyb | pjz: another thing to try is to load up the "mtr" package on the asterisk server, and then mtr back to the phone. It has all sorts of jitter and related statistics |
| 20:15.55 | pjz | johnnyb: how do I do that? |
| 20:16.01 | shido6 | asteirsk does suppor g729 |
| 20:16.03 | Bile_One | pjz try that too. That tools is invaluble |
| 20:16.17 | johnnyb | Download mtr, compile it, install it, then do "mtr XXX.XXX.XXX.XXX" to your phone. |
| 20:16.45 | Anti_Mike | I guess I should try to explain my situation, my senior project is to be running *, and I haven't the foggiest how to get everything implemented as my prof. wants |
| 20:16.57 | pjz | I love debian: 'apt-get install mtr' |
| 20:17.02 | obsidian-studios | Anti_Mike: you been to wiki |
| 20:17.12 | Anti_Mike | yes i have |
| 20:17.16 | laotan | apt-get is the shiznit |
| 20:17.18 | *** join/#asterisk Sift (~user@c-24-22-24-212.hsd1.or.comcast.net) |
| 20:17.18 | johnnyb | Anti_Mike, what do you want to do? |
| 20:17.23 | obsidian-studios | start small and add on from there |
| 20:18.05 | *** join/#asterisk samueltc (~samuel@196.203.26.86) |
| 20:18.16 | samueltc | hi |
| 20:18.17 | Anti_Mike | i also do not know linux very well |
| 20:18.32 | *** join/#asterisk fugitivo (~ajf@201.255.99.226) |
| 20:18.37 | johnnyb | pjz: You'll have to look in the help to figure out how to enable the jitter-based parameters. |
| 20:18.47 | johnnyb | Anti_Mike, that could be a big problem. |
| 20:19.08 | samueltc | anyone here can help about call progression/status detection on a TDM400 in tunisia? |
| 20:19.14 | Anti_Mike | johnnyb i can actually navigate and install things properly but getting the full functionality out of * is going to be a prolbme |
| 20:19.34 | obsidian-studios | what do you need out of * |
| 20:19.41 | Anti_Mike | johnnyb: we actually have 2 lines dialing each other |
| 20:19.44 | Sift | so is the sipura 3000 the best box to get? or will the 2100 work ok? |
| 20:20.14 | Anti_Mike | obsidian-studios: we need to have ACD, DID, Voicemail fowarding to email, agi scripts repeating the weather from weather.com |
| 20:20.16 | bsdfreak | is the spa-3000 fully compatable with asterisk? |
| 20:20.20 | icexx | anybody tried asterisk with iconnecthere and had positive results? |
| 20:20.42 | obsidian-studios | well voicemail forwarding to email is easy ;) everything else will require some time and work :) |
| 20:21.07 | los415 | and thats the truth |
| 20:21.10 | *** join/#asterisk Mike_TK (~Mike@bell.yes.net.ua) |
| 20:21.47 | Anti_Mike | obsidian-studios: also, we need to connect to some SIP connections via Libertel, have wireless SIP connectivity, and |
| 20:21.47 | samueltc | i'm trying to get my TDM400 work. I can dialout but the remote hangup is not detected |
| 20:21.49 | obsidian-studios | sip is not that bad, but I would not say easy either |
| 20:22.04 | obsidian-studios | sip is straight forward |
| 20:22.28 | Anti_Mike | obsidian-studios: indeed it isn't. setting up the phone extensions was rather easy. going into the extensions.conf, sip.conf and eh the other one.conf wasn't too bad |
| 20:23.09 | obsidian-studios | what is your problem then the other stuff, most of it I have not done yet, or had the need/want/demand so not much help from me there if you got all the basics in place |
| 20:23.14 | johnnyb | Anti_Mike, the rest is fairly easy. |
| 20:23.21 | johnnyb | Voice mail is simple. |
| 20:23.31 | Anti_Mike | johnnyb: i hope so |
| 20:23.31 | johnnyb | What's ACD? |
| 20:23.43 | zno | automatic call distribution |
| 20:23.44 | Anti_Mike | johnnyb: automated call distributing...like in call centers |
| 20:23.44 | johnnyb | Anti_Mike, no, it's REALLY simple. |
| 20:23.52 | durex | annonymous childrens dancing |
| 20:23.54 | obsidian-studios | if you put an I in there e I know :) |
| 20:23.55 | Sift | is asterisk compatible with suse? |
| 20:23.59 | Anti_Mike | durex: rofl |
| 20:23.59 | durex | :-) |
| 20:24.00 | johnnyb | I think you use groups.conf for that. |
| 20:24.11 | zno | Sift: with any linux |
| 20:24.18 | johnnyb | I haven't tried DID yet, but AGI scripts are very, very simple. |
| 20:24.21 | Sift | zno, thx |
| 20:24.21 | *** join/#asterisk kimc (~freenode@pcp09643046pcs.wbrmfd01.mi.comcast.net) |
| 20:24.26 | kimc | lo |
| 20:24.35 | Anti_Mike | johnnyb: i'm on my windows machine right now :) <power windows user> |
| 20:24.49 | johnnyb | However, for doing things like weather, you may want to enable Festival |
| 20:24.51 | obsidian-studios | Anti_Mike: till it crashes ;) |
| 20:25.05 | Anti_Mike | obsidian-studios does fine for me between crashes :D |
| 20:25.19 | johnnyb | Be sure to download and use Asterisk::AGI for your AGI scripts because it makes it a whole lot easier. |
| 20:25.19 | Anti_Mike | johnnyb: festival? |
| 20:25.24 | obsidian-studios | yep just save your work a hole lot like ever other key press |
| 20:25.31 | johnnyb | Anti_Mike, Festival is a text-to-speech program. |
| 20:25.32 | fugitivo | Anti_Mike: a TTS |
| 20:25.33 | obsidian-studios | I like festival but there is a huge need for more voices |
| 20:25.40 | Anti_Mike | tkx |
| 20:25.45 | fugitivo | Cepstral is $30 |
| 20:25.47 | obsidian-studios | some are easier for people to understand than other |
| 20:25.49 | durex | here I'm using festival with brazilian portuguese voice |
| 20:25.49 | fugitivo | and you have a lot of voices |
| 20:25.53 | obsidian-studios | really what is that |
| 20:25.58 | durex | and working perfectly (and robotic, of course) |
| 20:26.06 | Anti_Mike | the problem is that some of the things were already partially setup with this machine |
| 20:26.06 | obsidian-studios | I would love to go back to a TTS engine etc instead of making recordings |
| 20:26.18 | johnnyb | Anti_Mike, I found that installing it for use with asterisk is very difficult, so I built my own AGI script for doing festival instead of Asterisk's regular command. |
| 20:26.21 | durex | it's easiest than record every voice message u want |
| 20:27.03 | fugitivo | obsidian-studios: ceptral is a TTS, www.cepstral.com |
| 20:27.03 | obsidian-studios | cool I will look into any good? |
| 20:27.04 | fugitivo | obsidian-studios: you can hear examples online |
| 20:27.04 | kimc | can I get festival with 'hood voice :) |
| 20:27.04 | Anti_Mike | this is free labor for the guy that has us doing it :) |
| 20:27.13 | obsidian-studios | hell yeah I want taht |
| 20:27.30 | Anti_Mike | lol...i don't want to be envolved in the *nix world after i graduate :( |
| 20:27.39 | johnnyb | Anti_Mike, for a demonstration, I set up an asterisk office of 3 phones with Voicemail, an attendant, a meeting system, and a test AGI script in 1.5 hours. |
| 20:27.57 | obsidian-studios | what up mofo, who you trying to reach, with my extension being 420, and for some 187 :) |
| 20:28.00 | Anti_Mike | johnnyb now that is awesome johnnyb |
| 20:28.08 | fugitivo | johnnyb: that's a lot of time! |
| 20:28.16 | kimc | heh |
| 20:28.20 | johnnyb | fugitivo, I had to describe what I was doing for everyone |
| 20:28.29 | obsidian-studios | Anti_Mike: do not repeat anti *nix statements here if you want help :) |
| 20:28.40 | fugitivo | johnnyb: then it's ok :) |
| 20:28.46 | Anti_Mike | obsidian-studios: sorry :'(...us windows users don't really have manners |
| 20:29.11 | obsidian-studios | Anti_Mike: np but remember when you need to dial 911, it works party because of *nix |
| 20:29.19 | obsidian-studios | or when you call mom, or anyone else for a while |
| 20:29.22 | johnnyb | Anti_Mike, you would prefer to pay $10,000 for the same thing? |
| 20:29.37 | *** join/#asterisk chris_d (~chris@66.88.142.66.ptr.us.xo.net) |
| 20:29.38 | Anti_Mike | obsidian-studios: my only gripe is that the game support for *nix is kinda lacking and imma gamer |
| 20:29.43 | obsidian-studios | who do you think created Unix or the *nix foundation |
| 20:29.58 | Anti_Mike | johnnyb i dunno johnnyb |
| 20:30.01 | chris_d | Can somebody help me with dial out strings on Asterisk at Home? |
| 20:30.04 | durex | lol |
| 20:30.05 | obsidian-studios | hey man I play nwn and in spare time am bulding a cool module |
| 20:30.12 | fugitivo | me too |
| 20:30.17 | obsidian-studios | if it's a game worth playing or buying it gets ported to linux :) |
| 20:30.19 | bsdfreak | heh |
| 20:30.21 | fugitivo | nwn is the only game i need, and it works in linux |
| 20:30.23 | chris_d | Every time I try to dial, I get 484 Address Incomplete. |
| 20:30.41 | obsidian-studios | check out nwnx.org for my posts on stuff we are doing in linux |
| 20:30.59 | chris_d | I am trying to dial out through Broadvoice. |
| 20:30.59 | Anti_Mike | obsidian-studios: its about the gotta have it right now though..UT2K4 is my game of choice. never could get it runnin on the Linux box :( |
| 20:31.31 | johnnyb | Anti_Mike, you should probably go through all of the commands in this list, just so you get to know the full abilities of asterisk: |
| 20:31.35 | johnnyb | http://www.voip-info.org/wiki-Asterisk+-+documentation+of+application+commands |
| 20:31.49 | obsidian-studios | Anti_Mike: it's work but I got buddies who have machine that boot soley into certain games like nwn, running very minimal resources on bitchen hardware and it blows windows away |
| 20:32.06 | fugitivo | oh, It's UT2k4! |
| 20:32.16 | fugitivo | and it runs! |
| 20:32.19 | *** join/#asterisk nine76 (~t00r@cpe-69-135-184-24.woh.res.rr.com) |
| 20:32.22 | nine76 | hello all:) |
| 20:32.23 | obsidian-studios | Anti_Mike: just wait till you see what you can do on Linux with * and how reliable it will be |
| 20:32.40 | Anti_Mike | fugitivo: i've played it on others machines and its alright... |
| 20:32.56 | Anti_Mike | obsidian-studios: hopefully i can get it done with the help of my partners in 4 weeks |
| 20:32.58 | kimc | hmm 484 |
| 20:33.05 | obsidian-studios | cepstral tts is not to bad much better than festival voices |
| 20:33.25 | Anti_Mike | obsidian-studios: no built in tts prog? |
| 20:33.26 | obsidian-studios | Anti_Mike: should be enough time depending on how much you devote to it |
| 20:33.45 | Anti_Mike | obsidian-studios: well i have 2 jobs, and i have classes and this project too |
| 20:33.49 | obsidian-studios | *nix philosophy is small things that do their job :) |
| 20:34.06 | obsidian-studios | so some things are built in, most modules, even more functionality via external programs |
| 20:34.09 | Anti_Mike | i might just have to bring the box home and set it up on my separate monitor keyboard dooflickie :) |
| 20:35.01 | Anti_Mike | johnnyb: what should i do with those commands though...am i using the * CLI? |
| 20:36.52 | Anti_Mike | believe it or not ya'll i had problems registering with the server |
| 20:37.30 | sean | when setting my externip (on my LAN), can I set it dynamically, somehow? |
| 20:37.54 | nine76 | dhcp? |
| 20:37.59 | ManxPower | sean: not really. |
| 20:38.18 | sean | kind of DHCP |
| 20:38.23 | ManxPower | sean: you can set externip=your.host.name but that doesn't work very well if DNS ever goes down and I don't think it ever checks the address again. |
| 20:38.54 | sean | hmm.. I have DDNS set up, so my.host.name would work initially.. |
| 20:40.12 | Sift | will asterisk run on a PII 400? |
| 20:40.56 | Nugget | of course |
| 20:41.15 | *** join/#asterisk ace95 (~iceman-95@dsle108.ody.ca) |
| 20:41.19 | Nuxi | There have been rare reports of * laughing when you dial with hardware that is too old. :) |
| 20:41.26 | sean | I have asterisk on a static IP (192.168.144.10) on my LAN. My LAN has one public IP (assigned by my ISP). My asterisk server, however, receives all traffic to that IP. |
| 20:41.32 | tzanger | Nuxi: hahaha |
| 20:41.48 | laotan | Nuxi++ |
| 20:42.06 | Nugget | sean: why do you use NAT if you don't need to? |
| 20:42.11 | ace95 | i am trying to setup asterisk@home 0.8 with TDM400 card...does anyone have a working config?? |
| 20:42.40 | sean | Nugget: I do need to. I have 5 machines on my LAN. My * server happens to be one of those. |
| 20:42.42 | johnnyb | Anti_Mike: Those are for dialplan extensions |
| 20:42.46 | sean | (I only have 1 IP) |
| 20:42.49 | bsdfreak | Is the Sipura SPA-3000 fully supported by Asterisk? |
| 20:43.03 | Nugget | then your asterisk box isn't receiving all traffic? |
| 20:43.34 | *** join/#asterisk tld (~tld@80.203.70.227) |
| 20:43.41 | sean | Nugget: I have my firewall redirecting all traffic to * (DMZ host for my LAN) |
| 20:43.58 | tld | Anyone know of any good papers or resources on setting up number-plans? |
| 20:44.05 | critch | what are the various revs of the TE410P card, and any known issues between them |
| 20:44.11 | Nugget | then the other four machines don't have internet connectivity? |
| 20:44.21 | sean | Nugget: of course they do. |
| 20:44.28 | Nugget | then it isn't "all traffic", right? |
| 20:44.41 | sean | when a request comes in on (e.g.) port 5060 my firewall directs it to my * box |
| 20:44.53 | Nugget | that's not all traffic. that's one port. |
| 20:45.25 | sean | Nugget: I have the same redirect set up for all ports |
| 20:45.39 | Nugget | then how do your other machines receive traffic? |
| 20:46.07 | sean | Nugget: what I'm doing isn't rocket science. It's a simple config in any Linksys or DLink router. |
| 20:46.14 | sean | do you really not understand? |
| 20:46.21 | Nuxi | I'm trying to get dialup over voip working. When/If it works, will voip work over duovoip? |
| 20:46.25 | Nugget | but what you described isn't what you're doing, apparently. you've said two conflicting things. |
| 20:46.32 | sean | *sigh* |
| 20:46.41 | Nugget | I'm trying to figure out which of the things you've said is accurate so I can understand what you want to ask |
| 20:47.10 | obsidian-studios | gotta run catch you all later |
| 20:47.12 | sean | sorry if I've explained myself poorly. My asterisk box is my DMZ host on my LAN. Traffic coming to my public IP goes to * |
| 20:47.18 | *** part/#asterisk obsidian-studios (~obsidian-@c-66-177-188-197.hsd1.fl.comcast.net) |
| 20:47.19 | Nugget | either the asterisk box receives all the traffic for your public ip or it doesn't. it can't be both ways |
| 20:47.49 | ace95 | i am trying to setup asterisk@home 0.8 with TDM400 card...does anyone have a working config?? |
| 20:47.54 | Nugget | I'd suspect a static nat mapping, but you've already said you only have one public ip. |
| 20:48.00 | Nugget | so that's probably not what you're doing |
| 20:48.01 | sean | alright. I see we're being semantic. All incoming connects go to my * box |
| 20:48.05 | sean | how's that? |
| 20:48.21 | Nugget | so how do the other four boxes receive traffic? does asterisk forward on to them? |
| 20:48.28 | sean | No. My firewall does it. |
| 20:48.39 | Nugget | then the firewall receives all the traffic? |
| 20:48.45 | sean | My other boxes are outgoing only. |
| 20:49.00 | sean | yes, that's correct, but it seamlessly transfers connects to my asterisk box. |
| 20:49.02 | Nugget | udp traffic isn't "connects", so I don't know what you mean by that. |
| 20:49.23 | *** join/#asterisk Optic (dfraser@H31.C18.B96.tor.eicat.ca) |
| 20:49.25 | Optic | hey |
| 20:49.27 | sean | alright. apparently I'm explaining myself poorly. |
| 20:49.32 | sean | let's start from scratch. |
| 20:49.34 | Optic | what's a good basic business phone these days? |
| 20:49.42 | Nugget | it sounds to me like you have nat, and the asterisk box is just like the other four. |
| 20:49.43 | Optic | polycom IP300 work okay? |
| 20:49.55 | ManxPower | ~docs |
| 20:49.56 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 20:49.57 | ManxPower | ~mailinglist |
| 20:49.58 | jbot | rumour has it, mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
| 20:50.03 | Nugget | and you're sending unfamiliar tcp to it, which isn't particurlarly useful |
| 20:50.54 | sean | Yes, I have NAT. My * box IS like the other 4, with the exception that any traffic that comes IN to my public IP (ie, TCP connect, or a UDP destination of my public IP) goes to my asterisk box by way of my firewall. |
| 20:51.15 | Nugget | so your other four cannot pass udp through nat? |
| 20:51.21 | sean | correct |
| 20:51.24 | Nugget | ok. |
| 20:51.29 | sean | I don't care about them. Just my * box |
| 20:51.32 | Nugget | *nod* |
| 20:51.45 | sean | so, we're on the same page? (-: |
| 20:51.50 | Nugget | I think so :) |
| 20:52.01 | sean | alright. Sorry for getting frustrated. |
| 20:52.35 | sean | My public IP is assigned by my ISP, dynamically (it doesn't change every hour.. maybe every couple days -- basically whenever my ISP coughs) |
| 20:52.50 | sean | my asterisk box, though, has a static, private IP |
| 20:52.55 | sean | make sense? |
| 20:53.05 | Nugget | yup |
| 20:53.12 | sean | localnet = 192.168.144.0/255.255.255.0 |
| 20:53.18 | MeTaBSD | I have a question |
| 20:53.19 | sean | but to what do I set externip ? |
| 20:53.55 | Nugget | you're going to have to find a way to update externip when your public ip gets pulled out from under you |
| 20:54.19 | sean | ok. |
| 20:54.30 | shido6 | ask MeTaBSD |
| 20:54.37 | sean | does a "sip reload" do that? |
| 20:54.46 | ace95 | u can use dyndns.org |
| 20:54.47 | Nugget | I'm unsure of that. |
| 20:54.57 | sean | (provided I have a dynamic hostname in my sip.conf?) |
| 20:55.01 | sean | hmm.. alright |
| 20:55.01 | Nugget | I'd expect a full 'reload' to do it. |
| 20:55.04 | MeTaBSD | in asterisk |
| 20:55.16 | MeTaBSD | i try to use database put blacklist number 1 |
| 20:55.23 | ace95 | it updates your dynamic ip to a domain name |
| 20:55.26 | sean | fair enough.. will a full "reload" stop all current connections? |
| 20:55.31 | MeTaBSD | but i have much number what is the limite of this database |
| 20:55.35 | ace95 | and updates when your ip changes |
| 20:55.52 | sean | ace95: I have my own DDNS set up, yes. |
| 20:56.05 | Nugget | sean: reload is safe to do |
| 20:56.06 | sean | ace95: the problem is telling * when to refresh the IP it has resolved. |
| 20:56.17 | sean | Nugget: ok. Thanks very much. |
| 20:56.34 | ace95 | it should care...it only knows it ip and there should be a 1 to 1 nat on your firewall |
| 20:56.45 | Nugget | sorry if I was obtuse. |
| 20:56.53 | sean | Nugget: I'm sure it was my fault. (-: |
| 20:57.08 | ace95 | i am trying to setup asterisk@home 0.8 with TDM400 card...does anyone have a working config?? |
| 20:57.51 | ace95 | it is a type TDM11B card..only 2 modules...1fxo 1 fxs |
| 20:57.57 | *** join/#asterisk rg1 (~rg1@mail.airlinksystems.com) |
| 20:58.17 | ace95 | i have tried everything and it will not setup |
| 20:58.40 | pjz | sean: you might want to look into STUN |
| 20:58.46 | rg1 | what do we have to do on our gateway/firewall box to connect an internal asterisk machine to IAXTel? |
| 20:59.06 | pjz | sean: I don't know much about it other than it's a way to tell your outbound where you are |
| 20:59.32 | sean | pjz: I took a quick look at it. I'm not sure my SIP proxy supports it (my DID). |
| 20:59.43 | sean | also, latency is a problem; I want to minimize "hops" |
| 21:00.27 | rg1 | sean, what is your phone-co interface? |
| 21:00.51 | rg1 | anyone: what do we have to do on our gateway/firewall box to connect an internal asterisk machine to IAXTel? |
| 21:01.10 | sean | rg1: I'm very new to this. I don't understand the question. I have a DID that I connect to via SIP Proxy |
| 21:01.14 | ace95 | i am trying to setup asterisk@home 0.8 with TDM400 card...does anyone have a working config?? |
| 21:01.21 | MeTaBSD | Anyone can help me ? |
| 21:01.27 | ace95 | anyone?? |
| 21:02.17 | rg1 | sean - do you have a line from the phone company hooked into a card in asterisk? |
| 21:02.28 | *** join/#asterisk Grooby (~Grooby@66.160.105.186) |
| 21:02.58 | sean | rg1: nope. it's all via public internet |
| 21:03.13 | sean | DID -> SIP PROXY -> Softphone/Asterisk |
| 21:04.29 | ace95 | i am trying to setup asterisk@home 0.8 with TDM400 card...does anyone have a working config?? |
| 21:04.39 | ace95 | i could really use some help |
| 21:06.28 | *** join/#asterisk porche (~a@dsl81-215-20485.adsl.ttnet.net.tr) |
| 21:06.29 | porche | hi |
| 21:06.44 | porche | chan_oss.c: Unable to open /dev/dsp: No such devic |
| 21:06.54 | porche | what's the affect of chan_oss? |
| 21:07.10 | critch | oss = Open Sound System, sound cards |
| 21:07.29 | porche | yep, but on *, where does it affect on operation |
| 21:07.38 | porche | I dont use any sound card op'n |
| 21:07.43 | critch | no console, nothing really a problem |
| 21:07.56 | ace95 | does anyone have a idea how to setup *@home with tdm400 card?? |
| 21:07.57 | porche | but i do voice record, and tts |
| 21:08.14 | critch | just put a noload in the modules conf to get rid of the line |
| 21:08.21 | porche | weird |
| 21:08.23 | porche | cuz |
| 21:08.28 | porche | in the new cvs |
| 21:08.40 | porche | record ing sometimes crashes the whole * |
| 21:08.41 | Nivex | porche: chan_oss and chan_alsa give you the Console/dsp destination and the ability to use the "dial" command at the console. mostly useless except for niche testing cases. |
| 21:09.13 | porche | then alsolutely no_load, I dont need them |
| 21:09.40 | porche | I was testing the recording, suddenly realized it sometimes crashes or cannot detect silence |
| 21:09.54 | porche | tnx |
| 21:10.01 | ace95 | anybody? am i asking something i shouldnt?? i am really frustrated with this product as i have been wqorking on it for several days with no luck |
| 21:10.03 | critch | it is a good idea to noload any module, especially channels, that you don't use |
| 21:10.17 | porche | hm |
| 21:10.17 | critch | ace95: it isn't a product |
| 21:10.39 | ace95 | ok |
| 21:10.53 | porche | critch is there a list of the modules, that are not must for the op'n, may be some link? |
| 21:11.00 | ace95 | can u help me with my setup |
| 21:11.03 | shido6 | check your message ace95 |
| 21:11.04 | critch | ace95: the closest thing to a product is the TDM card |
| 21:11.46 | critch | porche: not that I know of, I just do a show modules when running and see what I want to get rid of. |
| 21:12.06 | critch | ace95: I do not use *@home, have no clue what problems you may encounter |
| 21:12.39 | *** join/#asterisk visik7 (~ciao@visik7.user) |
| 21:12.42 | *** join/#asterisk nextime (~nextime@213-140-6-96.fastres.net) |
| 21:12.55 | *** join/#asterisk P-Chan (~jpfingstm@68.142.66.200) |
| 21:13.01 | P-Chan | Hello. |
| 21:13.34 | *** join/#asterisk djMax (~chatzilla@artsalliancelabs.com) |
| 21:13.38 | P-Chan | If the error: WARNING[2523]: Unable to read password happens when someone tries to log into voicemail from a phone, what does that mean? |
| 21:13.50 | sean | Nugget: thank you; asterisk is now proxying my DID to my cellphone (-: |
| 21:13.55 | P-Chan | They can log in with comedian mail w/o any problems, just not from the phone. |
| 21:13.58 | porche | cool good idea |
| 21:13.59 | Nugget | yay |
| 21:14.23 | critch | P-Chan: how do you log in to voice mail witout a phone? |
| 21:14.41 | P-Chan | critch: Comedian Mail - part of AMP |
| 21:14.52 | critch | P-Chan: so a web page? |
| 21:14.57 | P-Chan | critch: yes |
| 21:15.11 | critch | Comedian Mail is the name of the voicemail app in asterisk |
| 21:15.21 | porche | hell |
| 21:15.25 | Nugget | there's also a crude but functional vmail.cgi as part of asterisk. |
| 21:15.27 | porche | record file .... s=1 |
| 21:15.30 | Nugget | it's some perl thing. works fine |
| 21:15.32 | porche | crashes the * |
| 21:15.37 | critch | P-Chan: could it be a non numeric password? |
| 21:15.45 | rg1 | well, just got an answer from digium: |
| 21:15.46 | rg1 | <PROTECTED> |
| 21:15.46 | rg1 | <PROTECTED> |
| 21:15.55 | P-Chan | Critch: the voicemail.conf line looks like this: |
| 21:15.56 | rg1 | straight from digium sales department |
| 21:16.11 | ionix | of course analog doesn't support DID |
| 21:16.24 | ionix | there is no signaling on a POTS |
| 21:16.39 | P-Chan | 19 => 19,Firstname Lastname, email@domain.com,,attach=no|... |
| 21:16.39 | critch | ionix: yes there is, it is called E&M wink |
| 21:16.39 | rg1 | ionix - well you can order it from ma bell |
| 21:16.54 | ionix | E&M is to send the callnig party |
| 21:17.00 | P-Chan | So the password is 19 for user 19 |
| 21:17.11 | critch | ionix: I used it for DID when I had it. |
| 21:17.25 | ionix | but you cannot have multiple lines on 1 analog pair with many DIDs with strictly E&M |
| 21:17.33 | rg1 | critch - did you use it on Asterisk system? |
| 21:17.50 | critch | rg1: I did but via channelized T1 |
| 21:17.57 | P-Chan | critch: I've tried the password 1234 just to see if 19 is too short - stil no go. |
| 21:18.08 | critch | P-Chan: maybe remove the space |
| 21:18.15 | *** join/#asterisk pr0f1t (~pr0f1t@toronto-HSE-ppp4021895.sympatico.ca) |
| 21:18.17 | rg1 | critch - how did you order that |
| 21:18.20 | CoaxD | ionix: There's that stupid dual-pots-multiplexing thing, btu thats about it |
| 21:18.29 | P-Chan | critch: Ok, I'll try that. |
| 21:18.36 | critch | rg1: I had a full T1 with E&M wink. |
| 21:19.00 | critch | But channelized T1 is just a digital form of the analog component |
| 21:19.27 | CoaxD | thing with CT1 is that they're pretty dumb. You can do a *lot* more with PRI. |
| 21:19.35 | CoaxD | I have a couple CT1s for modem banks. |
| 21:19.50 | critch | rg1: Have you looked into using a Adit or Adtran to connect an anolog DID line to asterisk? |
| 21:20.15 | rg1 | can i do that? |
| 21:20.16 | critch | CoaxD: I agree that CT1 is pretty dumb. Thats why I have PRI now |
| 21:20.31 | critch | rg1: I think so, look into the specs for the channel banks |
| 21:21.05 | critch | rg1: how many lines are you wanting to bring into asterisk? |
| 21:21.12 | rg1 | just one |
| 21:21.23 | critch | one line with DID? |
| 21:21.28 | rg1 | yeah |
| 21:21.28 | ManxPower | rg1: you need FXO ports on the Channel bank if you want to connect PSTN analog to the channel bank |
| 21:21.32 | rg1 | just to program and check it out |
| 21:21.50 | critch | rg1: that is a waste of your time and money |
| 21:21.52 | porche | absolutely, record file with s=1 crashes |
| 21:22.01 | djMax | odd behavior: I conference a second call on the polycom, put them back together, and I can only hear the second call. Then I put the conference on hold, take it off hold, and everybody is there. Is that * or poly? |
| 21:22.24 | ManxPower | djMax: Do you have each "line" on the polycom set up as a different SIP user? |
| 21:22.46 | samueltc | hi, how should I get a TDM400P working in a never used country? I got he dialtone but nothing is detected (hangup, ring, ...) |
| 21:22.51 | djMax | I don't *think* so but I'll go check. I mean, they're defnitely the same extension |
| 21:22.51 | critch | rg1: to really use DID, you need to be pulling many lines in anyways |
| 21:23.01 | shido6 | what country samueltc ? |
| 21:23.04 | samueltc | tunisia |
| 21:23.10 | shido6 | you can always do trial an error |
| 21:23.13 | shido6 | there arent that many |
| 21:23.33 | ManxPower | djflux: In my experience you can't conference on the Polycom unless each "line" is a seperate SIP user. |
| 21:23.52 | samueltc | shido: you are talking about tonezone? |
| 21:24.05 | djMax | fascinating. So in case anybody says it breaks, just tell em to put the conf on hold and take them back. :) |
| 21:24.08 | *** join/#asterisk ipso (~ipso@d207-81-249-35.bchsia.telus.net) |
| 21:24.14 | *** join/#asterisk r0d3nt (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 21:24.23 | djMax | So does that mean all my extension plans would now have to dial the first and then second user? |
| 21:25.06 | ManxPower | djflux: yup |
| 21:25.18 | rg1 | critch - are you saying that I can take my analog trunk line with DID, put it into an Adit or Adtran and then connect THAT into the asterisk? |
| 21:25.25 | ManxPower | But you get MUCH more control for the extra complexity. |
| 21:25.29 | ipso | I recently upgraded a couple Asterisk servers to CVS head, and now when I run iax2 show channels during a call, it shows the lag, jitter, and jitter buffer as all 0's. Is this normal? |
| 21:25.53 | samueltc | when i'm calling someone should I see 'Ring' in the CLI? i've already setup a TDM in canada but i can't remember if i had set debugging on or something... |
| 21:26.59 | ManxPower | rg1: you need a Digium T-1 card to go from the channel bank to Asterisk |
| 21:28.11 | mmlj4 | any issues with running on AMD-64? |
| 21:28.37 | djMax | guess I can put it all in the stdexten macro anyway. One other question: any gotchas when moving from a TDM04 to a TE110P? New drivers or anything? |
| 21:28.49 | rg1 | ManxPower - so if I get a Digium T1 card and run my trunk into the adtran box and that into the Digium T1 card, that should do it? |
| 21:30.35 | *** join/#asterisk ariel_ (Ariel@148.sub-70-213-80.myvzw.com) |
| 21:37.51 | porche | hate this |
| 21:38.03 | porche | can someone suggest me an iax2 provider? |
| 21:38.08 | kiokorobert | how do i connect a router with fxs cards to asterisk? |
| 21:38.09 | *** join/#asterisk bajanman (~william@cp66-203-194-32.cp.telus.net) |
| 21:38.47 | ManxPower | rg1: Yes, but it seems a little expensive and complicated. |
| 21:39.02 | ManxPower | Why not just get a CT1 or PRI from your provider and go directly into the Digium T-1 card. |
| 21:40.39 | pjz | how can a switched fabric suffer from bandiwdth problems? |
| 21:40.47 | pjz | ie. why does a switch need QoS ? |
| 21:41.31 | Manipura | how do I add two IP's to host= in sip.conf? |
| 21:42.57 | ManxPower | Manipura: for inbound or outbound calling? |
| 21:43.13 | ManxPower | host= is only use for calls going OUT of Asterisk. |
| 21:43.34 | ManxPower | permit=/deny= is used to limit what IPs can connect to Asterisk |
| 21:44.21 | Manipura | Its for our DID ManxPower |
| 21:44.30 | fugitivo | anyone else is having problems with stanaphone? |
| 21:44.30 | sean | is there a way to make an extension Dial() two numbers at once? |
| 21:44.34 | fugitivo | sip.stanaphone.com died? |
| 21:44.37 | sean | (make that "an easy way") |
| 21:44.46 | zip | porche, try nufone |
| 21:44.46 | Nivex | sean: yeah, seperate them with the | character |
| 21:44.57 | pjz | sean: Dial(channel1&channel2) |
| 21:45.05 | pjz | Nivex: I thought | was for delay |
| 21:45.08 | pjz | Nivex: er, not delay, but timeout |
| 21:45.16 | ManxPower | Manipura: then host= will do nothing for you. See permit/deny |
| 21:45.19 | Nivex | pjz: yeah, I was wrong. |
| 21:45.20 | sean | Nivex: cool.. does the second continue ringing? or does the channel just get assigned to the first to pick up? |
| 21:45.22 | nine76 | fugitivo: I'm registered |
| 21:45.28 | Nivex | sean: what pjz said |
| 21:45.35 | Manipura | okay, thanks manx |
| 21:45.38 | pjz | sean: I think it gets assigned to whichever picks up first |
| 21:45.39 | fugitivo | nine76: hmm, seems a problem with my dns |
| 21:45.46 | sean | pjz: awesome thanks |
| 21:45.49 | sean | Nivex: too (-: |
| 21:46.01 | nine76 | fugitivo: One of my colo'd boxes is also having dns problems. |
| 21:46.23 | fugitivo | nine76: oh, i hope it's not some kind of dns worm again |
| 21:47.13 | porche | zip got problems |
| 21:47.16 | porche | with nufone |
| 21:47.18 | nine76 | customer support is not responding... if anyone is pleased with there dedicated server provider pls drop me a line with there name:) |
| 21:47.25 | porche | no dtmf at all |
| 21:47.59 | critch | nine76: I prefer real colo's, My company owns all our machines. |
| 21:47.59 | porche | quality is good bu i need dtmf, and no answer from support |
| 21:48.02 | Manipura | Manx Permit Didn't work... It will only work if I put the IP in host= |
| 21:48.34 | porche | any alternative to nufone? |
| 21:48.37 | nine76 | what colo provider would you recommend? |
| 21:48.38 | critch | nine76: Inflow is a great company. Granted they where bought by Sungard and now Sungard was bought and is supposed to be split |
| 21:49.01 | critch | nine76: all that happened in about 4 months |
| 21:49.42 | critch | time to bounce out again as I reboot this machine |
| 21:49.57 | nine76 | thx for the info |
| 21:50.03 | johnnyb | pjz: a switch can need QoS when multiple sources are passing packets to the same destination -- it has to know which to pass first. |
| 21:51.05 | johnnyb | pjz: remember that the problem with IP phones is not _bandwidth_, but _latency_ |
| 21:51.57 | tld | Anyone know the level of fun stuff you can do with the Polycom phones? I mean, do they offer something to Ciscos XML-stuff? |
| 21:53.44 | niZon | tld: thats what i'd like to know |
| 21:53.44 | *** join/#asterisk file (~file@mctn1-3974.nb.aliant.net) |
| 21:54.28 | tld | niZon: First of all if it has anything I can make a custom interface against, and secondly if there's anywhere I can read up on it's capabilities, so I don't bother you all with all my questions. ;) |
| 21:55.34 | niZon | hm |
| 21:55.43 | zip | porche, probably your phone settings |
| 21:55.48 | zip | how are you doing dtmf? |
| 21:55.58 | bajanman | can someone ask a simple noob question? |
| 21:56.17 | bajanman | I've been following http://sprackett.com/asterisk/, and I would like to ask a few questions.... |
| 21:56.29 | niZon | bajanman just ask |
| 21:56.41 | niZon | we don't need your life's story :P |
| 21:56.45 | bajanman | well, I don't want to put forth text without asking....lol |
| 21:56.52 | bajanman | in extensions.conf: |
| 21:56.52 | bajanman | [outbound-local] |
| 21:56.52 | bajanman | exten => _NXXXXXX,1,Dial(SIP/${EXTEN}@mt-1204) |
| 21:56.53 | bajanman | [from-pstn] |
| 21:56.53 | bajanman | include => ext-local |
| 21:56.53 | bajanman | exten => 8000,1,Macro(fixcid) |
| 21:56.55 | bajanman | exten => i,1,Playback(pbx-invalid) |
| 21:56.58 | bajanman | exten => i,2,Hangup |
| 21:56.59 | bajanman | in sip.conf: |
| 21:57.01 | bajanman | [mt-1204] |
| 21:57.03 | bajanman | type=peer |
| 21:57.05 | bajanman | host=172.20.16.7 |
| 21:57.07 | bajanman | context=from-pstn |
| 21:57.10 | bajanman | my question is: |
| 21:57.33 | bajanman | if I want to do an outside call, with my sip phone, the outbould-local goes to the sip.conf for the mt-1204? right? |
| 21:57.44 | niZon | woah |
| 21:57.46 | shido6 | that my friend |
| 21:57.46 | niZon | pastbin |
| 21:57.49 | shido6 | is what pastebin.ca is for |
| 21:57.50 | niZon | +e |
| 21:57.56 | shido6 | http://pastebin.ca |
| 21:57.57 | fugitivo | lol |
| 21:58.07 | Nuxi | http://pastebin.ca |
| 21:58.07 | shido6 | and give us the URL it gives you when you paste it there :) |
| 21:58.20 | bajanman | ahhhh |
| 21:58.22 | bajanman | really? |
| 21:58.33 | ManxPower | bajanman: flooding the channel can get you banned. |
| 21:58.33 | bajanman | ok let me go check: thanks: tha's why I wanted to ask first... |
| 21:58.38 | rg1 | critch - you still online here? |
| 21:58.49 | bajanman | manxpower: that's why I wanted to ask first.... |
| 21:58.50 | bajanman | ok |
| 22:00.36 | bajanman | ok. sorry for the little flood |
| 22:00.45 | bajanman | the question I have: |
| 22:00.46 | bajanman | http://pastebin.ca/10533 |
| 22:00.53 | bajanman | I'm trying to figure out the flow.... |
| 22:01.51 | bajanman | and I THINK, that if I make a call via sip, it will goto [outbound-local], then it will goto the ext = > @mt-1204), which sends it to sip.conf right? |
| 22:02.43 | ManxPower | bajanman: any call you make from the device will go to the context in extensions.conf that is listed as context=whatever in the the section of sip.conf for that device. |
| 22:02.56 | bajanman | hmmmm |
| 22:02.57 | shido6 | bajanman, where is fixcid? |
| 22:03.02 | bajanman | fixcid? |
| 22:03.20 | bajanman | don't know, I got this from: http://sprackett.com/asterisk/ |
| 22:03.25 | shido6 | lord |
| 22:03.26 | bajanman | so I'm just cut and paste a few parts |
| 22:03.32 | bajanman | lord? |
| 22:03.33 | shido6 | build your own bajanman :) |
| 22:03.39 | bajanman | I'm trying!!! |
| 22:03.43 | shido6 | OK!!! |
| 22:03.45 | shido6 | :) |
| 22:03.46 | bajanman | lol |
| 22:04.01 | shido6 | so what ya doin here? |
| 22:04.03 | bajanman | its hard.. and I'm trying to figure this out, and haven't gotten then "click" yet on it |
| 22:04.18 | bajanman | hoping some ppl can be nice and answer some vague noob questions? |
| 22:04.25 | shido6 | peers dont have contexts |
| 22:04.32 | bajanman | I can't find a simple... dummy's guide to asterisk yet.... |
| 22:04.36 | bajanman | lol |
| 22:04.39 | Lee__ | that's cause it's not for dummies |
| 22:04.50 | shido6 | they are all 85 - 125/hr ( dummy guides) |
| 22:05.10 | Lee__ | true, you can find those |
| 22:05.17 | bajanman | anyways. |
| 22:05.19 | shido6 | I'll build one and shtick it on www.nufone.net |
| 22:05.40 | bajanman | it confuses me shido, that you say peers don't have contexts... |
| 22:05.50 | shido6 | look at your sip.conf on the pastebin |
| 22:05.54 | shido6 | you have a context tere |
| 22:05.56 | shido6 | in the peer |
| 22:05.59 | shido6 | you dont need that |
| 22:06.04 | shido6 | if it needs a context |
| 22:06.06 | shido6 | it isnt a peer |
| 22:06.07 | shido6 | its a user |
| 22:06.40 | shido6 | you want to talk to me? then damnit you're a USER to me |
| 22:06.48 | shido6 | if I wanna talk to YOU then you're a PEER to me |
| 22:06.49 | bajanman | yes right. I get that |
| 22:06.51 | file | actually, that's not always true |
| 22:06.53 | bajanman | ahhhh |
| 22:06.53 | bajanman | ok |
| 22:07.04 | shido6 | hush file :) |
| 22:07.17 | bajanman | so, I make a call from my computer: I go to sip.conf first: cause I have to auth * hass to know who I am first. |
| 22:07.30 | bajanman | hence I see: [01] |
| 22:07.30 | bajanman | type=friend |
| 22:07.34 | shido6 | lord |
| 22:07.39 | bajanman | what? |
| 22:07.48 | bajanman | is that a flood? |
| 22:07.58 | shido6 | no friend |
| 22:08.06 | shido6 | use users and peers, please :) |
| 22:08.13 | bajanman | then why did this guy have "friend"?? |
| 22:08.17 | bajanman | is he doing it wrong? |
| 22:08.17 | ManxPower | bajanman: Shido doesn't like type=friend. He's a bit anal sometimes. |
| 22:08.22 | bajanman | oh |
| 22:08.33 | bajanman | lol |
| 22:08.41 | shido6 | not wrong, just causing himself grief |
| 22:08.47 | ManxPower | bajanman: In MY opinion, type=friend is fine for a PHONE device, but NEVER for a SIP provider or SIP gateway. |
| 22:08.50 | shido6 | or users that want to follow his idea, grief |
| 22:09.20 | bajanman | shido: then can you please suggest a better simple example for someone to learn? |
| 22:09.34 | bajanman | you did mention you're gonna build a simple one on a website??? |
| 22:09.35 | shido6 | sure... |
| 22:09.40 | bajanman | I'll give you $20.... |
| 22:09.41 | bajanman | lol |
| 22:10.10 | shido6 | sure... |
| 22:10.16 | shido6 | billing@nufone.net (paypal) |
| 22:10.31 | shido6 | simple sip extension? |
| 22:10.36 | shido6 | w/ vmail? |
| 22:10.39 | bajanman | yes |
| 22:10.44 | shido6 | gimme 2 minutes |
| 22:10.49 | bajanman | with a sip inbound and outbound |
| 22:11.23 | *** join/#asterisk crypt^_^ (~cryptkeep@65.172.117.3) |
| 22:11.29 | bajanman | ok, let me go and pay...lol |
| 22:12.02 | bajanman | OH: shido, where ya from? US or Canada? |
| 22:13.46 | *** part/#asterisk FarrisG (~jrush@h-68-164-19-170.dllatx37.covad.net) |
| 22:14.19 | *** part/#asterisk Grooby (~Grooby@66.160.105.186) |
| 22:14.44 | bajanman | I'm ready, just let me know, if you're in us or canada... (I'm at paypal) |
| 22:16.26 | ManxPower | bajanman: Shido is in USA |
| 22:16.40 | ManxPower | or at least billing@nufone.net IS in the USA |
| 22:17.00 | bajanman | ahh ok |
| 22:17.06 | bajanman | I'll do it now |
| 22:19.35 | *** join/#asterisk Gunnar (~gunnar@34.80-203-209.nextgentel.com) |
| 22:19.39 | Trickyphillips | I need some recordings done for my PBX. Would anyone with a sexy voice be willing to record a welcome message and such, for me? :P |
| 22:19.43 | shido6 | http://pastebin.ca/10535 |
| 22:19.54 | ManxPower | Trickyphillips: Why not use thevoice.digium.com? |
| 22:19.58 | shido6 | my fingers hurt now |
| 22:20.08 | Trickyphillips | Manx: I'll take a look at that. :) |
| 22:20.16 | bajanman | shido: I'm waiting for it to go through.... |
| 22:20.44 | shido6 | waiting for what to go through? |
| 22:21.31 | bajanman | the payment: it just went... |
| 22:21.37 | bajanman | let me know if you got it |
| 22:21.42 | shido6 | where'd u send it |
| 22:21.44 | shido6 | billing? |
| 22:21.53 | bajanman | yess |
| 22:22.04 | bajanman | form carman osbourne: girlfriends credit card....lol |
| 22:22.09 | bajanman | I love her soooo much |
| 22:22.16 | *** join/#asterisk obsidian-studios (~obsidian-@c-66-177-188-197.hsd1.fl.comcast.net) |
| 22:22.31 | *** part/#asterisk jwitte (~jwitte_su@firefly.alpha-lab.net) |
| 22:23.17 | obsidian-studios | was looking around the wiki, I have seen people use variables before to control the amount of times something is played etc. Does anyone have a link to an example or can post a small one. I just need a single var named what ever that I can increment to a specified number then terminate the call or etc |
| 22:25.13 | ManxPower | Random Dialplan Example: http://pastebin.ca/10536 |
| 22:25.34 | ManxPower | obsidian-studios: you mean like the pastebin I just pasted? |
| 22:25.39 | obsidian-studios | yep ty |
| 22:26.13 | shido6 | you know you're surrounded by astmasters when they spit out "RANDOM DIALPLAN EXAMPLES" |
| 22:26.56 | Chotaire | aha |
| 22:27.00 | obsidian-studios | cool I had guessed SetVar(COUNT=${COUNT}+1) |
| 22:27.22 | *** join/#asterisk moy (~kvirc@201.129.134.130) |
| 22:27.41 | obsidian-studios | way more than I needed but allot better to ty very very much :_ |
| 22:27.43 | obsidian-studios | :) |
| 22:27.58 | ManxPower | THAT won't work. you need spaces around the + |
| 22:28.03 | *** join/#asterisk bjohnson (~bjohnson@66.11.188.76) |
| 22:28.07 | obsidian-studios | ok |
| 22:28.17 | obsidian-studios | so it cares about them good to know |
| 22:28.21 | Chotaire | GotoIf($[${COUNT} = 3]?exit,1) - hm... |
| 22:28.21 | ManxPower | They may have fixed that. I don't know. |
| 22:28.24 | obsidian-studios | I am not a big fan of spaces |
| 22:28.49 | Chotaire | count = 3!?!? |
| 22:29.00 | ManxPower | Chotaire: what's wrong with that? |
| 22:29.05 | Chotaire | can you explain that to me? |
| 22:29.20 | Chotaire | like explain what that line exactly does? |
| 22:29.38 | ManxPower | If COUNT = 3 then go to priority 1 of extension "exit" |
| 22:29.43 | ManxPower | extensions don't have to be numbers |
| 22:29.52 | Chotaire | ok but why 3? |
| 22:30.08 | Chotaire | like count 3 diggits`? |
| 22:30.13 | ManxPower | Chotaire: Because I only give them three chances to dial the extension that they want. |
| 22:30.21 | ManxPower | Chotaire: read the example carefully and put down the beer. |
| 22:30.22 | Chotaire | ah, attempts. |
| 22:30.22 | Chotaire | ok. |
| 22:30.52 | Chotaire | it's actually a whiskey-coke ;) |
| 22:31.10 | Manipura | Mmmmm..... Whiskey.... |
| 22:31.18 | Chotaire | 12 yrs old canadian whiskey. |
| 22:31.26 | Manipura | Darn and I just spent all my money on beer |
| 22:31.41 | Manipura | Yeah! Nothing like canadian whiskey |
| 22:31.53 | Chotaire | a friend from toronto brought it when we met in nyc last week. |
| 22:32.21 | Manipura | So I take it your not canadian? |
| 22:32.27 | Chotaire | german ;) |
| 22:32.31 | *** join/#asterisk ctooley (~ctooley@pc51.utati.net) |
| 22:32.35 | Manipura | Oh! Cool... |
| 22:32.38 | Druken | americans think canadian beer is like moonshine :) |
| 22:32.41 | Manipura | I'm canadian |
| 22:32.42 | ctooley | Anyone had any problems using Realtime with MySQL 4.1? |
| 22:32.56 | Manipura | I've never had american beer. But I'm sure its like water |
| 22:33.07 | Manipura | ctooley.. Nope |
| 22:33.15 | shido6 | tastes like carbonated tea |
| 22:33.19 | *** join/#asterisk nicuro (~X-pert@83.103.227.232) |
| 22:33.33 | Manipura | ctooley, whats your problem? |
| 22:33.33 | *** join/#asterisk sandnigg0r (~niggerplz@66-55-197-254.gwi.net) |
| 22:33.36 | ctooley | Manipura, only if you're drinking an A-B or Miller product. And for certain anything with the word "Light" in the name. |
| 22:33.41 | shido6 | coors light is not bad tho |
| 22:33.43 | Druken | carbonated tea? that just sounds nasty |
| 22:34.03 | ctooley | Manipura, 3.23 doesn't have subselects so we're looking at upgrading to 4.1 but don't want to break things. |
| 22:34.06 | Chotaire | any people been testing eyebeam with * lately? I still keep getting lag using that client. when I hold a call for a sec, it's gone. |
| 22:34.48 | Chotaire | btw urine... i actually drank some really good american beer on broadway, i was amazed. it's not all urine ;) |
| 22:35.02 | Manipura | ctooley, I've had no problems with 4.1 & realtime. The only problems I got with 4.1 is with the software I built to read cdr's |
| 22:35.26 | Manipura | I'm still yet to downgrade to 4.0, but again, don't want to break things... Seems difficult |
| 22:35.50 | ctooley | Manipura, we write our own CDR's with our agi to populate our necessary info. |
| 22:36.26 | *** join/#asterisk amir|away (~amir@195.226.9.186) |
| 22:36.27 | ManxPower | Chotaire: The small breweries can be very good. |
| 22:36.53 | ctooley | And things like : select b.city,b.state,a.area_code,a.prefix,count(distinct a.did) as count from inventory as a, npanxx as b where a.area_code = b.NPA and a.prefix = b.NXX and a.status = 0 group by city order by count asc; are starting to make my head hurt |
| 22:36.58 | ManxPower | mmlj4: I'm drinking a Guinness at the moment |
| 22:37.02 | Manipura | So do I, but I have a website (php) for billing purposes. It didn't like 4.1 - Asterisk had no problems |
| 22:37.11 | P-Chan | Ok, I've been having this problem for a while now, what would cause keypresses not to be registered when checking voicemail, which returns error: |
| 22:37.14 | P-Chan | Incorrect password '' for user '16' (context = <any>) |
| 22:37.31 | *** part/#asterisk obsidian-studios (~obsidian-@c-66-177-188-197.hsd1.fl.comcast.net) |
| 22:37.40 | ctooley | P-Chan, what kind of device are they using when they're checking voicemail? |
| 22:37.47 | Chuji | ~sip+nat |
| 22:37.48 | Chotaire | manxpower: yes indeed. I been to a brewery that is located on broadway. |
| 22:38.03 | Manipura | I'm drinking Bowen Island Beer, its the cheapest beer I've even seen. and tastes good |
| 22:38.09 | P-Chan | ctooley: Grandstream BudgeTones |
| 22:38.22 | bajanman | ahhhh Guinness |
| 22:38.23 | ctooley | P-Chan, make sure you're using the correct dtmfmode in sip.conf |
| 22:38.29 | Sift | what softphones work with asterisk? or does it depend on your provider? |
| 22:38.31 | Chuji | jbot: sip+nat is http://www.asteriskguru.com/natut.php <--- Zoa's tutorial |
| 22:38.32 | jbot | okay, Chuji |
| 22:38.34 | ctooley | I had to switch my Polycoms from inband to rfc |
| 22:38.37 | bajanman | I was told US peer is horse piss.. but. I'm sure I'm wrong...lol |
| 22:38.39 | P-Chan | ctooley: rfc2833 |
| 22:38.56 | Chotaire | Sift: eyebeam actually works great, besides of that lag problem. I see several people commenting about the same problem but I haven't yet found a solution. |
| 22:39.01 | Manipura | Sift, theres a good reference on voip-info.org |
| 22:39.10 | Chotaire | I have never seen a better softphone than eyebeam, it's not free though. |
| 22:40.03 | P-Chan | ctooley: And this phone is able to dial out without any problems, just not check voicemail - it doesn't detect anything but the # key. |
| 22:40.28 | ctooley | sounds like a dtmfmode to me... either that or the mode in voicemail.conf |
| 22:40.34 | ctooley | if there even is one. |
| 22:41.02 | *** join/#asterisk ZX81_Laptop (~ZX81@222-153-22-70.jetstream.xtra.co.nz) |
| 22:41.03 | ctooley | and dialing out has nothing to do with dtmf |
| 22:41.10 | *** part/#asterisk ZX81_Laptop (~ZX81@222-153-22-70.jetstream.xtra.co.nz) |
| 22:41.19 | ManxPower | Another Gem. Dialplan, Voicemail, Perl |
| 22:42.27 | Chotaire | gem... gemstones... off playing some ragnarok online. bbl. |
| 22:42.34 | ManxPower | http://pastebin.ca/10538 |
| 22:43.05 | ManxPower | This one should make your brain hurt and want to call the Society for the Prevention of Cruelty to Asterisk |
| 22:43.50 | ctooley | Manipura, am I going to need to reinstall anything other than updating the MySQL libraries? |
| 22:44.40 | Manipura | ctooley, I doubt it |
| 22:44.51 | *** join/#asterisk roche (~roche@216.194.173.2) |
| 22:45.01 | ctooley | Manipura, didn't think so. |
| 22:45.28 | Druken | spca eh? |
| 22:45.28 | Manipura | ManxPower the SPCA? |
| 22:46.02 | roche | Do you know If I can stop or erase the seconds counter of budgetone 100 phones? |
| 22:47.05 | ManxPower | Manipura: Of course. |
| 22:47.14 | moy | hi there, hopefullfy someone will know this: how can i do that asterisk hangup the calls that i do with Dial(ZAP/g1/123456) when the zaptel interfaces assigned to group 1 are not plugged to the FXO ports? |
| 22:47.22 | ManxPower | Read the pastebin, I think you'll agree |
| 22:47.39 | ManxPower | moy: you can't. |
| 22:48.17 | tzafrir | Hi, I'm trying to set up a FWD connection via IAX from behind a NAT. Outgoing works. For incoming I'm stuck on: |
| 22:48.34 | tzafrir | Apr 29 01:45:12 NOTICE[10847]: chan_iax2.c:5441 socket_read: Rejected connect attempt from 65.39.205.121 |
| 22:49.56 | moy | :( |
| 22:50.27 | tzafrir | A user is defined. The context is OK. I see packets coming on tcpdump. I searched google. |
| 22:50.36 | ManxPower | Chuji: I think he wants to have a call fail when there's no line plugged into the FXO port of a Digium card. |
| 22:50.56 | ManxPower | AND know the call failed so he can route the call differently. |
| 22:51.14 | *** join/#asterisk JunK-Y (~junky@modemcable174.107-81-70.mc.videotron.ca) |
| 22:51.28 | CoaxD | Manx: Aint a bad idea |
| 22:51.42 | tzanger | ManxPower: I still think it' dbe a minor driver change to have the DAA sense battery voltage |
| 22:51.42 | ManxPower | tzafrir: you have an issue with iax.conf (I don't know WHAT the issue is), not with your NAT or portforwarding. |
| 22:51.42 | CoaxD | Manx: Tho traditionally, asterisk has been abstracted from knowing such things |
| 22:51.51 | tzanger | ManxPower: and generate RA in that case |
| 22:52.12 | ManxPower | tzanger: one would think so. zttool can see when a line isn't plugged in, it's just that chan_zap.c doesn't seem to know. |
| 22:52.34 | ManxPower | CoaxD: It knows it on T-1 interfaces, VoIP interfaces. |
| 22:52.59 | tzanger | ManxPower: it's on my lengthy todo list :-) |
| 22:53.06 | CoaxD | Manx: I suppose it does |
| 22:53.16 | tzanger | CoaxD: it has? |
| 22:53.21 | CoaxD | Manx: it shouldn't be that hard to grab the knowledge of a red alarm |
| 22:53.33 | CoaxD | Manx: but, thats ioctls left and right. etc. |
| 22:53.44 | tzanger | CoaxD: it shouldn't be |
| 22:53.49 | tzanger | think of it as a Zap registration :-) |
| 22:54.02 | CoaxD | tzanger: Yeah, asterisk has no way to know what the hell is going on with an x100p, for instance |
| 22:54.11 | tzanger | yeah there needs to be something in the Zap driver |
| 22:54.13 | tzanger | actually there is |
| 22:54.14 | CoaxD | tzanger: Hell, * doesn't even know if the damn thing is on or off hook properly |
| 22:54.28 | tzanger | chan_zap can tell if a T1 chan is in RA or not |
| 22:54.32 | tzanger | thsi is the exact same thing, we just need to fill the existing structure |
| 22:54.33 | ManxPower | Hell, if you unplug the line from an X100P you'll even get a console notice/warning/somthing that the line went away! |
| 22:54.42 | Chuji | CoaxD : zap show channel doesn't show the right hookstatus? |
| 22:54.57 | CoaxD | Chuji: last time I checked, nope |
| 22:55.06 | CoaxD | Manx: That must be new |
| 22:55.12 | tzanger | Hookstate (FXS only): Onhook |
| 22:55.17 | tzanger | it's an FXS-only thing |
| 22:55.29 | Chuji | ahh |
| 22:55.30 | *** join/#asterisk PTG1234 (PTG123@ip68-106-24-139.ph.ph.cox.net) |
| 22:55.37 | ManxPower | Apr 28 17:55:14 WARNING[1141]: chan_zap.c:5653 handle_init_event: Detected alarm on channel 1: Red Alarm |
| 22:55.38 | ManxPower | Apr 28 17:55:17 NOTICE[1141]: chan_zap.c:5648 handle_init_event: Alarm cleared on channel 1 |
| 22:55.43 | tzanger | IIRC there should be an "in alarm" bit |
| 22:55.46 | ManxPower | That's my X100P when I unplugged the line for a moment |
| 22:55.56 | CoaxD | Manx: Nifty |
| 22:56.14 | tzafrir | ManxPower, I figured it is that, because I saw the IAX packets |
| 22:56.26 | moy | another one..... does anybody knows a way to get the caller ID into asterisk when the call is coming from PSTN?? |
| 22:56.32 | tzafrir | Where can I get more meat for debugging this |
| 22:56.40 | tzafrir | iax debug show nothing |
| 22:56.42 | ManxPower | moy: callerid=asreceived |
| 22:56.47 | cypromis | use a shovel |
| 22:56.52 | ManxPower | tzafrir: try iax2 debug |
| 22:57.09 | JunK-Y | zap show status wil be the answer. |
| 22:57.19 | moy | ManxPower: great, let me see |
| 22:57.33 | tzafrir | nada |
| 22:57.50 | moy | ManxPower: :( does not work |
| 22:58.03 | tzafrir | socket error. what could that be? |
| 22:58.03 | moy | i get strange chars in my SIP phone display |
| 22:58.21 | moy | i have a grandstream |
| 22:58.38 | tzanger | tzafrir: what are you working on? |
| 22:59.05 | tzafrir | tzanger, just trying to connect to FWD |
| 22:59.14 | ManxPower | moy: The grandstream does NOT support callerid name, only callerid number. |
| 22:59.34 | ManxPower | moy: I think you also need usecallerid=yes. It's all pretty obvious from the examples in the sample config file. |
| 22:59.44 | bajanman | hmmm hope shido comes back online... :( |
| 22:59.52 | tzanger | oh |
| 22:59.55 | tzafrir | from an * system, (1.0.5) |
| 22:59.55 | ManxPower | bajanman: he pasted the example URL |
| 22:59.58 | tzanger | FWD's fucked half the time |
| 23:00.08 | bajanman | manxpower: he didn't include an "inbound".. |
| 23:00.16 | bajanman | so I gave him something to do.. |
| 23:00.28 | bajanman | I'm sure he'll be back |
| 23:01.00 | tzafrir | tzanger, I use their "CallMe" script. And it seems to generate some IAX packets in my general direction |
| 23:03.09 | *** join/#asterisk Drel (~drel@dsl254-029-130.sea1.dsl.speakeasy.net) |
| 23:04.23 | moy | ManxPower: yep, layzie me, thanks |
| 23:06.28 | tzanger | does anyone here have a TE110P they'd be willing to trade a T100P for? I am going to hardware hack an optical E1 interface but already have a T100P |
| 23:08.25 | JunK-Y | who wants to trade 500$ for 2000$? |
| 23:08.30 | tzanger | eh? |
| 23:08.35 | tzanger | TE110P is $500, T100P is $500 |
| 23:08.42 | cypromis | was |
| 23:08.47 | tzanger | TE110 can do T1 and E1, T100P can only do T1 |
| 23:08.49 | JunK-Y | oups, read TE410P |
| 23:08.56 | ManxPower | Tes, but the TE110P does DMA and E-1 |
| 23:08.57 | cypromis | t100p suxx |
| 23:09.14 | tzanger | it's a perfectly fine T100P, I just need an E1 interface and already have teh $500 T100P |
| 23:09.22 | tzanger | cypromis: they work great for me |
| 23:09.43 | tzanger | I wonder if Digium'd give me a trade-in, I should contact 'em |
| 23:09.59 | ManxPower | tzanger: Digium DOES have a trade-in program, I think. |
| 23:10.08 | ManxPower | Not a very GOOD one, IIRC |
| 23:10.11 | tzanger | ha |
| 23:10.18 | tzanger | well I'm not going to piss away the cost of the card |
| 23:10.30 | tzanger | I'll use the A101u I have if I have to, but id' prefer to do this on a digium card |
| 23:10.33 | tzanger | simpler drivers |
| 23:11.15 | *** join/#asterisk kFuQ (~somedude@c-24-17-224-78.hsd1.wa.comcast.net) |
| 23:11.17 | *** join/#asterisk jhowardPA (~jhoward@12.25.177.120) |
| 23:11.35 | jhowardPA | Wooo! I got Asterisk working properly, at last! |
| 23:11.54 | jhowardPA | It took a couple days, but I think I get it, now. |
| 23:12.02 | ManxPower | Argh! The Supreme Court of the USA is not doing what the USA Govt wants them to so, so people are calling to impeach the judges. |
| 23:12.10 | tzafrir | Hmmm... I can't even find this exact error message in the code. Strange. |
| 23:12.15 | jhowardPA | Anybody here familiar with Adtran 550's? |
| 23:12.25 | tzanger | ha |
| 23:12.33 | tzanger | no wonder you want out of there, ManxPower |
| 23:12.39 | tzafrir | In chan_iax2.c I find a number of variations on this message, but each of them is longer |
| 23:12.42 | ManxPower | I predict that in 10 years the USA will be a christian theocracy in everything but name. |
| 23:12.49 | jhowardPA | In 10 years? |
| 23:13.06 | ManxPower | jhowardPA: yes, the supreme court is limiting it currently |
| 23:13.22 | jhowardPA | ManxPower: you're more optimistic than I. ;) |
| 23:13.38 | ManxPower | jhowardPA: it saddens me very much |
| 23:14.10 | jhowardPA | ManxPower: I'd be more than happy to take my talents elsewhere, if anyone knows a nice little country in Europe. |
| 23:14.24 | ManxPower | jhowardPA: Netherlands |
| 23:14.44 | jhowardPA | ManxPower: their platform toilets put the fear in my heart. ;) |
| 23:14.46 | tzafrir | Vatican. It answers the "little" part |
| 23:14.49 | ManxPower | jhowardPA: LOL! |
| 23:15.19 | jhowardPA | I can't, for the life of me, find any practical info on my Adtran 550. |
| 23:15.40 | ManxPower | jhowardPA: I've seen Adtran 650 and 750, but not 550 |
| 23:16.10 | jhowardPA | ManxPower: I'd wager that they're similar in a few ways. Did you get familiar? |
| 23:17.19 | jhowardPA | Oh, nope... neither of those is an "ATLAS" series. |
| 23:17.45 | *** join/#asterisk Legend (~legend@24.244.142.134) |
| 23:21.22 | ManxPower | jhowardPA: Hmm? |
| 23:23.34 | *** join/#asterisk obsidian-studios (~obsidian-@c-66-177-188-197.hsd1.fl.comcast.net) |
| 23:24.00 | sean | any suggestions for asterisk and voicexml integration? |
| 23:24.20 | obsidian-studios | anyone here familiar with cisco fxs ports? Can I get caller id info over sip to analog, or do I need MGCP/SGCP |
| 23:24.31 | *** join/#asterisk sandnigg0r (~niggerplz@66-55-197-254.gwi.net) |
| 23:24.41 | ManxPower | sean: Step 1: Learn C. Step 2: Read any VXML Specs Step3: ??? Step 4: Profit |
| 23:25.11 | sean | ManxPower: I was hoping for something a little more.. shall we say.. downloadable (-: |
| 23:25.36 | ManxPower | sean: HAHAHAHA!!!!!!!!!!!!!!!!!!!!!! |
| 23:25.44 | ManxPower | No such thing exists as far as I know. |
| 23:26.28 | *** join/#asterisk kielstirling (~kiel@knss.net) |
| 23:26.57 | sean | my brief research shows the same, unfortunately |
| 23:27.30 | kielstirling | can anyone help me with a asterisk h.323 termination problem? |
| 23:30.14 | kielstirling | nope OK i'll try later |
| 23:31.18 | *** join/#asterisk Blissex (~Blissex@82-69-39-138.dsl.in-addr.zen.co.uk) |
| 23:31.22 | *** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
| 23:35.03 | obsidian-studios | is caller id info typically sent as part of the sip data/packets? Not sure correct terms |
| 23:35.23 | tzafrir | my problem was not using the special name "iaxfwd" for the incoming iax username |
| 23:39.28 | *** join/#asterisk yaboo (~jsirucka@220.245.131.131) |
| 23:40.20 | *** join/#asterisk Sedorox (~Sed@Neptune.client.wlmsprt.pa.sed6.net) |
| 23:41.26 | *** join/#asterisk benw (benny@202-45-126-237.stabat.com) |
| 23:44.05 | *** join/#asterisk nDuff (~cduffy@64.128.31.220) |
| 23:45.54 | obsidian-studios | is there a way with * to change the caller id info displayed on the called parties end, like I am calling from behind * to a regular analog phone somewhere with caller id? Like fake or change the info |
| 23:47.37 | pjz | is there a way to make one context 'import' another? |
| 23:47.47 | obsidian-studios | yes include -> |
| 23:47.50 | obsidian-studios | include => |
| 23:47.52 | pjz | or to make one context have two names |
| 23:47.56 | pjz | ah, okay, thanks |
| 23:48.21 | obsidian-studios | like I specify times to include certain contexts and others at different times, like day and night stuff |
| 23:48.27 | pjz | for include => does the included context have to be defined before it in the file? |
| 23:48.33 | obsidian-studios | nope |
| 23:48.38 | pjz | obsidian-studios: ahhh, good idea |
| 23:48.40 | pjz | obsidian-studios: thanks |
| 23:48.43 | obsidian-studios | can be anywhere |
| 23:48.52 | obsidian-studios | not my idea but yes a good one ;) |
| 23:49.30 | yaboo | anyone using cisco call manager, like to ask some info about it please |
| 23:49.32 | obsidian-studios | all kinds of good examples here if you have not seen http://sprackett.com/asterisk/conf/extensions.conf |
| 23:49.41 | pjz | so I'm writing this toy python program to build my asterisk confg files |
| 23:50.01 | obsidian-studios | no but I use cisco equipment with * being the call manager, just not paying Cisco for it :) |
| 23:50.16 | obsidian-studios | you trying to get * to play with a call manager? |
| 23:50.22 | pjz | there's a bunch of classes and stuff, but the main defs look like: RGMUser("pj",510,"Paul Jimenez",510,phone=SipTone2), |
| 23:51.18 | pjz | so far I've got it generating the guts of extensions.conf, voicmeail.conf and sip.conf |
| 23:51.23 | obsidian-studios | do I need to pass along caller id info from a zap channel to sip or does that happen automatically? |
| 23:52.18 | *** join/#asterisk [hC] (~turnerd@209.153.195.141) |
| 23:53.58 | [hC] | hey, i dont know if this is possible or not, i have a master pbx in colocation where all of our DID's and master dialplan info is. I have some other pbx's out at remote offices that all do IAX trunks to the master pbx. What im looking to do is make RTP go point to point between the voip provider and the remote office asterisk box, rather than having to duplicate all the traffic thru the master pbx. (to save on bandwidth and latency) is there any way to d |
| 23:53.58 | [hC] | o this? |
| 23:54.27 | [hC] | Maybe by having sip set up the call thru the master pbx, then somehow insructing RTP to go direct to the two end points? I dont know if the protocol even permits this. |
| 23:55.10 | Moonwick | asterisk will usually try to take itself out of the call path if it's possible for it to do so |
| 23:55.26 | Moonwick | are you using Monitor() ? |
| 23:55.29 | [hC] | hm. If i can figure out what criteria it looks for, that would be great. |
| 23:55.32 | [hC] | No, not using monitor() |
| 23:55.35 | [hC] | I dont think? |
| 23:55.51 | jhowardPA | Can I plug a T1 into some kind of big-name box, which I then connect several Asterisk servers to via some kind of ethernet? |
| 23:56.13 | [hC] | The idea is the remote office places a call to the master pbx, and the master pbx has a dial plan for all the voip providers we peer with. the sip setup has to go thru the master pbx of course, but if RTP could somehow go direct, that would be excellent. |
| 23:56.36 | Moonwick | it should just work |
| 23:56.37 | obsidian-studios | jhowardPA: yes like a cisco router or other T-1 terminating router, or pci card in a linux machine :) |
| 23:56.42 | Moonwick | I'm not sure what else might prevent it from doing so |
| 23:56.45 | Sift | why in the world does the sipura 3000 have a 10base/t jack |
| 23:56.53 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 23:57.03 | ariel_ | Hello everyone |
| 23:57.11 | [hC] | Moonwick: It may act that way now, i havent looked closely, this is sort of a theory i was trying to solve before fully implementing it. Do you know what asterisk refers to this as, so i could maybe go look up some documentaion? |
| 23:57.22 | obsidian-studios | Sift: to daisy chain phones, or connect a computer to the network or other ip device not needing 100 |
| 23:57.42 | jhowardPA | obsidian-studios: Any suggestions on what kind of Cisco I can use? Also, what kind of networking would I use between the Asterisks and the Cisco? |
| 23:57.54 | Sift | but everything comes with 100 now days |
| 23:58.03 | Sift | just seems out of date |
| 23:58.12 | jhowardPA | obsidian-studios: I'm talking about connecting to POTS via the T1 - not the Net, btw. |
| 23:58.12 | obsidian-studios | depends on how much $ you want to spend etc |
| 23:58.15 | Sift | even my packet8 device... |
| 23:58.17 | Entegrity | anyone here got ccm up with asterisk? |
| 23:58.17 | Sift | is 100 |
| 23:58.19 | *** part/#asterisk bajanman (~william@cp66-203-194-32.cp.telus.net) |
| 23:58.27 | *** join/#asterisk bajanman (~william@cp66-203-194-32.cp.telus.net) |
| 23:58.35 | Entegrity | cisco callmanager to asterisk |
| 23:58.39 | obsidian-studios | jhowardPA: are you pots lines coming over T-1 or do you have actual pots lines just using your T-1's bandwidth? |
| 23:58.41 | Entegrity | I tried the cfg on voip info |
| 23:58.59 | Entegrity | but I'm an asterisk newb and dont uderstand extensions.conf yet |
| 23:59.06 | obsidian-studios | Sift: most voip stuff like phones do not need anything more than 10 |
| 23:59.10 | yaboo | is cisco call manager enterprise or carrier grade pbx? |
| 23:59.19 | Entegrity | enterprise |
| 23:59.25 | jhowardPA | obsidian-studios: My dept. fired the telecomm guy, and I'm trying to fill. I'm not sure... how can I determine that? |
| 23:59.40 | obsidian-studios | do you have pots lines some where? :) |
| 23:59.41 | Sift | obsidian-studios thats what bill gates said about 4 meg back in the day too :) :) |
