| 00:02.57 | *** join/#asterisk jks (jks@193.189.93.254) |
| 00:03.40 | bmoraca | interesting...callweaver can act as a T.38 gateway |
| 00:04.48 | russellb | FFA uses a commercially licensed fax stack, not spandsp. |
| 00:07.52 | hardwire | tries to figure out a good method of testing GotoIfTime a lot |
| 00:08.19 | hardwire | I suppose I could randomize the timezone |
| 00:10.40 | *** join/#asterisk devoid (~devnull@unaffiliated/devemo) |
| 00:12.17 | russellb | hardwire: going to write an automated test for the test suite?! |
| 00:12.26 | hardwire | f no |
| 00:12.31 | russellb | :-( |
| 00:12.44 | hardwire | russellb: I *am* the test suite |
| 00:13.44 | hardwire | heh.. there is a 'right' subdirectory in /usr/share/timezones |
| 00:13.49 | hardwire | I wonder wth that means. |
| 00:14.12 | hardwire | good.. posix/Etc/GMT(-+)x |
| 00:14.14 | hardwire | that should make it easier |
| 00:14.16 | hardwire | tests |
| 00:14.29 | *** join/#asterisk AndyML (~AndyML@pool-173-49-144-213.phlapa.fios.verizon.net) |
| 00:14.52 | AndyML | has anyone used twinstar with Xorcom Astribanks? |
| 00:16.22 | *** join/#asterisk romb_work (~romb@unaffiliated/romb-work/x-7222485) |
| 00:17.11 | *** join/#asterisk Katty (~Katty@mail.copi-rite.com) |
| 00:18.14 | *** join/#asterisk joobie (~joobie@mx01.anric.com.au) |
| 00:18.43 | joobie | guys anyone know of any good fxo units that are cheap and do multiple devices? ive been lookina tthe spa400 from linksys but they are liike 250-300$AUD.. |
| 00:19.34 | Katty | hi |
| 00:21.39 | joobie | hi katty |
| 00:21.44 | joobie | sup |
| 00:22.40 | tzafrir | russellb, anyway, here we just switched to DST |
| 00:23.09 | hardwire | tzafrir: where is here atm? |
| 00:23.24 | manxpower | joobie: that is cheap |
| 00:23.27 | hardwire | also.. why is there a GMT-14? |
| 00:23.37 | tzafrir | $ date -d '02:00' |
| 00:23.37 | tzafrir | date: invalid date `02:00' |
| 00:24.10 | Katty | joobie: nada, just tryin to get this headache to go away |
| 00:24.38 | joobie | manxpower, is it the cheapest option for a multiport fxo without degrading quality? |
| 00:24.55 | manxpower | joobie: it might be the cheapest option that will work well. |
| 00:25.01 | joobie | manxpower, im not too concerned about brand.. more concerned that it's multiport and doesn't screw the quality of the call.. and trying to go el'cheapo |
| 00:25.09 | joobie | cool |
| 00:25.19 | manxpower | joobie: cheap reliable good. Pick TWO. |
| 00:25.27 | joobie | Katty, "virtual headache" or an actual headache? |
| 00:25.36 | Katty | real headache |
| 00:25.45 | joobie | manxpower, i like all 3 though.. |
| 00:25.53 | manxpower | joobie: telecom is expensive. It's more expensive when you try to go cheap. |
| 00:25.55 | joobie | Katty, doh |
| 00:26.19 | p3nguin | tzafrir: $ date -d tomorrow |
| 00:26.19 | p3nguin | Fri Mar 26 19:26:11 CDT 2010 |
| 00:26.24 | p3nguin | works fine for me |
| 00:26.51 | joobie | tzafrir, what are you trying to do |
| 00:27.00 | Katty | goes upstairs for a bit |
| 00:27.17 | tzafrir | p3nguin, it's an invalid date because that specific time does not exist (due to DST timezone changes) |
| 00:27.33 | hardwire | somebody.. anybody.. I'm GMT-8 atm.. if I set the timezone in SayUnixTime to Etc/GMT-8 it gives me 8am not 4pm. |
| 00:27.39 | p3nguin | tzafrir: Do you really think that's why it is invalid? |
| 00:27.54 | hardwire | its 4pm :) |
| 00:28.00 | tzafrir | p3nguin, yes |
| 00:28.11 | p3nguin | tzafrir: I would have to disagree. |
| 00:28.20 | p3nguin | tzafrir: What do you think date -d is supposed to do? |
| 00:29.36 | *** join/#asterisk killfill (~killfill@200.63.96.244) |
| 00:29.40 | *** join/#asterisk mykhyggz (~col@evolone.org) |
| 00:29.44 | killfill | hi |
| 00:29.54 | tzafrir | p3nguin, first libc has to convert the string you gave it to "time" (seconds since epoch) |
| 00:30.09 | tzafrir | The string is a time in the current time zone |
| 00:30.14 | *** join/#asterisk W0OTM (~SAID@75-170-199-29.desm.qwest.net) |
| 00:30.18 | W0OTM | howdy |
| 00:30.39 | killfill | where can i get information about AMI commands an events?.. i.e. how to use them.. that parameters, etc. |
| 00:30.39 | tzafrir | But in the current time zone there's no 2AM today, due to switching to DST |
| 00:30.43 | p3nguin | So are you saying that your libc is broken? |
| 00:30.48 | tzafrir | no |
| 00:31.07 | tzafrir | it's a feature |
| 00:31.21 | p3nguin | Let me try 2 am on the day we switched to DST and see what happens. |
| 00:31.21 | killfill | i.e. the Events command is not working for me.. :S |
| 00:31.34 | hardwire | yeh... I'm doing something wrong |
| 00:31.35 | hardwire | http://hardwire.pastey.net/134595 |
| 00:31.36 | killfill | i get 500 Internal Error when i send "on" |
| 00:31.58 | killfill | im reading http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Events but i guess its too old? |
| 00:31.58 | bmoraca | hardwire has his own pastebin? you must screw up like crazy, man! |
| 00:32.40 | p3nguin | tzafrir: $ date -d "sunday march 14 02:00" |
| 00:32.40 | p3nguin | date: invalid date `sunday march 14 02:00' |
| 00:33.10 | p3nguin | but 03:00 is good: Sun Mar 14 03:00:00 CDT 2010 |
| 00:33.18 | W0OTM | will someone help me debug my sip configuration? |
| 00:33.19 | hardwire | http://bmoraca.pastey.net |
| 00:33.32 | p3nguin | http://asterisk.pastey.net |
| 00:33.44 | tzafrir | infobot, tell W0OTM about ask |
| 00:33.45 | p3nguin | http://flipflop.pastey.net |
| 00:34.42 | bmoraca | lol |
| 00:34.59 | p3nguin | What I really want to find out is what is accessing my hard disk continuously. |
| 00:35.15 | bmoraca | filemon |
| 00:35.17 | p3nguin | What tools/commands should I be looking for? |
| 00:35.29 | bmoraca | unless you're running linux...then, sorry |
| 00:35.32 | bmoraca | :P |
| 00:35.40 | p3nguin | Of course I'm running Linux. |
| 00:36.10 | hardwire | apparently I'm a boob |
| 00:36.11 | hardwire | http://sources.redhat.com/ml/glibc-bugs/2005-10/msg00071.html |
| 00:36.20 | hardwire | ALL IS WELL FOLKS! |
| 00:36.22 | hardwire | back to work. |
| 00:36.39 | *** join/#asterisk jtodd (lw9khlieav@asterisk/community-director-and-tie-dye-shirt-lover/jtodd) |
| 00:36.39 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 00:36.46 | bmoraca | WTB the freakin AT&T tech to show up, imo |
| 00:37.05 | bmoraca | my circuit's broken, they were supposed to be here at 4pm, and i want to go home and drink beer |
| 00:38.16 | hardwire | bmoraca: maybe they are using Etc/GMT* |
| 00:38.35 | hardwire | exten => 8463,n,SayUnixTime(,Etc/GMT+${RAND(0,12)},R) |
| 00:38.35 | hardwire | yay |
| 00:38.39 | hardwire | simple pleasures |
| 00:38.59 | hardwire | is there any builtin method of detecting how many times invalid has been called per channel? |
| 00:39.04 | hardwire | other than storing a variable? |
| 00:39.25 | hardwire | via the dialplan. |
| 00:40.48 | *** join/#asterisk diegomad (mad@186.0.4.182) |
| 00:41.33 | manxpower | Define "invalid" |
| 00:42.43 | *** join/#asterisk fofware (~chatzilla@186.125.110.227) |
| 00:43.33 | bmoraca | i believe he means the 'i' extension |
| 00:43.42 | hardwire | i |
| 00:43.43 | hardwire | sorry. |
| 00:43.52 | hardwire | i invalidly defined it |
| 00:44.22 | *** part/#asterisk randomuser (~pete@97-121-222-209.blng.qwest.net) |
| 00:45.54 | *** join/#asterisk lmsteffan (~laurent@reef.ac-noumea.nc) |
| 00:46.01 | *** join/#asterisk lmsteffan_ (~lmsteffan@reef.ac-noumea.nc) |
| 00:46.51 | manxpower | I don't think I've used exten => i in years. I usually define a wildcard extension to catch those and process them myself. |
| 00:47.52 | hardwire | well.. you are manxy enough to do things like that |
| 00:48.08 | hardwire | I just realized I was taking the counter I already have and replacing it with the same amount of code only using groups |
| 00:48.13 | Slugs_ | :) |
| 00:48.13 | hardwire | which does me very little good |
| 00:51.40 | hardwire | also.. /me hugs whoever implemented Transfer() |
| 00:51.55 | hardwire | I can actually move stuff off my network now. |
| 00:52.40 | hardwire | call comes into IVR over SIP.. Detect DTMF.. Transfer to another SIP provider. |
| 00:52.42 | hardwire | Ideally. |
| 00:53.12 | *** join/#asterisk arnotixe (~arno@190.131.122.24) |
| 00:59.07 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 01:12.36 | Slugs_ | hardwire, i changed my extension.conf, and sip.conf to understand 'context' better, i can dial 48707 from * sip ext 1001, but not the other way around. |
| 01:12.55 | hardwire | yeh |
| 01:13.09 | hardwire | err.. eh? |
| 01:13.22 | Slugs_ | ;0! |
| 01:13.41 | Slugs_ | sec. |
| 01:16.46 | Slugs_ | http://pastebin.com/0Tfrr2Af |
| 01:19.06 | p3nguin | Glad I was able to get to the bottom of that little problem. Software RAID array is rebuilding after the power outage, so of course there's continuous disk activity. :/ |
| 01:19.45 | *** join/#asterisk obnauticus (~obnauticu@about/windows/regular/obnauticus) |
| 01:26.59 | norrec | how can u force a peer to unregister? |
| 01:28.06 | hardwire | http://hardwire.pastey.net/134597 |
| 01:28.07 | hardwire | simple enough |
| 01:28.16 | Katty | hi |
| 01:28.17 | hardwire | no more statically defining those. |
| 01:28.28 | Katty | oh |
| 01:28.30 | hardwire | 15k+ files later |
| 01:28.33 | Katty | i wanted to share my fun little backup script |
| 01:28.38 | hardwire | do it! |
| 01:28.49 | p3nguin | slugs_: You cannot dial FROM extensions. Why are you having such a hard time understanding that? |
| 01:29.14 | p3nguin | slugs_: Devices dial extensions, extensions Dial() devices. |
| 01:29.57 | Katty | http://pastebin.com/xPScTs8a <- this is my lil postgres backup |
| 01:30.20 | p3nguin | slugs_: If SIP/1001 can dial 48707 and something happens, it's because you have exten => 48707,1,Somecommand(with,data,here) in your extensions.conf. |
| 01:30.36 | TJNII | Heh, I have a backup script like that for my Linux box at work. |
| 01:30.36 | Katty | http://pastebin.com/h89wjuRZ <- this is my other lil backup |
| 01:31.04 | TJNII | Dumps a tarball into a backup directory, which I then copy to the Windows box for the official corporate backup to pick up. |
| 01:31.09 | Katty | i've been debating adding a mutt to the end of it...but...meh |
| 01:31.21 | Katty | TJNII: the /backups directory is actually a mounted share |
| 01:31.42 | Katty | TJNII: so it actually dumps it to windows server. |
| 01:31.59 | Katty | TJNII: and both postgres.sh and backup.sh are cronjobs |
| 01:32.53 | Katty | TJNII: i've been debating setting up another server, and doing copy jobs from server a to server b every x minutes to maintain a battle ready secondary phone system should i need it |
| 01:32.59 | TJNII | Unfortunately my server config is the opposite, so I don't do that. The Windows machine is a laptop that floats between networks, whereas the Linux box has a static IP. Though your way would be more convenient. |
| 01:33.52 | TJNII | I don't have it in a cron job because it isn't fully automated. Though I really only need to backup that system when I modify my test scripts, so it isn't bad. |
| 01:33.54 | Katty | it's just laziness |
| 01:34.04 | Katty | yeah |
| 01:34.15 | Katty | i'm really not backing up much myself |
| 01:34.24 | Katty | the only stuff that changes is the voicemails and maybe extensions.conf |
| 01:34.39 | TJNII | I have a cold spare second * server, I should make some auto-config-sync goodness. |
| 01:34.49 | Katty | yeah |
| 01:34.50 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
| 01:34.51 | Slugs_ | i don't know y i have this mental block |
| 01:35.16 | TJNII | I really should set up wake on LAN on that so I can kick it and have it fire up without physical access. |
| 01:35.31 | Katty | TJNII: i think i'm just goign to have mine setup on another ip |
| 01:35.38 | Katty | TJNII: then set it up as 'server 2' on the polycom phones |
| 01:35.58 | Katty | TJNII: all that would really need to be done in my acense is to move the pri from one machine to another |
| 01:36.09 | TJNII | Mine is on IP n + 1 as well, both private and public. |
| 01:36.22 | p3nguin | slugs_: What are you trying to figure out now? |
| 01:36.23 | Katty | dear jenny, please move the big white cable from the shiny box labeled 1 to the shiny box labeled 2 |
| 01:36.46 | TJNII | I haven't thought og a graceful way to handle failover with my ITSP. |
| 01:36.59 | Katty | well |
| 01:37.10 | Katty | we call the telco and have them forward calls to the mobile phones |
| 01:37.20 | Katty | bandwidth.com is pretty good about doin that too |
| 01:37.25 | Katty | but it's not seamless by any means |
| 01:38.05 | TJNII | We've had 2 power failures here in the last 6 months. I might have to consider getting a cell phone again. |
| 01:38.07 | p3nguin | I'm fortunate enough that mine offers failover automatically. Just pre-configure it via web portal, and it Just Works. |
| 01:39.20 | p3nguin | After the power outage earlier, I decided I needed something else besides just a voice mail box on the failover server... I guess I'll make it dial cell phones. |
| 01:39.47 | TJNII | It isn't high priority for me, nobody calls. :( |
| 01:40.01 | Katty | yeah we don't have much call volume here either |
| 01:40.05 | arnotixe | hi I have a working GSM-SIP device; I can call to and through it and receieve calls. I also see SMS messages being recieved by asterisk when I turn on sip debug. But how can I send a MESSAGE packet to the GSM-SIP device? I tried Sendtext() but can't see it working on debug? |
| 01:40.13 | p3nguin | I was trying to make it dial out over PSTN via SPA-3102, but I don't know how to get it to dial from a VoIP call out the PSTN side. |
| 01:40.57 | p3nguin | As long as I can dial cell phones via VoIP, that'll be good enough for me. |
| 01:41.01 | p3nguin | (for now) |
| 01:41.07 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 01:41.14 | Katty | you could dump everyone into meetme |
| 01:41.24 | TJNII | I think my ultimate failure plan is going to be a UPS with lots of batteries. |
| 01:41.39 | Katty | a generator connected to the hamster wheel |
| 01:41.48 | TJNII | hehe. |
| 01:42.16 | hardwire | lol.. moving nanpa data into /etc/asterisk/nanpa/npa/nxx = fast lookup |
| 01:42.17 | p3nguin | I have one on the gateway and modem, and then another one for Asterisk, but sometimes the power stays off way too long. |
| 01:42.38 | Katty | that's where a generator would be handy |
| 01:42.44 | TJNII | Yea, I wish my UPS was 12v so I could jumper it to the car. |
| 01:42.58 | p3nguin | Yeah, but I'm not going to bother with a generator... my calls are not that critical. |
| 01:43.31 | p3nguin | If we took in hundreds of calls per day, then I would consider it. |
| 01:44.05 | *** join/#asterisk Chinorro (~Chino@202.219.27.77.dynamic.mundo-r.com) |
| 01:44.14 | TJNII | I have a friend who got a big (5kVa or something like that) UPS and batteries cheap. Last I heard he was wiring his house into it. |
| 01:44.24 | p3nguin | nice |
| 01:44.50 | TJNII | He has a gas furnace, so he can even have heat. |
| 01:45.03 | Katty | that would have been nice a coupel winters ago |
| 01:45.14 | Katty | southern missouri had about 2 weeks of no power areas. |
| 01:45.20 | p3nguin | I have three 1500s, a 650, and a 500 for my equipment. |
| 01:45.23 | TJNII | Nice. |
| 01:45.46 | Katty | it's quite an experience to go 2 weeks without any power |
| 01:45.51 | Katty | in the dead of winter |
| 01:45.56 | Katty | with ice all over everything |
| 01:46.00 | TJNII | I'll bet I remember one of those storms in Iowa. Woke up and half the apartment had power. |
| 01:46.12 | TJNII | And the power transformer outside was _angry_ about it. |
| 01:46.50 | TJNII | I like having periods without power, its fun. |
| 01:47.58 | Katty | :< |
| 01:48.16 | TJNII | Though the house I grew up in had a sump pump, and my Dad had it rigged so the smoke detector would go off when the pit filled up. I have not-so-fond memories of being woken up in the middle of the night to start the generator. |
| 01:48.25 | p3nguin | One 1500 on the gateway and modem, and the 500 on the main asterisk system at the primary location. One 1500 on my primary server (which is at the secondary location), one 1500 on the modem and gatway at the secondary location, along with the 650 on the backup asterisk box. |
| 01:48.29 | *** join/#asterisk LemensTS (~LemensTS@adsl-70-238-175-27.dsl.stlsmo.sbcglobal.net) |
| 01:48.41 | LemensTS | whats good |
| 01:48.43 | Katty | hi LemensTS |
| 01:48.46 | Katty | hugs LemensTS |
| 01:49.04 | p3nguin | So I really have better power protection on the backup system. :/ |
| 01:49.12 | TJNII | I don't think any of the servers at work are on UPSes. They are all protos in testing, though, so I guess it isn't worth it. |
| 01:49.16 | LemensTS | sup katty. |
| 01:49.25 | LemensTS | what no survivor on tv tonite grrr |
| 01:49.28 | Katty | LemensTS: just took some pain killers. |
| 01:49.32 | Katty | LemensTS: head's killin me |
| 01:49.45 | TJNII | p3nguin: You apparently do. |
| 01:49.55 | LemensTS | Katty: chase it with a beer and it wont |
| 01:50.11 | TJNII | Let me guess, you're prepared if the first goes out but not the second, right? |
| 01:50.12 | p3nguin | I'm actually wanting to get another 1500 just for phones. |
| 01:50.13 | Katty | dont like beer |
| 01:50.20 | TJNII | Wine then. |
| 01:50.22 | Katty | i think it was from dehydration |
| 01:50.36 | Katty | i drank literally nothing all day long |
| 01:50.47 | Katty | i should probably go get another glass of water |
| 01:50.51 | Katty | brb |
| 01:51.21 | TJNII | Make a white russian with creme de menthe instead of vodka. |
| 01:51.26 | TJNII | It is delicious. |
| 01:51.40 | p3nguin | What really annoys me is when the power is out all over town and my batteries last longer than those of the ISP. Then I'm sitting here still trying to use the internet and the ISP has no power. |
| 01:51.45 | LemensTS | white russians are dangerous they taste to good |
| 01:51.57 | TJNII | That tastes like melted mint ice cream. |
| 01:52.12 | TJNII | Use peppermint schnapps and it tastes like an Andes mint. |
| 01:52.31 | p3nguin | I have a box of those right here on my desk. |
| 01:52.36 | TJNII | p3nguin: At least you're on the right side of that problem. |
| 01:53.13 | p3nguin | I just hate losing power. Period. |
| 01:53.21 | LemensTS | p3nguin: lol i always race home to shut my servers off. good thing they have a fancy shutdown tool that i have not installed software for |
| 01:53.25 | p3nguin | <PROTECTED> |
| 01:53.33 | Katty | returns |
| 01:53.34 | p3nguin | Still going. |
| 01:53.51 | TJNII | /proc/mdstat whoo! |
| 01:54.09 | p3nguin | lemensts: Yeah, my box didn't shut down before the battery exhausted, now I'm waiting on the raid to rebuild. :( |
| 01:54.31 | Katty | bummer. |
| 01:54.35 | p3nguin | At least it's running at a reasonable speed. |
| 01:54.45 | p3nguin | 60 MB/s |
| 01:54.47 | LemensTS | raids suck blah |
| 01:55.04 | Katty | ryan's been raiding all night :< |
| 01:55.14 | Katty | didn't kind of raid, but still makes me sad. |
| 01:55.27 | Katty | s/didn't/different/ |
| 01:57.02 | LemensTS | s/raids/vacuum's |
| 01:57.24 | LemensTS | lol |
| 01:57.30 | LemensTS | a b c |
| 01:57.38 | LemensTS | s/b/is/ |
| 01:58.12 | p3nguin | Ugh, that reminds me that I need to go over to the farm supply store with some white spray paint. |
| 01:58.26 | TJNII | O.o |
| 01:58.33 | LemensTS | what reminds u of that |
| 01:58.35 | p3nguin | I was there earlier and the banners out front read, "Customer Appreciation Day's." |
| 01:58.35 | TJNII | I don't think they would care for that. |
| 01:58.42 | TJNII | Aah. |
| 01:58.43 | p3nguin | Day's |
| 01:59.05 | p3nguin | Not Days like normal people. |
| 01:59.20 | TJNII | Taco bell had a sign that said "99 cent taco's" for a long time. |
| 01:59.26 | TJNII | We would point and laugh. |
| 01:59.38 | p3nguin | I got one for ya. Just a second. |
| 01:59.46 | hardwire | tacois! |
| 01:59.50 | hardwire | I love tacois! |
| 02:00.10 | hardwire | -> home |
| 02:01.05 | LemensTS | We had a billboard of these 3 kids in a car drinking hanging out the window having an absolute hell of a time. And it said something bout Your actions influence others. And they want people not to drink? lol |
| 02:01.57 | TJNII | Time to go. |
| 02:02.09 | p3nguin | well hell. |
| 02:02.14 | *** join/#asterisk rossand (~aross@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 02:02.34 | p3nguin | I wanted him to see this: http://imagebin.org/90412 |
| 02:03.18 | p3nguin | I took the picture, then called them to let them know they shouldn't let the retarded kids do the marquis. |
| 02:03.45 | *** part/#asterisk AndyML (~AndyML@pool-173-49-144-213.phlapa.fios.verizon.net) |
| 02:04.01 | *** join/#asterisk hluesea (~hulusikah@88.247.127.66) |
| 02:16.37 | *** part/#asterisk Sipster (~18c8052d@gateway/web/freenode/x-nbnzaiuqqwymdrgd) |
| 02:23.42 | *** join/#asterisk the1_ (~x@122.52.175.49) |
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| 02:29.02 | p3nguin | That seems to work pretty good. When the ITSP goes to failover, now it rings cell phones before dumping to voicemail. |
| 02:29.46 | p3nguin | Heck, I could incorporate the IVR and everything on the secondary box, and then dial cell phones instead of local phones. |
| 02:32.45 | *** join/#asterisk darkdrgn2k (~darkdrgn2@bas2-toronto44-1242514614.dsl.bell.ca) |
| 02:33.22 | darkdrgn2k | Hi, how can im writing a short dial plan, i have it dialing a number. Is ther a way i can make it wait a few seconds then send a DTFM tone to it? |
| 02:34.00 | darkdrgn2k | i know i can make it wait using wait.. |
| 02:34.04 | darkdrgn2k | but how do i send a dtfm tone |
| 02:34.52 | Slugs_ | darkdrgn2k: SendDTMF(digits[|timeout_ms]) |
| 02:35.04 | darkdrgn2k | that would owork wouldnt it |
| 02:35.17 | arnotixe | darkdrgn2k, I'm trying the same with Dial(number,timeout,D(905)) but I can't seem to make it work. |
| 02:35.31 | arnotixe | neither M(macroname) works for me it seems. |
| 02:39.11 | darkdrgn2k | huh |
| 02:39.15 | darkdrgn2k | D might be what im looking for |
| 02:40.14 | darkdrgn2k | hmm |
| 02:40.33 | arnotixe | darkdrgn2k, please tell me if it works for you |
| 02:40.42 | darkdrgn2k | what do you put for timeout |
| 02:41.13 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 02:41.16 | darkdrgn2k | nm |
| 02:41.17 | darkdrgn2k | it worked find |
| 02:41.48 | darkdrgn2k | exten => s,n,Dial(SIP/9051953776@Line,,D(1)) |
| 02:42.21 | darkdrgn2k | is there any way i can make it delay a second or two before briding the call? |
| 02:44.39 | *** join/#asterisk OrNix (~ornix@host89-251-107-3.hnet.ru) |
| 02:49.17 | the1_ | darkdrgn2k 905 is the local? |
| 02:49.23 | the1_ | did it work? |
| 02:51.49 | *** join/#asterisk antiwire (~antiwire@unaffiliated/antiwire) |
| 02:52.38 | Katty | peeks in |
| 02:52.57 | ChannelZ | ah crap I forgot I need to go take out the trash |
| 02:53.27 | Katty | i never take out the trash |
| 02:53.40 | Katty | deligated that responsibility to ryan years ago :P |
| 02:54.56 | ChannelZ | Well I live alone so I guess you can call me Ryan |
| 02:55.32 | Katty | bummer :< |
| 02:56.14 | Katty | http://i.imgur.com/jWhvY.jpg <- not political. |
| 02:56.51 | Slugs_ | setup a goto(trash) |
| 02:57.02 | ChannelZ | Really? That cat looks like a democrat. |
| 02:57.32 | Slugs_ | lol |
| 02:57.46 | ChannelZ | I mean look at the confusion on his face. I doubt he has a job. |
| 02:57.49 | Katty | you mean, democat |
| 02:57.57 | ChannelZ | Maybe I could hire him to take out my trash |
| 02:58.03 | Katty | maybe he's |
| 02:58.05 | Katty | CATATONIC |
| 02:58.13 | ChannelZ | RAWR! |
| 02:58.44 | ChannelZ | ok brb |
| 02:59.44 | LemensTS | meow |
| 03:00.25 | Katty | herroes LemensTS |
| 03:03.36 | *** join/#asterisk phix (~threat@123-243-44-131.tpgi.com.au) |
| 03:03.37 | phix | hi gang |
| 03:04.02 | Katty | howdy |
| 03:04.22 | darkdrgn2k | the1_: sorry 1 is the local |
| 03:04.25 | darkdrgn2k | fo rme |
| 03:04.27 | darkdrgn2k | for me |
| 03:04.36 | darkdrgn2k | but i still hear a quick He from Hello |
| 03:07.50 | ChannelZ | http://icanhascheezburger.com/2010/03/25/funny-pictures-taeks-foreber/ |
| 03:08.13 | Katty | awww hehehehe |
| 03:09.38 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 03:14.12 | norrec | how can you tell what user asterisk is running as? |
| 03:15.57 | Slugs_ | ps -aux | grep asterisk |
| 03:16.31 | p3nguin | fail |
| 03:16.52 | p3nguin | slugs_: -aux is invalid. You probably mean aux. |
| 03:17.01 | ChannelZ | Depends on your ps |
| 03:17.27 | ChannelZ | It might complain but show you anyway |
| 03:17.37 | Slugs_ | indeed |
| 03:17.59 | norrec | oh yeah, i was forgetting x lol |
| 03:18.11 | p3nguin | It says STOP USING THE WRONG SYNTAX PLEASE, but then shows you what you wanted to see anyway. |
| 03:18.24 | ChannelZ | Sounds familiar |
| 03:18.30 | ChannelZ | Are you German? |
| 03:18.31 | Slugs_ | yeah.. |
| 03:18.36 | hluesea | hello |
| 03:18.52 | hluesea | is anyone try to chan_mobile here ? |
| 03:19.31 | *** join/#asterisk hluesea (~hulusikah@88.247.127.66) |
| 03:19.36 | hluesea | actually i want to how many mobile phone supported at 1 usb dongle ? |
| 03:20.31 | p3nguin | "Note that "ps -aux" is distinct from "ps aux". The POSIX and UNIX standards require that "ps -aux" print all processes owned by a user named "x", as well as printing all processes that would be selected by the -a option. If the user named "x" does not exist, this ps may interpret the command as "ps aux" instead and print a warning. This behavior is intended to aid in transitioning old scripts and habits. It is fragile, subject to ... |
| 03:20.37 | p3nguin | ... change, and thus should not be relied upon." |
| 03:21.33 | p3nguin | So I'm going to start creating user x on every system just to break people's habits. |
| 03:22.23 | antiwire | you SOB |
| 03:23.11 | ChannelZ | Well, he doesn't get out much.. |
| 03:23.31 | p3nguin | Why do I need to go out at all? |
| 03:24.01 | ChannelZ | So the rest of us have some relief |
| 03:24.02 | p3nguin | I can have beer and pizza catered... I have running water and electricity... |
| 03:26.49 | p3nguin | And just so you know, you can use ps all by itself to find out the same thing as displaying everything and then piping it into grep. ps -C asterisk u |
| 03:27.46 | Katty | peeks in |
| 03:28.41 | *** join/#asterisk geneticx (~geneticx@c-65-34-240-65.hsd1.fl.comcast.net) |
| 03:28.56 | *** join/#asterisk strungmind (~strungmin@18.sub-72-97-145.myvzw.com) |
| 03:29.02 | Katty | hmm |
| 03:29.15 | Katty | i've got dah munchies |
| 03:29.43 | Katty | let's go to steak n shake |
| 03:29.55 | p3nguin | I don't blame you, but I'm thinking I should skip the food and just go to bed. |
| 03:30.07 | Katty | yes it would probably be healthier |
| 03:30.16 | p3nguin | I went there for lunch yesterday, so I should skip it tonight. |
| 03:30.23 | Katty | but then the legs will start hurting :< |
| 03:30.31 | Katty | do your legs hurt when you get hungry? |
| 03:30.43 | p3nguin | not that I know of. |
| 03:31.14 | p3nguin | I just get stomach cramps, due to some stupid spastic colon thingy. |
| 03:31.44 | Katty | ahh |
| 03:31.54 | Katty | guess we all have our own responses |
| 03:31.58 | p3nguin | It's a bother. |
| 03:32.13 | Katty | my legs also ache when i get too cold |
| 03:32.18 | p3nguin | The solution is to keep mealtime consistent. |
| 03:32.33 | Katty | i just eat when i'm hungry |
| 03:32.39 | Katty | it's usually about the same time tho, really |
| 03:32.40 | *** part/#asterisk manxpower (~ewieling@216.186.151.147) |
| 03:34.06 | p3nguin | One time I was trying to explain to my dad about my stomach situation, and he told me about some woman he knows that had a similar problem. Her solution was to have six meals per day instead of the normal three. I've often wondered if I could adapt to six half-sized meals. |
| 03:34.39 | p3nguin | I hate to eat just to be eating. |
| 03:35.08 | antiwire | That's called "issues" |
| 03:35.10 | Katty | probably 300 calorie meals |
| 03:35.39 | p3nguin | I'm just wondering if it would be good for anyone or only those with certain conditions. |
| 03:35.43 | Katty | alot of people i know eat several little meals a day |
| 03:36.41 | p3nguin | I figure I would be tempted to eat regular sized portions instead of what I should be having. |
| 03:36.57 | antiwire | nuts, fruit, water |
| 03:37.20 | p3nguin | I like some nuts, but it's easy to have too much. |
| 03:37.48 | p3nguin | Some fruits I'm not fond of, like peaches. Blehck. |
| 03:38.13 | antiwire | Don't eat the fur |
| 03:38.22 | p3nguin | Shave it first? |
| 03:38.27 | antiwire | totally |
| 03:40.11 | *** join/#asterisk pawz (~pawz@ppp118-208-94-150.lns20.bne4.internode.on.net) |
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| 03:56.41 | *** join/#asterisk romb_work (~romb@unaffiliated/romb-work/x-7222485) |
| 04:06.41 | norrec | does any1 have any experience with fax for asterisk? |
| 04:12.45 | ChannelZ | I got my free license but have never actually set it up |
| 04:12.50 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 04:18.41 | darkdrgn2k | hi im briding a call using exten => s,n,Dial(SIP/18887775509@MyTrunk,,D(1)) |
| 04:18.58 | darkdrgn2k | is there a way i can delay the bridge a few seconds. i get the first 1/2 a word before the D(1) kicks in' |
| 04:21.49 | norrec | ChannelZ: hm, well idk if that helps too much lol |
| 04:32.30 | *** join/#asterisk b14ck (~comradeb1@75.80.14.233) |
| 04:32.31 | spartan07 | is away: I'm busy |
| 04:56.24 | *** join/#asterisk jameswf (~james@unaffiliated/jameswf-home) |
| 05:18.06 | *** join/#asterisk JAMMAN2110 (~james@unaffiliated/jamman2110) |
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| 05:51.08 | AeroCloud | anyone know what hardware if any other than upgrading the cpu helps with transcoding? |
| 05:58.18 | AeroCloud | darkdrgn2k: did you try the M() macro inside the dial.. and issue a Wait(2) inside that macro? |
| 05:59.28 | AeroCloud | <-- sleep |
| 06:09.17 | norrec | does any1 have any experience with fax for asterisk? |
| 06:11.21 | *** join/#asterisk romb_work (~romb@unaffiliated/romb-work/x-7222485) |
| 06:12.35 | *** join/#asterisk ltd_wk (~z@sixified.transact.net.au) |
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| 06:16.56 | *** join/#asterisk SouthSuburbTech (~Administr@184.78.19.77) |
| 06:17.34 | SouthSuburbTech | could someone help me config google voice and gizmo5 in asterisk |
| 06:17.42 | SouthSuburbTech | i've google and tryied everything , can't seem to get it working |
| 06:19.26 | *** join/#asterisk siva214215 (~Glass@122.169.220.182) |
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| 07:02.36 | sawgood | Where is everyone? |
| 07:05.50 | *** join/#asterisk Ad-Hoc (~nimbus@62.169.216.185) |
| 07:10.57 | siva214215 | When I try to run "console dial 1001" at CLI prompt the response is "No such command 'console dial 1001' ?? |
| 07:10.59 | siva214215 | :( |
| 07:11.06 | siva214215 | what is wrong with this?? |
| 07:13.50 | *** join/#asterisk smooth_penguin (~smoove@59.95.18.104) |
| 07:14.21 | *** join/#asterisk sawgood (~sawgood@173-13-158-25-sfba.hfc.comcastbusiness.net) |
| 07:16.09 | *** part/#asterisk sawgood (~sawgood@173-13-158-25-sfba.hfc.comcastbusiness.net) |
| 07:18.31 | *** join/#asterisk sawgood (~sawgood@173-13-158-25-sfba.hfc.comcastbusiness.net) |
| 07:21.17 | ChannelZ | siva214215: do you must not have it loaded |
| 07:22.58 | ChannelZ | siva214215: load chan_oss |
| 07:25.10 | *** join/#asterisk e-jones (~jkastner@nat/redhat/x-msbcwqxmqfqgpigj) |
| 07:25.57 | sawgood | how does one determine if they have the 'Asterisk add-ons' installed? |
| 07:28.22 | ChannelZ | well, add-ons comes with format_mp3, chan_mobile, chan_ooh323, some others.. so if you had any of those chances are you got them through asterisk-addons |
| 07:29.26 | ChannelZ | ok is there some drugged-out version of 'ls' that will let you specify which columns you want to see in what order, like being able to show accessed time and created time together too? |
| 07:29.38 | siva214215 | channel2 thanks :) |
| 07:31.03 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 07:33.07 | *** part/#asterisk joobie (~joobie@mx01.anric.com.au) |
| 07:34.13 | ChannelZ | guess I can use find.. |
| 07:35.07 | phix | http://xkcd.com/719/ |
| 07:35.30 | phix | IT IS VERY RELEVENT! |
| 07:36.16 | sawgood | I do not have the module format_mp3 |
| 07:36.30 | ChannelZ | well there you go then |
| 07:36.33 | sawgood | show module like format (lots listed but not format_mp3 |
| 07:36.50 | sawgood | do the Asterisk add ons come in RPM form? |
| 07:37.14 | *** join/#asterisk farkus_ (~chatzilla@cpe-72-225-212-219.nyc.res.rr.com) |
| 07:38.56 | ChannelZ | maybe |
| 07:39.09 | sawgood | I see them ... there are several add on RPM modules for 1.6.0 |
| 07:39.17 | ChannelZ | and 'show module' only shows *loaded* modules |
| 07:39.25 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
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| 08:07.25 | *** join/#asterisk Polysics (~Luca@host83-67-dynamic.30-79-r.retail.telecomitalia.it) |
| 08:07.47 | Polysics | hello |
| 08:07.59 | ChannelZ | yarr |
| 08:08.00 | Polysics | i have my queues configured in a MySQL db |
| 08:09.10 | Polysics | do i need to add and remove members when they log in, or can i just leave them in and they will be skipped if they are not there? |
| 08:12.31 | ChannelZ | What do you mean by 'if they are not there'? |
| 08:12.32 | *** join/#asterisk kruemeltee (~Maddin@port-92-198-62-82.static.qsc.de) |
| 08:12.37 | Polysics | not online |
| 08:12.47 | ChannelZ | If their device is active but they are physically not present, then their phones will just ring |
| 08:13.39 | Polysics | no, i mean, not logged in to the SIP system... this is supposed to be softphone-based |
| 08:14.20 | ChannelZ | If their devices are not accessable at all it's probably OK but might cause a bunch of noise in the console/logs |
| 08:14.32 | Polysics | so it would be better to add/remove them? |
| 08:14.46 | Polysics | i already have some AMI monitor that logs who logs in and out |
| 08:14.57 | ChannelZ | Ideally yes |
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| 08:17.26 | Polysics | ChannelZ, since you seem knowledgeable, can i ask you a big-picture question? |
| 08:17.40 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 08:17.56 | Polysics | ssytem we are developing is a web-based translation network, where foreigners living in Italy and working for the governmente provide translations to each other |
| 08:18.12 | Polysics | everything is web-based usgin Zoiper Web piloted thtough JS |
| 08:18.34 | Polysics | every op has one or more language he speaks, which is in a DB table |
| 08:18.49 | Polysics | i need an incoming call to be routed to a queue for each language |
| 08:19.05 | Polysics | i also need to be able to directly call one op, but that' easy |
| 08:19.18 | Polysics | would you use * queues or code it by hand in AGI/AMI? |
| 08:19.30 | Polysics | i am pretty proficient in Ruby/EventMachine |
| 08:19.39 | Polysics | actually way better than with * confs :-) |
| 08:20.31 | Polysics | each call is then logged and the "owning" company for each op gets credits added/deducted, to keep the service fair |
| 08:20.41 | Polysics | i'd say i have logging nailed down |
| 08:21.17 | Polysics | although i still have trouble keeping tab of each operators' status in a mysql table that can be displayed by PHP |
| 08:22.38 | *** join/#asterisk mpe (~mpe@gate.ipvision.dk) |
| 08:25.06 | Polysics | coding queues "by hand" looks like overkill, although i could see some benefits |
| 08:25.16 | creativx | are you using extension hints? |
| 08:25.19 | ChannelZ | Hmm I would probably just use queues, I'm not sure what you'd gain writing it yourself (unless there's just something queues can't do that you would need to do) |
| 08:25.39 | creativx | i use pausequeuemember when an operator changes their status to unavailable |
| 08:25.45 | Polysics | by the way, let me expose one thing: ops provide "reperibility times" where they are also available on their cellphone |
| 08:25.47 | creativx | i think thats what the ami command was |
| 08:26.02 | Polysics | writes down what creativx says :-) |
| 08:26.12 | creativx | so basically the members are always in the queue |
| 08:26.17 | creativx | but paused/resumed as needed |
| 08:26.17 | Polysics | cellphones could be a break-it problem |
| 08:26.29 | creativx | ive not tried to tie cellphones into queue handling |
| 08:26.32 | Polysics | can a queue call a cellphone instead of a SIP user based on time of day? |
| 08:26.33 | creativx | only on DID calls |
| 08:26.44 | creativx | yeah it can |
| 08:26.49 | creativx | but you'd have to handle the logic somewhere |
| 08:26.57 | creativx | or wait.. i think it can, hehe |
| 08:27.06 | ChannelZ | If you call a Local channel you can make the dialplan do whatever you want |
| 08:27.18 | creativx | yeah |
| 08:27.23 | Polysics | so i need to read up on Local channels? |
| 08:27.30 | ChannelZ | Local/999 and make extension 999 do whatever you want |
| 08:27.36 | creativx | local channels are a beast of their own :] |
| 08:27.54 | Polysics | oh, Local just means "call this ext"? |
| 08:28.20 | Polysics | cool, then the idea is: configure SIP and cellphone login in an AGI script that responds to a Local call |
| 08:28.42 | Polysics | then queues just call those |
| 08:28.51 | ChannelZ | Yeah a Local channel is sort of a way to feed a call back into the dialplan |
| 08:28.53 | Polysics | can a Local call tell the calling queue "skip me"? |
| 08:29.10 | creativx | yeah by modifying channel status i assume |
| 08:29.16 | creativx | unavailable perhaps |
| 08:29.30 | Polysics | souncs pretty good |
| 08:29.31 | ChannelZ | I'm not sure Local channels have status' because they aren't real channels |
| 08:29.35 | creativx | just like a normal agent that presses hangup on a queue call would make the call jump back into the queue |
| 08:29.47 | creativx | me neither ChannelZ :-) |
| 08:29.54 | creativx | last time i poked in our conf files were over 1 year ago |
| 08:29.55 | creativx | hehe |
| 08:30.42 | ChannelZ | has to go to bed - have fun y'all |
| 08:30.50 | Polysics | wil ltry :-) |
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| 09:25.01 | siva214215 | I've installed asterisk and when I try to connect with ekiga from other host |
| 09:25.29 | siva214215 | i'm having the error Registration failed |
| 09:25.52 | siva214215 | how can I resolve this?? |
| 09:26.40 | siva214215 | Any advice is highly appreciated :) |
| 09:26.48 | *** join/#asterisk ddefrenne (~ddefrenne@83.101.71.187) |
| 09:27.20 | *** join/#asterisk Toommi (~name@geldern.screenwork.de) |
| 09:29.48 | Toommi | can i use regex in hints ? |
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| 09:51.57 | TommyBotten | Toommi: Not regular expressions, but you can use the asterisk pattern matching in versions 1.6.1 and newer |
| 09:52.42 | Toommi | i tried : hint(Custom:text) _*51ZZ => { |
| 09:53.02 | Toommi | but it did not worked out |
| 09:53.23 | Toommi | but an extension : _*52ZZ => { is workin |
| 09:53.41 | Toommi | ofc i meant the asterisk pattern ;) |
| 09:57.53 | Toommi | ok i am just stupid :) |
| 10:07.18 | TommyBotten | Did it work out? |
| 10:08.35 | Toommi | yeah i just missed a } in a variable in the same line *gg |
| 10:09.45 | Toommi | but my blf still not working with this "more dynamic" hint ^^ |
| 10:10.10 | Toommi | but i think i can debug this :> |
| 10:10.32 | *** join/#asterisk Tim_Toady (~moi@77.49.45.81.dsl.dyn.forthnet.gr) |
| 10:10.56 | tuxx- | Hi again! :) Were trying to reinvite our rtp stream, but it somehow fails. We got all sippeers with the options: 'canreinvite=yes', 'nat=no'. Also, the dial statement we use has no extra paramaters. How is it possible that the RTP stream still goes through asterisk when all these options are set? |
| 10:11.18 | tuxx- | oh yeah, codecs are the same for all phones |
| 10:11.28 | tuxx- | read on voip-info.org that thats a well known 'bug' too |
| 10:11.29 | tuxx- | ;p |
| 10:11.44 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
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| 10:19.26 | Toommi | <PROTECTED> |
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| 10:36.11 | jarvis141 | Hi, mates. Anyone who tried to use Asterisk as a Voice Mail for Cisco Call Manager? |
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| 10:45.56 | Toommi | <PROTECTED> |
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| 10:47.42 | tuxx- | Hi again! :) Were trying to reinvite our rtp stream, but it somehow fails. We got all sippeers with the options: 'canreinvite=yes', 'nat=no'. Also, the dial statement we use has no extra paramaters. How is it possible that the RTP stream still goes through asterisk when all these options are set? |
| 10:47.48 | tuxx- | oh yeah, codecs are the same for all phones |
| 10:47.53 | tuxx- | read on voip-info.org that thats a well known 'bug' too |
| 10:48.07 | tuxx- | why does my question involve 3 lines, i wonder |
| 10:48.08 | tuxx- | hmmm |
| 10:48.08 | tuxx- | ;d |
| 10:48.44 | *** join/#asterisk romb_work (~romb@unaffiliated/romb-work/x-7222485) |
| 10:52.21 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-fpfvcufcydddgekw) |
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| 11:00.40 | *** join/#asterisk EmleyMoor (phil@topdeck.tinsleyviaduct.com) |
| 11:01.34 | EmleyMoor | Is there a way to create a log file of dialplan trace? Had a call get past a filter and am not sure why, and my console session was not connected at the time. |
| 11:03.15 | TommyBotten | Yes. Take a look at logger.conf |
| 11:03.53 | *** join/#asterisk k-man (~jason@unaffiliated/k-man) |
| 11:04.51 | k-man | any traps to watch for upgrading from 1.4 to 1.6? |
| 11:06.37 | *** join/#asterisk dennisG (~visionlab@84.30.136.208) |
| 11:07.45 | *** join/#asterisk Dovid (~annon@213.8.121.90) |
| 11:07.48 | Dovid | j #asterisk-il |
| 11:09.39 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-lvlbjpwlssiedgmq) |
| 11:10.13 | k-man | i upgraded to 1.6 and can no longer receive calls |
| 11:10.18 | EmleyMoor | TommyBotten: Thanks - now I should be able to see why some calls sneak round |
| 11:10.47 | EmleyMoor | k-man: Over what technology? |
| 11:10.54 | k-man | err... sip |
| 11:12.03 | k-man | when i call my number, i get a message from the voip provider saying the number is busy |
| 11:12.29 | k-man | asterisk is receiving info about the call, but not ringing internal phones |
| 11:12.35 | k-man | ill paste some info hang on |
| 11:14.01 | k-man | would the output of sip set debug on have any sensitive info in it? |
| 11:14.56 | EmleyMoor | I don't know - but you could probably tell by reading through it to make sure. |
| 11:15.02 | k-man | http://pastebin.ca/1852764 |
| 11:15.12 | k-man | that seems relevant |
| 11:17.28 | EmleyMoor | I'm no expert in 1.6 but I would be double-checking sip.conf and the provider's settings |
| 11:18.25 | k-man | EmleyMoor: ok |
| 11:18.40 | *** join/#asterisk Akiraa (~Akira@92.81.175.135) |
| 11:18.40 | k-man | has the format of extensions.conf changed between 1.4 and 1.6? |
| 11:19.32 | k-man | i don't understand the reason for this messgae "handle_request_invite: Failed to authenticate device "0414461104"<sip:0414461104@125.213.160.81:5060>;tag=34888250-co3008-INS001" as that is my mobile number that I am calling in from |
| 11:22.20 | k-man | ah, its something to do with insecure=very |
| 11:22.59 | *** join/#asterisk spenguin[work] (~penguin@59.162.86.164) |
| 11:23.08 | *** join/#asterisk Bryanstein (bryan@shellium/admin/bryanstein) |
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| 11:28.25 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 11:49.34 | *** join/#asterisk rare1980_ (~rare1980@pat.telespectrum.com) |
| 11:49.51 | rare1980_ | hi all .. i need some info on asterik AMI... |
| 11:53.30 | tuxx- | yay, we found the bug |
| 11:53.31 | tuxx- | :-) |
| 11:53.45 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
| 11:53.55 | tuxx- | when dtmf = rfc2833 canreinvite=yes doesnt work |
| 11:53.55 | tuxx- | :) |
| 11:58.10 | tuxx- | edited voip-info.org with the dtmfmode information |
| 11:58.11 | tuxx- | yay |
| 11:58.12 | tuxx- | \o\ /o/ |
| 11:58.14 | TommyBotten | tuxx-: What?? |
| 11:58.20 | TommyBotten | Ah... naturally |
| 11:58.47 | tuxx- | when you come to think of it, its pretty logic |
| 11:58.51 | tuxx- | the only problem was finding it |
| 11:58.52 | tuxx- | :-) |
| 12:03.45 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 12:05.57 | *** join/#asterisk rare1980_ (~rare1980@12.25.228.67) |
| 12:07.37 | *** join/#asterisk rossand (~aross@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 12:09.20 | Toommi | <PROTECTED> |
| 12:09.41 | *** join/#asterisk Skeeter- (skeeter@190-141.cgocable.ca) |
| 12:11.04 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 12:11.31 | *** join/#asterisk muiro (~muiro@unaffiliated/muiro) |
| 12:11.51 | Toommi | and another issue, that i googled allready without any success when my agi script send a print there comes this error : ERROR[3888]: utils.c:1128 ast_carefulwrite: write() returned error: Broken pipe |
| 12:12.00 | Toommi | print example: print 'EXEC AddQueueMember '.(string) $queue.',local/'.$callerId.'@queue-call-phone ' . "\n"; |
| 12:12.09 | Toommi | in php^^ |
| 12:12.13 | Toommi | 5.3.2 |
| 12:15.18 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 12:16.59 | *** join/#asterisk Fs0L (~Fs0L@136.223.19.60) |
| 12:17.50 | Fs0L | am looking to play around with asterisk and freepbx for my home again (through my HT-503 against the wall after playing with it for awhile... firmware upgrade after firmware upgrade to fix issues) |
| 12:18.03 | Fs0L | and I had moved from trixbox to asterisknow in the past |
| 12:18.27 | Fs0L | would like to use asterisknow again. But was wondering if any of the bugs that were found after the initial release were updated in a newer release |
| 12:18.46 | Fs0L | does anyone know if there is a development iso available for asterisknow |
| 12:19.04 | Fs0L | possibly with 1.6 on it? |
| 12:21.11 | kaldemar | someone at #asterisknow might know better |
| 12:21.30 | Fs0L | ahh... didn't know that existed. I'll try that |
| 12:21.30 | Fs0L | thanks |
| 12:24.35 | *** join/#asterisk ManxPower-work (~manxpower@216.186.151.147) |
| 12:28.39 | Toommi | any ideas on my problem?^^ |
| 12:29.45 | *** join/#asterisk pentanol (~pentanol@77-35-13-226.pppoe.primorye.net.ru) |
| 12:29.50 | *** join/#asterisk e-jones (~jkastner@nat/redhat/x-yboltkknbkaannmc) |
| 12:34.27 | Katty | hi |
| 12:34.46 | Toommi | hu |
| 12:38.30 | *** join/#asterisk contrabanda (~contr@188.123.128.2) |
| 12:38.34 | contrabanda | Helloo |
| 12:39.36 | Toommi | hu |
| 12:39.44 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 12:40.43 | contrabanda | I have 2 E1 cards. 1- Sangoma on slot 1 and 2 - Digium on slot 2. I would like Sangoma to work on ss7 and digion with PRI. I have installed libss7 to support ss7 on dahdi channels. can you please tell where i have to make separate configs for this devices? in chan_dahdi.conf ? /etc/dahdi/system.conf ? |
| 12:41.00 | *** join/#asterisk Bryanstein (~bryan@shellium/admin/bryanstein) |
| 12:42.13 | [TK]D-Fender | contrabanda: its just more channels and spans... look at any multi-port card config... same thing when adding more. Find out which device is loaded first and set the channels up appropriately |
| 12:46.36 | Toommi | any ideas on this error: "utils.c:1128 ast_carefulwrite: write() returned error: Broken pipe" when agi extecuts for example: print 'EXEC Playback agent-loggedoff ' . "\n"; |
| 12:50.56 | *** join/#asterisk hipitihop (~denis@203.132.229.18) |
| 12:52.08 | ManxPower-work | Toommi, It sounds like your script is not following the AGI spec by reading it's STDIN and processing that. |
| 12:52.20 | ManxPower-work | Toommi, Are you using an AGI library? |
| 12:52.38 | Toommi | nope |
| 12:52.46 | ManxPower-work | that would be it then |
| 12:52.47 | Toommi | i pastebin it mom |
| 12:52.48 | rare1980_ | hi all ... i need some info on asterisk AMI.. |
| 12:53.04 | ManxPower-work | Toommi, don't bother. I'll see if I can find the AGI spec. |
| 12:53.30 | rare1980_ | i want to make call using 3rd party software... i will want pass call using 3rd party calls to asterisk AMI.. |
| 12:53.42 | rare1980_ | i need to know what commands i can use? |
| 12:53.47 | rare1980_ | any help on this please? |
| 12:54.18 | ManxPower-work | rare1980_, a simple google search would have given you the info you are looking for. |
| 12:54.46 | ManxPower-work | ~answers |
| 12:54.47 | infobot | answers is, like, Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 12:54.56 | ManxPower-work | rare1980_, check the Wiki |
| 12:55.23 | ManxPower-work | Toommi, did you read the AGI info on voip-info.org. Why are you not using an AGI library? |
| 12:57.22 | rare1980_ | ManxPower-work: well i just need to know that is can be done through asterisk AMI?? |
| 12:57.28 | rare1980_ | rite? |
| 12:57.46 | Toommi | well i am just begun to build this pbx and wanted to put the logic outside in a dynamic language that i can controlle, so i just startet to get to know how agi works so i just tried a few comments but i alawys geht the error msg, but my script works perfectly besides that |
| 12:58.29 | ManxPower-work | rare1980_, You did not provide neough information to tell you that. |
| 12:59.11 | *** join/#asterisk jtodd (cgq7dvan44@asterisk/community-director-and-tie-dye-shirt-lover/jtodd) |
| 12:59.11 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 12:59.16 | Toommi | here is my script at this time: http://pastebin.com/0SJvxZdH it is just technical preview |
| 12:59.25 | ManxPower-work | Toommi, did you read the AGI info on voip-info.org |
| 12:59.29 | Toommi | yes |
| 13:00.17 | [TK]D-Fender | Silly errors and design failures |
| 13:00.29 | Toommi | and i read the agi chapter in the german "asterisk buch" |
| 13:00.38 | Toommi | with his examples i get the same errors :/ |
| 13:01.03 | [TK]D-Fender | Toommi: then maybe you shouldn't be following that site <- |
| 13:01.42 | rare1980_ | ManxPower-work: basically i have a predictive dialer on windows. which is based on intel dialogic .. now wht i want to remove dialogic and use SIP through asterisk ... now wht is want... i want to make predictive calls through my predictive dialer and pass those calls to asterisk AMI.. asterisk will dial those numbers using dial plan |
| 13:01.45 | Toommi | well i guess he has a lot of knowledge, it the maker of "gemeinschaft" if you know it ;) |
| 13:02.31 | [TK]D-Fender | rare1980_: Go read up on "AMI Originate" |
| 13:02.49 | rare1980_ | humm.. ok thanks |
| 13:02.54 | [TK]D-Fender | Toommi: Either his samples are bad.. or your adaptation of them is |
| 13:03.21 | rare1980_ | wht is differnce between asterisk AMI and asterisk AGI |
| 13:04.18 | [TK]D-Fender | rare1980_: NOTHIGN to do with each other |
| 13:04.43 | rare1980_ | rite |
| 13:04.47 | Toommi | [TK]D-Fender getting the same error with the samples provied by asterisk :P |
| 13:05.08 | [TK]D-Fender | rare1980_: AGI = external processing of a call instead of straight dialplan. AMI = random non call-specific * server manipulations |
| 13:05.31 | [TK]D-Fender | Toommi: Show me where you're reading .. |
| 13:05.44 | *** join/#asterisk socain (~socain00@74.255.249.66) |
| 13:07.04 | Toommi | dont getting you |
| 13:08.23 | [TK]D-Fender | TommyBotten: show me the site with the broken sample |
| 13:09.06 | *** join/#asterisk ManxPower-work (~manxpower@216.186.151.147) |
| 13:09.08 | *** part/#asterisk Fs0L (~Fs0L@136.223.19.60) |
| 13:09.08 | [TK]D-Fender | [09:04]<Toommi>[TK]D-Fender getting the same error with the samples provied by asterisk :P <- Provided by "Asterisk" huh? WHere? |
| 13:09.50 | ManxPower-work | Wow, Toommi you're an idiot |
| 13:09.52 | rare1980_ | [TK]D-Fender: short question.. for my task i would be using AMI or AGI? makeing outbound or inbound call using 3rd party software? |
| 13:10.08 | ManxPower-work | Toommi, you do not have an AGI script. |
| 13:10.15 | Toommi | ? |
| 13:10.17 | [TK]D-Fender | [09:02]<[TK]D-Fender>rare1980_: Go read up on "AMI Originate" <- are you even reading what we're telling you? |
| 13:10.48 | rare1980_ | yes i am... |
| 13:10.50 | TommyBotten | [TK]D-Fender: You didn't mean me? |
| 13:10.51 | ManxPower-work | Toommi, your program is not a valid AGI program. |
| 13:11.28 | [TK]D-Fender | TommyBotten: Correct. Autocomplete error from before. Please disregard. |
| 13:11.37 | TommyBotten | :) |
| 13:11.40 | ManxPower-work | Toommi, AGI programs must be written a specifc way. You did not write your program that way. |
| 13:12.01 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
| 13:12.02 | ManxPower-work | Now why don't you stop wasting everyone's time and go read up on how to write an AGI program. |
| 13:12.58 | ManxPower-work | rare1980_, Have you looked up the AMI info in the Wiki? |
| 13:13.04 | ManxPower-work | It's not sounding like you did. |
| 13:13.21 | rare1980_ | Maxpower-work: yes.. |
| 13:13.26 | rare1980_ | i just reading that |
| 13:13.33 | rare1980_ | http://www.voip-info.org/wiki/view/Asterisk+manager+API |
| 13:13.44 | rare1980_ | right now i am goign through this link |
| 13:13.46 | [TK]D-Fender | rare1980_: Show me the "Asterisk" sample you said was "broken". |
| 13:13.50 | Katty | ohai |
| 13:13.50 | ManxPower-work | rare1980_, so you understand that AMI is an interface to control Asterisk. It is not something you "send calls thru" |
| 13:13.56 | [TK]D-Fender | Toommi: rather... |
| 13:14.07 | ManxPower-work | [TK]D-Fender, he's not following the AGI spec |
| 13:14.22 | [TK]D-Fender | ManxPower-work: Oh I know full well where the first glaring error is |
| 13:14.32 | ManxPower-work | http://pastebin.com/0SJvxZdH |
| 13:14.45 | rare1980_ | manxpower-work: yeh i am getting the idea |
| 13:14.48 | ManxPower-work | [TK]D-Fender, he's not reading STDIN before sending stuff to Asterisk for one thing. |
| 13:14.59 | [TK]D-Fender | ManxPower-work: Oh no... far worse than that <- |
| 13:15.09 | *** join/#asterisk highvoltz (rogers@bling.bling.org) |
| 13:15.20 | ManxPower-work | [TK]D-Fender, I stopped at the first "It's obvious THIS one didn't read the docs" problem. |
| 13:16.03 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 13:16.24 | highvoltz | Hey guys, I have a phone thats on a remote subnet over vpn thats able to register and make outbound calls, however the status is "UNREACHABLE" in peer detail. When called it goes directly to voicemail. I checked DND and its not on. Any ideas? |
| 13:16.41 | rare1980_ | guys let me read and i will get back to you .. .thanks for ur help |
| 13:17.22 | ManxPower-work | ~answers |
| 13:17.23 | infobot | rumour has it, answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 13:19.12 | *** part/#asterisk ManxPower-work (~manxpower@216.186.151.147) |
| 13:19.22 | [TK]D-Fender | highvoltz: if its listed as "UNREACHABLE" then * won't even TRY to call it |
| 13:19.50 | highvoltz | what might list it as UNREACHABLE but be able to register? |
| 13:20.03 | *** join/#asterisk anonymouz666 (~anonymouz@189.24.87.110) |
| 13:20.06 | [TK]D-Fender | highvoltz: "qualify" failure |
| 13:20.29 | *** join/#asterisk ddefrenne (~ddefrenne@91.176.10.251) |
| 13:20.43 | norrec | is there a way to get asterisk to register via another port? |
| 13:20.56 | [TK]D-Fender | highvoltz: it could register. Then respond OK to which * doesn't necessarily expect an answer and then simply never know that outbound packets aren't routed right at all. |
| 13:21.02 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 13:21.23 | [TK]D-Fender | norrec: ":1234" after the host in your register statement |
| 13:21.36 | norrec | ah, thanks |
| 13:22.07 | Toommi | ManxPower-work: http://pastebin.com/YZ3LrpA2 is this "agi" script also not written in the specific way you meant, and yes i read the the informations of voip info |
| 13:22.13 | highvoltz | hmm weird. should I try turning off qualify? |
| 13:22.24 | Toommi | oh he leavt :X |
| 13:22.31 | highvoltz | I have another locaiton, exact type of setup with same hardware and its working fine with the same settings |
| 13:23.26 | norrec | [TK]D-Fender: that didnt seem to work... its still registering from 5060 |
| 13:23.37 | *** join/#asterisk ManxPower-work (~manxpower@216.186.151.147) |
| 13:24.15 | [TK]D-Fender | norrec: You want * to not listen on 5060, but rather something else? |
| 13:25.28 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 13:25.41 | highvoltz | I'm also seeing in peer details its using a weird port, 2069 - all the others are using 2051 and 5060 |
| 13:25.47 | [TK]D-Fender | Toommi: Apparently your ability to mimc that scrip failed as of the FIRST LINE |
| 13:25.47 | norrec | [TK]D-Fender: no, i want asterisk to register from, for example, 5061 to 5060 |
| 13:26.15 | [TK]D-Fender | norrec: I'm not sure I get your sample of 2 ports... |
| 13:26.27 | [TK]D-Fender | norrec: whats wrong with th resister it sends out? |
| 13:26.39 | [TK]D-Fender | mimic* |
| 13:26.47 | darkdrgn2k | morning guys |
| 13:26.49 | norrec | [TK]D-Fender: ok, well i have 2 asterisk servers, and as a patch to get though the day i need to connect the 2 of them via a sip trunk |
| 13:26.55 | ManxPower-work | Toommi, You MUST read the STDIN input to your script!! |
| 13:27.02 | norrec | however, i need 4 different sip trunks between the same 2 servers |
| 13:27.26 | darkdrgn2k | u have a dialplan |
| 13:27.27 | darkdrgn2k | exten => 18007775509,n,Dial(local/18007775509@outbound,,D(1)) |
| 13:27.29 | norrec | and since they are multicontextual and running on the same ip:port it causes issues |
| 13:27.47 | Toommi | i did but same result at all, script works in the way it meant but there is still this error |
| 13:27.48 | darkdrgn2k | it takes a few seconds to actualy dial it, and the end user hears silence.. |
| 13:27.49 | ManxPower-work | norrec, You should not run multiple asterisk servers on the same machine. |
| 13:28.02 | norrec | its not on the same machine |
| 13:28.03 | darkdrgn2k | i tried adding "exten => 18007775509,n,Ringing" just above it but it doesnt seem to work |
| 13:28.05 | darkdrgn2k | any ideas? |
| 13:28.08 | [TK]D-Fender | norrec: Then you need each running on a different listening port so they can RETURN to the proper server... Also they'll need separate RTP ranges, etc. |
| 13:28.11 | ManxPower-work | Toommi, use the phpagi library |
| 13:28.20 | [TK]D-Fender | norrec: change the bindport |
| 13:28.25 | [TK]D-Fender | ManxPower-work: first line <- |
| 13:28.28 | ManxPower-work | darkdrgn2k, adding "r" or "ringing" almost never helps. |
| 13:28.40 | ManxPower-work | [TK]D-Fender, I'm not going to debug this guy's program. |
| 13:28.53 | norrec | so just do bindport=5061 for the peer right? |
| 13:28.56 | darkdrgn2k | ManxPower-work: any idea how i could accomplis this? |
| 13:29.03 | ManxPower-work | darkdrgn2k, do you answer the call first? |
| 13:29.16 | ManxPower-work | darkdrgn2k, ringing should be heard automatically. |
| 13:29.27 | darkdrgn2k | ManxPower-work: yes the call is already answered |
| 13:29.36 | ManxPower-work | darkdrgn2k, why? |
| 13:29.43 | darkdrgn2k | ManxPower-work: becuase this call is comming from another context |
| 13:29.48 | ManxPower-work | You should never answer a call unless you MUST. |
| 13:29.48 | darkdrgn2k | (ivr) |
| 13:30.07 | ManxPower-work | darkdrgn2k, post answer ringback is handled by /etc/asterisk/indications.conf |
| 13:30.33 | [TK]D-Fender | Toommi: Youre script is bad from the very first LINE in it. |
| 13:30.41 | [TK]D-Fender | Toommi: Now go COMPARE them. |
| 13:30.43 | ManxPower-work | pre=answer indications are handled by the SIP device itself. |
| 13:31.09 | [TK]D-Fender | norrec: No... you need asterisk to listen on a different port altogether... the peer has nothing to do with registering. |
| 13:31.28 | darkdrgn2k | ManxPower-work: ok but how can i synthesize ringing, or even play an audio click WHILE its trying to connect |
| 13:31.35 | darkdrgn2k | (dead air sounds like somethings wrong) |
| 13:31.56 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 13:32.16 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:32.16 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:32.23 | Katty | hi leif |
| 13:32.32 | leifmadsen | hey! |
| 13:33.03 | Katty | let's hug |
| 13:33.05 | Katty | hugs leifmadsen |
| 13:33.10 | leifmadsen | hugs back! |
| 13:34.15 | ManxPower-work | darkdrgn2k, fix your /etc/asterisk/indications.conf |
| 13:34.43 | darkdrgn2k | ManxPower-work: it looks like indications just holds what kinda of ring to present, not any contexts |
| 13:35.03 | darkdrgn2k | ManxPower-work: i dont understand how it has anything to do with initialiting ring-like sounds on the line |
| 13:35.04 | ManxPower-work | darkdrgn2k, indications.conf specifics off hook audio indications. |
| 13:35.12 | ManxPower-work | ring, busy, ringback, etc. |
| 13:35.24 | ManxPower-work | darkdrgn2k, do you or do you not have a /etc/asterisk/indications.conf? |
| 13:35.54 | ManxPower-work | argueing with me will not change how Asterisk works. |
| 13:37.42 | Polysics | ManxPower-work, quote of the week right there :-) |
| 13:37.49 | ManxPower-work | Ah! I see you were on my /ignore list. |
| 13:42.22 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 13:42.54 | darkdrgn2k | ManxPower-work: im not argueing just dont understand. and yes i do hae one |
| 13:42.57 | darkdrgn2k | (soryr phone) |
| 13:43.22 | ManxPower-work | darkdrgn2k, then there must be something else wrong. |
| 13:43.38 | ManxPower-work | lacking /etc/asterisk/indications.conf is the most common reason for not getting ringback. |
| 13:43.52 | ManxPower-work | By far the MOST common reason. |
| 13:44.03 | darkdrgn2k | ManxPower-work: im sorry, there may be some miscommunication |
| 13:44.17 | [TK]D-Fender | [09:31]<darkdrgn2k>ManxPower-work: ok but how can i synthesize ringing, or even play an audio click WHILE its trying to connect <- wHERE IS THE failed ATTEMPT FOR US TO LOOK AT? |
| 13:44.33 | *** join/#asterisk lordoxide (~chatzilla@206.183.2.183) |
| 13:44.37 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 13:44.41 | darkdrgn2k | ManxPower-work: normaly i DO hear ringing when i do dia() under other situations |
| 13:44.48 | darkdrgn2k | exten => 18007775509,n,Dial(local/18007775509@outbound,,D(1)) |
| 13:44.59 | darkdrgn2k | when i dial that i get pause pause "he" pause pause Ring |
| 13:45.15 | darkdrgn2k | (he from the HELLOW that is said just before 1 gets pushed) |
| 13:45.25 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 13:45.48 | ManxPower-work | darkdrgn2k, when [TK]D-Fender asks for the failed call attempt, he's not asking for a single line. He's asking for the cli output of a failed call. |
| 13:45.49 | darkdrgn2k | from my understainding D(1) is sopposed to dial 1 BEFORE it bridges that call, no? |
| 13:46.06 | ManxPower-work | darkdrgn2k, remove the extra options for debugging |
| 13:46.20 | darkdrgn2k | ManxPower-work: sorry i missed that. but technically the call back does not FAIL |
| 13:46.37 | darkdrgn2k | ManxPower-work: it is simply dead air for about 5 or 7 seconds with a "he" in the middle |
| 13:46.44 | ManxPower-work | darkdrgn2k, You do not know enough to know what we need to see. |
| 13:46.56 | darkdrgn2k | the call ultimatly suceeds |
| 13:46.58 | ManxPower-work | now, either pastebin the output of a problem call or stop wasking our time. |
| 13:47.26 | ManxPower-work | darkdrgn2k, and yet, chances are the info we need to solve the issue may be in the pastebin info that we have not seen yet. |
| 13:48.43 | darkdrgn2k | one moment loging call to paste |
| 13:49.02 | lordoxide | sup all, I need a little asterisk help, I relatively new to asterisk. I have a sip provider which gives us 2 trunks for sip.conf (link2voip), both are setup and working. They gave a macro for extensions.conf which will try trunk2 then back to trunk1 if there is congestion. My only problem is if I'm trying to originate a call via a manager login, I cant get it use the macro for dialing. I'm... |
| 13:49.03 | lordoxide | ...sure what the proper syntax for the Channel Line is, "Channel: SIP/1xxxxxxxxxx@trunk1 works, but how do I use the dialing context instead? |
| 13:49.37 | *** join/#asterisk geneticx_wrk (~geneticx_@host-208-88-126-198.biznesshosting.net) |
| 13:50.49 | kaldemar | lordoxide: use a local channel, like Local/exten@context |
| 13:51.22 | lordoxide | i've tried, that works with call files but with the php script accessing the manager it was not =) im trying again now tho |
| 13:52.40 | *** join/#asterisk Circlefusion (~circlefus@74-132-116-73.dhcp.insightbb.com) |
| 13:53.01 | lordoxide | well now it works, so i don't care why it didnt before, thanks kaldemar =) |
| 13:53.08 | darkdrgn2k | Log -> http://pastebin.ca/1852860 |
| 13:53.40 | darkdrgn2k | i understand this is not a support for freepbx, my question originated from a custom dialplan that was not part of freepbx |
| 13:54.05 | ManxPower-work | darkdrgn2k, you are on your own. |
| 13:54.30 | ManxPower-work | Now if you want to create a simple diaplan without a gazillion lines of FreePBX crap, then I might change my mind. |
| 13:54.45 | darkdrgn2k | ManxPower-work: i figured as much, i guess i have to build a seperate astrisk box, with only my custom dial plan and come back to try to resolve it before i try to integrate it with the freepbx krap :) |
| 13:55.11 | ManxPower-work | darkdrgn2k, *nod* FreePBX is much more complicated than regular Asterisk. |
| 13:55.20 | darkdrgn2k | ManxPower-work: belevae me i know that |
| 13:55.39 | darkdrgn2k | ManxPower-work: im trying to learn dialplans.. but i cant remove myself from freepbx just yet.. hopefully soon |
| 13:55.42 | anonymouz666 | I am worried... that FreePBX crap keeping growing and growing. The -users list is totally infected |
| 13:56.32 | darkdrgn2k | anonymouz666: i agrea... hence me being emberased to even show that im using it :-S |
| 13:56.54 | darkdrgn2k | i was hopeing the answer was something as silly as "s,1,ring-user()" lol |
| 13:57.17 | [TK]D-Fender | darkdrgn2k: At wahat point to does the silence start, and where does it end? Why isn't SIP DEBUG included? |
| 13:59.01 | darkdrgn2k | oops |
| 13:59.05 | darkdrgn2k | its hangs at 70 |
| 14:01.01 | ManxPower-work | darkdrgn2k, you won't learn dialplans by using the FreePBX dialplan |
| 14:01.11 | *** join/#asterisk linuxcentos (~linuxcent@rhelbox.uio.no) |
| 14:01.20 | ManxPower-work | anonymouz666, I know. I'm close to abanding #asterisk and asterisk-users |
| 14:01.43 | darkdrgn2k | ManxPower-work: yes and no.... i do get to playu with some basing ideas on a live system... |
| 14:01.47 | ManxPower-work | darkdrgn2k, no, it is as simple as Dial(TECH/peer/dest) |
| 14:02.13 | [TK]D-Fender | darkdrgn2k: And you aren't looking at a relevant leg of this call. |
| 14:02.25 | *** part/#asterisk ManxPower-work (~manxpower@216.186.151.147) |
| 14:02.35 | darkdrgn2k | [TK]D-Fender: which would be |
| 14:03.05 | [TK]D-Fender | [09:57]<[TK]D-Fender>darkdrgn2k: At wahat point to does the silence start, and where does it end? Why isn't SIP DEBUG included? |
| 14:03.23 | darkdrgn2k | im pasting the sip debug |
| 14:03.26 | darkdrgn2k | and it starts at 70 |
| 14:03.34 | *** join/#asterisk coppice (~chatzilla@202.62.81.147) |
| 14:03.48 | Katty | hi coppice |
| 14:03.50 | darkdrgn2k | debug -> http://pastebin.ca/1852864 |
| 14:03.56 | darkdrgn2k | from line 70 sip debug |
| 14:06.03 | *** join/#asterisk Bryanstein (bryan@shellium/admin/bryanstein) |
| 14:07.02 | darkdrgn2k | it seems the the delay is from ME and the remote number. SO i guess i would need to find a way to play an audio file fofr x number of seconds WHILE the call was being made |
| 14:10.50 | gego | I'm trying to get the fax application working for me in * 1.6.2 ( interface is BN8S0 mISDN/lcr ) |
| 14:10.57 | gego | it gives this error: app_fax.c:337 fax_generator_generate: Only generating 240 samples, where 256 requested |
| 14:12.39 | gego | does anyone know what it means? can it be caused by a missing time source (since the test machine is not connected to PSTN) ? |
| 14:12.43 | *** part/#asterisk bsaxon (~bsaxon@12.68.234.174) |
| 14:14.30 | [TK]D-Fender | darkdrgn2k: sadly incomplete |
| 14:14.57 | *** join/#asterisk wcselby (~wcselby@216.110.88.194) |
| 14:15.21 | wcselby | o/ |
| 14:17.50 | Katty | glomps wcselby |
| 14:19.12 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:19.12 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:19.40 | darkdrgn2k | [TK]D-Fender: You know this is has gone into a whole debug of the dial plan and frankly i dont want to put anyone throuhg a debug of the FREEPBX dial plan. ITs an inconvenance but it works. thank you for tryining im just gonna live with it :) |
| 14:20.21 | [TK]D-Fender | darkdrgn2k: Who said the problem was the DIALPLAN? |
| 14:22.19 | *** join/#asterisk eppigy (~eppigy@c-69-180-16-188.hsd1.ga.comcast.net) |
| 14:22.32 | [TK]D-Fender | eppigy: HELLO YOU ARE DAVE! |
| 14:23.13 | *** join/#asterisk dennisG (~visionlab@84.30.136.208) |
| 14:23.33 | *** join/#asterisk Bryanstein (bryan@shellium/admin/bryanstein) |
| 14:24.16 | eppigy | HELLO |
| 14:24.17 | eppigy | YES |
| 14:24.20 | eppigy | IT IS TRUE |
| 14:25.03 | Katty | hugs eppigy |
| 14:26.32 | wcselby | what's a glomp? |
| 14:26.49 | [TK]D-Fender | wcselby: Like a blorp, only different |
| 14:27.05 | wcselby | ooh |
| 14:27.07 | wcselby | okay |
| 14:27.15 | darkdrgn2k | [TK]D-Fender: sory the problem is my laggy provider.. i just confirmed that... takes 6 seconds to connect! |
| 14:27.32 | darkdrgn2k | [TK]D-Fender: so now the question is, can i play an audio clip for a few seconds during the lag |
| 14:29.34 | [TK]D-Fender | darkdrgn2k: Dial with "r" option. |
| 14:30.04 | [TK]D-Fender | darkdrgn2k: And I'm nt going to clami "laggy" as a culprit here |
| 14:32.21 | darkdrgn2k | [TK]D-Fender: If i was not a guy i could kiss you1 |
| 14:32.33 | *** join/#asterisk mikkel (~mikkel@84-238-113-66.u.parknet.dk) |
| 14:32.37 | Katty | wcselby: http://media.photobucket.com/image/how%20to%20glomp/shikigami90001/t059art.jpg <- How To Glomp |
| 14:33.11 | darkdrgn2k | [TK]D-Fender: ill debug the "lag" when i get rid of freepbx.. really dont want to put any one through that :-P |
| 14:34.08 | wcselby | Katty - so I'm the bishi? and that makes me dead by the end of the description..... |
| 14:34.12 | wcselby | :P |
| 14:35.28 | Katty | >.< |
| 14:36.18 | Katty | has anyone tried calling i9technologies lately? |
| 14:36.23 | Katty | all i ever seem to get is their voicemail. |
| 14:36.30 | Katty | no one's returning my calls :< |
| 14:36.48 | Katty | and it's not even for support! |
| 14:37.04 | p3nguin | You're calling sales? |
| 14:37.09 | Katty | yes |
| 14:37.18 | p3nguin | You'd think they'd want to take that call. |
| 14:37.22 | Katty | i know, right? |
| 14:37.34 | Katty | infobot: seen seanmh |
| 14:37.35 | infobot | seanmh <n=johndoe@207.114.199.107> was last seen on IRC in channel #asterisk, 164d 19h 15m 54s ago, saying: 'Katty: how's the 1.6 testing going?'. |
| 14:37.41 | Katty | ^- that's also a bad sign. |
| 14:37.46 | p3nguin | long time |
| 14:38.02 | Naikrovek | 164d? wow |
| 14:41.29 | *** join/#asterisk ghenry (~ghenry@pdpc/supporter/monthlybyte/ghenry) |
| 14:41.32 | ghenry | Weird |
| 14:41.38 | ghenry | one of our customers got hacked into |
| 14:41.43 | ghenry | Lots of calls to 14112800043820919507 |
| 14:41.49 | ghenry | We don't allow anon sip calls |
| 14:41.54 | ghenry | sip is blocked from outside |
| 14:42.01 | *** part/#asterisk benngard (~benngard@213.88.138.230) |
| 14:42.02 | ghenry | all extension have very good passwords |
| 14:42.08 | ghenry | using apg |
| 14:42.11 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 14:42.41 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 14:43.42 | Katty | ghenry: we had that happen once too |
| 14:43.53 | Katty | ghenry: ended up locking down 5060 to only particular connecting, static, ip addresses |
| 14:43.58 | Dovid | ghenry: do you see the calls on your system or on ur bill |
| 14:44.36 | Katty | i'm sure he just sees it on his call logs, since he has sip calls blocked |
| 14:45.13 | ghenry | on system |
| 14:45.14 | ghenry | 00923224255617 |
| 14:45.17 | ghenry | india |
| 14:45.19 | ghenry | how did they get in? |
| 14:45.28 | ghenry | 00923234898085 |
| 14:45.36 | ghenry | lots of this 14112800043820919507 |
| 14:47.02 | Katty | check your firewall logs |
| 14:47.10 | ghenry | yeah |
| 14:48.13 | *** join/#asterisk Netgeeks (~chris@173.11.68.155) |
| 14:50.49 | *** join/#asterisk rttrey (~trey@209.208.18.121) |
| 14:52.00 | *** join/#asterisk devoid (~devnull@unaffiliated/devemo) |
| 14:56.08 | *** join/#asterisk jpeeler (~jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:57.26 | ghenry | http://paste.scsys.co.uk/41357 |
| 14:58.34 | ghenry | that's the logs |
| 15:01.57 | [TK]D-Fender | ghenry: Looks like a ZAP device placed those calls... got a DISA you left poorly secured? |
| 15:02.18 | [TK]D-Fender | ghenry: or a poorly laid out dialplan? |
| 15:02.20 | ghenry | nope |
| 15:02.26 | ghenry | PBXinAFlash with fail2ban on |
| 15:02.32 | ghenry | anon sip not allowed |
| 15:02.37 | ghenry | sip blocked at firewall |
| 15:02.39 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 15:02.50 | ghenry | inbound routes goes to day night mode or voicemal |
| 15:03.10 | [TK]D-Fender | ghenry: 2010-03-26 02:36:48 Zap/2-1... UNAVAILABLE 14112800043820919507 ANSWERED 22:10 <--- where do I see **SIP** in this? |
| 15:03.31 | ghenry | you dont ;-) |
| 15:04.06 | wcselby | ghenry - did the calls all go out from a single extension? |
| 15:04.11 | ghenry | nope |
| 15:04.56 | wcselby | were they from sip extensions? |
| 15:05.05 | ghenry | nope |
| 15:05.59 | wcselby | then why are you worried about sip? |
| 15:07.36 | ghenry | juset trying to work things out |
| 15:07.49 | ghenry | 102. 2010-03-23 17:41:25 Zap/3-1... WITHHELD 0034673259966 ANSWERED 12:31 |
| 15:07.56 | ghenry | Has to be a voicemail hack |
| 15:08.10 | ghenry | We have a mobile callerid |
| 15:08.51 | ghenry | Voicemail unavailable kicks in after 5.30 ish |
| 15:09.00 | ghenry | then goes to vm unavaible |
| 15:09.05 | ghenry | some hack there |
| 15:09.08 | ghenry | check logs |
| 15:09.24 | ghenry | Can you get an internal line via central voicemail? |
| 15:09.47 | wcselby | if your dialplan is bad, I suppose so...? |
| 15:10.09 | wcselby | there's two options that I know of to exit the voicemail app, pressing zero or *. |
| 15:10.19 | ghenry | this is freepbx |
| 15:10.31 | wcselby | your dialplan should have catches for these, and I think voicemail.conf defines whether they're able to use them or not, and where to send them |
| 15:10.52 | p3nguin | ~freepbx |
| 15:10.52 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 15:10.54 | ghenry | ckeching |
| 15:11.02 | ghenry | ta |
| 15:12.51 | *** join/#asterisk flapjacks (~flapjacks@wsip-70-166-201-90.ph.ph.cox.net) |
| 15:12.52 | *** join/#asterisk SuPrSluG (~SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 15:14.13 | freezey | hey for some reason i am getting all circuits are busy when doing a dundi transfer.. it was working earlier but not now |
| 15:14.58 | *** join/#asterisk Obeliks (obeliks@gentoo/contributor/Obeliks) |
| 15:16.51 | *** join/#asterisk moy (~chatzilla@74.12.121.207) |
| 15:16.56 | bmoraca_work | freezey: the temperature in Spain dropped by 2 degrees last night causing your issue. |
| 15:18.29 | *** join/#asterisk Pimmetje (~Pimmetje@83.119.156.35) |
| 15:19.31 | *** join/#asterisk Tim_Toady (~moi@77.49.45.81.dsl.dyn.forthnet.gr) |
| 15:20.39 | chazzam | bofh? |
| 15:22.44 | Naikrovek | bofh. |
| 15:23.41 | Katty | ^- bastard operator from hell |
| 15:27.01 | Katty | boggles |
| 15:27.12 | *** join/#asterisk kartik (~koolkarti@117.199.120.76) |
| 15:28.35 | coppice | Katty: AT&T? T-Mobile? |
| 15:31.00 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 15:31.38 | ghenry | I just managed to dial out via voicemail |
| 15:31.46 | ghenry | when you get to voicemail |
| 15:31.50 | adnc | is it possible within the cdr records to see if it was an incomming or an outgoing call? |
| 15:31.59 | ghenry | hit 0 but dial an external numebr right away |
| 15:32.05 | ghenry | and it passes it out |
| 15:33.05 | Katty | coppice: buwha? |
| 15:33.08 | Katty | coppice: i use sprint. |
| 15:36.23 | Katty | coppice: i was boggling over my isymphony config giles |
| 15:36.25 | Katty | giles? |
| 15:36.30 | Katty | sighs |
| 15:39.56 | wcselby | ghenry - let the folks at freepbx know what you've found |
| 15:40.04 | wcselby | or whatever you were using again |
| 15:40.36 | ghenry | yup |
| 15:40.42 | *** join/#asterisk Lord-Rahl (~quassel@173-162-45-177-michigan.hfc.comcastbusiness.net) |
| 15:41.34 | Lord-Rahl | ? for someone I trying to ring and ext but if they do answer go to next step not to voice mail. Is there a way to do that? |
| 15:42.02 | Lord-Rahl | do not answer* |
| 15:42.41 | [TK]D-Fender | Lord-Rahl: Yes. Dial. Then call voicemail. The end. |
| 15:42.54 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 15:42.57 | p3nguin | lord-rahl: You mean you want to call exten 100 and SIP/stan will ring for a while, then go to voicemail if there is no answer? |
| 15:44.43 | patrb | Lord-Rahl: dont forget to save Kahlan |
| 15:45.32 | Lord-Rahl | p3nguin: I do not want it to go to voicemail I want to goto next step in the dial plan. I am under the impression that is a dial sip/stan if it fails on time out asterisk send it to voicemail automatically |
| 15:46.19 | p3nguin | lord-rahl: If it goes to voicemail, the next step in the dialplan is probably Voicemail(). |
| 15:46.22 | Lord-Rahl | p3nguin: Do I need to send an over ride command to stop it from doing that? |
| 15:46.36 | p3nguin | no, just take out the Voicemail() line. |
| 15:46.46 | p3nguin | Dialplan 101 stuff. |
| 15:46.58 | [TK]D-Fender | Lord-Rahl: Your dialplan does what YOU tell it to. There is no such thing as "automatic" |
| 15:47.15 | [TK]D-Fender | Lord-Rahl: if it goes to VM, that's because you told it to |
| 15:47.33 | p3nguin | At least because someone told it to. |
| 15:47.39 | *** join/#asterisk Akiraa (~Akiraaaa@79.112.32.97) |
| 15:47.40 | p3nguin | He's probably using a GUI. |
| 15:48.09 | Lord-Rahl | p3nguin: yep macro's suck |
| 15:48.17 | [TK]D-Fender | Lord-Rahl: No, they don't |
| 15:48.32 | Lord-Rahl | true |
| 15:48.40 | [TK]D-Fender | Lord-Rahl: they do exactly what you told them to do and are meant to save you repeat code where the flow can profit from it |
| 15:49.45 | Lord-Rahl | i need to learn them |
| 15:49.45 | *** join/#asterisk imcdona (imcdona@173.160.189.69) |
| 15:50.42 | [TK]D-Fender | Lord-Rahl: Macro is a single dialplan app... "core show application macro". Extremely little to know |
| 15:52.43 | *** join/#asterisk RobH (~robh@cpe-173-169-30-118.tampabay.res.rr.com) |
| 15:53.49 | *** join/#asterisk Faithful (~Faithful@202.6.145.116) |
| 15:54.10 | Lord-Rahl | [TK]D-Fender: take this ' exten=s,1,Set(__DYNAMIC_FEATURES=${FEATURES})' what is _DYnamic_Features mean I know it a function but how do find this function and what it does |
| 15:54.23 | p3nguin | syntax error |
| 15:54.37 | Lord-Rahl | voip-info or something |
| 15:55.33 | [TK]D-Fender | Lord-Rahl: it is not a feature. its a var used by features.conf |
| 15:55.46 | [TK]D-Fender | Lord-Rahl: to determine DTMF functionality for a given call |
| 15:56.03 | [TK]D-Fender | function* |
| 15:56.45 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
| 15:57.28 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 15:57.54 | *** join/#asterisk andres833 (~andres833@190.144.139.78) |
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| 15:59.30 | flapjacks | can anyone think of what when I park a call im not hear the extension where the call was parked. Yes in the CLI i see it is playing the sound files |
| 16:00.53 | [TK]D-Fender | flapjacks: If you did a BLIND transfer instead of an ATTENDED one like you're supposed to |
| 16:01.02 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.157) |
| 16:02.07 | *** join/#asterisk hmmhesays (~hmmhesays@24-116-107-203.cpe.cableone.net) |
| 16:02.16 | p3nguin | When you do a blind transfer, your side of the call ends as soon as the phone number sends. There wouldn't be time to hear the spot where you parked it. |
| 16:02.25 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 16:03.26 | flapjacks | thanks guys, new it had to be user error |
| 16:11.38 | *** join/#asterisk asteriskATmarmuD (~mundt@193.158.65.23) |
| 16:12.49 | asteriskATmarmuD | what would be the best solution to get info in the dialed number(s) transferred to another server via LAN (using channel variables and AGI?) |
| 16:14.26 | ariel_ | transferred how? |
| 16:15.59 | asteriskATmarmuD | don't know... at least over the local network |
| 16:16.41 | asteriskATmarmuD | there is another server which needs to get the status of the "connections" |
| 16:17.06 | asteriskATmarmuD | and take immediate action depending on the connection/channel status |
| 16:17.36 | asteriskATmarmuD | I'm thinking of messages like (number, status) for exmaple (36363636, busy) |
| 16:18.01 | ariel_ | still don't understand if you transfer a call to another box it is via sip, iax2 or what ever. but you can just correct the info it sends via the dial plan |
| 16:18.13 | asteriskATmarmuD | no, sorry |
| 16:19.18 | asteriskATmarmuD | I don't want to do anything with the connections... only if the state of a connection changes from dialing to connected or busy or answered/not answered |
| 16:19.40 | asteriskATmarmuD | the other system should be informed immediately |
| 16:19.54 | *** join/#asterisk rare1980_ (~rare1980@115.186.24.103) |
| 16:20.13 | ariel_ | have a service run on other box to get your status via the ami, or post your status of the call to a DB like mysql and read it from the other system. |
| 16:20.27 | ariel_ | info from one system to other can be had many different ways |
| 16:20.34 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 16:20.45 | asteriskATmarmuD | hmm ok, is that possible without having the "external" system pulling for info? |
| 16:20.51 | wcselby | i think leifmadsen made a presentation about doing that sort of thing with xmpp at last year's astricon |
| 16:21.00 | asteriskATmarmuD | would be nice, if the server running asterisk would "push" the info on change |
| 16:21.23 | ariel_ | it can to a db then you just read them off the db |
| 16:23.38 | rare1980_ | i have assigned privilege read = call,log in manager.conf but i have designed an API on windows so i can see asterisk call events i can connect to askterisk AMI and get dialing results going on asterisk CLI... i can connect to astersk AMI .. but i am not getting all call events on my windows API..... |
| 16:23.55 | rare1980_ | i can get call events result but not all of them.. |
| 16:24.01 | rare1980_ | any knows the reason |
| 16:24.04 | rare1980_ | ? plz help |
| 16:24.31 | *** join/#asterisk ChUbB (~IceChat7@cpc2-aztw12-0-0-cust229.aztw.cable.virginmedia.com) |
| 16:25.42 | rare1980_ | ?????// |
| 16:29.24 | rare1980_ | hello? |
| 16:32.37 | hardwire | rare1980_: please attempt to hold your horses |
| 16:32.45 | *** join/#asterisk andres833 (~andres833@190.144.139.78) |
| 16:32.50 | hardwire | the little boys room is down the hall if you absolutely have to go. |
| 16:33.01 | Slugs_ | lol |
| 16:33.35 | *** join/#asterisk kazaa_lite (~eddie@78-86-126-14.zone2.bethere.co.uk) |
| 16:34.51 | Katty | file: so what do you do if your apple device locks up |
| 16:35.07 | hardwire | hammer |
| 16:36.33 | Katty | weird. |
| 16:36.46 | Katty | i've never seen a device without an off button |
| 16:37.00 | wcselby | the off button on the iphone is the lock button at the top |
| 16:37.05 | wcselby | you hold it down for five-ten secondw |
| 16:37.08 | wcselby | seconds* |
| 16:37.13 | Katty | well...this is an ipod nano |
| 16:37.16 | Katty | any idea where it's off button is? |
| 16:37.19 | wcselby | then you slide across the screen |
| 16:37.19 | Katty | there's a hold button at the top |
| 16:37.24 | *** join/#asterisk Z_God (~julius@schwartzenberg.xs4all.nl) |
| 16:37.30 | wcselby | probably hold that down for a five seconds |
| 16:37.36 | Katty | it does uhh |
| 16:37.38 | Katty | go down |
| 16:37.42 | Katty | it just slides from left to right |
| 16:37.46 | Katty | lock/unlock |
| 16:37.58 | wcselby | http://ipod.about.com/od/tes1/a/turn_off_nano.htm |
| 16:38.14 | *** join/#asterisk pzn (~pzn@pdpc/supporter/active/pzn) |
| 16:38.20 | Katty | erm doesnt' |
| 16:38.41 | p3nguin | That reminds me of a bunch of weirdos posting bad reviews on the Samsumg T260HD because it didn't have a power button on the monitor/TV... said they HAD to use the remote to power it on or off. |
| 16:38.44 | Katty | yeah see? |
| 16:38.45 | *** join/#asterisk fors1 (~forsen@pat-tdc.opera.com) |
| 16:38.46 | Katty | no off button |
| 16:38.48 | Katty | just sleep mode |
| 16:38.59 | Katty | and then the hold will keep it in sleep mode |
| 16:39.00 | wcselby | exactly Katty |
| 16:39.03 | Katty | but it's still not /off/ |
| 16:39.04 | p3nguin | The power button was a touch sensitive spot on the front panel rather than a button. |
| 16:39.10 | pzn | Hi. I have several extensions and several sips (around 50 each). I just want to deny sip/300 to call to extension 304. all other things are allowed. it there a simple way of doing that? |
| 16:39.30 | Katty | i've never seen a device that does have an OFF button |
| 16:39.36 | p3nguin | It even had the regular |O markings on it to indicate that it is a power button. |
| 16:39.37 | Katty | just a sleep button |
| 16:40.07 | Katty | pzn: yes you can check the callerid number before you dial 304 |
| 16:40.21 | Katty | pzn: and if the callerid number is 300, then route it somewhere else |
| 16:40.55 | Katty | pzn: like uhh, blacklist |
| 16:41.12 | pzn | Katty: ok, if caller id is 300, hangup()... but I think callerID can be configured by the user of sip/300 in his device |
| 16:41.25 | p3nguin | Blacklist usually is a common list that you'll be using on multiple extensions... not good for only one extension. |
| 16:41.30 | pzn | Katty: I should match the real user, not called id. |
| 16:41.33 | Katty | pzn: well then you can check it against the user |
| 16:41.51 | p3nguin | You could probably even match against the SIP information. |
| 16:41.54 | Slugs_ | Katty: it says the nano does not have on or off, it has awake and asleep |
| 16:41.59 | Katty | Slugs_: i know |
| 16:42.08 | Katty | Slugs_: which is why i'm so baffled |
| 16:42.16 | Katty | Slugs_: how does this thing NOT have a manual OFF setting >.< |
| 16:42.27 | pzn | ok, I'll search for variables that can have sip information (mainly the username). thanks! |
| 16:42.28 | Slugs_ | drain battery |
| 16:42.28 | Katty | Slugs_: there's not even a battery pack to get at |
| 16:42.31 | wcselby | because it's an apple product |
| 16:42.38 | Slugs_ | hehe |
| 16:42.39 | Katty | dear apple, stop being so confusing. |
| 16:42.42 | wcselby | and apple does things their own way |
| 16:42.52 | p3nguin | SIPCHANINFO(peer) |
| 16:43.05 | pzn | p3nguin: nice! thanks! |
| 16:43.17 | Katty | dear apple, please also stop this nonsense of trying to keep me from moving audio files from my ipod into itunes...you do realize your device is mass storage and i can see the mp3s right??? thx, Katty |
| 16:43.45 | p3nguin | pzn: Combine that with an ExecIf(), and you'll be good to go. |
| 16:43.46 | Slugs_ | i hate itunes more tha bill gates i think |
| 16:44.00 | Katty | well i haven't jailbreaked the ipod yet |
| 16:44.12 | Katty | jailbroke? |
| 16:44.15 | Katty | something like that |
| 16:44.19 | Slugs_ | ; |
| 16:44.22 | wcselby | i like most things about my iphone, but there are a few that makes we think about jailbreaking it |
| 16:44.26 | Katty | snickerdoodled firmware! |
| 16:44.30 | Slugs_ | my iphone is jailbroken/unlocked |
| 16:44.36 | Slugs_ | its the only way to go |
| 16:44.37 | Slugs_ | ;0 |
| 16:44.47 | Katty | well come jailbreak mine |
| 16:44.51 | Katty | i'll buy you lunch |
| 16:44.55 | Slugs_ | its easy |
| 16:45.05 | Katty | if i jailbreak it, will my nike app still run? |
| 16:45.11 | Slugs_ | yep |
| 16:45.15 | Katty | that's good. |
| 16:45.21 | Katty | else the ipod would be officially dead to me |
| 16:45.22 | Slugs_ | all apps still work |
| 16:45.31 | Katty | well, this is more than an app |
| 16:45.42 | Katty | it's a transmitter in the shoe, and a reciever bit that plugs into the ipod |
| 16:46.00 | Slugs_ | yeah it would still work |
| 16:46.23 | Katty | k |
| 16:47.18 | Katty | well maybe you can walk me through it later (= |
| 16:47.37 | Katty | i've put in 4 hours overtime this week and they're tellin me to go home early |
| 16:48.04 | Slugs_ | sure |
| 16:48.14 | Katty | :> |
| 16:49.46 | *** join/#asterisk hwa (~hwa@li117-222.members.linode.com) |
| 16:51.20 | p3nguin | If you put in 4 hours ot, I would expect you to get to leave 6 hours early. |
| 16:51.49 | p3nguin | I guess it doesn't work that way since you didn't actually hit the 40. |
| 16:55.05 | *** part/#asterisk asteriskATmarmuD (~mundt@193.158.65.23) |
| 16:56.12 | Katty | yeah i haven't hit 40 yet |
| 16:59.44 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 17:04.08 | *** part/#asterisk c0rnoTa (~c0rnoTa@178.176.167.140) |
| 17:04.53 | *** join/#asterisk outtolunc (~me@c-67-160-192-210.hsd1.ca.comcast.net) |
| 17:12.35 | *** join/#asterisk Skeeter- (skeeter@c216.218.2-65.clta.globetrotter.net) |
| 17:15.53 | angryuser | can someone recommend me Reliable T.38 provider |
| 17:15.58 | angryuser | ? |
| 17:18.27 | W0OTM | grrrrr |
| 17:18.41 | W0OTM | <PROTECTED> |
| 17:18.41 | W0OTM | <PROTECTED> |
| 17:18.41 | W0OTM | [Mar 26 12:18:23] WARNING[12380]: app_dial.c:1242 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) |
| 17:18.41 | W0OTM | <PROTECTED> |
| 17:18.42 | W0OTM | <PROTECTED> |
| 17:18.57 | p3nguin | ~pb |
| 17:18.58 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 17:19.14 | W0OTM | sry wrong window |
| 17:19.15 | W0OTM | oops |
| 17:22.33 | Slugs_ | W0OTM: pastebin your sip.conf and extension.conf |
| 17:27.01 | *** join/#asterisk aandrade (~aandrade@187.59.13.182) |
| 17:27.48 | *** join/#asterisk k- (~k-@189.205.227.242) |
| 17:27.55 | Nugget | haha. jamie hyneman emoticon: \:€ |
| 17:28.45 | devoid | haha |
| 17:28.54 | k- | has anybody had experience installing asterisk 1.6 with dahdi on gentoo linux? |
| 17:29.11 | Nugget | k- I run it on a gentoo-server box. no clue about the gooey desktop crap. |
| 17:32.24 | k- | Nugget, do you have any clue about this error? http://dpaste.com/176456/ |
| 17:32.32 | *** join/#asterisk b14ck (~comradeb1@75.80.14.233) |
| 17:32.48 | Nugget | did you compile and install dahdi? |
| 17:33.09 | Nugget | and configure it and load the module |
| 17:33.32 | *** join/#asterisk lost_soul (~noymfb@cpe-67-241-68-202.twcny.res.rr.com) |
| 17:34.44 | *** join/#asterisk hwa (~hwa@li117-222.members.linode.com) |
| 17:34.49 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 17:35.13 | k- | Nugget, yes, I have installed net-misc/dahdi-2.2.0.2 and configured it |
| 17:36.56 | Sargun | Who do you guys use for IP Geolocation? |
| 17:37.35 | k- | It seem like a problem creating /dev/dahdi when dahdi install performs |
| 17:37.42 | p3nguin | geodns or geoip |
| 17:37.44 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
| 17:39.44 | Nugget | /dev/dahdi is a cmponent of the kernel module. if it doesn't exist, that means that dahdi isn't configured properly |
| 17:40.09 | *** join/#asterisk lanning (~lanning@208.87.235.224) |
| 17:40.09 | Nugget | either it's misconfigured, the module isn't actually loaded, or both. |
| 17:40.15 | *** join/#asterisk bmoraca (bmoraca@66.242.162.254) |
| 17:40.52 | Nugget | I don't know anything about installing dahdi from binary, so that's all I know, sorry. I'm a suspender-wearing, bearded unix geek and I build stuff from tarball. |
| 17:43.12 | *** join/#asterisk coppice (~chatzilla@202.62.81.147) |
| 17:44.18 | k- | Nugget, I dont like binary so much either, thats why I prefer gentoo =) |
| 17:45.08 | Skeeter- | <PROTECTED> |
| 17:45.16 | k- | Nugget, I'm going to double check my config, then let you know, thank you! |
| 17:45.31 | Nugget | there's a dahdi-test script (forget the exact name) that should be helpful. |
| 17:45.42 | Nugget | dahdi_test |
| 17:45.45 | *** join/#asterisk Akiraa (~Akiraaaa@79.112.30.234) |
| 17:45.53 | Nugget | and check dahdi_genconf and see what it suggests |
| 17:47.36 | k- | Nugget, I will, thanks |
| 17:49.03 | Kobaz | what's a good car cdma booster |
| 17:49.23 | p3nguin | like an amplifier? |
| 17:49.27 | Kobaz | yeah |
| 17:49.39 | Kobaz | there's like 23987983724 of them |
| 17:49.42 | Kobaz | but which ones are good |
| 17:50.24 | p3nguin | Oh, hmm, I just had one of my consultants working on a CDMA MogFi project. I'll have to ask him what amps he prefers. What kind of phone do you have? |
| 17:50.33 | Kobaz | http://www.allproducts.com/manufacture100/tayx/product2.html |
| 17:50.42 | Kobaz | a client of mine just gave me a crackberry |
| 17:50.49 | p3nguin | which model? |
| 17:51.00 | *** join/#asterisk Heretic (~Fallen@41.133.210.50) |
| 17:51.04 | Kobaz | 8350 or something |
| 17:51.17 | Kobaz | 8330 |
| 17:51.17 | Heretic | lo all |
| 17:51.36 | p3nguin | That's a Curve, right? |
| 17:52.06 | Kobaz | yeah |
| 17:52.27 | *** join/#asterisk LemensTS (~LemensTS@adsl-70-238-175-27.dsl.stlsmo.sbcglobal.net) |
| 17:52.56 | Kobaz | http://www.newegg.com/Product/Product.aspx?Item=N82E16875995040&nm_mc=OTC-Froogle&cm_mmc=OTC-Froogle-_-Cell+Phones+Accessories-_-Wilson+Electronics++Inc.-_-75995040 |
| 17:53.12 | p3nguin | Are you going to use a magnet mount external antenna? |
| 17:53.31 | Kobaz | i suppose |
| 17:53.49 | Kobaz | i'm thinking of getting the data plan and tethering and all that |
| 17:54.01 | Kobaz | and i frequently travel to areas where i get like 1 bar of service |
| 17:54.35 | p3nguin | So far, he's said, "Well, for any CDMA/GSM phone you would just use something Wilson makes. For a Blackberry phone you would need something that works passively." |
| 17:55.02 | Kobaz | mmm |
| 17:55.38 | p3nguin | "Meaning, the amp/antenna wouldn't connect to the phone, it would basically add an antenna inside your car/home/office which your phone would use as a repeater, in which that then communicated with the towers." |
| 17:56.00 | Kobaz | yeah |
| 17:56.18 | Kobaz | isn't that what the amps do? |
| 17:56.32 | p3nguin | Amplifier is such a general term. |
| 17:57.00 | Kobaz | yeah |
| 17:57.20 | p3nguin | To me, an amplifier is any device that boosts the RF output of another device directly. |
| 17:57.25 | ChannelZ | hmmm. Can you not Pickup a ringing call that was transferred from another phone? |
| 17:57.52 | p3nguin | If all devices are in the right call groups and pickup groups, I would think so. |
| 17:58.32 | p3nguin | kobaz: I don't really consider a repeater to be an amplifier, but I'm also not into the whole cellular thing, either. |
| 17:59.26 | ChannelZ | oh I think I see what happened maybe.. when I do the transfer, it goes out in a different context |
| 17:59.56 | ChannelZ | although they are included in each other. hmmm. |
| 18:00.39 | ChannelZ | I guess it can't see through that. |
| 18:01.41 | p3nguin | kobaz: "WPSAntennas.com has a lot of reading information regarding amps, antennas, passive repeaters, etc. CDMA/GSM amps are expensive. A 3watt amp by Wilson is over $250" |
| 18:01.53 | Kobaz | yeah |
| 18:02.05 | p3nguin | Hope this is at least somewhat useful to you. |
| 18:02.37 | Kobaz | i think i've been to this site before |
| 18:02.47 | *** part/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 18:03.42 | p3nguin | kobaz: http://www.primecellular.com/811211 |
| 18:04.32 | p3nguin | "Antenna goes outside the home/office/car and goes inside to the "amp" and then from the amp it goes to what is called a passive repeater/antenna." |
| 18:05.03 | p3nguin | 3 Watts / 26dB gain on that one. |
| 18:05.49 | *** join/#asterisk pabelanger (~pabelange@yoda.kanatek.com) |
| 18:05.51 | Kobaz | preferably a non-direct connect |
| 18:07.10 | *** join/#asterisk antiwire (~antiwire@unaffiliated/antiwire) |
| 18:08.29 | bmoraca | if i recall correctly, adding w's before the number in a dial string causes it to take the line offhook, pause, and then dial, correct? |
| 18:08.39 | p3nguin | What does direct connect even mean on a passive system? |
| 18:09.55 | [TK]D-Fender | bmoraca: anywhere * issues DTMF |
| 18:10.11 | p3nguin | If you're using the D() option in Dial, I know you can put in pauses before sending the DTMF with w. |
| 18:11.18 | p3nguin | kobaz: http://www.primecellular.com/801232 |
| 18:11.32 | *** join/#asterisk socain (~socain00@74.255.249.66) |
| 18:12.07 | bmoraca | [TK]D-Fender, well, it's an E&M Wink trunk, so it's dialing via DTMF, so i guess that counts |
| 18:12.20 | bmoraca | getting the telco to support this stupid thing is a pain in the ass |
| 18:12.22 | [TK]D-Fender | bmoraca: should |
| 18:12.42 | Kobaz | p3nguin: very cool |
| 18:12.48 | Kobaz | so the antenna is seperate |
| 18:13.07 | p3nguin | kobaz: That first product isn't really direct connection because there is no antenna jack/connector. It just uses velcro to hold the passive receptor to the phone. |
| 18:13.15 | Kobaz | yeah |
| 18:13.32 | Kobaz | i would prefer not to have to connect/velrco stuff though |
| 18:13.38 | socain | Any polycom users know how to get 650 arrow keys to work so you can arrow through on-hold calls? I have a polycom 601, same sip ver 3.2.2, and it works on it. |
| 18:14.04 | bmoraca | the trunk is immediately answering and then giving an immediate disconnect about 90% of the time. that doesn't sound like an issue with the Digium hardware, i wouldn't think... |
| 18:14.41 | [TK]D-Fender | socain: arrowing through implies use of multiple calls/line-key. behavior shuold be identical if your linekeys are allocated identically |
| 18:14.46 | [TK]D-Fender | socain: Which I seriously doubt |
| 18:16.30 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:16.30 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 18:17.00 | p3nguin | kobaz: http://www.primecellular.com/801262 62 dB gain! |
| 18:17.32 | p3nguin | kobaz: If you don't want to do the velcro thing, that second link looks like it could be something you would like. |
| 18:18.26 | socain | [TK]D-Fender: i have side cars on the 650 with 6 like keys for the 1 and only registration. What is the best way to disply on hold calls if you have multiple key appearances. The display only shows the last call I touched and no name/etc. for the other on hold calls. |
| 18:18.35 | *** part/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 18:18.51 | p3nguin | kobaz: I guess it has an in-car antenna to pick up your phone, then repeats it back out the external antenna. |
| 18:19.23 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 18:19.37 | Kobaz | yeah |
| 18:19.42 | Kobaz | 62db, heh nutty |
| 18:19.43 | p3nguin | kobaz: It also looks like you get all the parts for only $182.71 |
| 18:19.57 | *** join/#asterisk jtodd (g1ky15qewk@asterisk/community-director-and-tie-dye-shirt-lover/jtodd) |
| 18:19.57 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 18:20.06 | p3nguin | Yeah, that one is a bit crazy. For buildings, though. |
| 18:20.18 | Kobaz | that's for a building, yeah |
| 18:20.37 | Kobaz | that would be nice for my house out in the boonies |
| 18:20.46 | Kobaz | i get one bar if i walk half way up the hill |
| 18:20.55 | Kobaz | or stand on the picnic table outside the house |
| 18:21.44 | AeroCloud | anyone know any way to increase transcoding performance? |
| 18:21.57 | *** join/#asterisk bsaxon (~bsaxon@12.68.234.174) |
| 18:22.00 | AeroCloud | reduce the cpu usage per ^ |
| 18:23.15 | *** join/#asterisk jkroon (~jkroon@dsl-244-51-04.telkomadsl.co.za) |
| 18:23.34 | *** join/#asterisk RobH (~robh@2620:0:860:2:21e:c2ff:fe03:2465) |
| 18:23.40 | jkroon | hi guys, so I've got asterisk in a deadlocked state ... what info do I need to grab for a proper bug report? |
| 18:24.45 | Kobaz | probably sigquit it, and get a core |
| 18:24.48 | Kobaz | #asterisk-dev |
| 18:25.05 | Kobaz | jkroon: what did you do to it? |
| 18:25.42 | jkroon | my monkey test. basically I take a quad pri digium card, put two ports in net mode, two in cpe, loop them with crosses. |
| 18:26.00 | Kobaz | and that's it? |
| 18:26.05 | Kobaz | started it up and it's deadlocked? |
| 18:26.07 | *** join/#asterisk aidinb (~Aidin@66-214-43-104.dhcp.lnbh.ca.charter.com) |
| 18:26.42 | Kobaz | and it's not a loopback if it's one port to another... that's just a crossover |
| 18:26.44 | jkroon | then I originate a call that has MOH on the one side and a recursive dial on the other, so basically if not all dahdi channels used, pass the call out, if all used (determined by Dial() failing with something other than ANSWER), Answer() the call, wait 0.1 seconds and hang up. |
| 18:26.54 | Kobaz | jkroon: #asterisk-dev |
| 18:26.56 | jkroon | initiate two of those, give it about an hour and *boom* |
| 18:27.15 | Kobaz | does it make a boom sound too? |
| 18:27.18 | jkroon | i said looped, not loop back :p |
| 18:27.37 | Kobaz | but looped doesn't even make sense in that context |
| 18:27.37 | jkroon | i wish. that would at least give me the satisfaction of seeing it blow up properly. |
| 18:27.47 | Kobaz | #asterisk-dev |
| 18:28.26 | jkroon | ok, so just crossed back to the same server? |
| 18:28.42 | Kobaz | jkroon: do i have to go to your computer and type /join #asterisk-dev for you |
| 18:28.58 | *** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no) |
| 18:29.28 | Kobaz | jkroon: generically it's considered a crossover yeah.. out and back to the same card |
| 18:30.15 | Kobaz | oh |
| 18:30.16 | Kobaz | you did join |
| 18:30.17 | jkroon | i've already joined. |
| 18:30.21 | Kobaz | i was looking at the wrong window |
| 18:30.22 | Kobaz | hah |
| 18:30.50 | jkroon | just busy typing an overly long message (trying to fit as much info into a single message as possible whilst still hopefully keeping it relevant) |
| 18:32.36 | SouthSuburbTech | i have asterisk set up to recieve calls in, how do i configure it to make calls out using gizmo |
| 18:33.28 | Kobaz | okay yeah, mm |
| 18:33.38 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 18:33.39 | kazaa_lite | hi all |
| 18:34.03 | kazaa_lite | how can i change the text colour of messages appearing on CLI? |
| 18:35.02 | kazaa_lite | i just installed asterisk 1.6.2.6 and it has poor grayish colour in output of commands |
| 18:35.09 | kazaa_lite | which is very hard to read |
| 18:35.23 | kazaa_lite | everything was ok with ast v.1.4.x and 1.6.1.x |
| 18:37.06 | Katty | hi |
| 18:37.34 | hardwire | hrm.. if you Goto from a Gosub extension.. return will never be met right? |
| 18:37.36 | Katty | tried a new burger place for lunch...it wasn't really new but it's the first time i've been there. |
| 18:37.41 | hardwire | unless the goto fails to find the extension/context |
| 18:37.51 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 18:41.45 | socain | [TK]D-Fender: you were right. If i change linekeys=1 then I can arrow through the multiple calls on hold. |
| 18:42.14 | [TK]D-Fender | socain: No, the real difference is "callseperlinkey" |
| 18:43.49 | AeroCloud | [TK]D-Fender: Do you know much about transcoding? |
| 18:44.10 | socain | [TK]D-Fender: I set it to 1, and 6, and they both had the same appearance. Nothing happens and I cant see the names of the on-hold calls. If I only set 1 line key, with multiple calls per key, then i can arrow through. is there a better way to display all of the calls when you have multiple line keys? |
| 18:44.14 | [TK]D-Fender | AeroCloud: just ask |
| 18:44.49 | [TK]D-Fender | socain: Typically with multiple keys yuo assign 1 per key and only use 1 reg for it |
| 18:45.07 | AeroCloud | [TK]D-Fender: I have a pretty beefy server, dual quad core, 8gb ram, and I can only transcode about 110 calls to g729 before the asterisk server explodes |
| 18:45.33 | AeroCloud | I know the server should be able to handle atleast 300 or more transcoded calls |
| 18:45.34 | socain | [TK]D-Fender: i'll try that and see what it looks like. |
| 18:45.51 | [TK]D-Fender | AeroCloud: Make sure to stand as close as possible to tighten the shrapnel pattern ;) |
| 18:45.54 | AeroCloud | the asterisk server restarts itself |
| 18:46.11 | AeroCloud | dropping all 110 callers |
| 18:47.03 | AeroCloud | right now we are limiting each server to only 100 callers because of this.. but there has to be something we can do |
| 18:47.07 | [TK]D-Fender | AeroCloud: I'd check your OS and module build format to make sure you're on the most optimised for your platform |
| 18:47.13 | AeroCloud | how does asterisk transcode? Ram? HD? |
| 18:47.37 | [TK]D-Fender | AeroCloud: RAM... this is realtime.. |
| 18:47.47 | Kobaz | AeroCloud: what do you mean it restarts itself? |
| 18:47.57 | *** join/#asterisk mr_ian (~mr_ian@S0106001b63f49383.du.shawcable.net) |
| 18:47.59 | Kobaz | AeroCloud: asterisk crashes? the machine reboots randomly? what? |
| 18:48.00 | AeroCloud | it restarts itself |
| 18:48.11 | AeroCloud | asterisk restarts itself after it crashes |
| 18:48.21 | Kobaz | okay so you don't have a load issue, you have an asterisk bug |
| 18:48.47 | Kobaz | have you gotten the core file and did a backtrace? |
| 18:48.54 | *** part/#asterisk ACK-NAK (~Miranda@home.chicagoventure.com) |
| 18:49.04 | AeroCloud | no, i have logging = off |
| 18:49.11 | Kobaz | that's not what i asked |
| 18:49.14 | AeroCloud | production servers cant afford too many logs |
| 18:49.19 | Kobaz | sure they can |
| 18:49.32 | Kobaz | the more logging the better |
| 18:49.32 | AeroCloud | nah we use minimal quick drives |
| 18:49.36 | jkroon | AeroCloud, i've started running production boxes with sipdebug=yes |
| 18:49.42 | Kobaz | jkroon: nice |
| 18:49.53 | jkroon | works wonders for informing clients that it's their phones that are broken. |
| 18:49.55 | AeroCloud | thats nice with the logging, maybe I'll throw 1 in the loop |
| 18:50.00 | Kobaz | anyways |
| 18:50.02 | Kobaz | you need the core file |
| 18:50.05 | Kobaz | run asterisk with -g |
| 18:50.08 | jkroon | need to logrotate it often though. |
| 18:50.11 | Kobaz | recompile with DONT_OPTIMIZE |
| 18:50.16 | Kobaz | jkroon: haha yeah |
| 18:50.24 | *** join/#asterisk p3nguin (gpz5GvdFkf@mtop-mpls.a2infotech.com) |
| 18:50.25 | AeroCloud | in the last month, we have over 750,000 calls on 1 server |
| 18:50.26 | Kobaz | jkroon: i forgot to set up logrotate on this one box... 5 gigs of logs after a week |
| 18:50.31 | AeroCloud | limiting to 100 concurrent |
| 18:50.43 | AeroCloud | thats alot of logging |
| 18:50.44 | jkroon | is that it? |
| 18:50.52 | jkroon | I do 13GB/day! |
| 18:50.57 | jkroon | with sipdebug that is :p |
| 18:51.01 | Kobaz | AeroCloud: follow my instructions, get your core file, post a bug on issues.digium.com |
| 18:51.07 | Kobaz | jkroon: i didn't have sip debug on |
| 18:51.15 | Kobaz | jkroon: that's just dialplan output |
| 18:51.15 | AeroCloud | I'll have to replicate it again with a server |
| 18:51.22 | AeroCloud | sucks to have to drop 100+ people |
| 18:51.38 | jkroon | no, core set verbose 0 :) |
| 18:51.55 | Kobaz | it does, but you'll have to dig and post a bug so you can get this fixed |
| 18:52.11 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 18:52.16 | AeroCloud | it might be an issue with the transcoder |
| 18:52.20 | AeroCloud | more than asterisk itself |
| 18:52.24 | Katty | if i wanted to launch an .sh from rc.local |
| 18:52.25 | Kobaz | it could be an issue with anything |
| 18:52.25 | AeroCloud | thats why I'm asking questions |
| 18:52.28 | Katty | what is the syntax i would use |
| 18:52.32 | Kobaz | asking questions is not going to help |
| 18:52.47 | Kobaz | getting a backtrace from your core file is the only thing that will point you in the right direction |
| 18:52.53 | *** join/#asterisk Alagar (~Administr@122.164.42.42) |
| 18:52.55 | Kobaz | doing anything else is useless and a complete waste of time |
| 18:53.23 | Katty | how do i put /path/to/my/thing.sh start into rc.local <- better question |
| 18:53.35 | Kobaz | Katty: put /path/to/my/thing.sh into rc.local |
| 18:53.48 | jkroon | my other open problem (still need debug data) is eyeBeam (mac) making a call into asterisk, if I bridge it to an IAX/2 channel, works, if I bridge it to SIP ... no voice. |
| 18:53.50 | Katty | Kobaz: i don't need any fancy usr bin whatever stuff? |
| 18:53.58 | Kobaz | katty: for what? |
| 18:54.00 | jkroon | tcpdump shows rtp going towards eyeBeam, nothing coming back. |
| 18:54.00 | socain | [TK]D-Fender: ok, when i put it back to 6 line keys, with 1 call per key, it puts a call on each line but the display only shows the name, number, and duration of the last call I touched. So if I have 6 calls on hold i cannot see the name/number next to each line key. would adjusting font size allow the display to show the names of all callers? |
| 18:54.04 | Katty | Kobaz: idk. |
| 18:54.12 | Katty | Kobaz: i've never put anything into rc.local :P |
| 18:54.12 | Kobaz | just put the thing in there |
| 18:54.14 | Katty | k |
| 18:54.14 | jkroon | also another day's problem. |
| 18:54.21 | Kobaz | and add a & if it doesn't spawn in the background by itself |
| 18:54.33 | Kobaz | otherwise it'll lock up your boot process and you wont get a login prompt |
| 18:54.55 | hardwire | is there a known tried/true method of checking to see if a variable exists for IF statements? |
| 18:55.00 | hardwire | exists + has content |
| 18:55.11 | hardwire | do the "${var}" != "" test? |
| 18:55.42 | AeroCloud | its wierd.. 8 cores.. so at 100 calls asterisk shows about 30% cpu usage.. but after the 110 mark it jumps up exponentially |
| 18:56.15 | Kobaz | sounds like a task scheduling issue |
| 18:56.47 | Katty | can i run rc.local |
| 18:56.50 | Katty | just to test it |
| 18:56.51 | Kobaz | sure |
| 18:56.54 | Kobaz | sh /etc/rc.local |
| 18:56.57 | Katty | k |
| 18:57.05 | Katty | hot |
| 18:57.56 | Katty | what does 'sh' mean? |
| 18:58.18 | Kobaz | it's usually a symlink to bash on most linux systems |
| 18:58.35 | Katty | is that the windows equivilent of exe? |
| 18:58.47 | Kobaz | if /etc/rc.local wasn't executable you would have to run it through something |
| 18:58.56 | Kobaz | so it's like the windows equivalant to batch |
| 18:59.25 | Kobaz | assuming /etc/rc.local is executable. which it probably should be... and you're running on a fairly normal linux system, which you probably are |
| 18:59.38 | Kobaz | then there would be no real difference between sh /etc/rc.local and just plain /etc/rc.local |
| 19:00.14 | Kobaz | it's not equivalent to exe |
| 19:00.17 | Katty | hrmm |
| 19:00.20 | Katty | bat? |
| 19:00.21 | Kobaz | chmod u+x would be the equivalent to exe |
| 19:00.27 | Kobaz | batch=bat |
| 19:00.28 | Kobaz | yeap |
| 19:00.29 | Katty | k |
| 19:00.32 | Kobaz | aka: shell script |
| 19:01.02 | Katty | well i added isymphony into and it works like a charm (= |
| 19:01.09 | Kobaz | sexy |
| 19:01.10 | Katty | i should also add a mutt in there |
| 19:01.15 | Katty | to tell me the server rebooted |
| 19:01.19 | Kobaz | sure |
| 19:01.22 | Kobaz | make sure to background it |
| 19:01.31 | Katty | background it? |
| 19:01.38 | Kobaz | you can use straight-up mail if you don't need anything fancy |
| 19:01.45 | Kobaz | echo "whatever" | mail ...options |
| 19:01.56 | Katty | i was just going to like echo "foo" | mutt -s "zomg server restart" |
| 19:01.58 | Kobaz | echo "whatever" | mail ...options & <-- background |
| 19:02.12 | Katty | k |
| 19:02.20 | Katty | will echo etcetcetc be fine in rc.local? |
| 19:02.24 | Kobaz | yeah |
| 19:02.25 | Katty | or do i need to turn it into an sh |
| 19:02.28 | Katty | and then execute sh |
| 19:02.29 | Kobaz | it's just a regular shell script |
| 19:02.34 | Katty | k |
| 19:02.38 | Kobaz | it's already a shell script |
| 19:03.52 | Katty | checks her phone for sms |
| 19:04.20 | Katty | frowns |
| 19:04.36 | hardwire | wonders if he's missing out by not using AEL |
| 19:04.38 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 19:04.40 | Kobaz | hardwire: yes |
| 19:04.49 | hardwire | Kobaz: you bastard.. I don't want to change my dialplan |
| 19:04.50 | Kobaz | Katty: tail -f /var/log/mail (or whatever mail server you're using) |
| 19:04.53 | Katty | TADA |
| 19:04.59 | Katty | it's there (= |
| 19:05.18 | Katty | does mutt let you specific the sending party? |
| 19:05.36 | Katty | digs through man |
| 19:05.36 | Kobaz | dunno |
| 19:05.42 | Kobaz | i like swaks |
| 19:05.46 | Kobaz | you can do anything with swaks |
| 19:05.56 | Kobaz | including sending attachments, do authentication, all that good stuff |
| 19:06.04 | Katty | i know mutt does attachments |
| 19:06.09 | Kobaz | swaks! |
| 19:07.15 | Kobaz | hardwire: dialplan without ael is like driving in a mustang gt with only two cylinders fireing |
| 19:07.45 | Kobaz | it works, and it'll get you from point a to point b... but man you're missing out |
| 19:08.32 | Kobaz | hardwire: but i found that for me, rewriting everything in AGI made things even better |
| 19:09.47 | Katty | waits for sms again |
| 19:10.20 | Katty | pokes phone |
| 19:10.31 | Katty | Dear Sprint, hurry it up. Love, Katty |
| 19:11.22 | *** join/#asterisk DennisG (DennisG@2002:541e:88d0:0:f9e4:43ec:892b:7e5a) |
| 19:11.37 | Katty | there's really no reason for this |
| 19:11.40 | Katty | it's just an email |
| 19:13.31 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 19:14.13 | Katty | alright so if i date it gives me the date |
| 19:14.16 | Katty | how do i date into an echo |
| 19:15.58 | p3nguin | Hmm, what do you need/want to do? |
| 19:16.03 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 19:17.41 | ChannelZ | date into an echo? |
| 19:18.54 | ChannelZ | like echo "Hey bitches, today is `date` and I feel fine" |
| 19:25.57 | p3nguin | We may never know. |
| 19:26.30 | paulc | Kobaz: So AEL instead of extensions.conf - big fan? |
| 19:26.50 | Kobaz | paulc: as long as you understand how it's converted... it's wonderful |
| 19:27.06 | paulc | Interesting.. cos I've never really played with it.. but might be worth a look.. |
| 19:27.14 | Kobaz | i really can't stand unstructured programming |
| 19:27.32 | paulc | right now I'm wondering about IVR with database integration and scalability.. pondering a few ideas.. and not really in the mood for work at the day job |
| 19:27.36 | Kobaz | i think extensions.conf should be phased out and just have ael be the standard... it seems silly |
| 19:28.02 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-84-111-21-244.red.bezeqint.net) |
| 19:28.03 | Kobaz | 10,print hi |
| 19:28.06 | Kobaz | 20,goto 10 |
| 19:28.23 | Kobaz | reminds me of the horrible days of BASIC programming |
| 19:29.47 | *** part/#asterisk muiro (~muiro@unaffiliated/muiro) |
| 19:31.54 | Corydon76-dig | Kobaz: If you have a problem with it, then you should be using pbx_lua, not pbx_ael |
| 19:31.57 | paulc | LOL but it's still a lot nicer than some of that XML configuration stuff you get with a different product a co worker of mine was a huge fan of |
| 19:32.06 | *** join/#asterisk rgsteele (~rgsteele@207.106.239.81) |
| 19:32.07 | paulc | XML = bloat |
| 19:32.27 | Kobaz | Corydon76-dig: i tried pba_lua.. didn't like it |
| 19:32.57 | Kobaz | i use ael for quick little dialplan stuff, menues and whatever |
| 19:33.05 | Kobaz | i write actual applications using AGI |
| 19:33.19 | Corydon76-dig | Someone will always need to be proficient in extensions.conf, because AEL is translated directly into it |
| 19:33.26 | Kobaz | yeah |
| 19:34.50 | Kobaz | jkroon: i can play later, i have a dual span in a test box |
| 19:35.12 | Kobaz | speaking of dual spans |
| 19:35.24 | paulc | Kobaz: What do you write your AGI in? |
| 19:35.31 | Kobaz | should i get a quad span, or two dual spans |
| 19:35.42 | Kobaz | paulc: perl |
| 19:35.50 | jkroon | single dual span in fine. |
| 19:36.01 | leifmadsen | jkroon: he's asking how he should configure 4 spans |
| 19:36.07 | leifmadsen | Kobaz: I'd say single quad-span |
| 19:36.10 | Kobaz | jkroon: this is for a new job |
| 19:36.19 | jkroon | oh, nm. this is getting confusing. |
| 19:36.22 | Kobaz | hehe |
| 19:36.39 | jkroon | well, according to digium they officially don't support more than three of their cards in a single box. |
| 19:36.44 | Kobaz | yeah |
| 19:36.58 | Kobaz | pci bus contention |
| 19:37.25 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 19:37.40 | jkroon | well, it's three BRIs and one PRI, but I can reproduce with a single BRI and a single PRI ... so their argument in this case doesn't stand ground. |
| 19:38.45 | Kobaz | i need a new phone/board vendor |
| 19:39.03 | Kobaz | the one i've been dealing with for the past two years is taking a nosedive in terms of support |
| 19:40.21 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 19:44.19 | paulc | I hate it when support nosedives. It's sad when good companies that you've loved previous go downhill |
| 19:44.48 | Kobaz | yeah |
| 19:44.55 | Kobaz | all i'm trying to do is rma two phones |
| 19:45.08 | Kobaz | i've been trying to get the rma numbers for a week now |
| 19:46.48 | paulc | You need some "Best of British Bolshyness" on the phone :) |
| 19:47.11 | paulc | "Look, this is ridiculous. I've got these phones, they're faulty, all I need is an RMA, why is this taking so long - what's the problem? is there anything I can do to help?" |
| 19:47.25 | paulc | ugh - nothing pisses me off more than bad customer service.. cos 99% of the time, it's avoidable |
| 19:47.51 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 19:48.20 | Kobaz | yeap |
| 19:53.20 | *** join/#asterisk |AnToS| (~31749@93-44-103-101.ip96.fastwebnet.it) |
| 20:25.52 | *** join/#asterisk infobot (ibot@rikers.org) |
| 20:25.52 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.6, 1.6.1.18, 1.6.0.26 (2010/03/12), 1.4.30 (2010/03/12), *-Addons 1.6.2.0 (2009/12/18), 1.6.1.2, 1.6.0.4, 1.4.10 (2009/12/02), dahdi-linux 2.2.1 + dahdi-tools 2.2.1 (2010/01/20), Libpri 1.4.10.2 (2009/10/20) -=- Related channels: #asterisknow,#switchvox,#freepbx,#asterisk-dev,#asterisk-bugs |
| 20:25.57 | Skeeter- | ah nice |
| 20:26.02 | Skeeter- | ~Qwell |
| 20:26.02 | infobot | it has been said that qwell is a patented liquid formula that contains three plant-based bio-active agents that work together in a perfectly balanced combination. These agents act synergistically to boost your good cholesterol and slash the bad. |
| 20:26.26 | Skeeter- | ~Flashtek |
| 20:26.33 | Skeeter- | umm |
| 20:28.06 | *** join/#asterisk nix8n82 (~nate@mo-65-41-196-62.sta.embarqhsd.net) |
| 20:28.51 | nix8n82 | Anyone know what the difference is in polycoms 3.2.2 combined and 3.2.2 split? |
| 20:29.13 | DelphiWorld | Skeeter-: hahaha |
| 20:29.27 | *** join/#asterisk jkroon (~jkroon@dsl-244-51-04.telkomadsl.co.za) |
| 20:29.31 | Skeeter- | nix8n82, |
| 20:30.07 | Skeeter- | combined is 1fil that contains all informations for all phones(bootrom firmware, etc) split = obvious |
| 20:30.22 | Skeeter- | nix8n82, remember that im most of the time wrong |
| 20:30.50 | nix8n82 | Skeeter-, good to know, thank you |
| 20:31.04 | DelphiWorld | where i can see the numbering plan of the world in ITU? |
| 20:31.25 | Skeeter- | ~where i can see the numbering plan of the world in ITU? |
| 20:31.38 | Skeeter- | task must be impossible |
| 20:32.08 | DelphiWorld | Skeeter-: respect yourself |
| 20:32.34 | Qwell | ITU doesn't handle numbering plans. |
| 20:32.55 | DelphiWorld | Qwell: http://www.itu.int/oth/T0202000003/en |
| 20:33.06 | *** join/#asterisk Oozio (~user@modemcable092.170-202-24.mc.videotron.ca) |
| 20:34.54 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-84-111-21-244.red.bezeqint.net) |
| 20:42.40 | nix8n82 | might be another dumb question, but can you have some polycom phones running sip 3.0 and another group 3.2.2 and still talk to each other with asterisk? |
| 20:43.22 | Qwell | sure, it's just SIP |
| 20:43.43 | wcselby | those are firmwares on the phones, and yes they can be different |
| 20:43.59 | nix8n82 | Thanks Qwell and wcselby |
| 20:45.33 | wcselby | np |
| 20:47.21 | megalomano | hi , someone have any howto or file , to configure a quintum tenor af series in asterisk? |
| 20:48.35 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 20:50.40 | *** join/#asterisk binbash_ (~peter@ip4da53781.direct-adsl.nl) |
| 20:52.41 | hardwire | huts ${ARRAY... |
| 20:52.43 | hardwire | hugs |
| 20:53.38 | hardwire | southSIIIIDE |
| 20:59.25 | voxter | This is strange... If i force a SIP trunks codec to ulaw only, passing calls in and to my fax server (hylafax) work great. If i set my sip peer to g729 AND ulaw to my provider (they pick g729 by default), and use SIP_CODEC to force the call to change to ulaw, that seems to work (by evidence of sip show channels, i see ulaw codec being used) yet all faxes fail, as though its not "really" ulaw. What gives? |
| 20:59.27 | *** part/#asterisk rttrey (~trey@209.208.18.121) |
| 21:00.11 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 21:00.30 | p3nguin | Where's the sip debug? |
| 21:00.52 | *** join/#asterisk kazaa_lite (~eddie@cpc6-slam5-2-0-cust171.2-4.cable.virginmedia.com) |
| 21:04.25 | *** join/#asterisk war9407 (war@liquidswords.org) |
| 21:17.13 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 21:17.24 | voxter | p3nguin: have to reproduce again after hours, it affected too many people to keep it going. |
| 21:17.50 | voxter | I suspect that asterisk 'thought' it was ulaw, but it was still using g729 to my trunk provider. |
| 21:17.59 | p3nguin | You do a lot of faxing, huh? |
| 21:18.16 | voxter | I dont, but my customers do. Inbound. |
| 21:18.18 | voxter | Its unfortunate. |
| 21:26.02 | *** join/#asterisk jsgoecke (~Adium@c-71-202-25-141.hsd1.ca.comcast.net) |
| 21:26.09 | jsgoecke | Hello all |
| 21:26.28 | paulc | hi hsgoecke |
| 21:27.37 | paulc | Any IVR type peeps in here? I'm wondering about the feasibility of replacing a 920 port IVR box with Asterisk.. polling a back end database (via CURL? or AGI?) to find out what to do next.. |
| 21:27.44 | paulc | dreaming dreams on a Friday afternoon at work... |
| 21:28.53 | *** join/#asterisk timholum (~chatzilla@64-91-67-5.stat.centurytel.net) |
| 21:29.01 | jsgoecke | Well, Asterisk could foot the bill based on requirements of course. You could use it in conjunction with Adhearsion (http://adhearsion.com) of course. Which provides a dev framework that supports both AGI and AMI. |
| 21:29.53 | jsgoecke | I keep getting this error when trying to compile a vanilla Asterisk 1.4.30 on Snow Leopard http://gist.github.com/345413 |
| 21:29.57 | jsgoecke | src/add.c:1: error: CPU you selected does not support x86-64 instruction set |
| 21:30.00 | jsgoecke | How do I get around this? |
| 21:31.01 | jsgoecke | paulc Which type of IVR are you looking to replace and what is it doing exactly? |
| 21:32.39 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 21:34.54 | hardwire | with gosub, how can you tell if an argument was used explicitely.. because it inherits ARG* |
| 21:35.10 | hardwire | so checking to see if an arg exists doesn't really work well. |
| 21:36.36 | hardwire | used/set |
| 21:38.11 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 21:49.24 | *** join/#asterisk Sparky-UK (~sparky@cpc1-stok5-0-0-cust42.bagu.cable.ntl.com) |
| 21:49.59 | *** join/#asterisk ManxPower-work (~manxpower@216.186.151.147) |
| 21:51.38 | Sparky-UK | Can anyone tell me why I can get a call to SIP/SER1/1234 working in a dial plan, but not from an originate command? |
| 21:55.04 | *** join/#asterisk moy (~chatzilla@74.12.123.160) |
| 21:57.11 | paulc | jsgoecke: delayed reply, people keep stopping by my desk. We have a chat line type product, people scrolling through greetings, messaging each other etc |
| 21:57.43 | jsgoecke | k, seems like you would need an async API for that to be done easily, something like the AMI of Asterisk |
| 21:57.48 | jsgoecke | Which IVR vendor do you use now? |
| 21:58.17 | paulc | CT-ADE, previously on top of Dialogic, but now on top of a Vail SIP stack pretending to be Dialogic (cos we're sneaky like that) |
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| 21:59.28 | jsgoecke | What is wrong with the way you are doing it now? |
| 21:59.52 | sbrath | I just had a wierd experience, I added a few new extensions to users.conf, and did a sip reload, dialplan reload and voicemail reload and then all the phones un-registered? I'm wondering that since I'm using users.conf instead of seperate files that this might be gunking me up ? |
| 22:00.56 | paulc | jsgoecke: we have some architectural issues with the way it's designed/works.. accounts belong to nodes, it's not truly meshed/scalable, a bunch of stuff |
| 22:01.23 | paulc | ack - I have a meeting - back in 30 or less |
| 22:01.25 | jsgoecke | Well, Asterisk will take some work too. For 920 ports of conferencing you are going to have to do some clustering. |
| 22:02.01 | paulc | jsgoecke: it's mostly message passing, not conferencing... 26 systems x 920 ports per box (but typically running at less than 50% capacity). I'll ping you when I'm back. |
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| 22:05.13 | ManxPower-work | ~users.conf |
| 22:05.14 | infobot | [~users.conf] users.conf is a flaming pile of sh1t that takes the fine control of several perfectly usable * config files and reduces them to the lowest common denominator and makes your system behave like a "toaster grade" PBX system. |
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| 22:16.01 | ManxPower-work | How are you today, [TK]D-Fender |
| 22:16.26 | [TK]D-Fender | happily home. |
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| 22:30.40 | hardwire | wonders if a threaded manager_http.c will ever happen |
| 22:31.30 | hardwire | manager_curl.c would be it I suppose |
| 22:32.39 | hardwire | ponders using trunk+CEL |
| 22:34.27 | hardwire | I told my boss a year ago that I would try it out |
| 22:34.30 | hardwire | sigh. |
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| 22:59.07 | jsgoecke | Well, Asterisk could foot the bill based on requirements of course. You could use it in conjunction with Adhearsion (http://adhearsion.com/) of course. Which provides a dev framework that supports both AGI and AMI. |
| 22:59.07 | jsgoecke | 2:29 |
| 22:59.07 | jsgoecke | I keep getting this error when trying to compile a vanilla Asterisk 1.4.30 on Snow Leopard http://gist.github.com/345413 |
| 23:01.39 | jsjc | looking for someone to provide around 60minutes of asterisk support for a decent price. I started reading the book but it is out of my hand the first configuration. I will be able to implement things after and here and there but first setup it is out of my hands |
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| 23:10.38 | paulc | jsjc: what kind of stuff you looking to do? |
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| 23:21.01 | ChUbB | what device do I need to connect a asterisk server to a BT phone line ?? whats best a device/box or an internal card and whats cheapiest ?? |
| 23:25.02 | Chainsaw | ChUbB: To connect to a BT phone line and make calls you would need an FXO port. |
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| 23:25.29 | Chainsaw | ChUbB: An internal PCI card like the Digium TDM410 with (at least) 1 FXO module can do that job. |
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| 23:28.27 | Mhaddog | hello, good night all |
| 23:30.38 | paulc | ChUbB: Or something like a Sipura SPA-3000 works well too |
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