00:02.11 | Zippoman | Just so you guys know I didn't mean asterisk web GUI |
00:02.20 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
00:02.39 | root52 | So what on earth do you mean? |
00:03.05 | drmessano | If this involves porn, I am all in |
00:03.12 | Zippoman | hahaha |
00:03.24 | drmessano | Unless it's pedo crap.. in which case, I need to punch you in the throat |
00:03.27 | Zippoman | Im making a site so I can have text to speech calls |
00:03.40 | drmessano | So out with it, Loose Zipper Larry |
00:03.44 | drmessano | Hmm |
00:04.00 | StanManCan | i had quite the grin on my face when i saw my grandma's call registering on the asterisk server |
00:04.01 | StanManCan | lol |
00:04.14 | StanManCan | going through the steps "enter pin" "thank you" "enter number" |
00:05.49 | StanManCan | now the quality was good, but a little fuzzy, i'm using ulaw, any other's i can try that may give a better result ? |
00:12.39 | [TK]D-Fender | StanManCan: Its likely your provider. ulaw is the highest practical quality codec. it then comes down to network conditions and your provider |
00:16.46 | StanManCan | thought so.. oh welll |
00:21.11 | carrar | ... and then as quickly as he came into the room, he was gone! |
00:21.54 | drmessano | carrar: I hate it when you act "like that" |
00:22.18 | carrar | You hate everything when this time of the month! |
00:25.54 | [TK]D-Fender | carrar: Actually he lasted far longer... and listened and compied and stuff too! |
00:26.24 | carrar | heh |
00:26.36 | [TK]D-Fender | carrar: Last time he tore off when I told him his attempt to mask everything in debug was fucking over attempts to fix the problem. |
00:28.23 | drmessano | Im glad his grandma can call him now |
00:28.28 | drmessano | Mission accomplished |
00:28.50 | drmessano | I would love it if my grandma called me, but Asterisk doesnt have that feature yet |
00:29.10 | carrar | I'd be a little scared if mine called |
00:29.30 | [TK]D-Fender | carrar: I'm suspecting your reasons are the same. |
00:29.38 | carrar | module load app_afterlife.so |
00:29.57 | [TK]D-Fender | dials 1-800-THE-LORD |
00:30.17 | carrar | or perhaps thats chan_afterlife.so |
00:31.11 | carrar | debugging that could be hell |
00:31.51 | drmessano | ba dump |
00:31.53 | drmessano | CHING! |
00:32.32 | [TK]D-Fender | takes his giant cane and yanks carrar off the stage |
00:32.40 | drmessano | I hear they just installed a large CCM network in hell.. |
00:33.43 | carrar | good dd, good dd, good dd errr Thats All Folks! |
00:34.02 | drmessano | MySQL hates me |
00:34.14 | carrar | PostgreSQL love you |
00:34.28 | carrar | loves even |
00:34.39 | [TK]D-Fender | goes running for his asbestos suit... |
00:35.40 | carrar | SUN says MySQL 5.1 is ready for production!! :) |
00:36.21 | drmessano | MySQL 5.1 is giving me a tumor |
00:36.34 | [TK]D-Fender | carrar: http://arstechnica.com/open-source/news/2009/02/unsatisfied-with-direction-mysql-creator-leaves-sun.ars |
00:36.57 | carrar | drmessano, be glade thats all MySQL 5.1 is doing too you |
00:37.13 | jupeterson | does anyone know of a good web voicemail application? So people can listen to voicemail via a browser.. login and check messages |
00:37.35 | carrar | heh yeah I read that on /. |
00:40.57 | *** join/#asterisk ManxPower (n=manxpowe@75.251.218.255) |
00:41.23 | [TK]D-Fender | My company uses MySQL for a few small things for which I could probably faily easily switch to PostgreSQL if things ever turn on us |
00:41.58 | carrar | jupeterson, did you look at the source? |
00:42.03 | carrar | jupeterson, asterisk/contrib/scripts/vmail.cgi |
00:42.46 | carrar | TK, no time like the working present! :) |
00:42.48 | jupeterson | no, I didnt look at that yet.. thanks.. I'll check it out |
00:42.59 | ManxPower | As I understand it vmail.cgi is not maintained and has not been for years |
00:43.30 | carrar | goes back to googling for jupeterson |
00:44.24 | *** join/#asterisk neurosys (n=vinix@c-71-196-9-142.hsd1.fl.comcast.net) |
00:45.26 | [TK]D-Fender | carrar: ... presently I have MySQL and it is working :) |
00:46.07 | [TK]D-Fender | ManxPower: As I understand it *'s voicemail file storage hasn't changed in years either :) |
00:46.27 | ManxPower | [TK]D-Fender: I'm working with FreeBSD. I've not had to recompile a Linux kernel in *years*. I've recompiled the FreeBSD kernel like 5 times already. |
00:46.27 | [TK]D-Fender | ManxPower: And apparently the CGI has been updated with references to ODBC... |
00:46.39 | ManxPower | [TK]D-Fender: in that case I was wrong. |
00:50.39 | carrar | jupeterson, I setup Asterisk to use UW Imap for vm and that works slick |
00:50.54 | carrar | people just add the VM imap server to their existing mail program |
00:50.56 | rob0 | One time, I thought I was wrong, but as it happened, I was mistaken. |
00:51.48 | jupeterson | ok.. yeah, I've got it emailing attachments right now.. wanted a point and click method via the web.. I found this iPhone app for Asterisk but havent gotten it working yet |
00:51.56 | jupeterson | was wondering if there was something turnkey out there |
00:52.04 | carrar | this is not "emailing attachement" |
00:53.05 | carrar | theres a few web apps out there |
00:53.21 | carrar | there is a nice one that comes with SwitchVox |
00:54.43 | [TK]D-Fender | And i'll only cost you your immortal soul! muahahhaahaha *cough* |
00:54.47 | carrar | hahha |
00:54.50 | carrar | yeah |
00:55.00 | carrar | too bad SwitchVox isn't open source |
00:55.07 | ManxPower | [TK]D-Fender: I sold that recently. I don't seem to miss it all that much. |
00:55.33 | [TK]D-Fender | ManxPower: I'm not sure who got gypped on that deal ;) |
00:55.36 | ManxPower | sorry, I thought you said "Immoral soul", not "immortal soul" |
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01:05.45 | neurosys | ManxPower: are you running asterisk on fbsDF? |
01:05.48 | neurosys | *fbsd? |
01:06.06 | drmessano | fbsDF, christ, another distro |
01:06.12 | neurosys | heheh |
01:08.14 | ManxPower | neurosys: no |
01:08.21 | carrar | I used to only use freebsd |
01:08.28 | ManxPower | *heart* Linux. |
01:08.55 | ManxPower | But Linux has some issues with high speed usb-serial, so I'm using FreeBSD. I'm starting to hate their packet filtering stuff |
01:09.17 | neurosys | fbsd is my platform of choice. |
01:09.24 | neurosys | ManxPower: Which one are you using? |
01:09.55 | ManxPower | neurosys: I use Linux for everything except my gateway, which is FreeBSD |
01:10.31 | neurosys | ManxPower: What are you using? IPW, IPFW, PF? |
01:11.24 | ManxPower | neurosys: ipfw I found that I needed option IPFIREWALL_FORWARD So I'm recompiling again |
01:12.16 | ManxPower | neurosys: think "captive garden", all connections to port 80 from the local network with a destination that is not the local network be redirected to 127.0.0.1, port 81, where a web server takes over |
01:12.28 | neurosys | ManxPower: you really should try PF. It is by far the best. It was ported over from OpenBSD. Far superior. |
01:12.30 | ManxPower | I need to allow ssh and irc thru the box with just basic nat |
01:12.48 | ManxPower | neurosys: I've spent days learning about ipfw, I would rather not throw out all that work. |
01:13.08 | ManxPower | neurosys: and PF is far, far, far too vague for my taste. |
01:13.43 | neurosys | ManxPower: Vague? How so? |
01:13.44 | ManxPower | allow web not to me and to not me nat to local 1.2.3.4 |
01:13.55 | ManxPower | sorry, that is not a filter that is "fisher price my first firewall"/ |
01:14.21 | ManxPower | But I come from the iptables world where you are expected to understand networking before you try to set up a firewall. |
01:14.30 | carrar | If you can deal with the bright colors it works great |
01:14.57 | neurosys | ManxPower: heh PF is much more robust than that. |
01:15.03 | carrar | red circle can only fit through the red circle fw rule |
01:15.09 | *** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu) |
01:15.29 | carrar | unless you paint the outside blue |
01:15.31 | rob0 | grabs the red circle from carrar and runs off! |
01:16.04 | [TK]D-Fender | carrar: But Miss the square peg DOES go in the round hole! *WHAM*wham*WHAM*wham*WHAM*wham*WHAM*wham*WHAM*wham |
01:16.22 | ManxPower | I can configure iptables in my sleep but ipfw and the rest are just kicking my ass |
01:16.58 | jaytee | so just take a nap and use iptables :-) |
01:16.59 | carrar | [TK]D-Fender, nothing a little a soldering iron can't melt away some protection! |
01:18.41 | jaytee | tries to calculate the risk factor in eating a Reese's peanut butter cup. |
01:18.55 | *** join/#asterisk RobertLaptop (n=rmiddle@96.244.48.200) |
01:19.21 | ManxPower | jaytee: Verizon EVDO on Windows = 1.5Mbps more or less. Verizon EVDO on FreeBSD = 0.6Mbps Verizon EVDO on Linux = 0.014Mbps. |
01:19.22 | carrar | 1000:1 |
01:20.18 | jaytee | ManxPower, what's up with the shitty performance of EVDO on BSD and Linux? crappy drivers? |
01:20.45 | [TK]D-Fender | ManxPower: Think their throughput math is bad.... jsut wait'll you look at their accounting ;) |
01:20.48 | ManxPower | jaytee: that is my thought. 0.6Mbps is OK for my use almost all the time. |
01:21.05 | [TK]D-Fender | ManxPower: just my .02 cents worth :p |
01:21.47 | carrar | penny for your thoughts? |
01:22.05 | ManxPower | jaytee: Almost all the USB Hight Speed (EVDO Cellular) internet devices act like serial devices. The Linux usb/serial drivers are pretty horrid with some devices. |
01:22.47 | drmessano | Why not just use an EVDO router? |
01:22.54 | carrar | My AT&T HSPDA works pretty good in the city |
01:23.05 | carrar | consistant 1 Mbps |
01:23.06 | ManxPower | carrar: on what OS? |
01:23.10 | carrar | OSX |
01:23.13 | jaytee | ManxPower, what does your EVDO modem show up as in linux? |
01:23.20 | carrar | I should test it again |
01:23.23 | carrar | been a fewmonths |
01:23.32 | drmessano | OSX isn't an OS.. it's a lifestyle |
01:23.33 | ManxPower | jaytee: /dev/ttyU0.0 using the option.ko driver |
01:23.37 | carrar | hahah |
01:23.39 | carrar | true |
01:23.47 | carrar | but makes a smoken desktop |
01:24.05 | drmessano | Well |
01:24.10 | jaytee | ManxPower, what distro? |
01:24.13 | ManxPower | We'll be right back, after this reboot. |
01:24.16 | drmessano | Windows XP does one thing OSX will never do |
01:24.18 | ManxPower | jaytee: centos |
01:24.23 | drmessano | Run on my PC |
01:24.34 | carrar | You can put OSX on a PC |
01:24.47 | drmessano | Not out of the box |
01:25.23 | jksM | anyone using the siemens gigaset SIP-phones and got direct call _to_ them working? |
01:27.49 | carrar | ok I'll try another speed test here |
01:29.49 | carrar | yeah still the same |
01:29.59 | carrar | 1 meg |
01:30.48 | jaytee | ManxPower, what brand of modem is it? |
01:31.41 | carrar | Merlin XU870 3G HSDPA 7.2 ExpressCard |
01:32.07 | jaytee | that's his card? |
01:32.14 | carrar | oh |
01:32.15 | jaytee | or yours |
01:32.16 | carrar | sorry |
01:32.18 | carrar | mine |
01:33.00 | carrar | Goes up to 7.3 Mbps |
01:33.11 | carrar | too bad AT&T will never let me do that |
01:34.06 | carrar | peaking out at 1.0768 Mbps |
01:37.12 | [TK]D-Fender | carrwhat's your upstream? |
01:37.20 | carrar | was just gonna test that |
01:39.08 | carrar | heh |
01:39.14 | carrar | not so good :) |
01:39.26 | carrar | so a wget down does 1.096 Mbps |
01:39.37 | carrar | a scp up is about 400 kbps |
01:40.23 | carrar | 100 meg file |
01:40.40 | carrar | so it has time to level off |
01:42.20 | carrar | we'll try scp down to see if they favor the web traffic |
01:42.37 | carrar | naw looks the same down |
01:43.20 | carrar | acceptable speeds |
01:43.59 | [TK]D-Fender | waits for ADSL2 to roll out in his area... |
01:45.09 | carrar | bonded ADSL2!! |
01:45.26 | [TK]D-Fender | carrar: 1 at a time :) |
01:45.33 | carrar | heh |
01:45.36 | [TK]D-Fender | carrar: I was considering bonded ADSL1 |
01:45.48 | carrar | we do bonded DSL |
01:46.00 | carrar | works pretty slick |
01:46.03 | [TK]D-Fender | carrar: I'm looking to get a better bang/buck on T1+ overall speeds |
01:46.32 | [TK]D-Fender | carrar: anything >= to 1.544 up/down (sync not req |
01:46.33 | carrar | dual 7meg DSL customers see about 12 megs down |
01:46.42 | [TK]D-Fender | CarUPSTREAM is what I care about |
01:47.06 | carrar | 1.5 up |
01:47.18 | carrar | tops |
01:48.09 | [TK]D-Fender | carrGAH! |
01:48.17 | [TK]D-Fender | carrar: dang autocomplete |
01:48.27 | [TK]D-Fender | anyway... off to play pool for a while.... BBL: |
01:48.39 | carrar | drink more! |
01:48.42 | carrar | have fun |
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03:15.44 | drmessano | have I mentioned in the last hour how much I hate mysql |
03:16.05 | ekat | whys that |
03:16.06 | ekat | ? |
03:18.48 | jaytee | no, you haven't mentioned it in at least two or three hours. |
03:18.48 | drmessano | mysqldump needs to be renamed to "dump some crap.non-restorable" |
03:18.59 | ekat | lol |
03:19.14 | drmessano | It doesnt replicate well |
03:19.17 | ekat | are you trying to restore a .sql file to a different mysql version? |
03:19.20 | drmessano | and you cant back it up and restore |
03:19.20 | *** join/#asterisk ipguy (n=ipguy@124-170-148-202.dyn.iinet.net.au) |
03:19.21 | drmessano | No |
03:19.37 | jaytee | I've used mysqldump to move data from a database on one server to a database on another with no problem. |
03:19.40 | drmessano | Im trying to dump from one server to another |
03:19.44 | drmessano | same version |
03:19.50 | drmessano | Same OS |
03:19.51 | ekat | yea, thats weird them |
03:19.54 | ekat | er then |
03:20.09 | drmessano | So I am unpacking a new, blank SQL DB |
03:20.09 | ipguy | stupid question time, how do i change the theme is trixbox ? |
03:20.21 | jaytee | you ask in #trixbox |
03:20.27 | drmessano | or go to their forums |
03:20.45 | ipguy | yep, just did |
03:20.48 | *** part/#asterisk ipguy (n=ipguy@124-170-148-202.dyn.iinet.net.au) |
03:21.06 | jaytee | and how'd that work out for you? better than asking in here? |
03:21.16 | ekat | too late |
03:21.19 | ekat | already gone |
03:21.50 | ekat | drmessano: does the new server support the same type of table types? innodb? |
03:22.07 | ekat | it should say where the .sql file breaks |
03:22.10 | ekat | that should give a hint |
03:22.16 | jaytee | trixbox users, meh! they'd friggin glue sequins on the interface if they could |
03:22.26 | ekat | ol |
03:22.28 | ekat | lol |
03:25.48 | drmessano | I think I got it now |
03:26.06 | drmessano | empty db for the win |
03:26.26 | drmessano | I had 2 servers.. which I wanted to set up to replicate |
03:26.29 | drmessano | and it never worked |
03:26.39 | drmessano | well, worked for a month, then stopped |
03:26.45 | drmessano | Then worked, stopped.. worked, stopped |
03:26.53 | drmessano | Damn relay log corruption |
03:27.19 | drmessano | So I gave up.. its a hot standby.. just want to dump the DB and restore it once a day |
03:27.24 | drmessano | But couldnt even get that working |
03:27.38 | ScribbleJ | What DBMS? |
03:27.50 | ekat | ahh |
03:27.53 | jaytee | what about using a cronjob to stop the daemon, rsync the files to another server and then restart the daemon? |
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03:27.58 | ekat | yea i haven't messed much with replication lately |
03:28.33 | drmessano | Thats not a bad idea, jaytee |
03:28.44 | drmessano | I had considered it.. and it would work well |
03:29.16 | jaytee | but I think you'd have to stop the daemon on the target server too but I think you could use rexec for that or a scripted ssh command line instruction. |
03:29.34 | jaytee | mysql replication definitely sucks ass |
03:30.09 | ScribbleJ | jaytee, hey, I wanted to mention to you... I built a 'north' 'south' 'east' 'west' grammar in my system. |
03:30.15 | drmessano | Nothing like feeling like you never REALLY have a backup |
03:30.27 | ScribbleJ | jaytee, and... wouldn't you knwo it, it /can't/ get 'south' most of the time. |
03:30.50 | ScribbleJ | But now I'm on a mission to solve it and I've got some help from the CMU guys. |
03:31.57 | jaytee | there's just something "odd" about south. |
03:32.35 | jaytee | I guess I should be glad that it's not just Lumenvox |
03:32.47 | ScribbleJ | You're not kidding. I even added several different pronunciations to try out |
03:32.59 | ScribbleJ | Speaking of which - I had better luck if I made a point of pronouncing the 'th' at the end. |
03:33.08 | ScribbleJ | But removing it from the pronunciation didn't help too much. |
03:33.39 | drmessano | Ok, got my MySQL dump moved over and inserted.. so thats good |
03:34.32 | jaytee | according to Lumenvox's analysis of my sre response files from test calls it seems that most of the time it was missing the S at the beginning or picking up noise. I think maybe the recognition engine mistakes sss sounds for noise and dumps it. |
03:34.52 | ScribbleJ | Huh |
03:34.57 | drmessano | Did you try OUTH |
03:34.59 | ScribbleJ | I wonder if I set u... |
03:35.08 | ScribbleJ | just what was going to say, basically, drmessano . |
03:35.32 | ScribbleJ | "OW TH" maybe. I went the other way and tried "S OW" and "Z OW" |
03:35.46 | drmessano | Dump the S if its getting clipped as noise |
03:36.02 | ScribbleJ | Well, I'll try that myself in a little while. |
03:36.04 | jaytee | drmessano, yep, that was Lumenvox's suggestion. I put in an outh and added phonetics for "AW TH" which is what their phonetic list suggests. Made recognition worse instead of better. |
03:36.13 | drmessano | OW TH |
03:36.23 | drmessano | AW TH is way different IMO |
03:36.25 | jaytee | not according to their phoneme list |
03:36.36 | jaytee | I'd agree with you though |
03:36.37 | ScribbleJ | drmessano, it depends on their phonome set.. |
03:36.40 | ScribbleJ | right. |
03:36.46 | drmessano | heh |
03:37.03 | drmessano | AW TH would be appropriate for NORTH, if youre from up NAWTH |
03:37.17 | drmessano | CAW FEE |
03:37.32 | jaytee | the odd thing was after I added that to the grammar it wouldn't recognize east or west, just north and they came after south in the grammar list. I couldn't find any typos so I'm not sure what was going on. |
03:37.38 | drmessano | GOT A CAW TER FOR SOME CAW FEE |
03:37.43 | ScribbleJ | jaytee, that does sound odd, though... |
03:37.58 | ScribbleJ | jaytee, I'll try it without the 's' on mine and see how it looks. |
03:38.05 | drmessano | That was our answering machine message my sister put on the home machine years ago |
03:38.30 | drmessano | "Sorry you wasted a CATWER on the CAWL but we're out getting CAWFREE right now" |
03:38.35 | drmessano | Funny shit |
03:38.51 | drmessano | CAWTER |
03:38.53 | drmessano | bah |
03:39.30 | jaytee | I just made a test call, if I stress it by saying sss oww thhh it recognizes it. |
03:40.00 | drmessano | Maybe it the accent |
03:40.02 | drmessano | its |
03:40.28 | drmessano | SATH? heh |
03:40.30 | jaytee | yeah, ya mean cuz I pahk my cah in the hahvahd yahd? |
03:41.06 | drmessano | Hey, you ova dare.. lemme come ova dare and wrap a wrench arand ya neck, ya little joik |
03:41.26 | jaytee | so for a southern drawl I'd do "sss ay ow th" |
03:41.54 | drmessano | Nah, you just leave off the South.. |
03:42.03 | drmessano | "Over there in Carolina" |
03:42.06 | drmessano | "WHICH ONE?" |
03:42.29 | drmessano | Sad but true |
03:42.36 | drmessano | "Which direction will you be coming from?" |
03:42.42 | jaytee | "for directions please say the direction you are coming from unless it's south. If you're coming from the south say, "BUMFUCK!!!" |
03:42.43 | drmessano | "We're coming up there" |
03:42.58 | drmessano | "We're headed up yonder" |
03:43.01 | drmessano | HA |
03:43.06 | drmessano | "down yonder" |
03:43.08 | drmessano | There you go |
03:43.28 | drmessano | "We're coming from down yonder" |
03:43.50 | drmessano | I love the system Comcast has |
03:43.57 | carrar | WHAT |
03:44.03 | jaytee | I bet if I added "Fuck!" to the grammar but left it out the prompt after the 2nd repeat the average user would say, "Fuck!" and it would then go, "Coming from the south, take I-65 to exit 17 and head left....." |
03:44.13 | drmessano | "Ok. Tell me what sort of problem you're having so I can direct your call" |
03:44.18 | carrar | Such a POTTY mouth! |
03:44.24 | drmessano | So theres me at work |
03:44.30 | drmessano | "Shits broke" |
03:44.39 | drmessano | "Im sorry, i didnt understand your response" |
03:44.43 | drmessano | "Tell me what sort of problem you're having so I can direct your call" |
03:44.47 | jaytee | hahahaa |
03:44.50 | drmessano | "Um, damn internets broke" |
03:44.53 | drmessano | "Im sorry, i didnt understand your response" |
03:44.56 | drmessano | "Tell me what sort of problem you're having so I can direct your call" |
03:45.05 | drmessano | "Cant surf, damnit" |
03:45.16 | drmessano | "Ok, let me connect you to someone that can help" |
03:45.53 | drmessano | It doesnt understand when I try something like "cant connect to the internet" |
03:45.56 | drmessano | So screw it |
03:45.57 | ScribbleJ | I used to get the IVR at Comcast and say "fuck" and it would put me through to a real person immediately. |
03:46.06 | ScribbleJ | DUnno if it still does, but it was a useful trick. |
03:46.22 | drmessano | We tried that |
03:46.44 | ScribbleJ | It's actually understood me every time... but it annoys the living shit out of me. |
03:46.46 | drmessano | "Customer service" doesnt work either.. nor does "representative" |
03:46.51 | drmessano | or the name of the competition |
03:47.02 | jaytee | just say stuff like, "switching to U-Verse". bet that goes through really quick |
03:47.10 | ScribbleJ | i suppose it's kind of funny I've been working on this speech API plugin when IVRs annoy me so much. |
03:47.14 | drmessano | Although yelling "Knology" or "AT&T" over and over is fun |
03:47.30 | ScribbleJ | I'd buy U-verse right now if they offered it in my area. :( |
03:47.33 | drmessano | "Tell me what sort of problem you're having so I can direct your call" |
03:47.45 | drmessano | "I hate you.. all of you... I want to... BURN you...." |
03:48.09 | carrar | jaytee is becoming the enimey! |
03:48.15 | carrar | err |
03:48.20 | carrar | ScribbleJ is |
03:48.22 | drmessano | "The world is burning. RUN" |
03:48.23 | carrar | sorry jayteee |
03:48.25 | carrar | heh |
03:48.36 | drmessano | Thats one of my fav XKCDs |
03:48.55 | drmessano | http://xkcd.com/78/ |
03:48.57 | jaytee | my favorite is the stove ownership one |
03:49.32 | drmessano | My all time fav is A minus minus |
03:49.38 | jaytee | http://xkcd.com/418/ |
03:49.49 | drmessano | ROFL |
03:50.04 | drmessano | "instead of office chair, package contained bobcat" |
03:50.09 | drmessano | "would not buy again" |
03:50.37 | jaytee | hahahaaaa |
03:50.37 | drmessano | I have that taped to the wall |
03:50.53 | drmessano | http://xkcd.com/325/ |
03:51.15 | drmessano | black hat guy is always = WIN |
03:51.32 | jaytee | I loved the clip of an Ebay feedback page that had one entry that said, "If we were in prison together, I'd protect you in the shower! Highly recommend!!!" |
03:52.27 | drmessano | http://xkcd.com/538/ <-- This reminds me of Mr "I encrypt my config files, my hold music, AND my .call files /just in case/" |
03:53.01 | drmessano | Its like "what if i walk up the console with a $6 thumb drive?" "Oh, shit.. BRB" |
03:53.43 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
03:53.53 | jaytee | http://xkcd.com/276/ |
03:54.36 | drmessano | I love that one |
03:54.51 | drmessano | http://xkcd.com/440/ |
03:54.54 | drmessano | That one rocked |
03:55.25 | drmessano | "she'd be alive if it weren't for you" "Oh, god" |
03:56.48 | jaytee | http://xkcd.com/111/ |
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03:57.47 | drmessano | http://xkcd.com/350/ |
03:58.37 | drmessano | Who's a good virus? |
03:58.39 | drmessano | Yes you are |
03:59.05 | jaytee | http://xkcd.com/186/ |
04:00.50 | phix | xkcd <3 |
04:01.41 | jaytee | ok, this is from another channel: temugen> isn't it also the summation from 0 to infinity of (4/2n+1)(-1)^(n) |
04:01.48 | *** join/#asterisk ruyo (n=sayo@81.84.17.144) |
04:01.53 | jaytee | <jdong> I don't want to figure out what arctan(1) is in maclaurin series :) |
04:02.04 | jaytee | what are the odds that these two have never had sex? |
04:03.58 | phix | lol |
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04:18.30 | drmessano | ROFL |
04:19.04 | drmessano | jaytee: dungeonmasters do not have levels |
04:19.53 | giovani | drmessano: how would you know? ;) |
04:20.36 | jaytee | lol |
04:21.43 | drmessano | I have a +20 against your sarcasm |
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04:43.16 | drmessano | jaytee: Check my facebook status |
04:43.59 | drmessano | nevermind.. shit |
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04:44.15 | StanManCan | Fender.... |
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04:45.19 | StanManCan | I borked something and I can't figure it out for the life of me.... I just get silence when I dial out, verbose 3 doesn't show any errors, it's just...silence... Let me know if anybody's here and i'll pastebin my confs and debug |
04:45.24 | jaytee | drmessano, what about it? |
04:45.53 | drmessano | hang on.. I had a good status update and it borked it |
04:47.29 | drmessano | danny defines "database" as "binary storage of important data in an unstable, and deletion prone format". |
04:47.53 | drmessano | Should add "See also: MySQL is the MySUQ" |
04:48.24 | drmessano | The world is burning. RUN. |
04:50.10 | StanManCan | symlink /usr/src/asterisk/contrib/init.d to /etc/init.d/ |
04:50.17 | StanManCan | is that the right way to make asterisk run on boot ? |
04:50.44 | jaytee | StanManCan, what distro? |
04:51.09 | StanManCan | debian |
04:51.42 | jaytee | probably, but if I were you I'd google it to make sure. On RHEL or CentOS it's chkconfig asterisk on |
04:54.53 | neurosys | Can asterisk handle SMS to cellular carriers? |
04:55.04 | StanManCan | hmm, excepting i don't have a /usr/src/asterisk.. |
05:04.49 | drmessano | jaytee: on debian, depends how you hold your mouth |
05:05.41 | jaytee | I don't run * on debian or ubuntu or any of the debian based distros, just RHEL and CentOS |
05:05.58 | sipy | ~book |
05:05.59 | jbot | i guess book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
05:06.23 | jaytee | and if you don't have a /usr/src/asterisk it probably means you installed from packages..... |
05:06.30 | jaytee | ~wglwat |
05:06.31 | jbot | rumour has it, wglwat is well, good luck with all that |
05:07.47 | *** join/#asterisk _abc_ (n=Peter@bas3-toronto06-1177698468.dsl.bell.ca) |
05:08.08 | _abc_ | is there a tiny iso with asterisk on it ? the 50mb business cd kind ? |
05:10.41 | rob0 | If so it would be tight, since asterisk alone would probably be more than half of that. |
05:11.03 | rob0 | well, I'm not sure, I installed some extra sounds |
05:11.46 | rob0 | mine is 26MB |
05:12.35 | StanManCan | yea i installed by hand |
05:12.48 | StanManCan | download packages and installed |
05:13.03 | StanManCan | er.. the tarballs |
05:13.43 | _abc_ | i heard of asterisk running in a wrt54gl. so there must be a way to squeeze it |
05:14.04 | drmessano | Apples and oranges |
05:14.25 | jaytee | packages and tarballs, two different animals |
05:14.46 | drmessano | 1 is talking about putting the install on a 50MB CD, I think.. the other is talking about putting a compiled binary asterisk on a WRT54GL |
05:15.00 | drmessano | Not even close to the same comparison |
05:15.08 | _abc_ | awe and wonder, and this is not the one i saw, this is what i found now: http://www.kvaes.be/unix-linux/installing-asterisk-on-a-linksys-wrtg/ |
05:15.35 | _abc_ | i was talking about a live iso with asterisk on it |
05:15.50 | _abc_ | the wrt54g has at most 16MB flash afaik, usually 4 or 8 |
05:16.25 | drmessano | and is also a completely different platform |
05:16.30 | _abc_ | please put _abc_ in the line if you want me to react i am busy otherwise |
05:16.46 | drmessano | No, I cant be bothered |
05:16.52 | _abc_ | drmessano, yes, mips binaries are larger |
05:17.05 | _abc_ | drmessano, just _a TAB please |
05:17.19 | rob0 | Um, I think better ettiquette is that people who ask questions are expected to read the channel. |
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05:17.30 | drmessano | rob0: No shit |
05:17.35 | rob0 | and did I spell that right? |
05:17.44 | _abc_ | i read the channel of course but /whois _abc_ will tell you i have a lot of channels to read |
05:17.55 | drmessano | heh |
05:18.17 | rob0 | http://en.wikipedia.org/wiki/Etiquette |
05:18.21 | rob0 | I guess not |
05:18.30 | _abc_ | netiquette :) |
05:18.30 | rob0 | one of the few I always get wrong |
05:18.49 | drmessano | rob0: Sorry, if you are gonna post hyperlinks, can you please VNC to my workstation and type them in my browser for me |
05:18.53 | drmessano | rob0: Very busy here |
05:19.02 | rob0 | Sorry drmessano, I will. |
05:19.53 | neurosys | I guess Email gateways are the best way to get my system to text cell carriers :P |
05:20.05 | drmessano | Yes |
05:20.10 | drmessano | less hassle |
05:20.25 | drmessano | They're actually very reliable |
05:21.15 | rob0 | How about inbound SMS, do you need a special service for that? I can email my cell, but it can't reply (no SMS-to-email gateway) |
05:21.34 | _abc_ | is there a gateway from asterisk iax to S7 ? |
05:21.42 | drmessano | Yeah, Asterisk |
05:22.11 | drmessano | IAX <> ASTERISK <> libss7 |
05:22.19 | _abc_ | asterisk does S7 ? i need to go read on it. i haven't touched a new version in 1 year |
05:22.32 | _abc_ | ok, thanks |
05:22.32 | Qwell | SS7, not S7. |
05:22.36 | _abc_ | sure |
05:23.23 | _abc_ | i wonder how hard it is to backport the wrt54 asterisk install to x86 tiny bootable iso. probably not hard. |
05:24.02 | Qwell | O.o |
05:24.31 | drmessano | lol |
05:24.33 | _abc_ | by backport i mean compile natively instead of cross and graft it on grml or similar ( http://grml.org/ ) |
05:24.55 | _abc_ | what is so funny |
05:24.57 | drmessano | By backport I mean, not using backport in any sense of the word |
05:26.02 | Qwell | So you want to install it normally |
05:26.24 | _abc_ | well wrt54gl oopen source firmware is a cross compiled set of packages that should be compilable for x86 as is just leaving out the platform specific tools. the core is busybox based anyway. |
05:26.39 | _abc_ | and no, it is not so normal imho. |
05:28.15 | neurosys | drmessano: smsm email gateways seem to be very quick too. Curious how quick duing peek hrs. |
05:29.08 | drmessano | neurosys: About the same.. I get a weather bulletin out to phones in about the same time as e-mails hit standard inboxes |
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05:54.15 | _abc_ | there is even a version for nslu2 apparently |
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05:54.47 | [TK]D-Fender | _abc_: Don't expect performance after something nick-named a "slug" |
05:54.58 | _abc_ | hehe |
05:55.05 | _abc_ | just reliability would be nice |
05:55.38 | [TK]D-Fender | _abc_: Buy a 10$ phone at the drug store and run it off your telco's analog service. Ther you go |
05:56.19 | _abc_ | lol |
05:56.23 | _abc_ | ok, i get the spirit |
05:56.56 | drmessano | Make it look like a sunflower |
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05:57.54 | drmessano | Effin Steve Jobs |
05:58.44 | drmessano | Sad they diagnosed him with Cancer.. That just shows me how crappy medical science is. So much they had to have missed. |
06:01.18 | [TK]D-Fender | "We don't what's wrong" = cancer |
06:07.26 | neurosys | He's been officially diagnosed with Cancer? |
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06:11.28 | drmessano | Like 2 years ago |
06:11.43 | drmessano | He's had pancreatic cancer |
06:11.52 | neurosys | Oh yeah. but didnt that go into remissio or something? Is that what the problem is again? |
06:12.44 | drmessano | Rumor has it he's HIV positive.. but so far just a couple liveleak documents that may be forged |
06:12.56 | neurosys | hehe |
06:15.07 | *** part/#asterisk neurosys (n=vinix@c-71-196-9-142.hsd1.fl.comcast.net) |
06:15.13 | *** join/#asterisk neurosys (n=vinix@c-71-196-9-142.hsd1.fl.comcast.net) |
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06:24.13 | baliktad | Asterisk 1.6.0.5 is crashing when I park a call: http://pastebin.ca/1330559 |
06:24.59 | neurosys | drmessano: LOL you still awake? |
06:25.15 | drmessano | yes |
06:25.22 | neurosys | http://www.aim-med.org/library/articles/1192638115/ |
06:25.27 | neurosys | founded in 2006 hehe |
06:25.50 | neurosys | those leaks say that company tested him in 04' :P |
06:28.34 | drmessano | The company has been around for years.. the claim is that they are a reissue of old results |
06:30.05 | drmessano | Im not saying I necessarily believe the documents to not be fake, but issue of the age of the company is the weakest argument |
06:31.07 | neurosys | Oh, saying the documents are simply under a new letterhead? |
06:31.24 | neurosys | i gotcha. well.. i hope for his sake it's not true :P |
06:33.10 | drmessano | Well, I honestly hope it's not.. Him covering up having AIDS with a pancreatic cancer story would tell me he's still living in the 1980s.. |
06:33.43 | drmessano | He may as well at that point issue a press release "I have AIDS... and I am not gay".. |
06:33.53 | drmessano | He'd get the drmessano WTF of the year award |
06:34.06 | neurosys | Too right. :) |
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07:03.02 | Zippoman | anyone ever used any sms gateways...to recommend? |
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07:53.58 | DelphiWorld | hi my friends. how are you ? please, try to give me a book (PDF or CHM format) about asterisk PBX. Thanks! |
07:54.16 | carrar | ~book |
07:54.17 | jbot | book is, like, probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
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08:17.05 | *** join/#asterisk DelphiWorld (n=Miranda@41.201.96.184) |
08:18.45 | DelphiWorld | my friends: please give me a asterisk book in CHM format if pocible |
08:24.00 | MaliutaLap | ~bool |
08:24.01 | jbot | i heard bool is bool, nothing else need be said. |
08:24.04 | MaliutaLap | ~book |
08:24.05 | jbot | rumour has it, book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
08:24.12 | MaliutaLap | 'tas all |
08:30.57 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
08:37.00 | DelphiWorld | MaliutaLap: are you able to download the PDF book ? |
08:37.52 | MaliutaLap | yes, I have copies everywhere |
08:39.34 | DelphiWorld | please, is it pocible to send to me a copy ? |
08:39.45 | DelphiWorld | bicose i'm no able to use it ofline |
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09:37.41 | estr4ng3d | How do you enable zap debug? |
09:38.31 | *** join/#asterisk oej (n=olle@LIS-DHCP-11.ulb.ac.be) |
09:48.54 | ScribbleJ | I zap debug with de bug zapper. |
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09:53.14 | DelphiWorld | hi my friends |
09:53.51 | DelphiWorld | please, what is the realy best linux distribution to run asterisk ? (simulare to TrixBox) |
09:54.06 | Maliuta | there are no friends, only people screaming RTFM! |
09:55.07 | DelphiWorld | Maliuta: i love open source users. then this is my friends. |
09:55.19 | ScribbleJ | BSD, this is the best linux distro for Asterisk. |
09:55.26 | Maliuta | DelphiWorld: run slackware |
09:55.33 | ScribbleJ | <- real jerk. |
09:55.34 | ScribbleJ | Heh |
09:55.37 | Maliuta | or build your own distro |
09:56.51 | DelphiWorld | ScribbleJ: BSD dont have asterisk by default |
09:57.03 | DelphiWorld | Maliuta: please ho to build my hone ? |
09:57.20 | DelphiWorld | i'm next to Linux from scratch but is no easy for me |
09:58.27 | Maliuta | distro is all about preference |
09:59.36 | carrar | DelphiWorld, you should use LFS (Linux From Scratch) |
09:59.38 | DelphiWorld | Maliuta: please try to help my friend to bring me to the open source community |
09:59.43 | DelphiWorld | i'm using only a windows PBX |
09:59.49 | DelphiWorld | but i want to switch to linux |
10:00.21 | ScribbleJ | In this case, I suggest something user-friendly, like Ubuntu, or something with paid corporate support, like RedHat. |
10:00.51 | carrar | CentOS works very well also |
10:01.00 | DelphiWorld | ScribbleJ: you dont understand me |
10:01.13 | ScribbleJ | CentOS is the paid of RedHat without the support.... :( |
10:01.14 | DelphiWorld | ScribbleJ: i want a distro simulare to trixbox (if you know it) |
10:01.18 | ScribbleJ | paid = pain |
10:01.29 | ScribbleJ | DelphiWorld, Sorry, I don't. Why this requirement? |
10:01.37 | carrar | CentOS works awesome |
10:01.56 | carrar | Who cars about paid |
10:01.58 | carrar | cares |
10:02.01 | DelphiWorld | i want to start a very very small PBX a my home |
10:02.05 | ScribbleJ | I'm sure it works great... I have personally issues with Redhat's package management and naming scheme. |
10:02.15 | ScribbleJ | DelphiWorld, I do this. I use Ubuntu. |
10:02.15 | carrar | like you are gonna get commercia support for opensource Asterisk |
10:02.40 | ScribbleJ | carrar, I believe you can get that all day long from Digium. |
10:02.46 | carrar | nope |
10:02.55 | carrar | They will not support open source |
10:03.02 | DelphiWorld | carrar: do you know trixbox ? |
10:03.04 | carrar | only their Buisness eddition |
10:03.08 | ScribbleJ | Oh, I see. |
10:03.20 | ScribbleJ | Hrm |
10:03.34 | carrar | DelphiWorld, you might try #trixbox |
10:03.39 | ScribbleJ | That makes me wonder if we bought Business Edition for our IVR. Guess we must have. |
10:04.14 | DelphiWorld | i dont wabnt to use trixbox |
10:04.20 | carrar | then stop talking about it |
10:04.24 | DelphiWorld | ui'm looking for other distro simulare to it |
10:04.31 | carrar | then use it |
10:05.39 | DelphiWorld | carrar: realyty: i want to use a distro based on debian no CentOS (RedHat) |
10:05.57 | carrar | I use Asterisk OpenSource on CentOS |
10:06.01 | ScribbleJ | DelphiWorld, so what was wrong with my Ubuntu suggestion? |
10:06.04 | carrar | works great |
10:06.14 | ScribbleJ | I use it on Ubuntu, and Debian both, works fine. |
10:06.15 | carrar | Ubuntu works great too |
10:06.21 | carrar | they all do |
10:06.29 | DelphiWorld | ScribbleJ: your ubuntu suggestion is realy cool. but asterisk is no pre configured! |
10:06.30 | carrar | DelphiWorld, it's personal pref |
10:06.37 | ScribbleJ | DelphiWorld, wrong, it is too. |
10:06.51 | carrar | if you go with pre configured why bother with Asterisk open source? |
10:06.52 | ScribbleJ | DelphiWorld, just click to install it in the package manager, comes complete witha working same config. |
10:06.56 | ScribbleJ | same = sample |
10:07.05 | carrar | You are missing a lot of things |
10:07.37 | DelphiWorld | ScribbleJ: but include a web interface (frontend) ? |
10:07.37 | carrar | google for building asterisk from source |
10:07.50 | carrar | again, wrong channel |
10:08.16 | DelphiWorld | carrar: then what you suggest to me ? |
10:08.28 | carrar | what are your requirements? |
10:08.45 | ScribbleJ | There is no web interface for ASterisk in Ubuntu, but honestly configuring it from the config files is not rocket surgery. |
10:08.51 | DelphiWorld | carrar: please heare me: |
10:08.57 | DelphiWorld | 1. a small linux distro |
10:09.00 | carrar | don't waste your time with web front ends |
10:09.11 | DelphiWorld | 2. setting up asterisk for home usage |
10:09.25 | DelphiWorld | 3. a web interface (frontend) to configure my asterisk |
10:09.32 | ScribbleJ | 1. Ubuntu Server. 2. aptitude install asterisk |
10:09.38 | ScribbleJ | 3. Doh! |
10:09.43 | *** join/#asterisk brunner (n=chris@68-119-87-106.dhcp.mtgm.al.charter.com) |
10:09.51 | carrar | DelphiWorld, why do you need web? |
10:09.59 | carrar | Why can't you learn to you command line? |
10:10.04 | carrar | use |
10:10.10 | brunner | I just got a phone call through IPKall, but they haven't told me my phone number yet! |
10:10.26 | ScribbleJ | brunner, so, who was calling?!? Ar telemarketers THAT good now? |
10:10.33 | DelphiWorld | carrar: probably you dont know me |
10:10.41 | DelphiWorld | carrar: i'm a blind user using a screen reader |
10:10.45 | brunner | the call was from sip:4153385516@66.54.140.46 |
10:10.48 | ScribbleJ | What! |
10:10.50 | ScribbleJ | What! |
10:10.53 | DelphiWorld | i want to access / configure asterisk from windows |
10:10.55 | ScribbleJ | I'm sorry, I'll say it once more. |
10:10.56 | ScribbleJ | What! |
10:11.01 | DelphiWorld | this is the frontEnd Usage reason |
10:11.17 | ScribbleJ | DelphiWorld, you're the first blind person I ever met who would prefer a web browser to a commandline. |
10:11.26 | brunner | I can't call them back through IPKall, of course, but I tried with my cell phone and it said it was an invalid pager number |
10:12.07 | ScribbleJ | Using a web browser with a screenreader or braille terminal is a nightmare. |
10:12.37 | DelphiWorld | ScribbleJ: i'm using only TTS no brail terminal |
10:12.43 | carrar | ssh into your asterisk box from your windows box that has a screen reader |
10:12.52 | carrar | problem solved! |
10:13.39 | ScribbleJ | DelphiWorld, yeah, I'm sighted, but I notice my blind friends prefer TTS to the braille terminal. It's just faster, I suppose. Do you use somehting like FEstival only with the speed cranked up to chipmunk-speed? |
10:13.43 | DelphiWorld | carrar: SSH i know only the populare "PUTTY" SSH client but no accessibkle to me! |
10:14.39 | carrar | Try SecureCRT |
10:14.45 | DelphiWorld | ScribbleJ: i know a screen reader for the GNOME Desktop named "orca" that use ESpeak |
10:14.50 | ScribbleJ | DelphiWorld, there are plenty of ways to set up Linux for blind folks; like I mentioned, FEstival can be used for TTS. You might try it all sometime, it's a better intro to linux than ASterisk, and once you go tthat down, ASterisk'll seem way easier. |
10:15.26 | ScribbleJ | Sorry, "once you got that down" - I bet parsing typos on a TTS is a real bother. |
10:15.30 | DelphiWorld | carrar: ho to download it ? |
10:16.28 | DelphiWorld | ScribbleJ: i want to use ubuntu server, but dont have a screen reader included |
10:17.04 | ScribbleJ | DelphiWorld, I can imagine getting it set up initially might be a bitch. Maybe get a sighted linux nut to help out. I'd think it would be an interesting project. |
10:18.55 | DelphiWorld | ScribbleJ: what you thank if i start a new linux distro that have the GNOME desktop and a pre included ORCA screen reader to setup it ? |
10:19.16 | DelphiWorld | bicose ubuntu have orca, but not working if i start the setup program |
10:19.35 | ScribbleJ | DelphiWorld, I think that would be cool, but possibly missing the boat. If I were blind, I would imagine I would like the fact that I can accomplish everything in linux from a non-graphical environment. |
10:19.43 | *** part/#asterisk Mog (n=mog@c-68-62-170-242.hsd1.al.comcast.net) |
10:20.11 | DelphiWorld | ScribbleJ but no easy |
10:20.42 | ScribbleJ | DelphiWorld, I wonder how easy it would be; in theory it couldn't be much more complicated than piping your tty through festival. |
10:21.04 | ScribbleJ | DelphiWorld, or you mean, the linux commandline isn't easy? Heh |
10:21.19 | ScribbleJ | DelphiWorld, All I can say to that is, it's easier than Windows once you know it as well. |
10:21.25 | DelphiWorld | ScribbleJ: faistival is only a TTS |
10:21.32 | DelphiWorld | require a program to use faistival |
10:21.55 | ScribbleJ | DelphiWorld, yeah, I'm surprised one isn't out there already for that though; seems like an obvious application of it. |
10:22.36 | DelphiWorld | orca use ESpeak by default but is pocible to setup faistival and configure it to use it |
10:22.57 | ScribbleJ | Espeak might be better for all I know, I'm not familiar with it. |
10:23.24 | DelphiWorld | ScribbleJ: no |
10:23.29 | DelphiWorld | faistival is best |
10:24.10 | ScribbleJ | DelphiWorld, I heard 'cepstral' recently, which is a clone of festival, but with better data, and it costs money to buy... but holy cow does it sound nice, almost like a real person. |
10:24.47 | DelphiWorld | ScribbleJ: but probably that dont have a driver for GNOME |
10:24.56 | ScribbleJ | DelphiWorld, Yeah, probably, I have no idea. |
10:25.17 | *** join/#asterisk masus (i=masus@88.248.14.186) |
10:25.35 | DelphiWorld | ScribbleJ: what about asterisk billing ? |
10:25.38 | DelphiWorld | how is work ? |
10:26.20 | ScribbleJ | I don't know much about it, actually. My involvement with Asterisk so far has been limited to a little bit of SIP and a lot of TTS and Speech Recognition. |
10:27.15 | DelphiWorld | then cool! |
10:27.23 | DelphiWorld | i dont know any informations about asterisk |
10:27.28 | DelphiWorld | i'm only a new user |
10:29.24 | DelphiWorld | ScribbleJ: i try trixbox and tel you later |
10:29.30 | ScribbleJ | Good luck! |
10:31.03 | DelphiWorld | ScribbleJ: thanks |
10:47.16 | DelphiWorld | please what is a zapcart ? |
10:54.30 | brunner | does anyone have an IPKall number I could use for testing? |
10:56.01 | DelphiWorld | brunner: do you want to make a call ? |
10:56.33 | brunner | DelphiWorld: no... I just to be able to receive a call on my asterisk box |
10:56.59 | DelphiWorld | brunner: then you want to i call you ? |
10:57.47 | brunner | DelphiWorld: well it would make testing easier if I could call it myself more than once as needed. I'll just wait for IPKall to process my application or sign up with a voip provider. |
10:57.54 | DelphiWorld | send to me your sip uri |
11:03.04 | ScribbleJ | Brunner, I tried voicepulse and broadvoice... broadvoice'll turn you on the minute you subscribe and if you never sign their 911 form they'll autocancel your account after a month and give you a refund. |
11:03.28 | ScribbleJ | I was pretty impressed with them to be honest... but I stuck with voicepulse instead. |
11:05.08 | brunner | ScribbleJ: so this is unlimited in-bound, right? http://www.broadvoice.com/rateplans_unlimited_state.html |
11:05.34 | ScribbleJ | brunner, both voicepulse and broadvoice are unlimited inbound - the niice thing about broadvoice is they also offer unlimited /outbound/ |
11:05.49 | brunner | not to be picky, but is there some low-quality provider that offers even cheaper inbound service that I could use for testing? |
11:06.10 | ScribbleJ | No, that's not picky. I do not know personally, those are the only two I have used. |
11:06.35 | ScribbleJ | How long do you need it for, brunner? |
11:06.48 | brunner | a week? maybe a month? |
11:07.00 | ScribbleJ | Ah, yeah. Well. |
11:07.30 | ScribbleJ | I signed up for broadvoice, paid the fee, never signed their 911 form they require, and they refunded me the full amount. |
11:07.31 | brunner | IPKall is probably the only thing out there. if they don't respond in another day or so, I'll just use broadvoice |
11:07.40 | ScribbleJ | So... doesn't get much freer than that. |
11:07.48 | brunner | yeah, that's cool |
11:07.55 | brunner | I appreciate the heads pu |
11:07.57 | brunner | up* |
11:08.04 | ScribbleJ | IT's kind of taking advantage to do that on purpose I guess, but... it is what it is. |
11:08.17 | brunner | I'll just pay the $10, heh |
11:08.29 | brunner | or do they require a contract? |
11:08.56 | ScribbleJ | I don't remember if they want you to sign up for long-term, I think it was month-to-month. |
11:09.03 | ScribbleJ | I never made it past the first month myself, of course. :) |
11:10.17 | brunner | yeah |
11:10.34 | brunner | well I don't mind paying the $10 for a month of service, as long as there's no contract |
11:10.36 | ScribbleJ | I'm going to get a 2nd number from VoicePulse tongiht, actually, to put on my website as a demo of this Speech Rec system I'm working on. |
11:11.20 | brunner | hell, I might just hack my roommate's vonage adapter and get him to switch so I can make use of the sip line at night to test asterisk |
11:11.39 | brunner | cool. what's your website? |
11:12.20 | ScribbleJ | It's scribblej.com - there's nothing there yet but check back in 6 hours and you'll either find a free replacement for LumenVox.... or... I'll be dead, I suppose, could happen anytime. |
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11:13.01 | brunner | k |
11:13.05 | ScribbleJ | Vonage adapter, hah. I /just/ got a Linksys PAP2T-NA today, haven't had time to do more than hook it up. |
11:14.14 | brunner | woah, perfect: http://www.broadvoice.com/rateplans_byod.html |
11:14.37 | ScribbleJ | Oh yeah, that's the plan I signed up for I guess. |
11:15.00 | brunner | oh, I didn't see all the fees |
11:15.15 | ScribbleJ | Oh, except I did BYOD-Unlimited-World, I think |
11:15.17 | brunner | that sucks. I wish IPKall would just hurry up and approve the application |
11:15.54 | ScribbleJ | Huh |
11:16.15 | ScribbleJ | I haven't heard of IPKall but I've been looking for some way to get a SIP termination for free... |
11:16.20 | ScribbleJ | How can they do that?? |
11:16.40 | brunner | terminative minutes |
11:16.58 | brunner | on the PSTN, whoever the call terminates with gets paid |
11:17.03 | brunner | by the other telcos |
11:17.07 | ScribbleJ | Oh, I see. |
11:17.17 | ScribbleJ | So basically they are gambling on more inbound thn outbound? |
11:17.18 | brunner | so they actually make money on it |
11:17.28 | brunner | they won't do outbound at all |
11:17.31 | ScribbleJ | That's pretty clever. |
11:17.31 | brunner | except toll free |
11:17.32 | brunner | bbr |
11:17.57 | ScribbleJ | HUH! |
11:18.05 | ScribbleJ | I wonder what it takes to set up a system like that. |
11:18.14 | ScribbleJ | That would be a fun service to provice and 'free money' |
11:18.40 | brunner | yeah |
11:18.48 | brunner | they don't offer customer support at all |
11:19.02 | ScribbleJ | Right, that kind of thing, you can basically set it up and forget it. |
11:19.06 | ScribbleJ | Automate the whole thing. |
11:19.20 | brunner | yeah, until the spammers discovered it |
11:19.21 | ScribbleJ | I'd have no idea where to start on the PSTN side though. |
11:19.29 | brunner | now there's a long waiting period |
11:19.54 | ScribbleJ | There's ways to try to automate some of that anyhow... we automate background checks and things for our business, but of course that gets cost prohibitive. |
11:20.26 | brunner | a company in Pennsylvania is basically letting me use a quad xeon asterisk box for the same reason |
11:20.30 | ScribbleJ | Still, you could do a some simple tests... and ACH is cheap, depending on how profitable it is you could ACH a coupel pennies to each person to verify them. |
11:20.39 | brunner | just so they make money on the inbound calls, I mean |
11:20.49 | ScribbleJ | Of course that takes at least a day. |
11:20.52 | brunner | and it's not costing me a penny |
11:21.24 | brunner | you wouldn't need to do ACH |
11:21.41 | brunner | just require they confirm they have a real alternate phone number in the US |
11:21.54 | ScribbleJ | Well, you could automate /that/ easy with ASterisk. |
11:21.58 | brunner | you call them, they enter a code online, and it's done |
11:22.02 | brunner | right |
11:22.02 | ScribbleJ | Give them a code # to punch in, cl... yeah |
11:22.07 | ScribbleJ | Heh |
11:22.38 | ScribbleJ | I don't get how you have to be hooked into the phone network to a) get phone numbers to give out and b) get on the dole for incoming calls. |
11:23.45 | brunner | yeah, I don't really understand it, either... I'll have to ask my friend about it again sometime |
11:24.01 | brunner | he works for a phone company that makes money just on that |
11:24.14 | brunner | they offer free conference services and such |
11:24.14 | ScribbleJ | I'd guess you have to pay someone something for the block of #s up front, I wonder if you could optimize that though - people are getting a free service, maybe they'd be cool with sharing numbers and only getting an /extension/ on your automated phone tree. |
11:24.38 | brunner | nah, numbers are super cheap wholesale |
11:25.32 | brunner | there'd be no need for sharing |
11:25.45 | ScribbleJ | Hrm...! |
11:25.50 | *** join/#asterisk Maliuta (n=scooby@kiev.lusan.id.au) |
11:25.52 | ScribbleJ | I'm going to have to find out more about this. |
11:25.58 | *** join/#asterisk estr4ng3d (n=sm4rt@196.219.96.141) |
11:26.13 | ScribbleJ | If it didn't require too much capital I might do it myself.... and if it did I know some people I could pitch it to. |
11:27.18 | brunner | I really don't think it's worth it |
11:27.37 | brunner | you'd have to physically exist in a state with screwed up regulations |
11:27.37 | ScribbleJ | Not that much money in the incoming calls? |
11:27.41 | ScribbleJ | Hah |
11:27.44 | brunner | and then you'd have fraud out the ass |
11:28.02 | brunner | it just really wouldn't be worth it |
11:28.12 | brunner | otherwise, there'd be a lot more people doing it |
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11:42.51 | estr4ng3d | How do you enable zap debug? |
11:49.04 | MaliutaLap | you zap de-bug wiss de bug zapper |
11:49.25 | MaliutaLap | takes loose interpretations |
11:56.31 | ScribbleJ | Dude. |
11:56.39 | ScribbleJ | I so made that joke hours ago. |
11:57.08 | MaliutaLap | ScribbleJ: I improved on it |
11:57.15 | ScribbleJ | 2 hours ago: <estr4ng3d> How do you enable zap debug? <ScribbleJ> I zap debug with de bug zapper. |
11:57.26 | ScribbleJ | Hah, sheesh |
11:57.40 | MaliutaLap | ScribbleJ: I _could_ have told him the compiler options for gdb |
11:57.58 | MaliutaLap | ScribbleJ: bah! _your_ telling was weak :P |
11:58.03 | ScribbleJ | I suppose actually being helpful is beyond either of us. |
12:01.09 | MaliutaLap | for something documented? |
12:01.37 | MaliutaLap | if people actually read some manuals and tried things on the CLI ... |
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12:02.29 | SparFux | regular phones suck! I wish I only had IP phones :-( All a mess here in my stupid house! |
12:02.59 | MaliutaLap | hugs his cisco |
12:03.24 | MaliutaLap | I scared a troll off on #linpeople! |
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12:24.33 | markit | I'm looking for a gsm plugin for kaffeine or other player in GNU/Linux, any idea where to find it? |
12:25.02 | markit | (debian sid) |
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12:29.36 | [gnubie] | anyone knows if there is a repository on binary debian etch package for asterisk 1.4.23.1? |
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12:54.15 | nvrpunk | which sound is the ringback sound in /var/lib/asterisk/sounds ? |
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13:01.41 | *** join/#asterisk Dovid (n=annon@tony09-118-62.inter.net.il) |
13:02.05 | Dovid | is the presentation number on a PRI the same as CID on callerid(num) on sip ? |
13:09.24 | nvrpunk | how come when I have a voice prompt context for inbound calls to dial an extensin, the ring isn't passed to the caller? |
13:09.35 | nvrpunk | ie it's blank until the person answers their extension |
13:12.33 | Maliuta | nvrpunk: because you didn't pass the right option to dial? |
13:12.44 | Maliuta | nvrpunk: because you don't have any tones on the system? |
13:13.00 | Maliuta | nvrpunk: because the planets are out of alignment? |
13:13.16 | Maliuta | nvrpunk: you haven't given us anything to base our advice on |
13:13.29 | nvrpunk | it |
13:13.42 | Maliuta | lets start with some dialplan action\ |
13:13.48 | nvrpunk | its and AGI Script setting the voice prompt system |
13:13.53 | nvrpunk | it's not just a config |
13:14.04 | Maliuta | then we need to see the script |
13:14.14 | nvrpunk | if you would like to see 10 pages of php sure |
13:14.15 | nvrpunk | :) |
13:14.22 | Maliuta | it's that simple, you give us info an we'll see if we can help |
13:14.25 | Maliuta | ~pastebin |
13:14.26 | jbot | [~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
13:14.44 | nvrpunk | ill be back, im going to struggle with it for a bit |
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13:14.47 | nvrpunk | before I give up |
13:14.48 | nvrpunk | :P |
13:15.18 | nvrpunk | I do have one question though |
13:15.21 | nvrpunk | that should be simple |
13:15.22 | nvrpunk | [Feb 8 16:06:35] WARNING[20997]: channel.c:2930 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x40 (slin) |
13:15.25 | Maliuta | look at the manual for dial |
13:15.35 | Maliuta | nvrpunk: that's simple |
13:15.36 | nvrpunk | I don;t know what it's trying to transcode or why |
13:15.43 | nvrpunk | there should be nothing |
13:15.48 | Maliuta | nvrpunk: you don't have g729 installed |
13:15.48 | nvrpunk | to rings have to be transcoded? |
13:15.54 | nvrpunk | of course not |
13:15.59 | Maliuta | everything has to be transcoded |
13:16.02 | Maliuta | why not? |
13:16.02 | nvrpunk | im using g729 all the way through |
13:16.04 | nvrpunk | no |
13:16.13 | Maliuta | what codec are the files in? |
13:16.19 | nvrpunk | g729 |
13:16.20 | nvrpunk | all of em |
13:16.21 | nvrpunk | :P |
13:16.29 | Maliuta | obviously not |
13:16.41 | Maliuta | otherwise it wouldn't need to transcode |
13:16.42 | nvrpunk | but its not pointing me to what its looking for |
13:16.47 | nvrpunk | thats not g72 |
13:16.51 | nvrpunk | which is bullocks |
13:17.10 | nvrpunk | i cant fix something without proper debug messages! |
13:17.11 | nvrpunk | :P |
13:17.43 | nvrpunk | so how am I supposed to find which file is not g729? |
13:17.44 | nvrpunk | :p |
13:20.16 | Maliuta | and the output of 'module show like sln' is??? |
13:20.43 | Maliuta | it is a useful message if you use you brain |
13:21.07 | nvrpunk | 1 modules loaded |
13:21.19 | nvrpunk | raw sign linear audio support |
13:21.25 | nvrpunk | signed* |
13:21.48 | nvrpunk | if I wanted to use my brain I would never use a computer :) |
13:24.53 | Maliuta | I don't have time for this, I have to be at the hospital in 7 hours for my hip replacement |
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13:30.29 | ScribbleJ | Very beta version of the free Asterisk Generic Speech API engine I've put together: http://scribblej.com/svn/ |
13:31.56 | ScribbleJ | Uh, that's not actually a subversion repo... maybe I should rename that directory. |
13:55.19 | estr4ng3d | How do you enable zap debug? |
14:02.29 | nvrpunk | - Executing [11076@makecall:1] Dial("IAX2/jackal-8426", "SIP/192.168.2.8/11076||S(6000)r") in new stack |
14:02.37 | nvrpunk | it's still not rnging back to the caller |
14:03.19 | *** part/#asterisk markit (n=marco@88-149-177-66.static.ngi.it) |
14:04.15 | Daejeo | nvrpunk: if you want use computer, you also need to use your brain |
14:05.00 | nvrpunk | hmm |
14:05.18 | nvrpunk | so if i have a gotoif that passes a call to another context that has a dial |
14:05.22 | nvrpunk | and using the r option |
14:05.26 | nvrpunk | will it pass back the ring? |
14:05.49 | nvrpunk | what I am really wondering is what stops the ring from passing back to the caller |
14:05.52 | nvrpunk | in general |
14:05.58 | nvrpunk | what specifically stops it, ever |
14:06.41 | nvrpunk | Daejeo, and as I don't have a lot of time to do all the things I need to do in a day and considering my salary is quite high, this is just a side thing to figure out |
14:06.44 | nvrpunk | a minor annoyance |
14:06.58 | nvrpunk | and in the end, I could just pay someone to do it :) |
14:07.18 | nvrpunk | money > brain power |
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14:22.19 | [gnubie] | tzafrir_laptop: hello.. are you already freezing to version 1.4.21.2? |
14:22.34 | tzafrir_laptop | hi |
14:22.37 | tzafrir_laptop | probably |
14:23.11 | [gnubie] | tzafrir_laptop: what about the fixes made from the upstream from 1.4.22 and the current one? |
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14:45.36 | masus | Hi all, Newbie Question. Is it possible to get the dialstatus if we place a call to /var/spool/asterisk/outgoing ? Thanks All. |
14:50.47 | masus | Or how to handle outgoing.call files ? |
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15:38.16 | Dovid | anyone use the $ams-> "function" with phpAgi ? |
15:48.08 | *** join/#asterisk BadHAL (n=nn@cpe-72-179-194-139.stx.res.rr.com) |
15:52.44 | *** join/#asterisk oej (n=olle@LIS-DHCP-11.ulb.ac.be) |
15:54.23 | *** join/#asterisk markit (n=marco@88-149-177-66.static.ngi.it) |
15:55.33 | markit | hi, I'm updating italian suond set for asterisk 1.4.x (previous was 1.2.x). I need some help in understanding the meaning of some sounds. conf-adminmenu-162.gsm, what is the meaning of "extend the conference"? let more people enter, or last more time? |
15:55.42 | markit | (never used conferences myself, sigh) |
15:58.04 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
16:00.28 | *** join/#asterisk jicksta (n=jicksta@c-67-169-165-162.hsd1.ca.comcast.net) |
16:07.24 | *** join/#asterisk elux (n=pak@24.114.233.2) |
16:07.26 | elux | hey guys |
16:07.58 | elux | im trying to understand what asterisk does in the voip-stack. i understand its at the lowest level, which is like an SIP-daemon? |
16:08.01 | elux | am i getting that right |
16:12.02 | [gnubie] | waves to all.. gtg.. thanks.. ;-) |
16:14.10 | *** join/#asterisk LtScarr (i=benno@palm.hoeg.nl) |
16:15.02 | LtScarr | ey everyone |
16:15.10 | elux | hi |
16:15.36 | LtScarr | is this a right channel to ask end-user questions about asterisk? |
16:15.51 | LtScarr | or should i do that somewhere else? |
16:16.12 | elux | beats me. first time im here. but im sure your free to ask questions but it doesnt seem like anyone is around |
16:16.37 | LtScarr | oh heh :) |
16:16.47 | LtScarr | ah well |
16:16.51 | LtScarr | i can try then |
16:17.19 | markit | what is a "representative"? |
16:18.15 | LtScarr | i can seem to be able to "dial" from an analog phone |
16:18.20 | nvrpunk | devs are in here |
16:18.21 | nvrpunk | :P |
16:18.22 | nvrpunk | so yeah |
16:18.29 | LtScarr | i can make calls to it |
16:18.32 | LtScarr | but not from it |
16:18.36 | nvrpunk | there are people who know an awful lot about asterisk |
16:18.38 | LtScarr | i do get a dial tone |
16:18.46 | nvrpunk | asterisk -r |
16:18.49 | nvrpunk | core set verbose 9 |
16:18.52 | nvrpunk | check the output |
16:18.55 | LtScarr | but when i dial a number |
16:18.57 | nvrpunk | see where it's hanging |
16:19.04 | LtScarr | it just keeps giving a dial tone |
16:19.22 | nvrpunk | sounds like your contexts aren't right |
16:19.32 | nvrpunk | and that it's not grabbing the number at all |
16:20.01 | LtScarr | well what i did |
16:20.09 | LtScarr | i backupped the default config |
16:20.14 | nvrpunk | pastebin your configs |
16:20.17 | nvrpunk | and post the url here |
16:20.19 | LtScarr | and then stripped everything |
16:20.20 | nvrpunk | ill look at em |
16:20.30 | LtScarr | maybe a bit too much :) |
16:20.40 | LtScarr | okay hold on a sec |
16:21.18 | nvrpunk | oddly, I can't get a ring to passback between openser and asterisk |
16:22.05 | elux | what kind of provider do i need to make/accept multiple calls with asterisk? |
16:22.20 | elux | i guess there are many channel protocols supported, im looking at SIP or IAX2 |
16:22.33 | nvrpunk | pstn gateway |
16:22.35 | nvrpunk | is the norm |
16:22.48 | LtScarr | http://pastebin.com/d62563ece |
16:22.56 | LtScarr | those are my extensions |
16:23.09 | elux | so i hook up * to a pstn, and then enable access to my services (ie. adhearsion) to be able to connect via SIP? |
16:23.27 | LtScarr | http://pastebin.com/d498c3f5 |
16:23.34 | LtScarr | and that's my SIP config |
16:23.41 | elux | the PSTN will take my asterisk exchange and link it to the global telephony system? |
16:23.51 | elux | so i can make calls through it etc.? |
16:24.34 | LtScarr | ow and i configured my router to be the ATA |
16:24.52 | LtScarr | and i registers with asterisk |
16:28.26 | elux | what is the difference between having my clients connect directly to a SIP using some provider or making my own asterisk box that connects to a PSTN? .... i guess its just cost? |
16:30.16 | markit | trying to translate asterisk sound files: is there some subtle differences between agents and representatives? |
16:30.33 | nvrpunk | sorry back |
16:30.37 | nvrpunk | looking at your configs |
16:31.07 | LtScarr | thanks |
16:31.10 | elux | im trying to wrap my head around how voip/asterisk works |
16:31.16 | elux | nvrpunk: have what i been saying makes sense? |
16:31.51 | nvrpunk | one min |
16:31.53 | nvrpunk | lemme read |
16:31.57 | nvrpunk | doing multiple things at once |
16:32.15 | elux | thx |
16:32.17 | nvrpunk | elux, the differences vary |
16:32.24 | nvrpunk | a pstn allows you to have a DID |
16:32.33 | nvrpunk | you can do what a pstn does with additional cards |
16:32.42 | nvrpunk | and a trunk into your home |
16:32.54 | elux | what is a DID? |
16:33.01 | nvrpunk | Direct Inward Dial |
16:33.07 | nvrpunk | check out voip-info.org |
16:33.13 | nvrpunk | it has a lot of useful info |
16:33.17 | nvrpunk | that you should read through |
16:33.26 | elux | i simply want to use make/accept multiple calls using voip through a SIP/IAX2 client |
16:33.49 | dwery | hello. with a sip channel with call-limit = 1, I need to be able to st the cause code that goes back to chan_lcr to busy. is there a way to do it? |
16:33.54 | nvrpunk | you would think this helicopter above my building is looking for my rogue radio station |
16:33.55 | nvrpunk | hmm |
16:34.19 | LtScarr | :) |
16:34.24 | nvrpunk | LtScarr, where is your outbound calling context? |
16:34.28 | nvrpunk | i dont see one at all |
16:34.40 | nvrpunk | or are you trying to make calls from phone to phone |
16:34.40 | LtScarr | there isn't any :) |
16:34.45 | LtScarr | just internal calls |
16:35.29 | LtScarr | i'm trying this as an experiment for future implementations |
16:35.40 | nvrpunk | ok so you have extension 1 2 and 3 |
16:35.44 | LtScarr | yes |
16:36.33 | *** join/#asterisk hi365_m (n=hi365@85.130.230.240) |
16:36.37 | LtScarr | two of them are softphones |
16:36.40 | nvrpunk | LtScarr, lemme point you to http://www.asteriskguru.com/tutorials/ |
16:36.41 | nvrpunk | :) |
16:36.47 | nvrpunk | great place to start |
16:36.55 | nvrpunk | also, as I said asterisk -r |
16:37.00 | nvrpunk | core set verbose 9 |
16:37.16 | nvrpunk | watch the asterisk server handle the call :) |
16:37.24 | nvrpunk | it's better than just giving an answer hehe |
16:37.36 | LtScarr | ok :) |
16:37.52 | nvrpunk | i dont see your sip.conf either |
16:37.57 | nvrpunk | which i would need to see |
16:38.12 | LtScarr | well i did paste 2 urls |
16:38.16 | nvrpunk | oh |
16:38.17 | nvrpunk | sorry |
16:38.21 | *** part/#asterisk markit (n=marco@88-149-177-66.static.ngi.it) |
16:38.48 | *** join/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
16:39.06 | *** part/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
16:39.33 | LtScarr | one note though about the tutorials online |
16:39.33 | elux | so i can only get a phone number and accept inward calls from the global telephony network if i have DID? |
16:40.05 | LtScarr | they don't start with zero configuration |
16:40.42 | LtScarr | so it's hard to see what is needed and what isn't |
16:44.01 | LtScarr | okay i think this problem isn't asterisk related |
16:44.17 | LtScarr | i think my ATA isn't functioning the way it should |
16:44.41 | LtScarr | so thanks for the time :) |
16:46.57 | *** join/#asterisk jtodd (n=jtodd@jetblue.colorbroadband.com) |
16:46.57 | *** mode/#asterisk [+o jtodd] by ChanServ |
16:49.28 | *** join/#asterisk yondaime (n=Yamato@unaffiliated/yondaime) |
17:01.48 | *** join/#asterisk joako (n=joako@99-153-162-33.lightspeed.miamfl.sbcglobal.net) |
17:05.51 | nvrpunk | haha, its fun to call from a cell phone in iraq to our pstn in the states which then routes to our asterisk box in sweden and then to our asterisk box in iraq |
17:06.01 | nvrpunk | walkie talkie ftw |
17:06.04 | dwery | is there a way to force a sip peer to use plaintet authentication? |
17:07.03 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
17:07.17 | *** join/#asterisk Khratos (n=Khratos@190.166.111.76) |
17:07.18 | nvrpunk | hmm, http://books.google.com/books?id=9_wRFy5OGw4C&pg=PA140&lpg=PA140&dq=plaintext+authentication+sip+asterisk&source=web&ots=7hPikKrL-d&sig=ulh9sDMViL7sq3AB3066Fpen3_A&hl=en&ei=IBGPSdSVI8H7tgecy7CrCw&sa=X&oi=book_result&resnum=4&ct=result |
17:07.45 | nvrpunk | that will answer your question, somewhere in it :P |
17:08.25 | dwery | nice, that mean shttp://www.voip-info.org/wiki/view/Asterisk+phone+grandstream+budgetone is completely wrong |
17:08.53 | *** join/#asterisk bminish (n=bminish@2001:770:180:0:219:d1ff:fe80:ea64) |
17:09.19 | dwery | I have the problem where my sip phone generates a busy tone after the first digit when in early dial mode. |
17:09.35 | dwery | it will however keep transmitting the digits and the call will go thru |
17:09.40 | dwery | but the bsy tone is annoying |
17:09.47 | [TK]D-Fender | kill off early dial |
17:10.08 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
17:10.21 | dwery | I can't, I need asterisk to provide a dial tone when I press the key for the external isdn line |
17:10.26 | dwery | ad without early dial |
17:10.33 | nvrpunk | I have a problem with the "r" option not creating a ring and don't know why |
17:10.34 | nvrpunk | :( |
17:10.35 | dwery | I would have a 4 second timeout |
17:11.04 | nvrpunk | on the bright side people are still paying for calls |
17:11.07 | nvrpunk | which is a good thing |
17:11.57 | *** join/#asterisk estr4ng3d (n=sm4rt@196.219.96.141) |
17:13.30 | [TK]D-Fender | dwery: what key for external line? Show us the dialplan & CLI output for your call attempt |
17:14.25 | nvrpunk | anyone care to tell me if there's anything wrong with these two contexts? http://www.pastebin.ca/1330837 |
17:14.41 | nvrpunk | it's not generating a ring to the caller |
17:16.13 | giovani | nvrpunk: try specifying the ring time? |
17:16.21 | giovani | I've never tried using double-commas like that |
17:16.24 | [TK]D-Fender | nvrpunk: you need to ANSWER teh call first |
17:16.49 | nvrpunk | [TK]D-Fender, its an IVR |
17:16.59 | nvrpunk | they dial in the extension |
17:17.03 | nvrpunk | it rings the extension |
17:17.05 | nvrpunk | hmm |
17:17.08 | nvrpunk | ah |
17:17.09 | [TK]D-Fender | nvrpunk: [makecall] <- isn't |
17:17.12 | nvrpunk | Answer before it dials |
17:17.12 | nvrpunk | duh |
17:17.51 | *** join/#asterisk Bonix (n=Bonix@212-lo1.rt2.isimples.com.br) |
17:17.52 | dwery | [TK]D-Fender: the actual dialpan has only one line: exten => _2XX,1,Dial(SIP/${EXTEN}) |
17:18.21 | [TK]D-Fender | dwery: What does that have to do with ISDN? |
17:18.51 | dwery | that come later. I plan to have the 0 key to access the first free line. and I need to signal that w have one using a dial tone |
17:19.12 | *** join/#asterisk ikevin_ (n=kevin@ANancy-256-1-53-94.w90-26.abo.wanadoo.fr) |
17:19.18 | ikevin_ | hello |
17:19.23 | dwery | So I want the user to press 0 and have a dial tone immediately. and that works only with early dial |
17:19.55 | ikevin_ | anyone know how can i have sound quality in moh? |
17:20.00 | ikevin_ | i use mpg12 |
17:20.04 | ikevin_ | mpg123* |
17:20.08 | [TK]D-Fender | dwery: fine record a continuous dialtone and background it in an IVR |
17:20.20 | [TK]D-Fender | ikevin_: HUH? |
17:20.41 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
17:20.41 | Maliuta | ikevin_: you are aware that it doesn't transmit all the frequencies |
17:20.55 | ikevin_ | while i try to setup music on hold the sound is bad, cutting, ... |
17:21.01 | ikevin_ | yes |
17:21.33 | dwery | [TK]D-Fender: there must be an easier way |
17:21.37 | ikevin_ | so i think that can be played in the good speed and without cutting? |
17:21.44 | [TK]D-Fender | ikevin_: Stop using mpg123 and use Native MoH |
17:22.08 | [TK]D-Fender | ikevin_: For MP3's you also have to make sure they are non VBR and have no ID3 tags |
17:22.12 | ikevin_ | i have tryed to play an mp3 width 'mp3' or 'quietmp3' that are not played |
17:22.35 | nvrpunk | [TK]D-Fender, even with exten => _XXXX.,1,Answer and then the Dial, it's still not ringing back |
17:22.37 | ikevin_ | how can i check vbr? |
17:22.38 | [TK]D-Fender | ikevin_: See above |
17:22.46 | [TK]D-Fender | ikevin_: Get a good audio program |
17:23.05 | ikevin_ | k |
17:23.19 | ikevin_ | does the bitrate limit is 8kb/s ? |
17:23.40 | [TK]D-Fender | dwery: If there is it has to be offered by the phone... * can't slowly pass early dial info on for you to your ISDN. |
17:24.00 | [TK]D-Fender | dwery: * will pass the first thing it can directly on and you're DOA from there |
17:24.06 | [TK]D-Fender | ikevin_: No. |
17:24.22 | [TK]D-Fender | ikevin_: standard bitrates are fine. Use mode=files |
17:24.31 | dwery | [TK]D-Fender: even without ISDN, I want early dial to work for SIP without the bsy tone on the first digit |
17:24.37 | ikevin_ | ok |
17:24.48 | dwery | ill try playtones |
17:24.57 | [TK]D-Fender | dwery: Show us what is happening. |
17:27.00 | dwery | [TK]D-Fender: sure. what you need me to show? sip set debug ? |
17:27.23 | [TK]D-Fender | dwery: A complete call attempt and what it's doing |
17:27.50 | *** join/#asterisk hi365_m (n=hi365@85.130.230.240) |
17:28.03 | dwery | [TK]D-Fender: ok, logging now |
17:28.22 | ikevin_ | what kind of software are you using to encode mp3 for moh? |
17:28.38 | ikevin_ | all i test are putting tags on the file |
17:30.24 | dwery | [TK]D-Fender: http://privatepaste.com/820oxi3sgf I elieve the problem lies on line 130. * trasmits 404 instead of 484 |
17:30.45 | dwery | I have probably configured somethig wring |
17:30.49 | [TK]D-Fender | ikevin_: If you encode it yourself don't use MP3 |
17:31.43 | [TK]D-Fender | dwery: Looking like * itself dos not support early dial |
17:31.50 | ikevin_ | what format is bester? |
17:31.56 | ikevin_ | can i use ogg? ^^' |
17:32.15 | [TK]D-Fender | ikevin_: use the codec your CHANNEL will use |
17:32.40 | *** join/#asterisk Talkradio (i=talkradi@linuxgeneration.ca) |
17:33.48 | dwery | [TK]D-Fender: that's strange, I'm pretty sure I read it was supported |
17:34.03 | [TK]D-Fender | dlewis: People always are |
17:34.12 | dwery | ouch! |
17:34.17 | dwery | allowoverap=yes |
17:34.54 | dwery | [TK]D-Fender: now it's prefect. thaks for helping out |
17:35.12 | [TK]D-Fender | dwery: glad you found it |
17:36.02 | ikevin_ | [TK]D-Fender, i tryed to make a wave and using mode=files, sound is already cuted, slow, ... |
17:37.03 | *** join/#asterisk Linuturk (n=linuturk@fluxbuntu/developer/Linuturk) |
17:38.48 | [TK]D-Fender | ikevin_: have you fixed your FILES? |
17:38.59 | [TK]D-Fender | ikevin_: what are you testing with? Describe the call |
17:40.13 | ikevin_ | i use ekiga as sip client, i use codec gsm for the call, i have make a wave file (bitrate 8k/s) and using asterisk native format |
17:40.48 | [TK]D-Fender | ikevin_: Try Ekiga with ULAW or ALAW instead and test |
17:41.14 | ikevin_ | ok |
17:43.15 | ikevin_ | ekiga still don't have this one |
17:44.04 | sipy | ifconfig |
17:44.13 | sipy | aaaarrrrggghhh |
17:46.18 | dwery | [TK]D-Fender: now I have to tell * to buffer the digits when using the ISDN line, bcause it does not support early dial |
17:47.57 | [TK]D-Fender | ikevin_: yes it does |
17:48.06 | [TK]D-Fender | ikevin_: g711u, g711a |
17:48.28 | [TK]D-Fender | dwery: that is the part I figured you would get screwed by |
17:48.37 | ikevin_ | i only have g721 |
17:49.01 | [TK]D-Fender | ikevin_: there is no way you only have GSM & G721 <- what is that?) |
17:49.10 | [TK]D-Fender | ikevin_: look harder |
17:49.43 | ikevin_ | gsm seems the bester availlable, i have put a lowest priority on, and he's used |
17:49.43 | [TK]D-Fender | ikevin_: they are listed as PCMA and PCMU in Ekiga |
17:50.00 | ikevin_ | ok, found |
17:50.12 | ikevin_ | so, there are the first activated in the list |
17:50.26 | ikevin_ | do i need to declarer anything on asterisk to use them? |
17:50.38 | [TK]D-Fender | ikevin_: your allow/disallow by peer |
17:51.09 | ikevin_ | i don't have any entry allow or disallow |
17:51.27 | [TK]D-Fender | ikevin_: Then go set them |
17:51.38 | ikevin_ | ok |
17:51.48 | [TK]D-Fender | ikevin_: "disallow=all" followed by "allow=ulaw" (for PCMU) |
17:53.43 | ikevin_ | ok i try |
17:54.48 | *** join/#asterisk Pazzo (n=ugelt@reserved-225136.rol.raiffeisen.net) |
17:54.54 | ikevin_ | waou, there are a big difference :D |
17:55.05 | [TK]D-Fender | ikevin_: Improvement? |
17:56.47 | ikevin_ | yep |
17:56.52 | ikevin_ | thx very much :) |
17:57.19 | [TK]D-Fender | ikevin_: this is likely your problem : |
17:57.21 | [TK]D-Fender | ~gsmbug |
17:57.43 | jbot | [~gsmbug] there is a bug compiling Asterisk with GCC 4.2 where optimization errors cause GSM transcoding to distort heavily. Compile with GCC 4.1 or lower, or follow this patch : http://bugs.digium.com/view.php?id=11243 Also please read : http://forums.digium.com/viewtopic.php?p=116731&sid=3c65e49ec840380ed20f1c8426382b39 |
17:57.46 | [TK]D-Fender | ^^^^^^^^ |
17:57.50 | *** join/#asterisk estr4ng3d (n=sm4rt@196.219.96.141) |
17:58.03 | ikevin_ | yep, 13k/s is maybe too short for what i need :) |
17:58.13 | [TK]D-Fender | ikevin_: No.... its a transcoding error |
17:58.28 | [TK]D-Fender | ikevin_: if ULAW came out OK, this is almost guaranteed to be why |
18:02.41 | *** join/#asterisk cesau (n=cesau@207.58.225.93) |
18:03.31 | *** join/#asterisk markit (n=marco@88-149-177-66.static.ngi.it) |
18:03.56 | markit | hi, conf-extended... extended means "more duration" or "more people allowed in"? |
18:08.42 | [TK]D-Fender | markit: A sound file alone means nothing |
18:10.37 | markit | [TK]D-Fender: there are some commands to "extend" the conference. I have to translate sounds related, but since I don't use conference, don't know the meaning of "extend" a conference |
18:11.03 | [TK]D-Fender | markit: usually it would imply TIME, not # of participants. |
18:11.17 | markit | [TK]D-Fender: thanks a lot :) |
18:11.40 | markit | [TK]D-Fender: also can't understand the difference, if any, between "agent" and "representative" |
18:12.03 | *** join/#asterisk ManxPower (n=manxpowe@11.sub-70-214-34.myvzw.com) |
18:12.10 | markit | do you have some hint about it? |
18:12.18 | markit | hi ManxPower |
18:12.18 | [TK]D-Fender | markit : Their meaning is usually the same. |
18:12.54 | [TK]D-Fender | markit: "representative" is more gentle & welcoming. Agent sounds a little harsher |
18:13.16 | markit | ok, fine. Last question: there are some minor errors (very trivial) in core-sounds-en.txt, do I have to use the bugtracker or try the -bugs channel? |
18:13.43 | ManxPower | [TK]D-Fender: I prefer "Your worst customer service nightmare" |
18:13.45 | ikevin_ | does sound at 32kb/s@16khz is better than a sound at 64kb/s@8khz ? |
18:14.11 | ManxPower | ikevin_: very few devices support wideband (16Khz) |
18:14.31 | ikevin_ | ok |
18:15.41 | ManxPower | Polycom calls wide band "HD voice" |
18:16.04 | ManxPower | that's the only company that supports it as far as I know, and only a few polycom models even support it |
18:16.19 | Corydon76-dig | but in answer to your question, yes, 16kHz almost universally sounds better than 8kHz |
18:16.43 | [TK]D-Fender | Half the bitrate and better quality.... odd |
18:17.11 | ManxPower | Corydon76-dig: Even at 32K? I would think since you have more data to compress the same bit rate would sound worse at 32K compared at 8Khz? |
18:17.22 | Corydon76-dig | [TK]D-Fender: what matters is not the number of samples, but the depth of each sound |
18:18.27 | ManxPower | quickly parents a wide band extension to the LPC coded! |
18:18.30 | Corydon76-dig | Above a certain number of samples per second, you won't be able to tell a difference anyway |
18:18.41 | ManxPower | and paTents it too! |
18:19.40 | [TK]D-Fender | Corydon76-dig: Yeah.. there looks to be a # missing. sample frequency, audio range, bitrat... those seem to include only 2 |
18:19.42 | Corydon76-dig | The only advantage LPC10 ever had was the size of the resulting bitstream |
18:21.31 | Corydon76-dig | It's like the gamers who try to get 110fps... anything above about 30fps is wasted CPU |
18:22.05 | Corydon76-dig | In this case, we're talking about the perceptual limitation of the ears, not the eyes, but the same principle applies |
18:22.58 | ScribbleJ | Good morning friends. I promised to post a version of my Generic Speech Engine API plugin yesterday, even though I am ashamed of the code at this point. And I did - you can get it at http:/scribblej.com/svn/ and there is a phone number on that page too where you can try it. |
18:26.11 | [TK]D-Fender | Corydon76-dig: 60 fps :) the 30 deal well... thats a whole story |
18:26.44 | [TK]D-Fender | Corydon76-dig: nifty trick how movies get away with 24 because of the concept of persistence .\ |
18:30.38 | *** join/#asterisk hi365_m (n=hi365@85.130.230.240) |
18:30.53 | ScribbleJ | 24 fps is how Hollywood makes it's money |
18:30.57 | *** join/#asterisk taso (n=hfaohh@c-67-180-35-251.hsd1.ca.comcast.net) |
18:31.01 | ScribbleJ | You thought they were only gouging you on concession stand prices! |
18:31.12 | ScribbleJ | But no, not only that, but they /keep/ 6 frames out of every second that YOU PAID FOR! |
18:32.10 | taso | hey guys, quick question. Suppose I have a cell phone w/ an active #... Let's say the # is 228-492-2249 ... can I have that # forwarded to a Asterisk server that acts as an answering machine? |
18:33.15 | ScribbleJ | Yes, but if your next question is 'how' the answer is 'ask your cell phone provider' - they have to arrange the forwarding. |
18:33.53 | taso | okay, so, that's possible great. Next question. Let's say that I forward my calls to an Astierk server, then, if it's a number I want , it calls me, is that possible or no? |
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18:34.36 | ScribbleJ | I'd assume so, althugh it might be impossible for your provider to forward some calls and not others (i.e. when you call yourself from asterisk, it has to /not/ get forwarded, right?() |
18:34.39 | taso | I would think no since it would has to call you back.... and when it went to call you back...it would be calling itself |
18:34.43 | ScribbleJ | Right |
18:34.46 | ScribbleJ | I think no also. |
18:34.58 | ScribbleJ | Here's how I'd solve it. |
18:35.03 | taso | 2 #'s ? |
18:35.05 | taso | same phone? |
18:35.07 | ScribbleJ | GEt a SIP provider, hand out /that/ phone number to my frineds. |
18:35.12 | taso | ah I see |
18:35.20 | ScribbleJ | Then when my friend calls they are calling directly to ASterisk |
18:35.24 | taso | yeap |
18:35.35 | ScribbleJ | And from there I canf orward to the cell # they do not know, if they meet my quality standards. |
18:35.47 | taso | :) |
18:35.49 | ScribbleJ | They don't, by the way - |
18:36.01 | taso | :P |
18:36.04 | ScribbleJ | I would not be friends with anyone who would have a loser like me for a friend. |
18:36.19 | taso | unfortunate |
18:38.27 | *** part/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
18:41.37 | taso | so here's another question, can you move your cell phone # to a land line? or to any line for that matter? |
18:42.12 | taso | I know you can move a home phone # to a cell phone |
18:42.17 | taso | but can you do the inverse |
18:43.23 | ScribbleJ | You can in the US |
18:43.37 | ScribbleJ | YOu can even move it to SIP provider if you do not want a real phone line |
18:44.22 | taso | wild |
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18:48.29 | ManxPower | Porting a number to a SIP provider usually takes 10 - 14 days. |
18:48.41 | taso | yea, I just got off the phone with my provider |
18:48.46 | taso | they said it was possible |
18:48.57 | taso | is there anyway to automate this service? |
18:49.14 | ManxPower | define "automate" |
18:49.42 | ManxPower | It takes as long as it takes. Nothing you can do about it. |
18:49.52 | taso | i.e. , let's say you were a telco, and you had people signing up for your SIP service, and it said "Please tell use the # you would like to use" , and you enter in your cell phone # and it automatically starts the switch over |
18:50.01 | taso | or send an email to your provider |
18:50.06 | taso | etc |
18:50.18 | ManxPower | taso: not a chance. Telcos need LOTS of paperwork to prove you authorized the change |
18:50.25 | taso | ah |
18:50.49 | ManxPower | AND your old telco has to accept the loss of the customer. |
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18:51.49 | ScribbleJ | Yeah, if they want to, your old telco can bitch about it and require papers submitted. |
18:51.57 | ScribbleJ | Although it turns out that sometimes they do not. |
18:52.11 | taso | well, it actually wouldn't be a loss, you would get a new # assigned to your cell phone, and your old cell phoen # would go to the SIP provider |
18:52.18 | ScribbleJ | I transferred a number to AT&T from Sprint; all AT&T needed was my word, Sprint just trusted them. |
18:52.53 | ManxPower | It generally works like this. Customer requests a number port from their new telco. Their new telco gets all the required information for your old account, then contacts the new provider for a number port, the old provider says something like "REJECTED! You did not dot that i", the process repeats once or twice and then your number is (usually) ported. |
18:53.14 | ManxPower | ScribbleJ: cellphone to cellphone ports are different |
18:53.34 | ManxPower | I was referring to porting from a land line. |
18:53.48 | ManxPower | ScribbleJ: how long did the port take? a few days? |
18:54.13 | ScribbleJ | Are they different in some real way, or are they just different int hat the cell phone companies tend to know eeach other and go easy on the requirements? |
18:54.13 | ScribbleJ | ManxPower, as I recall it took about 4 hours, actually. |
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18:54.17 | ManxPower | ScribbleJ: I don't think there is an anti-slamming law for cell companies. |
18:54.31 | ScribbleJ | Oh, I see. |
18:54.58 | ManxPower | With land lines if you move your number and then claim you did not authorize it -- if the telco can't prove you authorized it there can be big fines. |
18:55.20 | ManxPower | for slamming long distance service it's $10,000 per incident if I recall correctly. |
18:55.22 | taso | it's a law that you have to state "This phone call with be recorded, correct ? " |
18:55.47 | ManxPower | taso: call recording within a state is goverened by the laws of that state. |
18:55.48 | taso | if you're going to record a call that is |
18:55.54 | taso | interesting |
18:56.12 | ManxPower | some states only require ONE of the parties to consent to recording. |
18:57.04 | taso | hrm |
18:57.37 | taso | meaning that if there is Person A and Person B... and Person A calls person B, Person A can be okay with it, without person B knowing |
18:57.40 | taso | ? |
18:59.58 | taso | hrm reading up on it now, that's pretty wild |
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19:11.39 | giovani | taso: wild? |
19:11.49 | giovani | it just allows you to record your own calls ... reasonable to me |
19:12.25 | ScribbleJ | In my opinion, it's easy to just always hav your system say "this call will be recorded" - people don't even pay attention to this anymore. |
19:13.01 | giovani | I think most companies go with "may be recorded" for legal reasons, so that they're not obligated to provide it, should someone, say, a lawyer, request the recording |
19:13.04 | [TK]D-Fender | ScribbleJ: "may" be recorded :) Less forceful and equally legitimizing :) |
19:13.44 | giovani | maybe obligated was the wrong word -- so that they can claim they never recorded it :) |
19:14.08 | taso | but if it was a personal phone call people would be thinking wtf? |
19:14.29 | taso | you call your friend Dave, and it says "This call will be recorded" |
19:14.31 | taso | that's different |
19:14.32 | giovani | if it's a personal phone call ... you won't be facing any litagation unless you're publishing the recordings |
19:14.45 | giovani | so, it's really a non-issue other than a hypothetical, and pedantic legal one |
19:14.52 | taso | heh |
19:15.42 | giovani | if, however, you call up some company, like say, customer support, and plan on recording it for some reason, say, to embaress the company on your blog ... I'd announce it to ensure no legal trouble :) |
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19:25.44 | jaytee | usually the standard phrase in the states is "This call may be recorded for "quality assurance" purposes." |
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19:33.09 | [TK]D-Fender | jaytee: AND "training" |
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19:48.26 | pseubodot | hi there, I'm trying to get asterisk running on debian with a Wildcard TDM400P REV E/F Board 1, running thru the examples in the oreilly book. seems no matter what I do, I always get "4 channels to configure." even if zttool reports the card as 'OK' |
19:49.05 | ScribbleJ | I don't suppose any helpful person wants to tell me if I set things up properly for incoming sip calls, if you tried something like sip:2000@home.scribblej.com you /might/ get my Speech Engine PLugin |
19:50.13 | ScribbleJ | Or you might get nothing if I screwed up. Heh. |
19:54.57 | [TK]D-Fender | pseubodot: that message is fine. |
19:55.27 | [TK]D-Fender | pseubodot: it is not negative. Its saying that Zaptel is initialize and ready for * to allocate channels agains |
19:55.47 | [TK]D-Fender | +t |
19:55.56 | pseubodot | [TK]D-Fender: thanks |
19:56.25 | pseubodot | i get to the point where my dial plan looks like what it says in the book, and 'dialplan show' seems to come out right |
19:56.38 | pseubodot | but I'm not getting dialtone on my phone (plain phone) |
20:01.11 | *** part/#asterisk bgmarete (n=marebri_@196.201.217.80) |
20:01.39 | [TK]D-Fender | pseubodot: What modules do you have on your card? Pastbin your configs. |
20:01.41 | [TK]D-Fender | ~pb |
20:01.52 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
20:01.55 | pseubodot | thanks |
20:02.31 | pseubodot | http://paste.debian.net/27970 |
20:04.00 | pseubodot | 2FXOs, 2FSOs |
20:04.12 | [TK]D-Fender | pseubodot: I see. Now for your configs... |
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20:04.32 | pseubodot | [TK]D-Fender: which specifically? |
20:04.59 | [TK]D-Fender | pseubodot: zaptel/zapata, dialplan, etc |
20:06.03 | pseubodot | [TK]D-Fender: http://paste.debian.net/27971/ <- all here |
20:06.48 | [TK]D-Fender | pseubodot: Your zaptel doesn't match your ztcfg |
20:06.54 | [TK]D-Fender | pseubodot: You commented the FXO's out |
20:07.15 | [TK]D-Fender | pseubodot: Sorr, 1 each |
20:07.19 | pseubodot | standby |
20:08.13 | [TK]D-Fender | pseubodot: And you are defining your channels twice because of that include |
20:08.14 | pseubodot | [TK]D-Fender: http://paste.debian.net/27972/ (fixed zaptel.conf file, did ztcfg -vv) |
20:08.26 | pseubodot | [TK]D-Fender: I should remove the include? |
20:08.44 | [TK]D-Fender | pseubodot: include is fine but you can see it doing them twice |
20:08.53 | [TK]D-Fender | pseubodot: fix tha |
20:09.21 | [TK]D-Fender | pseubodot: Also not that you do not have anything you can dial in your [default] context. Not good... |
20:09.31 | [TK]D-Fender | pseubodot: Also ensure that you have the molex connected to the card |
20:10.32 | *** join/#asterisk beek (n=klinebl@pdpc/supporter/professional/beek) |
20:10.40 | pseubodot | Not sure I follow - I'm not sure that I can see it being added twice |
20:11.08 | pseubodot | the molex is on the correct port (afaik), cord from wall goes in 3rd receiver from the top of the card |
20:11.34 | pseubodot | cord from the telephone on my desk is going into the 1st receiver from the top of the card. |
20:16.59 | pseubodot | this is the third time I'm trying to set this up |
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20:19.35 | [TK]D-Fender | pseubodot: No, the molex POWER connector from your power supply to the PCI card |
20:19.45 | pseubodot | [TK]D-Fender: ah yes, I do have power connected |
20:19.54 | pseubodot | to the card |
20:20.10 | [TK]D-Fender | pseubodot: Only connect an analog phone and try each port. IIRC they are not sequential on the back of the card |
20:20.32 | pseubodot | [TK]D-Fender: what should I be listening for? |
20:20.52 | [TK]D-Fender | pseubodot: dial-tone, and watch * CLI for "starting simple switch |
20:21.52 | pseubodot | found one port where I seem to be getting the echo |
20:22.53 | pseubodot | and when I lift the receiver I see in zttool the Total/Conf/Act line change from 4/4/0 to 4/4/1 |
20:23.08 | pseubodot | woo! progress! |
20:23.22 | [TK]D-Fender | pseubodot: Get out of zttool and startup * |
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20:23.42 | pseubodot | [TK]D-Fender: debian*CLI> |
20:24.02 | [TK]D-Fender | pseubodot: "zap show channels" |
20:24.14 | masus | hi all, can anybody say me the possibility to get the dialstatus of a call maded by /var/spool/ast.../outgoing directory .. |
20:24.29 | pseubodot | [TK]D-Fender: "No such command 'zap show channels'" |
20:24.53 | [TK]D-Fender | pseubodot: pastebin "show modules" |
20:25.05 | pseubodot | [TK]D-Fender: will do. standby |
20:25.17 | masus | ;( |
20:26.09 | pseubodot | [TK]D-Fender: http://paste.debian.net/27973/ |
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20:26.58 | [TK]D-Fender | masus: for the "Channel:" you aren't going to be able to for sip/zap/iax, etc. You need to be dialing a LOCAL channel and you can process the dial in there. |
20:27.24 | [TK]D-Fender | pseubodot: "module reload chan_zap.so" |
20:27.34 | [TK]D-Fender | pseubodot: the do "zap show channels" again |
20:28.31 | pseubodot | [TK]D-Fender: http://paste.debian.net/27975/ |
20:28.53 | masus | [TK]D-Fender: Thank you. I'll try . |
20:29.16 | [TK]D-Fender | pseubodot: Looks like your modules are configured backwards |
20:29.31 | pseubodot | [TK]D-Fender: I would change these in /etc/zaptel.conf? |
20:30.33 | [TK]D-Fender | pseubodot: Seriously double check the module order on the card physically and see how they really map to the port #'s. IIRC its simething like 4,1,2,3 or something like that... then make sure zapata & zaptel both match |
20:31.19 | pseubodot | [TK]D-Fender: two green modules on left (towards card bracket), two red modules on right (away from bracket) |
20:31.47 | [TK]D-Fender | pseubodot: Read your card docs for the functional order, not just left-right |
20:32.27 | cesau | ok - i have a box with no dialers, just astersik -- im trying to make an outbound call via my itsp and play a file -- my itsp tells me i dont need to register with them -- when i originate a call, would i use Local/somerandomeext@context extension SIP/provider@context? if so, which context is being performed, the local or the providers? |
20:34.00 | pseubodot | [TK]D-Fender: okay, thanks |
20:34.09 | pseubodot | I have no docs, so I'll have to figure that out. |
20:34.35 | [TK]D-Fender | cesau: SIP/provider@context is not an EXTENSION. Your DIALPLAN holds your extensions. |
20:34.41 | [TK]D-Fender | pseubodot: www.digium.com |
20:34.54 | [TK]D-Fender | pseubodot: should have the install guides, etc |
20:36.46 | cesau | ok, if i have a simple extensions (like _X) setup, would it then be: originate sip/provider@context extension numbertodial <-- with the dial command in the first context? (this seems like it should be really easy and im overcomplicating it) |
20:37.34 | [TK]D-Fender | cesau: "go read the instructions for originate |
20:37.38 | masus | [TK]D-Fender: i have do it with local channel , but there is something wrong its calling the number twice , here are some debug and config files -> http://rafb.net/p/jMyto215.html if u can take a look please . :) |
20:38.26 | cesau | the instructions seem to imply that i have a hard phone to originate from... =/ |
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20:38.52 | [TK]D-Fender | masus: Of course it is... YOU are telling it to call that local channel and dump it into the SAME EXTENSION |
20:39.23 | pseubodot | [TK]D-Fender: found it, according to this my card should be as I expect. |
20:39.38 | pseubodot | [TK]D-Fender: should and is are two different things tho |
20:39.41 | masus | hmmm , i dont understand the logic :s |
20:40.25 | [TK]D-Fender | masus: http://pastebin.com/m7ac11036 |
20:41.27 | [TK]D-Fender | masus: And you didn't specify the context |
20:42.28 | masus | [TK]D-Fender: Thank you i'll try a little more |
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20:44.00 | *** join/#asterisk badcfe (i=cso@78.156.5.233) |
20:44.15 | badcfe | i have a question actually related more to sip than to * |
20:44.26 | badcfe | since UPDATE is a non rfc3261 it should be treated as OPTION and thus not fork. |
20:44.31 | badcfe | a proxy may have parallelly forked an INVITE and now receives an UPDATE on that early dialog. |
20:44.34 | badcfe | my question is: what should this statefull proxy then do? |
20:45.43 | pseubodot | hmm. |
20:46.01 | pseubodot | removes 1xFXO and 1xFXS to make figuring this out simpler. |
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21:00.33 | badcfe | no SIP experts around huh |
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21:04.02 | ManxPower | badcfe: try the SER/OpenSer/OpenSIPS channels. |
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21:06.25 | cesau | assuming "Contact: <sip:asterisk@172.18.55.20>" should really be a class C ip address, which setting in sip.conf would change that? |
21:06.47 | [TK]D-Fender | ~sipnat |
21:06.54 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
21:06.55 | [TK]D-Fender | cesau: READ ^^^ |
21:07.16 | badcfe | ManxPower: thanks. never heard of opensips by the way.. |
21:07.16 | cesau | D-Fender, thank you! =D |
21:07.17 | ManxPower | 1) there are really no "classes" anymore, and anything like that would need you to know the netmask |
21:07.33 | ManxPower | badcfe: It is a fork of OpenSER |
21:07.45 | ScribbleJ | Does anyone have a second to try calling sip:2000@home.scribblej.com and let me know if they get an answer? Or is there some clever way I can test my SIP externally myself? |
21:07.50 | [TK]D-Fender | ManxPower: its a whole friggen cutlery set by now... |
21:08.16 | ScribbleJ | Maybe if I route my own call through my itsp that'd work |
21:08.23 | [TK]D-Fender | ScribbleJ: Dial(SIP/2000@home.scribblej.com) |
21:11.24 | ScribbleJ | [TK] my only * box is the one serving that, and I also need to know if I got my nat and routing correct to let in calls from the universe. |
21:11.43 | ScribbleJ | So I have to originate the call from someplace other than my own LAN, and * server. |
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21:12.04 | *** join/#asterisk HermesNeto (n=HermesNe@189.71.50.127) |
21:12.07 | HermesNeto | Hi |
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21:24.16 | ghenry | any progress on the best way to run asterisk on an xen guest without a patch i.e. ztdummy or dahdi_dummy |
21:26.15 | hardwire | install. use asterisk 1.6 |
21:26.21 | hardwire | HEAD |
21:26.33 | hardwire | rejoice |
21:28.29 | ghenry | is anyoen using 1.6 yet? I think it's way too early in the cycle |
21:29.16 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
21:29.36 | hardwire | ghenry: you're right.. sorry I mentioned it.. continue having issues please. |
21:30.14 | ghenry | will do hardwire. Thanks ;-) |
21:30.33 | hardwire | Not accepting my word as the golden law of the land has made you an enemy of Ra.. |
21:30.47 | hardwire | Be prepared to deal with the consequences.. My Anubis army is already on it's way. |
21:30.47 | *** join/#asterisk russellb_ (n=russell@asterisk/digium-open-source-team-lead/russellb) |
21:30.47 | *** mode/#asterisk [+o russellb_] by ChanServ |
21:30.56 | [TK]D-Fender | ghenry: Careful or hardwire will go nova on your ass! |
21:31.12 | phix | :D [TK]D-Fender! |
21:31.24 | ghenry | I got a pretty thick ass ;- |
21:31.45 | ghenry | will try iy |
21:31.47 | ghenry | it |
21:31.57 | hardwire | good boy |
21:31.59 | ghenry | I havent tried since workign on the ldap stuff |
21:32.16 | ghenry | I've got a few tickets I need to attend to when I can |
21:32.23 | hardwire | what project? |
21:32.30 | ghenry | res_ldap |
21:32.52 | hardwire | what functions does that expose? |
21:33.27 | hardwire | I rarely use LDAP so I'm curious how it's integrating. |
21:33.34 | ghenry | all the noraml realtime stuff |
21:33.41 | ghenry | that can be put in a directory server |
21:33.48 | hardwire | weird |
21:34.02 | ghenry | for some things, not for sip users etc. |
21:34.09 | hardwire | so contexts are ldap groups? |
21:34.16 | hardwire | how are dialplan entries sorted? |
21:34.24 | ghenry | exactly |
21:34.30 | ghenry | I've got to update the docs |
21:34.35 | ghenry | take a peak |
21:34.58 | ghenry | http://bugs.digium.com/view.php?id=13660 |
21:35.00 | hardwire | is there a priority kv in your schema? |
21:35.22 | hardwire | ah.. so for endpoints only? |
21:35.51 | hardwire | russellb: .. |
21:35.57 | russellb | orly? |
21:36.03 | hardwire | you're spamming my hosstom |
21:36.05 | hardwire | hosstop |
21:36.12 | hardwire | I demand sacrifice. |
21:36.44 | hardwire | is an agent of the sun god today. |
21:36.46 | russellb | o.o |
21:36.52 | russellb | offers file |
21:36.58 | hardwire | woot |
21:37.21 | hardwire | I'd like to thank whoever left a half eaten bag of doritos on my desk |
21:37.32 | hardwire | much obliged. |
21:37.38 | hardwire | your offering is acceptable |
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21:38.29 | [TK]D-Fender | sacrifices... HARDWIRE :p |
21:38.37 | [TK]D-Fender | actually... I'm not sure that would count ;) |
21:38.41 | hardwire | oh i'd like to see you try! |
21:38.53 | hardwire | is a fast lil bugger |
21:40.07 | hardwire | I should have gone home long ago |
21:40.18 | hardwire | I just did /etc/init.d/networking stop on a remote machine |
21:40.18 | ghenry | hardwire: so with 1.6 dahdi_dummy is not needed |
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21:40.35 | [TK]D-Fender | grabs his athamé and chases after hardwire |
21:40.38 | hardwire | what's funny is there is a serial cable enroute to site.. but nobody is on site today to plug it into the backup admin modem. |
21:40.40 | hardwire | lul |
21:41.08 | hardwire | [TK]D-Fender: for shame.. making me google |
21:41.26 | hardwire | ooh shiny. |
21:41.46 | hardwire | ghenry: you know.. I don't actually know. |
21:42.03 | hardwire | I just keep hearing how it's being/been/possibly/wishlisted into the most current code. |
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21:42.29 | hardwire | but setting youself up to accept a 1.6 update sounds like a fancy idea |
21:42.52 | ghenry | will try for now then. |
21:43.13 | ghenry | if timing doesn't need dahdi then for meetme etc. then good to go I guess |
21:43.33 | ghenry | suppose so; http://www.jeremy-mcnamara.com/2008/06/17/a-new-timing-api-for-asterisk-silencing-digium-critics/ |
21:43.33 | hardwire | russellb: ? know anything about that? |
21:43.48 | hardwire | hears a groan in the distance. |
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21:44.20 | russellb | know anything abotu what |
21:44.26 | russellb | the timing API? |
21:44.26 | russellb | yes. |
21:44.40 | hardwire | Good.. ghenry that should answer that. |
21:44.43 | hardwire | somebody knows about it. |
21:44.56 | ghenry | I'm doing asterisk with xen with no pci cards etc. |
21:45.07 | russellb | the timing API is only in 1.6.1, not 1.6.0 |
21:45.08 | ghenry | just wondering if I need dahdi at all |
21:45.15 | russellb | ghenry: depends what you want to do. |
21:45.25 | russellb | if you want conferencing, yes, but that is the only reason. |
21:45.36 | ghenry | even in 1.6.1? |
21:45.39 | russellb | yes. |
21:45.46 | russellb | the timing API does not replace the need for DAHDI for conferencing |
21:45.49 | ghenry | for dahdi_dummy right |
21:45.53 | russellb | as DAHDI contains the entire conference mixing engine. |
21:46.00 | russellb | it's not just used as a timing source |
21:46.04 | rob0 | I thought the dahdi_dummy was needed as a timing source, but only for conferencing? |
21:46.12 | rob0 | I didn't know that part |
21:46.12 | ghenry | so, should be DAHDI, sorry |
21:46.20 | russellb | However, we have code that replaces the need for it for conferencing, as well |
21:46.27 | russellb | it is on the roadmap for inclusion in 1.6.2 |
21:46.33 | ghenry | ok |
21:46.48 | hardwire | woot |
21:46.55 | hardwire | I use asterisk under openvz |
21:46.57 | hardwire | timing works fine |
21:47.03 | ghenry | so with 1.6.2 you can, if no hardware needed say for use in xen, no DAHDI is needed. |
21:47.04 | hardwire | it's actually somewhat evil |
21:47.30 | ghenry | I mean, no need for analgoue or digital connections etc. |
21:48.28 | *** join/#asterisk ThreeSevenths (n=ThreeSev@exchngsvr.new-atom.net) |
21:49.57 | ghenry | I could just add a pri card to the box for timing to save all this I suppose |
21:50.28 | ghenry | russellb: can Digium Business Edition run under xen? |
21:50.45 | russellb | it does not have additional functionality that is not open source |
21:51.00 | ThreeSevenths | hello all, I am trying to implement dundi to connect two asterisk boxes together. One is on the internet with a public IP and iptables. The other is behind a cisco/linksys hardware firewall with a private IP. UDP port forwarding for port 4520 has been added on both systems. Each system will successfully perform a query for the other's entity ID. only the system which is publicly available can receive a lookup from the other one behind the |
21:51.22 | ThreeSevenths | does anyone have any idea what the problem may be? Any help or hints would be appreciated |
21:51.32 | ghenry | russellb: I mean is the rpath xen kernel patched or is zaptel? |
21:52.14 | russellb | ghenry: i'm not sure what you mean ... |
21:52.16 | ghenry | if it can, that might be quite interesting |
21:52.17 | russellb | the issues are dahdi/zaptel issues |
21:52.31 | russellb | ThreeSevenths: if you have port forwarding on, it should work ... |
21:52.49 | ghenry | yeah, but I thought you can patch ztdummy etc. Maybe I googled wrong |
21:52.57 | russellb | ThreeSevenths: do you have registration turned on from the private box? |
21:53.08 | russellb | ghenry: yeah, you can i think |
21:53.15 | russellb | ghenry: i think there might be a patch on bugs.digium.com for it ... |
21:53.23 | De_Mon | lol. xkdc is great. |
21:53.30 | De_Mon | oops xkcd |
21:53.32 | russellb | ThreeSevenths: I mean the "register=yes" option |
21:53.38 | ghenry | Anyone tried the 51i Aastra phone and experience after a couple of mins "no service" even though it registers first time? |
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21:53.43 | ghenry | russellb: OK, thanks. |
21:54.44 | ghenry | http://bugs.digium.com/view.php?id=9592 |
21:54.47 | ThreeSevenths | russellb: no yet |
21:54.50 | ThreeSevenths | let me try that now |
21:55.34 | ThreeSevenths | russellb: register=yes has not helped |
21:56.30 | russellb | ThreeSevenths: have you done a packet capture to see if the query from the public box is even getting there? |
21:56.40 | russellb | that is where i would look to see if it is a network problem or an asterisk problem |
21:56.49 | ThreeSevenths | russellb: yes we have, both boxes are communicating |
21:57.16 | russellb | Alright, next thing to look at is the "dundi debug" output, to see what Asterisk does with the query |
21:57.21 | ThreeSevenths | when i do a lookup from one to the other, we receive a reply with HINT DONTASK|UNAFFECTED |
21:57.34 | russellb | Ah, so it's a configuration issue of some kind. |
21:58.13 | ThreeSevenths | yes, and i'm just not sure where it is, because the lookup works one way, but not the other, and we have the exact same config at each end |
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21:59.49 | russellb | ThreeSevenths: I can sanity check the configuration if you'd like. just pastebin it. |
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22:01.50 | alfa202 | hello |
22:02.07 | ThreeSevenths | http://pastebin.com/d2a12889 |
22:02.14 | russellb | looks |
22:02.57 | ThreeSevenths | we are using the e164 context while we debug and plan to move to the priv context later |
22:03.40 | russellb | nods |
22:04.45 | russellb | ThreeSevenths: and in extensions.conf, you have the extensions you want to be able to lookup available in dundi_local, right? |
22:05.26 | ThreeSevenths | nope |
22:05.33 | ThreeSevenths | actually, you just pinpointed the issue |
22:05.35 | russellb | that would be the problem, then. |
22:05.37 | russellb | :-) |
22:05.43 | ThreeSevenths | it's dundi-local in ext.conf |
22:05.48 | russellb | ah ha! |
22:06.14 | *** join/#asterisk kink0 (n=xchat@212.170.176.86) |
22:06.16 | kink0 | hello |
22:06.28 | ThreeSevenths | et voilla, it works |
22:06.32 | russellb | yay! |
22:06.47 | kink0 | there any way to return NO DURATION and/or preseted q931 cause code from Background() ? |
22:06.58 | ThreeSevenths | russellb: thanks alot, pebkac |
22:07.06 | russellb | you're welcome. |
22:07.46 | kink0 | i.e. I want to play a message "the number is wrong..." and returns Cause 27 or some other cause, at same time not duration to avoid billing |
22:08.35 | russellb | kink0: well, I know you can control the cause code. |
22:08.43 | russellb | kink0: as an argument to the Hangup() application. |
22:09.01 | russellb | I don't think there is anything you can do beyond that |
22:09.34 | *** part/#asterisk badcfe (i=cso@78.156.5.233) |
22:12.12 | kink0 | russellb I tryed Playback(wrong_number + Hangup(27) , but I faced these problems... |
22:12.36 | kink0 | first the call has duration , seconds the Cause seems ignored |
22:13.04 | russellb | kink0: did you Answer() before the Playback()? |
22:13.08 | russellb | if so, don't. |
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22:14.48 | russellb | kink0: also make sure you set the 'noanswer' option for Playback() |
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22:16.18 | kink0 | russellb, no, I have not Answer() |
22:16.41 | kink0 | I did not put "noanswer" option for playback ... let me try it now with noanswer option |
22:16.49 | russellb | k |
22:17.12 | russellb | without it, Playback implicitly answers |
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22:24.03 | kink0 | I am trying.... but with noanswer option file is not played, just fast busy |
22:25.00 | russellb | ok, one more thing to try ... |
22:25.10 | russellb | run Progress() right before the Playback() using 'noanswer' |
22:25.41 | russellb | after that I give up, as I need to go to bed soon :-) |
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22:27.05 | [TK]D-Fender | russellb: Ah yes... you're in Sprouts-land right? :) |
22:27.39 | russellb | yup |
22:31.15 | kink0 | sprouts ? |
22:32.16 | WilliamK | sprouts are good, didn't your mother teach/tell you that? :) |
22:32.39 | russellb | I'm in Brussels, heh |
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22:34.12 | WilliamK | same as - |
22:34.14 | WilliamK | The Brussels (or brussels) sprout (Brassica oleracea Gemmifera Group) of the Brassicaceae family, is a Cultivar group of wild cabbage cultivated for its small (typically 2.54 cm or 11.5 in diameter) leafy green buds, which resemble miniature cabbages. |
22:34.55 | WilliamK | sorry russell - no escaping! |
22:35.46 | kink0 | ahhhh here Spain |
22:36.09 | kink0 | How is dressed the Petit Juilane today ? :) |
22:36.58 | russellb | I didn't make it over to see that .. |
22:37.50 | russellb | i'm out. good night, everyone! |
22:37.55 | kink0 | good night |
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22:42.38 | wonderworld | is there a dedicated linux server hoster that offers an E1 / T1 as well? |
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23:09.41 | keith4_ | I'm converting a system from cdr-csv to postgres, and I want to push the existing CSV records into the database. The fields aren't in the same order... is there documentation on the CSV format somewhere? |
23:10.01 | keith4_ | most of the fields are obvious, but there are a few that i'm not sure about |
23:10.59 | [TK]D-Fender | keith4 : Yes.. in your TARBALL |
23:13.15 | keith4_ | my fault. i asked the wrong question... |
23:13.24 | keith4_ | is there documentation anywhere *other* than in the source code? |
23:14.05 | seanbright | do you have to enable something magical in zapata.conf to get CID name over PRI? |
23:14.23 | seanbright | i don't even see the IE coming over on an inbound call, so i'm assuming it's on the telco side |
23:14.46 | [TK]D-Fender | keith4 : Maybe somebody cut & paste it on the WIKI... until then go look in the damn source tarball :p |
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23:16.14 | cesau | in sip.conf, should it be "[general]" or "[global]" ? |
23:16.18 | cesau | v1.4 |
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23:18.31 | jaytee | [general] |
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23:26.30 | keith4_ | [TK]D-Fender: despite your best efforts, i think you pointed me in the right direction |
23:26.46 | keith4_ | sure, the validity of this is questionable... but it looks right to me http://www.voip-info.org/wiki/view/Asterisk+CDR+csv+conversion+mysql |
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23:49.24 | joako | seanbright: Yes, but first and foremost you need the provider to send the name over the PRI: |
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