00:01.14 | citywok | Hello |
00:01.23 | r0m|u | hola |
00:08.48 | hardwire | helloha |
00:10.07 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
00:11.30 | Naikrovek | holy moly it is hard to convince mgmt that used cubes aren't a ripoff. "$800 per desk?! Are they made of gold?!" me: "New they're $5k per seat." |
00:11.52 | hardwire | yeh |
00:12.38 | hardwire | Naikrovek: this is where you request a brand new water cutter machine and a home depot card. |
00:12.43 | hardwire | and a sewing machine |
00:12.45 | hardwire | and profit |
00:13.17 | Naikrovek | I set down $41k of material requests on my CFOs desk today. cubes were $24k of that. |
00:13.35 | Naikrovek | he didn't even blink at anything else. the cubes made his head pop off his shoulders. |
00:13.49 | Naikrovek | i should have asked him what he thought they cost. |
00:14.05 | Naikrovek | even though I didn't, I know what his answer would be: "why can't we go to walmart?" |
00:15.36 | p3nguin | Would they prefer $99 desks and $49 chairs from Walmart instead of cubes? |
00:19.03 | Naikrovek | i'm beginning to thinkso |
00:19.13 | Naikrovek | everything is too expensive |
00:19.30 | Naikrovek | it's not that we don't have the money, it's that the CEO gets involved in every little decision and says "that's too much." |
00:19.53 | F2Knight | what are these cubes? |
00:20.00 | p3nguin | I can't understand how spending 4 times as much is a money saving plan. |
00:20.05 | p3nguin | But that's what they do. |
00:20.45 | Naikrovek | SteelCase Series 9000 is what I want to get. They are inexpensive used, and common. |
00:20.54 | [TK]D-Fender | F2Knight: They're like rectangles... only equally long on all sides ;) |
00:21.24 | F2Knight | [TK]D-Fender, so not like a TARDIS then |
00:21.33 | Naikrovek | replacement parts (if ever needed) will be easy to obtain, and if we grow our office again, more of that make/model will be easy to obtain. |
00:21.51 | hardwire | just get some wood and cement board |
00:21.53 | hardwire | and nails! |
00:21.57 | hardwire | then! |
00:22.01 | hardwire | get some appliance epoxy |
00:22.04 | hardwire | and spray it on |
00:22.06 | p3nguin | Screw planning for the future, you're just spending money on something you don't even yet need! |
00:22.09 | hardwire | and make indestructible cubes |
00:22.10 | p3nguin | :/ |
00:22.20 | hardwire | better yet |
00:22.21 | hardwire | truck bed liner. |
00:22.26 | Naikrovek | lol |
00:22.27 | hardwire | cheap.. effective. |
00:22.40 | Naikrovek | well i have requirements driving this, i'm not spending for spending's sake |
00:22.55 | Naikrovek | it doesn't matter to me so much, but the stakeholders will shit themselves if this doesn't happen |
00:23.16 | hardwire | I can't wait until you can just put a bunch of used banana peels (don't ask) in the middle of a room and have nanobots turn it into a cube farm. |
00:23.26 | Naikrovek | hardwire: also, it must meet fire code. |
00:23.37 | hardwire | I'm pretty sure indestructble meets fire code. |
00:23.44 | Igneous | p3nguin: sorry, mind if I bug you with another dumb question? |
00:23.56 | *** join/#asterisk tmrhmdv (4575afcd@gateway/web/freenode/ip.69.117.175.205) |
00:24.13 | p3nguin | igneous: If you already asked it and I didn't answer, don't expect me to answer if you ask me directly. |
00:24.21 | Naikrovek | lol |
00:24.22 | Igneous | I know you're regarded as the dialplan guru in here. |
00:24.23 | Naikrovek | just ask |
00:24.24 | Naikrovek | maybe i know, hell |
00:24.49 | Naikrovek | p3nguin? dialplan guru? really? |
00:24.52 | F2Knight | or maybe I do too. |
00:24.59 | *** part/#asterisk pietro (~pietro@88-149-224-154.dynamic.ngi.it) |
00:25.08 | p3nguin | ~p3nguin |
00:25.08 | infobot | p3nguin strives to maintain his elitist reputation by refusing to use GUIs where they aren't beneficial. See: FreePBX. |
00:25.13 | F2Knight | hands p3nguin a fish |
00:25.19 | p3nguin | hmm |
00:25.22 | Igneous | I'm just trying to figure out why ${QEHOLDTIME} isn't set unless it's called by membermacro (or after the channel is established). |
00:25.33 | p3nguin | Interesting... but no mention of dialplan guru. |
00:26.14 | F2Knight | ~F2Knight |
00:26.35 | F2Knight | yep insignificant |
00:26.50 | tmrhmdv | Hi folks! I am trying to achieve the best install. I've installed Asterisk over 16x just today and it... sucks. I'm a linux noob, but I know that it's possible to automate the install.process. So, can someone point me in the right direction, please? |
00:26.55 | hardwire | heheh |
00:27.14 | p3nguin | tmrhmdv: What distro do you use? |
00:27.24 | tmrhmdv | p3nguin: Amazon Linux |
00:27.38 | p3nguin | Is that derived from something more... mainline? |
00:27.54 | Naikrovek | he's on EC2 sounds like |
00:27.59 | Naikrovek | ...maybe |
00:27.59 | tmrhmdv | Yes, it's like CentOS/RedHat/Fedora etc. uses yum |
00:28.04 | tmrhmdv | Yes Iam |
00:28.08 | Naikrovek | knew it. |
00:28.17 | F2Knight | Source install... : cd asterisk ; ./configure ; make ; make install ; make samples |
00:28.22 | p3nguin | Okay, so install the digium/asterisk repo, and install asterisk and friends with yum. |
00:28.29 | Naikrovek | asterisk will work, but you should hone your craft on a virtual machine before you pay to learn it on EC2 |
00:28.34 | citywok | do what F2Knight said. wget it, tar it, cd in to it, ./config; make; make install |
00:28.48 | citywok | Naikrovek: you can learn for free on ec2 w/ a tiny instance |
00:29.02 | Naikrovek | citywok: ah yeah forgot about that. |
00:29.17 | citywok | although i've never tried * in EC2 i only use it for web hosting |
00:29.20 | Naikrovek | still - you have to select the proper kernel or you'll have timing issues |
00:29.31 | Naikrovek | timing issues = audio quality issues |
00:29.36 | F2Knight | I actually keep an SVN repo upto date locally, and run a bash script every night to rebuild and then create a deb package that I put on machines. |
00:29.42 | Naikrovek | audio quality issues = preception of crap phone system. |
00:29.46 | Naikrovek | so, try on local hardware |
00:29.52 | citywok | F2Knight: i hope that is a dev environment you are testing in |
00:30.03 | F2Knight | citywok, yeppers. |
00:30.29 | tmrhmdv | Let me correct my question. I'm not having any issue with installing. It's just that I'm installing it over and over. Therefore just wanted to know if I could automate it with some (bash?) script or whatever it's called. + I'll have to install it on 4 server |
00:30.53 | Naikrovek | tmrhmdv: with packages you can do it really quickly and easily |
00:30.54 | p3nguin | Why are you installing it over and over and over? |
00:30.54 | F2Knight | Source install... : cd asterisk ; ./configure ; make ; make install ; make samples |
00:30.59 | Naikrovek | p3nguin: he's experimenting |
00:31.01 | *** join/#asterisk troyt (~troyt@PariahZero.broker.freenet6.net) |
00:31.07 | Naikrovek | local virtual machines would solve this. (snapshots) |
00:31.08 | p3nguin | I think THAT is more of the problem than the fact that it takes a while to do. |
00:31.19 | tmrhmdv | Yes |
00:31.25 | tmrhmdv | Exactly, local VMs don't work |
00:31.30 | p3nguin | Unless you are changing the acutal code, there is no reason to keep installing again and again. |
00:31.31 | tmrhmdv | you've to install the OS and blah blah |
00:31.49 | citywok | what p3nguin, you shouldn't need to reinstall unless you are doing dev in the source |
00:32.02 | citywok | s/p3nguin/what p3nguin/ |
00:32.03 | p3nguin | Changing the configuration does not require reinstallation of the software. |
00:32.04 | Naikrovek | local VM will work. You can use them for testing. You're a linux newb and you want to deploy asterisk? I assume for production? Practice locally |
00:32.10 | F2Knight | tmrhmdv, your choices are easy.. yum install asteirsk |
00:32.15 | Naikrovek | yep |
00:32.17 | F2Knight | or if from source. |
00:32.19 | F2Knight | Source install... : cd asterisk ; ./configure ; make ; make install ; make samples |
00:32.31 | Naikrovek | yum install asterisk18 asterisk18-addons asterisk18-config |
00:32.31 | tmrhmdv | I'm comiling it myself |
00:32.32 | Naikrovek | then wait |
00:32.32 | F2Knight | and you run that after you install the OS |
00:32.33 | Naikrovek | then done |
00:32.35 | tmrhmdv | compil* |
00:32.41 | tmrhmdv | compiling* dammit |
00:32.57 | p3nguin | I still don't see an actual problem. |
00:33.02 | tmrhmdv | No, no guys :D |
00:33.02 | F2Knight | oh a better way would be this... |
00:33.36 | Naikrovek | tmrhmdv: okay take your time and explain because we're missing something |
00:33.40 | F2Knight | cd asterisk ; contrib/scripts/install_req install ; ./configure ; make ; make install ; make samples |
00:33.42 | autofsckk | p3nguin: hello, can you help me with my incoming calls from the trunk? it is already registering, i can make calls, but when receiving my ITSP says that it looks like busy signal |
00:33.42 | p3nguin | If you explain an actual problem, maybe you can get an actual answer. |
00:33.50 | tmrhmdv | There's no problem with installing, this is more of a Linux/CLI question. You know you can install many software with *.sh script that include all of your 'yum install yada yada' |
00:33.54 | F2Knight | that would get all the 'extra' files and dependency you need |
00:34.07 | Naikrovek | tmrhmdv: yes |
00:34.14 | tmrhmdv | I know all the packages/dependencies I need |
00:34.17 | Naikrovek | okay |
00:34.29 | F2Knight | tmrhmdv, do you know what a .sh file is? |
00:34.32 | Naikrovek | he knows |
00:34.47 | p3nguin | autofsckk: Show me some configuration and I'll tell you what I think is wrong with it. |
00:34.56 | Naikrovek | i dont' think english is his first language; give him a break |
00:34.58 | Naikrovek | let him say it |
00:35.06 | F2Knight | Naikrovek, I don't know that he does.. if he did he would now how it works. |
00:35.24 | p3nguin | autofsckk: I see the pastebin. |
00:35.35 | tmrhmdv | *facepalm* english isn't my 1st lang. but i'm fluent, i just can't explain what i'm trying to do xD |
00:35.41 | p3nguin | autofsckk: I need to also see your register statement. |
00:36.14 | tmrhmdv | F2Knight: I don't. I am a HS student, I was just playing around with asterisk and install it on 4 servers. |
00:36.34 | tmrhmdv | But I don't want to install it from yum packages |
00:36.39 | tmrhmdv | I want to install it from source |
00:36.54 | citywok | tmrhmdv: then ./configure; make; make install in a shell / bash script |
00:36.57 | p3nguin | autofsckk: I want to see your entire sip.conf, too. Hide only your passwords. If there is anything else changed or hidden, I'm not going to waste my time. |
00:37.41 | F2Knight | tmrhmdv, okay you want me to send you my script? |
00:38.01 | tmrhmdv | F2Knight: can you please, I just want to see how it's done |
00:38.08 | p3nguin | autofsckk: Change your register statement. |
00:38.21 | p3nguin | autofsckk: Append /phonenumber |
00:38.32 | p3nguin | autofsckk: user:pass@host/phonenumber |
00:40.01 | p3nguin | autofsckk: Your ITSP is probably not behind nat, so change nat=yes to nat=no |
00:40.23 | autofsckk | ok done |
00:40.27 | p3nguin | autofsckk: insecure=very shouldn't even work. If you need to use it, it would be insecure=port,invite |
00:40.39 | *** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net) |
00:41.14 | p3nguin | autofsckk: Then define an extension in your incoming_calls context for your phone number. Make it do something useful. |
00:42.01 | Naikrovek | tmrhmdv: ah so you want to install from source, is all? that's not too hard. |
00:42.08 | tmrhmdv | "If you need to install Asterisk onto several machines, you may wish to build a set of scripts to help automate this process." That "build a set of scripts" is what I am interested in |
00:42.11 | tmrhmdv | :D |
00:42.11 | Naikrovek | i had a good link for this, let me see if i can find it. |
00:43.00 | F2Knight | Naikrovek, I just sent him a install script |
00:43.22 | Naikrovek | ah |
00:43.23 | Naikrovek | tyvm |
00:43.57 | tmrhmdv | thank you! ooh, finally :D |
00:44.05 | tmrhmdv | was able to explain |
00:44.21 | F2Knight | http://pastebin.com/x9yuBAHR |
00:44.41 | F2Knight | as you will notice tmrhmdv, it is the exact same thing you would type at the command line |
00:44.45 | F2Knight | nothing special at all |
00:45.02 | p3nguin | autofsckk: All I see in paste 18 is a failed attempt at an extension pattern. |
00:45.14 | autofsckk | p3nguin: i tried what you told me, but it still doesnt work, i cant see any incoming call al CLI :S |
00:45.27 | p3nguin | autofsckk: sip set debug on |
00:45.47 | p3nguin | autofsckk: Make a call to your system from your mobile or something. |
00:45.52 | tmrhmdv | Oh, ok. I just found typing same lines of command on 4 machines would be time consuming, anyways, thank you F2Knight, Naikrovek, citywok and p3nguin |
00:46.02 | p3nguin | autofsckk: Pastebin the entire thing. It may be very long. |
00:46.02 | F2Knight | If you wanted to get a litlte more fancy you could get the latest name from the website and automaticly extract it and such using variables, but that will require a litle more learning on your part. |
00:46.40 | autofsckk | p3nguin: the only way i can call myself right now is with the same voip phone :S |
00:47.01 | F2Knight | 4 machines one single line of commands, not a waste... 400 machines ... well either way you have to get the script on to the box. so thats why most people dont bother with it. |
00:47.21 | autofsckk | from the sip debug i get may info about destroying SIP dialog and diferent ips |
00:48.01 | p3nguin | autofsckk: Without seeing what's going on, I can't help further. You're limiting me by not giving me entire configs from the start, and you're limiting me further by not being able to make test calls to troubleshoot. |
00:48.42 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
00:48.54 | p3nguin | I've seen the extensions.conf, but I haven't seen the sip.conf. |
00:50.51 | tmrhmdv | F2Knight: I never knew what a Linux was :P I got started only a few days ago...excuse my noob-ness |
00:51.22 | Naikrovek | we were all newbs once |
00:51.29 | Naikrovek | many still are |
00:52.08 | tmrhmdv | :) |
00:53.02 | p3nguin | autofsckk: There is no autofallthrough setting in sip.conf. Remove that line. |
00:53.48 | p3nguin | autofsckk: You've defined a localnet, which tells me your asterisk is behind NAT, but you have not configured nat=yes nor any externaddr/externhost values. |
00:53.49 | Naikrovek | linux is pretty awesome as a learning tool |
00:53.59 | Naikrovek | i hate it as a desktop, but as a server it's tolerable |
00:54.13 | tmrhmdv | I agree, I 'm already falling in love with it |
00:54.41 | zyphlar | "everything's a file" is a wonderful concept. i'm still struggling with automating configuration of my windows servers and i've been doing windows for years |
00:55.00 | tmrhmdv | Ooh, btw, if any onf these: Leif Madsen Jim Van Meggelen Russell Bryant gentlemen are here, I wanna thank them for their book, Asterisk™: The Definitive Guide, it taught me a lot! |
00:55.08 | autofsckk | p3nguin: well i think that for now i can get rid of that line and just make asterisk work internally right? after that i can make it receive connections from the outside? |
00:55.17 | Naikrovek | zyphlar: that's no fault of yours; windows doesn't really facilitate this well |
00:55.19 | p3nguin | autofsckk: I would also name my peer for my ITSP something a bit more descriptive, such as the name of the provider. |
00:55.45 | p3nguin | autofsckk: If your asterisk is behind nat, configure it correctly for working behind nat or don't bother configuring it at all. |
00:55.54 | p3nguin | Your problem was that calls are not coming in from outside. |
00:56.00 | p3nguin | That involves nat. |
00:56.42 | p3nguin | Make sure you forward the necessary ports at the firewall: UDP 5060, and whatever UDP port range is defined in rtp.conf (usually 10000-20000). |
00:57.31 | Naikrovek | on recent Windows OSs, powershell goes a LONG way to reaching that goal, though, zyphlar |
00:57.38 | Naikrovek | especially on the server |
01:02.42 | *** join/#asterisk ketas (~ketas@ketas6-sixxs.si.pri.ee) |
01:03.16 | *** join/#asterisk Hanumaan (~Hanumaan@132.230.17.77) |
01:03.53 | autofsckk | p3nguin: i think its working now :D |
01:04.20 | autofsckk | thanks a lot |
01:04.30 | p3nguin | I guess that's a good thing. Confirm for a fact that it works. |
01:06.54 | autofsckk | p3nguin: well the thing is that i dont have credit on my cell, so the only way i can test it is by auto calling me with the same voip account, so i did it and i now receive the call on my pap and twinkle here on this computer, calling from my netbook running asterisk and twinkle too, so yes, it works now, thanks a lot :D |
01:07.12 | p3nguin | Great. |
01:09.54 | *** part/#asterisk osas (~osas@nslu2-linux/osas) |
01:10.45 | *** join/#asterisk LiuYan (~liu.yan@211.154.128.135) |
01:11.55 | autofsckk | p3nguin: is something very strange, remember 2 days ago that i asked for help too? i was trying to configure an asterisk running from a vm on centos, that couldnt register, nor the asterisk running here on this computer, using the same info that im running on the netbook, what could it be wrong on those boxes that couldnt register with mi ITSP ? any ideas? |
01:12.26 | *** join/#asterisk timahvo1 (~rogue@197.176.218.193) |
01:13.04 | p3nguin | autofsckk: Without seeing the evidence, I can't even begin to guess. |
01:13.23 | p3nguin | If you would have given me the sip debug, maybe I would have been able to say what's wrong. |
01:14.30 | *** join/#asterisk jasonwert (~w3rt@66-227-208-111.dhcp.trcy.mi.charter.com) |
01:14.53 | *** join/#asterisk raden (~J@66-168-15-100.dhcp.stpt.wi.charter.com) |
01:15.01 | raden | Naikrovek, Yo !!! you around bro ? |
01:15.08 | Naikrovek | yo bro |
01:15.10 | Naikrovek | gi joe |
01:15.11 | autofsckk | i will try now with this files that are working too see if it still doesnt work, so for sure ill will be asking for help maybe later or tomorrow |
01:15.17 | tmrhmdv | autofsckk: I experienced your problem few times and it was because of NAT for me, but it may be something else that's causing your problem |
01:15.20 | Naikrovek | knowing is half the battle-o |
01:15.34 | Naikrovek | my uncle was the voice of lion-o (not joking there) |
01:15.42 | hardwire | nuhuh |
01:15.45 | autofsckk | tmrhmdv: i disabled iptables on both boxes, and it still wasnt working |
01:15.49 | raden | Naikrovek, for wisp I want to offer FTP backup service or something , 10 GB limit per user , how would I set this up in ftp ? |
01:15.51 | Naikrovek | yeah-huh |
01:15.56 | p3nguin | NAT usually doesn't prevent registration to your ITSP. |
01:15.58 | hardwire | nuhuh |
01:16.21 | Naikrovek | raden: ftp server software may offer this. i don't know how quotas work in linux but that woudl be the other way |
01:16.26 | Naikrovek | hardwire: yeah-huh |
01:16.28 | autofsckk | i know, you can register, but you have audio problems, or not receiving calls right? |
01:16.32 | hardwire | cool |
01:16.41 | raden | could I make each user directory a partition ? |
01:16.48 | raden | ( seems a lil extreme lol ) |
01:17.18 | WIMPy | raden: Quota |
01:17.18 | p3nguin | autofsckk: Usually that's what happens with misconfigured NAT settings. |
01:17.19 | Naikrovek | hardwire: well, it's my wife's uncle. Her cousin is trudy weigel on reno 911 |
01:17.29 | Naikrovek | raden: linux kernel supports user quotas |
01:17.41 | tmrhmdv | p3nguin: yeah, as I said there were many factors, like enabling 5060-5061, 10000-20000 and etc ports |
01:18.06 | tmrhmdv | autofsckk: When I tried installing and setting up Asterisk on local VMs, all failed and I couldn't get anything to work. Now, I'm just using EC2 and everything works with no problem + it doesn't cost too much |
01:18.09 | p3nguin | I have no idea why you would "enable" 5061. |
01:18.23 | hardwire | Naikrovek: claimin to faimin.. I like it |
01:18.23 | p3nguin | since SIP is 5060, and all. |
01:18.25 | tmrhmdv | Me neither, but I had to on EC2 |
01:18.35 | hardwire | my uncle was on taxi |
01:18.39 | p3nguin | I doubt you had to. |
01:18.44 | p3nguin | since SIP is 5060, and all. |
01:19.00 | Naikrovek | hardwire: who did he play |
01:19.02 | tmrhmdv | I read a 'guide' that said I 'had' to :) |
01:19.04 | F2Knight | raden, disk quota's... however... you have to run a script to determine the disk usage for each user. It is cpu intensive action and you must run it when ever you want to check. The end result is that you could have a user use more then 10GB of data up until the point that you check |
01:19.15 | p3nguin | tmrhmdv: Post a comment that informs them of their mistake. |
01:19.16 | hardwire | john Burns (Randall Carver) |
01:19.22 | Naikrovek | hardwire: nice |
01:19.34 | F2Knight | 5061 is used for sip over TCP i believe |
01:20.03 | Naikrovek | hardwire: he was in There Will Be Blood |
01:20.06 | Naikrovek | I love that scene |
01:20.07 | p3nguin | I guess forwarding UDP 5061 wouldn't do much good for that, then. |
01:20.14 | F2Knight | nope |
01:20.30 | Naikrovek | "Now, I'm not going to waste your time, Mr. Bankside; I'd appreciate it if you didn't waste mine." |
01:20.34 | Naikrovek | LOVE that movie |
01:20.39 | hardwire | that's all I got for fame |
01:20.48 | hardwire | oh.. and I ended up in the kernel source more than once.. |
01:20.51 | hardwire | so that's cool. |
01:21.03 | hardwire | but that's not mainstream :) |
01:21.36 | Naikrovek | if you got in there early enough you would have been part of valinux ipo. that would have netted you a lot of dougth |
01:21.38 | Naikrovek | dough& |
01:21.40 | Naikrovek | * |
01:21.46 | hardwire | yeh.. no. |
01:21.50 | hardwire | I'm a late bloomer |
01:22.13 | Naikrovek | they used the linux kernel source to gather part of their list of pre-ipo offers. many millionaires were made that day |
01:22.41 | Naikrovek | watched it open at $100 and close at $400 (if memory serves) |
01:23.27 | Naikrovek | eh wikipedia says i'm full of shit |
01:23.30 | Naikrovek | doesn't disagree. |
01:23.37 | hardwire | haha |
01:23.53 | tmrhmdv | :D |
01:24.13 | tmrhmdv | sometimes, memory fails |
01:24.29 | Naikrovek | sometimes becomes usually eventually |
01:24.43 | tmrhmdv | true |
01:25.54 | Naikrovek | hmm. how do i convince my mgmt that these silly cubes are worth the money |
01:25.55 | hardwire | my buddy Gareth worked for va.. thats aall I remember. |
01:26.09 | *** join/#asterisk seraphie (~erin@75.76.38.159) |
01:26.14 | hardwire | Naikrovek: piss on the other offers.. while they sit there watching. |
01:26.22 | Naikrovek | heh |
01:26.25 | hardwire | to make it more dramatic.. drink a lot of 151 something |
01:26.27 | hardwire | then light it. |
01:26.56 | Naikrovek | well they were the least expensive by a little bit, and i've compared the price of these cubes to new. dramatic difference there |
01:27.02 | Naikrovek | not sure how else i can lay it out honestly |
01:27.07 | *** join/#asterisk coppice (~chatzilla@m121-202-79-203.smartone-vodafone.com) |
01:27.38 | Naikrovek | middle management needs 16 more cubes in the space provided. we have 12, but to fit 28 i have to replace everything. this is what that costs |
01:27.49 | Naikrovek | up to the middle managers who need the seats to do the selling at this point |
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01:39.38 | bluregard | good evening all |
01:40.36 | bulletrtr | Hello |
01:40.42 | bluregard | does anyone know how the AMI Uniqueid: is calculated? |
01:40.59 | bluregard | I'm wondering if it's suitable as a DB primary key |
01:50.18 | F2Knight | bluregard, the UniqueID if I recall is not really all that unique. What is the application that you are requiring a unique ID for? |
01:51.48 | F2Knight | but it is composed of epoch time of when a call starts, plus a monotonically incrementing interger. |
01:52.56 | F2Knight | they will only be unique for calls on that box... this is using the ${UNIQUEID} |
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01:58.07 | bluregard | f2knight: I'd be using it as a primary key for a mysql table containing a list of calls |
01:58.55 | bluregard | its not that big of a deal if it's not unique enough, I can always just use an auto_increment if I have to. I'd rather be safe than risk a duplicate key |
02:07.11 | F2Knight | bluregard, I would just create your own key. You are assured that will work. If you want to store the unique ID from asterisk for later searches, you can set an index on it to make looks eaiser. but I would not use it personally. |
02:12.51 | bluregard | f2knight: yeah, I'll probably just end up using it to relate events with individual calls rather than as a DB key |
02:16.20 | bluregard | did google voice break something again? I can get calls to go out from asterisk to my cell phone, but when I answer my cell it doesn't acknowledge I answered it, it just times out. |
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03:24.43 | andyoutside | remind me how do you have asterisk read text please. More the url for that info |
03:24.52 | dijib | festival |
03:25.10 | dijib | using text2speech in a system command or the festival command |
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03:45.32 | bulletrtr | Can anyone recommend a good quality non HP/Dell/Gateway rackmount server that can support RAID, redundant power supplies and have capability to hold a Sagnoma or Digium card? |
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04:29.23 | andyoutside | bulletrtr: why not any of those price? |
04:38.57 | SwK | bulletrtr: what not just get a dell off the secondary market theres tons of them cheap out there |
04:39.33 | SwK | like a dell 1950, complete with raid, remote lights out management redundant PSUs and hold various PCI/PCIe cards |
04:47.34 | bulletrtr | Price is an issue, but I am more concerned about redundancy and long term service to customers. |
04:48.03 | bulletrtr | I have a 2950 now which I like, but they don't seem to make them anymore. |
04:48.24 | bulletrtr | Has anyone heard of supermicro? |
04:48.39 | SwK | bulletrtr: theres a ton of that stuff on the secondary market |
04:48.43 | SwK | thats the great part about dells |
04:48.45 | bulletrtr | Dell's service has been weak. |
04:49.12 | bulletrtr | I looked on Ebay for new stuff. The selection was a bit weak. |
04:50.26 | andyoutside | I use 2950 from ebay |
04:51.00 | andyoutside | normally we will call up a company that has a lot of them and tell them what we want in it and they give us a price |
04:52.05 | SwK | theres better places then ebay heh |
04:52.22 | SwK | bulletrtr: are you in the states? |
04:52.34 | bulletrtr | Yes. |
04:52.42 | SwK | check out stikc.com |
04:53.23 | SwK | let me refer you to my sales guy there if you see something you like (disclaimer; the do a referal credit which I will use heh) |
04:53.47 | SwK | ever server I have purchased in the last 4 years i have purchased from therem |
04:53.50 | SwK | errr them |
04:54.13 | bulletrtr | They look well put together and have been high reliability? |
04:54.48 | andyoutside | nice |
04:55.39 | bulletrtr | It looks like they have some reconditioned stuff too. Thay might be an option. |
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05:00.24 | andyoutside | and is this for asterisk? |
05:01.00 | andyoutside | who is your tel provider |
05:01.55 | bulletrtr | No, it is for a email server/DNS/website, but I want it as a backup in the future for my present Asterisk server. Centurytel. |
05:03.24 | bulletrtr | I am looking at starting a WISP and selling VOIP through my Asterisk server. |
05:08.47 | andyoutside | in that case look at maybe two servers if you really want it to stay up |
05:08.59 | irroot | bulletrtr not always the easiest running voip over wifi ... but can be done |
05:09.26 | andyoutside | I do it all the time |
05:09.45 | andyoutside | plus across country to the server |
05:10.02 | irroot | bulletrtr what wifi kit you want to use we have some experiance doing it arround town |
05:10.12 | SwK | bulletrtr: i dont nothing but ITSP and CallCenter stuff |
05:10.53 | irroot | andyoutside bulletrtr it works no doubt about it Johannesburg and Durban have plenty sites on wifi |
05:11.31 | SwK | bulletrtr: and these guys sell refurb stuff... very little new stuff... the con is its not brand new so you wont get the lastest thing dell just launched yesterday... but the stuff is refurb'd/reconditioned before they ship it out |
05:11.37 | andyoutside | The biggest thing that you have control of is your isp |
05:11.54 | andyoutside | we use level3 |
05:12.04 | SwK | andyoutside: how was your internet this morning? heh |
05:12.09 | bulletrtr | We are planning on using the new UBNT AirMax and installing many sites close to the populations to reduce interference and offer better throughput (low latency back to Asterisk) |
05:12.37 | andyoutside | internet was working but another network messed up their routing tables |
05:12.51 | andyoutside | so only part of the internet was avalable |
05:13.16 | SwK | latest I heard they botched a juniper MX upgrade again |
05:13.31 | SwK | which isnt hte first time i have heard of such things from L3 |
05:13.59 | andyoutside | heck I will go see what the log says |
05:16.25 | dijib | http://ontario.kijiji.ca/c-buy-and-sell-computers-Dell-PowerEdge-2950-II-Server-2x-DC-3-0GHz-8GB-4x-73GB-15k-W0QQAdIdZ327676121 |
05:17.06 | irroot | andyoutside mmm how another network can mess up there tables mmmm sounds like they did not set up BGP filtering properly |
05:17.28 | andyoutside | The IP NOC has advised that multiple links network wide are bouncing, which is affecting IP traffic. |
05:18.15 | SwK | <PROTECTED> |
05:18.28 | bulletrtr | dijib: Thanks! |
05:18.32 | SwK | irroot: Level3 admitted to doing a router code update this morning |
05:18.39 | dijib | np |
05:19.07 | irroot | ah ok ouch that has to hurt thx Swk |
05:20.11 | SwK | irroot: yeah... the problem is once that started happening it started cascading... then you had route fapping and triggering flap hold downs heh |
05:20.16 | SwK | issues all over the place |
05:20.19 | andyoutside | what ISP do you like better than level3? |
05:20.31 | SwK | andyoutside: use more then 1 |
05:20.41 | SwK | we use Cogent, L3, and XO right now |
05:20.42 | andyoutside | nods |
05:21.03 | andyoutside | if we were to use a second one it would be cox |
05:21.33 | SwK | depends on where you colo or if you are trying to backhaul the connection to your site |
05:21.51 | SwK | if you are colo'd the choices are better and usually cheaper due to lack of backhaul costs |
05:22.06 | andyoutside | they are alwired and ready. We lease the land of one of their biggest if not biggest users |
05:22.24 | SwK | nice |
05:22.37 | SwK | yeah I colo in NYC and MIA |
05:22.41 | irroot | SwK we have only 2 International carriers and at most 4 providers so not much choice |
05:23.10 | andyoutside | where are you swk |
05:23.34 | SwK | andyoutside: I'm in MS ... used to live in Huntsville (down the street from digium almost quite litterally) |
05:24.22 | andyoutside | Some of the bigger ones you just have to call and ask how much for them to do it. |
05:24.31 | SwK | yeah |
05:27.40 | andyoutside | so who here uses snort while we are off topic |
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05:37.39 | SeRi | quite night... |
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06:30.52 | ChannelZ | farts loudly |
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06:39.14 | irroot | ChannelZ for a minute i thought it was the dog |
06:44.58 | SeRi | lol |
06:45.35 | SeRi | is reading The Book... Silence I KILL YOU! |
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06:59.22 | ChannelZ | http://failblog.org/2011/11/01/epic-fail-photos-family-fun-fail-3/ |
07:00.36 | SeRi | lol |
07:09.40 | autofsckk | night, anybody have used spa3102? a friend of mine told me that he have a lot of delay, jitter |
07:10.09 | autofsckk | i have read somethings about that spa3102 and it says it is not so a good idea to use it as an F |
07:10.18 | autofsckk | FXO sorries |
07:12.02 | ChannelZ | I have one |
07:12.59 | ChannelZ | I always had a problem with echo |
07:13.02 | autofsckk | ChannelZ: is it good? i have read bad things about it, is it really that bad= |
07:13.36 | ChannelZ | It has a LOT of settings |
07:13.48 | autofsckk | i've read that ther are some things you can try to fix it, but it cant be solved completly |
07:14.00 | ChannelZ | I'm using it as an FXS now, works great for that |
07:14.40 | autofsckk | i think its like a pap2 as a FXS, i have a pap2 and it works well |
07:15.07 | autofsckk | ChannelZ: what do you use now as FXO? |
07:16.02 | ChannelZ | well nothing.. I got rid of my home phone line and do it VOIP |
07:16.46 | ChannelZ | I have a TDM800 at work for FXO but a bit expensive for a home system :) |
07:17.05 | SeRi | guys is this worth the money? http://www.voiplink.com/Polycom_550_OB_p/polycom-550-ob.htm |
07:17.45 | andyoutside | hm |
07:17.50 | autofsckk | i have VOIP on my home too, im new to asterisk, today is the first day i could receive and make calls through my * box :D |
07:17.58 | andyoutside | lets see how much is the 650 |
07:18.46 | ChannelZ | What do you need/want FXO for then? |
07:19.21 | SeRi | autofsckk, cool. |
07:19.23 | autofsckk | not for me |
07:19.28 | ChannelZ | ah |
07:19.30 | autofsckk | for a friend of mine |
07:19.44 | autofsckk | SeRi: thanks |
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07:19.53 | SeRi | autofsckk, I have a PAPt2-NA and works grate. no echos no issues. |
07:20.54 | ChannelZ | But that's FXS |
07:20.56 | autofsckk | i have a pap2 too, and i used to connect that to my itsp, used to have a little delay, but now its connected to my asterisk box and it sounds better |
07:21.31 | andyoutside | I forget off hand what the difference is between the 550 and the 650 but I would go with the 650 for 20 more dollars |
07:21.35 | SeRi | well it seems like a good deal. cant find it for no less than 250 and i like the fact that it has a backlit lcd |
07:21.51 | SeRi | andyoutside, you think so? |
07:22.07 | autofsckk | ChannelZ: i need to connect to pstn with fxo because where i live, there are no available numbers, i mean, my itsp doesnt have tel numbers from here |
07:22.33 | ChannelZ | wow |
07:22.55 | SeRi | andyoutside, where do you se it for 20 more dollars? |
07:22.56 | andyoutside | SeRi: for the 20 if you are buying a few I would go for the 650 I want to say it has a few better things in it |
07:23.23 | autofsckk | i would like to buy a little usb sangoma FXO to connect it to my box |
07:23.36 | andyoutside | http://compare.ebay.com/like/120807144401?var=lv<yp=AllFixedPriceItemTypes&var=sbar&_lwgsi=y |
07:23.44 | andyoutside | http://www.google.com/products/catalog?q=polycom+650&hl=en&prmd=imvns&biw=1280&bih=655&um=1&ie=UTF-8&tbm=shop&cid=1354935552959657688&sa=X&ei=5Ni4TvHjDMG1tweeqpnGBw&ved=0CHwQ8wIwADg8 |
07:24.10 | ChannelZ | I don't really know of any other FXO that work well in the same price range |
07:25.01 | SeRi | andyoutside, thanks |
07:25.06 | andyoutside | np |
07:25.51 | autofsckk | i saw digium clones at an excellent price, having the same issues as digium cards heheheheh, i forgot the name of those cards |
07:26.31 | autofsckk | they also have the same irq problems as the original ones |
07:26.41 | ChannelZ | my TDM works great, once you tune it.. I'm not even using hardware echo cancellation |
07:26.55 | ChannelZ | you might be talking about the old single-port cards |
07:27.52 | autofsckk | thats what i was going to ask, what about the echo with those cards, is it really needed the echo cancellation card? or it depends on the land lines of your country? |
07:28.48 | autofsckk | ChannelZ: no im talking about the openvox cards, have you tried them? |
07:29.04 | ChannelZ | I suppose, and/or if you have not a very powerful computer to do software cancel |
07:29.15 | ChannelZ | no sorry |
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07:31.03 | autofsckk | for soft echo cancel what do you need the most? CPU? or RAM? or both? ha |
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07:33.21 | ChannelZ | CPU. It's not really a huge burden for small numbers of channels. |
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07:34.44 | ChannelZ | If you've got a big analog installation with one or more of those 24-port cards and high utilization, it's more worth it to get the HWEC module. |
07:34.48 | autofsckk | do atom CPU work well? |
07:35.22 | ollii | autofsckk: how many extensions? |
07:35.50 | andyoutside | for server or destop or embeded |
07:36.46 | autofsckk | ollii: 5 maybe, or how much extensions can handle with good quality? |
07:37.25 | SeRi | autofsckk, my alix handles up to 6 calls simultaneously with no issues... I have not tried more. |
07:37.35 | SeRi | Alix 2D3 |
07:37.46 | ollii | depends on your needs...5 extensions should be fine...transcoding could be not good enoug |
07:38.36 | ollii | with a generic atom we handle about 20 extensions with ~ 10 calls simultaneously and offering a php/mysql gui |
07:39.20 | SeRi | g/n all! |
07:39.36 | autofsckk | ollii: how much ram? |
07:39.40 | autofsckk | SeRi: good night |
07:40.06 | ollii | 1024 |
07:40.16 | ollii | ram should not be a problem |
07:40.51 | ollii | if you do some transcoding (switching codecs...maybe from g729 to g711) could bring your cpu in trouble |
07:40.55 | autofsckk | i think so, i have read about the capability of some old processors with so little ram doing good jobs |
07:42.15 | autofsckk | ive been reading for the last month or so about asterisk, i can understand a lot more now, and it rocks |
07:43.36 | ollii | but please secure it... |
07:43.47 | ollii | otherwise it will be a very expensive try ;) |
07:43.57 | autofsckk | hehehe yes i know |
07:44.54 | autofsckk | my voip is limited, i just get the amount of credit i put it, so isnt really very dangerous |
07:46.13 | autofsckk | but i have read about security too, i think is not so insecure |
07:47.30 | autofsckk | ollii: what linux distro do you use to put * on? |
07:47.49 | ollii | autofsckk: ubuntu server / centos |
07:48.01 | ollii | with asterisk from source and own patches |
07:49.26 | autofsckk | i installed centos on a vm and built ast from source too, but i dont use centos, so i had to look for some things i didnt know, it took me a lot of time |
07:50.15 | autofsckk | and i didnt like centos, it consumes a lot of ram i think |
07:50.50 | autofsckk | but i see that centos is like the distro often used |
07:51.14 | ollii | its very near to redhat |
07:51.20 | ollii | so it might be a good choice |
07:51.42 | ollii | but if you use * from source you could almost use what you want |
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07:52.22 | autofsckk | do you run ast as non-root user? |
07:53.11 | ollii | unfortunately no |
07:53.36 | ollii | we ran into some issues dont remember what they were |
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07:58.23 | autofsckk | well thanks for the help again everybody, good night |
07:58.33 | autofsckk | see you tomorrow |
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08:30.13 | ik_5 | hello |
08:31.01 | andyoutside | hello |
08:31.15 | andyoutside | please enter more data |
08:31.36 | ik_5 | when a bridged call is hangup, I'm loosing all of the set variables of both channels. Does anyone know on a way to keep this information and access it after hangup ? |
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08:35.27 | bulkorok | did you trie to send it to the h-extension? |
08:37.22 | bulkorok | ik_5: description can be found here: http://www.the-asterisk-book.com/unstable/besondere-extensions.html#h-extension |
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08:38.19 | *** join/#asterisk _N1x (~n1x@mail.orient-logic.com) |
08:39.07 | _N1x | hi guys , anybody help me about asterisk stress test? |
08:40.08 | ik_5 | bulkorok, i did try the h extension, but it DumpChan does not display my variables |
08:40.35 | ik_5 | bulkorok, I think this is because of the bridge cmd |
08:42.40 | bulkorok | ik_5 maybe you have to set the variables after bridge again!? just a guess.... |
08:43.19 | ik_5 | bulkorok, I'll try, thanks |
08:43.32 | bulkorok | :-) |
08:44.16 | singler | you could try using __variable while setting, this way variables will propagate to subchannells |
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08:45.14 | ollii | but is there any subchannel? |
08:45.31 | bulkorok | _N1x you can make stress-tests with sipp and a good dialplan. more info is here: http://www.voip-info.org/wiki/view/Asterisk+dimensioning |
08:45.37 | singler | he is using bridge, so I guess yes |
08:45.56 | ik_5 | singler, I'm already using __variable |
08:46.20 | singler | oh |
08:46.41 | bulkorok | yeah... the underscore is the correct option |
08:47.28 | _N1x | bulletrtr: yep i making right now , but i need advices and recommendations at advance level :) |
08:47.53 | _N1x | bulletrtr: sorry . |
08:48.02 | _N1x | bulkorok: |
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08:50.31 | *** join/#asterisk krotos (~d3v1l@87.13.66.170) |
08:50.37 | krotos | hi :) |
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08:58.40 | schmidts | good morning |
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08:59.15 | th0mz | hi |
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09:01.53 | krotos | i'm using ami for making a controll on active channel, and its duration |
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09:04.07 | krotos | if i need a list of active channel and then i want to hangup someone of this |
09:04.45 | krotos | which command i need to use for listing channels (using ami)? Sip show channels give me an incomplete name of channels for use channel request hangup SIP/ |
09:06.06 | kaldemar | krotos: CoreShowChannels |
09:06.56 | kaldemar | krotos: if that doesn't show complete channel names, use Command to execute "core show channels concise". |
09:09.59 | krotos | kaldemar: thankyou, concise is that i need |
09:10.00 | bulkorok | krotos you can use "core show channels verbose" too... you get some more information with this... |
09:18.27 | *** join/#asterisk magicrhesus (~magicrhes@aether.hipocoon.be) |
09:19.03 | *** join/#asterisk magicrhesus (~magicrhes@aether.hipocoon.be) |
09:34.06 | *** join/#asterisk tuxxie (~Ryan@rrcs-70-63-90-226.midsouth.biz.rr.com) |
09:34.32 | dym | Hey - im using a Primux 2S2M ISDN Controller - which is the best channel driver to use for faxing these days - chan_capi doesnt seem to be available in 1.8.X |
09:35.06 | WIMPy | Do you have any other driver than capi for that thing? |
09:35.18 | dym | Good question |
09:35.34 | dym | chan_capi compile seems to fail anyways |
09:35.40 | WIMPy | guesses no |
09:35.54 | WIMPy | Did you get the git version? |
09:36.11 | WIMPy | Or svn rather |
09:36.19 | irroot | dym i had it work on mISDN before had to hack the hfcmulti driver to recognise the id and i cant remember if i got the LED's right but it worked |
09:36.52 | WIMPy | It's a HFC thing? |
09:37.15 | irroot | if its the same chip WIMPy had a primux 2S it was a HFC chip indeed |
09:37.17 | *** join/#asterisk garymc (~chatzilla@host81-139-152-192.in-addr.btopenworld.com) |
09:37.25 | irroot | but will need to see the PCID |
09:38.03 | WIMPy | Patching in the ID is probably easier than to use capi. |
09:38.16 | dym | irroot: fuck |
09:38.16 | irroot | <PROTECTED> |
09:38.18 | irroot | <PROTECTED> |
09:38.25 | dym | hack the hfcmulti driver? :/ |
09:38.30 | dym | doesnt sound convenient |
09:38.45 | irroot | dym lol check if its the id above |
09:38.49 | irroot | lspci |
09:38.52 | dym | sec |
09:39.02 | WIMPy | Or check if it's a HFC chip at all. |
09:39.32 | irroot | WIMPy i have the primux card in my tree seen it only once but still in there |
09:39.49 | dym | doesnt show up - yet to install drivers. |
09:39.59 | dym | irroot: did you use the primux drivers from the website? |
09:40.02 | irroot | should see the device |
09:40.11 | irroot | nope dym |
09:40.24 | irroot | just had the card it was hfc so i hacked it |
09:41.49 | WIMPy | From the pictures, there's no HFC on that card. |
09:43.05 | dym | lspci: 01:00.0 ISDN controller: Lattice Semiconductor Corporation Device e236 |
09:43.40 | irroot | dym nope dym not same id |
09:43.47 | dym | well |
09:43.50 | dym | this is the "big" card |
09:44.09 | dym | 60 channels |
09:44.20 | dym | fuck - doing my head in.. |
09:44.23 | WIMPy | sees TI logos |
09:44.53 | irroot | dym that is 2XE1 ?? the card i had was 2XBRI |
09:45.45 | dym | Its the 2402 |
09:45.49 | dym | errr |
09:46.12 | WIMPy | Well, the name 2s2m is reather obvious. |
09:46.14 | dym | 2 ports. Whats XE Exactly? |
09:46.26 | WIMPy | XE? |
09:46.31 | dym | E1* |
09:46.50 | WIMPy | The line that carries the PRI. |
09:47.09 | dym | well, yes then its 2XE1 |
09:47.35 | *** join/#asterisk _N1x (~n1x@mail.orient-logic.com) |
09:47.38 | _N1x | guys how i can increase simultaneosly calls amount? |
09:47.42 | WIMPy | S2M is what youget behing the NT. |
09:48.30 | WIMPy | thinks keyboards are getting too complicated for him :-( |
09:48.43 | dym | the faq says install asterisk 1.4 - install chan_capi, install card driver |
09:48.48 | dym | outdatedmuch :( |
09:49.47 | dym | dahdi? |
09:50.06 | dym | i doubt chan_misdn would apply here |
09:50.09 | WIMPy | What do you dream about at night? |
09:50.13 | dym | :D |
09:50.21 | dym | so i actually have to use 1.4? |
09:50.23 | WIMPy | Most probably not. |
09:50.24 | dym | :( |
09:50.33 | kaldemar | _N1x: do something about the bottleneck that limits them. |
09:51.38 | WIMPy | No, it does not look like Linux has drivers for that card. |
09:51.50 | WIMPy | You will have to stick with what Gerdes supplies. |
09:52.30 | dym | Well, there is capi drivers for linux |
09:52.35 | dym | http://www.primuxisdn.de/primux/inhalt/download.htm |
09:52.42 | _N1x | kaldemar: what you mean?... |
09:53.19 | dym | WIMPy: if i use that driver with the card - what would be the equivalent in asterisk? chan_capi only? |
09:53.25 | dym | and therefore => 1.4 ? |
09:53.42 | WIMPy | Yes, chan_capi. |
09:53.53 | dym | okay. back to 1.4 then,. |
09:53.59 | WIMPy | But I told you yesterday that it works with 1.8, just not 10. |
09:54.17 | *** join/#asterisk beccara (~beccara@180.222.64.254) |
09:54.37 | dym | chan_capi works with 1.8? |
09:54.39 | kaldemar | _N1x: that you need to provide more information. that kind of a question has no answer. |
09:54.40 | dym | i cant seem to compile it |
09:55.04 | WIMPy | Did you use the svn version? |
09:55.38 | dym | the current website one |
09:55.54 | WIMPy | That's for 1.6. |
09:56.23 | dym | http://pastebin.com/zRF1GJu1 |
09:56.30 | _N1x | kaldemar: i making sipp stress test , and i cant making more than 200 sip channels |
09:56.35 | WIMPy | svn co svn://svn.chan-capi.org/chan-capi/trunk chan-capi-trunk |
09:56.52 | _N1x | and need to increase |
09:57.30 | kaldemar | _N1x: what prevents you from making more than 200? what happens when you try to make more? |
09:57.48 | _N1x | kaldemar: i 'll show you , wait a sec. (pastebin) |
09:58.08 | dym | WIMPy: ill try. |
09:58.43 | WIMPy | And I will ad a not to that effect. |
10:01.46 | _N1x | kaldemar: http://pastie.org/2830030 |
10:02.31 | kaldemar | _N1x: did you notice the "Try increasing max file descriptors with ulimit -n" part in you pastebin? |
10:02.35 | WIMPy | _N1x: "too many open files" |
10:03.07 | _N1x | kaldemar: i see this debug but , how to do this command? or where? |
10:03.11 | _N1x | in asterisk cli? |
10:03.11 | WIMPy | I know it's mean, but sometimes it helps to read. |
10:03.27 | kaldemar | _N1x: ulimit is a system command. "man ulimit" |
10:03.48 | _N1x | kaldemar: not in my debian |
10:03.54 | wdoekes2 | (man ulimit will get you the lib call, you want man bash) |
10:04.08 | _N1x | root@debian:/etc/asterisk# unlimit |
10:04.08 | _N1x | -bash: unlimit: command not found |
10:04.21 | kaldemar | _N1x: who said unlimit? ulimit. |
10:04.49 | _N1x | root@debian:/etc/asterisk# ulimit -n |
10:04.49 | _N1x | 1024 |
10:04.58 | *** join/#asterisk markusl (~markus@carbon.gonicus.de) |
10:05.07 | wdoekes2 | _N1x: adjust your init script to set ulimit -n 8192 just before starting the asterisk daemon |
10:05.18 | kaldemar | wdoekes2 was right, man bash or man sh will get you the help that has ulimit. |
10:07.34 | _N1x | hm go to test. |
10:08.40 | WIMPy | Ouch. Gerdes still advertise that they support 1TR6 even on the leaflet telling you about their new PCI-e cards. |
10:09.51 | _N1x | kaldemar: wdoekes2 working thanks guys :) |
10:10.41 | beccara | Is anyone able to tell me what the difference between local and remote bridging on RTP in calls is and is this now whats known as packet2packet bridging? |
10:11.41 | *** join/#asterisk mathi (~Matthew@78.129.48.220) |
10:16.26 | mathi | hi |
10:20.08 | ollii | beccara: mediastream over asterisk and mediastream directly between to peers |
10:20.54 | bulkorok | can anyone help with compiling errors of ptlib for t38modem!? |
10:21.04 | ollii | http://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite |
10:21.33 | beccara | ollii, cheers, so local bridge = rtp going via the asterisk server, remote bridge = rtp doing directly to the peer? If so does the local bridge mean the RTP is going via the asterisk core? I'm trying to get to a state where it's doing p2p bridging to reduce the load |
10:29.23 | *** join/#asterisk happylife (~happylife@212.92.145.7) |
10:38.02 | dym | WIMPy: still aroundß |
10:38.39 | dym | == Parsing '/etc/asterisk/capi.conf': == Found [Nov 8 11:37:25] WARNING[11944]: chan_capi.c:8273 cc_init_capi: CAPI not installed, chan_capi disabled! |
10:38.42 | dym | :/ |
10:39.10 | WIMPy | Dosn't look bad. |
10:39.15 | WIMPy | So get the capi going. |
10:39.24 | dym | i installed the cards capi driver |
10:39.43 | WIMPy | Did you start it? |
10:39.56 | dym | (: |
10:40.12 | WIMPy | Can't remember the sames. Look at what the capi utils package offfers. |
10:40.21 | WIMPy | capiinit or something. |
10:40.41 | WIMPy | It needs a conf file defining your interfaces. |
10:43.55 | dym | http://pastebin.com/d3uTBhvJ |
10:44.52 | WIMPy | Hmm. rmmod *capi* and try capiinit again? |
10:45.28 | dym | apparently doesnt exist. |
10:45.35 | dym | well - this is the oddish primux capi driver probably |
10:45.58 | *** join/#asterisk timahvo1 (~rogue@197.176.198.173) |
10:46.06 | WIMPy | But it moans about the main capi.ko, not the cards driver. |
10:46.31 | dym | this is the systems capi driver? |
10:46.41 | WIMPy | The framework. |
10:46.56 | WIMPy | What does modprobe capi give? |
10:47.08 | WIMPy | Anything useful in dmesg? |
10:48.11 | dym | i havent installed any |
10:48.12 | dym | FATAL: Error inserting capi (/lib/modules/2.6.32-5-amd64/kernel/drivers/isdn/capi/capi.ko): Device or resource busy |
10:48.27 | WIMPy | any what? |
10:49.11 | dym | disregard that |
10:49.28 | dym | http://pastebin.com/ayXmwzB9 |
10:49.51 | WIMPy | Looks like you're missing the devices. |
10:50.01 | _N1x | kaldemar: wdoekes2 guys http://pastie.org/2830204 |
10:50.06 | dym | as in /dev/CAPI ? |
10:50.10 | _N1x | what is this? |
10:50.20 | WIMPy | Or the device numbers are used by something else. |
10:50.29 | WIMPy | yes |
10:50.55 | dym | inexistant |
10:50.58 | dym | no such devices in /dev |
10:51.12 | WIMPy | grep 68 /proc/devices |
10:52.18 | dym | urgh |
10:52.20 | dym | http://pastebin.com/6kW1TmLJ |
10:52.22 | dym | that cant be good |
10:52.36 | kaldemar | _N1x: did you do what the output suggested? |
10:53.05 | WIMPy | Depends on the minors, but as it does mona, that looks rather unpleasant. |
10:53.20 | WIMPy | moan |
10:53.33 | _N1x | kaldemar: sorry? |
10:53.53 | dym | WIMPy: any idea on correcting this issue? |
10:54.20 | kaldemar | _N1x: read what it says to read? |
10:55.06 | WIMPy | Get rid of whatever claims 68 for sd or patch one of them to use another major. |
10:55.15 | WIMPy | Pretty uncool. |
10:55.33 | dym | sd must be a hd |
10:55.46 | _N1x | kaldemar: how i can show current calls? |
10:55.49 | _N1x | sip show channels? |
10:55.50 | dym | this means i'd have to hardcode another major into the driver? |
10:56.29 | WIMPy | If I look in to /proc/devices here, I've got tons of sd. All but "8" are unused. |
10:56.32 | kaldemar | _N1x: "core show channels" and "core show calls" |
10:57.33 | dym | this is really fucked up |
10:57.40 | WIMPy | yes |
10:58.08 | dym | WIMPy: also capiinit is not necessarily needed |
11:00.33 | irroot | dym www.unfuckitup.com :P |
11:00.44 | WIMPy | Looks like we have seriousely run out of device numbers. |
11:01.08 | dym | Okay, any idea how i could fix this issue? |
11:01.23 | dym | i r equipped with too basic linux knowledge for this. |
11:01.26 | irroot | dym not sure but enjoy that site |
11:01.49 | _N1x | kaldemar: when i making more than 2000 requests to asterisk from sipp , output is http://pastebin.com/mzuKQF1v |
11:01.56 | WIMPy | You will have to either modify sd or capi and recompile your kernel. |
11:02.04 | _N1x | what is this? |
11:02.21 | dym | omfg |
11:02.22 | WIMPy | Unless there's soe magic parameter for sd that tells it hwich/how many majors to claim. |
11:02.36 | _N1x | i see my cpu is not full loaded |
11:02.48 | dym | WIMPy: what about modding the major within the capi drivers code and recompiling that? |
11:02.48 | irroot | dym the problem with any system FOSS in particular if the manufacturer does not supply a solution that is taken up or does not supply the drivers then its hard to get anywhere |
11:03.03 | dym | irroot: so i have to install suse? :( |
11:03.08 | WIMPy | That is part of the kernel. |
11:03.11 | dym | dont say yes - ill cry |
11:03.22 | dym | it seems suse uses different major ranges then |
11:03.39 | irroot | dym no not at all unless suse has the bits you after but then you could get them from suse either way |
11:03.54 | dym | fuck |
11:04.07 | dym | so either kernel recompile with some modifications i have no idea about |
11:04.09 | irroot | use of udev ?? |
11:04.09 | dym | or changing to suse? |
11:04.16 | dym | irroot: care to explain? |
11:04.19 | dym | i know udev |
11:04.40 | irroot | if its using udev then this should be taken care of |
11:04.50 | irroot | not all systems use it properly |
11:05.06 | dym | well |
11:05.09 | dym | udev is installed |
11:05.13 | dym | can i enforce a major change? |
11:05.18 | WIMPy | Load capi before sd *eg* |
11:06.15 | WIMPy | It would be possible. |
11:06.51 | dym | mhhh |
11:07.01 | dym | from K01lcapiinit: |
11:07.03 | dym | <PROTECTED> |
11:08.25 | dym | WIMPy: couldnt i just change it there? |
11:09.03 | WIMPy | Yes, but your capi wouldn't know to dind to it. |
11:09.06 | WIMPy | bind |
11:09.40 | dym | excuse me? if i changed it there, things would stop working? |
11:10.16 | WIMPy | CONFIG_SD_EXTRA_DEVS |
11:10.44 | WIMPy | Soory. That's obsolete. |
11:11.35 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
11:12.46 | dym | WIMPy: Save my day! |
11:12.47 | dym | :( |
11:13.08 | WIMPy | Are you using an initrd? |
11:15.16 | dym | This is Debian Squeeze Disk Install with grub2 |
11:15.23 | dym | but i guess so. |
11:15.50 | WIMPy | Then try to load capi before you load sd. |
11:15.55 | kaldemar | _N1x: http://svn.digium.com/svn/asterisk/tags/1.8.0/doc/sip-retransmit.txt |
11:16.11 | WIMPy | Maybe that's finally a use for an initrd :-) |
11:17.02 | dym | WIMPy: where would this be done? |
11:17.25 | WIMPy | Somewhere in your init scripts. |
11:17.25 | *** join/#asterisk JuanCri (~JuanCri@pc-205-210-86-200.cm.vtr.net) |
11:17.37 | WIMPy | But I have NFI what they look like on Debian. |
11:19.35 | _N1x | kaldemar: i already see it , but how to resolve? |
11:20.57 | dym | WIMPy: http://pastebin.com/yxvfwSsX |
11:21.08 | dym | im not sure where the sd deviced are initialized. |
11:21.37 | WIMPy | There will be a lot of modprobing somewhere. |
11:22.49 | WIMPy | I whould think there must be a parameter to sd, but I cant find any. |
11:24.56 | kaldemar | _N1x: maybe your HW hit its limits. buy a beefier box. |
11:25.37 | dym | mhhh |
11:26.25 | *** join/#asterisk mathi (~Matthew@78.129.48.220) |
11:26.30 | mathi | hi |
11:27.11 | carrar | HARRO |
11:27.28 | dym | WIMPy: if i changed the major in the capi init, would that disturb some other software cause of it addressing capi by its major? |
11:28.13 | carrar | GOODBYE Belgium |
11:28.38 | WIMPy | The device inodes need to match the ones the kernel drivers bind to. |
11:29.15 | dym | and on compile of the capi driver, it selected whatever the kernel needs |
11:29.42 | WIMPy | The kernel capi, yes. |
11:29.47 | *** join/#asterisk mathi (~Matthew@78.129.48.220) |
11:30.00 | mathi | hi |
11:30.11 | dym | fudge |
11:30.15 | carrar | HARRO |
11:30.27 | dym | carrar: shut up. |
11:30.30 | dym | we've seen you |
11:30.30 | mathi | why does Asterisk output WAV files and not for ex. MP3 files which takes less space on disk ? |
11:30.32 | WIMPy | mathi: Jumping for joy today? *eg* |
11:30.40 | carrar | It's dark here |
11:30.43 | dym | mathi: cause you told it to. |
11:30.43 | carrar | You can't see me |
11:30.49 | mathi | WIMPy, hi, it's "dokg". this is my other nick |
11:31.11 | mathi | dym, I told it to ? |
11:32.40 | carrar | mathi, which output are you talking about? |
11:32.45 | carrar | voicemail? |
11:33.16 | *** join/#asterisk coppice (~chatzilla@m121-202-67-187.smartone-vodafone.com) |
11:34.15 | *** join/#asterisk Nasga (~Nasga@186.212.10.93.rev.sfr.net) |
11:34.20 | dym | mathi: course you did. |
11:34.35 | dym | WIMPy: any idea on how to proceed now? |
11:34.59 | mathi | no, actually in my IVR he could leave a message if he press some menu (I don't need to record the whole call). Then I record this message in an audio file, and I have to send it to a remote server (I am restricted to FTP). That's why I would like the best quality/size compromise. WAV files are quite huge it seems to me, thus sending it by FTP may take some time, plus huge disk space on the remote server. |
11:35.07 | WIMPy | No better idea that the module loading order or patching so far. |
11:35.25 | carrar | defaulted to that |
11:35.28 | carrar | change it |
11:35.30 | carrar | read the docs |
11:35.44 | carrar | core show application Record |
11:35.46 | mathi | carrar, me? |
11:35.59 | carrar | See anyone else asking that question? |
11:36.12 | mathi | carrar, but what format do you suggest in my case ? |
11:36.24 | carrar | I'd just go with what you have personally |
11:36.33 | carrar | but you don't seem to like it |
11:36.50 | mathi | I have nothing now, I am just studying the case) |
11:36.52 | carrar | You sox to change it to a mp3 |
11:36.58 | mathi | sox ? |
11:36.59 | carrar | or use ogg or flac |
11:37.04 | carrar | man sox |
11:37.12 | mathi | I can only use mp3, wav, or ogg |
11:37.13 | carrar | you=use |
11:37.24 | carrar | Sounds great then |
11:37.33 | carrar | convert it to one of those |
11:38.06 | mathi | carrar, but look I'm confused that in the reference guide we talk so much about WAV, as I could use MP3 and use less space seems to me |
11:39.07 | WIMPy | mathi: If it's really about size, save as wav and convert to mp3 or ogg with a fine-tuned set of filters. They make quite some impact for voice. |
11:39.36 | mathi | WIMPy, like which converter ? |
11:39.56 | WIMPy | The one you like. |
11:40.41 | mathi | WIMPy, why would I need a "fine-tuned set of filters" ? |
11:40.53 | WIMPy | What does the following line mean? |
11:40.55 | WIMPy | ERROR[32019]: lock.c:258 __ast_pthread_mutex_lock: chan_sip.c line 14833 (register_verify): 'peer' really deep reentrancy! |
11:41.16 | dym | WIMPy: kinky ;) |
11:41.36 | WIMPy | mathi: Because you can get much better size/quality ratio. |
11:41.53 | mathi | WIMPy, are you using one ? |
11:42.06 | carrar | mathi |
11:42.09 | carrar | MAN SOX |
11:42.17 | WIMPy | I experimentd with it many years ago. |
11:42.18 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
11:42.32 | carrar | record it as a wav, then on the next line use sox t convert it to mp3 |
11:42.42 | carrar | boom |
11:42.45 | carrar | your done |
11:42.46 | WIMPy | mp3enc, lame, oggenc |
11:43.11 | carrar | or one of those too |
11:43.14 | carrar | YOU PICK |
11:43.17 | carrar | :) |
11:43.32 | mathi | carrar, omg I thought you were talking about a man and his socks, I had no idea what ou were talking about |
11:43.32 | dym | carrar: require chill pills? |
11:43.36 | carrar | You ahve a PLETHORA of options! |
11:43.38 | WIMPy | And keep in mind, that you need a licence for mp3. |
11:43.50 | carrar | UNIX command line "man sox" |
11:43.56 | carrar | wihtout the quotes :) |
11:44.10 | carrar | if you have sox installed |
11:44.16 | carrar | which you probably don't |
11:44.17 | carrar | heh |
11:44.35 | carrar | yes you also need a license to be on the INTERNET |
11:44.42 | carrar | please quire |
11:44.44 | carrar | aquire |
11:44.44 | mathi | uhm? |
11:44.51 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
11:44.59 | dym | carrar: do you always photograph yourself in shorts? |
11:45.02 | mathi | WIMPy, licence to convert or to use MP3 ? |
11:45.04 | carrar | ALWAYS |
11:45.10 | carrar | cause thats all I wear |
11:45.15 | WIMPy | mathi: Both. |
11:45.23 | carrar | in fact |
11:45.27 | carrar | I have em on right now |
11:45.30 | dym | nice |
11:45.42 | mathi | WIMPy, and if I don't buy the licence ? |
11:45.44 | carrar | I just had real sushi with my shorts on |
11:45.51 | WIMPy | carrar: How many? 2 or 3? ecnr |
11:45.53 | WIMPy | scnr |
11:46.04 | carrar | scnr? |
11:46.07 | carrar | plates? |
11:46.17 | carrar | I had 13 plates |
11:46.22 | WIMPy | mathi: You're an evil thief. |
11:46.39 | carrar | mathi, probably nothing |
11:46.49 | carrar | so use OGG |
11:46.59 | mathi | WIMPy, where do I buy it? |
11:47.08 | carrar | heh |
11:47.31 | WIMPy | Fraunhofer Institure |
11:47.42 | carrar | That is a good question by the way |
11:47.43 | WIMPy | Or use ogg/vorbis |
11:48.27 | mathi | I wonder how many users use MP3 without licence ? millions ? |
11:48.40 | carrar | 7 billion |
11:48.46 | WIMPy | Playback is free for private use. |
11:49.22 | carrar | use FLAC |
11:49.27 | carrar | it's free |
11:49.55 | mathi | carrar, I can't, I need <audio> in HTML5 for playback, I am limited to ogg, mp3 and wav for chrome |
11:50.11 | mathi | it's a private web app, users use chrome |
11:50.13 | carrar | ogg |
11:50.14 | WIMPy | Then use ogg. |
11:50.18 | mathi | ok:) |
11:52.03 | carrar | Any other questions? |
11:54.29 | WIMPy | Yes, what are next weeks lottery numbers? |
11:54.49 | carrar | 23 44 18 12 88 44 -- 42 |
11:55.07 | WIMPy | Out of range |
11:55.29 | WIMPy | And a dupe |
11:55.40 | carrar | You chances of winning could be lower |
11:55.41 | WIMPy | Now I can't trust you any more. |
11:55.59 | carrar | probably as a result of never playing lottery |
11:57.10 | WIMPy | I usually try every few months, but this time it must have been more than half a year ago. |
12:00.28 | mathi | carrar, yes about speech recognition in french, in my IVR I would like people to pronounce a date, without typing on phone. typing a date is not that easy |
12:01.19 | carrar | SIRI |
12:03.29 | cusco | hello folks |
12:04.31 | cusco | question: can I make cdr log an extra variable without the need to use Set(CDR(myVar)=${myVar}); in every dialplan that it is on? |
12:04.47 | carrar | mathi, try google? "speech recognition asterisk french" |
12:04.54 | cusco | can I somehow bind a var to cdr everytimt that it is set? |
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12:21.23 | patrickod_ | I'm tying to use SayAlpha in a test extension yet when I dial that extension I get no audio |
12:21.28 | mathi | carrar, there are many alternatives, anything to suggest ? |
12:21.35 | patrickod_ | the asterisk CLI shows it playing the required letters without error |
12:21.43 | patrickod_ | any ideas as to what might be causing the silence ? |
12:22.37 | singler | did you answer the call? |
12:23.08 | patrickod_ | yep I just figured that out. thanks :) |
12:23.15 | singler | np :) |
12:23.53 | carrar | mathi, I don't use speech recognition |
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12:29.19 | puzzled | hi |
12:29.40 | leifmadsen | ohai |
12:29.55 | carrar | konnbanwa! |
12:31.05 | ixyd_ | hi guys, is it possible, to configure CEL in a way that it works in some kind of a batch mode like CDR? |
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12:39.59 | leifmadsen | ixyd_: I don't think that's the point of CEL though |
12:50.22 | ixyd_ | i dont see why using a batchmode with CEL would be more strange than it is with CDR? |
12:53.38 | dym | greets leifmadsen |
12:55.12 | leifmadsen | dym: ohai |
12:55.30 | leifmadsen | ixyd_: oh wait, I think I misunderstood what you meant |
12:55.48 | ixyd_ | ah :) |
12:56.45 | leifmadsen | ixyd_: and yes I understand what you mean now, I just did a search in the conf files, and it doesn't appear to exist (batch mode) |
12:57.12 | leifmadsen | and I searched all the CEL .c code for batch, and no hits there either |
12:57.38 | ixyd_ | ah ok thank you very much! :) |
13:01.01 | leifmadsen | np! |
13:01.15 | leifmadsen | ixyd_: would be a good feature I agree, I wonder how hard it would be to implement... |
13:02.02 | ixyd_ | would be very nice to have definitely! |
13:02.23 | leifmadsen | just for fun I'm going to see if I can find the batch code in CDR and see how much there is of it |
13:02.32 | leifmadsen | if you wanted to implement it that's where I'd start |
13:02.35 | leifmadsen | (I'm also not a programmer) |
13:03.43 | ixyd_ | iam not a real programmer to, i have to wait for the request of the customer...maybe you'll get a patch from us later ;) |
13:06.05 | leifmadsen | ixyd_: ya, so you want to look in main/cdr.c and add the code to main/cel.c for batch mode. There is a lot of code, but none of it looks that complicated. Most of cdr.c seems to actually be batch code. I'd start there and see if you can make cel.c contain batch mode as well |
13:06.18 | leifmadsen | ixyd_: good luck! ping me if you end up filing an issue with a patch |
13:06.33 | ixyd_ | i will do so! thank you leif! :) |
13:06.40 | leifmadsen | np |
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13:26.47 | ollii | is their a list where all soundfiles for * 1.8 are named with its content? |
13:28.17 | p3nguin | Yes. Look at the .txt files in the sounds directory. |
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13:29.12 | ollii | that is to easy... :X |
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13:29.26 | ollii | thank you.. ;-) |
13:29.32 | p3nguin | You should have the core-sounds and the extra-sounds text files. |
13:30.15 | ppeejjaayy | Hi everyone, has anyone tried DYNAMIC_FEATURES in Asterisk 1.4.36 ? i'm trying to use the application Goto as DYNAMIC_FEATURE |
13:30.29 | ollii | p3nguin: yeah found it...that was way to easy |
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13:33.39 | ppeejjaayy | while in a call i need to redirect the customer to an IVR, so i'm trying the DYNAMIC_FEATURES as a solution. I was able to launch a macro. In the macro I can't do waitExten, so i need to use the Goto app directly from the feature |
13:34.11 | *** join/#asterisk mandla (~mandla@168.167.180.161) |
13:34.48 | ppeejjaayy | featureTest => *77, Goto,myContext|myExtension|1 |
13:35.46 | mandla | irroot, can you help me identify a line in my extension.conf where should add ,,t |
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13:36.28 | irroot | mandla its the line with a "dial" to the extension you looking for |
13:36.44 | irroot | the problem with your extensions.conf is there many comments |
13:37.10 | kaldemar | ppeejjaayy: looks like your feature is missing the <ActivateOn>[/<ActivatedBy>] part entirely. |
13:37.13 | mandla | irroot, man there are many lines with dial. |
13:37.26 | irroot | yes the one with the extension number |
13:37.32 | mandla | irroot, ok, let me look it up. |
13:37.34 | ppeejjaayy | featureTest => *77, peer,Goto,myContext|myExtension|1 |
13:37.38 | leifmadsen | ppeejjaayy: try GoSub |
13:37.44 | ppeejjaayy | sorry i forgot the peer |
13:38.00 | leifmadsen | ppeejjaayy: nevermind, I just read you're using 1.4 |
13:38.18 | ppeejjaayy | in fact the the feature is working but the goto isn't |
13:38.34 | ppeejjaayy | i can do launch playbacks, macros, but no goto |
13:38.37 | p3nguin | I think I would have just used my transfer key. |
13:38.50 | ppeejjaayy | what's the problem with * 1.4 ? |
13:38.55 | p3nguin | Or if I didn't have one, I'd configure DTMF transfers. |
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13:39.31 | timeshell | what's the command to kill a sip channel in CLI? |
13:39.51 | leifmadsen | channel request hangup |
13:40.26 | kaldemar | ppeejjaayy: your problem is most likely with more than one argument. try featureTest => *77,peer,Goto,"myContext|myExtension|1" or featureTest => *77,peer,Goto(myContext|myExtension|1), that's how its documented in newer versions. |
13:40.57 | p3nguin | It seems like if you want to transfer a call to some arbitrary extension, the transfer key would be a good way to accomplish it. |
13:41.03 | timeshell | ty |
13:41.46 | leifmadsen | kaldemar: in 1.4 I'm pretty sure arguments are separated with commas like he is doing |
13:41.52 | leifmadsen | newer versions are more sane |
13:42.05 | ppeejjaayy | kaldemar: i'll try it right away, i thought the syntax is : AppName,Argument1|Argument2 |
13:42.08 | leifmadsen | although I haven't used 1.4 in years |
13:42.15 | mandla | irroot, the only line with the extension is exten = _X.,1,Goto(default,917,1) |
13:42.17 | leifmadsen | ppeejjaayy: look at the sample file, and follow that |
13:42.36 | mathi | why are software for asterisk usually sold per port/channel ? |
13:42.53 | [TK]D-Fender | <ppeejjaayy> while in a call i need to redirect the customer to an IVR, so i'm trying the DYNAMIC_FEATURES as a solution. I was able to launch a macro. In the macro I can't do waitExten, so i need to use the Goto app directly from the feature <--- wrong idea. |
13:42.58 | irroot | cool mandla now look in default for a priority 1 that matches 917 |
13:43.00 | kaldemar | leifmadsen: 1.4 has the MOH class option also. depends on how the parser handles pipes. |
13:43.01 | [TK]D-Fender | ppeejjaayy, Just transfer the call. |
13:43.15 | irroot | it could be exten => _XXX,1,...... |
13:43.18 | [TK]D-Fender | ppeejjaayy, Dynamic features are for in-line little bits not "transfer the call" |
13:43.40 | leifmadsen | oh ya, if the purpose is to transfer the call... just do that. There is built-in transfers or you can use the transfers from the phone. |
13:45.12 | leifmadsen | ppeejjaayy: oh, and this: http://pastebin.com/4NqZi8pR |
13:45.21 | [TK]D-Fender | ppeejjaayy, Multiple failures : The call stays bridged with the dynamic feature. Also your IVR will end up using waitexten which will fail regardless. |
13:45.43 | ppeejjaayy | during a call i want to launch an IVR to get Credit Card numbers by DTMF, so i thought using the dynamic_feature to a context where i'll put playbacks and waitExten |
13:46.05 | mandla | irroot, all entries in [default] are commented out, lol. |
13:46.20 | irroot | mandla is there an include ?? |
13:46.29 | irroot | look in the included section |
13:46.37 | ppeejjaayy | leifmadsen, thx for the link |
13:46.53 | leifmadsen | ppeejjaayy: thats from features.conf.sample fyi |
13:47.02 | mandla | irroot, in my dialplan?? |
13:47.23 | mandla | irroot, or in default?/ |
13:47.37 | irroot | yes in extensions.conf either a file is included or a context is included in [default] |
13:48.51 | mandla | irroot, true, it includes demo, but its commented by asterisk gui. |
13:48.57 | leifmadsen | yuck [default] |
13:49.34 | ppeejjaayy | leifmadsen: indeed, i haven't read that part. so in my case what should i do ? transfer the call ? |
13:49.43 | [TK]D-Fender | ppeejjaayy, Yes |
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13:50.18 | ppeejjaayy | and when the customer finishes filling the data, retransfer to the original channel |
13:50.33 | timeshell | Is there a way in CLI to tell asterisk to stop trying to register a trunk? |
13:50.34 | [TK]D-Fender | ppeejjaayy, Sure |
13:50.38 | timeshell | Without removing the trunk? |
13:50.56 | [TK]D-Fender | timeshell, nope |
13:51.09 | timeshell | Well, there's a feature request. |
13:51.11 | irroot | mandla is it a file include or a context include ? |
13:51.14 | ppeejjaayy | D-Fender, thx |
13:51.35 | irroot | mandla looking at the calltrace it ends up in default so should be there |
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13:54.03 | mandla | irroot, its a context include, include=demo |
13:54.20 | irroot | mandla now find it in there .... |
13:54.27 | irroot | look for [demo] |
13:54.38 | mandla | irroot, in demo, right |
13:54.43 | mandla | irroot, thanx |
13:55.54 | mandla | irroot, t should be the 3rd entry right?? |
13:56.57 | irroot | mandla what you mean by 3 ? its Dial(<CHAN>,<TIMEOUT>,<OPTIONS>) so the 3 opt yes |
13:57.34 | ppeejjaayy | D-Fender, how do i trigger the transfer ? |
13:57.40 | mandla | irroot, thats what i meant. |
13:58.19 | ppeejjaayy | can it be done from the dynamic features ? |
13:58.28 | irroot | mandla when you done do "dialplan reload" |
13:58.50 | irroot | and then maybe "dialplan show <NUM>@default" |
13:59.00 | [TK]D-Fender | ppeejjaayy, No... transfer is one of the most boring basic features of your phone |
13:59.15 | [TK]D-Fender | ppeejjaayy, What are yuo using? |
13:59.24 | irroot | sorry no @ |
14:00.06 | irroot | with it works i typo here |
14:00.11 | ppeejjaayy | as softphone ? a homemade |
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14:01.35 | [TK]D-Fender | ppeejjaayy, If you did it yourself.. you didn't make a SIP transfer feature? |
14:01.48 | [TK]D-Fender | ppeejjaayy, If not use the dial options for transfers via DTMF |
14:01.54 | [TK]D-Fender | ppeejjaayy, Thsi is not "dynamic features" |
14:06.54 | WIMPy | What does this mean or what should I watch out for? - ERROR[32019]: lock.c:258 __ast_pthread_mutex_lock: chan_sip.c line 14833 (register_verify): 'peer' really deep reentrancy! |
14:09.42 | [TK]D-Fender | WIMPy, Dunno ... sounds kinky ;) |
14:09.53 | *** join/#asterisk master_of_master (~master_of@p57B54F47.dip.t-dialin.net) |
14:10.03 | WIMPy | It says ERROR. That's usually not a good thing. |
14:10.56 | irroot | WIMPy lol its a lock on the same resource in a loop without unlock ?? |
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14:12.14 | WIMPy | Hmm. I have 3 locks on sip_send_mwi_to_peer. But I got quite some of the above messages. |
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14:20.07 | hudony | Hi : sorry to ask it here but cant find a final answer : using cisco spa-303 and a poe switch. The phone doesn't use poe from ethernet, I still have to use ac adapter. When googling, I found that an adapter seems to be required to opereate POE on the phone. IS that right? |
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14:22.33 | [TK]D-Fender | hudony, What switch? |
14:23.00 | hudony | cisco sf300 24 ports |
14:23.12 | hudony | managed switch and I have enable poe |
14:23.32 | hudony | all ports have it enabled but I can see 0 consumed mW |
14:23.38 | mandla | irroot, still not working. |
14:23.51 | irroot | mandla is it set |
14:24.01 | mandla | irroot, still doing that native bridging thing. |
14:24.08 | irroot | and when you dial you see it in the output |
14:24.17 | mandla | irroot, yah i think its set. |
14:24.24 | kaldemar | hudony: why did you think that the spa-303 would use PoE in the first place? |
14:25.05 | hudony | ah |
14:25.08 | hudony | dman |
14:25.12 | [TK]D-Fender | http://www.cisco.com/en/US/prod/collateral/voicesw/ps6788/phones/ps10998/data_sheet_c78-601648.html |
14:25.15 | hudony | just noticed that not |
14:25.31 | [TK]D-Fender | Power over Ethernet (PoE) Support SPA 303G No |
14:25.33 | hudony | guess ive been confused by the possibility to add it through the optional adapter |
14:25.36 | hudony | my bad |
14:27.08 | hudony | thx anyway guys |
14:27.16 | hudony | Thx god only ordered 2 of these |
14:27.21 | hudony | :S |
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14:30.52 | [TK]D-Fender | hudony, You really really should read the specs on products you order... |
14:31.03 | hudony | I can tell :( |
14:31.36 | [TK]D-Fender | hudony, God-aweful expensive switch ... to power cheap-ass phones with no PoE. Prioritize man! |
14:31.50 | leifmadsen | I really like the D-Link switches with PoE in them |
14:32.19 | hudony | Well.. the guy here only wanted cisco switch so we ordered that one for POE |
14:32.31 | leifmadsen | powers Polycom phones quite well. The problem is that (and this seems to be an issue with all PoE switches) is that if you order a 24 port switch, don't expect to have enough power load in the switch to power 24 PoE phones |
14:32.36 | hudony | But the phone choice is a total fuck up |
14:32.38 | hudony | hehe |
14:33.02 | leifmadsen | when you do the math, you can do about 8-10 phones at full power draw (I assume full draw happens at boot up) |
14:33.30 | hudony | oh |
14:33.32 | hudony | ok |
14:33.35 | hudony | we have 16 phones |
14:33.51 | hudony | so I guess some will have to use ac |
14:34.22 | Naikrovek | test it before you make that assumption |
14:34.36 | Naikrovek | the phones actually draw very little once they're booted up. |
14:34.57 | [TK]D-Fender | Naikrovek, The spec sheet says "don't bother" |
14:34.57 | leifmadsen | right |
14:35.10 | [TK]D-Fender | "<[TK]D-Fender> Power over Ethernet (PoE) Support SPA 303G No" |
14:35.22 | leifmadsen | the only issue would be if you had all 24 phones boot up at the same time (like after a power outage) |
14:35.37 | hudony | oh ok |
14:35.46 | hudony | Found 504g and was looking for his power consumption |
14:35.48 | leifmadsen | wonder if Polycoms have a staggered boot sequence you could enable, then that'd not be a problem |
14:36.20 | Naikrovek | the switches stagger by a few milliseconds in my experience |
14:36.26 | WIMPy | I guess you'd need that feature in the switch. |
14:36.28 | Naikrovek | they don't all offer full power at once |
14:36.40 | Naikrovek | as the ports are brought online they are powered |
14:36.55 | coppice | PoE switches have considerable complexity in them to avoid trouble like that |
14:36.57 | leifmadsen | Naikrovek: right, but what happens when all the phones request power at the same time |
14:37.06 | Naikrovek | they have to be powered on to request more power |
14:37.20 | leifmadsen | huh, learn something new every day |
14:37.21 | Naikrovek | the switch doesn't supply power to all ports at once when the power comes back on |
14:37.37 | leifmadsen | I've never actually tested all that to see what happens, so that'd be an interesting experiment |
14:37.46 | Naikrovek | yeah |
14:38.11 | leifmadsen | I applied theory to the solution, and played it safe in a couple of installs. Now I want to try plugging all the phones in and power them at once and see what happens :D |
14:38.13 | coppice | its all laid down in the PoE spec |
14:38.13 | [TK]D-Fender | My D-Link DES-1526's stagger the power-up. |
14:38.14 | Naikrovek | they don't actually "request" power, it's a simple circuit that measures resistance that determines the amount of power to send down the wire |
14:38.28 | leifmadsen | coppice: I've never read the PoE spec |
14:38.40 | Naikrovek | only on a 300ft run will a polycom phone need the full 15.4w or whatever it is. |
14:38.44 | leifmadsen | [TK]D-Fender: cool, I think those are the same ones I'm using |
14:38.49 | Naikrovek | and most of that will be due to losses in the cable |
14:38.57 | coppice | it defines an elaborate procedure to managing the loads |
14:39.07 | Naikrovek | put in 22gauge instead of 24 and you'll use less power. |
14:39.17 | Naikrovek | (ethernet cable, i mean) |
14:39.35 | leifmadsen | Naikrovek: I only use audio cable for my ethernet deployments |
14:39.41 | Naikrovek | lol |
14:39.56 | leifmadsen | 16 gauge bwaaaaaa! |
14:40.19 | Naikrovek | i've found that almost everything that people communally assume is true, is actually false. i test everything i can anymore, just because i almost always learn something and i've always been a serial contrarian. |
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14:44.04 | Naikrovek | it comes with age, i think. I used to believe just about anything that sounded reasonable. now i realize that a whole lot of that was 100% fiction. |
14:44.42 | Naikrovek | so now i don't believe anything, and whenever someone says something with conviction, I always just presume they're wrong. Turns out that's the approach that most reflects reality. |
14:45.14 | WIMPy | I only believe what I see. |
14:45.28 | WIMPy | Since the invention of TV I believe everything. |
14:46.29 | [TK]D-Fender | "believe half of what you see and nothing that you hear" |
14:46.32 | irroot | Naikrovek WIMPy according to the dictionary a pessimist is a experianced optomist |
14:46.55 | [TK]D-Fender | irroot, No, a realist is an experienced pessimist :) |
14:48.12 | WIMPy | The pessimist calls a realist an optimist, while the optimist calls a realis a pessimist. |
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14:51.44 | Naikrovek | everyone seems to call me dorkwad lately. |
14:51.53 | Naikrovek | s/dorkwad/"dorkwad"/ |
14:52.08 | Naikrovek | they hate it when i'm right all the time and resort to calling me names |
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14:52.54 | Naikrovek | "install that phone, dorkwad," and "I'm late for pottery class, dorkwad," etc. |
14:53.21 | leifmadsen | Naikrovek: "Poor planning on your part does not constitute and emergency on mine" <-- best cubicle sign ever. |
14:53.33 | leifmadsen | efff.... I continually type 'and' when I mean 'an' |
14:53.42 | leifmadsen | stupid auto pilot typing |
14:54.07 | Naikrovek | yeah that is an awesome weapon against stupidity |
14:54.07 | p3nguin | Cisco PoE actually does request more or less power, and it is done via cdp. |
14:54.41 | WIMPy | You mean "inline power"? |
14:54.55 | [TK]D-Fender | <WIMPy> The pessimist calls a realist an optimist, while the optimist calls a realis a pessimist. <- Sounds about right :) |
14:55.46 | garymc | I get the error : retrieve_conf failed, config not applied |
14:55.52 | garymc | asterisk wont reload |
14:56.04 | garymc | I must have broken something and i cant get it to work |
14:56.06 | *** join/#asterisk treborsux (~treborsux@75-144-117-117-Jacksonville.hfc.comcastbusiness.net) |
14:56.09 | [TK]D-Fender | garymc, Not an Asterisk problem.... |
14:56.24 | treborsux | linux problem? |
14:56.28 | [TK]D-Fender | FREEPBX |
14:56.29 | *** join/#asterisk nuxinc (~nux@87.120.139.81) |
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15:00.14 | p3nguin | No, I do not mean inline power. I mean power. Cisco CDP allows a device to say, "Hey, I don't need this much power; please turn down your available power." |
15:00.38 | Naikrovek | except it doesn't actually turn down power, it just changes the power reported as being used |
15:01.06 | Naikrovek | current is drawn, not forced. volts are forced, but poe is a standard voltage always |
15:01.38 | Naikrovek | that CDP negotiation tells the switch how much is actually being used, which tells the switch how much capacity it has left |
15:02.18 | Naikrovek | and it uses that capacity to continue to power phones until capacity gets to zero |
15:02.29 | Naikrovek | at which point the next phone won't turn on. the port will appear dead. |
15:03.58 | Naikrovek | but yes, your general point is correct. cdp does negotiate |
15:04.06 | p3nguin | Yes it does. |
15:04.35 | p3nguin | And it can negotiate a lower or a higher available power amount. |
15:04.40 | p3nguin | Which is what I said before. |
15:17.21 | *** join/#asterisk endemic (~endemic@2001:49f0:400e::5) |
15:21.48 | SeRi | p3nguin, do you create xml files in your nix box? |
15:22.06 | p3nguin | Uh, sure, I guess. |
15:22.18 | SeRi | do you use any specific tool? |
15:22.22 | p3nguin | vim |
15:22.38 | SeRi | vim sets the encoding for xml? |
15:22.49 | p3nguin | Encoding? It's just text. |
15:22.53 | SeRi | I thought xml had some type of encoding |
15:22.56 | SeRi | oo ok |
15:23.01 | SeRi | well burn me :) |
15:23.04 | SeRi | lol |
15:23.05 | SeRi | Thanks |
15:26.01 | *** join/#asterisk endemic (~endemic@2001:49f0:400e::5) |
15:26.54 | [TK]D-Fender | SeRi, "Build a fire for a man and he's warm for a day. Light a man on fire and he's warm for the rest of his life." - Terry Pratchett |
15:27.28 | SeRi | [TK]D-Fender, very nice. thank you :) |
15:34.35 | SeRi | [TK]D-Fender, would this be a valid polycom softkey assignment? http://pastebin.com/raw.php?i=9CrUje1z |
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15:36.58 | lal00 | is it possible to load a codec without restarting asterisk? |
15:37.07 | p3nguin | Yes. |
15:37.08 | [TK]D-Fender | SeRi, Haven't touched the EFK stuff yet, though I definitely should start |
15:37.20 | p3nguin | module load codec_ulaw.so for example. |
15:38.28 | SeRi | [TK]D-Fender, I am just going to try and see what heppens :P Do you know how I can disable the default "Forward" label in the screen? |
15:39.08 | [TK]D-Fender | SeRi, Not off-hand... |
15:39.24 | SeRi | Thanks. Back to experimenting :) |
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15:40.05 | lal00 | p3nguin: thanks |
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16:19.21 | hardwire | dislike nickserv sometimes |
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16:23.06 | SeRi | I give up |
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16:23.08 | SeRi | lol |
16:23.10 | SeRi | Configuration file "softkey.cfg" is from template Unknown, revision Unknown |
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16:24.11 | LemensTS | anyone here use phpagi swift function |
16:24.17 | in0cula | hi, i'm on fedora, wich softphone i need to use for testing asterisknow? |
16:28.43 | SeRi | in0cula, you could use any softphone |
16:28.51 | SeRi | a popular one is ekiga |
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16:32.45 | [TK]D-Fender | SeRi, Sounds ike you are missing some XML headers |
16:33.11 | SeRi | yea thats what I just figured |
16:33.19 | SeRi | but its looking for a revision |
16:34.50 | [TK]D-Fender | SeRi, Copy the heard tags off a sample |
16:35.00 | SeRi | just did. trying now |
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16:37.45 | SeRi | it took the file but I dont see the softkey.... not sure why. I am wondering if I am overlapping it or something |
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16:57.06 | LemensTS | [Nov 8 04:46:08] NOTICE[20856]: app_swift.c:429 app_exec: DTMF = 1 |
16:57.07 | LemensTS | Does this mean it is setting DTMF =1 as a channel variable? |
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17:23.57 | hudony | Hi : Been trying for 2 days to get it work by myself but it seems like I can't and I need advice : here is my config : asterisk 1.8 on machine with 2 nic (public and lan) with shorewall on it allowing everything outbound but only 10000-20000 udp and 5060 tcp/udp inbound. Within the lan, it all goes well but when calling form the outside, the call is initated with no error in the console but... |
17:23.59 | hudony | ...i got no audio |
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17:24.11 | Qwell | ~sipnat |
17:24.11 | infobot | [~sipnat] Quick guide on configuring Asterisk + 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 . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, canreinvite, externhost or externip, and localnet. |
17:24.24 | hudony | Yesterday, D-fender helped me by telling me about nat=yes for my phones and careinvite=no |
17:24.26 | hudony | But still.. |
17:24.36 | Qwell | and externip |
17:24.39 | hudony | I read it |
17:24.43 | hudony | externip and local... |
17:24.45 | hudony | forgot it |
17:24.51 | hudony | localnet |
17:26.07 | hudony | And I have qualify = yes |
17:26.18 | hudony | Tried to follow various tutorials |
17:29.05 | hudony | Here is my sip.conf configuration file : http://pastebin.com/28wSLqk4 |
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17:41.23 | autofsckk | hello, iax2 needs natting too? |
17:42.28 | Qwell | hudony: yeah you might want to change your primus password. |
17:42.57 | Qwell | and I don't see nat=yes and canreinvite=no in the same peers |
17:43.59 | hudony | this is not the good password |
17:44.35 | hudony | oh my bad, I removed careinvite=no a moment ago for testing purpose |
17:45.08 | hudony | he's back in but I still have the same proble |
17:45.16 | Qwell | what ports are set in rtp.conf? |
17:45.17 | hudony | I'm trying to see with sip set debug on |
17:45.21 | hudony | 10000:20000 |
17:48.30 | hudony | but since its a rtp problem...and I don't think I'' find anything interesting |
17:52.49 | hudony | Here is what I get : http://pastebin.com/Wew2vW8B |
17:53.25 | [TK]D-Fender | hudony, add nat=yes to [general] , nat=no to [5142252220] |
17:53.46 | [TK]D-Fender | hudony, Also what ver of * are you using? |
17:54.39 | hudony | 1.8.7.0 |
17:54.59 | hudony | and i saw <--- SIP read from UDP:209.183.11.198:5060 ---> |
17:55.00 | hudony | SIP/2.0 401 Unauthorized |
17:55.06 | hudony | ligne 88 of the pastebin |
17:56.01 | hudony | modemcable082*CLI> sip show registry |
17:56.03 | hudony | Host dnsmgr Username Refresh State Reg.Time |
17:56.04 | hudony | st-01.bvoice.primus.ca:5060 N 5142252220@b 74 Registered Tue, 08 Nov 2011 12:36:55 |
17:56.06 | hudony | 1 SIP registrations. |
17:56.11 | [TK]D-Fender | hudony, should be "directmedia=no" for all sections |
17:56.15 | hudony | ok |
17:56.19 | hudony | I'll try that |
17:56.37 | *** join/#asterisk michael-i (~anonymous@204.11.230.58.static.etheric.net) |
17:56.46 | hudony | It removes the careinvite=no from what i have read so do i remove the instruction |
17:56.53 | hudony | or do I keep both ? |
17:57.36 | hudony | oh nice |
17:57.47 | hudony | I have progress! : I can now hear incoming audio |
17:59.24 | hudony | The only part still down is that the caller doesn't hear the person who answered the call |
17:59.59 | hudony | I did a tcpdump on the asterisk box and udp packet are sent back from my external interface to primus when a call is occuring |
18:00.00 | hudony | :S |
18:00.12 | hudony | So I don't see why this isn't working |
18:01.13 | [TK]D-Fender | PB new configs |
18:01.27 | hudony | Sorry? |
18:02.29 | [TK]D-Fender | Show us what you've got now |
18:02.51 | eppigy | date with neuroscience girl |
18:02.55 | eppigy | life is interesting |
18:03.38 | hudony | you mean a sip log or my sip.conf file? |
18:05.17 | michael-i | Is there a magic flag somewhere to automatically exclude the calling channel from a multiple-party Dial()? E.g, SIP/201, SIP/202 and SIP/203 are in group 200. When 201 dials group 200, I want it to not call 201. |
18:07.02 | [TK]D-Fender | PB new configs |
18:07.22 | [TK]D-Fender | michael-i, Don't put it in your dial. |
18:07.39 | [TK]D-Fender | michael-i, You exclude it by not putting it there |
18:08.08 | michael-i | [TK]D-Fender: obviously…but I didn't want to dial n permutations of a single group of n extensions |
18:08.37 | [TK]D-Fender | michael-i, Whatever you put in Dial() is what you're going to get. |
18:08.55 | michael-i | s/dial/generate |
18:09.16 | michael-i | [TK]D-Fender: i seemed like a common problem and I thought there could be a magic flag in the dial app somwhere |
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18:09.42 | [TK]D-Fender | michael-i, Of course not. It does what you tell it. |
18:09.55 | [TK]D-Fender | michael-i, Just like ever other app |
18:10.36 | michael-i | [TK]D-Fender: yes, and one of the things you could tell it to do is to ignore the calling channel. But this is not the case |
18:11.07 | [TK]D-Fender | michael-i, What app gives you an option to ignore what you filled into another parm? |
18:11.29 | [TK]D-Fender | DoSomething(yes,IMeanNo) |
18:11.50 | hudony | New conf : http://pastebin.com/xrNhR2eZ |
18:12.29 | [TK]D-Fender | 12:53 <[TK]D-Fender> hudony, add nat=yes to [general] , nat=no to [5142252220] <------------- |
18:12.34 | michael-i | [TK]D-Fender: can't think of one, just seemed a common problem. Other solutions as to how "call all other extensions in a call group" would be welcome |
18:12.58 | [TK]D-Fender | michael-i, I don't see how its a problem. Don't put it in there. Sounds pretty easy. |
18:13.24 | hudony | oh |
18:13.27 | hudony | my bad.. |
18:14.02 | michael-i | [TK]D-Fender: I'm generating each group as an extension in the context "groups". So, instead I must generate a specific version of the group for each phone in the system. It's ugly, just that |
18:14.07 | hudony | Wow, it works! |
18:14.18 | [TK]D-Fender | \o/ |
18:14.19 | hudony | Dunno why but it does |
18:14.33 | hudony | Seriously...you guys are god... |
18:14.42 | [TK]D-Fender | hudony, Because * didn't know it was behind NAT because those other settings only come in when you actually tell it they matter |
18:14.58 | hudony | Can you tell me the nat=no directive to a phone that is behind a nat? |
18:15.07 | hudony | oh |
18:15.34 | [TK]D-Fender | <hudony> Can you tell me the nat=no directive to a phone that is behind a nat? <- ... huh? |
18:15.36 | hudony | Cause i have the asterisk 1.6 book and it doesn't says you can put nat=yes in general |
18:15.39 | hudony | only in peer :( |
18:16.03 | p3nguin | The sample config would have told you. |
18:16.04 | [TK]D-Fender | Just because it doesn't say you can doesn't mean you can't |
18:16.14 | hudony | I can tell :( |
18:16.42 | hudony | But what about putting nat=no to my peer (sip phone). The phone is nat. |
18:16.52 | hudony | I don't understand this part |
18:17.10 | [TK]D-Fender | hudony, * will tell your provider the wrong address if you don't put it in [general] |
18:17.16 | hudony | oh |
18:17.24 | p3nguin | If the phone is behind NAT, but it is the same NAT that Asterisk is behind, use nat=no for that phone. |
18:17.26 | [TK]D-Fender | "Where am I", vs "where are they" |
18:17.29 | [TK]D-Fender | Both matter. |
18:17.34 | hudony | ok I see |
18:17.39 | hudony | Thanks again to all of you! |
18:17.47 | hudony | You guys are really helpfull |
18:17.58 | [TK]D-Fender | michael-i, 2 lines of dialplan. Hardly "messy" |
18:17.59 | hudony | Have a good day! |
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18:56.55 | Naikrovek | Anyone here have access to Polycom UCS SIP 4.0? |
18:59.05 | *** part/#asterisk libryder (~david@209.33.214.243) |
19:02.19 | tm1000 | Naikrovek: yes |
19:02.52 | Naikrovek | have you use it yet? I want to try it but I don't want to wait for it to be released publicly |
19:03.09 | Naikrovek | It could be some time before that happens. |
19:03.30 | tm1000 | Yes. I've used it |
19:03.33 | tm1000 | its on my 550 |
19:03.39 | Naikrovek | your thoughts? |
19:03.45 | tm1000 | it just looked nicer |
19:03.50 | tm1000 | everything else is functionally the same |
19:04.28 | tm1000 | configuration files from 3.3.x work on 4.0 just fine |
19:05.19 | Naikrovek | yes, but SIP REASON header is supposed to be supported. Have you tried that? Also corp directory no longer requires the productivity license. |
19:05.57 | Naikrovek | new Web UI, etc. tons of stuff added |
19:06.04 | Naikrovek | it's that stuff I'm curious about |
19:07.49 | Naikrovek | also, the .tgz links on provisioner.net all go to non-existant topics. |
19:07.55 | Naikrovek | fyi. |
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19:08.49 | tm1000 | yes I know. considering the tgz are mainly for freepbx anyways and are hosted in a directory for direct access the links are kinda irrelevant |
19:09.01 | tm1000 | not sure why youd want them on the #asterisk channel |
19:09.05 | tm1000 | unless you are using freepbx? |
19:09.08 | tm1000 | Naikrovek: ^^ |
19:09.29 | Naikrovek | well i'm looking for a way to provision phones more intelligently, without using freepbx. |
19:09.39 | Naikrovek | guess your solution isn't what i'm looking for. no biggie. |
19:09.40 | tm1000 | well you should look at the repo |
19:09.44 | Naikrovek | alright. |
19:09.47 | tm1000 | uhhh? |
19:10.03 | tm1000 | it's used in freepbx, blue.box, whistle, and independently |
19:10.06 | Naikrovek | i have perl scripts that i use now but there are limitations and it needs rearchitected anyway. |
19:10.09 | tm1000 | it surely IS what you are looking for |
19:10.18 | Naikrovek | <-- not using freepbx |
19:10.30 | Naikrovek | or blue.box or whistle. |
19:10.35 | tm1000 | Naikrovek: did you skip over what I am talking about? |
19:10.43 | tm1000 | It's distro independent |
19:10.47 | tm1000 | it doesn't need any gui |
19:10.56 | tm1000 | eg "independently" |
19:10.58 | tm1000 | :-) |
19:11.03 | p3nguin | (1308.47) <tm1000> yes I know. considering the tgz are mainly for freepbx <-------- |
19:11.05 | Naikrovek | no, we're circling the drain of this conversation. we're responding to .. nevermind i'll chekc the repo done |
19:11.31 | tm1000 | p3nguin: Naikrovek the tarballs, tgz are for freepbx, the actual project is not |
19:11.43 | p3nguin | tgz IS a tarball. |
19:11.48 | tm1000 | omg |
19:11.50 | tm1000 | yes |
19:11.53 | tm1000 | read what I am saying |
19:11.58 | tm1000 | "the actual project is not" |
19:11.59 | *** join/#asterisk xpot-mobile (~james@70-91-210-237-BusName-Utah.hfc.comcastbusiness.net) |
19:12.04 | tm1000 | " the tarballs, tgz are for freepbx" |
19:12.06 | tm1000 | p3nguin: ^^ |
19:12.08 | irroot | furball or hairball |
19:12.12 | Naikrovek | you're lagging or something. lol you're responding to 2 lines back all the time. |
19:12.16 | p3nguin | furries! |
19:12.34 | Naikrovek | anyway you're caught up now. everyone is on the same page |
19:12.39 | Naikrovek | geepers |
19:12.42 | tm1000 | p3nguin: Naikrovek https://github.com/tm1000/Provisioner |
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19:13.19 | Naikrovek | yup already there |
19:13.59 | tm1000 | Naikrovek: i sent you a pm, check |
19:17.42 | tm1000 | note to self. fix links on provisioner.net |
19:17.49 | tm1000 | re-do site |
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19:31.22 | WIMPy | What does this mean or what should I watch out for? - ERROR[32019]: lock.c:258 __ast_pthread_mutex_lock: chan_sip.c line 14833 (register_verify): 'peer' really deep reentrancy! |
19:31.50 | WIMPy | I have no idea when it happens verbose and debug both at 9 don't show anything. |
19:32.02 | p3nguin | OH NOSE! YOU'VE ENTERED THE ABYSS! |
19:32.54 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-azvmcmkpsmrinukc) |
19:33.03 | Pio | when i use a certain softphone (qutecom) and asterisk transfers a call from hold music to the sip phone, i get a lot of distortion and choppy audio for the first second or two, then it goes away.. |
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19:34.04 | irroot | Pio this could be jitterbufers and or some form of echo can the phone may support |
19:34.37 | irroot | pio or even pc load when a call is made |
19:34.37 | Pio | what i dont understand is, if i have directmedia=no in my sip.conf.. to prevent the softphone from ever talking to anyone but asterisk.. shouldnt that mean that, despite transferring or any other activity on asterisk, its one unbroken audio stream? |
19:35.11 | p3nguin | It's not necessarily unbroken. |
19:36.17 | Pio | its weird, ekiga seems to be the only softphone which i have no issues with.. but ekiga's build on ubuntu oneiric is crappy, it segfaults half the time when you try to start it, other issues |
19:36.42 | Pio | twinkle and qute both have audio problems, but not the SAME audio problems.. its very odd |
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19:39.28 | irroot | Pio i use blink its quite usfull |
19:39.53 | Pio | googles.. no linux client :( |
19:40.06 | Pio | oh wit |
19:40.09 | irroot | i use it on ubuntu |
19:40.16 | Pio | yeah its just not on the front page here.. lets see |
19:41.30 | SeRi | anybody experience with polycom phones? I am trying to disable a softkey and add a new one but no matter what I do the "Forward" softkey is all ways there |
19:41.35 | Pio | i'll try these natty packages |
19:41.56 | SeRi | better said remove one |
19:43.01 | irroot | SeRi been a while all the magic in sip.conf is pain to edit |
19:43.38 | SeRi | irroot, my fingers are bleeding! |
19:43.52 | SeRi | all morning at it and nothing. |
19:46.34 | irroot | SeRi <sip><softkeys ..... |
19:46.48 | LemensTS | SeRi: you are editing the cfg files right |
19:48.06 | SeRi | guys here is what I have |
19:48.08 | SeRi | http://pastebin.com/raw.php?i=XigK7V9A |
19:48.23 | SeRi | softkey.cfg gets called from mac.cfg |
19:49.02 | SeRi | i remove all <softekeys> entries from sip_317.cfg |
19:49.15 | SeRi | This is a polycom 501 |
19:49.33 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
19:49.35 | SeRi | LemensTS, Yes. |
19:49.49 | irroot | Seri looks like some spaces got mixed up |
19:50.54 | SeRi | irroot, Like what? |
19:51.09 | irroot | softkey.1.use.active="1"softkey.feature.newcal |
19:51.20 | irroot | they args run into each other |
19:51.36 | irroot | this may be a problem |
19:51.37 | LemensTS | SeRi: http://pastebin.com/yXi3RPMV |
19:51.55 | Pio | irroot, blink is working very well so far, thanks |
19:52.20 | LemensTS | SeRi: that is the config I run on some of mine, should give you some tips |
19:52.34 | [TK]D-Fender | SeRi, I think I see it : If you attempted to "export" that section to a separate file you need to recreate the XML tree up to that tage, IE : <sip/> .... etc |
19:52.34 | irroot | Pio its not perhaps ideal but it seems stable |
19:52.45 | SeRi | LemensTS, Thanks. irroot Thanks for the point out. |
19:52.50 | [TK]D-Fender | SeRi, Because EFK isn't a base-level element |
19:53.24 | Pio | irroot, yeah, no audio problems, less buggy than ekiga, its a net improvement for me :) |
19:54.43 | Pio | no tray icon? thats kinda weird |
19:55.25 | SeRi | [TK]D-Fender, I am not following.... :/ sorry. so I have to move my tag? |
19:55.39 | [TK]D-Fender | SeRi, You have to add the oter ones. |
19:55.42 | [TK]D-Fender | outer* |
19:57.34 | SeRi | [TK]D-Fender, here is what I was following... some what |
19:57.35 | SeRi | http://wiki.sipfoundry.org/display/sipXecs/Polycom+Phone+Customization |
19:58.21 | [TK]D-Fender | SeRi, "This XML snippet must be between the <sip></sip> tags which are at the top and bottom of the configuration file" |
19:58.23 | [TK]D-Fender | ^^^ |
19:58.43 | [TK]D-Fender | add those around your file |
19:58.44 | SeRi | ops |
19:59.13 | SeRi | Thanks for the point out :) |
20:00.01 | irroot | SeRi grab a xml tree viewer from the net should help you |
20:00.25 | SeRi | I was looking for one for nix and couldnt find one :( |
20:00.29 | SeRi | slackware here |
20:01.23 | *** join/#asterisk JuanCri (~JuanCri@200.72.190.92) |
20:01.49 | SeRi | here is what I got now |
20:01.50 | SeRi | http://pastebin.com/raw.php?i=TuiAgL4B |
20:02.32 | SeRi | fires up a vm |
20:03.38 | irroot | SeRi <sip> must come after <?xml header |
20:04.17 | p3nguin | conglomerate, editix, kxmleditor, qxmledit, serna, tx, xmlcopyeditor, xxe |
20:04.24 | p3nguin | all XML editors |
20:04.28 | *** join/#asterisk kpettit (~kpettit@99-116-144-138.lightspeed.hstntx.sbcglobal.net) |
20:05.47 | leifmadsen | likes xmlmind |
20:05.57 | leifmadsen | wrote multiple Asterisk books using that editor |
20:05.58 | p3nguin | aka xxe |
20:06.07 | leifmadsen | p3nguin: oh I totally missed that on the end |
20:06.57 | p3nguin | Last, but not necessarily least. |
20:07.10 | leifmadsen | or first! |
20:07.16 | leifmadsen | or best! |
20:07.34 | leifmadsen | decisions?! we don't need no stinkin' decisions! |
20:07.46 | SeRi | lol |
20:07.52 | Pio | irroot, with blink, does it not support history when you use your own sip server? the 'history' section is just always grayed out |
20:11.41 | Katty | peeks in |
20:14.52 | SeRi | I give up. I am tried. |
20:16.08 | irroot | Pio no clue :P i just use it on the run and as a test phone |
20:16.25 | irroot | checks Katty out |
20:17.46 | cusco | hi |
20:18.05 | cusco | in cdr, how can I have a unique identifier for the whole call? as uniqueid keeps changing... ? |
20:18.46 | p3nguin | There should only be one unique identifier for the channel. |
20:19.54 | cusco | well the channel changes |
20:20.04 | *** join/#asterisk simNIX (~simNIX@156-60.bbned.dsl.internl.net) |
20:20.15 | cusco | it is Local/bla and SIP/bla etc |
20:20.17 | cusco | for the same call |
20:20.19 | p3nguin | As a call comes into asterisk, that channel remains the same until that call ends. |
20:20.29 | cusco | hu? no! |
20:20.58 | cusco | as it enter the queue, queue dials SIP/1 then SIP/2 tehn SIP/3 |
20:21.03 | cusco | so it creates a new channel |
20:21.22 | cusco | in the same call ! |
20:21.22 | p3nguin | If I pick up my phone and dial some extension, my phone brings up a channel and makes the call. The channel that is created by my phone does not change until I hang up. |
20:21.40 | *** join/#asterisk Netgeeks (~chris@gw1.netgeeks.net) |
20:21.49 | cusco | unless that ext I dial, is a local dialplan that goes to a queue first |
20:21.57 | cusco | and it tries several dials |
20:22.01 | cusco | until one is answered |
20:22.07 | p3nguin | Any call you make begins on some extension. |
20:22.08 | cusco | so it created and destroyed several channels |
20:22.17 | cusco | right... |
20:22.24 | cusco | well.. |
20:22.38 | p3nguin | So if you're talking about someone calling inbound from the PSTN to asterisk, that's just one channel. |
20:22.44 | p3nguin | There is a uniqueid for it. |
20:22.52 | p3nguin | It remains until the caller hangs up. |
20:22.56 | cusco | the uniqueid changes too |
20:23.08 | *** join/#asterisk scubes13 (~scubes13@24.168.196.0) |
20:23.22 | p3nguin | The channel created does not change until that leg of the call is disconnected. |
20:23.23 | cusco | for every dial to the several queuemembers a new id |
20:23.52 | p3nguin | I'm tired of repeating myself, so I'll leave you to figure out what I'm saying. |
20:24.10 | *** join/#asterisk simNIX (~simNIX@156-60.bbned.dsl.internl.net) |
20:24.31 | SeRi | is building xmlcopyeditor |
20:24.39 | cusco | I (think I ) understand you, but what I'm trying to say is: there are several channels created and destroyed in a whole call |
20:25.34 | p3nguin | Set a variable on the inbound leg of the call and see how far you can carry it. |
20:25.41 | *** join/#asterisk grandpapadot (~grandpapa@99.175.248.81) |
20:25.41 | Netgeeks | in asterisk 1.6 you could set a variable in a subroutine (gosub) like __channel_wide_variable = monkeys and you could read that variable from outside the subroutine, in asterisk 1.8, this appears to have changed, and you are not able to read the variable as set in the sub from outside the sub. Is there a way to set a variable inside a sub in ast 1.8 that makes it accessable outside the sub? |
20:25.56 | p3nguin | Set(__myUniqueID=12345) |
20:26.14 | grandpapadot | Hey guys, in 1.8.x, is the "h" extension no longer valid in a Macro? I'm noticing my calling context's "h" extension being fired ... can't find anything discussing ... |
20:26.31 | cusco | p3nguin: im dong that, im taking the first uniqueid and keeping it |
20:26.42 | cusco | problem is i cannot set __CDR(UID)=bla |
20:26.44 | p3nguin | h shouldn't normally be run in a macro. |
20:26.59 | p3nguin | Why can't you set it? |
20:27.08 | leifmadsen | 'h' has never been valid in a Macro() |
20:27.11 | [TK]D-Fender | Netgeeks, * dialplan has no sense of scop from Gosub <- |
20:27.11 | cusco | __CDR(UID) CDR(UID) cannot be read after |
20:27.17 | cusco | only CDR(UID) |
20:27.25 | cusco | I tried, but CDR is a function, right= |
20:27.28 | cusco | not a var? |
20:27.29 | [TK]D-Fender | Netgeeks, This isn't a high-level language. all vars are global to the channel; |
20:27.30 | grandpapadot | @leifmadsen - tnx |
20:27.40 | p3nguin | Correct, the CDR function is a function. |
20:28.02 | cusco | so I cannot set __CDR() only __VAR |
20:28.09 | leifmadsen | Netgeeks: ya, GoSub() allows you to read it anywhere -- there is nothing stopping you from assigning a variable outside a GoSub() and reading from the GoSub(), and vice-versa (unless you use LOCAL()) |
20:28.15 | Qwell | CDR() is a function.. |
20:28.17 | leifmadsen | cusco: right, that is not valid |
20:28.31 | leifmadsen | cusco: set a channel variable with the data, then set it on the other channel with the CDR() function |
20:28.33 | cusco | but I would like CDR(UID) to be inherithed too |
20:28.34 | p3nguin | But you can set a __variable and then set CDR() to the value of the variable. |
20:28.59 | leifmadsen | cusco: you may need to look at using something like U() or M() flags to Dial() |
20:29.08 | Netgeeks | [TK]D-Fender: well, i just upgraded an asterisk with a diaplan that is doing what I said. in 1.6 it works just fine, I set a variable __dialed_exten=1234566 and the next line in the code after the return is a verbose(1,${dialed_exten}) . when I load the same diaplan in 1.8, the variable comes up empty in the verbose statement |
20:29.09 | leifmadsen | it'll execute dialplan on the called channel |
20:29.32 | [TK]D-Fender | <PROTECTED> |
20:29.36 | leifmadsen | Netgeeks: there is no reason that wouldn't work as nothing should be changed |
20:29.39 | [TK]D-Fender | that's part of the problem. |
20:29.44 | [TK]D-Fender | Set like normal, use like normal |
20:29.49 | leifmadsen | ^^ that |
20:29.50 | cusco | ok thanks, I'll look that up |
20:29.57 | Netgeeks | roger |
20:30.13 | p3nguin | When the call comes in, Set(__myUniqueID=${UNIQUEID}) ... and then on other channels, Set(CDR(UID)=${myUniqueID}). |
20:30.48 | cusco | I actually tried that... lol I need to review it |
20:30.51 | cusco | thanks |
20:31.51 | Netgeeks | [TK]D-Fender so you are saying don't use Set(__dialed_exten=${<some number string>}), use Set(dialed_exten=${....})? |
20:32.05 | [TK]D-Fender | yes |
20:32.26 | Netgeeks | or in reality, in my case the real code is an ARRAY() but I assume the same for it |
20:32.27 | [TK]D-Fender | Netgeeks, In a call vars aren't local to a context |
20:32.49 | Netgeeks | but if I don't use __ before the var, it isn't passed to child channels, correct? |
20:33.09 | [TK]D-Fender | channel inheretence is another matter |
20:33.30 | p3nguin | If you use one underscore, it will be inherited by one newly created channel off the main channel. |
20:34.10 | *** join/#asterisk michael-i (~anonymous@204.11.230.58.static.etheric.net) |
20:34.23 | Netgeeks | so, in 1.8 if I want to set a variable in a gosub, and have that variable read anywhere in the dialplan by that channel, and also have it inherited by subsequent channels, what form of set statement would I use? |
20:34.36 | leifmadsen | the same as any other place |
20:34.40 | leifmadsen | GoSub() is not magical |
20:34.43 | Netgeeks | but it's not working leif |
20:34.45 | leifmadsen | it is just a Goto() with memory |
20:34.47 | Netgeeks | I just tested |
20:34.55 | leifmadsen | then provide enough information showing the error and ability to reproduce |
20:35.38 | Netgeeks | kk, I'll be back with a simplifed dialplan that reflects this, or I'll be back with an embarrasing oops, I see what I did wrong |
20:37.08 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
20:39.01 | leifmadsen | Netgeeks: I suspect a typo |
20:39.28 | p3nguin | or failure to "dialplan reload" after making minor changes. |
20:39.56 | Netgeeks | leif, I wish, the thing that bothers me is that 1.6 the dialplan works just fine, 1.8 it doesn't and I can at this point state that the dialplans are identical (copied & diffed for extra sureness) |
20:40.13 | p3nguin | Show us. |
20:40.20 | leifmadsen | Netgeeks: works fine here: http://pastebin.com/DhzaBzVs |
20:40.24 | p3nguin | I moved my 1.4 dial plan to my 1.8 system, and it didn't break. |
20:40.28 | leifmadsen | Netgeeks: I just tested on Asterisk 10 |
20:40.41 | p3nguin | I had a few warnings about syntax change, but I don't remember anything breaking. |
20:43.17 | p3nguin | If I have a complaint about choppy sounding audio on one side of a call, should I concentrate on bandwidth/network usage or is there any chance that an underpowered asterisk computer could cause it? CPU usage remains very low, and there is no transcoding (using ulaw on both legs of the call). |
20:43.35 | *** join/#asterisk lcat (~lcat@187.45.254.21) |
20:43.51 | *** join/#asterisk jplank (~G_Bove@208-104-67-27.dyn.fttp.comporium.net) |
20:43.56 | *** part/#asterisk jplank (~G_Bove@208-104-67-27.dyn.fttp.comporium.net) |
20:44.55 | leifmadsen | p3nguin: sounds like a network issue to me if CPU isn't the problem, especially with no transcoding. What about I/O for recording calls? Typically though my experience with choppy audio nowadays stems from either a provider problem, or a network issue or some sort (i.e. I had a problem where my ITSP was getting DDOS'd -- that caused audio issues :)) |
20:45.02 | *** join/#asterisk jplank (~G_Bove@208-104-67-27.dyn.fttp.comporium.net) |
20:46.31 | p3nguin | I'll have to look into the I/O while MixMonitor() is recording the call. When I review the call, I hear crystal clear audio on both sides, which makes me think the loss is between my upstream side of my CPE and my ITSP, rather than between me and the phone. |
20:47.12 | *** join/#asterisk lhfnet (~lhfnet@net-cdd-fw01.cddlasmercedes.com) |
20:47.23 | *** join/#asterisk GameGamer43 (~GameGamer@li119-60.members.linode.com) |
20:48.13 | p3nguin | My initial thought was that it could have been due to some torrent traffic... since it was during the weekend not during business hours. |
20:48.17 | lhfnet | Hi, I am having this message continuously in the CLI chan_sip.c:6343 sip_write: Asked to transmit frame type alaw, while native formats is 0x4 (ulaw) read/write = 0x4 (ulaw)/0x4 (ulaw), I set disallow = all allow=ulaw in all my users in users.conf and in the sip.conf and iax.conf. Using Asterisk 1.8.7.1 |
20:48.43 | *** part/#asterisk jplank (~G_Bove@208-104-67-27.dyn.fttp.comporium.net) |
20:48.49 | p3nguin | I obviously can't make that determination after the fact, but I can check it next time when I'm told as it happens. |
20:49.08 | vader-- | hmmm, i need to come up with a recommendation for a guy |
20:49.26 | p3nguin | can recommend a guy for vader-- |
20:49.43 | vader-- | he has 3 pots lines for his little store... right now he has a samsung prostar 816 plus pbx that isn't hooked up |
20:49.48 | vader-- | he has the pbx and all the phones |
20:50.43 | Netgeeks | aha! I reproduced it in a simplified dialplan |
20:50.56 | Netgeeks | it's broken when using ARRAY() to set the vars |
20:51.14 | vader-- | so im not sure if i should recommend he stay with the prostar systema nd ill try and hook it up for him |
20:51.27 | vader-- | or build some sort of small freepbx system, or myabe a talkswitch system or something |
20:54.34 | Netgeeks | http://pastebin.com/uRHYTsQv |
20:57.08 | Netgeeks | hrm, ignore that pastebin, it's author is braindead at the moment |
21:00.06 | Netgeeks | [TK]D-Fender so what I was doing wrong is that the original author of this dialplan wrote an array assignment like Set(ARRAY(var1,var2)=value1\,value2) |
21:00.36 | [TK]D-Fender | Netgeeks, Well, I can't speak for hte use of ARRAY as I've never touched it... but OK :) |
21:00.47 | Netgeeks | well, in 1.6 that works just fine, in 1.8, the / escapes the comma, and .... all the values get put in var1 |
21:00.52 | Netgeeks | and I was looking at var2 |
21:01.09 | SeRi | LemensTS, I use your code with an xmleditor and the phone still does not change. |
21:01.10 | Netgeeks | well, var2 is now empty, because array thought it only had one long string |
21:01.54 | Netgeeks | so I just need to go remove all the / from in front of commas in his array statements |
21:07.33 | *** join/#asterisk LittleFool (~LittleFoo@95.129.212.120) |
21:10.52 | leifmadsen | Netgeeks: oh ya, you don't need to escape things anymore |
21:11.27 | Netgeeks | I'm quite happy right now, this was nearly the easiest fix I've ever had to do |
21:11.27 | leifmadsen | the dialplan is smarter about that now |
21:12.07 | leifmadsen | Set() is not longer a multi-Set() application (which is why commas needed to be escaped before). Even easier might have been s/Set/MSet/g :) |
21:12.32 | leifmadsen | then you probably could have left the escape character there |
21:12.42 | leifmadsen | (what you did is better though) |
21:12.58 | Netgeeks | searching and replacing \, with , was simple enough |
21:13.01 | michael-i | [TK]D-Fender: does this look elegant enough to filter group members from group calls? Set(FILTERED_DIAL_CHANNELS=${STRREPLACE(${DIAL_CHANNELS},SIP/${CONTEXT})}) |
21:14.07 | michael-i | It keeps on returning an empty string…so I'm wondering if I can even use a variable like that in STRREPLACE |
21:15.00 | [TK]D-Fender | michael-i, you just issues a replace with 2 parms. Doesn't look kosher. Perhaps you should reread each function's instructions. |
21:15.48 | michael-i | [TK]D-Fender: replace-with is optional |
21:16.11 | [TK]D-Fender | michael-i, Well I don't ee what those vars start with .... |
21:17.22 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
21:17.49 | [TK]D-Fender | michael-i, I don't see that function in 1.6.2 ... that 1.8 new? |
21:17.59 | michael-i | [TK]D-Fender: 10 |
21:18.05 | *** join/#asterisk navaismo (~navaismo@187.170.0.233) |
21:19.52 | [TK]D-Fender | michael-i, PB the function instructions and your complete code |
21:20.44 | *** join/#asterisk garymc (~chatzilla@host86-176-88-100.range86-176.btcentralplus.com) |
21:21.35 | vader-- | any recommendations on a 8-16 port cheap switch with poe? |
21:24.10 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
21:24.10 | leifmadsen | michael-i: you don't use ${DIAL_CHANNELS} -- just use DIAL_CHANNELS |
21:24.23 | leifmadsen | michael-i: you give it the name of the variable to work with, not the value of the variable |
21:24.27 | [TK]D-Fender | leifmadsen, that's what I was suspecting from similar functions.... |
21:25.24 | leifmadsen | michael-i: Set(thisResult=${STRREPLACE(DIAL_CHANNELS,SIP/${CONTEXT},abc,CBA)}) |
21:27.36 | michael-i | leifmadsen: thanks! |
21:28.14 | michael-i | I get a bit lost in escaping, etc since I'm writing code which generates this dialplan logic |
21:30.44 | leifmadsen | michael-i: yuch :) |
21:31.41 | garymc | in asterisk cli what command shows me the sip phones trying to connect and failing? |
21:31.58 | garymc | ive tried "sip set debug" but not working |
21:32.05 | garymc | is that command out dated now? |
21:37.05 | michael-i | leifmadsen: moving the logic into a one-time generation is the only way I can cut down on resource requirements :) it's an art unto itself sometimes |
21:37.27 | leifmadsen | michael-i: I've gone the route of writing code to write code, and it always ended up in misery for me |
21:37.36 | leifmadsen | garymc: sip set debug on |
21:37.45 | leifmadsen | garymc: tab completion ftw |
21:37.59 | garymc | thanks |
21:38.29 | garymc | sip set debug on = no such command |
21:41.02 | garymc | leifmadsen: what is tab comletion ftw? |
21:41.10 | garymc | *completion |
21:41.16 | leifmadsen | sip set<tab> |
21:41.26 | leifmadsen | garymc: what version of asterisk? is chan_sip.so loaded? |
21:41.38 | garymc | not sure 1.6 |
21:41.41 | leifmadsen | sip<tab |
21:41.46 | leifmadsen | will show you optoins |
21:41.53 | leifmadsen | core show version |
21:42.09 | garymc | 1.6.2.11 |
21:42.50 | garymc | no joy with any of those commands |
21:45.08 | navaismo | try with module reload chan_sip.so |
21:45.31 | *** part/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
21:45.44 | garymc | how do i do that? |
21:45.52 | navaismo | module reload chan_sip.so |
21:46.46 | garymc | i did that nothing happened |
21:47.11 | navaismo | error warning something? |
21:47.13 | irroot | garymc it could be its snarled up |
21:47.20 | irroot | core show locks ?? |
21:47.33 | irroot | it may need a reastart |
21:47.39 | leifmadsen | ya sounds like a deadlock |
21:48.38 | garymc | core show locks = nothing |
21:49.06 | leifmadsen | may not be enabled in menuselect |
21:49.09 | navaismo | stop asterisk, then asterisk -vvvvvcg and check if the module load |
21:51.59 | garymc | yeah its working now |
21:52.04 | navaismo | leifmadsen: which option in menuselect? |
21:52.42 | leifmadsen | navaismo: https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace |
21:52.52 | navaismo | thx |
21:53.02 | leifmadsen | navaismo: see section "Getting Information for a Deadlock" |
21:53.27 | irroot | DHCP server gone bad <- horror movie for next halloween i haz one ATM |
21:53.38 | garymc | anyone know why my softphone is failing to register http://pastebin.com/Uk4UHqrz |
21:53.43 | navaismo | thx leifmadsen |
21:56.47 | navaismo | garymic i see some Got SIP response 405 "Method Not Allowed" and SIP/2.0 403 Forbidden (Bad auth) |
21:56.48 | navaismo | brb |
21:58.21 | garymc | do i have to create an rtp.conf file or is there already one |
22:00.48 | garymc | getting this in asterisk cli now: |
22:00.50 | garymc | Got SIP response 405 "Method Not Allowed" back from 81.1 |
22:11.56 | *** join/#asterisk ruied (~ruied@po-217-129-154-119.netvisao.pt) |
22:12.54 | ruied | hello, how can I do a "case" like function in asterirsk? |
22:13.48 | ruied | is there any "case" function? |
22:14.27 | Qwell | several if checks |
22:14.32 | navaismo | gotoif or agi |
22:14.36 | navaismo | too late |
22:14.39 | Qwell | or gotoif, yeah |
22:14.45 | Qwell | or just goto, for that matter |
22:16.26 | ruied | ok, thanks :) |
22:18.38 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
22:20.09 | *** join/#asterisk irroot (~irroot@197.172.248.230) |
22:21.08 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
22:22.41 | *** join/#asterisk saisoma (~saisoma@client72.jdcc.edu) |
22:22.43 | ruied | I think goto is best for what I need, I'm making a macro with blf custom lamps. Press once to redirect incoming calls to my store (green light buton), press again, incoming calls to my house (red button), press again to redirect to my mobile (red blinking)... |
22:23.40 | ruied | and when I press the button it will say a voice of what it is doing ex: "Retirecting calls to Mobile"... |
22:23.43 | *** join/#asterisk timeshell (~chatzilla@gw.lusi.on.ca) |
22:23.47 | timeshell | leifmadsen ping |
22:23.52 | leifmadsen | o/ |
22:23.55 | timeshell | Ola |
22:23.59 | timeshell | Where do I find LOG_DEBUG |
22:24.05 | leifmadsen | what do you mean? |
22:24.18 | timeshell | Per dumphistory = yes|no : Enable support for dumping of SIP conversation's transaction history to LOG_DEBUG. Default no. (New in v1.2.x) |
22:24.26 | leifmadsen | logger.conf |
22:24.27 | *** join/#asterisk Greenlight (~Wullie@cpc2-dund11-2-0-cust994.sgyl.cable.virginmedia.com) |
22:24.53 | leifmadsen | log to a file with debug level logging, and 'core set debug 10' |
22:26.20 | Greenlight | Evening folks. I've an issue with my voip provider (voiptalk). At certain perioids during the day their servers stop replying to my SIP Invites in a timely fashion (sometimes no replies at all, sometimes takes 6 retries). This is causing delays of 20+ seconds for calls to start ringing, or sometimes causing complete timeout. Has anyone else ever had issues like this, and if so what was the |
22:26.21 | Greenlight | solution? |
22:26.52 | Qwell | solution: fire them |
22:27.03 | Qwell | You've already called and complained, right? |
22:27.07 | wdoekes2 | ruied: Goto(s-${value),1) and then have exten s-CASE1..., s-CASE2... |
22:27.16 | Greenlight | Indeed - they've had 3 days to look over the traces now |
22:27.40 | Greenlight | They asked me to increase the T1 timeout up to 1500ms (the SIP RFC states it should be 500ms) |
22:27.45 | Greenlight | It didn't help ;/ |
22:28.03 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
22:28.20 | Greenlight | I'm leaning towards binning them, only thing is they have a really nice relseller system which we've got a bunch of our customers using |
22:29.08 | ruied | wdoekes2, that seems to be the best structure |
22:30.02 | Greenlight | Anyone recommend decent SIP provider in the UK, who ideally offer reseller/wholesale options. Only a 3 or so million minutes a month at present. |
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22:30.56 | *** mode/#asterisk [+o mnicholson] by ChanServ |
22:31.08 | saisoma | hey guys, having an issue with voicemail and a branch office: http://pastebin.com/1zZfMmZ9 any assistance is greatly appreciated |
22:31.59 | *** join/#asterisk mnicholson (~mnicholso@nat/digium/x-jlhmqhwkgixbgnss) |
22:31.59 | *** mode/#asterisk [+o mnicholson] by ChanServ |
22:32.24 | timeshell | leifmadsen Should I strip out personal IP's and such from the file? Or can I submit it somewhere where it will remain confidential? |
22:33.23 | *** join/#asterisk nunne (~nunne@static-213-115-116-75.sme.bredbandsbolaget.se) |
22:34.42 | Greenlight | While I'm in here - Where would be the best place to go to find Asterisk consultants? |
22:39.00 | leifmadsen | timeshell: strip it because making it confidential greatly reduces the number of eyes who can look at it |
22:39.10 | leifmadsen | Greenlight: asterisk-biz mailing list I guess |
22:39.14 | timeshell | k |
22:39.27 | Greenlight | Okay, thanks |
22:39.45 | _Corey_ | Greenlight: You can search online or put in a request with Digium's website for a referral too |
22:42.15 | Greenlight | I've had a brief look online, but wasn't too sure. I might try the Digium website. Am looking for a person/company who we can have investigate any issues with our Asterisk systems that we dont have the time or resources to deal with internally. We were thinking of recruiting someone for the post but I think our budget might be better spent on a reliable consultant who we can use as needed |
22:42.53 | _Corey_ | There are a number of dCAPs that can be found on LinkedIn |
22:43.03 | _Corey_ | (search for the dCAP group) |
22:43.04 | SeRi | Greenlight, talk to p3nguin |
22:43.26 | Greenlight | _Corey_: Cool will check out LinkedIn, ty |
22:43.38 | Greenlight | SeRi: Is he UK based? |
22:43.45 | SeRi | ops. no sorry |
22:43.58 | Greenlight | US ? |
22:44.03 | SeRi | Yes |
22:44.21 | *** join/#asterisk jrose_atDigium (~jon@nat/digium/x-wnprsiohpjmqjzfs) |
22:44.29 | Greenlight | Might still work actually - guess it may even be helpful if "out of hours" here is during working day there |
22:45.01 | _Corey_ | Greenlight: My recommendation in the UK would be David Duffet, just google him |
22:45.23 | Greenlight | Ok, that's perfect, Thanks again, I shall |
22:46.40 | _Corey_ | No problem |
22:46.57 | Greenlight | _Corey_: Just wondering if you've worked with him, or just know of him? |
22:49.58 | _Corey_ | We've both spoken at Astricon for the last few years, so I've known him at least that long. He runs on the of the best Digium Resellers in Europe |
22:50.25 | _Corey_ | Really good guy though, if I were in the UK and needed someone he'd be my first call |
22:50.45 | Greenlight | Sounds like a solid endorsement! :) |
22:51.07 | Greenlight | Will be getting in touch with him I think, thanks again, it's appriciated |
22:51.27 | _Corey_ | I'm not sure what sort of engagement terms his shop offers... we mostly do maintenance contracts here |
22:51.33 | _Corey_ | Best of luck though |
22:51.54 | Greenlight | Hopefully he can offer something that is a fit with out needs |
22:51.59 | Greenlight | *our |
22:53.08 | leifmadsen | David Duffett is awesome, if you need someone in Europe, ya, call him |
22:53.38 | Greenlight | He certainly sounds like the right chap |
22:54.36 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-145.cablep.bezeqint.net) |
22:57.52 | Greenlight | Anyone experience using AQL for voip termination in the UK ? |
23:01.31 | vader-- | Whats a good phone for a small business? Cost effective but not crippled? |
23:01.55 | Greenlight | As in hard phone? |
23:01.59 | vader-- | ya |
23:02.03 | vader-- | IP phone |
23:02.09 | Greenlight | Depends what features you're needing |
23:02.28 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-catefqwfgqdqvpfj) |
23:02.36 | vader-- | hold mainly |
23:02.36 | vader-- | hehe |
23:02.41 | Greenlight | We use snom300's and never had any problems |
23:02.51 | Greenlight | I'd guess they're somewhat midrange |
23:03.20 | vader-- | PoE? |
23:03.40 | Greenlight | Yea - they also come with AC adaptors though |
23:04.34 | vader-- | can you intercom with them? |
23:05.05 | Greenlight | Never tried to be honest, but I'd imagine so. Most hard phones generally support the required sip header |
23:05.51 | vader-- | i guess the biggest thing this guy would want to be able to do is put a call on hold and pick up the call from another phone |
23:06.17 | Greenlight | Well that's more parking than hold |
23:06.34 | Greenlight | So you'd just use a featurecode on Asterisk to handle that |
23:07.40 | Greenlight | http://downloads.snom.net/documentation/data_snom300_en.pdf |
23:08.07 | Greenlight | Works great, and doesn't feel "cheap" |
23:10.11 | vader-- | how are the cisco SPA phones? |
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23:16.57 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-catefqwfgqdqvpfj) |
23:22.57 | SeRi | [TK]D-Fender LemensTS I got it!!!!!!!!!!!!!!! :D |
23:23.08 | SeRi | so much blood spilled.... |
23:23.13 | SeRi | but I got it :D |
23:26.23 | paulc | Looking for a recommendation on SIP DECT phones.. Panasonic vs Gigaset S675 IP.. leaning towards the latter, perhaps.. anyone got any thumbs up or down for either? |
23:45.15 | *** join/#asterisk ketas-ts (~ketas@82.131.20.5.cable.starman.ee) |
23:45.16 | ruied | wdoekes2, done! :) |
23:54.27 | *** join/#asterisk michael-i (~anonymous@204.11.230.58.static.etheric.net) |
23:54.59 | michael-i | Is there a way to jump directly into VoiceMailMain() to record a new greeting? |
23:57.08 | ChannelZ | wonders if there is a shortcut as part of MiniVM |
23:57.41 | michael-i | Do they use the same directory structure? |
23:59.30 | ChannelZ | Not positive. |
23:59.59 | ChannelZ | You could probably "roll your own" easily enough by constructing the path to the greeting and just using Record |