| 00:01.23 | CrashHD | are priority labels broken in 1.4? |
| 00:01.31 | Qwell | be more specific |
| 00:01.50 | *** join/#asterisk mavior (n=Miranda@88-149-160-22.f5.ngi.it) |
| 00:01.54 | CrashHD | extension,n(label),NoOP |
| 00:01.57 | mavior | hello everybody |
| 00:02.03 | CrashHD | Goto(label) |
| 00:02.15 | CrashHD | is droping channel as if there is no (label) |
| 00:02.23 | Strom_C | i smell a typi |
| 00:02.25 | Strom_C | er, typo |
| 00:02.53 | Strom_C | pastebin that extension |
| 00:03.02 | CrashHD | http://www.pastebin.ca/316802 |
| 00:03.03 | mavior | can somebody tell me how to exit from the asterisk CLI, after started by /usr/sbin/asterisk -cvvv ? |
| 00:03.27 | fetcher | mavior: 'quit', but that will shut down asterisk as well |
| 00:03.29 | Strom_C | CrashHD, your gotoif is wrong |
| 00:03.38 | fetcher | mavior: or 'stop now' |
| 00:03.40 | Strom_C | try |
| 00:04.05 | Strom_C | GotoIf($["${VARNAME}foo" != "foo"]?label) |
| 00:04.20 | mavior | without stop asterisk, is it possible? |
| 00:04.21 | JT | mavior: i know closing the controlling terminal will do it |
| 00:04.31 | CrashHD | k brb |
| 00:04.37 | mavior | how to run it in background? |
| 00:04.39 | Strom_C | mavior, start asterisk in the background by just using /usr/sbin/asterisk |
| 00:04.53 | Strom_C | then type /usr/sbin/asterisk -r to connect to the console |
| 00:05.14 | mavior | and if I want to re-call the CLI by asterisk -r , then how can I exit it? |
| 00:05.21 | Strom_C | exit |
| 00:05.26 | JT | <PROTECTED> |
| 00:05.31 | mavior | i mean the CLI asterisk one |
| 00:05.34 | fetcher | and 'set verbose 4' from the CLI to get the equivalent of -vvvv |
| 00:05.38 | Strom_C | mavior, exit |
| 00:06.04 | mavior | exit or quit gives me something like "command not found" |
| 00:06.11 | Strom_C | mavior, I would seriously suggest you read the documentation |
| 00:06.13 | nick125_lappy | Anyone here order from voxilla before? |
| 00:06.26 | JT | just use ctrl + c then |
| 00:06.37 | mavior | ok |
| 00:06.43 | mavior | thank you guysù |
| 00:06.44 | CrashHD | Strom_C: it's evaluating to 0 but ${SCREEN_FILE} is not set |
| 00:07.15 | Strom_C | pastebin your new extension |
| 00:07.50 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 00:07.53 | CrashHD | http://www.pastebin.ca/316804 |
| 00:08.25 | CrashHD | ahh |
| 00:08.26 | CrashHD | 2nd line |
| 00:08.28 | CrashHD | SetVar |
| 00:08.47 | Strom_C | yeah, why the hell are you using SetVar? |
| 00:09.06 | CrashHD | retarded copy/paste from an old app |
| 00:09.12 | CrashHD | fixed |
| 00:09.13 | *** join/#asterisk dwmw2_gone (i=ctrlprox@baythorne.infradead.org) |
| 00:09.14 | CrashHD | seems ok now |
| 00:09.23 | CrashHD | thanks for the extra set of eyes |
| 00:09.47 | CrashHD | is asterisk going quote sensitive? |
| 00:10.22 | CrashHD | seemed as long as there was no spacing I could get away without quotes |
| 00:10.26 | CrashHD | that no longer the case? |
| 00:11.36 | *** join/#asterisk lunaphyte (n=lunaphyt@static-71-120-128-10.gdrpmi.dsl-w.verizon.net) |
| 00:12.46 | *** join/#asterisk lunaphyte (n=lunaphyt@static-71-120-128-10.gdrpmi.dsl-w.verizon.net) |
| 00:13.09 | *** join/#asterisk coppice (n=chatzill@129.168.17.210.dyn.pacific.net.hk) |
| 00:14.47 | anonymouz666 | why PAP2 has an option called Jitter Buffer "up and down"? JB isn't only on incoming packets? |
| 00:14.54 | *** join/#asterisk yassine (n=yassine@xdsl-84-44-152-79.netcologne.de) |
| 00:14.57 | yassine | hi everyone |
| 00:15.25 | *** join/#asterisk fnordus (n=dnall@24.85.128.203) |
| 00:15.48 | yassine | lspci shows me : 01:07.0 Communication controller: Motorola Wildcard X100P but ztconfig claims this : line 0: Unable to open master device '/dev/zap/ctl' |
| 00:15.51 | yassine | any ideas ? |
| 00:17.52 | *** join/#asterisk X-Rob (n=rob-x@CPE-61-9-217-229.static.qld.bigpond.net.au) |
| 00:17.55 | nick125_lappy | Anyone have any ideas on how to do LCR and Call billing? |
| 00:18.29 | JT | lcr is acheivable in the dialplan |
| 00:18.45 | JT | call billing, you'll need to look into using the CDRs |
| 00:18.50 | nick125_lappy | Doing both is where I'm getting confused though |
| 00:20.52 | SwK | dont look at it is doing both |
| 00:20.58 | SwK | look at it as doing one then the other |
| 00:21.22 | SwK | LCR doesnt have to do with billing unless you are adjusting the rates based on the termination route chosen |
| 00:21.40 | nick125_lappy | SwK: that's basically what I need to do.. |
| 00:22.09 | SwK | then its still not a problem the outbound route is in the DR |
| 00:22.11 | SwK | err CDR |
| 00:22.17 | KuJaX | Is there a bootable project out there that someone did that installs linux and all of the appropriate stuff for asterisk and NOTHING else? |
| 00:22.31 | CrashHD | does Read() in 1.4 accept verbal numbers said or only dtmf? |
| 00:22.33 | SwK | kujax: asteriskNow, trixbox |
| 00:22.38 | nick125_lappy | (I'm setting up a small voip setup for myself, my family, and, a few of my friends, and I need to know the rate that they are using) |
| 00:22.54 | KuJaX | SwK: seen Trixbox, way too many features (GUI). Does AsteriskNow have a GUI? I want to keep it simple. |
| 00:23.04 | SwK | they both have a gui |
| 00:23.25 | SwK | you arent going to find one that doesnt have a gui |
| 00:23.25 | CrashHD | nm |
| 00:23.31 | JT | wrong |
| 00:23.33 | JT | astlinux |
| 00:23.34 | SwK | just do a base gentoo install then install just asterisk on that |
| 00:23.36 | KuJaX | Swk: Easy way to disable the GUI by simply overwriting all of the CONF files? Or more to it then that? |
| 00:23.46 | *** join/#asterisk bagpuss_thecat (n=bagpuss_@lodge.glasgownet.com) |
| 00:23.54 | SwK | jt didnt kristin add a gui to that? |
| 00:23.56 | Qwell | KuJaX: You don't need to use the GUI in asterisknow |
| 00:24.09 | KuJaX | Qwell: Do you know what linux distro it uses and php version? |
| 00:24.12 | JT | KuJaX: gui will overwrite conf files unless disabled |
| 00:24.13 | bagpuss_thecat | evening all |
| 00:24.17 | JT | but yeah |
| 00:24.21 | Qwell | rpath, and probably doesn't have php |
| 00:24.33 | JT | i don't see what's so hard about installing a linux distro then asterisk |
| 00:25.02 | KuJaX | JT: Nothing is, did it already today with CentOS 4.4. However, I also need SugarCRM to work off of the same system, and I couldn't get the right PHP version (centOS repository didn't have it) |
| 00:26.22 | JT | compile :) |
| 00:26.29 | KuJaX | don't know how. :) |
| 00:26.43 | KuJaX | Trixbox is great, but I don't want all of the GUI and behind the scenes (trixbox includes a working sugarcrm) |
| 00:27.25 | JT | compiling stuff isn't hard |
| 00:27.40 | KuJaX | JT: I don't know how and I need this thing up tomorrow. :( lol |
| 00:27.41 | JT | read the INSTALL or README file in the base directory of an extracted tarball |
| 00:27.59 | KuJaX | JT: Interested in some money for services? :) |
| 00:28.35 | JT | i'm not familiar with CentOS and might be too far away for your liking |
| 00:29.30 | KuJaX | JT: I'll use whatever you want to use, I just need basic Asterisk installed correctly (I can develop the dialplan to just ring to one phone until this next weekend). And then SugarCRM installed: My brother just picked up a Panasonic 42 inch LCD TV of which has a SD Card reader in the front of it. It is pretty neat, you can plug in a SD card, and use the remote to browse through the images on the SD card and you can also setup a slideshow wit |
| 00:29.30 | KuJaX | I went to BestBuy the other day to check out on this technology and found that they only had one model at BestBuy with USB OR SD Card reader, and it was made by Panasonic and it was a 50 inch. When I spoke with two BestBuy home theare employees about it, they had no idea what I was talking about. |
| 00:29.30 | KuJaX | Basically I was thinking of doing this with a similar 42 inch LCD TV which doesn't have a USB nor SD card reader, but it has the usual, S-Video ports and VGA or DVI port to plug a PC in. |
| 00:29.33 | KuJaX | My question is this: |
| 00:29.35 | KuJaX | Is there a device out there that is small that you can plug into a normal TV to play the digital media on the device, such as a SD, Flash, Compact Flash, ETC? Something that uses S-Video or VGA/DVI cable device? I think it would be a cool effect for a local wedding I am helping out with here shortly where we can just put the bride and grooms images on a SD card and then put onto their large 50 LCD TV they have in their living room without actu |
| 00:29.41 | KuJaX | Whohps. |
| 00:29.50 | KuJaX | Lol that got mangled. |
| 00:29.51 | KuJaX | http://www.sugarcrm.com/wiki/index.php?title=Platform_Requirements_and_Configurations |
| 00:30.05 | *** join/#asterisk cygar (n=cygar@201.37.249.222) |
| 00:30.09 | KuJaX | JT: Need basic asterisk and zaptel installed, and then sugarcrm working. Above is a link to the sugar requirements. |
| 00:31.29 | JT | i think you can buy devices that read flash devices and output to vga or similar on ebay |
| 00:31.40 | KuJaX | JT: That is another subject. :) hehe |
| 00:31.45 | KuJaX | i'll check that out though. thanks |
| 00:33.35 | cygar | I'm facing a dead lock with asterisk using 99% of CPU , I have compiled it with "-g" and used gdb doing a backtrace to try to find out where it's getting stuck... there's almost 30 threads running... my question is how can I know what's the thread that's using all the CPU ? [ i know it's not an * question but i'm trying to find a bug on it] |
| 00:34.03 | nick125_lappy | Dangit, there used to be a place that had free hold music (mostly public domain stuff), but, I can't remember where it is, anyone here know of what I'm talking about? |
| 00:34.22 | Qwell | nick125_lappy: freeplaymusic? |
| 00:35.09 | *** join/#asterisk BSDTech (n=RNeese@ppp-69-238-75-25.dsl.irvnca.pacbell.net) |
| 00:35.34 | nick125_lappy | Qwell: I don't think that's it.. |
| 00:40.28 | DrukenLPY | anyone got experince with protel payphones? |
| 00:41.09 | aptura | Anyone know of software that can allow remote activation of door solinoids ect and work with asterisk? |
| 00:41.11 | JT | nick125_lappy: "royalty free" |
| 00:41.59 | *** join/#asterisk zimdog (n=zimdog@c-24-9-69-162.hsd1.co.comcast.net) |
| 00:42.01 | robin_sz | sigh ... OK, so any bright ideas on how to make mISDN work acceptably with DTMF? |
| 00:42.14 | zimdog | #freepbx |
| 00:42.27 | zimdog | \join #freepbx |
| 00:42.50 | JT | mm, i never had any luck with getting mISDN working acceptably |
| 00:42.52 | robin_sz | try a / instead of \ |
| 00:43.09 | robin_sz | JT, it works fine ... except for DTMF tone detection |
| 00:43.21 | JT | it doesn't work fine for me |
| 00:43.30 | robin_sz | what card? |
| 00:43.38 | JT | it's pretty alpha level |
| 00:43.42 | JT | junghanns octoBRI |
| 00:44.05 | *** part/#asterisk mog (n=mog@c-71-207-215-93.hsd1.al.comcast.net) |
| 00:44.08 | robin_sz | im just using a HFC/Cologne card |
| 00:44.11 | robin_sz | seems OK |
| 00:44.19 | JT | it's also quite immature |
| 00:44.24 | robin_sz | it is. |
| 00:44.31 | JT | no group dial on NT ports, ridiculous, imho |
| 00:44.52 | robin_sz | group dial? |
| 00:45.07 | JT | yeah, dialling a group instead of a line |
| 00:45.11 | JT | like G1 |
| 00:45.17 | robin_sz | right |
| 00:45.36 | robin_sz | so it just picks an unused channel? |
| 00:45.56 | robin_sz | actually, that seems to "just work" for me ... |
| 00:46.28 | robin_sz | admittedly I only have 2 channels, but it just seems to find the unused one and dial out on that |
| 00:46.34 | JT | you probably won't experience the problem if you only use TE mode, or use a small card |
| 00:46.55 | robin_sz | NT mode is where you connect phoens to the card right? |
| 00:46.56 | aptura | Does anyone know of a outdoor phone intercom that can can have its signaling pass across common house wiring? |
| 00:47.00 | JT | what do you use to make it do that? even the readme says group dial on NT ports is unsupported |
| 00:47.01 | KuJaX | Can someone help me install Asterisk on a linux distro that doesn't have YUM? |
| 00:47.03 | JT | yes |
| 00:47.22 | KuJaX | There are some requirements that are missing when I tried to install Asterisk and Zaptel. |
| 00:47.23 | robin_sz | i only use TE mode then |
| 00:47.29 | JT | ah |
| 00:47.45 | robin_sz | KuJaX, which distro? |
| 00:48.21 | KuJaX | robin_sz, I have no idea, I downloaded it off of SugarCRM's website. It installed SugarCRM, php, mysql and apache. |
| 00:48.26 | KuJaX | How can I determine what distro it is? |
| 00:48.52 | robin_sz | does it have apt-get? |
| 00:48.58 | KuJaX | let me check.. |
| 00:49.02 | robin_sz | uname -a might help too |
| 00:49.59 | KuJaX | uanem -a "Linux localhost.localdomain 2.6.17.14-0.3.x86.i686.cmov |
| 00:50.16 | robin_sz | and doe it have apt-get? |
| 00:50.27 | KuJaX | Command not found |
| 00:50.34 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 00:50.36 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 00:51.15 | JT | KuJaX: thought you already said you had centos |
| 00:51.28 | robin_sz | well, no idea then ... if it doesnt have apt-get you might as well delete it anyway |
| 00:51.42 | KuJaX | JT: Did, but it didn't have the correct PHP version I needed. It seems that SugarCRM is harder to setup than Asterisk. So now I am starting with a distro that SugarCRM loves. |
| 00:52.06 | JT | do you know what that distro is? |
| 00:52.19 | KuJaX | not a clue, it was a rpath linux installer. |
| 00:52.35 | JT | trixbox? |
| 00:52.36 | robin_sz | well, that should make installing * a bit of a git then :) |
| 00:52.58 | KuJaX | JT: Trixbox is an issue, because I don't want FreePBX or any of the GUI, it has jacked me before (unstable) |
| 00:53.09 | JT | so is that what you have now? |
| 00:53.21 | KuJaX | Yeah, and it doesn't work properly |
| 00:53.26 | KuJaX | Asterisk will shutdown for no reason out of no where |
| 00:53.30 | JT | trixbox uses centos |
| 00:53.41 | KuJaX | right... |
| 00:53.52 | KuJaX | and I have another hard drive that has this SugarCRM rpath installer on it that doesn't have apt-get or YUM |
| 00:53.59 | KuJaX | for testing purposes (trying to go to a new platform) |
| 00:54.14 | robin_sz | I tried sugar crm ... waste of space in my opinion |
| 00:54.16 | KuJaX | all I want is Asterisk and SugarCRM. I am about to go and just buy the Linksys SP9000 PBX. lol :) |
| 00:54.24 | KuJaX | robin_sz: It works in my business structure very well. |
| 00:54.32 | robin_sz | you already use it? |
| 00:54.41 | KuJaX | yes |
| 00:54.53 | robin_sz | we tried it ... found it way too heavy |
| 00:55.22 | KuJaX | What do you use now? |
| 00:55.55 | anonymouz666 | SugarCRM ou Vtiger? |
| 00:56.09 | KuJaX | What other distro's work very well with Asterisk? I have tried CentOS, can't update PHP to the version I need, so through that out. Tried Debian, wouldn't detect NIC, through that out. |
| 00:56.26 | JT | debian is fine |
| 00:56.48 | JT | nic detection is unnecessary |
| 00:57.31 | KuJaX | It wouldn't work JT. I even tried a millino things in the #debian and #linux channel and no one could figure it out |
| 00:57.49 | JT | what sort of nic is it? |
| 00:57.54 | robin_sz | KuJaX, we use xrms now, works well for us |
| 00:58.03 | KuJaX | it detected it just fine, but it wouldn't talk to the DHCP server. They concluded that the cat5 wasn't plugged in (lol, even though I swap hard drive in there to a CentOS version and it works just fine) |
| 00:58.07 | aptura | KuJaX I had issues with debian not enabling asterisk apon bootup. Gave up and went fedora. |
| 00:58.26 | KuJaX | fedora.redhat.com ? |
| 00:58.34 | JT | aptura: did you use the debian packaged version of asterisk? |
| 00:58.35 | robin_sz | ? |
| 00:58.36 | KuJaX | err fedoraproject.org |
| 00:58.37 | aptura | Any comments on netlink wireless wifi phones? |
| 00:58.49 | aptura | KuJaX dont recall it was long ago. |
| 00:58.51 | robin_sz | all wifi phoens suck |
| 00:59.14 | aptura | robin_sz mmm really have you used them all in testing???? |
| 00:59.22 | aptura | sarcastic comment :) |
| 00:59.23 | JT | KuJaX: what sort of night? |
| 00:59.30 | *** join/#asterisk RoyK (n=roy@217-175-39.100710.adsl.tele2.no) |
| 00:59.37 | KuJaX | JT: What do you mean? |
| 00:59.40 | robin_sz | aptura, no i have used 2 sorts and now know why they suck |
| 00:59.41 | JT | aptura: the concept just seems a bit lacking |
| 00:59.47 | JT | KuJaX: NIC sorry, falling asleep ;) |
| 00:59.54 | JT | i wish it was night |
| 01:00.09 | KuJaX | JT: Intel onboard. |
| 01:00.09 | robin_sz | the big problem with wifi is jitter |
| 01:00.14 | KuJaX | It found it just fine, just wouldn't detect the network |
| 01:00.17 | robin_sz | its inherent in the topology |
| 01:00.31 | aptura | I would not put the blame on the phone nessesarly. It could be the fact the signaling from wifi is weak and does not pass though walls easily. |
| 01:00.37 | robin_sz | KuJaX, welll confihure manually then |
| 01:00.52 | JT | it could be the fact that wifi is inappropriate for reliable voip |
| 01:00.58 | KuJaX | robin_sz: We did, we edited the conf, we set it static IP, we set it to DHCP. IT wouldn't read for some reason. |
| 01:01.05 | JT | robin_sz was blaming the technology not the phone |
| 01:01.26 | robin_sz | aptura, which part of "inherent in the topology" dont you understand? |
| 01:01.59 | robin_sz | wifi basically relies on randomising its time slots to avoid collisions with other dvices in the channel |
| 01:02.14 | robin_sz | thus you get a lot of jitter |
| 01:02.16 | aptura | I figure it the low Dbm of the tranciver base that is a issue when trying to pass the signaling around HVAC ducting. |
| 01:02.21 | robin_sz | and thats baaaaaad |
| 01:02.47 | robin_sz | secondly, all wifi phones are crap |
| 01:03.00 | aptura | :) |
| 01:03.01 | robin_sz | they have a pocket life of about 8 hours |
| 01:03.05 | aptura | Well wimax has kicked off in seattle. |
| 01:03.34 | robin_sz | wimax? its a phone? |
| 01:03.41 | BSDTech | can the webserver built into asterisk serve up cgi files |
| 01:03.41 | aptura | wimax.org |
| 01:03.45 | BSDTech | or only html |
| 01:03.48 | JT | another wireless standard |
| 01:04.04 | JT | asterisk has a built in webserver? |
| 01:04.08 | robin_sz | sigh ... its not even fscking wifi |
| 01:04.09 | BSDTech | yes |
| 01:04.36 | JT | what is the webserver for? |
| 01:04.37 | BSDTech | its how they serve the digium asterisk gui |
| 01:04.52 | BSDTech | for the webgui they are working on |
| 01:04.59 | JT | most people don't use that |
| 01:05.10 | JT | sounds like you're talking about asterisknow, not asterisk |
| 01:05.28 | BSDTech | asterisk has the server built in |
| 01:05.36 | JT | from what version? |
| 01:05.43 | robin_sz | mine doesnt |
| 01:05.43 | BSDTech | according to everything I was told here amonth ago |
| 01:05.53 | JT | asterisknow might |
| 01:06.34 | BSDTech | I wish there was a way to serve cgi files with out a full html serve |
| 01:07.42 | robin_sz | ? |
| 01:07.52 | robin_sz | just install apache, whats so hard about that? |
| 01:08.22 | WAudette | I think he is working on towards an embedded system where resources are tight. |
| 01:09.15 | robin_sz | well, cgis arent goingto help him much then |
| 01:09.32 | *** join/#asterisk Mattwj2005 (n=Matt@c-76-17-131-68.hsd1.mn.comcast.net) |
| 01:09.34 | robin_sz | unless he has room for Perl or whatever |
| 01:09.42 | Mattwj2005 | guys I am so happy :) |
| 01:10.04 | robin_sz | rban WiMAX, A London based operator using the 4.9 and 5.8GHz bands selling up to 10Mbps symmetrical access to businesses and next day delivery. |
| 01:10.08 | Mattwj2005 | I for the first time got meetme working using Trixbox or AsteriskNOW |
| 01:10.09 | Mattwj2005 | :) |
| 01:10.20 | Mattwj2005 | *working without |
| 01:10.20 | robin_sz | what do they mean "next day delivery"? |
| 01:10.49 | JT | robin_sz: modem arrives, you unpack it, plug it in, starts to work |
| 01:10.49 | robin_sz | you request a packet from a remote machine and it turns up next day? |
| 01:10.53 | DrukenLPY | anyone know if alison is still available for recordings? |
| 01:10.58 | JT | don't wait for techs to install a cabled service |
| 01:11.08 | *** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 01:11.41 | TripleFFFF | <PROTECTED> |
| 01:11.41 | robin_sz | oh, I assumed it was talking about packets delivered across the network :) |
| 01:12.48 | anonymouz666 | "I am sorry that's not a valid extension" |
| 01:12.58 | robin_sz | and ... ive yet to see a wifi phoen that can keep itself allive all day in my pocket |
| 01:13.48 | DrukenLPY | robin_sz: how much use? |
| 01:14.20 | robin_sz | matters not ... |
| 01:14.28 | robin_sz | say 5 minutes |
| 01:14.49 | TripleFFFF | hm this not good |
| 01:14.49 | BSDTech | it needs to be more securw |
| 01:14.58 | BSDTech | and as small as possible |
| 01:15.03 | TripleFFFF | - Executing Dial("SIP/frank1232-cae3", "SIP/asdadsasd|20") |
| 01:15.04 | TripleFFFF | <PROTECTED> |
| 01:15.04 | TripleFFFF | noone answered in 20000 ms |
| 01:15.19 | TripleFFFF | why not return with peer not exist ? when a t/o |
| 01:15.51 | TripleFFFF | even without a t/o |
| 01:16.01 | robin_sz | BSDTech, http://www.mathopd.org/ |
| 01:17.42 | robin_sz | DrukenLPY, so, go on then, do you know of any WiFi phone that can stay alive for 10 hours or so in my pocket?? [yes|no] |
| 01:18.18 | BSDTech | hmm ok |
| 01:18.47 | TripleFFFF | just not front pockets..wifi is bad for balls |
| 01:19.08 | robin_sz | why is that? |
| 01:19.44 | robin_sz | and if you say "because of the radio transmisions" Im afraid I shall have to ridicule you |
| 01:19.52 | TripleFFFF | !google wifi health risks |
| 01:20.14 | TripleFFFF | you know what ? you have a pap2 ? right ? |
| 01:20.33 | TripleFFFF | open a line to where ever.. then ring your cell and put it near it.. or near your TV.. |
| 01:20.51 | robin_sz | sigh ... |
| 01:20.54 | TripleFFFF | if you think its harmeles.. then you know microwaves would be good for you too.. stick your head in it |
| 01:21.06 | robin_sz | sigh |
| 01:21.15 | robin_sz | question for you: |
| 01:21.38 | robin_sz | what is the output power of a WiFi card? what is the output power of a cell phone? |
| 01:22.12 | robin_sz | answer, wifi about 20mw |
| 01:22.17 | robin_sz | phone about 2W |
| 01:22.36 | TripleFFFF | hey.. power is not all |
| 01:22.39 | robin_sz | cell phoens are 100 times the power |
| 01:22.43 | TripleFFFF | just look at mythbusters last night test |
| 01:22.53 | robin_sz | theatre |
| 01:22.55 | TripleFFFF | and wifi = 250mw.. |
| 01:23.01 | TripleFFFF | boostable via nay firmware. |
| 01:23.11 | TripleFFFF | yeah it all theater |
| 01:23.12 | TripleFFFF | ;) |
| 01:23.14 | *** part/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 01:23.56 | *** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 01:24.01 | TripleFFFF | sorry got flunked |
| 01:24.02 | TripleFFFF | A Federal standard limits the amount of microwaves that can leak from an oven throughout its lifetime to 5 milliwatts (mW) of microwave radiation per square centimeter at approximately 2 inches from the oven surface. This limit is far below the level known to harm people. Microwave energy also decreases dramatically as you |
| 01:24.12 | TripleFFFF | http://www.fda.gov/cdrh/consumer/microwave.html |
| 01:24.35 | robin_sz | well, thats bollocks for a start |
| 01:24.43 | TripleFFFF | so we talking 5 miliwat.. per square centimeter.. your clel is up to your head and is 4 times that .. and for 5 to 50 minutes |
| 01:24.44 | robin_sz | ovens leak WAY more than that |
| 01:24.50 | TripleFFFF | that the LEAK |
| 01:24.53 | robin_sz | ovens leak WAY more than that |
| 01:24.54 | TripleFFFF | allowed for ovens |
| 01:24.56 | robin_sz | ovens leak WAY more than that |
| 01:25.04 | TripleFFFF | LEAK = even less authorized then your cell emits |
| 01:25.08 | Mattwj2005 | it is only 5 milliwatts? |
| 01:25.08 | TripleFFFF | exact.. |
| 01:25.20 | TripleFFFF | so go put your head by the oven for 1 year.. |
| 01:25.25 | TripleFFFF | and ill meet your kids.. |
| 01:25.41 | TripleFFFF | that FDA.. so its not a fucking parano website |
| 01:25.45 | Mattwj2005 | oh that must be energy that leaks out |
| 01:26.02 | CrashHD | can someone answer this: (convo pasted) |
| 01:26.03 | CrashHD | http://www.pastebin.ca/index.php |
| 01:26.10 | Mattwj2005 | inside is like 900 Watts though |
| 01:26.13 | robin_sz | nah, thats bollocks, ovens leak way more than 5mw, go get a neon bulb, cut the leads to 1/4 wave at 2.540 ghz and put it near the door, it lights up :) |
| 01:26.16 | TripleFFFF | yeah |
| 01:26.22 | TripleFFFF | inside you surely dont want to be |
| 01:27.08 | robl^ | if you put a small pet inside of a microwave, they light up too |
| 01:27.08 | robin_sz | cell phoens are fairly worrying, 2W next to your brain, but the health risks appear to be small, but they do exist |
| 01:27.09 | TripleFFFF | alot of china companies are going away from wifi.. they where getting sicker and sicker and the policy is work so they saw in reports it did have lessen the sick employees by removinglol |
| 01:27.51 | JT | sounds like quackery |
| 01:27.52 | TripleFFFF | just RING your cell near TV.. now get a medical degree and learn how your spinal system works.nerve system and brain , its all electric impulses.. . |
| 01:28.12 | TripleFFFF | if you do 1 day of electrical eng..youll learn that maggnetic fields DISRUPT electric current |
| 01:28.22 | JT | your shouting and blind handwaving prooves you have no medical degree |
| 01:28.33 | Strom_C | pseudoscience wins again! |
| 01:28.37 | robin_sz | why do you keep coming back to CELL phones to prove that WIFI os bad? |
| 01:28.39 | TripleFFFF | yep never said i did.. its just commen sense.. |
| 01:28.46 | *** part/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 01:28.50 | robin_sz | what an idot |
| 01:28.53 | JT | unfortunately, most people are fucking stupid |
| 01:28.57 | robin_sz | seems so |
| 01:28.59 | JT | so common sense is not very common |
| 01:29.05 | robin_sz | now, thats true. |
| 01:29.19 | Mattwj2005 | how do I configure Asterisk for an FXO or FXS |
| 01:29.33 | Mattwj2005 | the one that connects to the public phone network? |
| 01:29.55 | robin_sz | strangely enough I was about to tell him that I have a degree in electronic engineering, a ham license and my wife is an intnesive care nurse ... ;) |
| 01:30.29 | robin_sz | Mattwj2005, you'll need an FXO card or a shoky intel modem that look like an FXO card anyway |
| 01:30.37 | robin_sz | shonky |
| 01:30.38 | *** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 01:30.41 | TripleFFFF | also Less is known about what happens to people exposed to low levels of microwaves. Controlled, long-term studies involving large numbers of people have not been conducted to assess the impact of low level microwave energy on humans. M |
| 01:30.50 | Mattwj2005 | yeah I have a modem clone.....I have zaptel installed |
| 01:31.13 | Mattwj2005 | if someone has a nice web link I would really appreciate it :) |
| 01:31.28 | robin_sz | Mattwj2005, have you configured zaptel? |
| 01:31.46 | TripleFFFF | actually i dont really care just pointing out that im neither on one side or other , but prefer to hmm see all sides |
| 01:31.49 | BSDTech | why did gnomephone die off |
| 01:31.52 | Mattwj2005 | I just installed zaptel....no real configuration |
| 01:32.00 | BSDTech | gnophone sorry |
| 01:33.42 | robin_sz | Mattwj2005, edit your /etc/zaptel.conf |
| 01:33.50 | robin_sz | Mattwj2005, and run zttest |
| 01:34.20 | Mattwj2005 | okay then what.....extension.conf somewhere? |
| 01:35.07 | robin_sz | when and only when you can get that working and looking like its happy, consider editing /etc/asterisk/zapat.conf to use the zaptel devices |
| 01:35.18 | robin_sz | zapata.conf |
| 01:35.32 | Mattwj2005 | okay |
| 01:35.36 | Mattwj2005 | thank you :) |
| 01:36.30 | robin_sz | theres a config generator for zaptel somewhere that sometimes works perfectly ;) |
| 01:36.33 | robin_sz | sometimes |
| 01:36.48 | *** join/#asterisk Mad|Cow (i=Mad_@74.92.109.201) |
| 01:38.01 | Mad|Cow | Does anyone know if RTPproxy can work with a NAT'd connection? I'm getting one way audio with SER, I'm hoping someone out there might have figured out a way around this |
| 01:38.31 | Mattwj2005 | where is zttest? |
| 01:38.46 | Mattwj2005 | I see ztcfg and zttool |
| 01:38.54 | robin_sz | try zttool |
| 01:39.06 | robin_sz | thats the config viewer thing |
| 01:39.24 | Mattwj2005 | ok |
| 01:39.37 | robin_sz | zttest is a speed and jitter test thing ... you need that to be stable and better than97% ... |
| 01:39.50 | robin_sz | less than 97% and the audio is unuseable |
| 01:40.22 | robin_sz | try running the test and dicking around with alll the apps on the machien if it runs as a webserver etc to see if it croaks |
| 01:40.39 | Mattwj2005 | okay |
| 01:40.41 | *** part/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 01:41.58 | robin_sz | TripleFFFF, well, im going to log off now, im pretty worried about the induced voltages from the charger lead on the laptop giving me CANCER or the dangerous radiation from the LCD display making my hair fall out. its common sense. |
| 01:42.03 | Mattwj2005 | it says unconfigured |
| 01:42.14 | robin_sz | get configuring then :) |
| 01:42.16 | BSDTech | get real |
| 01:42.22 | Mattwj2005 | ztcfg? |
| 01:42.37 | robin_sz | after editing /etc/zaptel.conf yes |
| 01:43.36 | robin_sz | I assume you have an /etc/init.d/zaptel script? |
| 01:43.56 | robin_sz | oh, night .. bedtiem |
| 01:44.33 | Mattwj2005 | yeah I do |
| 01:44.39 | Mattwj2005 | thanks you have a good night |
| 01:46.14 | CrashHD | does 1.4 have a voicemail detection function? |
| 01:46.30 | CrashHD | so channels are not connected unless an actual person answers? |
| 01:46.57 | JT | that sounds pretty technically unfeasible |
| 01:47.04 | CrashHD | I was thinking that |
| 01:47.06 | CrashHD | but I remember |
| 01:47.06 | JT | unless the voicemail service has a special tone |
| 01:47.13 | CrashHD | reading somewhere in the 1.4 docs |
| 01:47.18 | CrashHD | it was a function or new feature or something |
| 01:48.08 | Mattwj2005 | zttool sees the card as a "Generic Clone Board 1" |
| 01:48.23 | Mattwj2005 | it is just a modem |
| 01:49.08 | Mattwj2005 | should I configure like how I would configure a Wildcard X100P |
| 01:49.09 | Mattwj2005 | ? |
| 01:49.14 | Mad|Cow | Does anyone have any experience with SER/RTPproxy? My internal Asterisk server is replying back to my SIP client (on the internet) with it's internal IP address. As a result, I have one way audio becuase my SIP client on the i-net cant communicate with Asterisk's internal IP. Can I rewrite the SDP URI within SER? |
| 01:50.33 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 01:50.33 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 01:52.13 | *** join/#asterisk paavum (n=dorphals@pcsp163-73.supercabletv.net.co) |
| 01:52.31 | paavum | Hello |
| 01:52.41 | paavum | I'm trying to connect a remote * to my box |
| 01:52.53 | paavum | when I turn debug on I can see it hitting my box |
| 01:53.05 | paavum | but the remote end says "all circuits are busy" |
| 01:53.18 | paavum | I have the debug output here |
| 01:53.30 | paavum | could anyone help me out? |
| 01:54.43 | Strom_C | paavum, are you using iax or sip? |
| 01:55.16 | paavum | sip |
| 01:55.36 | paavum | (actually I've tried both, but right now I'm using SIP) |
| 01:55.58 | *** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net) |
| 01:55.58 | Strom_C | let's try IAX; it's simpler |
| 01:56.14 | paavum | Ok lemme change the trunk to IAX |
| 01:56.17 | Strom_C | pastebin the extensions.conf and iax.conf for both machines |
| 01:56.56 | paavum | ure gonna hate me, but one of the machines is freepbx :sç |
| 01:57.19 | Strom_C | I can't debug that |
| 01:57.20 | Strom_C | sorry |
| 01:57.24 | Strom_C | go to #freepbx |
| 01:57.26 | paavum | yes |
| 01:57.34 | paavum | but I think the problem is on my end |
| 01:57.39 | paavum | which is standard asterisk |
| 01:57.45 | Strom_C | ok, pastebin that then |
| 02:01.34 | *** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com) |
| 02:03.25 | paavum | http://www.pastebin.ca/316878 <-- IAX.conf |
| 02:04.46 | paavum | http://www.pastebin.ca/316879 <-- extensions.conf |
| 02:04.51 | Strom_C | and what about the iax.conf entry on the remote machine? |
| 02:05.41 | Strom_C | well, the first problem is that the goto in [chris-in] is wrong |
| 02:05.50 | *** join/#asterisk xlnc2002 (n=root@203-97-110-166.cable.telstraclear.net) |
| 02:06.06 | Strom_C | you must go to [[context,]extension,]priority |
| 02:07.52 | CrashHD | hey storm |
| 02:07.55 | CrashHD | any ideas about this: |
| 02:07.55 | CrashHD | http://www.pastebin.ca/316881 |
| 02:08.01 | CrashHD | (convo pasted) |
| 02:08.08 | Strom_C | and also, there is no such priority as "s" - you must have priority "1" |
| 02:08.13 | [TK]D-Fender | Strom_C : Never shall they get your name right :| |
| 02:08.23 | CrashHD | hah |
| 02:08.31 | CrashHD | my apologies |
| 02:08.53 | CrashHD | my name recognition algorithm must be on the fritz.... |
| 02:09.12 | Strom_C | yeah, and you apparently can't be bothered to use tab-complete |
| 02:09.23 | [TK]D-Fender | CrashHD : and NO, you can't have BOTH ends answer the call. Thats not how Dial works. ONE person gets it. |
| 02:09.48 | [TK]D-Fender | CrashHD : this isn't a MeetMe you know... |
| 02:10.24 | paavum | Strom_C --> Isnt the 's' extension supposed to be for incoming calls? :$ |
| 02:10.35 | Strom_C | you don't have the s extension |
| 02:10.39 | Strom_C | you have the s priorityw |
| 02:10.42 | Strom_C | which makes NO sense |
| 02:10.55 | paavum | D'oh! |
| 02:10.58 | BSDTech | ok its the gui that has the web server in it |
| 02:11.00 | paavum | ure right |
| 02:20.26 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 02:20.28 | *** join/#asterisk robl^ (n=robl@dsl093-025-218.hou1.dsl.speakeasy.net) |
| 02:22.18 | *** join/#asterisk tengulre (n=tengulre@61.185.220.249) |
| 02:24.05 | CrashHD | Strom_C: but dial shouldn't consider the channel answered until the macro is finished executing....? |
| 02:24.21 | Strom_C | CrashHD, huh? |
| 02:24.35 | Strom_C | dial should consider the channel answered when the called party answers |
| 02:24.54 | paavum | Strom_C --> No luck |
| 02:24.59 | paavum | I still get the same msg |
| 02:25.15 | Strom_C | paavum, what do you have in that context now, and what are you dialing from the other system> |
| 02:25.32 | CrashHD | Strom_C: with the M() option |
| 02:26.19 | Strom_C | yeah - one channel answers the call and gets connnected to the macro |
| 02:26.22 | paavum | chris-in] |
| 02:26.22 | paavum | <PROTECTED> |
| 02:28.12 | CrashHD | dial should drop the other channels when a call is bridged...or have that option |
| 02:28.38 | Strom_C | CrashHD, what exactly are you trying to accomplish? |
| 02:29.08 | CrashHD | just a followme find me app that hits multiple people |
| 02:29.15 | CrashHD | I Don't like the new one in 1.4 |
| 02:29.27 | Strom_C | why don't you describe specifically what you're trying to do |
| 02:30.04 | CrashHD | basically a call needs to go out to multiple outside extensions and the first extension that presses 1 to accept the call should have the call |
| 02:30.15 | CrashHD | if a call goes out and connects to say...a cell voicemail |
| 02:30.32 | CrashHD | the other people it may be trying to reach are left unable to take the call |
| 02:31.28 | JT | Strom_C: are you certain on the M() behaviour? |
| 02:31.47 | CrashHD | I will be back in a few minutes |
| 02:31.51 | CrashHD | have to deal with the kids |
| 02:32.15 | Strom_C | CrashHD, that's kind of a.....weird way of doing followme, to put it mildly |
| 02:32.22 | Strom_C | but why not do it this way instead |
| 02:32.31 | Strom_C | put the calling party in a meetme conference |
| 02:32.39 | Strom_C | generate .call files as necessary |
| 02:32.44 | JT | i thought M() executed the macro just prior to bridging |
| 02:32.52 | JT | and the call did not get bridged to the macro |
| 02:33.11 | Strom_C | the first respondent is also connected the meetme; the others lose |
| 02:33.33 | Strom_C | JT, the point I was making is that the macro is only executed for the first caller that answers the ringing phone |
| 02:33.34 | paavum | Strom_C --> Now the [Chris-in] looks like this: exten => _9XXXXXXX,1,goto(LOCAL,${EXTEN},1) |
| 02:33.45 | Strom_C | paavum, what are you dialing from the other system? |
| 02:33.50 | JT | Strom_C: definately, i was just clarifying macro behaviour |
| 02:34.32 | Mattwj2005 | what were you guys talking about with all that jitter and stuff for a Zap channel |
| 02:34.35 | Mattwj2005 | I got it configured |
| 02:34.38 | Strom_C | why the fuck can no one spell "definitely" anymore? |
| 02:35.05 | Mattwj2005 | I defenitiely don't know |
| 02:35.06 | Mattwj2005 | haha |
| 02:37.25 | JT | must be because definitely is mostly used in informal communication so people don't have publications showing them the right way to spell it that they read day to day |
| 02:37.30 | paavum | Strom_C --> 96446800 |
| 02:37.45 | paavum | <PROTECTED> |
| 02:37.46 | Strom_C | paavum, i need the full Dial() statement, not just the extension |
| 02:37.58 | paavum | thats the Cli output |
| 02:38.16 | Strom_C | ok, and what do you have in the iax.conf entry on the calling system> |
| 02:38.17 | Strom_C | ? |
| 02:38.38 | Strom_C | btw, you should never use the "r" option unless you have a very good reason to |
| 02:39.06 | *** join/#asterisk ai-a (n=jake@cpc5-oxfd1-0-0-cust523.oxfd.cable.ntl.com) |
| 02:39.15 | paavum | Strom_C --> http://www.pastebin.ca/316898 |
| 02:39.23 | paavum | Thats the Iax of the caller |
| 02:39.51 | ai-a | my iax2 doesnt seem to be opening its port,, i have used 'make samples' and restarted asterisk, iax2 show users shows guest.. but nmap doesnt show port 4569 open... any ideas ? |
| 02:40.16 | ai-a | when i use my iax client to log in,, it fails. |
| 02:40.40 | Strom_C | paavum, remove qualify=yes from both ends |
| 02:41.58 | *** join/#asterisk lunaphyte (n=lunaphyt@static-71-120-128-10.gdrpmi.dsl-w.verizon.net) |
| 02:42.07 | paavum | Strom_C --> Done, reloaded, called, "All Circuits are busy now" |
| 02:42.29 | paavum | -- Executing Dial("SIP/888-084cdbb0", "IAX2/888/96446800|300|r") in new stack |
| 02:42.30 | paavum | <PROTECTED> |
| 02:42.38 | paavum | That is what the remote system shows |
| 02:43.37 | Strom_C | pastebin an iax debug on both systems |
| 02:44.51 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 02:46.26 | paavum | Strom_C --> http://www.pastebin.ca/316903 <-- IAX debug Remote System |
| 02:46.42 | Strom_C | "remote" is the called system? |
| 02:47.07 | paavum | "remote" is the caller, "local" is the called |
| 02:47.12 | Strom_C | ok |
| 02:47.18 | Strom_C | well the call isnt even setting up |
| 02:47.46 | paavum | and why could that be? |
| 02:47.48 | Strom_C | you /sure/ you took out qualify=yes and did a full reload on both systems? |
| 02:48.03 | paavum | sure but let me do it again |
| 02:48.10 | paavum | I have the smoothqualify in there tho |
| 02:48.14 | paavum | should I take it away as well? |
| 02:48.17 | Strom_C | yes |
| 02:50.22 | paavum | http://www.pastebin.ca/316907 <-- Again, the caller system IAX debug |
| 02:50.33 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 02:50.35 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 02:53.08 | paavum | doing a IAX2 Show peers I get 888 as UNREACHABLE |
| 02:53.19 | Strom_C | you /sure/ you took out qualify=yes and did a full reload on both systems? |
| 02:54.33 | paavum | 100% positive |
| 02:54.50 | Strom_C | just for shits and giggles, try restarting them both |
| 02:54.51 | paavum | Called System is behind a NAT |
| 02:55.11 | Strom_C | restarting asterisk, not rebooting |
| 02:55.23 | paavum | dont worry, I'm not THAT stupiod :P |
| 02:56.58 | paavum | Strom_C --> Working now |
| 02:56.59 | paavum | Thanks |
| 02:57.15 | Strom_C | good |
| 02:57.23 | Strom_C | now get that freepbx shit off your system |
| 02:57.25 | Strom_C | :) |
| 02:57.37 | paavum | Hahaha, I have it off Mine :P |
| 02:58.06 | paavum | I need to get a good Interface with the BASIC features add/remove extens add/remove inbound routes and add/remove follow mes |
| 02:58.15 | paavum | which one would you suggest? |
| 02:58.40 | Strom_C | "inbound routes" |
| 02:58.46 | Strom_C | you've been using freepbx too much |
| 02:59.32 | Strom_C | paavum, why do you need a gui for simple crap like that? |
| 03:00.53 | nick125_lappy | Strom_C: Now now, just because its simple for you... |
| 03:03.09 | [TK]D-Fender | Strom_C : Just for the record, which side was the error on? |
| 03:04.20 | Strom_C | [TK]D-Fender, the calling system |
| 03:04.39 | [TK]D-Fender | Strom_C : Which was which? |
| 03:04.47 | Strom_C | the one with freepbx |
| 03:05.20 | [TK]D-Fender | "In the case of the People vs FreePBX, we find the defendant GUILTY!" |
| 03:06.19 | [TK]D-Fender | ^%$@# FreePBX, the horse it rode in on, and its staunch supporters who all clearly always do everything right and think it's always an * bug. |
| 03:06.31 | *** join/#asterisk DocHolliday (i=RogerRab@gateway/gpg-tor/key-0x0E4F6D6C) |
| 03:06.33 | *** join/#asterisk X-Rob (n=rob-x@CPE-61-9-217-229.static.qld.bigpond.net.au) |
| 03:15.15 | *** join/#asterisk alamantia (n=Anthony@65.4.1.65) |
| 03:15.45 | xheliox | Sing it brotha [TK]D-Fender! |
| 03:17.12 | xheliox | Wow, that's impressive. |
| 03:17.50 | [TK]D-Fender | xheliox : You should hear me on guitar :) |
| 03:18.06 | xheliox | [TK]D-Fender: Me too, we should jam. |
| 03:19.20 | [TK]D-Fender | xheliox : I've been working on sweeping since Christmas, and am doing pretty decent since learing so very little from 1 15 minute video and just playing..... |
| 03:19.58 | xheliox | cool cool. I'm pretty lame, but I suppose that comes as no news. |
| 03:21.37 | [TK]D-Fender | xheliox : c'mon, do share.... |
| 03:22.11 | xheliox | :) |
| 03:23.22 | [TK]D-Fender | xheliox : Here's just a quick crappy scale sample I passed someone else here learning the basics on use of a singe scale : http://aocomputing.net/E-Harmonic-Minor.mp3 |
| 03:23.41 | xheliox | heh |
| 03:23.51 | [TK]D-Fender | xheliox : And a "goofing off" solo I did about 4 years ago : http://aocomputing.net/newguitarsolo.mp3 |
| 03:24.10 | xheliox | spiffyness. |
| 03:25.33 | BSDTech | ok the asterisk-now gui is ok but leaves alot to be desired |
| 03:25.49 | xheliox | Good thing it's just a work in progress then. |
| 03:26.59 | xheliox | Even though guis are for girls and men with non-functioning/missing testicals. I understand the consumer demand. |
| 03:28.15 | nick125_lappy | I used to use AMP.. |
| 03:29.07 | xheliox | Prosecution rests. |
| 03:29.39 | nick125_lappy | I eventually got tired of AMP screwing everything up, and, I permanently punted it from my system |
| 03:30.16 | BSDTech | the fact is that major busniesses want guis and preconfigured dialplans for the most part |
| 03:30.32 | BSDTech | with the pbx ready to go and easy to configure out of the box |
| 03:31.10 | BSDTech | but freepbx is not good for embedded devices |
| 03:31.43 | JT | fact is, most businesses are stupid if that's what they think they need |
| 03:31.47 | JT | traditionally |
| 03:31.53 | JT | when a business gets a pbx |
| 03:31.58 | BSDTech | and there are a few but they all lack in 1 way or another |
| 03:32.05 | JT | the pbx company installs and configures it |
| 03:32.06 | nick125_lappy | They usually hire someone to setup a PBX system.. |
| 03:32.14 | JT | asterisk shouldn't be any different |
| 03:32.18 | [TK]D-Fender | xheliox : Here's a brand new sample : http://aocomputing.net/sweeps1.mp3 |
| 03:32.28 | JT | telephony is not for noobs, especially at a business level |
| 03:32.40 | BSDTech | but alot of comapanis are looking to cute out the technician now days and want self install systems |
| 03:32.50 | *** join/#asterisk DocHolliday (i=RogerRab@gateway/gpg-tor/key-0x0E4F6D6C) |
| 03:33.02 | JT | they have to deal with their own ineptitude then |
| 03:33.15 | BSDTech | where it comes loaded andthey plugin and add the exten and go |
| 03:33.22 | BSDTech | yes |
| 03:33.52 | DocHolliday | is HA the best method these days for asterisk failover/clustering? |
| 03:33.59 | nick125_lappy | BSDTech: Then they deserve what they get with FreePBX or whatever it is this week.. |
| 03:34.25 | file | [TK]D-Fender: ! ! ! |
| 03:34.28 | BSDTech | lol so true |
| 03:34.37 | JT | HA refers to High Availability, not a specific scheme for facilitating it |
| 03:34.54 | [TK]D-Fender | file : I don't want to meet your mom! |
| 03:35.01 | [TK]D-Fender | file : Go download my latest crap! |
| 03:35.10 | file | is that an order? |
| 03:35.37 | [TK]D-Fender | file : No, but the masked men ouside with the truckload of 2L water bottles can be very convincing! |
| 03:36.17 | file | eep |
| 03:36.24 | DocHolliday | sorry heartbeat |
| 03:37.10 | paavum | Strom_C --> Actually, Appealing in favor or FreePBX (which I hate), the error was on my side :P |
| 03:37.20 | paavum | *of |
| 03:37.31 | paavum | which does NOT have freepbx |
| 03:37.40 | DocHolliday | hah |
| 03:37.45 | nick125_lappy | paavum: That still doesn't dispute the fact FreePBX....sucks? |
| 03:38.24 | paavum | Just wanted to speak on the side of justice |
| 03:38.25 | paavum | :P |
| 03:38.46 | paavum | I just need a GUI so I can have an aidiot add and remove exrensions, and "follow-me"'s and DIDs and whatever |
| 03:39.05 | DocHolliday | JT, any thoughts? |
| 03:39.27 | paavum | While I concentrate in the important part. |
| 03:39.35 | paavum | Relaxing and making money :P |
| 03:39.39 | nick125_lappy | I think FreePBX has its place though |
| 03:41.07 | [TK]D-Fender | nick125_lappy : As kindling at the bottom of the real log :) |
| 03:41.27 | nick125_lappy | [TK]D-Fender: Probably so. |
| 03:42.16 | nick125_lappy | Wow...in the last 10 minutes, I've sent/received around 43 text messages |
| 03:42.52 | nick125_lappy | Gee, I surely love unlimited text messaging :-) |
| 03:44.38 | nick125_lappy | My PAP2T-NA should be here on...tuesday or wednesday! |
| 03:44.40 | nick125_lappy | Wewt! |
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| 03:50.33 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 03:56.29 | DocHolliday | so anyone know what the best clustering setup is for asterisk right now? |
| 04:01.34 | *** join/#asterisk Grnd-Wire (n=groundwi@71-217-127-175.tukw.qwest.net) |
| 04:06.15 | nick125_lappy | DocHolliday: for load balancing or reliability? (or both?) |
| 04:07.35 | DocHolliday | nick125_lappy, preferably both.. provided a decent solution exists |
| 04:07.59 | JT | really depends what asterisk is doing |
| 04:09.28 | *** join/#asterisk HondaKid86 (n=veryadam@cpe-70-116-138-105.houston.res.rr.com) |
| 04:10.11 | HondaKid86 | Just wanted to drop by to say you guys rock! |
| 04:10.25 | HondaKid86 | I read about the project and it's really cool. You're making corporate monoliths passe! |
| 04:11.04 | Strom_C | I think playing with telephony is far more interesting than bringing down "the man" |
| 04:11.23 | HondaKid86 | if you're smart enough, i guess! |
| 04:11.31 | robl^ | Strom_C: but using monitor() you can do both ;-) |
| 04:11.38 | Strom_C | hahahaahahaha |
| 04:11.58 | [TK]D-Fender | Strom_C : Unless you're using telephony to bring downt he man! Double-whammy! |
| 04:12.15 | *** part/#asterisk BSDTech (n=RNeese@ppp-69-238-75-25.dsl.irvnca.pacbell.net) |
| 04:12.27 | [TK]D-Fender | robl^ : Damn you and your shorter retorts beating me to the punch! |
| 04:13.21 | robl^ | [TK]D-Fender: I love you too. ;-) |
| 04:14.21 | DocHolliday | nick125_lappy, any ideas? |
| 04:14.33 | paavum | Bye! |
| 04:14.41 | JT | DocHolliday: more info already damnit |
| 04:15.12 | HondaKid86 | Keep up the good work guys... I might drop in from time to time to check in and say hi. :) |
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| 04:15.21 | *** mode/#asterisk [+o russellb] by ChanServ |
| 04:15.37 | DocHolliday | JT, sorry didnt see your comment.. my bad. |
| 04:16.32 | DocHolliday | well about 25-30 IP Phones per location, voice mail, roughly 6 pots lines, conferencing and inter-office calling, with a total of 125 phones |
| 04:16.58 | DocHolliday | via IAX as well as possibly SIP DIDs from a service provider |
| 04:17.05 | DocHolliday | JT, is that enough info? |
| 04:18.21 | JT | hard to do failover on pots lines |
| 04:18.30 | JT | ip phone termination failover is fairly possible |
| 04:19.33 | DocHolliday | i was hoping 1-2 servers per location with 2 central servers |
| 04:20.16 | DocHolliday | yeah *understood* if we switched over to PRIs i guess it would be easier.. but how do you tackle the loadbalancing / failover? |
| 04:20.40 | DocHolliday | traditionally i've used heartbeat for basic stuff, but is there more? |
| 04:21.24 | JT | you can do failover with heartbeat, or load balancing, but that might not take care of the phones on one server if it's down |
| 04:24.05 | DocHolliday | right |
| 04:24.22 | DocHolliday | so it hasn't progressed much then to the level of call manager |
| 04:25.00 | JT | it's up to you to come up with whatever solution you desire |
| 04:26.29 | DocHolliday | JT, well i've decided on asterisk.. just dont want the nasty suprises. |
| 04:39.58 | russellb | there is currently no way to do failover for active calls on a server right now |
| 04:40.14 | russellb | they will be lost |
| 04:41.00 | DocHolliday | russellb, yeah i figured.. kinda sucks. |
| 04:41.23 | russellb | it's an extremely complicated problem to solve |
| 04:41.28 | JT | generally even telco switches can't do failover on active calls if the line card fails |
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| 04:41.45 | russellb | believe me, we've talked about it multiple times ... |
| 04:43.06 | DocHolliday | russellb, i definetely dont disagree that its difficult to solve, just from a marketing standpoint its hard to live with since asterisk doesnt quite have the credibility yet |
| 04:43.32 | DocHolliday | JT, yes but a Lucent 5ESS or DMS100 has an extremely low chance of failing |
| 04:43.45 | JT | DocHolliday: list me one thing that can do live failover on an ACTIVE call that directly terminates the medium? |
| 04:43.52 | JT | sure |
| 04:44.14 | JT | asterisk is not a telco switch, and you're deluding yourself if you think it is :) |
| 04:44.26 | russellb | i think it is! |
| 04:44.28 | russellb | :-p |
| 04:45.29 | DocHolliday | JT, i agree again :) but i'd be happy to use an intermediary device that terminates the call and then a few backend servers that act as redundant handlers |
| 04:45.58 | DocHolliday | yes its a POF but hopefully with a lesser chance of dying a tragic death |
| 04:46.15 | JT | as long as you accept you'll lose active calls |
| 04:46.29 | JT | with load balancing, you'd only lose some active calls |
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| 04:46.48 | DocHolliday | JT, yeah i'd lessen the loss :) |
| 04:47.18 | JT | there's quite a few solutions for PRI failover |
| 04:47.33 | JT | none of them involve not losing active calls however |
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| 04:48.07 | DocHolliday | JT yeah rather unfortunate.. i can live with it. |
| 04:48.26 | JT | you can get external pri failover boxes |
| 04:48.30 | DocHolliday | JT, my other question is do you support centralized interconnect servers or a meshed topology? |
| 04:48.30 | JT | plain Layer 1 |
| 04:48.35 | JT | or TDMoE converters |
| 04:48.35 | DocHolliday | yep :) |
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| 04:49.08 | DocHolliday | heh, none of that does what we are trying to achieve.. |
| 04:49.08 | JT | you'd have to have a pretty big setup to go meshed |
| 04:49.10 | Grnd-Wire | How are we supposed to transfer calls around via an ATA and the flash button? |
| 04:49.20 | JT | what are you trying to acheive? |
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| 04:50.56 | DocHolliday | JT, well we have about 25 phones per location ~ with a total global count of 125, we want interoffice calling.. so the question is do we centralize the exchange point or mesh the machines together |
| 04:51.23 | [TK]D-Fender | Grnd-Wire : Depends on the ATA. Linksys/Sipura's are very well documented and very functional. |
| 04:51.45 | Grnd-Wire | TKD: What about a Grandstream 286? I kinda figured it'd be standard and generic, but it's not eh? |
| 04:52.02 | Qwell | russellb: tomorrow, eh?! |
| 04:52.14 | JT | DocHolliday: centralised would be easier to manage |
| 04:52.53 | DocHolliday | JT, but with international latency am i screwing myself? |
| 04:53.08 | DocHolliday | i.e. Australia --> Canada --> NY .. instead of Australia --> NY |
| 04:53.25 | JT | why not put the core at NY then? |
| 04:53.34 | JT | or LA, even better, if you have an LA locationm |
| 04:53.36 | Qwell | DocHolliday: If you have one machine in NY, and two callers in Australia... |
| 04:53.48 | Qwell | the two callers in Australia would both have to go over the ocean |
| 04:53.49 | Grnd-Wire | TKD: I'm reading the grandstream manual, now that I know it's generic.. :) |
| 04:54.03 | Qwell | If, however, you also had a machine in Oz... |
| 04:54.09 | JT | LA in many respects is the centre of the Internet |
| 04:54.18 | DocHolliday | JT, because there are 5 locations internationally :P that was just an example. |
| 04:54.19 | JT | easy enough to get the .au users to call each other directly |
| 04:54.51 | DocHolliday | JT, doesnt really matter which location you pick.. the connections will come in throughout the globe :) |
| 04:55.13 | DocHolliday | Qwell, no they wont.. there will be an existing box at each location.. |
| 04:55.19 | JT | sure, but if they are offices like you say, people in the same office can call each other directly |
| 04:55.29 | DocHolliday | Australia --> internal Asterisk --> Australia |
| 04:55.34 | DocHolliday | yep |
| 04:55.36 | JT | and if they have to call overseas, it's going to be a problem anyway |
| 04:55.44 | JT | bit of latency |
| 04:55.53 | DocHolliday | JT, but when you consider its nearly 200MS from Australia direct to Canada.. never mind another hop |
| 04:56.04 | JT | it's not that noticable unless you're doing something silly like .au > .us > .au |
| 04:56.05 | [TK]D-Fender | Grnd-Wire : I could have sworn you'd have been warned against touching anything GrandSuck... |
| 04:56.22 | JT | 200ms is fine |
| 04:56.31 | [TK]D-Fender | DocHolliday : My vote is on "mesh". Rather than cascade latency. |
| 04:56.37 | JT | and the one way trip time should be less than 200ms for good routes |
| 04:56.48 | JT | from .au to .ca |
| 04:57.14 | JT | you need > 250ms one way trip time to start to screw peoples' talking up |
| 04:57.42 | DocHolliday | JT, sure |
| 04:58.13 | DocHolliday | good to know. |
| 04:58.34 | DocHolliday | [TK]D-Fender, well we can try both.. if it sounds terrible centralized it doesnt take much work to setup a bunch of IAX connections |
| 04:58.44 | JT | so i don't know if you'd call it meshed, but my vote is for a server at each office, and to only go outside that if necessarily |
| 04:59.08 | DocHolliday | JT, yeah we will do that anyway.. need to demarc the POTS at each location anyway |
| 04:59.28 | JT | where you can go to one or two central servers, or you can just make it route directly to the destination server, depending on locations i guess |
| 05:00.07 | DocHolliday | JT, sounds about right to me.. obviously the cost of 2 servers.. not a huge deal though |
| 05:00.12 | JT | if it gets complicated, it might be easier to use something like dundi, than fixed dialplan logic |
| 05:00.59 | DocHolliday | i dont think it will be too bad, just a lot of repeat configuration |
| 05:02.47 | DocHolliday | the dialplan should be pretty standard, 4xxx (one location), 5xxx (second location) and so on |
| 05:03.03 | Grnd-Wire | TKD: huh? No, I've never been "warned", and I've owned this 286 for years - it's now at a friend's house, and it's the only way he's able to learn asterisk right now |
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| 05:03.23 | JT | yeah but if you add a new loction, all server configurations will need to be updated unless you have a more dynamic system in place |
| 05:03.37 | [TK]D-Fender | Grnd-Wire : Soft phone = free |
| 05:03.51 | battini | x-lite |
| 05:03.52 | Grnd-Wire | TKD: Yea, he's got two of those running as well |
| 05:09.07 | JT | so people can write "abbrv." ;) |
| 05:09.54 | [TK]D-Fender | JT : I blame people's ignorance on auto-complete (or use of shitty clients) |
| 05:10.07 | JT | yeah |
| 05:10.35 | Mattwj2005 | I got free world dialup working |
| 05:11.48 | DocHolliday | JT, no i wont i'll send all calls 4XXX to a specific server |
| 05:11.56 | DocHolliday | if there is a new location just add one more line to each server |
| 05:12.20 | JT | you're contradicting yourself :) |
| 05:12.24 | Qwell | DocHolliday: and when you have 10 servers? |
| 05:12.34 | Qwell | or even 9 |
| 05:12.39 | drray | or even 2 |
| 05:12.44 | DocHolliday | JT, i just meant i wont add each phone to each server |
| 05:12.50 | JT | you're either adding a new line to the config of every server, or you're not |
| 05:12.55 | JT | sure, that's just be stupid |
| 05:13.02 | JT | i meant new office |
| 05:13.09 | DocHolliday | Qwell, i probably will have about 7 heh :( |
| 05:13.19 | DocHolliday | (for voice) |
| 05:13.25 | JT | and it'd be more than one line, as you'll need to modify sip or iax.conf too |
| 05:14.04 | DocHolliday | extensions.conf and iax.conf :) |
| 05:14.25 | DocHolliday | yes it would be a few lines of code.. but nothing too troublesome.. |
| 05:15.15 | DocHolliday | Qwell, i think it would be dumb to add each phone to each config? |
| 05:15.47 | [TK]D-Fender | DocHolliday : Yeah it'd be dumb. That means configuring 150 phones to talk to each server. |
| 05:16.09 | [TK]D-Fender | DocHolliday : Also meaning each phone would ahve to SUPPORT that many registration. |
| 05:16.33 | [TK]D-Fender | DocHolliday : Far better the use site-masks like you showed above. |
| 05:17.11 | [TK]D-Fender | DocHolliday : Though I might extend it so far as use XXYY for site/exten. you're close to breaking the single digit prefix. |
| 05:20.44 | DocHolliday | [TK]D-Fender, yeah i agree.. i just figured users might find it easier to think 4xxx = China 5xxx = US |
| 05:20.48 | *** join/#asterisk paavum (n=dorphals@pcsp163-73.supercabletv.net.co) |
| 05:20.52 | paavum | Hi me again :) |
| 05:20.55 | paavum | this time with a simple Q |
| 05:21.10 | paavum | can I enable sip debugging in only one channel? |
| 05:21.30 | [TK]D-Fender | DocHolliday You can still follow that. 4YXX = china by site... |
| 05:21.47 | CunningPike | paavum: 'sip debug peer <name|ip_address> |
| 05:21.48 | [TK]D-Fender | paavum : "sip debug peer [peername] |
| 05:22.05 | CunningPike | Why doesn't IRC autoclose quotes? |
| 05:22.07 | CunningPike | :) |
| 05:22.11 | [TK]D-Fender | CunningPike : latter should be "sip debug ip [ipaddress]" |
| 05:22.26 | CunningPike | [TK]D-Fender: Right - as always :) |
| 05:22.54 | [TK]D-Fender | CunningPike : No, just enough to pass scrutiny :) |
| 05:23.01 | CunningPike | lol |
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| 05:26.59 | DocHolliday | [TK]D-Fender, yep i can |
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| 05:35.24 | Mattwj2005 | hey guys I am going to take off |
| 05:35.33 | Mattwj2005 | nice talking to everyone and thanks for the help :) |
| 05:35.44 | DocHolliday | [TK]D-Fender, yeah i'd agree with you.. |
| 05:35.49 | *** part/#asterisk Mattwj2005 (n=Matt@c-76-17-131-68.hsd1.mn.comcast.net) |
| 05:36.06 | DocHolliday | *but* still using centralized boxes for the email in addition to voicemail to email |
| 05:37.12 | DocHolliday | there will 2 extensions per phone + intercom.. so the prefix will be used up pretty quick |
| 05:37.39 | [TK]D-Fender | DocHolliday : Why 2 ext's per phone? |
| 05:38.22 | DocHolliday | in case they are busy on line 1, will set to failover to line 2, so they can decide if the call is important |
| 05:39.14 | DocHolliday | line 1, line 2, cell, voicemail |
| 05:39.16 | [TK]D-Fender | DocHolliday : You're understanding is a little backwards. You don't make 2 registrations to do that, you get a phone that can handle multiple calls per reg. |
| 05:39.43 | [TK]D-Fender | DocHolliday : So you're now back to 1 ext per desk. |
| 05:40.16 | DocHolliday | [TK]D-Fender, with the cisco 7940 it has 2 line buttons, you have to register each with the asterisk server? (thats how i have done it in the past) |
| 05:40.44 | [TK]D-Fender | DocHolliday : They aren't really 2 DIFFERENT reg's, they just compound 1 reg. |
| 05:41.12 | DocHolliday | uhoh.. well typically i have registered the phones twice |
| 05:41.13 | [TK]D-Fender | DocHolliday : thats a config "confusion factor". Mind you I would never suggest cisco (with 2 other choices possible). |
| 05:41.32 | [TK]D-Fender | DocHolliday : That should be to the SAME registration, hence not needing 2 #'s to dial in * |
| 05:42.07 | DocHolliday | okay that makes sense. |
| 05:42.33 | DocHolliday | well i dont anticipate anyone actually dialing the second extension.. it just being there to facilitate a second call while the first line is in-use |
| 05:43.39 | [TK]D-Fender | DocHolliday : Have the phones aready been purchased? |
| 05:43.59 | DocHolliday | [TK]D-Fender, nope. |
| 05:44.25 | [TK]D-Fender | DocHolliday : Good, then there's still time to save yourself a lot of greif :) Polycom FTW! |
| 05:44.50 | DocHolliday | why grief? if you mean better phones i'll listen.. but cisco never gives me grief :) |
| 05:47.03 | *** join/#asterisk acehunky (n=chat_jok@203.124.217.98) |
| 05:47.07 | [TK]D-Fender | DocHolliday : Polycom has far better SIP support, typcailly cost a fair bit less, and have better call handling. |
| 05:47.44 | DocHolliday | yeah, [TK]D-Fender whats going to happen is we will probably buy 1 of every (system wide) and test out what we feel works best |
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| 05:48.16 | DocHolliday | *everything :< |
| 05:48.50 | fetcher | [TK]D-Fender: what do you think of Aastra SIP phones? |
| 05:49.13 | CunningPike | DocHolliday: Save yourself the time - we already went through that and Polycom won hands down |
| 05:49.15 | [TK]D-Fender | fetcher : The 480i is a great phone for *, the rest of their series just doesn't make the cut. |
| 05:49.17 | danp | is there a way to increase the volume of the call waiting beep on the polycoms? |
| 05:49.32 | CunningPike | danp: Yes - we actually _increased_ ours lol |
| 05:49.44 | [TK]D-Fender | danp : Sure, you can upload a dirrent WAV to use for it. |
| 05:49.47 | DocHolliday | CunningPike, which phones did you guys go with? |
| 05:49.49 | danp | hmm |
| 05:49.54 | CunningPike | danp: If you are around tomorrow, I can dig out the settings for you |
| 05:50.12 | CunningPike | danp: It's a sip.cfg setting |
| 05:50.12 | danp | that would be great, thanks! |
| 05:50.18 | [TK]D-Fender | DocHolliday : Model should be dependant on your users needs, wiring per site, and budget |
| 05:50.20 | danp | any idea which area? |
| 05:50.26 | CunningPike | DocHolliday: IP501s |
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| 05:50.58 | CunningPike | danp: It's somewhere in levels I think - search for levels and see what you find |
| 05:51.08 | acehunky | can anyone over here help me with MFC/R2 ? |
| 05:51.17 | [TK]D-Fender | DocHolliday : For new PoE general deployments I'd suggest the IP 430. If you need speakerphone and don't have PoE, the IP 501. If you don't need speakerphone and no PoE, IP 301. IP 601 for receptionist. |
| 05:51.34 | acehunky | can anyone over here help me with MFC/R2 ? |
| 05:51.50 | DocHolliday | [TK]D-Fender, need speakerphone and PoE |
| 05:52.33 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 05:52.47 | DocHolliday | [TK]D-Fender, i have heard the call quality with the polycom's is superb.. cant wait to try one out |
| 05:52.51 | CunningPike | danp: I think it's tone.chord something |
| 05:53.11 | CunningPike | DocHolliday: The 501 does PoE too......... |
| 05:53.12 | [TK]D-Fender | DocHolliday : IP 430 is for you then. |
| 05:53.38 | [TK]D-Fender | CunningPike : at extra cost, both int he phone itself, AND the extra cable needed. |
| 05:53.59 | [TK]D-Fender | CunningPike : Not to mention the bulk (annoying at best) of that adapter cable... |
| 05:54.03 | CunningPike | [TK]D-Fender: Nicer phone, imho - does the IP430 have a speakerphone - I forger |
| 05:54.05 | danp | ahh, yes...think i found it |
| 05:54.10 | CunningPike | s/forger/forget/ |
| 05:54.13 | acehunky | i am getting my /var/log/messages filled up with "Double miss (168, 168)..." |
| 05:54.18 | acehunky | can anyone help me with that ? |
| 05:54.25 | danp | tone.chord.x.level.y |
| 05:54.32 | [TK]D-Fender | CunningPike : Oh, I don't disagree the IP 501 is NICER, but generally messier, and definately most expensive in a PoE environment. |
| 05:54.33 | acehunky | why does this come up in /var/log/messages .... Double miss (168, 168)... |
| 05:54.44 | CunningPike | danp: Great - I think there should be an entry for callwaiting |
| 05:54.47 | [TK]D-Fender | CunningPike : Yes, speakerphone and built-in PoE |
| 05:54.53 | danp | CunningPike: yep, thanks fir the tip |
| 05:54.54 | DocHolliday | yeah they are reasonably priced too.. hmmm, this will get interesting |
| 05:55.07 | CunningPike | acehunky: Because you posted the same question twice in one minute |
| 05:55.11 | [TK]D-Fender | DocHolliday : How much did you find them for? |
| 05:55.31 | DocHolliday | i just did a quick check on voipsupply.com they seem to be around 170~ |
| 05:55.40 | acehunky | gee CunningPike that message is getting posted 50 times in 1 min on my messages file :P |
| 05:55.41 | DocHolliday | have you used the conference phones? |
| 05:55.50 | [TK]D-Fender | DocHolliday : 150$ - http://www.telephonydepot.com |
| 05:56.03 | CunningPike | DocHolliday: IP4000? |
| 05:56.03 | acehunky | CunningPike can ya tell me why does that happen ? |
| 05:56.11 | [TK]D-Fender | DocHolliday : You meant he "boomerang"-like ones? |
| 05:56.16 | DocHolliday | thanks |
| 05:56.18 | CunningPike | acehunky: No - never heard of it - is it even asterisk related? |
| 05:56.24 | DocHolliday | yeah thats what i mean :) |
| 05:56.36 | acehunky | yp CunningPike |
| 05:56.37 | [TK]D-Fender | DocHolliday : I have a SoundStation 2W (wireless analog) one on an ATA. its what I suggest |
| 05:56.47 | CunningPike | DocHolliday: We have one - it doesn't provision properly with vsftpd, but otherwise it's great |
| 05:56.53 | acehunky | MFC/R2 + TE110P ... |
| 05:57.12 | acehunky | those are kernel messages .. though |
| 05:57.12 | [TK]D-Fender | CunningPike : Wow.... makes no sense.... Works for all the rest though. |
| 05:57.27 | DocHolliday | [TK]D-Fender, looks nice |
| 06:00.46 | CunningPike | [TK]D-Fender: It's a firmware issue - the IP501 used to have the same problem |
| 06:00.55 | [TK]D-Fender | CunningPike : What versions you running? |
| 06:01.03 | CunningPike | 1.6.7 and 3.whatever.the.current.one.is |
| 06:01.04 | CunningPike | 3.1.3, iirc |
| 06:01.12 | [TK]D-Fender | CunningPike : Upgrade to 3.2.2 & 2.0.3. |
| 06:01.12 | DocHolliday | [TK]D-Fender, the main reason i like cisco is because im used to throwing the config file up and walking out of the room.. dont ever even have to wait and see if it works, always does :) |
| 06:01.13 | CunningPike | [TK]D-Fender: We will - as soon as we can get our reseller organized to provide it to us |
| 06:01.14 | [TK]D-Fender | DocHolliday : Thats my normal install. Create config file, unpack phone, assemble, plug in, walk away 100% functional. |
| 06:01.14 | CunningPike | DocHolliday: Well, Polycoms do that at half the cost |
| 06:01.14 | [TK]D-Fender | CunningPike : I jsut need to upload it to my server and I'll link you tomorrow |
| 06:01.14 | DocHolliday | CunningPike, not at half.. the 7940 retails at 170 ish |
| 06:01.15 | DocHolliday | its a very marginal savings |
| 06:01.15 | CunningPike | [TK]D-Fender: Beauty - thanks! |
| 06:01.18 | DocHolliday | but [TK]D-Fender showed me a place thats quite a bit cheaper, so who knows |
| 06:01.47 | DocHolliday | [TK]D-Fender, what im looking forward to most is that we are setting up a wifi net so you should be able to walk around with a wireless handset and have access from everywhere |
| 06:01.47 | [TK]D-Fender | DocHolliday : And consider the Cisco you have to pay SmartNet for firmware support, etc. THEN tack on the extra $ that Cisco (non 802.3af PoE) entails. |
| 06:01.48 | CunningPike | DocHolliday: afaik, the SIP implementation is more complete in Polycoms, and you don't have to shell out for a SIP license |
| 06:02.03 | DocHolliday | [TK]D-Fender, dont have to pay cisco for firmware.. free off the website |
| 06:02.09 | [TK]D-Fender | DocHolliday : And then Cisco's inferior call handling, lack of presence support, and buggy firmware... THEN call me in the morning :) |
| 06:02.15 | DocHolliday | we are using cisco switches anyway |
| 06:02.31 | CunningPike | DocHolliday: Polycoms speak CDP........ |
| 06:02.44 | DocHolliday | no the firmware (7.x) is damn stable, 8.x not so much |
| 06:03.06 | CunningPike | DocHolliday: You're allowed to run non-Cisco gear on a Cisco network, despite what your rep tells you ;) |
| 06:03.25 | DocHolliday | ;_;.. bbbbbut *Kidding* |
| 06:03.30 | CunningPike | lol |
| 06:04.02 | DocHolliday | [TK]D-Fender, have you ever setup video conferencing? |
| 06:04.42 | [TK]D-Fender | DocHolliday : Nothing more than 2 eyeBeam H.264 phones.... |
| 06:05.19 | *** join/#asterisk zeeesh (i=aadilism@9-237-154-202.wol.net.pk) |
| 06:05.48 | DocHolliday | is that a softphone? |
| 06:06.52 | DocHolliday | ah got a picture |
| 06:07.25 | DocHolliday | has that proven to be the best solution? |
| 06:07.50 | [TK]D-Fender | DocHolliday : No, I think electrolysis is the way to go these days..... |
| 06:08.12 | DocHolliday | that sounds like a medical treatment |
| 06:08.29 | [TK]D-Fender | DocHolliday : Nope, still unregulated :) |
| 06:09.01 | DocHolliday | haha |
| 06:09.27 | DocHolliday | [TK]D-Fender, so much to think about :P doing-it will be the easy part |
| 06:10.08 | DocHolliday | i like the cisco video phone, but $$$ |
| 06:10.51 | [TK]D-Fender | DocHolliday : If you want to try it, go ahead, but I'd check to see if its supported in * first... |
| 06:11.02 | [TK]D-Fender | DocHolliday : You know Cisco's proprietary ways... |
| 06:11.36 | DocHolliday | yeah i question whether it would..for that price i'd rather have a real conference room setup |
| 06:11.38 | DocHolliday | even if it costs a bit more |
| 06:13.08 | [TK]D-Fender | file : jst about to hit the sack actually. |
| 06:13.17 | [TK]D-Fender | file : this is my norm don't forget... |
| 06:13.20 | file | I knew that, I'm psychic you see |
| 06:13.30 | [TK]D-Fender | file : Typo.... make that PSYCHOTIC! |
| 06:13.34 | file | :D |
| 06:13.52 | DocHolliday | [TK]D-Fender, im already in bed :) |
| 06:13.54 | DocHolliday | beat ya to it |
| 06:14.10 | [TK]D-Fender | .... |
| 06:14.33 | [TK]D-Fender | DocHolliday : I don't really think I need to hear about you beating it in bed..... |
| 06:14.34 | DocHolliday | laptop |
| 06:14.43 | [TK]D-Fender | EW! |
| 06:14.48 | DocHolliday | LOL |
| 06:15.00 | DocHolliday | that does sound wrong actually... |
| 06:15.19 | [TK]D-Fender | Especially with my value-added emphasis :) |
| 06:15.27 | DocHolliday | but naw i read ebooks in bed... meh |
| 06:16.51 | DocHolliday | [TK]D-Fender, usually i just bring a bluetooth keybard (computer interfaces with TV) |
| 06:18.14 | [TK]D-Fender | DocHolliday : My * server is my "everything" server hooked up to my 52" HDTV :) |
| 06:18.35 | DocHolliday | sweet |
| 06:18.49 | [TK]D-Fender | DocHolliday : Used to make me coffee too :D |
| 06:19.05 | DocHolliday | how does that work? :O |
| 06:19.20 | [TK]D-Fender | DocHolliday : X-10 + *. Dial-a-coffee |
| 06:19.33 | DocHolliday | nooh way |
| 06:19.40 | [TK]D-Fender | DocHolliday : Way. |
| 06:19.54 | DocHolliday | thats.. so awesome. wish i thought of that |
| 06:20.16 | [TK]D-Fender | DocHolliday : Kettle on a SuperSocket. Heyu2 w/ *, and a CM11A |
| 06:20.37 | [TK]D-Fender | DocHolliday : Cool stuff I leave in a box now :) |
| 06:20.53 | DocHolliday | heh, not bad :) |
| 06:20.54 | [TK]D-Fender | DocHolliday : can't really be bothered to set it all up in my new apt... |
| 06:21.21 | DocHolliday | [TK]D-Fender, why not set it up on a timer for the morning? |
| 06:21.58 | paavum | exit |
| 06:22.16 | DocHolliday | setup an asterisk macro :) |
| 06:22.20 | [TK]D-Fender | DocHolliday : Because I don't get up at a fixed time and refuse to rush so it doesn't get cold :) |
| 06:22.45 | DocHolliday | damn, im on a routine every morning |
| 06:23.08 | DocHolliday | 7:30 wakeup, 8 am get out of bed.. 8:15 leisurely stroll to shower |
| 06:23.11 | [TK]D-Fender | DocHolliday : Yes it would be remarkably easy to do. But you see I sleep in as late as humanly possible and get out of bed and get ready enough to leave immediately. I do coffee at work for free :) |
| 06:23.29 | [TK]D-Fender | DocHolliday : Give me more sleeping time, and costs less. |
| 06:23.38 | DocHolliday | nice |
| 06:23.45 | DocHolliday | *shakes fist* :) |
| 06:24.04 | [TK]D-Fender | I am not lazy... I spend a lot of time planning my life this way! |
| 06:24.04 | DocHolliday | damn you and your sleep-ins :( |
| 06:24.21 | DocHolliday | haha ya right |
| 06:24.26 | [TK]D-Fender | I like to think of it more like an "efficient" use of my time |
| 06:24.51 | DocHolliday | oh i as well, no question about it |
| 06:25.12 | DocHolliday | [TK]D-Fender, i do an afternoon nap around 1:30 usually |
| 06:25.14 | [TK]D-Fender | Speaking of which.... consciousness limit attained... time to hit the sack. |
| 06:25.27 | [TK]D-Fender | DocHolliday : My employer might "frown" upon that :) |
| 06:25.33 | DocHolliday | likewise, till next time .. thanks for the help |
| 06:25.38 | [TK]D-Fender | np |
| 06:25.41 | DocHolliday | haha, self-employed! |
| 06:29.12 | acehunky | can any one help me with this : http://www.pastebin.ca/317026 |
| 06:30.20 | *** join/#asterisk paavum (n=dorphals@pcsp163-73.supercabletv.net.co) |
| 06:30.22 | paavum | hi |
| 06:30.29 | paavum | Just mty last question for a while |
| 06:30.30 | paavum | <PROTECTED> |
| 06:30.30 | paavum | <PROTECTED> |
| 06:30.30 | paavum | <PROTECTED> |
| 06:30.42 | paavum | What does the 484 mean? |
| 06:30.49 | JT | address incomplete |
| 06:30.57 | paavum | yes ... and that means? |
| 06:31.14 | JT | you probably didn't send enough matches to yield a dialplan match |
| 06:31.19 | JT | send enough digits |
| 06:31.21 | JT | i mean |
| 06:32.27 | paavum | yeah |
| 06:32.30 | paavum | no dialplan match |
| 06:32.31 | paavum | thnx! |
| 06:32.33 | paavum | bye |
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| 06:59.13 | Witch_Doc | anyone here have expierence with integrating asterisk with a pbx using Qsig? |
| 06:59.39 | CunningPike | Witch_Doc: Witch PBX? |
| 07:00.07 | Witch_Doc | panasonic |
| 07:00.36 | Witch_Doc | CunningPike but it shouldn't really matter which pbx does it? I though qsig is a standard protocol |
| 07:01.18 | CunningPike | Witch_Doc: Guess not |
| 07:04.52 | perd | what is a good cheap sip phone? |
| 07:08.12 | justdave | perd: I don't think you can get one. |
| 07:08.21 | justdave | you either get a good one or you get a cheap one. |
| 07:08.26 | perd | ah |
| 07:08.34 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 07:08.34 | perd | well f it then. software it is |
| 07:09.27 | justdave | Grandstream BudgetTone series is the cheapest I saw when I was shopping for phones last. They're not bad, but they have a few quirks that are annoying. :) |
| 07:09.50 | justdave | like no way to turn off the speakerphone once you turn it on without hanging up the call |
| 07:10.11 | justdave | and the callerid display is number only, doesn't show the name |
| 07:10.32 | Qwell | justdave: just like an analog speakerphone, you can probably lift the handset to go off speaker |
| 07:10.56 | justdave | Qwell: not if the handset is already lifted |
| 07:11.00 | Qwell | not that I'm defending them or anything...they suck ;) |
| 07:11.06 | justdave | (like if you picked up the handset first, then switched to speaker :) |
| 07:11.11 | Qwell | justdave: hanging up the handset doesn't keep it on speaker? |
| 07:11.40 | justdave | no, that hung up, too |
| 07:11.40 | Qwell | if not, then that's a bug :p |
| 07:11.40 | Witch_Doc | anyone here have expierence with integrating asterisk with a pbx using Qsig? |
| 07:11.46 | justdave | Polycom phones are *nice*. |
| 07:11.47 | Qwell | it should only hangup if you've gone onhook then offhook while on speaker |
| 07:11.53 | justdave | but they're freaking expensive, too. |
| 07:12.01 | Qwell | justdave: look at the ip430 |
| 07:12.07 | Qwell | not that expensive |
| 07:12.17 | Qwell | and much better than the barbietones |
| 07:12.32 | justdave | heh. IP501 is the one I have on my desk. |
| 07:12.40 | Qwell | bed time |
| 07:12.43 | justdave | provided by my employer, I could never afford one |
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| 07:31.22 | arcanine | cn sum1 comfirm dis: asterisk now works with dialogic JCT600-2e1 |
| 07:34.48 | perd | i know nothing!!! |
| 07:37.12 | Nugget | arcanine: i cn cnfrm asterisk cn wrk with dlg bt mb 2 it can't unless u make sre 2 edt main cnf with ln frl sak. |
| 07:39.19 | DrCron | b& for both of you |
| 07:40.45 | *** join/#asterisk flenders (n=fserto@unaffiliated/flenders) |
| 07:45.33 | flenders | anyone alive? |
| 07:46.02 | flenders | I have a question about caller ID on PSTN lines |
| 07:46.35 | flenders | is it possible to change the outgoing caller ID on all our PSTN lines on asterisk? |
| 07:48.22 | *** join/#asterisk alamantia (n=Anthony@65.4.1.65) |
| 07:49.41 | Nugget | flenders: not that I'm aware of. |
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| 07:50.59 | Witch_Doc | anyone here have expierence with integrating asterisk with a pbx using Qsig? |
| 07:51.05 | flenders | Nugget: do you know if carriers would be to do it? |
| 07:51.10 | arcanine | Nugget: sorry but wat is 'dlg bt mb' |
| 07:52.12 | arcanine | Nugget: is this intel's dialogic cards |
| 07:54.57 | Nugget | arcanine: I have no idea, I just didn't want your pidgin english to pass unmocked. |
| 07:56.39 | Nugget | Your AOLbonics made my head hurt so I thought I'd return the favor. |
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| 08:19.53 | JT | flenders: no, it's analogue, there is no signalling mechanism for doing so |
| 08:20.20 | *** join/#asterisk svenna_ (n=svenna@p548D2873.dip0.t-ipconnect.de) |
| 08:20.40 | JT | carriers have telco switches and digital lines internally |
| 08:20.56 | JT | the pstn is an archaic system reserved for low end consumers |
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| 08:41.51 | Ahrimanes | anyone have good voip termination in somalia? |
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| 08:46.28 | arcanine | have someone used a dialogic card on asterisk and is working |
| 08:49.28 | Corydon76-home | The dialogic cards are only supported via a proprietary driver, and even then, most of them aren't |
| 08:50.28 | Corydon76-home | My understanding is that the licensing fees cost as much as a brand new Digium card |
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| 09:02.38 | tzafrir | Not to mention the requirement for ABE, due to the proprietary drivers |
| 09:21.35 | monsted | gah! |
| 09:22.22 | Ahrimanes | monsted, now what? |
| 09:22.49 | *** join/#asterisk af_ (n=af@ip-173-157.sn1.eutelia.it) |
| 09:23.12 | monsted | one customer used 60 ISDN channels until wednesday where i upgraded to 90, which they immediately used. on friday i added 60 more and they're now using 135 of their 150 channels and are attaching another large office building to it today... |
| 09:23.49 | Ahrimanes | hehe |
| 09:23.55 | Ahrimanes | thats money in the bank |
| 09:24.17 | monsted | yeah |
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| 09:52.05 | zeeesh | hi |
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| 09:54.35 | santibiotico | hi |
| 09:54.48 | santibiotico | is there any bristuff version for use with asterisk 1.4 ?? |
| 09:58.15 | JT | don't believe so |
| 09:58.31 | santibiotico | uhh |
| 09:58.36 | santibiotico | so do i :lol |
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| 10:10.54 | tzafrir | santibiotico, not that I know of |
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| 10:13.07 | tzafrir | anybody has been in touch with kapejod lately? |
| 10:13.57 | tzafrir | nobody else seems interested, and I haven't heard anything from kapejod in quite a while |
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| 10:24.37 | susinths | Wikipedia writes that asterisk has some timing problem when used for VOIP only, is that right? |
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| 10:39.05 | perd | no, you just need ztdummy |
| 10:39.36 | perd | and it's only for certain things, like conference calls |
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| 10:40.21 | santibiotico | i'm getting a warning in the cli console whenever i try to make a call |
| 10:40.22 | santibiotico | Jan 15 11:34:02 WARNING[9191]: channel.c:1602 ast_waitfor_nandfds: Thread -1225696336 Blocking 'SIP/station42-081ba650', already blocked by thread -1226744912 in procedure ast_waitfor_nandfds |
| 10:40.36 | santibiotico | does anybody know what does it mean?? |
| 10:40.38 | perd | beats me |
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| 11:04.32 | Aurs | I always forget.. url to asterisk svn? |
| 11:04.34 | clive- | can anyone tell me if the iptables rule " udp --dport 5004:5082" means from port 5004 to 5082, or only 5004 and 5082 ? Its for SIP |
| 11:04.48 | Aurs | svn.digium.com, ok |
| 11:05.49 | Aurs | clive-: it is from A to B |
| 11:05.49 | Aurs | when A:B is used |
| 11:05.49 | clive- | Aurs thanks a lot |
| 11:17.56 | *** join/#asterisk penguinFunk (n=penguin@87.224.86.46) |
| 11:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
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| 11:25.35 | *** join/#asterisk PhilKC (i=greece@freenode/staff/about.linux.philkc) |
| 11:25.38 | *** join/#asterisk Kigh (n=kai@ciphron.de) |
| 11:25.42 | Kigh | hi there |
| 11:25.47 | *** join/#asterisk agallo (n=Antonio@88.34.216.63) |
| 11:25.49 | *** join/#asterisk bkw_ (n=brian@80.187.147.53) |
| 11:26.35 | agallo | Hello, wich is the correct syntax to monitor status of mISDN channell? exten=>1000,hint,mISDN/5 is working correctly only if both lines are busy but hwich is the syntax tomonitor each of the 2 channels? |
| 11:27.58 | Kigh | you may create a group containing your 2 channels and refer to that group in your exten => 1000,hint,mISDN/g:mygroup |
| 11:28.03 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:29.02 | Kigh | to configure groups, you need to edit asterisks misdn.conf |
| 11:29.53 | *** join/#asterisk zeppelin_ (n=zeppelin@201.66.140.22) |
| 11:30.13 | agallo | Kigh, ah sorry i meant i want to monitor both of the 2 channels of every port into a separate way :) |
| 11:30.24 | Kigh | Does somebody experience a short silence (1-2 sec) at the beginning of a conversation? i cant hear the callers name when he talks fast |
| 11:31.20 | agallo | Kigh, i tried hint,mISDN/5-1 and hint,mISDN/5/1 but didn't worked :-( |
| 11:31.27 | Kigh | echo training is turned "off", so its most likely not causing the rproblems |
| 11:31.37 | *** join/#asterisk random_wolf (n=boolean1@adsl-68-73-153-41.dsl.ipltin.ameritech.net) |
| 11:31.49 | Kigh | agallo: what is mISDN/5 .. a group? :) |
| 11:31.59 | Kigh | ah |
| 11:32.09 | Kigh | sry now i got the point |
| 11:32.29 | Kigh | hm. |
| 11:33.51 | Kigh | dont know how to do this in dial plan |
| 11:36.06 | random_wolf | So. |
| 11:36.09 | random_wolf | Hi. |
| 11:38.05 | *** join/#asterisk in-pt (n=lokesh@estrela.nortenet.pt) |
| 11:38.39 | *** join/#asterisk awannabe (n=gti@ip24-251-135-202.ph.ph.cox.net) |
| 11:42.00 | agallo | also, is there a way to start monitoring a call from CLI or with AGI or from ChanSpy() ? Actually i want to monitor a call on the fly, ofc a call that is not generated by one of my phones |
| 11:42.27 | zeeesh | hi |
| 11:42.34 | random_wolf | hi |
| 11:42.37 | random_wolf | :p |
| 11:42.38 | zeeesh | facing this problem ... during ... Everyone is busy/congested at this time (1:0/0/1) |
| 11:42.38 | zeeesh | <PROTECTED> |
| 11:44.44 | *** join/#asterisk Stephnie (i=Stephnie@u15157627.onlinehome-server.com) |
| 11:44.46 | Stephnie | hi |
| 11:46.33 | Stephnie | Dial() ==> 1 Second ===ANSWERED .......then Hangup .....In simple words...if Asterisk Gets Answered in 1 or 2 seconds after DIAL() ..then want to hangup that call.. |
| 11:46.54 | Stephnie | any idea to make it possible? |
| 11:47.03 | awannabe | hey guys, if i have a server dialing out thru another server, what the best way to set tougoing caller ID? all the calls come in on the same context, so that creates a problem since I cant do it by context |
| 11:47.12 | clive- | stephie use a timeout |
| 11:47.52 | Stephnie | clive-: like? |
| 11:48.24 | clive- | use the T option I think...see wiki |
| 11:48.40 | Stephnie | no...thats not what I want... |
| 11:48.51 | agallo | Stephnie, you want hangup if answered too early? |
| 11:48.57 | Stephnie | yes |
| 11:49.11 | Stephnie | agallo: correect ..if answered is in 1 or 2 secs |
| 11:49.37 | clive- | id use an agi for that |
| 11:50.34 | Stephnie | I have already been using agi... |
| 11:50.37 | agallo | Stephnie, maybe use M() option of the Dial command to spawn a macro when the call answered and check the time difference? :-) |
| 11:50.51 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 11:52.23 | Stephnie | agallo: hmmm never tried M and Macro....:) but going to check it.. |
| 11:52.41 | agallo | Stephnie, i never tested it too... so good luck :) |
| 11:53.10 | random_wolf | Has anyone successfully interfaced any dialogic cards? |
| 11:53.23 | Stephnie | if it doestn't after spending 1 hour...then I will be at ur head ;) |
| 11:55.58 | agallo | Stephnie, there are high mountains here... it will be hard to spot me :-P |
| 11:56.47 | Stephnie | hehe |
| 11:57.30 | Stephnie | can I do some matematics in dial PLAN ? |
| 11:57.54 | acehunky | anyone over here has worked on chan_ss7 ? |
| 11:58.01 | acehunky | i get this in my CLI " MTP2 CRC error (CRC=0x6c0 != 0xf0b8) on link" |
| 11:58.51 | awannabe | they have a chan_ss7? nice! |
| 12:06.53 | *** join/#asterisk andrebarbosa (n=andrebar@83.240.148.215) |
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| 12:11.36 | *** part/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 12:12.06 | agallo | urgh... anyone knows why if i get a 2nd ISDN call the 1st one drop with a message (illegal seek... bridge failed...etc.) ? |
| 12:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 12:18.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 12:20.38 | tzanger | kernel panic on the asterisk box |
| 12:20.41 | tzanger | no wonder the fucker wouldn't respond |
| 12:20.52 | random_wolf | XD haha. |
| 12:20.58 | random_wolf | Well worded. |
| 12:21.01 | tzanger | unable to dereference pointer at 0xb6b6b6b6 |
| 12:21.24 | tzanger | I think that might be pointing to a bad driver or iffy system |
| 12:21.35 | tzanger | and the ip-kvm and ip-switches are due in this week |
| 12:21.46 | xheliox | I give my servers Xanax when they panic. |
| 12:21.56 | Ahrimanes | tzafrir, perfect timing, eh? |
| 12:24.24 | tzafrir | tzanger, at which module? that pointer value looks like garbage |
| 12:24.38 | tzafrir | (only here due to Ahrimanes's call) |
| 12:24.40 | tzanger | tzafrir: precisely |
| 12:24.59 | tzanger | tzafrir: not sure yet. came in, bigass panic on the screen, nothing in logs |
| 12:25.04 | tzanger | so I also have to get serial logging going |
| 12:25.38 | *** join/#asterisk Aurs (i=Aurs@81.191.121.187) |
| 12:27.42 | tzanger | ugh I hate asterisk registrations |
| 12:27.44 | Ahrimanes | ah |
| 12:28.09 | tzanger | *CLI> [Jan 15 07:27:54] NOTICE[16222]: chan_sip.c:7190 sip_reg_timeout: -- Registration for 'andrew-home@foo' timed out, trying again (Attempt #1) |
| 12:28.19 | tzanger | bullshit, foo sees you and replies |
| 12:28.39 | Kigh | Does somebody experience a 1-2 second long silence when establishing a call? this also happens when calling from ext to ext (ext<->asterisk<->ext on local network). |
| 12:29.23 | Kigh | its really annoying to not hear the name of the caller =) |
| 12:29.33 | Gido-E | :-) |
| 12:29.37 | tzanger | Kigh: get a packet trace and see what's going on |
| 12:29.57 | *** join/#asterisk infernix (i=nix@spirit.infernix.net) |
| 12:31.12 | agallo | "res_features.c: Bridge failed on channels" O.o is that a codec problem? i did't used any features... just answered call O.o |
| 12:31.21 | *** join/#asterisk Nobbie (n=no@fwb003.fw.is.co.za) |
| 12:31.21 | Aurs | is ooh323 any good? |
| 12:31.26 | Aurs | from asterisk-addons |
| 12:31.46 | x86 | H323 should be avoided like the plague |
| 12:31.52 | qdk | Aurs: its quite useless for SIP calls. |
| 12:32.18 | xheliox | ... |
| 12:32.21 | xheliox | Captain Obvious |
| 12:32.33 | x86 | rofl |
| 12:33.21 | Nobbie | on SIP operator phones (Cisco 7960, SNOM360), how does one control the incoming calls so that a new incoming call doesn't interrupt you while busy doing a transfer ? |
| 12:34.21 | Ahrimanes | Nobbie, disable call-waiting on the phone and set call-limit on the sip peer is a good start |
| 12:34.30 | tzanger | yes H323 is like SIP in menopause |
| 12:35.16 | Nobbie | Ahrimanes: call-limit is in sip.conf ? |
| 12:37.08 | Nobbie | Ahrimanes: that could work, but you wouldn't be able to see that there are calls waiting then. i would like to see that there are other calls, but they mustn't be rude and take over focus of the phones interface |
| 12:37.23 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 12:37.32 | awannabe | Nobbie: thats a phone issue, not * then |
| 12:37.52 | *** join/#asterisk Formater (i=Formater@cable-87-116-140-142.dynamic.sbb.co.yu) |
| 12:38.38 | Nobbie | i figured as much |
| 12:38.52 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 12:38.53 | x86 | anyone ever have a problem with Polycom IP601's and DTMF? |
| 12:39.00 | Formater | hi! how to limit simultaneous calls using sip_buddies realtime stuff? in past in sip.conf there was limit or call-limit variable, now i tried to add those fields to sip_buddies table, but not helped |
| 12:39.30 | x86 | if i call an auto attendant using G.729 from my IP601 (for example, when I call my bank), it doesnt send DTMF correctly |
| 12:39.45 | awannabe | dtmfmode is wrong |
| 12:39.51 | x86 | i can, however, use my grandstream BT101 and call the same number, and have it work flawlessly |
| 12:39.55 | x86 | awannabe: nope |
| 12:40.07 | x86 | awannabe: works just fine with a grandstream phone |
| 12:40.08 | awannabe | oh, it is though lol |
| 12:40.18 | awannabe | different phones use differnt dtmf modes |
| 12:40.26 | x86 | it's set to RFC |
| 12:40.29 | Grnd-Wire | awannabe: Should it be rfc ? |
| 12:40.35 | Grnd-Wire | Or SIP INFO ? |
| 12:40.38 | awannabe | i thought poly used info |
| 12:40.47 | awannabe | i havent set one up in a while, heh |
| 12:40.51 | x86 | well it sends _some_ of my digits _some_ of the time |
| 12:41.06 | x86 | might take 7 or 8 presses of a button to get it to send the correct tone ;) |
| 12:41.20 | x86 | hmm |
| 12:41.57 | awannabe | intresting |
| 12:42.22 | x86 | so should i try info? |
| 12:42.38 | awannabe | cant hurt |
| 12:43.40 | Formater | hi! how to limit simultaneous calls using sip_buddies realtime stuff? in past in sip.conf there was limit or call-limit variable, now i tried to add those fields to sip_buddies table, but not helped |
| 12:43.42 | Grnd-Wire | ya - It'll either break it better, or make it better.. :D |
| 12:43.51 | x86 | lol, now it wont even dial at all |
| 12:43.57 | awannabe | im looking though, and i used rfc2833 on my poly |
| 12:44.12 | Grnd-Wire | Well there you go.. hee hee.. and you can't use inband because you're using g.729, correct? |
| 12:44.50 | x86 | right |
| 12:45.00 | x86 | so i _have_ to use rfc2833 |
| 12:46.33 | awannabe | lovely |
| 12:47.35 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
| 12:48.39 | x86 | odd, wonder why it doesnt work |
| 12:49.48 | awannabe | typcal monday it is! ha |
| 12:49.55 | *** join/#asterisk HarryR (n=Harry@host-83-146-53-46.bulldogdsl.com) |
| 12:50.16 | *** join/#asterisk bkw_ (n=brian@82.153.201.145) |
| 12:51.17 | x86 | any way to reboot a polycom phone from the asterisk CLI? |
| 12:51.26 | awannabe | not that i know of |
| 12:51.32 | awannabe | only snom |
| 12:51.44 | agallo | i have some analog lines using zaptel get stuck at Busy() command in dialplan.. i've to use CLI Soft Hangup to close them. any idea? |
| 12:52.27 | *** join/#asterisk Skarmeth (n=Skarmeth@201009061013.user.veloxzone.com.br) |
| 12:53.04 | *** join/#asterisk neuwald (n=felipe@200.199.251.131) |
| 12:53.20 | neuwald | hi folks |
| 12:53.21 | *** join/#asterisk shinux__ (n=shinux@196.220.24.177) |
| 12:53.58 | neuwald | I'm having a strange problem here. No sounds under /var/lib/asterisk/sound is played |
| 12:54.01 | neuwald | <PROTECTED> |
| 12:54.01 | neuwald | <PROTECTED> |
| 12:54.01 | neuwald | <PROTECTED> |
| 12:54.28 | neuwald | in this example, the sound custom/bemvindo-Andrea isn't played, but the moh is played. Anybody knows what does it should be? |
| 12:55.55 | Marty-OTT | Can you play other music? |
| 12:56.18 | Marty-OTT | b11d: you around? prolly not eh.. |
| 12:56.54 | *** join/#asterisk infernix (i=nix@spirit.infernix.net) |
| 12:57.02 | Marty-OTT | Anyways, when you see this message, I got around to rebooting last night (after the kernel recombiling) and ran into a few issues but the most important one being the following: |
| 12:57.25 | Marty-OTT | Jan 14 17:49:58 asterisk kernel: wanpipe1: ERROR: Unsupported T1 CHIP (0x07) |
| 12:57.37 | Marty-OTT | That doesn't look good... |
| 12:58.52 | neuwald | Marty-OTT in my full.log I got this: |
| 12:58.54 | neuwald | Jan 15 10:56:47 VERBOSE[10129] logger.c: -- Executing Playback("SIP/2001-0821afe0", "custom/bemvindo-Andrea") in new stack |
| 12:58.54 | neuwald | Jan 15 10:56:47 WARNING[10129] file.c: File custom/bemvindo-Andrea does not exist in any format |
| 12:58.54 | neuwald | Jan 15 10:56:47 WARNING[10129] file.c: Unable to open custom/bemvindo-Andrea (format ulaw): No such file or directory |
| 12:58.54 | neuwald | Jan 15 10:56:47 WARNING[10129] app_playback.c: ast_streamfile failed on SIP/2001-0821afe0 for custom/bemvindo-Andrea |
| 12:58.58 | neuwald | but the file is there |
| 12:59.27 | neuwald | santocristo1:/etc/asterisk# ls -laF /var/lib/asterisk/sounds/custom/bemvindo-Andrea.wav |
| 12:59.27 | neuwald | -rw-rw---- 1 asterisk asterisk 149036 2007-01-12 17:46 /var/lib/asterisk/sounds/custom/bemvindo-Andrea.wav |
| 12:59.42 | *** join/#asterisk infernix (i=nix@spirit.infernix.net) |
| 13:00.14 | Marty-OTT | Let me check the permission on my files.. (fyi.. I'm a newbie to Asterisk)... and I'm only playing gsm |
| 13:02.10 | *** join/#asterisk dasenjo (n=dasenjo@63.245.86.43) |
| 13:02.42 | *** part/#asterisk Stephnie (i=Stephnie@u15157627.onlinehome-server.com) |
| 13:02.50 | Marty-OTT | Hmm .. my permission are read for other as well: |
| 13:02.50 | Marty-OTT | -rw-r--r-- 1 root wheel 957 Jan 9 14:30 your.gsm |
| 13:05.01 | Marty-OTT | But it definitely looks like a permission or path issue though |
| 13:06.43 | *** join/#asterisk infernix (n=nix@spirit.infernix.net) |
| 13:11.23 | in-pt | hi all |
| 13:11.57 | in-pt | can anyone please tell me if chan_sccp patch can work with asterisk-1.4.0 ?? |
| 13:12.46 | zeeesh | hi |
| 13:13.02 | Nobbie | what difference does user=phone make ? |
| 13:13.28 | zeeesh | what is the reason for ............. == Everyone is busy/congested at this time (1:0/0/1) |
| 13:13.28 | zeeesh | <PROTECTED> |
| 13:13.28 | zeeesh | ????? |
| 13:13.49 | Thazza | Hey all. |
| 13:13.49 | *** join/#asterisk potsboy (n=chrisg@196.33.37.1) |
| 13:13.54 | Nobbie | zeesh: the line is busy or unavailable. what's the result of: sip show peers like 102 |
| 13:13.57 | Marty-OTT | morning |
| 13:14.10 | Thazza | Have a interesting question for asterisk guru's |
| 13:14.11 | *** join/#asterisk markit (n=konversa@host119-245-static.72-81-b.business.telecomitalia.it) |
| 13:15.13 | Thazza | Is it poss to send an inbound call back out, with the callerid, comming from the original callerid. |
| 13:15.30 | neuwald | Marty-OTT I changed the permission to o+r and I got the same error... |
| 13:15.52 | Thazza | For example, joe calls the office, which has an exten to dial(somemobile) can the mobile get joes callerid, not the callerid of the office voip line? |
| 13:16.11 | Nobbie | Thazza: yes. but if you send the call out via PSTN, the telco provider may not accept the call or won't send the clid |
| 13:16.30 | Nobbie | Thazza: doesn |
| 13:16.38 | Nobbie | Thazza: doesnt sound posible |
| 13:17.01 | Thazza | Nobbie: OK, so it is to do with telco.. ok.. thought since if you forward your home line to your mobile, it transfers the callerid. |
| 13:17.07 | markit | I would need a different logic for a "dialer" application.. at the moment, through the Manager, if I want to call someone I have asterisk make my extension ring and when I pick-up, I have just to wait the other to answer. I would like to have the same, but be able to pickup the extension without having it to ring, is it possible? |
| 13:17.53 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 13:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 13:18.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 13:18.19 | Thazza | Nobbie: I guess if the telco allowed spoofing of the callerid this would fix it.. Yet spoofing callerid, is bad, and i agree all telco's should not support. |
| 13:18.33 | Nobbie | pobbiel legal issues |
| 13:18.44 | Nobbie | possible even .... pretending to be someone else |
| 13:19.09 | Nobbie | like Visa/Mastercard division, asking for Credit card verifcation |
| 13:19.16 | Thazza | Nobbie: yes thats was i was researching 10 mins ago.. very bad.. atm mostly only friends joking with friends. |
| 13:19.25 | zeeesh | Nobbie: if i m dialing .. my own cell from user agent ... is that a right extension.... " exten => 9154XXXXXX,1,Dial(sip/9154XXXXX@9154XXXXXX)????? |
| 13:19.29 | Thazza | Nobbie: Has the poss of being very bad |
| 13:19.33 | Nobbie | Thazza: although, a prefix/suffix would be nice |
| 13:20.44 | Ahrimanes | calls.. |
| 13:20.56 | Nobbie | zeesh: you probably want to remove everything from, and including the @ |
| 13:20.57 | Ahrimanes | flaky laptop keyboard :( |
| 13:21.04 | Thazza | Nobbie: Perhaps once day when voip has replaced the telco's pstn network. we will all have wireless voip pda's.. so no need of redirecting to mobile. |
| 13:21.39 | Ahrimanes | hm global wifi coverage sounds well... not feasible |
| 13:21.39 | DrukenLPY | Thazza: i don't agree at all, i think all telco's should allow spoofing, however, i also belive you should be a registered clec in order to get it |
| 13:21.53 | Nobbie | Ahm: but useful ... |
| 13:21.56 | Thazza | clec? |
| 13:22.29 | zeeesh | <Nobbie> :then what shud be ?/??/ |
| 13:22.34 | Ahrimanes | Nobbie, but the range of wifi access points makes it un-possible.. wimax on the other hand.. |
| 13:22.48 | DrukenLPY | competitive local exchange carrier |
| 13:23.04 | Nobbie | zee: Dial(sip/9154XXXXX) |
| 13:23.14 | zeeesh | ok |
| 13:23.20 | Nobbie | zee: Dial(SIP/9154XXXXX) |
| 13:23.29 | zeeesh | trying |
| 13:24.09 | Thazza | We have global country internet access to 97%, just not faster than dialup for 60% of it. |
| 13:24.22 | Thazza | sorrt that was global wireless/sat tech |
| 13:24.31 | Thazza | s/sorrt/sorry |
| 13:25.23 | Thazza | If a laptop can have a pcmcia card that allows it net access, i can't see why a pda/voip phone can't do the same thing. |
| 13:26.12 | Ahrimanes | depends on latency etc on the connection |
| 13:26.20 | DrukenLPY | laptop has more processing power... whereas a voip phone is pretty dumb for the most part |
| 13:26.20 | Thazza | Perhaps not complete world coverage.. yet i am sure, most tech advantged countrys |
| 13:26.37 | Thazza | DrukenLPY: Give 'em time |
| 13:26.45 | Nobbie | who's running 1.4.0 yet ? |
| 13:26.56 | DrukenLPY | Thazza: not to mention, aircards for a laptop has fucken expensive |
| 13:27.05 | DrukenLPY | s/has/are |
| 13:27.08 | markit | Nobbie: me, but not in the office,play sometime at home |
| 13:27.35 | Thazza | DrukenLPY: Don't think they are here. |
| 13:27.49 | DrukenLPY | i KNOW they are here... |
| 13:27.49 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-140-124.ks.ok.cox.net) |
| 13:27.49 | Nobbie | i want to roll it out simply for the autofill=yes option in queues.conf. current setup with 1.2.x is not working well for us |
| 13:29.18 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 13:30.02 | Thazza | DrukenLPY: Is that the states? |
| 13:30.26 | puzzled | hi |
| 13:32.13 | lilalinux | for debugging reasons I need to log each request IMMEDIATELY (even without status code etc.) e.g. to errorlog |
| 13:33.25 | Marty-OTT | how bad is this: |
| 13:33.25 | Marty-OTT | Jan 14 17:49:58 asterisk kernel: wanpipe1: ERROR: Unsupported T1 CHIP (0x07) |
| 13:36.32 | puzzled | Marty-OTT: does not look encouraging. why don't you ask Sangoma support? |
| 13:37.46 | DrukenLPY | Thazza: canada |
| 13:38.27 | agallo | Where i can find a doc with the diferrence migrating from 1.2 to 1.4 ? |
| 13:38.45 | DrukenLPY | changes? |
| 13:38.48 | Marty-OTT | puzzled: Yeah, think I'll have to .. I wish b11d was around - he's doing the exact same as me. |
| 13:38.49 | *** join/#asterisk sudhir492 (n=sudhir@c-71-63-59-45.hsd1.va.comcast.net) |
| 13:38.53 | sudhir492 | Hi all |
| 13:39.07 | puzzled | hi |
| 13:41.55 | sudhir492 | anyone using FOP here |
| 13:41.55 | agallo | DrukenLPY, yup |
| 13:41.55 | sudhir492 | specifically, Channel_Context in fop |
| 13:41.55 | DrukenLPY | agallo, i ment the changes file ... |
| 13:41.55 | DrukenLPY | theres your document |
| 13:43.46 | agallo | DrukenLPY, yup i mean what i have to change to migrate from 1.2 to 1.4 (dialplan, AGI, and config files) |
| 13:44.29 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 13:46.23 | *** join/#asterisk UlbabraB (n=salama@host241-43-static.72-81-b.business.telecomitalia.it) |
| 13:47.01 | robl^ | agallo: there is a README.txt file that tells you what all needs to be done to migrate. Tells you what commands or functions have been removed, etc. |
| 13:49.48 | agallo | robink, only a text file? any URL to a more detailed review or article? :) |
| 13:56.37 | robl^ | agallo: actually. it is UPGRADE.txt http://svn.digium.com/view/asterisk/branches/1.4/UPGRADE.txt?rev=48349&view=markup |
| 13:58.55 | SheriF_SpacE | i'm trying to use asterisk-gui i have TDM400P installed i have a TDM400P installed and ztcfg -vv sees it but when i add a analog provider i got no analog line hardware configruared on the system. |
| 13:59.34 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:00.33 | agallo | Thank you! |
| 14:00.34 | *** join/#asterisk emiquelito (n=evandro@200-155-185-1.static.spo.ifx.net.br) |
| 14:03.24 | *** join/#asterisk IPmonger (n=ipmonger@c-68-84-208-206.hsd1.pa.comcast.net) |
| 14:07.50 | *** join/#asterisk jmacz (n=jmacz@201.244.173.61) |
| 14:13.12 | *** join/#asterisk _VoicePulse (n=contact@unaffiliated/voicepulse) |
| 14:13.56 | puzzled | SheriF_SpacE: afaik the gui is not ready yet. someone else please correct me if I'm wrong |
| 14:15.53 | *** join/#asterisk jmacz (n=jmacz@201.244.173.61) |
| 14:16.02 | jmacz | #freeswitch |
| 14:16.08 | *** part/#asterisk jmacz (n=jmacz@201.244.173.61) |
| 14:16.36 | *** join/#asterisk jmacz (n=jmacz@201.244.173.61) |
| 14:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 14:18.03 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 14:18.51 | tzafrir | puzzled, basic stuff does work |
| 14:19.03 | puzzled | tzafrir: ok thanks |
| 14:19.08 | tzafrir | buggy, but works |
| 14:19.43 | tzafrir | For instance: with Asterisk 1.4.0, IAX2 extensions won't ring for a very silly reason (already fixed in the svn) |
| 14:19.48 | *** join/#asterisk toot (i=chris@84.19.254.50) |
| 14:20.18 | *** join/#asterisk apardo (n=apardo@87.217.144.32) |
| 14:20.29 | *** join/#asterisk lerat (n=dnormand@bas2-montreal19-1177749681.dsl.bell.ca) |
| 14:20.40 | lerat | i everyone |
| 14:20.52 | lerat | i have a SIP problem |
| 14:21.03 | lerat | here it is |
| 14:21.24 | lerat | Peer 'xxx' is now UNREACHABLE! Last qualify: 72 |
| 14:21.34 | toot | hey folks - does dustin (vecsector/phonecall) still come on here? trying to track him down :) cheers |
| 14:21.36 | lerat | Can anybody help ???? |
| 14:21.54 | Flusher | hi |
| 14:21.55 | *** join/#asterisk Hello2007 (n=wardeh20@mail.splendor.net) |
| 14:22.55 | Katty | morning. |
| 14:23.20 | Flusher | i'd like to make asterisk answering and playing music (mp3) while dialing internal sip phones (and going to answering machine on no answer) : what's the key command please ? |
| 14:24.09 | Katty | is everyone having a beautiful, rainy, morning? |
| 14:24.37 | lerat | no for me it s a snowy morning! |
| 14:24.44 | coppice | well its night, and dry, but wifey is beautiful :-) |
| 14:24.57 | Ahrimanes | Katty, grey and rainy afternoon :) |
| 14:25.15 | DrukenLPY | Katty:, uhm... if you mean freezing rain, mixed with ice pellets and snow... then yes... |
| 14:25.20 | tzafrir | lerat, I guess that the peer is unreachable. A registration to it has failed? |
| 14:25.24 | Katty | well that's just peachy. |
| 14:25.53 | tzafrir | run 'sip show registry' . What is the line for that peer? |
| 14:26.50 | lerat | tzafrir, i had a complete office with that trouble |
| 14:27.00 | lerat | at the same time |
| 14:27.26 | Katty | and might i say, from the bottom of my heart.... /train > * |
| 14:27.38 | tzafrir | lerat, I'm not a big sip expert, just with the habbit of asking questions... |
| 14:27.43 | lerat | tzafrir, for about 1 min |
| 14:28.15 | DrukenLPY | lerat: what do you have the qualify= set to ? |
| 14:28.19 | DrukenLPY | yes? or a number? |
| 14:29.33 | Ahrimanes | lerat, sounds a lot like the network connection to that office might have been down... |
| 14:29.43 | lerat | DrunkenLPY , registry = yes and the default registration period is 2000 |
| 14:30.18 | DrukenLPY | lerat: that's not that i asked... i asked about the qualify |
| 14:30.34 | DrukenLPY | that = what |
| 14:31.22 | lerat | qualified = yes |
| 14:32.53 | *** join/#asterisk simplexio (n=sim@ogw.oamk.fi) |
| 14:33.27 | lerat | DrunkenLPY, i just need to put a check for the qualified and then enter a time value |
| 14:34.38 | lerat | DrunkenLPY, so i set it to qualify= yes and 2000 ms for the time value |
| 14:36.02 | lerat | does it ring you anything ???? |
| 14:36.05 | Hello2007 | hi, everybody |
| 14:36.20 | Hello2007 | anyone know a softphone that support h264 |
| 14:37.05 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-221.ph.ph.cox.net) |
| 14:37.46 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 14:42.13 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 14:44.34 | tzanger | ping |
| 14:44.40 | [TK]D-Fender | pong |
| 14:45.36 | simplexio | ehi. anyidea how i could make queue for queue ? |
| 14:47.14 | [TK]D-Fender | simplexio: HUH? |
| 14:48.06 | *** join/#asterisk cnet2 (n=nada@190.10.0.120) |
| 14:48.25 | Dr-Linux | anybody is using IAX trunk between 2 servers? |
| 14:48.46 | Marty-OTT | b11d: you around? |
| 14:49.26 | *** join/#asterisk tzanger (n=tzanger@208.68.91.47) |
| 14:49.26 | cnet2 | hi, I have a sipura 3102 reg to a *, and i wanted to foward pstn lines to a voip softphone, with no extra dialing.. (i was thinking of DP at the sipura conf, don't know how.) also tried s,1,Dial at the context..) |
| 14:49.45 | simplexio | [TK]D-Fender: :) |
| 14:50.01 | [TK]D-Fender | Katty: Mew. |
| 14:50.07 | [TK]D-Fender | Katty: Got speakers there yet? |
| 14:50.21 | Dr-Linux | [TK]D-Fender: i'd need your suggestion, there are 4 examples. Which one you would suggest for iax trunk between 2 asterisk servers 1.2.x |
| 14:50.22 | Dr-Linux | http://www.voip-info.org/wiki/view/Asterisk+-+dual+servers |
| 14:50.43 | simplexio | [TK]D-Fender: cant add new member in realtime to queue so that new memeber is shown for old memeber is queue, so solution make queue for queue |
| 14:50.46 | [TK]D-Fender | Dr-Linux: Tons of people. Thts sort of the point of IAX2 you know. |
| 14:51.11 | Dr-Linux | hhm.. |
| 14:51.23 | Dr-Linux | [TK]D-Fender: but i'm confused which one i should try now :S |
| 14:51.41 | Dr-Linux | and which method is recommended , since i don't know IAX |
| 14:51.54 | [TK]D-Fender | simplexio: You can't queue up a queued call. It always counts anss answered. |
| 14:51.58 | Dr-Linux | How about Example 2 ? |
| 14:52.07 | [TK]D-Fender | Dr-Linux: Just Do It (tm) |
| 14:52.24 | Dr-Linux | hhm.. ok |
| 14:57.03 | *** join/#asterisk bkw__ (n=brian@truphone.plus.com) |
| 14:57.07 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 15:00.34 | *** join/#asterisk JuanKerr (n=gouranga@c-71-236-93-168.hsd1.pa.comcast.net) |
| 15:01.29 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:01.33 | blitzrage | morning all |
| 15:01.50 | *** part/#asterisk lerat (n=dnormand@bas2-montreal19-1177749681.dsl.bell.ca) |
| 15:03.02 | [TK]D-Fender | blitzrage: ! ! ! |
| 15:03.24 | blitzrage | is there the ability to have Voicemail() set the permissions of the files it's writing to the filesystem? i.e. I want -rwxr--r-- instead of just -rwx------ |
| 15:05.10 | *** join/#asterisk andew (n=andew@84-45-170-202.no-dns-yet.enta.net) |
| 15:05.30 | Ahrimanes | blitzrage, hm that's doable with filesystem settings |
| 15:05.32 | Ahrimanes | umask |
| 15:07.18 | *** part/#asterisk slima (i=slima@unaffiliated/slima) |
| 15:07.47 | *** join/#asterisk slima (i=slima@unaffiliated/slima) |
| 15:07.53 | blitzrage | Ahrimanes: ahhh... so if I run umask on a directory, then that will set the permissions files in that directory should be created with? |
| 15:08.10 | neuwald | folks |
| 15:08.20 | neuwald | I'm trying to make a local sip call here and I got: |
| 15:08.21 | neuwald | <PROTECTED> |
| 15:08.31 | neuwald | anybody knows what does it should be? |
| 15:08.35 | Ahrimanes | blitzrage, yeah, that's what umask is for afaik :) |
| 15:08.52 | blitzrage | Ahrimanes: ok -- I read man umask, and it wasn't super obvious that's what it did :) |
| 15:09.19 | Ahrimanes | blitzrage, http://www.tech-faq.com/umask.shtml :) |
| 15:09.40 | blitzrage | Ahrimanes: merci! |
| 15:09.53 | *** join/#asterisk olivier__ (n=olivier@obs92-4-82-239-116-113.fbx.proxad.net) |
| 15:10.55 | Ahrimanes | blitzrage, yw :) |
| 15:10.55 | Marty-OTT | Hey... has anyone ever setup an ATA to make several phones in a house ring? |
| 15:11.51 | [TK]D-Fender | neuwald: ... |
| 15:11.54 | [TK]D-Fender | ~freepbx |
| 15:11.55 | jbot | somebody said freepbx was unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 15:11.58 | tzafrir | blitzrage, you should set the umask of the Asterisk process |
| 15:12.12 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 15:12.21 | [TK]D-Fender | Marty-OTT: Plent of people, whats your specific question? |
| 15:12.27 | blitzrage | tzafrir: oh yah? |
| 15:12.52 | blitzrage | that'll basically tell the system to set the permissions as what I state in the umask when that process creates a file? |
| 15:13.11 | Marty-OTT | TK: Well, usually ATAs are used to plug in just that one analog phone. Instead of plugging in one analog phone, I'm thinking.. could I just hook it up to hte home's phone network to make all the phones ring? |
| 15:14.03 | tzafrir | "-rwxr--r-"? to a file or a directory? |
| 15:15.44 | [TK]D-Fender | Marty-OTT: You'd have to unplug your home from the old telco completely or you'll fry it when the telco rings. |
| 15:17.19 | Marty-OTT | Yeah, that's what I gathered but otherwise, it should work.. |
| 15:17.48 | [TK]D-Fender | Marty-OTT: That is how places like Vonage tell consumers to install it to replace their old home service. Keep in mind the REN rating of the ATA your using and the # of phones you're going to place on it. |
| 15:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 15:18.03 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 15:18.43 | Marty-OTT | I'm deliberating again... (talking with qdk) for an appartment building... do the: Channel Bank + Asterisk or give VOIP service straight to the appartment (but if they have 2 phones, we have to provide 2 phones or ATA) |
| 15:18.54 | blitzrage | tzafrir: yah, I basically want the voicemail files in the /var/spool/asterisk/voicemail directory set to -rwxr--r-- when they are created / modified by asterisk |
| 15:21.01 | *** join/#asterisk expat_iain (n=iain@217.15.107.200) |
| 15:22.02 | Flusher | Hi again. Is there a way to make Asterisk playing (mp3) music rather than a ring tone at caller side when he's calling please ? It would be a soft of "automatic" MoH. |
| 15:22.17 | blitzrage | Flusher: show application dial |
| 15:22.41 | ChicagoBud | For a small office that needs 4 fxo would you go with a card in a pc or an extrenal gateway? |
| 15:22.53 | blitzrage | assuming you are doing this when calling another device attached to asterisk |
| 15:24.54 | qdk | Marty-OTT: you either have a channel bank OR ATAs for your old phone or just IP-phones, no matter what option you pick asterisk will still be able to dial them all at the same time. |
| 15:25.20 | Marty-OTT | yes, but if I don't get channel banks... |
| 15:26.03 | [TK]D-Fender | Marty-OTT: All ATA's typically support at least 5 phones worth of REN. |
| 15:26.33 | Marty-OTT | See, if an appartment has say 2 phones... and we give one voip phone (wired) but they also need another voip phone in another room - it would have to be wireless? If I did a straight voip service, we'd have to do an ATA |
| 15:26.41 | [TK]D-Fender | ChicagoBud: 4 Port TDM w/ HWEC. |
| 15:27.23 | ChicagoBud | <[TK]D-Fender>Such as? |
| 15:27.26 | [TK]D-Fender | Marty-OTT: For an appartment building you don't WANT to put VoIP phones in the units. Use an analog port and let them use normal phones or you'll freak people out. |
| 15:27.36 | [TK]D-Fender | ChicagoBud: Sangoma A200d |
| 15:28.08 | Marty-OTT | yeah.. |
| 15:28.15 | Marty-OTT | agreed |
| 15:28.52 | [TK]D-Fender | Marty-OTT: AND drive up costs. |
| 15:29.22 | ChicagoBud | <[TK]D-Fender> Thanks. Are they pretty friendly to install in most PCs? Any PCI issues? |
| 15:29.54 | [TK]D-Fender | Marty-OTT: What CAN do (and perhaps SHOULD), ensure they have cat5e to all jacks, and in your basement patch-panel, overlay 10/100 on the std 2 pairs, and use the midlle for RJ11. |
| 15:30.15 | [TK]D-Fender | ChicagoBud: None I've ever heard of. They share interrupts rather nicely. |
| 15:30.29 | ChicagoBud | <[TK]D-Fender>Nice. |
| 15:31.35 | ChicagoBud | <[TK]D-Fender>Will I need hardware echo cancel via the optional DSP card? |
| 15:33.34 | [TK]D-Fender | ChicagoBud: Yes you definately want the on-board EC version (with the "d" on the end) |
| 15:34.03 | danp | i love the sangoma card i'm using |
| 15:34.19 | *** join/#asterisk sasch (n=sasch@host102-30-static.107-82-b.business.telecomitalia.it) |
| 15:34.33 | *** join/#asterisk reber (i=reber@gateway/tor/x-30fc9a28a34496c8) |
| 15:34.37 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 15:34.37 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:35.01 | *** join/#asterisk ManxPower (n=manxpowe@116.sub-75-202-33.myvzw.com) |
| 15:39.07 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:40.04 | Marty-OTT | TK: Hmmm... you mean put regular Ethernet on the 2 pairs going to the appartments? It's Cat3 to the apparments. |
| 15:41.45 | Marty-OTT | However, that being said, I was thinking .. if we really didn't want to use a Channel bank, maybe we could terminate the apartements on an ATA in the basement.... that would remove the channel bank and a voip gateway at each location |
| 15:42.20 | Marty-OTT | thing is... does the ATA power enough to reach from the basement of a building all the way to a 20th floor appartment for ecample? |
| 15:42.52 | *** join/#asterisk mkl1525 (n=qwertz@35.206.27.217.static.versanetonline.de) |
| 15:43.14 | mkl1525 | Hi, does anybody know what the "announceoverride" option in the queue app is good for? |
| 15:43.21 | [TK]D-Fender | Marty-OTT: You DON'T typically want to use a channel bank. Use a 24 port SIP gateway |
| 15:43.45 | Dr-Linux | [TK]D-Fender: i'm trying iax2 trunk |
| 15:43.53 | Dr-Linux | [TK]D-Fender: please have a look: http://phpfi.com/193887 |
| 15:43.54 | [TK]D-Fender | Marty-OTT: And I mean to WIRE a split of the 4 pairs going to the unit at the BASEMENT side. |
| 15:44.05 | [TK]D-Fender | Marty-OTT: That way you can offer hybrid services |
| 15:44.06 | Dr-Linux | [TK]D-Fender: looks like some codec issue or something |
| 15:44.39 | [TK]D-Fender | Dr-Linux: "Jan 15 20:38:13 WARNING[18250]: chan_iax2.c:7900 iax2_request: Unable to create translator path for unknown to g729 on IAX2/ca-pbx-6" <- you think? |
| 15:44.45 | Marty-OTT | We had probblems with the 24 port Rj21 connectors on voip gateways we tested anyways so far. We're considering trying the CAC ADIT 600 - it talks to Asterisk via MGCP |
| 15:44.58 | [TK]D-Fender | Dr-Linux: Pick a built-in codec for your testing and just keep with it. |
| 15:45.04 | Marty-OTT | and it has a "built-in" channel balnk |
| 15:45.05 | Marty-OTT | bank |
| 15:45.27 | Dr-Linux | hhm.. |
| 15:45.33 | Dr-Linux | [TK]D-Fender: ok |
| 15:45.41 | [TK]D-Fender | Marty-OTT: I cant speak for MGCP's functionalitty, but its a dying breed... avoid if possible |
| 15:45.59 | [TK]D-Fender | Marty-OTT: I've had great success with the Mediatrix 1124. Very nice unit |
| 15:46.14 | Sweeper | I think D-fender works for mediatrix :P |
| 15:46.19 | Sweeper | or at least gets commission |
| 15:46.31 | *** join/#asterisk s1gny|wrk (n=s1gny@p54915827.dip.t-dialin.net) |
| 15:46.39 | *** part/#asterisk s1gny|wrk (n=s1gny@p54915827.dip.t-dialin.net) |
| 15:46.49 | Sweeper | RJ-21? huh? |
| 15:47.19 | fetcher | RJ-21X == 50-pin Amphenol, right? |
| 15:47.21 | [TK]D-Fender | Sweeper: No, you're supposed to think that of my relationship with POLYCOM! You've got your FUD all screwed up! |
| 15:47.28 | [TK]D-Fender | fetcher: correct |
| 15:47.41 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 15:48.08 | Sweeper | oh, that thing |
| 15:48.24 | Sweeper | that's what the ADIT600 has on the back of it |
| 15:48.32 | Sweeper | well, it has two of them |
| 15:48.48 | Sweeper | never had problems with the connector tho... |
| 15:49.58 | Marty-OTT | TK: Someone here th eother day complained about the Mediatrix and we were going to ship it back... you can vouch for it? We have that exact unit but neveer got around to testing it. |
| 15:51.03 | Sweeper | TK will vouch for mediatrix all day :P |
| 15:51.21 | Marty-OTT | Sweeper: Have you used it? |
| 15:51.26 | Sweeper | nope! |
| 15:51.36 | [TK]D-Fender | Marty-OTT: I've only worked with 1, but it worked great. |
| 15:51.43 | Sweeper | all I use are t1 cards + channel banks of one sort or another |
| 15:51.54 | ManxPower | I thought RJ-21X was commonly known as "66 block" |
| 15:52.07 | Marty-OTT | hmmmm.... Well that's my connundrum... elimate the channel banks or not.... |
| 15:52.08 | Sweeper | 66 block is the punchdown block |
| 15:52.23 | ManxPower | I never order them, I just steal unused ones from customers |
| 15:52.40 | simplexio | shouldnt manager api work with telnet connection ? just sendinf thise Action: login and so on ? |
| 15:52.40 | Nugget | telnet is eeeeeeevil! |
| 15:52.47 | Marty-OTT | TK - and it's stable? no crashes? This is for a live deployment. |
| 15:53.00 | [TK]D-Fender | Marty-OTT: Nothing I've heard of including other users... |
| 15:53.04 | [TK]D-Fender | Marty-OTT: Google it.... |
| 15:53.14 | Sweeper | you can get a 66 block that comes with an RJ-21 wired into it |
| 15:53.15 | Marty-OTT | i will - hmmm. |
| 15:53.26 | [TK]D-Fender | simplexio: Yes, thats the point. Go read about it in THE BOOK, and on the WIKI |
| 15:53.30 | [TK]D-Fender | ~book |
| 15:53.37 | jbot | i guess book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 15:53.38 | [TK]D-Fender | ~wikis |
| 15:53.39 | jbot | i guess wikis is http://www.voip-info.org |
| 15:53.42 | Marty-OTT | TK: it would make my life simpler... |
| 15:53.56 | Marty-OTT | tk: works good with Asterisk? |
| 15:54.16 | ManxPower | simplexio: yes, it should, but not in some versions of 1.4 |
| 15:54.38 | ManxPower | Sweeper: yes. I have several |
| 16:00.49 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 16:02.44 | Sweeper | Jan 15 10:05:30 WARNING[7533]: chan_zap.c:10875 setup_zap: Ignoring signalling <-- um, huh? these are pri channels, YOU NEED SIGNALLING |
| 16:03.17 | tzafrir | Sweeper, is this a reload? |
| 16:03.21 | Sweeper | yea |
| 16:03.38 | [TK]D-Fender | Marty-OTT: Thats what I used it with... |
| 16:03.41 | wunderkin | signalling = smoke |
| 16:03.47 | tzafrir | chan_zap cannot change signalling on reload. And it is verbose enough to let you know |
| 16:03.52 | Sweeper | aha! |
| 16:03.54 | Sweeper | good to know |
| 16:03.57 | ManxPower | Sweeper: You cannot change the signaling when you do a reload, therefore that option is ignored on a reload |
| 16:04.36 | simplexio | well if manager api answer, reason is that port or source address i denied, and all those Action: login [enter] Username: plop [enter] Secret: secret [Enter] ActionID: 1 ... al i get from those command is auth failed |
| 16:04.57 | ManxPower | simplexio: What version of Asterisk? |
| 16:05.08 | [TK]D-Fender | simplexio: What port are you using? |
| 16:05.14 | wunderkin | you need another new line after each block, so after the secret you need to press enter again |
| 16:05.31 | simplexio | correct one if it answers Escape character is '^]'. |
| 16:05.31 | simplexio | Asterisk Call Manager/1.0 |
| 16:05.31 | wunderkin | ooh.. |
| 16:05.44 | wunderkin | actionid with a login .. ? |
| 16:05.45 | ManxPower | simplexio: We do not support Asterisk Call Manager here. |
| 16:06.23 | ManxPower | We do support the Asterisk Manager Interface, but that was not my question. My question was "What version of ASTERISK are you using?" |
| 16:06.51 | [TK]D-Fender | simplexio: And my question was "Whic port"? |
| 16:06.57 | tzafrir | asterisk -rx 'show version' |
| 16:07.15 | tzafrir | [TK]D-Fender, why is that relevant? |
| 16:07.15 | *** join/#asterisk Qwell[] (i=qwell@unaffiliated/qwell) |
| 16:07.15 | *** mode/#asterisk [+o Qwell[]] by ChanServ |
| 16:07.15 | ManxPower | Also, in a recent release of Asterisk you could not log into the manager interface because of a bug |
| 16:07.17 | simplexio | <PROTECTED> |
| 16:07.24 | tzafrir | [TK]D-Fender, he gets a reply |
| 16:07.25 | *** join/#asterisk marv[work] (n=timr@24.214.206.254) |
| 16:07.31 | ManxPower | simplexio: 1.2.13 does NOT have the bug I am referring to. |
| 16:07.41 | [TK]D-Fender | tzafrir: By his wording I'd lay bets we was trying to connect to AMI through Port 21 :) |
| 16:07.50 | simplexio | i get answer from that port |
| 16:08.00 | [TK]D-Fender | tzafrir: Ok, MAYBE I missed that part :) |
| 16:08.04 | ManxPower | simplexio: Asterisk Manager Interface does not run on port 21 |
| 16:08.10 | *** part/#asterisk Hello2007 (n=wardeh20@mail.splendor.net) |
| 16:08.20 | simplexio | and port is 5038.. but that sould not be a problem |
| 16:09.37 | *** join/#asterisk ellisdee (n=ellisdee@69.15.174.114) |
| 16:10.15 | ellisdee | what would cause polycom 501 telephones to not be able to perform "blind transfers" between each other. |
| 16:10.29 | ManxPower | simplexio: pastebin what you get. pastebin.ca |
| 16:10.32 | ellisdee | sip v2.0.1 |
| 16:10.38 | ellisdee | ast 1.2.7.1 |
| 16:10.39 | ManxPower | ellisdee: not pressing "blind" soft button during transfer |
| 16:10.55 | ellisdee | lol |
| 16:10.58 | [TK]D-Fender | ellisdee: Thats a new one... what happens? |
| 16:11.09 | ellisdee | incoming call. |
| 16:11.18 | ellisdee | user answers call |
| 16:11.40 | ellisdee | user presses transfer, blind transfer, extension #, and then the call is dropped. |
| 16:11.46 | danp | hmm |
| 16:12.11 | ellisdee | bootrom 3.2.2 |
| 16:12.22 | ManxPower | ellisdee: that is odd. Did it ever work? |
| 16:12.26 | danp | i have users doing that...on 601's, also using 2.0.1. using a very recent build of 1.2-stable from svn |
| 16:12.28 | ManxPower | Is there any messages on the CLI? |
| 16:12.51 | ellisdee | i would have to get the end user to duplicate the problem. |
| 16:12.53 | ManxPower | My users do about 50 - 80 blind transfers per day with 1.2.13 and Polycoms 1.6.7 |
| 16:13.12 | ellisdee | but i have never seen this one before. |
| 16:13.16 | danp | definitely check the console |
| 16:14.10 | Dr-Linux | Parsing '/etc/asterisk/iax.conf': Found |
| 16:14.10 | Dr-Linux | Jan 15 08:12:45 NOTICE[10279]: chan_iax2.c:8613 set_config: Ignoring bindport on reload |
| 16:14.20 | Dr-Linux | what's wrong :S |
| 16:15.43 | wunderkin | probably refer to the response to Sweeper |
| 16:15.43 | simplexio | ManxPower: http://pastebin.ca/317393 |
| 16:16.18 | ManxPower | Dr-Linux: you can't change the bindport on reload? |
| 16:17.14 | Dr-Linux | ManxPower: i don't know about iax2, i just edit the iax.conf file and now reloaded and found this notice |
| 16:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 16:18.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 16:18.19 | *** join/#asterisk adde (i=daemon@1-1-1-25a.vhe.sth.bostream.se) |
| 16:18.32 | *** join/#asterisk bkw_ (n=brian@truphone.plus.com) |
| 16:18.35 | Dr-Linux | ManxPower: please have a look >> http://phpfi.com/193887 |
| 16:18.56 | *** join/#asterisk codefreeze (n=steve_mu@216.166.159.235) |
| 16:19.16 | adde | I am new to Asterisk and want to install it... What distribition is recomended to perpare the system with? |
| 16:20.18 | in-pt | adde: i used fc3 and fc5 |
| 16:21.39 | Nugget | Asterisk doesn't care. |
| 16:22.02 | Nugget | Heck, you don't even need to run Linux. Anything remotely POSIX is perfectly fine unless you need to support digium hardware. |
| 16:22.27 | Nugget | On a personal note, I'd suggest you select a distribution that includes a spell checker. ;) |
| 16:22.28 | Corydon-w | Except Windows |
| 16:22.33 | simplexio | Gentoo ad Suse seems to work pretty nice |
| 16:22.35 | ellisdee | erm |
| 16:22.41 | ellisdee | had the end user duplicate the problem |
| 16:23.09 | monsted | Corydon-w: well, windows isn't terribly POSIX'y :) |
| 16:23.16 | ellisdee | end user presses transfer, i get musiconhold, user press blind transfer, types extension, and i get dropped. |
| 16:23.24 | simplexio | adde: run it on system wich you know best. as long as it isnt windows :) |
| 16:23.41 | Corydon-w | monsted: actually, it is remotely POSIX compliant |
| 16:28.47 | tzafrir | adde, if you're familiar with a certain linux distro, use it. If not, use $tzafrirs_favorite_distro |
| 16:30.55 | Nugget | because that way tzafrir wins the internet. |
| 16:32.49 | Sweeper | um, tip-tip/ring-ring, or tip-ring/ring-tip, for a t1? |
| 16:33.04 | *** join/#asterisk thoughtpolice (n=austin@ip70-185-140-61.lu.dl.cox.net) |
| 16:33.19 | Marty-OTT | Hey, check out this EXCELLENT link on using Asterisk with Channel Banks - it's absolutely superb! |
| 16:33.21 | Marty-OTT | http://www.voip-info.org/tiki-index.php?page=Asterisk%20Channel%20Bank |
| 16:33.49 | Qwell[] | Sweeper: I thought tip/ring was an analog thing... |
| 16:34.26 | *** join/#asterisk cbullock81 (n=cbullock@adsl-068-213-099-052.sip.jan.bellsouth.net) |
| 16:36.19 | ManxPower | Qwell: The terms are also used in T-1s |
| 16:36.23 | Qwell[] | silly |
| 16:36.25 | [TK]D-Fender | Sweeper: Typically an RJ45 cat5 straight cable |
| 16:36.42 | Sweeper | mm |
| 16:36.56 | Sweeper | well, I'm crossconnecting two t1 ports for testing |
| 16:37.10 | Marty-OTT | TK: You said MGCP was going out the door but I'm reading rave reviews on the CAC ADIT 600 not to mention the notes in the link i just pasted. |
| 16:37.21 | [TK]D-Fender | Sweeper: In that case you need to make a crossover cable. Go google it |
| 16:37.26 | Marty-OTT | And it uses MGCP to talk to Asterisk though.. |
| 16:37.30 | wunderkin | t1 crossover |
| 16:37.31 | Sweeper | been there, done that |
| 16:37.45 | Sweeper | just crimped last connector :P |
| 16:38.37 | in-pt | hi all |
| 16:38.45 | in-pt | does asterisknow supports skinny channels |
| 16:39.12 | [TK]D-Fender | in-pt: Not the place to ask.... try #asterisk-gui , or just tryit yourself. |
| 16:39.12 | Qwell[] | in-pt: it does, but the gui does not |
| 16:39.45 | in-pt | ok thnx |
| 16:40.26 | Dr-Linux | Jan 15 08:38:58 NOTICE[25129]: chan_iax2.c:5041 register_verify: Peer 'lhr-pbx' is not dynamic (from 202.125.141.10) |
| 16:40.31 | Dr-Linux | what's wrong? :S |
| 16:40.55 | *** join/#asterisk samy_b1 (i=samy_b1@2001:49f0:1000:0:0:0:0:7) |
| 16:41.50 | cbullock81 | i have an MWI question. What should you have in SIP.conf to enable MWI? |
| 16:42.01 | file | I count 2 times so far that I look in here and the message tells you |
| 16:42.50 | *** join/#asterisk mivck (i=1000@ip-70-228.telesat.com.co) |
| 16:43.47 | ManxPower | cbullock81: mailbox=voicemailbox@voicemailcontext under each SIP device |
| 16:44.20 | Sweeper | AWSOME |
| 16:44.25 | ManxPower | Dr-Linux: You do not have host=dynamic but the remote device is still trying to register |
| 16:44.31 | Sweeper | so I plug my crossconnect cable in, FREEZES THE SYSTEM |
| 16:44.36 | ManxPower | if a device registers then you want host=dynamic |
| 16:44.50 | Qwell[] | Sweeper: is it a T1 crossover cable? |
| 16:44.55 | Sweeper | yea it is |
| 16:44.58 | [TK]D-Fender | If you have fixed IP, you shouldn't BOTHER with registereing |
| 16:45.01 | Qwell[] | because it's very different from RJ45 crossover |
| 16:45.11 | Dr-Linux | ManxPower: yes that's done |
| 16:45.18 | ManxPower | Qwell: you mean very different from an ethernet crossover |
| 16:45.24 | Dr-Linux | a little issue can't find the solution |
| 16:45.25 | Qwell[] | T1 != RJ45 |
| 16:45.29 | *** join/#asterisk mog (n=mog@c-71-207-215-93.hsd1.al.comcast.net) |
| 16:45.29 | *** mode/#asterisk [+o mog] by ChanServ |
| 16:45.31 | Dr-Linux | ManxPower: please have a look >> http://phpfi.com/193887 |
| 16:45.32 | Sweeper | um |
| 16:45.37 | Qwell[] | RJ45 has a specific pin layout |
| 16:45.39 | Sweeper | rj45 is a connector |
| 16:45.46 | ManxPower | Qwell: NO IT DOES NOT |
| 16:45.58 | ManxPower | RJ-45 is a type of connector. It does NOT specify pinouts |
| 16:46.01 | Sweeper | see, rj stands for "registered jack" |
| 16:46.04 | Qwell[] | Isn't T1 like RJ48? |
| 16:46.13 | monsted | RJ45 is just a row of pins :) |
| 16:46.18 | ManxPower | Qwell: T-1s can have coax connectors too |
| 16:46.28 | Qwell[] | ManxPower: okay, I'll give you that |
| 16:46.30 | Sweeper | a t1 just needs 4 conductors.... |
| 16:46.47 | Sweeper | bantam plugs, rj45, coax, whatever |
| 16:47.17 | ManxPower | perhaps you are thinking of TIA-568A/TIA-568B ( I think its 568) |
| 16:47.26 | cbullock81 | ManxPower: thanks |
| 16:47.38 | Sweeper | at any rate |
| 16:47.48 | Qwell[] | "RJ45": 8P8C, informal designation for T568A/T568B, including Ethernet; not the same as the true RJ45/RJ45S |
| 16:48.10 | Qwell[] | listed under "Unofficial" (incorrect) plug names |
| 16:48.11 | Sweeper | my cable goes like so: 1->4, 2->5, 4->1, 5->2 |
| 16:48.25 | ManxPower | Sweeper: you need a straight thru cable for telco/Asterisk, T-1 crossver for Asteirsk/channelbank |
| 16:48.31 | Qwell[] | The true RJ45(S) is an extremely uncommon registered jack, but the name "RJ45" is also used quite commonly to refer to any 8P8C modular connector. |
| 16:48.39 | Qwell[] | so, nyah nyah :P |
| 16:48.43 | Sweeper | I'm going from asterisk to asterisk |
| 16:48.59 | RoyK | http://img.4chan.org/b/src/1168876307109.jpg |
| 16:49.10 | ManxPower | Sweeper: then you need T-1 crossover. No matter what it should not lock Asterisk. You have some OTHER problem in Asterisk |
| 16:49.10 | Sweeper | 4 ports, setting one as master, one as slave, and connecting them for a bit of testing |
| 16:49.18 | *** part/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 16:49.26 | Sweeper | really? cause it locks up the instant I plug the cable in |
| 16:49.36 | Sweeper | did it twice already |
| 16:49.43 | Sweeper | total lock man |
| 16:50.06 | Sweeper | three fingered salute doesn't work, no ttys, nothing |
| 16:50.16 | [TK]D-Fender | Sweeper: Cool! Dit it again, I wanna watch! |
| 16:50.29 | [TK]D-Fender | do* |
| 16:51.06 | cbullock81 | D-Fender: can I bother you again with a polycom phone question |
| 16:52.38 | Sweeper | nothing in /var/log/ either :/ |
| 16:52.40 | [TK]D-Fender | cbullock81: shoot |
| 16:53.31 | [TK]D-Fender | Sweeper: if(crash && tryingt1) { dontdreamofgettingalog ==1; } |
| 16:53.41 | cbullock81 | D-Fender: Ok. I think i have everything setup properly in sip.conf and voicemail.conf, but i cannot get the MWI light blinking on the polycom phone... i have added what I thought was the necessary info in the polycom .cfg files, but still no luck |
| 16:54.08 | [TK]D-Fender | cbullock81: You don't HAVE to set anything on the phone. its all in your sicfg, and voicemail.conf |
| 16:54.13 | Sweeper | I suppose :/ |
| 16:54.45 | cbullock81 | d-fender: hmmm. i thought you had to setup mwi stuff in the phone... what are the required settings in sip.conf? |
| 16:57.57 | Hmmhesays | mailbox=123456@my_vm_context |
| 16:58.08 | [TK]D-Fender | cbullock81: ManxPower already TOLD you. Pay attention. |
| 16:58.10 | cbullock81 | i have that, but still no go |
| 16:58.18 | cbullock81 | i read what he said... its just not working for me |
| 16:58.26 | [TK]D-Fender | Hmmhesays: Go download my latest crap! http://aocomputing.net/sweeps1.mp3 |
| 16:58.38 | *** join/#asterisk bkw_ (n=brian@truphone.plus.com) |
| 16:58.48 | [TK]D-Fender | cbullock81: Pastebin your configs, don't assume we trust you did it right. |
| 16:59.00 | [TK]D-Fender | cbullock81: Because for the record, we never do :) |
| 16:59.06 | wunderkin | also make sure you do a sip reload |
| 16:59.28 | cbullock81 | k |
| 17:00.29 | *** join/#asterisk alamantia (i=Anthony@nat/digium/x-0440346151ebe024) |
| 17:01.04 | Dr-Linux | not sure wtf is this >> http://phpfi.com/193887 |
| 17:03.32 | *** join/#asterisk alamantia (i=Anthony@nat/digium/x-49ff8d519712e006) |
| 17:05.03 | xheliox | Does anyone have any (extensive) experience with the Patton SmartNode 4900? I'd be willing to pay someone for an hour of their time to bend their ear a bit about it. |
| 17:05.05 | cbullock81 | i've never used pastebin... does it usually take a while to submit |
| 17:05.35 | [TK]D-Fender | cbullock81: not long if you're using .ca |
| 17:07.11 | cbullock81 | ah. thx |
| 17:07.11 | *** join/#asterisk Nukemizer (n=Nuke@67.137.28.165) |
| 17:07.13 | cbullock81 | ok here it is... now i will preface this by letting you know I'm very new w/ asterisk, so if my .cfgs are screwball, thats why :) http://www.pastebin.ca/317438 |
| 17:08.11 | *** join/#asterisk qdk (n=qdk@0x535eae17.boanxx9.adsl-dhcp.tele.dk) |
| 17:09.14 | [TK]D-Fender | cbullock81: remove all of these : subscribemwi=yes |
| 17:10.09 | samy_b1 | hey is there a way to protect my voip calls for been screened ? |
| 17:10.49 | cbullock81 | k |
| 17:11.03 | samy_b1 | i know there is a way to tcpdump and convert them back to sound |
| 17:12.13 | *** join/#asterisk bkw_ (n=brian@truphone.plus.com) |
| 17:12.26 | cbullock81 | still no go so far, but is it correct that when I try to check messages on the phone that it calls my extension |
| 17:13.54 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 17:14.51 | *** join/#asterisk alamantia (i=Anthony@nat/digium/x-13cd9a38232a71f1) |
| 17:14.58 | [TK]D-Fender | cbullock81: MWI light has nothing to do with the "messages" button on your phone dialing what it will |
| 17:15.24 | [TK]D-Fender | cbullock81: To change the number that it dials, THAT is something to set in sip.cfg, but not at all necessary |
| 17:15.37 | andrebarbosa | It's possible to change the To field on SIP Header? |
| 17:15.49 | andrebarbosa | It's needed to implement call forward |
| 17:15.50 | andrebarbosa | :s |
| 17:15.54 | Dr-Linux | what's the difference between sipura 2000 and sipura 21000? |
| 17:15.57 | Dr-Linux | sory |
| 17:16.00 | Dr-Linux | 2100 |
| 17:16.12 | andrebarbosa | we need to keep original To dont we? |
| 17:16.53 | cbullock81 | D-Fender: thanks. I know i'm bugging the heck out of you right now & I apologize :) But do you have any other suggestions as to why mwi isint working |
| 17:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 17:18.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 17:18.36 | [TK]D-Fender | cbullock81: dump "show voicemail" and confirm that there ARE new VM's to be announced, and maybe reboot a phone after that. |
| 17:19.46 | cbullock81 | hey! it just beeped at me :) |
| 17:19.48 | cbullock81 | it worked |
| 17:19.52 | Dr-Linux | what's the difference between sipura 2000 and sipura 2100? |
| 17:19.54 | cbullock81 | i was not waiting long enough |
| 17:20.02 | cbullock81 | is there a way to make it check for messages sooner? |
| 17:23.21 | [TK]D-Fender | cbullock81: Here, because I'm feeling generous today : http://www.pastebin.ca/317460 |
| 17:25.52 | *** join/#asterisk jarg (n=jarg@200.56.225.61) |
| 17:26.55 | Qwell[] | [TK]D-Fender: What DOES the TK stand for? |
| 17:27.03 | Qwell[] | gamer clan? heh |
| 17:27.19 | [TK]D-Fender | Qwell[]: My old Actio:Half-Life clan. |
| 17:27.23 | [TK]D-Fender | Action* |
| 17:27.24 | [TK]D-Fender | yup |
| 17:27.41 | [TK]D-Fender | sure we are half/decade defunct, but I still wear it :) |
| 17:27.46 | Qwell[] | :p |
| 17:28.53 | [TK]D-Fender | Qwell[]: "Team Killers". We were very trigger happy and even in team-play, its a dog-eat-dog game, so if you mow down a teammate instintually to ensure your survival we did it with STYLE :) |
| 17:29.45 | [TK]D-Fender | Qwell[]: And afterwards we took on "Tall Killers" as a pseudo-clanname as we were all over 6' :) |
| 17:30.04 | Qwell[] | So, you killed...other Tall folks? |
| 17:30.25 | [TK]D-Fender | Qwell[]: We don't discriminate :) |
| 17:30.28 | *** join/#asterisk Hmmhesays (n=Neg@24-117-135-28.cpe.cableone.net) |
| 17:30.30 | Qwell[] | heh |
| 17:30.41 | [TK]D-Fender | Qwell[]: In death, all are equal :) |
| 17:30.48 | Hmmhesays | is there a something similar to cmd mysql for postgres? |
| 17:31.02 | [TK]D-Fender | Hmmhesays: pgsql, no? |
| 17:31.10 | Qwell[] | Hmmhesays: func_odbc is way cooler, FYI |
| 17:31.23 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqia.cable.mindspring.com) |
| 17:31.47 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.149) |
| 17:31.51 | Hmmhesays | Qwell: does that allow me to execute a custom query in the dialplan and user the returned data? |
| 17:31.56 | Qwell[] | yep |
| 17:32.04 | Qwell[] | func_odbc <3 |
| 17:35.43 | `Sauron | What directory do the * sounds go in again? |
| 17:36.05 | [TK]D-Fender | `Sauron: /var/lib/asterisk/sounds |
| 17:36.21 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:37.01 | `Sauron | Thanks |
| 17:37.26 | blitzrage | afternoon all! :) If I transfer someone into a conference.. is there any method I can use to pull that user back out of the conference and transfer them somewhere else? I can use MeetMeAdmin to kick a user, so I'm curious if I can do something with that user once they are kicked, or if Asterisk just drops the channel. |
| 17:37.32 | *** join/#asterisk suge (i=sugesuge@c-66-176-120-45.hsd1.fl.comcast.net) |
| 17:38.20 | suge | Hello.. its been over a year since I've played with Asterisk. Is Centos still regarded as the most popular distro for asterisk? Or has ubuntu replaced it? |
| 17:38.29 | `Sauron | damn bloody openpbx documentation sucks |
| 17:38.32 | Qwell[] | suge: use whatever you know |
| 17:38.36 | `Sauron | I wish I could use *. |
| 17:39.09 | suge | ok.. I just want to set up the most capatible for a production box. I guess ill stick with centos |
| 17:39.28 | blitzrage | suge: I prefer CentOS, but like Qwell said, just use whatever you're comfortable with -- Asteirsk doesn't care |
| 17:40.35 | suge | cool |
| 17:42.38 | [TK]D-Fender | suge: Many * users use is and it has been popular with the all-in-one * distro's |
| 17:42.51 | [TK]D-Fender | suge: I ahve personally done very well with it with a variety of hardware |
| 17:43.52 | suge | does it come out with the latest asterisk build? |
| 17:44.16 | suge | i think i was looking at one a year ago.. it had centos built in and was configged with asterisk |
| 17:45.01 | [TK]D-Fender | suge, its always best to download and compile * yourself. |
| 17:45.15 | drray | even a caveman could do it |
| 17:45.30 | suge | what is that supposed to mean? |
| 17:45.35 | suge | sheesh |
| 17:45.45 | suge | lol those commercials are great |
| 17:45.48 | drray | yeah |
| 17:45.50 | wunderkin | so easy a therapist could do it |
| 17:46.22 | drray | I skip commercials a lot with the tivo, but I have made a habit of watching them |
| 17:46.26 | suge | Also, is it possible to run 2 instances of Asterisk (easily)? I want to be able to run a production instance, as well as a dev one that i can start and stop |
| 17:47.39 | simplexio | suge: then you need use non standart port for -dev versio, which is that great. easier is just install asterisk to second dev box |
| 17:47.57 | drray | or a vm |
| 17:48.03 | drray | since vmware is free |
| 17:48.12 | suge | oh it is? good deal |
| 17:48.17 | suge | vmware for linux and windows? |
| 17:48.20 | Rhizome | or xen, since it's better ;) |
| 17:48.33 | Qwell[] | Rhizome: xen is not better, IMO... |
| 17:48.41 | Qwell[] | vmware is an amazing piece of software. |
| 17:48.55 | simplexio | well thats option too if you have fast computer, i usually use older hardware for dev |
| 17:49.53 | Hmmhesays | vmware workstation has like 25% overhead which isn't bad considering what it is doing |
| 17:50.02 | Rhizome | yea vmware is amazing! |
| 17:50.37 | Rhizome | Still, for running only various linux distros I like xen, and "no performance overhead" |
| 17:51.23 | *** join/#asterisk h0 (n=fakhir@unaffiliated/fakhir) |
| 17:51.35 | *** join/#asterisk [hC] (n=hardcore@S0106000fb51cc225.vf.shawcable.net) |
| 17:51.52 | simplexio | i havent tested Xen, dosent it need "own" isntall, like you dont get it easily to existing linux box ? |
| 17:52.26 | nick125_lappy | simplexio: You can install xen on a existing linux box.. |
| 17:52.35 | Rhizome | yea, you'll need to boot a xen specific kernel |
| 17:52.57 | nick125_lappy | basically, you install a xen-enabled kernel, change your bootloader to boot the xen hypervisor, and, install the xen userspace tools (such as xend, etc) |
| 17:53.03 | simplexio | nick125_lappy: i know i can install. i mean install it easily without much hassle |
| 17:53.08 | *** join/#asterisk evisu (i=hIRC@bzq-88-153-110-128.red.bezeqint.net) |
| 17:53.18 | nick125_lappy | simplexio: It depends on your definition of "easily" |
| 17:53.19 | simplexio | nick125_lappy: that sound easy |
| 17:53.25 | monsted | everything is a hassle on linux |
| 17:53.37 | Rhizome | monsted: no it's not. |
| 17:53.41 | simplexio | nick125_lappy: emerge xen && etc-updateconf is easy :) |
| 17:53.44 | *** join/#asterisk Ebola (n=Ebola@host81-152-204-157.range81-152.btcentralplus.com) |
| 17:53.47 | monsted | err, no |
| 17:53.48 | *** join/#asterisk PMantis (n=pmantis@66.251.89.34) |
| 17:53.49 | Rhizome | :P |
| 17:54.05 | nick125_lappy | simplexio: more like emerge xen-sources xen-tools |
| 17:54.11 | Nukemizer | hey hey |
| 17:54.12 | nick125_lappy | unless they made a metapackage |
| 17:54.18 | Nukemizer | was dreaming |
| 17:54.28 | simplexio | have to test it someday |
| 17:54.30 | PMantis | Hello... I've tried SET(CDR(userfield)-blahblahblah) and SETCDRUserField(blahblahblabh), but userfield is blank in cdr.. Asterisk 1.4.0 |
| 17:54.40 | simplexio | not much power in my linux box. |
| 17:54.42 | monsted | Nukemizer: oi, go back to sleep :) |
| 17:55.20 | simplexio | probably i look it when i update my desktop hw and move this x2 to linux server |
| 17:55.28 | *** join/#asterisk rene- (n=rene-@200.34.66.137) |
| 17:55.39 | rene- | hey |
| 17:56.04 | nick125_lappy | rene-: NO! JUST NO! |
| 17:56.05 | *** join/#asterisk jtexter3 (n=jtexter3@COX-66-210-197-34-static.coxinet.net) |
| 17:56.14 | [TK]D-Fender | rene-: ho |
| 17:56.18 | rene- | sorry to be here asking annoying stuff but i have a problem with chanspy, it basically is working for me as a 3 way call with echo |
| 17:56.43 | rene- | i have 1.4 -r50468 |
| 17:56.50 | *** join/#asterisk krondorl (n=me@acid.auricnet.ca) |
| 17:57.01 | rene- | calling it like this: exten => _*10X.,1,ChanSpy(SIP/${EXTEN:3}) |
| 17:57.05 | simplexio | but thats takes sometime, beacause i dont play much anymore and i have few 2000+ amd xp waiting memory. so no need to get faster playing machine or more servers ... |
| 17:57.45 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 17:58.14 | jtexter3 | Anyone here have experience with an Audiocodes Mediant? |
| 17:58.26 | simplexio | and when i have needed server at work. i have got one dedicated server for it |
| 18:00.27 | codefreeze | PMantis: Can you pastebin the snippet of your dialplan where you do this, and explain a little what the situation is? |
| 18:00.53 | krondorl | anyone know of a way to have one extension dial another and force the other to actually be picked up in speaker mode so the person calling can say something? |
| 18:01.23 | [TK]D-Fender | krondorl: Lookup "auto answer" on the WIKI |
| 18:01.44 | [TK]D-Fender | krondorl: Its phone-dependant |
| 18:01.59 | krondorl | k.. hmmm.. |
| 18:02.19 | *** join/#asterisk jaike (n=jaike@58.71.16.46) |
| 18:07.28 | *** part/#asterisk jaike (n=jaike@58.71.16.46) |
| 18:08.34 | *** join/#asterisk Redth (n=JonD@c-76-19-240-194.hsd1.ma.comcast.net) |
| 18:09.25 | *** join/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) |
| 18:10.36 | EmleyMoor | In Asterisk 1.2, must I have a numbered "1" priority in each extension or can I replace it with n(Start)? (or is that only allowed when the extension is only reached by Goto, not by answering or dialing?) |
| 18:12.32 | [TK]D-Fender | EmleyMoor: "no". |
| 18:12.55 | [TK]D-Fender | As it is, "n" is next to worthless, and not worth the code it took to implement. |
| 18:13.28 | *** join/#asterisk Aurs (n=Aurs@81.191.112.190) |
| 18:13.43 | *** join/#asterisk awannabe (n=brad@216.161.165.212) |
| 18:13.59 | EmleyMoor | So I do need a 1 - fair enough |
| 18:14.24 | awannabe | when you want to ring multiple phones, you can use exten => 550,1,Dial(SIP/peer/111&SIP/peer/112,20tr) corect? |
| 18:14.35 | EmleyMoor | awannabe: Yes |
| 18:15.16 | awannabe | that has been working fine, now all of a sudden it rings one phone perfect, and then second phone it rings for like a thousand of a second and shows a missed call, but thats it |
| 18:15.17 | [TK]D-Fender | awannabe: Correcting th missing comma for your 3rd parm, yes |
| 18:15.22 | Dr-Linux | exten => _7XXX,1,Dial(IAX2/serverB/${EXTEN:1},30,r) |
| 18:15.29 | Dr-Linux | anything work with this? |
| 18:15.36 | EmleyMoor | I use IAX2/phil&SIP/phil&Zap/1 for mine... and can put r and a number on the end to make the Zap phone ring distinctively |
| 18:15.50 | [TK]D-Fender | awannabe: And the fact you should never have to use the "r" option to begin with, and that "t" is an ugly hack only needed by those using X-Lite in production. |
| 18:16.05 | awannabe | lol, lovely!! so what should i be using then?/ |
| 18:16.09 | awannabe | ive alwasy used that :( |
| 18:16.21 | Dr-Linux | awannabe: heh , i'm stuck with same case |
| 18:16.43 | awannabe | so, no options at all |
| 18:16.43 | [TK]D-Fender | awannabe: using for what? |
| 18:17.14 | awannabe | instead of the tr |
| 18:17.18 | awannabe | just nothing at all |
| 18:17.20 | Dr-Linux | EmleyMoor: my both servers are connected with each other via IAX trunk, but i can't make a call and get this >> http://phpfi.com/193887 |
| 18:17.55 | Dr-Linux | tried alot to fix it, but maybe i don't know anything about iax |
| 18:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 18:18.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 18:19.50 | [TK]D-Fender | awannabe: That'd be a good idea. What kind of phones/lines are you running? |
| 18:20.01 | tzanger | Dr-Linux: give them both the same list of codecs |
| 18:20.06 | awannabe | inbound PRI, and the phones are snom's |
| 18:20.13 | *** part/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 18:20.16 | [TK]D-Fender | Dr-Linux: Set the codec to ULAW on both sides already, |
| 18:20.42 | Dr-Linux | tzafrir: both side codecs are the same |
| 18:20.49 | [TK]D-Fender | awannabe: Then you definately shouldn't be using "tr" at all. |
| 18:21.10 | awannabe | ok, ill remove that, and try it, heh |
| 18:21.10 | [TK]D-Fender | Dr-Linux: Please paste your configs. We don't like working blind here. |
| 18:21.23 | tzanger | Dr-Linux: it doesn't appear so |
| 18:21.43 | Dr-Linux | [TK]D-Fender: okey lemme paste |
| 18:22.39 | awannabe | Spawn extension (sip, 304, 2) exited non-zero on 'SIP/192.168.168.22-b775c9e0' |
| 18:23.12 | awannabe | intresting |
| 18:23.17 | cbullock81 | hey. if you're using ${CALLERIDNUM} for a SIP channel, how can you strip the extra info off the end of it... instead of SIP/101 i get like SIP/101-blahblah |
| 18:23.21 | suge | has anyone messed with speech recognition and *? |
| 18:23.49 | CunningPike | cbullock81: ${CUT} is your friend |
| 18:24.06 | Strom_M | cbullock81, also, that information should be in ${CHANNEL}, not ${CALLERID(num)} |
| 18:24.30 | CunningPike | cbullock81: That, too |
| 18:24.33 | cbullock81 | k |
| 18:24.53 | Qwell[] | CunningPike: ${CUT()} |
| 18:25.03 | Qwell[] | CunningPike: just to be explicit that it's a function |
| 18:25.23 | Strom_M | what about ${MOHEL()} ? |
| 18:25.26 | CunningPike | Qwell[]: I'm aware of that :) |
| 18:25.30 | Qwell[] | Strom_C: eh? |
| 18:25.55 | Corydon-w | Strom_M: is that a partial cut? |
| 18:25.57 | Strom_M | Qwell[], a mohel is a rabbi who does circumcisions |
| 18:26.04 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 18:26.11 | Qwell[] | umm, okay |
| 18:26.21 | Strom_M | Corydon-w, it only works on recently assigned variables |
| 18:26.30 | cbullock81 | so what would be the syntax for using it ${CUT(${CALLERIDNUM}) ? |
| 18:26.36 | Corydon-w | Make sure you keep chan_mohel away from codec_whiskey |
| 18:26.48 | Qwell[] | res_pork |
| 18:26.59 | Corydon-w | Why would you cut ${CALLERID(num)}? |
| 18:27.07 | Qwell[] | why not? |
| 18:27.08 | Strom_M | cbullock81, ${CALLERIDNUM} is deprecated |
| 18:27.13 | Corydon-w | There are no delimiters within ${CALLERID(num)} |
| 18:27.16 | blitzrage | ok... someone de-nub me... I'm looking for the changes that have happened between Asterisk releases (1.2 series) -- and I can't seem to find them on asterisk.org |
| 18:27.30 | Dr-Linux | [TK]D-Fender: there are alot of comments, lemme remove them from the config file |
| 18:27.30 | cbullock81 | k |
| 18:27.37 | Qwell[] | blitzrage: svn log ;) |
| 18:27.43 | blitzrage | gah |
| 18:27.47 | Strom_M | blitzrage, arent those in the changes.txt file? |
| 18:28.07 | jtexter3 | I'm having trouble with attended transfers, but not sure where to look. My setup is PSTN -> Audiocodes Mediant 1000 -> Asterisk -> Polycom phones. If I do a blind transfer on an inbound call, all works as expected. However, if I do an attended transfer, the inbound caller doens't hear any ringback during the transfer, voice only gets connected when the other person answers |
| 18:28.07 | blitzrage | Strom_M: CHANGES file in 1.2 hasn't been updated since 11/25/2005 |
| 18:28.08 | Corydon-w | Use of ${CUT()} absolutely REQUIRES that there be delimiters |
| 18:28.15 | Qwell[] | Corydon-w: 2 could be a delim |
| 18:28.32 | EmleyMoor | Qwell[]: In some bizarre set of circumstances? |
| 18:28.37 | Corydon-w | Qwell[]: I fail to see the use of that, though |
| 18:28.43 | Strom_M | cbullock81, you should be using ${CHANNEL} in this application, not ${CALLERID(num)} |
| 18:28.44 | Qwell[] | Corydon-w: never know |
| 18:28.44 | cbullock81 | here's what im trying to do... i just want to call VoiceMailMain(s) and have it plug in the SIP ID... what's the best way to do that |
| 18:29.04 | Strom_M | cbullock81, you should have a callerid= setting in sip.conf |
| 18:29.11 | jtexter3 | I've tried progressinband=yes, but that doesn't seem to help |
| 18:29.12 | [TK]D-Fender | Strom_M: Books and their authors : "Russian Circumcision" by Ivanna Kutchacockoff |
| 18:29.18 | Corydon-w | cbullock81: ${CUT(CHANNEL,-,1}) |
| 18:29.19 | Strom_M | hahahahhaa |
| 18:29.57 | Corydon-w | cbullock81: it's the channel name, not the callerid that you want |
| 18:30.41 | [TK]D-Fender | cbullock81: What you want to do is "VoicemailMail(${CALLERID(num)}@internal)" |
| 18:30.59 | Marty-OTT | Hey, has ayone used the CAC ADIT600? |
| 18:31.29 | [TK]D-Fender | cbullock81: What you want to do is "VoicemailMain(${CALLERID(num)}@internal)" |
| 18:31.33 | [TK]D-Fender | (typo) |
| 18:31.39 | cbullock81 | i love you guys! :) |
| 18:31.42 | cbullock81 | thanks so much |
| 18:31.46 | cbullock81 | that worked like a champ |
| 18:31.55 | Corydon-w | [TK]D-Fender: teach a man to fish... |
| 18:32.32 | Strom_M | what the hell does fishing have to do with dialing a phone? :) |
| 18:32.39 | [TK]D-Fender | Corydon-w: "Light a fire for a man, and he's warm for a night. Light a man on fire, and he's warm for the rest of his life" |
| 18:32.59 | cbullock81 | i'll stick with fishing... dont wanna burn :P |
| 18:33.13 | [hC] | [TK]D-Fender: hah. brilliant. |
| 18:34.19 | *** join/#asterisk Marty-OTT (n=marty@host-209-50-87-3.dyn.295.ca) |
| 18:39.22 | Dr-Linux | [TK]D-Fender: please go here > http://phpfi.com/193954 |
| 18:39.31 | Flusher | Hi again. Is there a way to make Asterisk playing (mp3) music rather than a ring tone at caller side when he's calling please ? I'm think about a sort of "automatic" MoH. |
| 18:39.51 | [TK]D-Fender | Dr-Linux: Congratulations, thats *1* side of this call. Now the other..... |
| 18:40.03 | `Sauron | Drop them in a conference playing MOH, then set up a call from the conference to the end user |
| 18:40.11 | `Sauron | when end user picks up, drop them in the conference |
| 18:40.14 | `Sauron | voila p |
| 18:40.15 | `Sauron | :p |
| 18:40.15 | [TK]D-Fender | Flusher: AAnd like you were told earlier "show application dial" |
| 18:40.25 | `Sauron | yeah, it sucks, but hey. it might work |
| 18:40.29 | Dr-Linux | [TK]D-Fender: other is just the same, excpet ip/user |
| 18:40.37 | [TK]D-Fender | `Sauron: That is a very wrong thing to do..... |
| 18:40.39 | Katty | mew. |
| 18:40.46 | [TK]D-Fender | Dr-Linux: You KNOW I don't want to hear that.... |
| 18:40.48 | Flusher | [TK]D-Fender: yes, but it provides no help for mp3 file |
| 18:40.50 | `Sauron | Fender: When was I known for doing things "right" |
| 18:40.54 | [TK]D-Fender | Katty: Mew. |
| 18:41.13 | Dr-Linux | [TK]D-Fender: okey lemme fetch the other one |
| 18:41.15 | [TK]D-Fender | Katty: Got speakers there yet? |
| 18:42.34 | [TK]D-Fender | Dr-Linux: Not that you didn't shoot yourself in the foot already... you have disallow=all ***AFTER*** your allows, hence you have NO codecs avialable. |
| 18:44.06 | *** join/#asterisk shift (n=shift@82-39-32-254.cable.ubr01.benw.blueyonder.co.uk) |
| 18:44.34 | *** join/#asterisk slima (i=slima@unaffiliated/slima) |
| 18:45.18 | Dr-Linux | [TK]D-Fender: okey sir lemme check it out |
| 18:47.29 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-154-64-105.red.bezeqint.net) |
| 18:47.38 | DrukenLPY | afternoon [TK]D-Fender |
| 18:48.28 | Dr-Linux | [TK]D-Fender: here is other end >> http://phpfi.com/193957 |
| 18:48.57 | Katty | [TK]D-Fender: i've had speakers. |
| 18:49.44 | [TK]D-Fender | Dr-Linux: See above... |
| 18:50.13 | Dr-Linux | ok |
| 18:54.01 | Dr-Linux | [TK]D-Fender: okey great, i'm in business now, but after one ring i see this >> http://phpfi.com/193959 |
| 18:55.41 | bkruse_home | sounds like hes trying to call but isnt registered as an iax peer/friend |
| 18:57.02 | Dr-Linux | Jan 15 10:56:09 NOTICE[25129]: chan_iax2.c:7183 socket_read: Host 202.125.141.10 failed to authenticate as lhr-pbx |
| 18:57.19 | bkruse_home | Dr-Linux: dude, wrong username/pass |
| 18:57.21 | bkruse_home | fo sho |
| 18:57.37 | file | bkruse_home: home? 'tsk 'tsk |
| 18:58.05 | bkruse_home | :[ |
| 18:58.14 | bkruse_home | file: im taking my mom to surgery, kthx |
| 18:58.20 | file | :( |
| 18:58.27 | bkruse_home | ya, it sucks man |
| 18:58.47 | [TK]D-Fender | Dr-Linux: Your account concept is BACKWARDS. compare your registers with whats on the OTHER side. |
| 18:59.18 | bkruse_home | [TK]D-Fender: exactly what im thinkin |
| 18:59.31 | Dr-Linux | [TK]D-Fender: hhm.. but it registered |
| 19:00.04 | [TK]D-Fender | Dr-Linux: Also make sure you are specifying the context in your Dial statements. |
| 19:00.27 | Dr-Linux | aww |
| 19:00.37 | EmleyMoor | Can I give my "1" priority a label? |
| 19:00.40 | Dr-Linux | [TK]D-Fender: i'm not specifying context hhm.. |
| 19:01.04 | [TK]D-Fender | EmleyMoor: Sure, but why bother... just Goto the # and be done with it. |
| 19:01.20 | Dr-Linux | [TK]D-Fender: here is dial string: exten => _74XXX,1,Set(CALLERID(NUM)=16505935400) |
| 19:01.20 | Dr-Linux | exten => _74XXX,2,Dial(IAX2/lhr-pbx/${EXTEN:1},30,r) |
| 19:01.37 | [TK]D-Fender | Dr-Linux: STILL not specifying the context... |
| 19:01.53 | [TK]D-Fender | Dr-Linux: And by that I mean the TARGET context on the OTHER side. |
| 19:02.04 | [TK]D-Fender | Dr-Linux: Not referring to your peer entry |
| 19:02.18 | bkruse_home | Dr-Linux: now i know why, its saying no authority because its probably defaulting to context default |
| 19:02.52 | Dr-Linux | [TK]D-Fender: aww i'm sorry, how can i specify context in the dial stamement, since i'm just using as WIKI says |
| 19:02.52 | bkruse_home | DIAL(iax2/ihr-pbx/${EXTEN:1}@incoming,30,r) |
| 19:03.02 | [TK]D-Fender | *SIGH* |
| 19:03.02 | Dr-Linux | hhm..ok |
| 19:03.10 | bkruse_home | or whatever context your iax statement said |
| 19:03.33 | Dr-Linux | [TK]D-Fender: no bcoz , i'm using context in the user account info |
| 19:03.52 | bkruse_home | uh........thats why you specify that context. |
| 19:03.56 | bkruse_home | whats the context in the user info? |
| 19:05.29 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:05.47 | *** join/#asterisk nick125_ (n=nick@atarack/staff/nick125) |
| 19:05.56 | adde | Anyone who could just sling me a little help at compiling Asterisk? |
| 19:06.07 | Nugget | ./configure && make && make install |
| 19:06.10 | Nugget | boom. |
| 19:06.21 | adde | configure: error: *** termcap support not found |
| 19:06.21 | Marty-OTT | hey, what's better... going from 711 between one device and a middle device and then G729 between the middle and device and the end device OR G729 all the way (for 24 calls) |
| 19:06.28 | *** join/#asterisk nortex (n=breeves@snapper.titanspecialties.com) |
| 19:06.34 | *** join/#asterisk tsurko (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 19:07.38 | adde | anyone? |
| 19:07.57 | Nugget | Doesn't appear to be an asterisk problem, adde. Fix your termcap stuff. |
| 19:08.04 | DrukenLPY | adde: install termcap |
| 19:08.15 | `Sean | adde |
| 19:08.23 | `Sean | man i dont get people like adde |
| 19:08.26 | `Sean | when they post this [2:04pm] <adde> configure: error: *** termcap support not found |
| 19:08.42 | `Sean | the error msg clearly states whats the problem but they still come into the channel asking for help |
| 19:08.43 | `Sean | :| |
| 19:09.11 | DrukenLPY | `Sean: least they are easy to answer questions :) |
| 19:09.18 | `Sean | haha yes. |
| 19:10.41 | DrukenLPY | uhg... i r bored |
| 19:11.10 | *** join/#asterisk bkw_ (n=brian@truphone.plus.com) |
| 19:12.02 | Dr-Linux | bkruse_home: the context in the user info is "users" |
| 19:12.05 | `Sean | i cant wait |
| 19:12.13 | `Sean | till prison break season 3 airs :D |
| 19:12.20 | Qwell[] | `Sean: 22nd? |
| 19:12.32 | `Sean | uhm i beleive so |
| 19:12.37 | `Sean | one sec let me get you a correct date |
| 19:13.35 | Dr-Linux | bkruse_home: now i deleted the contexts from both end (from the user info's) |
| 19:14.08 | *** join/#asterisk DocHolliday (i=RogerRab@gateway/gpg-tor/key-0x0E4F6D6C) |
| 19:14.11 | Dr-Linux | now i get this: |
| 19:14.11 | Dr-Linux | Jan 15 11:13:42 WARNING[25129]: chan_iax2.c:6967 socket_read: Call rejected by 202.125.141.11: No authority found |
| 19:14.12 | Dr-Linux | <PROTECTED> |
| 19:14.59 | DocHolliday | hey [TK]D-Fender |
| 19:15.12 | `Sean | Qwell[] its jan 29 |
| 19:15.22 | Qwell[] | suck |
| 19:15.56 | DocHolliday | Qwell, how goes? |
| 19:16.36 | Qwell[] | it goes |
| 19:17.50 | `Sean | Qwell[] ya i know it does suck |
| 19:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 19:18.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 19:18.03 | `Sean | alls i know is if someone pre's it before its official air date im downloading it :D |
| 19:18.04 | Qwell[] | `Sean: I missed the last couple episodes... I should catch up |
| 19:18.11 | Qwell[] | but damn that show is good :P |
| 19:18.12 | `Sean | yea they where awesome dude |
| 19:18.16 | `Sean | you should catch up |
| 19:18.19 | `Sean | ive got some of them |
| 19:18.22 | `Sean | if you want i can send |
| 19:18.29 | `Sean | but my upload is 100kB/s |
| 19:18.37 | Qwell[] | nah, I've got it covered |
| 19:18.43 | jtexter3 | Anybody around who can help troubleshoot a ringback issue with an Audiocodes gateway? |
| 19:21.55 | Dr-Linux | hhm.. |
| 19:22.08 | Dr-Linux | can i use same context in sip.conf and iax.conf? :S |
| 19:22.12 | Dr-Linux | maybe there would be an issue :S |
| 19:22.12 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 19:22.36 | [TK]D-Fender | DocHolliday: y0 |
| 19:23.02 | DocHolliday | good sleep? |
| 19:23.28 | syzygyBSD | all sleep is good, just how much good sleep... |
| 19:24.10 | adde | Well i understand it got problems with termcap, but everywere in yum it says its installed... I really wanna get asterisk installed and checked out.... |
| 19:24.34 | adde | Totally stuck now :( |
| 19:26.55 | Dr-Linux | that i get on dialed server >> Jan 15 11:26:31 NOTICE[25129]: chan_iax2.c:6772 socket_read: Rejected connect attempt from 202.125.141.11, who was trying to reach '5000@users' |
| 19:27.33 | Dr-Linux | not sure what's wrong |
| 19:28.31 | file | read what it says. |
| 19:28.51 | file | adde: you need the development package too |
| 19:30.02 | Dr-Linux | file: but both servers are registered with each other |
| 19:30.10 | Dr-Linux | i'm not sure hat i'm doing wrong |
| 19:30.15 | adde | thanks file |
| 19:30.29 | file | Dr-Linux: does extension 5000 exist in context users? |
| 19:30.48 | Dr-Linux | file: yes, sip user |
| 19:31.44 | file | I'll only accept a pastebin of the context in extensions.conf |
| 19:32.56 | Dr-Linux | http://phpfi.com/193972 |
| 19:33.19 | file | that's not what I asked for |
| 19:33.20 | Dr-Linux | file: ok hold on |
| 19:35.09 | Dr-Linux | file: that's here like this : exten => 5000,1,Macro(stdexten20,5000,SIP/5000) |
| 19:35.21 | file | yet again, that's not what I asked for |
| 19:35.21 | Dr-Linux | having cisco 7960 phone |
| 19:35.47 | file | oh - misread |
| 19:35.57 | file | your issue is with authentication still |
| 19:36.23 | Dr-Linux | file: which authenticaion? |
| 19:36.28 | Dr-Linux | but i'm registered with both side |
| 19:36.32 | adde | After installation of Asterisk how do i run it? |
| 19:36.38 | Qwell[] | adde: type asterisk |
| 19:36.39 | file | registration does not mean you can place calls between two points |
| 19:36.42 | Qwell[] | ~book |
| 19:36.50 | jbot | extra, extra, read all about it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 19:36.50 | Qwell[] | adde: go read that |
| 19:36.53 | file | Dr-Linux: you have user=user, while it should be username=user |
| 19:36.55 | Dr-Linux | file: - Registered IAX2 to '202.125.141.11', who sees us as 70.89.66.123:4569 with no messages waiting |
| 19:37.09 | file | registration simply means the server knows how to reach you |
| 19:37.13 | Dr-Linux | file: it's mean, WIKI is wrong |
| 19:37.16 | file | it doesn't mean it can authenticate |
| 19:37.21 | Dr-Linux | i see |
| 19:37.35 | Dr-Linux | lemme change user to username |
| 19:39.57 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 19:40.21 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 19:43.17 | Dr-Linux | file: username didn't make difference |
| 19:43.46 | Dr-Linux | file: here is dial string: |
| 19:43.47 | Dr-Linux | exten => _85000,1,Set(CALLERID(NUM)=111000911) |
| 19:43.47 | Dr-Linux | exten => _85000,2,Dial(IAX2/ca-pbx/${EXTEN:1}@users,30,r) |
| 19:44.00 | file | I'm going to make you a basic setup |
| 19:44.11 | Qwell[] | file: ?! |
| 19:44.58 | file | so, tell me this... why are you doing registrations? |
| 19:45.09 | wunderkin | itz kewl |
| 19:45.16 | Dr-Linux | file: actually it's iax which i don't know |
| 19:45.18 | [TK]D-Fender | file: He can't affor crack... |
| 19:45.35 | Dr-Linux | file: i wanna connect my both servers with iax trunk |
| 19:45.50 | file | that doesn't mean you have to use dynamic hosts |
| 19:46.25 | Dr-Linux | file: if i put IP address there ir doesn't work |
| 19:46.35 | adde | Asterisk is running... well if i try to run it it says its running but to connect to it use -r... How come i cant see it when running ps -a... And most cruicial how do i get to configure it... Webserver doesnt seem to be up and running... :S |
| 19:46.40 | Dr-Linux | it asks for dynamic |
| 19:47.00 | wunderkin | ... you turn off registrations.. |
| 19:47.04 | file | that's because you don't take out the registration line |
| 19:47.13 | file | and thus it keeps trying to register, which it doesn't allow since it's not dynamic |
| 19:47.16 | Nugget | adde: you configure it by editing the config files. |
| 19:47.45 | Dr-Linux | file: hhm.. do you suggest me to put static host there |
| 19:47.49 | *** join/#asterisk h0 (n=fakhir@unaffiliated/fakhir) |
| 19:47.59 | adde | I have used AstaroNow there i used the web interface.. |
| 19:48.15 | Nugget | Asterisk isn't AstaroNow. |
| 19:48.23 | adde | AsteriskNow... |
| 19:48.30 | Nugget | It isn't that either. |
| 19:48.33 | file | http://www.pastebin.ca/317650 |
| 19:48.36 | adde | Sorry got my firewall in front of me aswell ...lol |
| 19:49.21 | adde | Nugget what do you mean... |
| 19:49.36 | Nugget | I mean that you configure Asterisk by editing the files in /etc/asterisk. |
| 19:50.04 | Dr-Linux | file: okey lemme see |
| 19:50.06 | adde | Whats the difference between Asterisk and AsteriskNow besides it being a live cd...? |
| 19:50.37 | Qwell[] | AsteriskNOW is a full distro |
| 19:50.40 | Qwell[] | Asterisk is a PBX |
| 19:50.55 | Nugget | Among other things, AsteriskNOW includes a GUI. |
| 19:51.33 | Dr-Linux | file: very very thanks :) |
| 19:51.42 | adde | But does Asterisk come with the webinterface...`? |
| 19:51.46 | Nugget | No. |
| 19:52.48 | file | Dr-Linux: all it takes is using your brain to figure out what you want to do |
| 19:53.19 | Dr-Linux | file: yes, i understand this |
| 19:53.34 | Dr-Linux | file: but what i was doing, was on WIKI |
| 19:53.43 | file | doesn't mean it's right |
| 19:53.46 | Dr-Linux | http://www.voip-info.org/wiki/view/Asterisk+-+dual+servers |
| 19:53.57 | adde | There must be some addons for asterisk that enables this thoiugh? |
| 19:54.00 | Dr-Linux | file: i'd say example 2 is wrong then |
| 19:54.13 | file | Dr-Linux: then change it |
| 19:54.39 | Dr-Linux | yeah, i'm changing it |
| 19:55.54 | *** join/#asterisk awannabe (n=brad@216.161.165.212) |
| 19:56.18 | awannabe | hey guys, ive having problems with parked calls, can anyone help me out? |
| 19:57.19 | *** join/#asterisk flot (n=flot@ppp85-140-166-252.pppoe.mtu-net.ru) |
| 19:57.27 | *** part/#asterisk flot (n=flot@ppp85-140-166-252.pppoe.mtu-net.ru) |
| 19:58.01 | *** join/#asterisk dasenjo (n=dasenjo@63.245.86.43) |
| 19:58.01 | file | awannabe: only if you actually elaborate |
| 19:58.24 | awannabe | ha, well i can park the call properly, but when I dial 701 or whatever, i get a not found |
| 19:58.32 | *** join/#asterisk sb_mx (n=sb_mx@200.78.229.18) |
| 19:58.33 | awannabe | im guessing i need something else in my phone context, right? |
| 19:58.37 | Marty-OTT | Hey, say I ahve the following: GATEWAY -------- ASTERISK ------------ PEERS |
| 19:58.38 | bcnl | what's it say on the console when you park it |
| 19:58.48 | bcnl | does 'show parkedcalls' actually list the call? |
| 19:59.00 | awannabe | Parked SIP/306-0819cfd0 on 701. Will timeout back to extension [sip] , 1 in 45 seconds |
| 19:59.06 | Marty-OTT | The gateway and asterisk are besides each other |
| 19:59.28 | Marty-OTT | i'm wondering about having G729 between the GATEWAY and ASterisk as well as ASTERISK and PEERS |
| 19:59.28 | Marty-OTT | OR |
| 19:59.32 | bcnl | you need to put include => parkedcalls |
| 19:59.35 | awannabe | and yes it shhows it |
| 19:59.38 | bcnl | into whatever context your phones are in |
| 19:59.39 | awannabe | and i have the include as well |
| 19:59.41 | awannabe | oh wait |
| 19:59.45 | *** join/#asterisk karmatronic (n=karmatro@84.77.152.124) |
| 19:59.47 | Marty-OTT | Do G711 between the GATEWAY and ASTERISK and G729 between ASTERISK and PEERS |
| 20:00.05 | Marty-OTT | The 2nd option ... I have to do transcoding |
| 20:00.42 | *** part/#asterisk Redth (n=JonD@c-76-19-240-194.hsd1.ma.comcast.net) |
| 20:00.54 | Dr-Linux | opss |
| 20:00.59 | Dr-Linux | file: my back luck |
| 20:01.06 | Dr-Linux | s/back/bad |
| 20:01.07 | Dr-Linux | Jan 15 12:00:26 NOTICE[25129]: chan_iax2.c:6772 socket_read: Rejected connect attempt from 202.125.141.11, who was trying to reach '5000@users' |
| 20:01.13 | Nugget | adde: you can check out the asterisk-gui beta code from SVN and shoehorn it into a plain asterisk install. It's not going to be simple or as useful as you're hoping, but you can do it. |
| 20:01.36 | Marty-OTT | anyone? Is it better to have half the path G711 (where there's tons of bandwidth) and the other G729 or G729 all the way to elimincate transcoding |
| 20:01.48 | awannabe | bcnl, then i just dial the number it says its parked to, right? |
| 20:01.52 | Nugget | the GUI just isn't up to the task of configuring asterisk quite yet. It's got a ways to go (which is largely why it's not included with the main asterisk code yet) |
| 20:02.11 | file | Dr-Linux: show the EXACT iax.conf, extensions.conf, and an iax2 debug from each side |
| 20:02.12 | bcnl | awannabe: well on my setup we prefix internal dials with a # |
| 20:02.16 | bcnl | but basically, yup |
| 20:02.38 | Dr-Linux | file: okey |
| 20:03.00 | awannabe | bcnl, ok, i see |
| 20:03.21 | Marty-OTT | anyone? Is it best to have G729 all the way OR, where bandwidth permits.. use G711 part of the way? |
| 20:04.27 | perd | i suppose it depends on your system and bandwidth limitations |
| 20:04.39 | Dr-Linux | file: wow great |
| 20:04.40 | Marty-OTT | Bandwidth between the 2 first devices is 100 megs |
| 20:04.48 | Marty-OTT | non-issue |
| 20:04.58 | Dr-Linux | file: Thanks again |
| 20:05.06 | Dr-Linux | file: once side worked for me :) |
| 20:05.08 | Marty-OTT | GATEWAY --- ASTERISK ---------------------------------------------------- WAN-------------------- SIP PEER |
| 20:05.15 | file | then figure out the diference. |
| 20:05.19 | perd | so dont bother transcoding then, just make sure qos gives those sip packets priority |
| 20:05.19 | Dr-Linux | file: so other side would be not a problem for me |
| 20:05.26 | Dr-Linux | file: thank you very much :) |
| 20:05.27 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 20:05.34 | perd | hell use alaw |
| 20:05.42 | perd | :) |
| 20:06.08 | Marty-OTT | perd: :) ... so just use g729 all the way so I don't have to transcode? |
| 20:06.18 | perd | i would |
| 20:06.24 | perd | but im not an expert |
| 20:06.38 | Marty-OTT | Well, it's a good start |
| 20:08.39 | *** join/#asterisk maverickbna (i=sentinel@wikipedia/Shadowhntr) |
| 20:08.53 | danp | hurray, bug 8416 got fixed...maybe i can try 1.4 again |
| 20:10.08 | *** join/#asterisk markit (n=konversa@host119-245-static.72-81-b.business.telecomitalia.it) |
| 20:11.43 | sp0n9e | seems like every outbound call i make has static, but analog phones work fine when hooked straight in |
| 20:11.51 | *** join/#asterisk h0 (n=fakhir@unaffiliated/fakhir) |
| 20:14.30 | awannabe | bcnl, that was it, i didnt include it in my other context, duh! |
| 20:14.45 | bcnl | :> |
| 20:14.50 | awannabe | hrmm, now can I make a light on the phone lightup with the hint() function i wonder |
| 20:14.56 | awannabe | so i can visually show when its parked |
| 20:16.05 | bcnl | if you figure it out |
| 20:16.09 | bcnl | privmsg me :> |
| 20:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 20:18.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 20:18.10 | [TK]D-Fender | exten => #701,hint,Local/701@parkedcalls |
| 20:18.23 | awannabe | yeah i am seeing that on google |
| 20:18.34 | awannabe | now i jsut got to get this work on the snoms :) |
| 20:22.38 | ManxPower | awannabe: maybe you should get regular BLF working first? |
| 20:22.56 | awannabe | ManxPower, yeah those work |
| 20:23.05 | awannabe | what the hell, parking only works one way? |
| 20:23.28 | Sweeper | arg. so my tormenta2 card is causing the entire server to lock the fuck up |
| 20:23.50 | Sweeper | as soon as the drivers for it are loaded |
| 20:23.57 | Sweeper | and the interface is looped to itself |
| 20:23.59 | Sweeper | BAM |
| 20:25.29 | ManxPower | Sweeper: Well contact the company you bought that cheap ass clone card from. |
| 20:25.43 | awannabe | dang it, ih open i canget the BLF to work on this snom |
| 20:25.49 | awannabe | other knowing when a call is parked is not cool lol |
| 20:26.02 | Sweeper | PCI Computer Telefony \o |
| 20:26.04 | ManxPower | Sweeper: My bets would be placed on an IRQ issue. cat /proc/interrupts |
| 20:26.33 | ManxPower | Sweeper: Why DID you buy a Zaptel card that is at least THREE generations behind the current ones. |
| 20:26.41 | Sweeper | I didn't |
| 20:26.46 | Sweeper | lol existing hardware |
| 20:27.06 | *** join/#asterisk [hC] (n=hardcore@S0106000fb51cc225.vf.shawcable.net) |
| 20:27.07 | ManxPower | ah, ok |
| 20:27.08 | [TK]D-Fender | Sweeper: Unexist it :) |
| 20:27.10 | Sweeper | tor2 has irq 11 |
| 20:27.16 | [hC] | anyone know much about cisco service contract pricing? |
| 20:27.26 | ManxPower | Sweeper: is anything else on IRQ 11? |
| 20:27.30 | Sweeper | no |
| 20:27.32 | awannabe | [TK]D-Fender, have you used that hint comand in the past? |
| 20:27.39 | [TK]D-Fender | [hC]: I'd puta contract out on you far cheaper :) |
| 20:27.41 | ManxPower | [hC]: what do you need to know? |
| 20:28.02 | [TK]D-Fender | awannabe: Not specifically. |
| 20:28.03 | ManxPower | Cood choice, [TK]D-Fender |
| 20:28.14 | [TK]D-Fender | ManxPower: :D |
| 20:28.30 | Sweeper | wow is that fucker spinning out the interrups |
| 20:28.31 | [hC] | ManxPower: im showing someone why they shouldnt go with call manager, and im curious what elements of the system need to be renewed ever... just smartnet contracts? or some sort of licenses as well? and.. do they lock you in to any terms for smartnet when you buy the whole call manager? |
| 20:28.43 | ManxPower | Sweeper: 1,000 interrupts per second |
| 20:28.50 | Sweeper | yea |
| 20:28.58 | [hC] | ManxPower: and lastly, does the number of seats affect your support contract pricing? if so how? |
| 20:29.13 | [hC] | [TK]D-Fender: ;) |
| 20:29.20 | ManxPower | Sweeper: Digium cards, especially the ones based on the old GPL Tormeta design are incredibly sensitive to interrupt issues. |
| 20:29.22 | *** part/#asterisk krondorl (n=me@acid.auricnet.ca) |
| 20:29.37 | awannabe | bastage lights |
| 20:29.38 | Sweeper | mkay |
| 20:29.47 | Sweeper | well, I'm not showing any errors |
| 20:29.53 | ManxPower | [hC]: Get a quote for call manager and find out. There's nothing wrong with torturing sales people by getting their hopes up. |
| 20:30.02 | Sweeper | is it possible my looping one t1 to another is inherently bad? |
| 20:30.10 | Nugget | I have to power cycle my asterisk box every time my cat sneezes because the TDM400P stops answering calls. |
| 20:30.22 | ManxPower | And while you are at it maybe a member of upper mnagement will have a heart attack at the price on the quote and that is almost always a good thing. |
| 20:30.33 | Nugget | heh |
| 20:30.38 | ManxPower | Nugget: Now you know why I can never recommend TDM400Ps |
| 20:30.50 | Nugget | yup yup |
| 20:31.01 | markit | I would need a different logic for a "dialer" application.. at the moment, through the Manager, if I want to call someone I have asterisk make my extension ring and when I pick-up, I have just to wait the other to answer. I would like to have the same, but be able to pickup the extension without having it to ring, is it possible? |
| 20:31.07 | [TK]D-Fender | ManxPower: Except the odd time when they become persistant and keep friggen nagging you every week, pushing a project that YOU clearly aren't..... I've got this one nut-bar who's eyes show white all around... |
| 20:31.19 | ManxPower | Several years ago we got a quote for Call Manager. I think the price was right around a quarter of a million US dollars for a 40 seat system |
| 20:31.36 | [hC] | ManxPower: yeah, i have a quote from a competitor in front of me, ii just dont know exactly which things need renewing or not.. i need to send my competitive quote out within the hour, or i would call cisco myself |
| 20:31.54 | ManxPower | [hC]: it sucks to be you. |
| 20:32.01 | [hC] | ManxPower: these guys got a call manager express setup, for about 40 grand right in the door with 33 seats.. which seems stupidly low. |
| 20:32.08 | [hC] | ManxPower: so i figure the support contract must be hidden in there |
| 20:32.21 | ManxPower | [hC]: or NO support contract. |
| 20:32.43 | Sweeper | yay technician at $400 an hour if you have issues :) |
| 20:32.45 | ManxPower | Do you know what I do when someone comes in and says "We can do it cheaper", I say "Go for it!" |
| 20:32.49 | [hC] | ManxPower: all i see is smartnet on the call manager and switch, which i presume entitles you to upgrades, but no actual "support" |
| 20:33.02 | ManxPower | Actually, I say "Give me an official BINDING QUOTE" |
| 20:33.02 | Qwell[] | smartnet is everything |
| 20:33.07 | ManxPower | We seldom hear from them again |
| 20:33.36 | [hC] | ok.. smartnet gets you help too.. |
| 20:33.45 | Qwell[] | [hC]: supposed to ;) |
| 20:33.50 | ManxPower | [hC]: that depends on the smart net level. |
| 20:34.05 | Sweeper | so anyways. is looping the t1 ports to eachother what's causing my issue, or is there something else going on? |
| 20:34.07 | [hC] | "smart net extended service" |
| 20:35.05 | Sweeper | ha, holy crap this thing has a spartan processor XD |
| 20:36.50 | nick125_lappy | Wewt, hopefully my PAP2T-NA will ship today! |
| 20:36.52 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 20:36.53 | [hC] | it would appear that smartnet does not include onsite help. |
| 20:36.57 | [hC] | only telephone and email |
| 20:37.05 | Corydon-w | As opposed to an Athenian processor? |
| 20:37.21 | [TK]D-Fender | Sweeper: mmmm Apples..... |
| 20:38.03 | Sweeper | Corydon-w: I...guess? |
| 20:38.14 | Sweeper | I mean, spartan is a brand of proc :v |
| 20:38.20 | Corydon-w | Are there also Roman processors? |
| 20:38.28 | *** join/#asterisk oQPa (n=roque@15.Red-83-40-197.dynamicIP.rima-tde.net) |
| 20:39.08 | awannabe | whoaa, parking only works in one direction, ever seen that before? |
| 20:40.18 | Sweeper | fpga, I suppose would be moar specific |
| 20:42.26 | Sweeper | o, ir dumb... |
| 20:42.43 | [TK]D-Fender | Sweeper: O s u r |
| 20:47.35 | Sweeper | mmm |
| 20:47.41 | Sweeper | port one works fine |
| 20:48.17 | Sweeper | alrighty then |
| 20:48.33 | Sweeper | so apparently, cross-connecting ports on a tor2 card is a no-no |
| 20:48.55 | awannabe | thanks guys for help, i let you know on the parking stuff with snoms!! thanks! |
| 20:48.56 | *** join/#asterisk tsurko (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 20:50.38 | *** join/#asterisk rpm (n=russell@66.119.170.34) |
| 20:53.41 | wunderkin | Sweeper, it probably just doesn't like you, join the club, digium and polycom don't like me :P |
| 20:53.49 | Sweeper | nah |
| 20:54.01 | *** join/#asterisk af_ (n=af@ip-173-157.sn1.eutelia.it) |
| 20:54.02 | Sweeper | the ports are happy and run clean with my test set |
| 20:54.13 | *** join/#asterisk russellb (i=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 20:54.13 | *** mode/#asterisk [+o russellb] by ChanServ |
| 20:54.16 | Sweeper | timing is set to internal/exteral |
| 20:54.20 | [TK]D-Fender | wunderkin: No... Tor2 didn't like anybody :) |
| 20:56.26 | wunderkin | [TK]D-Fender, polycom apparantly has NFI about my problems :P hopefully..sometime.. i will get new phones in... |
| 20:56.56 | [TK]D-Fender | wunderkin: 2 words : Plausible Deniability :) |
| 20:59.09 | wunderkin | i will have to go back to my barbietone while i wait for a replacement... fun |
| 20:59.59 | *** join/#asterisk Strom_C (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 21:01.21 | *** join/#asterisk jm|laptop (n=jamie@dilbert.jamiem.com) |
| 21:02.05 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 21:07.31 | variable_office | is there a way i can check to see if dtmf is even being transmitted, whether by packet sniffer or internally on asterisk? |
| 21:08.46 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 21:08.47 | Dr-Linux | file: wow much better voice quality with IAX trunk, i'd like to add my all other servers |
| 21:11.53 | *** join/#asterisk X-Rob (n=rob-x@203.37.172.162) |
| 21:12.22 | *** part/#asterisk oQPa (n=roque@15.Red-83-40-197.dynamicIP.rima-tde.net) |
| 21:12.43 | *** join/#asterisk oQPa (n=roque@15.Red-83-40-197.dynamicIP.rima-tde.net) |
| 21:13.13 | *** part/#asterisk oQPa (n=roque@15.Red-83-40-197.dynamicIP.rima-tde.net) |
| 21:13.58 | *** join/#asterisk oQPa (n=roque@15.Red-83-40-197.dynamicIP.rima-tde.net) |
| 21:14.14 | [TK]D-Fender | Dr-Linux: Should have difference unless you were choked by bandwidth in your other config. Here's where you should enable trunking as well. |
| 21:14.16 | *** join/#asterisk oQPa (n=roque@15.Red-83-40-197.dynamicIP.rima-tde.net) |
| 21:15.37 | Dr-Linux | [TK]D-Fender: that is correct, actually my pakistan site bandwidth is poor, and my most of users are here |
| 21:16.09 | Dr-Linux | [TK]D-Fender: now maybe i registered all users locally |
| 21:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 21:18.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 21:24.09 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 21:25.26 | *** join/#asterisk _IPmonger_ (n=ipmonger@c-68-84-208-206.hsd1.pa.comcast.net) |
| 21:26.41 | *** join/#asterisk bkw_ (n=brian@88-110-74-241.dynamic.dsl.as9105.com) |
| 21:27.25 | [TK]D-Fender | heading home, bbiab |
| 21:28.38 | Dr-Linux | [TK]D-Fender: is there any opensource program which provide redundancy between 2 identify asterisk servers? |
| 21:29.27 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 21:29.29 | Hmmhesays | 2 identify asterisk servers? |
| 21:29.39 | Dr-Linux | Hmmhesays: i'm sorry |
| 21:29.45 | Dr-Linux | s/identify/identical |
| 21:29.47 | Hmmhesays | identical? |
| 21:29.55 | Dr-Linux | Hmmhesays: yes |
| 21:30.28 | Hmmhesays | openser can be setup to do such a thing |
| 21:30.58 | *** part/#asterisk dasenjo (n=dasenjo@63.245.86.43) |
| 21:31.03 | Hmmhesays | i use it with round robin dns to handle calls to a set of asterisk servers |
| 21:32.48 | Dr-Linux | hhm.. |
| 21:33.10 | Dr-Linux | Hmmhesays: how your syncronise the data between servers? |
| 21:33.25 | Hmmhesays | which data do you speak of? |
| 21:33.43 | jtexter3 | Is the list server down today? |
| 21:33.47 | Dr-Linux | hhm.. let's suppose voicemails or events etc |
| 21:34.32 | Hmmhesays | i use external notifies for voicemail |
| 21:34.47 | *** join/#asterisk EinsteinTaylor (n=dtaylor@208.50.194.173) |
| 21:34.53 | Dr-Linux | Hmmhesays: is there any manual for round robin DNS redundancy? |
| 21:35.11 | Dr-Linux | Hmmhesays: i see |
| 21:35.23 | Hmmhesays | Dr-Linux: not really, i use BIND and SRV dns records |
| 21:35.27 | Hmmhesays | my ata's support them |
| 21:35.42 | Dr-Linux | cool |
| 21:35.46 | EinsteinTaylor | using Realtime, if my SIP phone is unplugged, is there a way to make it keep parsing the rest of the Dialplan? specifically the part that has it forward to my cell phone? |
| 21:36.08 | Dr-Linux | Hmmhesays: someone said, www.linux-ha.org can help |
| 21:36.09 | EinsteinTaylor | the forward on no answer works fine if the phone is plugged in, but otherwise not |
| 21:36.09 | Hmmhesays | when an ata calls asterisk bind sends back an address of one of the servers, if that server doesn't respond, it grabs the netx one |
| 21:36.51 | *** join/#asterisk squish102 (n=squish10@cpe-024-074-100-250.carolina.res.rr.com) |
| 21:37.38 | *** join/#asterisk _IPmonger_ (n=ipmonger@c-68-84-208-206.hsd1.pa.comcast.net) |
| 21:39.38 | ManxPower | USA standard loop start POTS lines are 600 Ohm? |
| 21:40.34 | *** join/#asterisk Zefk (n=Zefk@81.181.249.106) |
| 21:40.56 | Strom_C | ManxPower: that sounds about right |
| 21:42.23 | Zefk | Does anyone used updatecdr=yes in agents.conf? For me it is not working at all :( |
| 21:42.24 | nick125_lappy | Ugh, I hate this stupid crappy DSL modem |
| 21:42.38 | variable_office | dtmf is not working for me in any way, inband, rfc2833, info; none work; what are some common problems that could make this happen? |
| 21:43.11 | Qwell[] | variable_office: on SIP I hope? |
| 21:43.16 | variable_office | yes, sip |
| 21:43.19 | Qwell[] | using dtmfmode=blah? |
| 21:43.23 | variable_office | yes |
| 21:43.31 | Qwell[] | How are you testing dtmf? |
| 21:43.32 | variable_office | well, its realtime, but same effect |
| 21:43.47 | variable_office | dialing phone menus that work normally and they dont |
| 21:46.01 | EinsteinTaylor | using Realtime, if my SIP phone is unplugged, is there a way to make it keep parsing the rest of the Dialplan? specifically the part that has it forward to my cell phone? |
| 21:46.07 | ManxPower | Strom_C: I''m FINALLY starting to play with the Sage Instruments 930A Comms Test Set |
| 21:46.11 | *** join/#asterisk jeebusroxors (n=jeebusro@cpe-75-80-231-237.dc.res.rr.com) |
| 21:46.18 | Strom_C | cool |
| 21:46.21 | variable_office | Qwell[] is there a better way/ have any suggestions? |
| 21:47.00 | *** part/#asterisk jtexter3 (n=jtexter3@COX-66-210-197-34-static.coxinet.net) |
| 21:55.10 | EmleyMoor | I'd like to be able to set up one of the Zap channels in here so that the user can dial a number and get straight into his voicemail without having to login or enter a password |
| 21:55.25 | EmleyMoor | Is that possible? Can someone point me in the right direction? |
| 21:58.15 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
| 22:00.35 | *** join/#asterisk BarnacleBob (n=karl@38.99.18.98) |
| 22:04.46 | ManxPower | EmleyMoor: exten => 444,1,VoicemailMain(s${CALLERIDNUM}} assuming the callerid is the same as their mailbox |
| 22:05.05 | EmleyMoor | Ah, yes, got it |
| 22:05.24 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 22:05.27 | *** join/#asterisk die_z (n=dieeasy@host163-115-static.104-80-b.business.telecomitalia.it) |
| 22:06.02 | EmleyMoor | I've done it by detecting when the call is made on Zap/2 |
| 22:06.55 | De_Mon | ah hell. an agent tried to xfer a caller and now asterisk CLI stops responding if I do a show agents, queue or queues |
| 22:07.36 | Sweeper | restart \o |
| 22:08.14 | De_Mon | ya cant do that with 5 active calls :/ |
| 22:08.49 | rene- | should a TE412p show in lspci as a TE410p or a TE412 ? |
| 22:08.53 | De_Mon | I cant unload chan agent either damnit! time to upgrade from 2.4.11 |
| 22:09.57 | rene- | because i have two servers and was told that both had TE412 cards but lspci actually reads TE410p on both |
| 22:10.32 | Strom_C | it's the same card |
| 22:10.40 | Strom_C | one just has an echo canceller module |
| 22:12.11 | rene- | Strom_C: is there a way i can determine if the card has or has not the echo canceling module and if there is anything i should do to enable it? |
| 22:12.12 | ManxPower | De_Mon: What verison of Asterisk? |
| 22:12.21 | rene- | i have echo cancel, echo training yes in zapata |
| 22:12.30 | Strom_C | well, then that should enable it |
| 22:12.38 | rene- | i still have echo |
| 22:12.49 | rene- | not on local calls |
| 22:12.53 | rene- | but on international calls |
| 22:12.59 | rene- | over tdm |
| 22:12.59 | Strom_C | are you sure the echo cancellation module is seated firmly on the card? |
| 22:13.37 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 22:14.49 | rene- | Strom_C: it was installed by digium so it should |
| 22:15.01 | rene- | i will check when we are finished for the day |
| 22:15.12 | Strom_C | it may have come loose during shipping |
| 22:15.29 | rene- | Strom_C: is the echo cancel module is enabled it should deal with any echo coming in from the carrier? |
| 22:15.35 | Strom_C | when you load the modules, does it make mention of "VPM450M" or "VPM128M"? |
| 22:16.04 | rene- | should any of those come up on a lsmod? |
| 22:16.12 | Strom_C | check dmesg |
| 22:16.57 | rene- | Strom_C nothing in dmesg |
| 22:17.37 | rene- | but i see this in dmesg |
| 22:17.39 | rene- | zaptel Disabled echo canceller because of tone (rx) on channel 56 |
| 22:17.48 | Strom_C | how long has the system been up? |
| 22:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 22:18.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 22:18.04 | rene- | 2 days |
| 22:18.11 | Strom_C | do |
| 22:18.15 | Strom_C | dmesg | grep VPM |
| 22:18.46 | rene- | Strom_C: nada |
| 22:18.54 | rene- | wait |
| 22:19.03 | rene- | VPM400: Not Present |
| 22:19.03 | rene- | VPM450: Not Present |
| 22:19.16 | rene- | weeird |
| 22:19.20 | rene- | so that is the problem |
| 22:19.23 | Strom_C | check that the module hasn't come loose in transit |
| 22:19.30 | rene- | ok will do |
| 22:19.58 | De_Mon | ManxPower 2.4.11 |
| 22:20.26 | De_Mon | 1.2.11 rather |
| 22:21.12 | ManxPower | De_Mon: best to upgrade to 1.2.14 before anything else. |
| 22:21.20 | Strom_C | rene-: and you /are/ running the latest version of zaptel 1.2 branch, right? |
| 22:21.26 | ManxPower | I just got a seg fault in 1.2.14 (getGroupcount) |
| 22:21.45 | *** join/#asterisk nahirean (n=nahirean@unaffiliated/nahirean) |
| 22:22.11 | rene- | Strom_C: i am in 1.4 branch around (50500) |
| 22:22.22 | Strom_C | for zaptel also? |
| 22:22.27 | rene- | yes |
| 22:22.31 | rene- | zaptel branch |
| 22:22.42 | rene- | dont know the zaptel revision |
| 22:23.01 | Strom_C | ok |
| 22:23.44 | bkruse_home | rene-: svn info /usr/src/zaptel err wherever ur zap dir is |
| 22:27.59 | *** part/#asterisk markit (n=konversa@host119-245-static.72-81-b.business.telecomitalia.it) |
| 22:28.03 | *** join/#asterisk StephenL (n=steve@216.62.85.65) |
| 22:28.45 | rene- | Revision: 1808 |
| 22:28.53 | rene- | thanks |
| 22:29.17 | rene- | actually i might have compiled zaptel trunk and use it with asterisk branch |
| 22:29.18 | *** join/#asterisk smithj (n=smithj@gentoo/developer/smithj) |
| 22:29.23 | *** part/#asterisk smithj (n=smithj@gentoo/developer/smithj) |
| 22:29.57 | rene- | i dont have zaptel branch just zaptel 1.4 but zaptel 1.4 wasnt compiling with asterisk trunk 1.4 |
| 22:30.13 | rene- | might be also part of the problem |
| 22:32.06 | rene- | weird after i reloaded zaptel in my other box echo cancelling showed up in dmesg |
| 22:32.25 | StephenL | What are the best free soft phones available right now? It's been over a year since I last used asterisk and at the time we had good success with x-lite. Is there something better now? |
| 22:32.37 | rene- | the problem is that i cant reload zaptel in the box that is running/.. thanks Strom, bkruse |
| 22:33.11 | Strom_C | rene-: learn this lesson - never put something into production unless you KNOW it works :) |
| 22:33.34 | *** join/#asterisk CrashHD (n=crashhd@c-67-182-167-222.hsd1.ca.comcast.net) |
| 22:34.25 | rene- | that sounds right |
| 22:34.55 | b11d|bbl | I dunno.. my dad told me this bit of invaluable advice: If you dont know what you're doing, just get started. |
| 22:35.02 | b11d|bbl | so far.. it's worked out :) |
| 23:01.53 | *** join/#asterisk duckz (n=duckz@141.85.3.18) |
| 23:01.57 | *** join/#asterisk DocHolliday (i=RogerRab@gateway/gpg-tor/key-0x0E4F6D6C) |
| 23:01.58 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 23:01.59 | *** join/#asterisk CrashHD (n=crashhd@c-67-182-167-222.hsd1.ca.comcast.net) |
| 23:02.03 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 23:02.04 | [TK]D-Fender | ^%@$#^%#$ white *&#^ |
| 23:02.05 | ManxPower | StephenL: All Softphones still Suck. |
| 23:02.05 | rene- | jeje |
| 23:02.05 | rene- | ManxPower: would you use any softphone for anything that was high volume? |
| 23:02.06 | DocHolliday | [TK]D-Fender, i can see you are real productive :P |
| 23:02.14 | ManxPower | rene-: I would not use any softphone at all. They suck so bad they give VoIP and Asterisk a bad name. |
| 23:02.37 | b11d|bbl | aye |
| 23:02.48 | StephenL | ManxPower: could you recommend any that don't suck quite as bad as others? |
| 23:02.48 | [TK]D-Fender | DocHolliday : I am! I live close to work and spend as much time unburying my car as driving it home! |
| 23:02.49 | DocHolliday | snow? |
| 23:02.49 | [TK]D-Fender | DocHolliday : and am in the middle of preparing a batch of sushi for supper |
| 23:02.49 | rene- | is polycom still the dear of the asterisk community? |
| 23:02.50 | [TK]D-Fender | StephenL : Forget soft-phones as "daily-use" phones... serious PITA |
| 23:02.50 | nick125_lappy | softphones are a pain in the rear |
| 23:02.50 | [TK]D-Fender | rene- : Until something better comes along, yeah :) |
| 23:02.56 | ManxPower | StephenL: X-Lite seems to suck somewhat less than the others, but I don't use softphones....they suck |
| 23:03.02 | *** join/#asterisk queuetue (n=scott@MTRLPQ02-1177745854.sdsl.bell.ca) |
| 23:03.02 | [TK]D-Fender | rene- : In order of preference : Polycom (Any), Aastra 480i, Linksys (Any), Cisco. |
| 23:03.03 | queuetue | Hi. Does anyone have an opinion of the Uniden UIP200? Amazon has them for 45 bucks... |
| 23:03.03 | rene- | queuetue: i like them |
| 23:03.03 | queuetue | Oh, used. |
| 23:03.04 | [TK]D-Fender | ManxPower : I'll ammend that by it works rather easily, though lacking internal transfer, conference and a lot more features, still royally sucks |
| 23:03.04 | queuetue | Still... |
| 23:03.04 | DocHolliday | [TK]D-Fender, with you i thouught it was polycom, aastra, linksys, buggy, beta, less than working, cisco |
| 23:03.04 | rene- | queuetue: yes, they have good quality and sound good |
| 23:03.06 | [TK]D-Fender | queuetue : I own 2. Do yourself a favour, DON'T. |
| 23:03.08 | rene- | D-Fender: why? |
| 23:03.12 | rene- | they dont have a web interfase tho |
| 23:03.16 | queuetue | It's a phone I'm sending to a remote office for a technophobe to use, so... It has to work pretty well and seem like "just a phone". :) |
| 23:03.19 | [TK]D-Fender | queuetue : handset quality is barely passable, speakerphone is HORRID. LCD is crap. |
| 23:03.22 | [TK]D-Fender | queuetue : they're better off with an ATA |
| 23:03.24 | queuetue | Does anyone have abetter sub-150 USD suggestion? |
| 23:03.26 | [TK]D-Fender | queuetue : The will seem "just like a phone" :) |
| 23:03.27 | rene- | queuetue: there are sub 150 polycoms i believe |
| 23:03.29 | queuetue | [TK]D-Fender: Like a spa2002? |
| 23:03.33 | [TK]D-Fender | queuetue : ATA does is. Polycom IP430 does as well. IP501 ($170) is very solid. |
| 23:03.36 | ManxPower | There is a nat=route option in Asterisk SPECIFICALLY to make NAT work with the Uniden phone |
| 23:03.39 | [TK]D-Fender | queuetue : Sure. thats +/- $70 USd for 2 ports |
| 23:03.43 | b11d|bbl | you can get brand new polycoms for less than $150 from cxtec.com |
| 23:03.46 | queuetue | [TK]D-Fender: We use spa-2000's her ein the office, and I love 'em. |
| 23:03.50 | b11d|bbl | 501's that is |
| 23:03.54 | [TK]D-Fender | queuetue : IP 430 = $150 |
| 23:03.59 | queuetue | I thought the spa's were hard to find unlocked now, though. Or am I 2 years behind? :) |
| 23:04.01 | [TK]D-Fender | b11d|bbl : Can you link that specifically? |
| 23:04.05 | [TK]D-Fender | queuetue : that was only the PAP2. |
| 23:04.06 | queuetue | Why does amazon's image of the Sipura 2002 have a linksys logo on it? |
| 23:04.06 | b11d|bbl | no.. you gotta talk to their sales rep. |
| 23:04.07 | *** join/#asterisk stctheman123070 (n=scurtis@65.77.64.98) |
| 23:04.07 | [TK]D-Fender | queuetue : And only idiots who read the big print and though it didn't matter. |
| 23:04.07 | b11d|bbl | i never get any list prices from their website. |
| 23:04.07 | stctheman123070 | newbie needs help with install |
| 23:04.07 | [TK]D-Fender | b11d|bbl : Ah |
| 23:04.08 | queuetue | [TK]D-Fender: Ok, thanks. |
| 23:04.10 | b11d|bbl | but they have been ROCK SOLID for me over the last few years.. |
| 23:04.15 | nahirean | linksys did produce q 2002 for a short time |
| 23:04.17 | b11d|bbl | everything delivered within 3 days.. |
| 23:04.22 | nahirean | a* |
| 23:04.30 | [TK]D-Fender | b11d|bbl : Their Web prices are unimpressive.... guess you've got a good rep |
| 23:04.33 | nahirean | right before they discontinued them |
| 23:04.46 | stctheman123070 | anyone willing to help? |
| 23:04.49 | b11d|bbl | yeah.. i guess i must have.. |
| 23:11.01 | Strom_C | ~ask |
| 23:11.04 | jbot | well, ask is Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily. See also http://catb.org/~esr/faqs/smart-questions.html |
| 23:11.04 | stctheman123070 | thanks |
| 23:11.05 | [TK]D-Fender | stctheman123070 : Ask a specific question, and you might jsut get a specific answer. |
| 23:11.05 | [TK]D-Fender | Strom_C : Jbot has forsaken you :| |
| 23:11.05 | stctheman123070 | I have installed asterisk (by the book) but when I run "make" I keep getting |
| 23:11.05 | stctheman123070 | root@scurtis-linux:/usr/src/asterisk-1.4.0# make install |
| 23:11.05 | stctheman123070 | menuselect/menuselect --check-deps menuselect.makeopts |
| 23:11.05 | stctheman123070 | <PROTECTED> |
| 23:11.05 | stctheman123070 | make[1]: *** [stereorize] Error 127 |
| 23:11.05 | stctheman123070 | make: *** [utils] Error 2 |
| 23:11.05 | ManxPower | Avenue Computer Supply, Baton Rouge, Louisiana 504-523-7874, Ask for Gaston. I don't know how GOOD their Polycom Prices are, but they are the reseller that I use. |
| 23:11.06 | stctheman123070 | What am I missing? |
| 23:11.06 | JT | pastebin |
| 23:11.06 | JT | ~pb |
| 23:11.08 | jbot | i guess pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 23:11.08 | *** join/#asterisk zapp-branigan (n=zapp-bra@81-202-140-56.user.ono.com) |
| 23:11.08 | stctheman123070 | what is pastebin? |
| 23:11.08 | JT | pastebin.ca |
| 23:11.09 | JT | if you are pasting over 3 lines, it's a thing to use to not flood over 300 users |
| 23:11.09 | b11d|bbl | holy shit TK.. you werent kidding.. those prices on thier website are ridiculous |
| 23:11.09 | [TK]D-Fender | Strom_C : ALL forsaken :( |
| 23:11.10 | *** join/#asterisk Asterman (n=newkinet@shell4.sea5.speakeasy.net) |
| 23:11.10 | stctheman123070 | ohh ok |
| 23:11.10 | [TK]D-Fender | b11d|bbl : Yup. www.telephonydepot.com |
| 23:11.10 | stctheman123070 | new here |
| 23:11.10 | b11d|bbl | $210 for a 501.. I get them for $144 |
| 23:11.10 | [TK]D-Fender | b11d|bbl : WOW |
| 23:11.10 | b11d|bbl | im telling you.. might be worth the call to cultivate a relationship with them :) |
| 23:11.10 | [TK]D-Fender | b11d|bbl : SURE they aren't reconditioned? They do list they offer such things... |
| 23:11.10 | nahirean | stctheman123070: go to pastebin.ca and past it in there, then link it |
| 23:11.10 | ManxPower | I don't know what we pay for Polycoms. I say "Order another Polycom" and in a few days they arrive. |
| 23:11.10 | b11d|bbl | yeah I notice that.. im 99% certain they arent refurb'd |
| 23:11.10 | nahirean | paste* |
| 23:11.10 | b11d|bbl | I do buy refurbished catalyst switches from them.. |
| 23:11.11 | b11d|bbl | and they have been great.. no way to tell they are used.. |
| 23:11.11 | b11d|bbl | lifetime warranty.. etc.. |
| 23:11.11 | b11d|bbl | nope.. i'll stand with cxtec.. |
| 23:11.11 | zapp-branigan | hi i'm installed the speex codec library and xmms-speex-0.9.1-8.fc6 |
| 23:11.11 | zapp-branigan | speex-devel-1.2-0.1.beta1.fc6 |
| 23:11.11 | zapp-branigan | speex-1.2-0.1.beta1.fc6 |
| 23:11.12 | b11d|bbl | always been impressed with them.. although, not with those web prices.. |
| 23:11.12 | zapp-branigan | but not compile in the asterik |
| 23:11.12 | zapp-branigan | 1.4 |
| 23:11.12 | stctheman123070 | http://pastebin.ca/317901 |
| 23:11.12 | Asterman | I'm having difficulties setting up zaptel with just the zaptel and ztdummy modules. I have no telephony card in my * box (external calls are all done through a SIP origination/termination provider) and so I'm trying to use the ztdummy module so that I can use meetme. The problem is, I get an error thrown up by ztcfg saying that it cannot init tones for channel x..... anyone any ideas? |
| 23:11.13 | zapp-branigan | ./configure fin it but the .so do not apear why ? |
| 23:11.13 | ManxPower | b11d|bbl: Try Avenue Computer |
| 23:11.14 | nahirean | stctheman123070: you ran a ./configure first, right? |
| 23:11.14 | b11d|bbl | for what? |
| 23:11.14 | stctheman123070 | no I did not |
| 23:11.14 | ManxPower | b11d|bbl: Polycoms |
| 23:11.14 | b11d|bbl | why? |
| 23:11.14 | Asterman | does anyone in here have a working zaptel.conf/zapata.conf that works without any card and just uses ztdummy so that meetme works? |
| 23:11.14 | ManxPower | I do NOT get a comission, but it is run by a friend. They may have decent prices |
| 23:11.14 | b11d|bbl | I'm sticking with cxtec as my supplier.. |
| 23:11.15 | angler | Asterman, you don't need any configuration if just using ztdummy |
| 23:11.15 | [TK]D-Fender | Asterman : lookup ztdummy on the WIKI. You'll see you have to uncomment it in the makefile... |
| 23:11.15 | ManxPower | b11d|bbl: Oh! I thought you were looking for a supplier |
| 23:11.15 | Asterman | TK - I've built the ztdummy module, that's not the prob (I think), it's my config that seems hosed |
| 23:11.15 | *** join/#asterisk Hmmhesays (n=Neg@24-117-135-28.cpe.cableone.net) |
| 23:11.15 | b11d|bbl | no.. thanks though :) |
| 23:11.50 | stctheman123070 | <nahirean> I just ran ./configure |
| 23:11.51 | Hmmhesays | anyone have any experience setting up a linux box to use two isp's? |
| 23:11.51 | stctheman123070 | then I tried make install again |
| 23:11.51 | stctheman123070 | same error |
| 23:11.51 | b11d|bbl | do you want to load balance, or maintain one peer for reduundancy? |
| 23:11.51 | stctheman123070 | should I run make clean, make then make install? |
| 23:11.51 | Asterman | angler - what do you mean, you don't need any config for ztdummy? meetme seems to need one |
| 23:11.51 | nahirean | yes, but i doubt it will fix it.. :) |
| 23:11.52 | stctheman123070 | same error after make |
| 23:11.52 | stctheman123070 | thanks for your help |
| 23:11.52 | nahirean | wait.. |
| 23:11.52 | nahirean | haha |
| 23:11.52 | ManxPower | Asterman: remove /etc/zaptel.conf |
| 23:11.53 | nahirean | /bin/bash: g++: command not found |
| 23:11.53 | nahirean | that's your bloody problem |
| 23:11.53 | stctheman123070 | ohh |
| 23:11.53 | nahirean | hah |
| 23:11.53 | stctheman123070 | no clue |
| 23:11.53 | Hmmhesays | b11d|bbl: peer redundancy |
| 23:11.53 | Asterman | Manx: - ok, I've just done that, then what? |
| 23:11.53 | nahirean | stctheman123070: what distro are you using |
| 23:11.54 | stctheman123070 | debian or ubuntu |
| 23:11.54 | ManxPower | Asterman: try it again |
| 23:11.54 | nahirean | drop to prompt and type apt-get install g++ |
| 23:11.54 | nahirean | as root |
| 23:11.54 | nahirean | you're missing the compiler |
| 23:11.54 | Hmmhesays | i know you can't set up two default gateways on a pc without it freaking out |
| 23:11.54 | stctheman123070 | under /usr/src ? |
| 23:11.54 | nahirean | doesnt matter where you do the apt-get |
| 23:11.54 | Asterman | Manx : now that I've removed /etc/zaptel.conf, when I try to do a modprobe ztdummy it complains that it is unable to open /etc/zaptel.conf :) |
| 23:11.54 | [TK]D-Fender | ..and sushi assembly begins.... |
| 23:11.55 | Asterman | Manx : same too when I run ztcfg |
| 23:11.55 | nahirean | [TK]D-Fender: Don't tempt me to order some.. please don't.. my wallet cannot take anymore $40 platters |
| 23:11.55 | Hmmhesays | i'd google for what i'm looking for but i'm not sure exactly what it is called |
| 23:11.55 | ManxPower | Asterman: ztcfg is for setting up cards, not ztdummy. Does lsmod show ztdummy loaded? |
| 23:11.55 | [TK]D-Fender | nahirean : I'm MAKING mine. infinitely cheaper |
| 23:11.55 | nahirean | [TK]D-Fender: interesting concept.. do something yourself.. hmm.. |
| 23:11.55 | *** join/#asterisk Bobthehunter (n=Bobthehu@145-27.mc.cite.net) |
| 23:11.55 | Bobthehunter | I was wondering if we can load agents.conf from extconfig ? |
| 23:11.56 | [TK]D-Fender | nahirean : Its another form of artistic expression for me. I can't draw for shit, but I've got martial arts, music, and now food :) |
| 23:11.56 | [TK]D-Fender | ... and I think pool counts for me as well :) |
| 23:11.56 | Asterman | Manx : yup, but it gave me that error when I did the modprobe. (btw, I could load the mod prior to this), I'll give meetme a whirl though, brb |
| 23:11.56 | nahirean | [TK]D-Fender: Nice. I have binge drinking and "getting shot down" as my two art forms. ;) |
| 23:11.56 | [TK]D-Fender | nahirean : Nice synergy :) |
| 23:11.56 | nahirean | [TK]D-Fender: Indeed. Perhaps they are related after all.. another revelation.. hmm.. |
| 23:11.56 | Asterman | Manx : still no joy, I'm getting the dreaded "No channel type registered for 'zap'" warning message whenever I call meetme, and it reckons my pins are invalid |
| 23:11.57 | nahirean | stctheman123070: did the make install work this time? |
| 23:11.57 | Asterman | Manx : /dev/zap/ctl exists :) |
| 23:11.57 | [TK]D-Fender | nahirean : I have "don't try", "rarely get", and "unexpected instant relationships" on my repetoire.... |
| 23:11.57 | ManxPower | Asterman: that sounds like you installed Asterisk before you installed Zaptel. |
| 23:11.57 | ManxPower | If you don't have zaptel installed, asterisk won't build chan_zap. |
| 23:11.57 | nahirean | [TK]D-Fender: You may be on to something. It seems that once you give up, you subsconciously convey somethinf desirable to that which is woman.. |
| 23:11.57 | nahirean | something* |
| 23:11.57 | stctheman123070 | nahirean - Thanks for your help! It looks like it is working now |
| 23:11.57 | ManxPower | nahirean: I think the word you are looking for is "desperation" |
| 23:11.58 | [TK]D-Fender | nahirean : could be... I've stopped looking for a reason personally. I'm a little too "tired" to look or care. |
| 23:11.58 | nahirean | ManxPower: Not me! Getting shot down is my art, but that doesn't mean I am always shot down. ;) |
| 23:11.58 | Asterman | Manx : this was an install off of open suse 10.1. The weird thing was, was that for just one day I did have this up and running, but then befor I could lock down everything someone rebooted the machine and I lost it |
| 23:11.58 | stctheman123070 | nahirean - yes it did!! Thanks :) |
| 23:12.25 | nahirean | stctheman123070: have fun ;) and say "No!" to 942s. |
| 23:12.26 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 23:12.48 | stctheman123070 | ??? |
| 23:12.48 | nahirean | nevermind ;) |
| 23:14.59 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 23:16.52 | *** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com) |
| 23:18.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 23:18.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 23:18.28 | *** part/#asterisk die_z (n=dieeasy@host163-115-static.104-80-b.business.telecomitalia.it) |
| 23:20.05 | De_Mon | hrm I see restart for asterisk, but not shutdown.. I thought there was a shutdown |
| 23:20.56 | CunningPike | Welcome back, jbot! |
| 23:21.18 | Qwell[] | De_Mon: stop |
| 23:21.18 | CunningPike | De_Mon: From the asterisk CLI? |
| 23:22.20 | *** join/#asterisk h0 (n=fakhir@unaffiliated/fakhir) |
| 23:23.06 | queuetue | Does anyone have a clue where to buy more of this phone? We have 9 or 10 in the company, they are great phones, and I'd like to just keep using them as we expand, but it seems they have become hard to find all of a sudden... http://www.amazon.com/gp/product/images/B000JJI6TY (It's a GE 27977, I believe) |
| 23:23.40 | De_Mon | ya stop, thats what I needed |
| 23:23.44 | *** join/#asterisk Mad|Cow (n=thirt@74.92.109.206) |
| 23:23.59 | CunningPike | De_Mon: Yup - 'stop when convenient' usually |
| 23:24.56 | queuetue | Or maybe something similar? The "only a headset" and changeable faceplate are both big winners, for us. |
| 23:27.15 | *** join/#asterisk robin_sz (n=robin@rapid2.gotadsl.co.uk) |
| 23:27.22 | *** join/#asterisk osas (n=nnnnnnnn@CABLE-72-53-75-244.cia.com) |
| 23:27.26 | *** join/#asterisk Dr-Linux|home (n=Dreamer@DSL-202-59-73-131.nexlinx.net.pk) |
| 23:27.27 | robin_sz | meep? |
| 23:27.49 | robin_sz | so, mISDN gurus? |
| 23:27.54 | robin_sz | DTMF detection?? |
| 23:27.58 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
| 23:29.06 | bkruse_home | :X |
| 23:29.18 | bkruse_home | :P |
| 23:29.23 | *** join/#asterisk malverian (n=malveria@gentoo/developer/malverian) |
| 23:30.19 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 23:31.08 | [TK]D-Fender | bkruse_home : Message to subway "performers" : Just because nobody understands you... doesn't mean you're an "artist" :) |
| 23:31.40 | bkruse_home | [TK]D-Fender: i like it, i like it |
| 23:32.58 | *** join/#asterisk h0 (n=fakhir@unaffiliated/fakhir) |
| 23:33.26 | [TK]D-Fender | Speaking of which, here's a sample of my latest crap :) http://aocomputing.net/sweeps1.mp3 |
| 23:33.51 | xheliox | MY EARS MY EARS! |
| 23:33.57 | nahirean | grr, fix your pipe |
| 23:34.10 | xheliox | Okay, now it's time to listen. |
| 23:34.44 | xheliox | neato. |
| 23:34.48 | [TK]D-Fender | I started to learn this technique about 2 weeks ago, and am just goofing off with it right now. |
| 23:35.03 | xheliox | This should be Asterisk's default on hold music. ;) |
| 23:35.09 | bkruse_home | [TK]D-Fender: nice, ima do that |
| 23:35.13 | nahirean | Wait! I need to find a wife to beat, stand by :) How long you been playing? |
| 23:35.48 | [TK]D-Fender | nahirean : About 15 years, now, and I'm just starting lessons :) |
| 23:36.16 | nahirean | [TK]D-Fender: I am trying to teach myself, I am terrible at it.. but I've a background in electronic music production.. actually serves to hinder me, given the phyiscal aspect of playing the geetar. |
| 23:36.37 | *** part/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) |
| 23:36.40 | nahirean | i know what sound I want to come out of the thing, but my fingers betray me |
| 23:36.52 | Dr-Linux|home | [TK]D-Fender: what technique? |
| 23:37.05 | *** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com) |
| 23:37.10 | [TK]D-Fender | Dr-Linux : Sweeping |
| 23:37.18 | nahirean | sounds like an arpeggio to me |
| 23:37.30 | nahirean | sort of |
| 23:37.35 | bkruse_home | </3 arpeggio |
| 23:37.44 | Dr-Linux|home | sweeping is a technique? :S |
| 23:37.52 | [TK]D-Fender | nahirean : in essence, its picked through in a sontinuous "sweep", but not allowed to ring through like a chord. |
| 23:38.08 | [TK]D-Fender | Dr-Linux : Wax on... wax off! |
| 23:38.20 | nahirean | i see |
| 23:38.43 | [TK]D-Fender | nahirean : http://www.youtube.com/watch?v=3aZZBvqdtCs |
| 23:38.57 | [TK]D-Fender | nahirean : the only video I got to watching on instruction. |
| 23:39.09 | bkruse_home | [TK]D-Fender: that a video on you? |
| 23:39.18 | bkruse_home | [TK]D-Fender: i need to show you a cool vid |
| 23:39.24 | Asterman | Question to anyone : is there a way to see if chan_zap was compiled into * ? I want to see if the problem I've been having was down to that |
| 23:39.34 | [TK]D-Fender | Dr-Linux : its msuci terminology, its completely understandable. the 2nd bit was a jok from tThe Karate Kid (80's movie) |
| 23:39.48 | bkruse_home | Asterman: ls /usr/lib/asterisk/modules | grep zap |
| 23:40.26 | Asterman | bkruse : thanks :) |
| 23:40.27 | Dr-Linux|home | i see |
| 23:40.31 | bkruse_home | Asterman: no ;[ |
| 23:40.35 | bkruse_home | np ;]* |
| 23:40.51 | [TK]D-Fender | bkruse_home : no, the video is a guy teaching the basics of sweeping. |
| 23:40.58 | bkruse_home | ya, ive seen it ;] |
| 23:41.01 | [TK]D-Fender | music* |
| 23:41.02 | nahirean | pretty interesting |
| 23:41.03 | bkruse_home | [TK]D-Fender: check out http://youtube.com/watch?v=HCnPpn1OFTU |
| 23:41.19 | nahirean | I'm sure you've all seen that kid playing Cannon right? |
| 23:41.21 | Dr-Linux|home | who is developer for AGI application? |
| 23:41.24 | [TK]D-Fender | bkruse_home : Andy Mckee yeah I know piece :) |
| 23:41.42 | bkruse_home | <3 andy mckee |
| 23:41.43 | [TK]D-Fender | bkruse_home : I found the tab and tuning for it... whacked... |
| 23:42.04 | bkruse_home | [TK]D-Fender: i know!!! i was like man, what happend to standard and drop D tuning |
| 23:42.34 | [TK]D-Fender | bkruse_home : Yeah this guy is like "drop EVERYTHING and just wonder what the hell I'm doing! |
| 23:45.25 | *** join/#asterisk karmatronic (n=karmatro@84.77.152.124) |
| 23:46.53 | [TK]D-Fender | bkruse_home : that is a ver "steath" playing video. Between tuning, and the angle of his hands because of the style you can't really tell whats going on... |
| 23:47.24 | *** part/#asterisk karmatronic (n=karmatro@84.77.152.124) |
| 23:47.43 | bkruse_home | ya, i saw |
| 23:47.46 | bkruse_home | but very cool |
| 23:50.15 | riddlebox | [TK]D-Fender, wow that youtube link was awesome, wish I could play anywhere near that good |
| 23:50.28 | b11d|bbl | practice :) |
| 23:53.15 | [TK]D-Fender | riddlebox : Save it and practice. I started from 0 unable to pick it properly, and then in 2 weeks, the recording I linked above |
| 23:53.25 | [TK]D-Fender | riddlebox : I'll come, just practice. |
| 23:53.29 | [TK]D-Fender | it'll |
| 23:53.48 | riddlebox | yeah, I have found youtube to be a great source for figuring out songs now |
| 23:56.01 | [TK]D-Fender | riddlebox : I'm more interested in the basic techniques and theory. I prefer to express "me", than to try to do it through someone elses music. |
| 23:56.33 | riddlebox | yeah, I have been working on theory but have lost the interest, due to other needs |
| 23:56.49 | [TK]D-Fender | riddlebox : Been there... |
