00:13.38 | *** join/#asterisk dysinger_ (n=tim@71-20-35-99.war.clearwire-wmx.net) |
00:15.22 | ayeso | raden_work: yes it is in /etc/logger.conf... it should allready be logging to /var/log/asterisk/messages |
00:17.04 | *** join/#asterisk netpro25_ (n=mmanning@fl-71-0-164-16.sta.embarqhsd.net) |
00:18.51 | netpro25_ | hello, can someone tell me more about this GSM bug and if it is in Asterisk 1.4.21.2~dfsg-1ubuntu3 |
00:19.03 | KyleK | what gsm bug? :) |
00:19.16 | russellb | ~gsmbug |
00:19.17 | infobot | [~gsmbug] there is a bug compiling Asterisk with GCC 4.2 where optimization errors cause GSM transcoding to distort heavily. Compile with GCC 4.1 or lower, or follow this patch : http://bugs.digium.com/view.php?id=11243 Also please read : http://forums.digium.com/viewtopic.php?p=116731&sid=3c65e49ec840380ed20f1c8426382b39 |
00:19.26 | netpro25_ | thanks russellb |
00:19.30 | netpro25_ | ;-) |
00:19.33 | russellb | infobot knows all |
00:19.34 | infobot | and don't you forget it |
00:19.39 | russellb | O.O |
00:19.40 | netpro25_ | hah |
00:20.22 | netpro25_ | russellb: do you know if that bug still exists in the most recent release |
00:20.46 | russellb | not AFAIK |
00:21.06 | russellb | I haven't heard complaints about it in a long time |
00:21.30 | netpro25_ | yea Ubuntu 9.04 uses an older 1.4.21 version |
00:21.37 | russellb | lammmmme |
00:21.44 | netpro25_ | hah |
00:22.58 | netpro25_ | okay so I have another ubuntu server with 1.4.17 and I did not have these gsm problems |
00:23.01 | KyleK | compile and package a newer version and send it to them? |
00:23.31 | netpro25_ | well I was looking to see if there was a repo that I can just add usually there are bleeding edge repos |
00:23.38 | netpro25_ | for things like this |
00:23.40 | netpro25_ | did not see one |
00:24.06 | KyleK | well asterisk is a bunch of "compile your own" people |
00:24.17 | carrar | YEAH!! |
00:24.23 | netpro25_ | heh, yea... It just makes things less standard |
00:24.52 | netpro25_ | debs are so much easier to deal with. If i could package my own then I would, just dont know how |
00:24.54 | KyleK | well if you really want a standard, step up and start packaging asterisk for debian |
00:25.15 | KyleK | (im assuming ubuntu just autocopies that package from debian) |
00:25.23 | netpro25_ | possibly |
00:26.39 | netpro25_ | I assume most asterisk people use other distros as ubuntu is just too easy to use |
00:26.49 | netpro25_ | based on your earlier statement |
00:27.08 | KyleK | i use ubuntu |
00:27.20 | netpro25_ | cool |
00:28.07 | grandpapadot | dear god, it takes like 5 minutes to install asterisk from source |
00:28.30 | grandpapadot | if you want to upgrade via deb you gotta jump through about 30 hoops and waste half a day |
00:28.31 | netpro25_ | grandpapadot: okay I am gonna take the plunge |
00:28.47 | netpro25_ | is 1.6 stable? |
00:28.58 | netpro25_ | nm |
00:28.58 | netpro25_ | rc |
00:29.01 | KyleK | lol |
00:29.01 | grahamsaa | netpro: yes |
00:29.02 | grandpapadot | Possibly, I run 1.4.26.1, very stable *in most cases* |
00:29.11 | KyleK | netpro25_: theres lots of versions of asterisk right now |
00:29.19 | netpro25_ | yes there are |
00:29.26 | KyleK | 1.6.2 is the svn trunk, im using 1.6.1 svn |
00:35.47 | carrar | Oh my |
00:35.55 | file | actually trunk is trunk, and 1.6.2 is 1.6.2 - they are different |
00:36.26 | carrar | more junk in your trunk |
00:38.15 | thehar | opens file's trunk |
00:38.34 | netpro25_ | lol |
00:38.34 | russellb | cartwheels |
00:38.45 | thehar | tosses cheese its at russellb |
00:38.55 | russellb | nom nom nom nom |
00:39.33 | thehar | indeed, sir. |
00:39.41 | netpro25_ | Hey kinda off topic but what do you guys think about the LPI Certs for linux |
00:39.50 | netpro25_ | anyone heard of it or done it? |
00:39.52 | thehar | i think certs are crap. |
00:39.59 | netpro25_ | So do I |
00:40.06 | carrar | They are not Scottish! |
00:40.11 | thehar | my workplace also thinks the same |
00:40.11 | netpro25_ | but i have a BS in CS and I cant find crap for work |
00:40.23 | thehar | qwest is hiring locally, lol |
00:40.24 | netpro25_ | (just graduated) |
00:40.29 | thehar | ah |
00:40.37 | thehar | eBay is hiring a local telcom engineer as well |
00:40.46 | thehar | but that is avaya or cisco |
00:40.48 | netpro25_ | yea local as in cali? |
00:40.52 | thehar | salt lake city |
00:41.03 | netpro25_ | yea I am married |
00:41.16 | thehar | oh here i can fix that |
00:41.18 | carrar | and mormon? |
00:41.20 | thehar | cuts off the ball and chain from netpro25_ |
00:41.22 | netpro25_ | kinda hard to just pick up and go |
00:41.25 | carrar | heh |
00:41.26 | thehar | is not mormon |
00:41.45 | *** join/#asterisk Deeewayne (n=dwayne@c-71-228-179-90.hsd1.al.comcast.net) |
00:41.46 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
00:41.55 | netpro25_ | Now I would rather do a LPI cert then a Microsoft cert anyday |
00:42.11 | carrar | Internet Certified! |
00:42.16 | thehar | you are probably having a hard time going against people with work experience with your brand new degree. |
00:42.27 | thehar | experience > degree w/out exp. |
00:42.42 | netpro25_ | yea exactly. I do have 5 years experience though |
00:42.42 | netpro25_ | as a network admin |
00:42.43 | thehar | previous to your degree? |
00:42.45 | thehar | or concurrently |
00:42.46 | netpro25_ | yea |
00:42.49 | thehar | well there you go |
00:42.50 | netpro25_ | well concurrently |
00:42.52 | thehar | ah. |
00:43.06 | thehar | talk to your university. they can probably help =) |
00:43.14 | netpro25_ | lol yea a masters |
00:43.23 | carrar | tell them you want your money back |
00:43.38 | netpro25_ | hah i knew that was coming |
00:43.44 | thehar | most universities will help find placements or jobs.. internships, something. |
00:44.25 | netpro25_ | yea I have been applying for shit jobs there |
00:44.31 | netpro25_ | $10 hr |
00:44.33 | netpro25_ | and no go |
00:44.48 | *** join/#asterisk scalex000 (n=chatzill@181.120.88.200.f.sta.codetel.net.do) |
00:44.52 | thehar | welcome to the economy =) |
00:45.09 | netpro25_ | yes, you are kinda blind to it when you are in school |
00:45.13 | netpro25_ | then you get out and bam |
00:45.14 | thehar | i help hire techs in our company and we got thousands of resumes last job posting |
00:45.23 | netpro25_ | damn |
00:45.50 | netpro25_ | yea I have been doing contract work for the time being. Trying to build up my business. Maybe it will turn out for the better |
00:46.45 | carrar | get yourself on linked-in |
00:46.49 | netpro25_ | so yea when I compile from source on like ubuntu, I should first uninstall any repo packages right? |
00:46.54 | netpro25_ | carrar: I am on there |
00:47.00 | carrar | network! |
00:47.06 | netpro25_ | heh. I have. |
00:47.15 | netpro25_ | Actually I think I am in the asterisk group |
00:47.35 | carrar | always stuff on CL too |
00:47.44 | netpro25_ | yea thats hit or miss |
00:48.02 | netpro25_ | only thing I have not tried is temp agencies |
00:48.11 | netpro25_ | saving those for last resort |
00:48.25 | carrar | recruiters are nice to know when you are looking |
00:48.38 | carrar | annouying as hell when you're workin |
00:48.46 | netpro25_ | hah yea |
00:48.47 | thehar | nah.. |
00:48.52 | thehar | it's nice to know you're wanted when working |
00:50.10 | carrar | s/when working// |
00:50.35 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
01:05.33 | *** join/#asterisk scalex000 (n=chatzill@181.120.88.200.f.sta.codetel.net.do) |
01:12.57 | netpro25_ | hey before building asterisk from scratch should I uninstall the existing debian package? |
01:13.26 | jaytee | sure, why not |
01:16.23 | netpro25_ | lol |
01:16.49 | *** join/#asterisk TJNII (n=TJNII@207.189.199.58) |
01:16.50 | *** join/#asterisk ming_zym (n=ming_zym@124.127.101.0) |
01:19.56 | netpro25_ | sweet asterisk ascii art |
01:21.23 | drmessano | netpro25_: yeah, love it.. if only the effin devs spent their time fixing bugs and adding features instead of working on Asterisk ASCII art, imagine what we could have |
01:21.48 | netpro25_ | hah... Yea that had to have taken a good amount of time |
01:21.59 | netpro25_ | its original though |
01:22.03 | netpro25_ | have not seen that before |
01:22.23 | drmessano | "hey, we should fix that bug drmessano reported" "Nah, fuck him, lets have some Dr Pepper" "RIGHT ON!" |
01:22.23 | netpro25_ | maybe I have not been compiling from source enough to notice |
01:22.25 | netpro25_ | heh |
01:22.42 | netpro25_ | lol |
01:23.15 | netpro25_ | drmessano: what phones do you use |
01:25.41 | netpro25_ | any ideas of the top of your head for this error: Unable to open pid file '/var/run/asterisk.pid': Permission denied |
01:25.50 | drmessano | Well, Linksys and Polycom |
01:25.51 | netpro25_ | when trying to start asterisk |
01:25.58 | drmessano | Yeah |
01:25.59 | netpro25_ | yea I have linksys its nice |
01:26.05 | drmessano | Which Linksys? |
01:26.21 | *** join/#asterisk cviniciusm (n=cviniciu@189.27.12.215.dynamic.adsl.gvt.net.br) |
01:26.28 | drmessano | You need to set perms .. chown that dir |
01:26.29 | netpro25_ | linksys SPA941 |
01:26.41 | drmessano | I have a few SPA941s.. they're nice |
01:26.42 | cviniciusm | Hello. |
01:26.50 | netpro25_ | yea |
01:26.51 | drmessano | Hi circumcision |
01:26.55 | drmessano | Wassup? |
01:27.00 | netpro25_ | lol |
01:27.08 | netpro25_ | circumcision |
01:27.14 | cviniciusm | HAHAHA. |
01:27.25 | drmessano | Sorry, <TAB> key is broken |
01:27.28 | drmessano | had to paste from google |
01:27.35 | *** join/#asterisk freakazoid0223 (n=knoppix@pool-71-246-17-206.phlapa.fios.verizon.net) |
01:27.37 | drmessano | Did you mean: Circumcision |
01:27.38 | drmessano | :( |
01:27.39 | drmessano | Sorry |
01:28.07 | netpro25_ | drmessano: chown the run directory? |
01:28.54 | cviniciusm | What´s the Sg8obX2Q |
01:29.01 | drmessano | useradd -c "Asterisk PBX" -d /var/lib/asterisk -s /bin/false asterisk |
01:29.02 | drmessano | openssl rand -base64 10 | passwd asterisk |
01:29.02 | drmessano | mkdir /var/run/asterisk |
01:29.02 | drmessano | mkdir /var/log/asterisk |
01:29.02 | drmessano | chown -R asterisk:asterisk /var/run/asterisk |
01:29.02 | drmessano | chown -R asterisk:asterisk /var/log/asterisk |
01:29.07 | drmessano | Thats from my playbook |
01:29.08 | netpro25_ | ah |
01:29.44 | netpro25_ | ah same shit |
01:29.52 | netpro25_ | let me try something |
01:29.56 | cviniciusm | Error...error... |
01:30.04 | cviniciusm | ...sorry. |
01:30.06 | drmessano | You have /var/run |
01:30.15 | drmessano | Check your asterisk.conf |
01:30.20 | netpro25_ | okay |
01:32.29 | *** join/#asterisk Kumbang (n=whazzup@125.163.83.153) |
01:33.18 | netpro25_ | drmessano: sweet. For some reason the default ubuntu asterisk.conf had just directories and not other info |
01:33.23 | netpro25_ | changed that |
01:33.30 | cviniciusm | I have installed Asterisk 1.6.1.4, but I don't known what's the port udp/5000 ? |
01:33.48 | netpro25_ | not sure what you mean |
01:33.53 | netpro25_ | what it is used for? |
01:33.59 | drmessano | udp 5000? |
01:34.20 | drmessano | Did you get the tarball from The Pirate Bay? |
01:34.25 | netpro25_ | lol |
01:34.44 | netpro25_ | yay no more GSM bugs |
01:34.45 | cviniciusm | Yes, netstat -lunp shows port udp/5000 for the process asterisk. |
01:35.18 | cviniciusm | No, I get one from Digium site. |
01:35.20 | drmessano | Does it have any open connections to russellbryant.net? |
01:35.36 | netpro25_ | lol |
01:36.49 | netpro25_ | drmessano: okay so I have some packages that are haning out in the auto-remove from the old asterisk ubuntu install. Do you think there is any possibility that the new asterisk install is dependent on those? |
01:37.18 | drmessano | They're probably old, obsolete shit.. So, doubtful |
01:37.27 | netpro25_ | okay gonna auto-remove |
01:38.09 | drmessano | Friends dont let friends install from apt-get |
01:38.09 | cviniciusm | russelbryant.net is a domain from godaddy.com . |
01:38.25 | *** part/#asterisk FoxValley (n=foxvalle@Elgn-Mlnm-WiFI.FoxValley.net) |
01:38.52 | drmessano | Awesome, you're just one step away from sleeping in the bushes outside Russell's house |
01:39.01 | drmessano | Note: bring blankets |
01:39.21 | drmessano | and lots of cat food |
01:39.38 | netpro25_ | drmessano: I still have some sound hickups |
01:39.50 | netpro25_ | and I installed the most recent trunk release |
01:40.01 | drmessano | Bribe the cats, and maybe you'll get your hands on his toothbrush too |
01:40.05 | netpro25_ | any suggestions? |
01:40.08 | netpro25_ | lol |
01:40.12 | drmessano | netpro25_: Yeah, you dont want trunk |
01:40.20 | netpro25_ | you are scaring circumcision away |
01:40.27 | netpro25_ | okay so I will recompile regualr |
01:40.43 | netpro25_ | how can I uninstall trunk |
01:40.57 | drmessano | make uninstall ? |
01:41.12 | netpro25_ | lol |
01:42.08 | *** join/#asterisk moy (n=moy@CPE001cdfec4cee-CM00080dab8485.cpe.net.cable.rogers.com) |
01:42.47 | netpro25_ | yay more ascii art |
01:43.26 | netpro25_ | so I assume if you delete the source directory then you are screwed |
01:43.31 | netpro25_ | which I did not |
01:43.36 | netpro25_ | but have in the past |
01:43.43 | KyleK | screwed for what |
01:43.46 | netpro25_ | uninstall |
01:43.57 | KyleK | oh i installed to /home/asterisk for that reason |
01:44.21 | netpro25_ | yea I want to make debs to avoid that |
01:44.29 | KyleK | well --prefix=/home/asterisk --localstatedir=/var because I'm crazy insane |
01:44.29 | netpro25_ | but have not found an easy way to do it |
01:44.38 | netpro25_ | hah |
01:44.48 | KyleK | find a debian packaging guide? |
01:45.01 | netpro25_ | yea have not looked into that |
01:45.09 | netpro25_ | guess I have some time know that I am unemployeed |
01:45.16 | netpro25_ | ;-) |
01:45.25 | netpro25_ | asterisk is a neat toy |
01:45.44 | netpro25_ | i find joy in designing IVR's |
01:45.48 | netpro25_ | and extensions |
01:46.40 | KyleK | well basically you just need to hijack the make install part |
01:46.47 | cviniciusm | Is there a patch from 1.6.1.4 to 1.6.1.5 ? |
01:47.04 | netpro25_ | whoa 1.6 |
01:47.09 | netpro25_ | sounds risky |
01:47.48 | drmessano | 1.6 pwns |
01:47.57 | netpro25_ | any major improvements? |
01:47.59 | cviniciusm | How to move from Asterisk 1.6.1.4 to 1.6.1.5 ? |
01:48.50 | drmessano | 1.2 was like Windows 95.. 1.4 was like Windows 98.. 1.6 is like Google OS |
01:49.00 | drmessano | Its a nirvana eutopia |
01:49.10 | netpro25_ | hah, so we went from a full pc to a netbook with a limited OS |
01:49.18 | netpro25_ | nice |
01:49.22 | netpro25_ | sounds like a downgrade |
01:49.39 | drmessano | Communist |
01:49.51 | netpro25_ | hah |
01:50.01 | drmessano | You know how I convince people to use 1.6 |
01:50.02 | drmessano | ? |
01:50.03 | netpro25_ | no I like google, no wait, I love google |
01:50.04 | *** join/#asterisk TJNII (n=TJNII@207.189.199.58) |
01:50.19 | netpro25_ | drmessano: no because if I did I would probly be using it |
01:50.39 | drmessano | I call them on the phone, talk up 1.6 a little.. then I abruptly end the call with "hey, sorry, we have 1.2 on this system here, and I need to reboot".. then I hang up |
01:50.50 | netpro25_ | hah |
01:50.56 | netpro25_ | sweet |
01:51.04 | KyleK | is google os out yet? |
01:51.11 | cviniciusm | hahaha. |
01:51.18 | drmessano | Nope |
01:52.08 | netpro25_ | i am waiting for the day they take over the world |
01:52.14 | netpro25_ | those jerks |
01:52.24 | KyleK | i could use some google back pain relief |
01:52.28 | netpro25_ | hah |
01:52.38 | drmessano | netpro25_: No need to wait |
01:52.47 | netpro25_ | google eye doctor |
01:52.56 | KyleK | google eye doctor beta |
01:52.58 | KyleK | :D |
01:54.07 | netpro25_ | hah |
01:54.07 | drmessano | Go to google |
01:54.15 | netpro25_ | they reinvented the beta |
01:54.18 | netpro25_ | term |
01:54.22 | netpro25_ | whats the url? |
01:54.23 | drmessano | Enter: "What date will google take over the world?" in quote, and hit "Im feeling lucky" |
01:55.30 | netpro25_ | is that your site? |
01:55.31 | netpro25_ | lol |
01:56.36 | netpro25_ | do you know if the new google voice uses asterisk? |
01:56.57 | drmessano | No |
01:57.13 | netpro25_ | no you dont know? |
01:57.21 | drmessano | I dont know |
01:57.27 | netpro25_ | I doubt anyone will ever know |
01:57.34 | netpro25_ | except the engineers |
01:57.36 | KyleK | whats the svn command to check for updates but not do an update |
01:58.04 | netpro25_ | speaking of google |
01:58.22 | drmessano | svn donothing |
01:58.56 | russellb | i wish i knew ... I think grandcentral used it |
01:59.13 | russellb | but they took enough time between acquisition and launching google voice to rebuild the whole thing |
01:59.33 | netpro25_ | yes |
01:59.40 | netpro25_ | are you using it? |
01:59.45 | russellb | google voice? yeah. |
01:59.52 | netpro25_ | I find it kinda a pain |
01:59.59 | netpro25_ | cause you end having two numbers |
02:00.05 | cviniciusm | I think Google uses libjingle. |
02:00.08 | russellb | KyleK: there isn't really a good command for that, actually. :-/ |
02:00.54 | russellb | I suppose what you could do is run "svn info" to see what revision you're at, and let's call that <current_rev> |
02:01.03 | russellb | svn log -r <current_rev>:HEAD |
02:01.38 | KyleK | k i'll give that a shot |
02:05.10 | cviniciusm | So, the development cycle of Asterisk is evolution based? |
02:05.37 | *** join/#asterisk denon (i=root@synapse.subneural.net) |
02:05.37 | *** mode/#asterisk [+o denon] by ChanServ |
02:05.42 | russellb | I'm not terribly sure what you mean by that .. |
02:06.30 | mk12pickle | i preferr the orbital palm sander to the face |
02:06.54 | netpro25_ | okay |
02:06.56 | netpro25_ | ? |
02:07.00 | russellb | blinks |
02:07.04 | drmessano | russellb: FYI, the bushes need a little pruning, and the spicket in the back of the house leaks a little |
02:07.26 | drmessano | russellb: You'll be happy to know I put all you mail back, after I finished smelling it |
02:07.38 | drmessano | your* |
02:07.50 | *** kick/#asterisk [drmessano!n=russellb@asterisk/digium-open-source-team-lead/russellb] by russellb (+17 creepy) |
02:07.51 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
02:07.59 | drmessano | [21:38] <cviniciusm> russelbryant.net is a domain from godaddy.com . |
02:08.04 | drmessano | [21:39] <drmessano> Awesome, you're just one step away from sleeping in the bushes outside Russell's house |
02:08.12 | drmessano | ahhaha |
02:09.11 | drmessano | russellb: Do you own any polydactyl's? |
02:09.33 | cviniciusm | At the end of each evolution cycle, we get a new code ready for use, then the several iterations are patched apart. |
02:10.22 | russellb | patched apart ... is that like picked apart or torn apart? |
02:10.29 | russellb | or is that something different :-) |
02:10.34 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
02:11.17 | russellb | we have feature frozen releases that we maintain with bug fixes, and we release bug fix releases on a fairly regular basis |
02:11.37 | netpro25_ | any ideas on this error |
02:11.38 | netpro25_ | Unable to connect to remote asterisk (does /var/run/asterisk//var/run/asterisk.ctl exist?) |
02:11.48 | russellb | For additional information on asterisk releases: http://www.asterisk.org/node/48602 |
02:11.57 | cviniciusm | The Asterisk has several branches 1.6.0.x, 1.6.1.y, 1.6.2.z and so on. |
02:12.09 | russellb | Yeah, see the link I just posted. That will explain it. |
02:12.15 | russellb | netpro25_: that path is seriously messed up. |
02:12.19 | netpro25_ | hah |
02:12.21 | netpro25_ | it is |
02:12.32 | netpro25_ | where is it pulling that from?> |
02:12.48 | russellb | asterisk.conf? |
02:12.50 | drmessano | You can also ask leif madsen ... he can into it in great detail, and I have his home phone number |
02:13.04 | netpro25_ | bingo |
02:13.18 | netpro25_ | sweet |
02:13.26 | netpro25_ | even more gratifying then google |
02:13.49 | russellb | that will cost you $9.95 |
02:14.07 | drmessano | Yes |
02:14.15 | netpro25_ | damn and I thought adwords was a rip off |
02:14.18 | russellb | drmessano: did you get your SfA license btw? |
02:14.31 | drmessano | Leif will cost you.. if you want something free, we can give you file's.. nobody ever wants to call him |
02:14.48 | russellb | harsh |
02:15.04 | drmessano | russellb: Yeah.. it never came.. I e-mailed customerservice and it was resent in about an hour |
02:15.14 | russellb | cool, glad it got resolved. |
02:16.02 | drmessano | Thanks.. Now I just need some time to play with it |
02:16.40 | *** join/#asterisk OrNix (n=ornix@l151-249-47.static.cn.ru) |
02:18.59 | netpro25_ | well okay so yea what are some other causes of jittery sound |
02:19.13 | raden_work | i want to be able to dial *67 before a number and have it set the caller id to unavailable or unknown how would i go about that |
02:19.38 | *** join/#asterisk KuASha (n=ahmed@203.189.242.181) |
02:19.39 | KyleK | I hope skype gets its wideband codec sorted out soon |
02:19.50 | KuASha | hi guys. |
02:19.57 | drmessano | KyleK: What do you mean? |
02:20.02 | KuASha | i am a newbie, looking for an asterisk expert to work with me on a project :) |
02:20.09 | KuASha | is anyone here for paid work ? |
02:20.17 | russellb | I'll do it for 1 million dollarz! |
02:20.30 | netpro25_ | lol |
02:20.37 | drmessano | guesses its probably illegal |
02:20.53 | KuASha | aww :( |
02:21.04 | russellb | drmessano: ha |
02:21.19 | KuASha | illegal to work in some paid project for opensource software ? |
02:21.56 | KuASha | i belive guys are in thousand of data center where linux are being used are doing illegal job then, such as system administrators :P |
02:22.20 | KyleK | drmessano: my interest in skype is its wideband codec, so before i plunk down for SFA it needs to support wideband, probably have the asterisk server transcode to G.722 or somesuch |
02:22.25 | manxpower | ~answers |
02:22.25 | infobot | it has been said that answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
02:23.44 | russellb | ~questions |
02:23.45 | infobot | remember, there are no stupid questions, just stupid people. <http://www.catb.org/~esr/faqs/smart-questions.html> |
02:23.53 | russellb | oic. |
02:24.34 | russellb | KyleK: do you have wideband endpoints in use already? |
02:24.47 | *** join/#asterisk cyberfab007 (n=cyberfab@CPE001b11cf4f69-CM0014f85c3ada.cpe.net.cable.rogers.com) |
02:25.08 | drmessano | russellb: it never fails.. i defend 5 or 6 guys who need help with high call volume startups using Asterisk that get accused of being spammers, and number 7 comes right out with "I am new to asterisk... I want to spam millions with auto warranty extension calls.. can you help me with my dialplan" |
02:25.12 | drmessano | :( FAIL |
02:25.20 | iflux | anyone here know of some cheap ip kvm solutions? |
02:25.25 | cyberfab007 | I LOVE ASTERISK!!!!!!!!!! |
02:25.34 | iflux | I know that there was one made by dlink for awhile that you could get for like $50 |
02:25.37 | cyberfab007 | just had to get that out |
02:25.41 | iflux | but it was EOL'd |
02:25.43 | russellb | cyberfab007: <3 |
02:25.45 | drmessano | ASTERISK ROCKS |
02:25.50 | drmessano | ... probably |
02:25.56 | russellb | does it also roll? |
02:26.16 | cyberfab007 | Man you know I have been configuring and building asterisk systems for 7 years and this is the first time I have been in the freenode room |
02:26.27 | KyleK | russellb: nope i just have 3 SPA3102's in use right now |
02:26.29 | russellb | does the math ... |
02:26.32 | *** join/#asterisk blkry (n=chatzill@96.37.27.72) |
02:26.34 | russellb | cyberfab007: since 2002, really? |
02:26.49 | cyberfab007 | yegh , like the first system I install were sooooooooooooo basic |
02:26.55 | drmessano | I try not to claim asterisk is the best shit since sliced bread or else the devs get a bad case of LEAD BOTTOM |
02:27.07 | drmessano | "ASTERISK IS.. REALLY NOT BAD" |
02:27.25 | netpro25_ | russellb: any ideas on what other things I should do to diagnose jittery sound? Or anyone else. |
02:27.28 | KyleK | the idea of using a single board x86 for a phone has crossed my mind :) |
02:27.32 | drmessano | cyberfab007: Long time listener, first time caller? Named your first kid Qwell? |
02:27.44 | mk12pickle | jitter is variation in latency |
02:27.49 | iflux | sup drmessano |
02:27.55 | drmessano | I've been using asterisk for 11 years now |
02:27.57 | mk12pickle | if all the latency is the same.. no jitter |
02:28.00 | russellb | netpro25_: I suggest a hammer |
02:28.08 | russellb | (my helpfulness after hours goes down drasitcally) |
02:28.13 | cyberfab007 | Actually First kid is a girl Named Nebula LOL |
02:28.21 | netpro25_ | russellb: let me find one |
02:28.24 | drmessano | cyberfab007: Put down the weed |
02:28.31 | cyberfab007 | I try to , |
02:28.52 | netpro25_ | mk12pickle: latency as in the internet connection latency? |
02:29.01 | cyberfab007 | but yegh man I did my first asterisk install in fall 2002 , it was a small business like 5 employees |
02:29.13 | drmessano | What version? |
02:29.14 | cyberfab007 | was my junior year at ASU |
02:29.17 | KyleK | netpro25_: internet, wifi, overloaded lan |
02:29.26 | drmessano | Augusta State University? |
02:29.29 | drmessano | wowow |
02:29.41 | iflux | ***D00D!*** <-- me using asterisks like 20 years ago on efnet |
02:29.45 | raden_work | anyone have a called id script to block calls with like *67 they could share |
02:30.16 | netpro25_ | damn see ya |
02:30.16 | drmessano | throws a rope over the divide |
02:30.17 | cyberfab007 | I think it was 0.2.0 |
02:30.31 | iflux | ya know.. if I hadn't sent +++ instead of ***.. I totally would have thought that netsplit was suspicious |
02:30.37 | KyleK | raden_work: something like that should be easily available |
02:30.38 | cyberfab007 | Arizona State man |
02:30.42 | iflux | r/hadn't/had/ |
02:30.46 | KyleK | odd that it isn't |
02:30.57 | cyberfab007 | home of the SUN DEVILES WE ROCK !!!!!!!!!! |
02:31.15 | KyleK | raden_work: you make an *67 extension that sets caller id and then Waitexten(10) |
02:31.18 | drmessano | cyberfab007: Marko was sending me Betas over ICQ back when it was called "Star" |
02:31.23 | cyberfab007 | well 0.3.0 did not come till the end of the year there |
02:31.27 | drmessano | cyberfab007: I guess that was around 97 or so |
02:31.40 | raden_work | KyleK, how can i do it so they can dial the whole thing like *671234567890 |
02:31.51 | KyleK | oh god not more pasturbating |
02:31.54 | cyberfab007 | nagh man 0.3.0 was around jan -feb 2003 |
02:31.59 | raden_work | strip first 3 digits |
02:31.59 | KyleK | raden_work: regular expressions match it |
02:32.05 | drmessano | pasturbating? <--- <3 |
02:32.19 | cyberfab007 | LOL , I guss that is some kind of computer sickness |
02:32.20 | cyberfab007 | lol |
02:32.24 | KyleK | ${EXTEN:3} or something |
02:33.00 | cyberfab007 | anyways I thought you all should know my company is making a Joomla extension that will easily provide a front for customer thorough joomla off a elastix box |
02:33.28 | drmessano | You know you've been using Asterisk too long when someone accuses you of taking something out of context and you insist "No, I used a Goto!!!" |
02:33.29 | cyberfab007 | I think 0.4.0 cam like late april early may or somthing like that |
02:33.37 | cyberfab007 | LOl |
02:33.53 | russellb | oh yeah, well, i actually taught mark spencer how to code |
02:34.00 | russellb | so you know, no big deal |
02:34.07 | KyleK | holy crap i may have gotten that syntax right |
02:34.27 | cyberfab007 | Haha , |
02:34.32 | iflux | ya know.. the asterisk box in my house is one of the two things I've put in that have extremely high wife acceptance factor |
02:34.45 | iflux | my wife loves it because I figured out how to use it to save us $50/mo |
02:34.50 | drmessano | russellb: I helped him code Gaim while chatting with him over early Gaim alphas |
02:34.51 | cyberfab007 | who remember blowing the soder off the old 56k modem boards? |
02:35.10 | cyberfab007 | man I was like 17 back then , how time flies |
02:35.12 | manxpower | If you never got the wife I bet you'd be saving much more than $50/month |
02:35.18 | iflux | cyber: you kidding.. I still have an original unsoldered Apple |
02:35.25 | cyberfab007 | LOL |
02:35.33 | drmessano | russellb: It was like the picture in "Back to the future".. i would feed him a bad line of code, and one of my buttons would disappear |
02:35.35 | cyberfab007 | man thats cool |
02:35.37 | KyleK | manxpower: hookers are cheaper in the long run? ;) |
02:35.38 | drmessano | russellb: good times |
02:35.47 | manxpower | KyleK: friends with benefits |
02:35.55 | KyleK | even better |
02:36.03 | cyberfab007 | So listen guys I am rallying the troops soon |
02:36.07 | *** join/#asterisk SaiSoma (n=SaiSoma@adsl-074-167-136-030.sip.mob.bellsouth.net) [NETSPLIT VICTIM] |
02:36.07 | *** join/#asterisk Naikrovek (n=jjohnson@63-252-251-77.ip.mcleodusa.net) [NETSPLIT VICTIM] |
02:36.07 | *** join/#asterisk luckyaba (n=lucky@ip72-194-215-55.sb.sd.cox.net) [NETSPLIT VICTIM] |
02:36.08 | *** join/#asterisk m477au (n=m477au@60.241.150.14) [NETSPLIT VICTIM] |
02:36.09 | *** join/#asterisk lirakis_ (n=lirakis@ool-45760ef7.dyn.optonline.net) [NETSPLIT VICTIM] |
02:36.09 | *** join/#asterisk qdk (n=qdk@0x573d8d8b.bynqu1.dynamic.dsl.tele.dk) |
02:36.09 | *** join/#asterisk fnordus (n=dnall@70.70.0.215) [NETSPLIT VICTIM] |
02:36.09 | *** join/#asterisk skymeyer (n=skymeyer@mailout.itconnect.be) [NETSPLIT VICTIM] |
02:36.09 | *** join/#asterisk Zhad (n=tom@server30261.uk2net.com) [NETSPLIT VICTIM] |
02:36.16 | iflux | cyber: it's in my parents attic.. my mom was like.. "hey can we throw out this super old computer that you're never taking home?" |
02:36.19 | drmessano | THE REVOLUTION IS HERE? |
02:36.29 | drmessano | GREAT, I HAVE THE GRENADES |
02:36.30 | iflux | I was like "*cough*sputter*NO!" |
02:36.33 | KyleK | I guess i gotta stop putting off this caller id view |
02:36.37 | cyberfab007 | any good C programmers in here with php experience and asterisk hard-on? |
02:36.44 | russellb | looks around |
02:36.52 | KyleK | i dont have a hardon |
02:36.54 | drmessano | cyberfab007: you want to code a GUI for Asterisk in PHP? |
02:36.58 | iflux | cyber: got a chan you want developed? |
02:37.00 | cyberfab007 | No , that is done |
02:37.08 | KyleK | you want someone to fix it? |
02:37.13 | drmessano | cyberfab007: you want to code a CLI for Asterisk in PHP? |
02:37.17 | cyberfab007 | No , fixing is for morons |
02:37.23 | cyberfab007 | coding is for pros |
02:37.32 | cyberfab007 | No, |
02:37.46 | KyleK | if it needs fixing the pros weren't pro enough |
02:37.49 | cyberfab007 | I am taking the elastix system and making moduals for it |
02:37.57 | cyberfab007 | excatly |
02:38.04 | drmessano | AHAHHAH |
02:38.08 | drmessano | Elastix? |
02:38.12 | iflux | elastix is another centos based distro, right? |
02:38.16 | cyberfab007 | Elastix |
02:38.17 | cyberfab007 | yes |
02:38.21 | cyberfab007 | but more than that , |
02:38.30 | cyberfab007 | has fram work for the ultimate god box |
02:38.41 | drmessano | Elastix has the worst asterisk binaries ever |
02:38.41 | cyberfab007 | did I say god box |
02:38.46 | iflux | if it were debian or ubuntu based I'd probably be slightly interested but if it's just centos based then I'd have no reason to not use piaf |
02:38.56 | drmessano | or AsteriskNOW |
02:38.59 | drmessano | COUGH COUGH |
02:39.06 | cyberfab007 | people , it is more than that , |
02:39.18 | cyberfab007 | any asshole can install and use elastix , this is important |
02:39.26 | cyberfab007 | Right? |
02:39.36 | drmessano | Any asshole can code and package elastix too, apparently.. |
02:39.38 | iflux | I just don't like the package management system that all the fedora based systems use |
02:39.55 | drmessano | I ran Elastix here for a week |
02:40.10 | cyberfab007 | I have about 40+ business running elastix , for 2 years now |
02:40.14 | drmessano | The next weekend, I put that box to better use and installed Vista on it |
02:40.26 | cyberfab007 | OHHH MAN THAT IS HARSH TALK |
02:40.30 | cyberfab007 | HARSH TALK |
02:40.57 | cyberfab007 | i have elastix cluster that i have not updated in a year cause I am scared and it still runs like a rock |
02:41.26 | drmessano | It's a cluster alright |
02:41.34 | drmessano | ba-dump CHING |
02:42.01 | cyberfab007 | Actually I am gonna upgrade with elatix 2.0 that will be running asterisk 1.6 |
02:42.11 | drmessano | Seriously, Elastix is slowsterisk.. They compile their binaries on a 286 |
02:42.49 | iflux | drmessano: and they don't even use -O3 I bet |
02:42.56 | raden_work | am i doing something wrong caller id still coming through http://pastebin.com/d20e819f5 |
02:43.02 | iflux | -O3 -funroll-loops, etc |
02:43.19 | drmessano | iflux: Damnit, you went over my head.. Now I need to google |
02:43.34 | cyberfab007 | But it is a solid distro , better than any other out there unless you install your own asterisk box running freepbx on you own distro |
02:43.35 | KyleK | raden_work: my itsp has a setting in the web interface to force caller id |
02:43.46 | cyberfab007 | But this is the call |
02:43.51 | cyberfab007 | Free networks , |
02:43.52 | KyleK | or you're setting the caller id at *67 and in @to-callcentric ;) |
02:43.53 | iflux | drmessano: compiler flags that optimize the hell out of the code.. |
02:43.56 | drmessano | cyberfab007: AsteriskNOW and PIAF are far more stable.. Elastix is slower than Trixbox |
02:44.04 | KyleK | pbx in a fire |
02:44.16 | KyleK | who burns a pbx? |
02:44.25 | iflux | I should make a super optimized version some time.. I bet you could squeeze a few hundred more calls out of an optimized system |
02:44.25 | cyberfab007 | Humm , before I ran asterisk @ home |
02:44.31 | cyberfab007 | than trixbox |
02:45.06 | raden_work | KyleK, trying to set it to something anonymous |
02:45.06 | iflux | i hate the green machine |
02:45.10 | cyberfab007 | than they went F***ed , and went astray of GPL , I could never leave trix box running turn the key and come back to delete logs , Elastix let me do that |
02:45.26 | KyleK | raden_work: 6042809000 |
02:45.36 | cyberfab007 | Anyways , |
02:46.21 | KyleK | then set the name to be Dialup Modem Pool ;) |
02:46.28 | drmessano | You never had to worry about Elastix being exploited because the boxes were too slow with their unoptimized binaries to make them worth using, even over dialup |
02:46.50 | raden_work | KyleK, ? |
02:46.53 | cyberfab007 | I am making some software that will do alot of things , one of them is letting someone install a joomla extension that will give them a customer frontend in joomla from a elastix box with a gateway to DIDX.net , set up a complete phone business in less than a day I say , but thats where it starts I pay a development team to do this |
02:46.55 | KyleK | joking |
02:47.02 | raden_work | lol |
02:47.03 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
02:47.13 | drmessano | AHHHHHH |
02:47.15 | KyleK | i always put in the number for my old dialup isp when sites ask for my number |
02:47.17 | drmessano | Youre an idea guy |
02:47.22 | drmessano | Well |
02:47.28 | drmessano | ~happyclownpbx |
02:47.29 | infobot | [HappyClownPBX] is currently in closed beta, approaching 12GB in size, uses Asterisk for it's core, it pwns, is also now compatible with the Diahatsumashiniriki Keyotason 200LP-A11 SIP phone |
02:47.38 | drmessano | ^^^ Bidding starts at 1 BILLION dollars |
02:47.58 | raden_work | and i cant dial out with vitelity for some reason :( |
02:47.59 | raden_work | http://pastebin.com/d3bdfe579 |
02:48.02 | KyleK | erm |
02:48.11 | raden_work | <PROTECTED> |
02:48.15 | raden_work | without caller id |
02:48.16 | cyberfab007 | i love vitelity , 2 years and not one issue |
02:48.25 | KyleK | raden_work: why are doing a Dial to local btw |
02:49.26 | cyberfab007 | But hey who would like to take and asterisk box , and provide a front end for people to govern them selfs with asterisk running as the communications engine? |
02:49.38 | drmessano | cyberfab007: You remind me of this guy I used to talk to on Undernet years ago.. he could tell you 100 different household items you could grind up and throw in a hookah, and would finish with "Its all legit, man" |
02:49.45 | raden_work | SIP dont work cause the rest of the dial plan in outbound context |
02:49.49 | raden_work | i use SIP it dont work |
02:50.15 | KyleK | cant use goto? |
02:50.20 | iflux | drmessano: hahahahaha I remember someone like that |
02:50.27 | KyleK | i guess it doesn't matter |
02:50.34 | drmessano | Damn, what was his nick |
02:50.59 | drmessano | Was a food item, I think |
02:51.01 | KyleK | drainbamaged? |
02:51.12 | cyberfab007 | Funny |
02:51.37 | iflux | it wasn't ratsalad was it? |
02:51.41 | *** part/#asterisk mk12pickle (n=johaan99@dhcp063-129.navigonet.com) |
02:51.48 | drmessano | Flapjack |
02:51.52 | drmessano | That was it |
02:52.03 | drmessano | That guy was... burned |
02:52.55 | *** join/#asterisk LiNeTuX (n=LiNeTuX@rrcs-71-43-111-78.se.biz.rr.com) |
02:53.37 | cyberfab007 | I am working not only on the joomla app , but also on a smart house app that lets you use Luminvox to speak commands to your house and lets asterisk runs scripts that produce results such as food status in your fride , IM contact on line and anything else you can think of to ask your asterisk box to figure out for you LOL 6 months in development this package is |
02:54.34 | drmessano | cyberfab007: Working as in "writing code" or "idea"? |
02:54.56 | iflux | cyber: do you get a 5D6 savings roll for 'forced out of parents basement'? |
02:54.57 | raden_work | anyone have any idea why i cant dial out with vitelity ? http://pastebin.com/d3bdfe579 |
02:55.03 | cyberfab007 | Imagin being on your laptop and asking your asterisk system to bring up your survalince equipment , whether you need eggs and milk or your cousin recently has posted on facebook |
02:55.05 | KyleK | I'm not an idea man but i come up with enough ideas to keep me busy |
02:55.06 | iflux | :-P |
02:55.22 | KyleK | raden_work: incorrect password? |
02:55.45 | drmessano | cyberfab007: Id prefer a freePBX facebook application so I only have to keep one browser window open at home |
02:55.47 | raden_work | I can register inbound works fine |
02:56.03 | *** join/#asterisk mumtazah (n=mumtazah@203.82.91.103) |
02:56.03 | cyberfab007 | There is code, but I have investors and stupid policeys and stuff , it will be GPL but not till it is released |
02:56.23 | drmessano | So what percentage is done.. Estimated lines of code? |
02:56.30 | drmessano | "Napkins do not count" |
02:56.31 | KyleK | drmessano: facebook uses xmpp doesn't it? |
02:56.39 | drmessano | KyleK: Not currently |
02:56.45 | KyleK | oh |
02:56.46 | cyberfab007 | raden-work , if you have half a brain and cant get vitelity workinf with there auto config script on their site , it is porbley the Distro your using |
02:57.13 | drmessano | KyleK: They talked a big game about XMPP interop, but it turned out to be bullshit vaporware |
02:57.27 | cyberfab007 | but what I am here for is to rally the troops |
02:57.46 | iflux | drmessano: hahaha.. it'd be fun to have your asterisk box post up facebook wall messages about who called you |
02:57.49 | iflux | I should do that |
02:57.49 | drmessano | cyberfab007: Sorry, fresh out of Kool Aid, and my sneakers are in the wash |
02:58.09 | drmessano | iflux: Easy.. Twitter.. > Facebook |
02:58.30 | cyberfab007 | drmessano do you believe in GPL , like fucking religion man ? |
02:58.32 | iflux | drmessano: yeah that'd work.. |
02:58.34 | drmessano | Twitter API is super simple.. ive played with Twitter + Asterisk quite a bit |
02:58.56 | cyberfab007 | gforce.ichorcom.com |
02:59.09 | iflux | cyber: personally.. I love fucking religion.. those catholic chicks are always shouting Oh God! |
02:59.20 | cyberfab007 | my public devel site |
02:59.53 | cyberfab007 | Ohh they are the best , here in Toronto I lived in Forrest Hill , man those girls were out of contorl |
03:00.23 | KyleK | hey can i access cdr unique id from within app_voicemail? |
03:00.34 | drmessano | Seems like the only project is a week old and it has one dev |
03:01.07 | KyleK | hehe gforce |
03:01.10 | iflux | "All connections are made at the rear" - adder.us |
03:01.49 | cyberfab007 | Well they are the openprojects |
03:01.53 | drmessano | hans paperbag is the only dev, and that Jeremy guy keeps fucking blogging.. I bet hans is bitter jeremy cant be bothered to commit one stinkin line of code |
03:01.55 | cyberfab007 | I have about 8 that work for me |
03:02.15 | drmessano | http://gforce.ichorcom.com/gf/project/elajoom/ |
03:02.25 | raden_work | <PROTECTED> |
03:02.39 | cyberfab007 | You should install the extension fuckign awsome |
03:03.07 | cyberfab007 | next week , most of the basic pageviews will be done inside of joomla |
03:03.39 | drmessano | By hans? |
03:03.40 | cyberfab007 | the week after joomla, vtiger, a2billing will be synced |
03:03.48 | drmessano | When do we see jeremy commit? |
03:04.10 | KyleK | raden_work: you're probably calling out via the sip account thats attached to right? |
03:04.11 | cyberfab007 | Well no Hanes actually has had to quite becasue of personal emerganices with his lady friend |
03:04.12 | drmessano | Hans already seems bitter.. It wont be long now before total meltdown |
03:04.22 | drmessano | Oh, only dev quit? |
03:04.28 | drmessano | BRIGHT future |
03:04.38 | cyberfab007 | Well he is a Joomla core developer |
03:04.47 | cyberfab007 | so he has much on his plate already |
03:04.47 | drmessano | and your only coder |
03:04.56 | cyberfab007 | well no as I said this is my open site |
03:04.57 | cyberfab007 | lol |
03:05.05 | raden_work | KyleK, http://pastebin.com/d49c0b549 |
03:05.14 | drmessano | and now you're here, because you have blog posts and need someone to write 10,000 lines of code to go with them? |
03:05.18 | drmessano | Right? |
03:05.28 | iflux | bahahaha drmessano.. that's not nice |
03:05.49 | cyberfab007 | Actually , most the code you see on that site I made and Hannes put it in Joomal fram work |
03:06.01 | cyberfab007 | and no for those project I have lots of developers |
03:06.09 | cyberfab007 | about 10 that I pay |
03:06.31 | drmessano | Oh? |
03:06.39 | drmessano | Public repos we can look at? |
03:06.56 | cyberfab007 | for this project I am thinking a E- Democracy Project , |
03:07.13 | *** join/#asterisk geneticx (n=geneticx@70.146.116.10) |
03:07.18 | *** join/#asterisk shinao1 (n=shinao1@41.219.205.197) |
03:07.50 | KyleK | raden_work: so vitelity-outbound is configured to be a different sip peer than vitelity-inbound? |
03:07.55 | drmessano | Which project? The one with no devs, or the mutlimilliondollar 10 dev coding tank we can't get a link to? |
03:08.06 | raden_work | KyleK, yeah thats what they sent me |
03:08.08 | cyberfab007 | a project that can only be done by people for people |
03:08.37 | raden_work | but callcentric is my 866 -452-3565 number that keeps getting forbidden i dont understand whats going on |
03:09.08 | drmessano | You're recoding Obama? |
03:09.09 | raden_work | [Aug 31 21:47:20] WARNING[10802]: chan_sip.c:15268 handle_response_invite: Received response: "Forbidden" from '"866-452-3565 x101" |
03:09.21 | raden_work | im x103 so im really confussed |
03:09.30 | cyberfab007 | Ichorcom is Deleware corporation with 20 million shares ammended and has a million dollars in funding. Is registerd with the SEC and is filling it FORM D offering for you ino |
03:09.32 | cyberfab007 | info |
03:09.58 | KyleK | raden_work: who is 69.179.99.17 |
03:10.07 | KyleK | oh its you |
03:10.09 | raden_work | WAN connection |
03:10.28 | raden_work | WTF the aastra phone set the called id itself |
03:11.06 | KyleK | well getting that forbidden response is separate from dialing vitelity |
03:11.24 | KyleK | before i checked your ip i thought vitelity was that ip |
03:11.44 | raden_work | vitel-outbound/tanning 64.2.142.29 5060 OK (45 ms) |
03:11.44 | raden_work | vitel-inbound/tanning 64.2.142.15 5060 OK (45 ms) |
03:11.45 | raden_work | callcentric/17772445766 204.11.192.38 5060 OK (45 ms) |
03:12.08 | *** join/#asterisk ketema (n=ketema@ketema.net) |
03:13.17 | cyberfab007 | http://www.edgarcompany.sec.gov/servlet/CompanyDBSearch?page=detailed&cik=0001465570&main_back=2 |
03:13.21 | cyberfab007 | if you were looking |
03:14.01 | drmessano | www.gigllc.com was more interesting |
03:14.12 | drmessano | www.gigpllc.com was more interesting |
03:14.14 | drmessano | My bad |
03:15.14 | cyberfab007 | HA i know the joomla template the are using LOL |
03:15.59 | drmessano | GLOBAL INVESTMENT GROUP, LLC, owner of telenity.net, which points to your site? |
03:16.20 | raden_work | [Aug 31 22:16:04] WARNING[10802]: chan_sip.c:15268 handle_response_invite: Received response: "Forbidden" from '"103" <sip:tanning@69.179.99.17>;tag=as2f4e3703' |
03:16.21 | cyberfab007 | if they do I dont know about it LOL |
03:16.36 | drmessano | Are you serious? |
03:16.46 | drmessano | http://telenity.com/gf/ |
03:16.52 | *** join/#asterisk blackest_mamba (n=blackest@c-71-239-160-143.hsd1.il.comcast.net) |
03:17.24 | drmessano | telenity.com, rather |
03:17.32 | drmessano | which, BTW.. redirects to /gf |
03:17.48 | raden_work | http://pastebin.com/d47cb8712 |
03:17.49 | *** join/#asterisk blackest_mamba (n=blackest@c-71-239-160-143.hsd1.il.comcast.net) |
03:18.13 | drmessano | cyberfab007: Have you ever used G726? |
03:18.16 | cyberfab007 | LOL , |
03:18.29 | hesco | is it unusual for multiple calls to have the same channel ID? |
03:18.41 | cyberfab007 | I get this VPS through ultrahosting , they must own this IP before or somthing lol |
03:18.49 | KyleK | hesco: at the same time? yup |
03:18.58 | hesco | sequentially |
03:19.19 | KyleK | I've had some of that |
03:19.28 | hesco | I'm doing a series of tests and I keep getting the same Channel ID |
03:19.51 | hesco | I would have thought that each would be unique |
03:20.09 | KyleK | well CDR(uniqueid) should be unique |
03:20.33 | *** join/#asterisk zerko (i=zerko@srv1.techality.com) |
03:20.49 | cyberfab007 | will only for very low bandwidth situations is has the backwards compatibility with asterisk 1.2 but I am no expert |
03:21.11 | cyberfab007 | it is old codec from like 90 or 91 |
03:21.15 | cyberfab007 | one of the first |
03:21.40 | KyleK | DECT uses G726 |
03:21.54 | drmessano | Actually, its not backwards compatible with 1.2.. The implementation in 1.4 and later changes, requiring a unique codec name |
03:21.56 | KyleK | so I'd think its use is on the uprise right now |
03:21.59 | *** part/#asterisk mumtazah (n=mumtazah@203.82.91.103) |
03:22.24 | drmessano | G726 in 1.2 != G726 in 1.4 |
03:22.56 | *** join/#asterisk yidiyuehan (n=yidiyueh@bb116-14-76-211.singnet.com.sg) |
03:23.32 | cyberfab007 | yes you are right , it has NO back wards compatiblity in 1.2 |
03:23.47 | cyberfab007 | but is older codec |
03:24.02 | iflux | I don't get why companies don't just spend the money on the g.729 license.. is it really THAT expensive? |
03:24.47 | cyberfab007 | yes , it is when you get to a call load of like 30,000 and they want like 250,000 for a license fee , |
03:25.04 | *** join/#asterisk netpro25_ (n=mmanning@c-71-226-86-184.hsd1.fl.comcast.net) |
03:25.06 | cyberfab007 | be different if you had milliions of customer but for start up makes no sense |
03:25.22 | zerko | anyone here have servers in dallas? |
03:25.24 | cyberfab007 | hard to pitch to investors |
03:25.43 | zerko | i have a few if anyone needs one |
03:25.45 | netpro25_ | Anyone ever use Colopronto? |
03:26.47 | cyberfab007 | drmessano pm me your skype you are obiousily the regular here |
03:26.49 | iflux | see.. it seems like you could build something so that g.729 is an addon module that is easily purchased by the customer.. basically enabled via a license key |
03:27.06 | iflux | cyber: drmessano's first day here was like 3 days ago.. |
03:27.10 | cyberfab007 | g729 should be opensource , |
03:27.22 | cyberfab007 | your shitting me it is my first day he is knowlageable |
03:27.23 | iflux | g729 is open source.. for um.. europe |
03:27.31 | cyberfab007 | g729 should be opensource |
03:27.36 | netpro25_ | lol |
03:27.45 | iflux | cyber: yeah.. I've been here like 7 years and I've only seen him this last week |
03:28.08 | KyleK | cyberfab007: its patent encumbered til 2012 or 2014 |
03:28.09 | cyberfab007 | really , I have avoided this room for some years but I finally need some help |
03:28.37 | cyberfab007 | its bull shit and another attempt by the power that be to be greedy and not publish GPL , will they ever get it? |
03:28.40 | iflux | cyber: yeah.. drmessano is actually 3 guys.. you never know which you're talking to at the time.. |
03:28.50 | iflux | drmessano: are you bill, rich, or harry tonight? |
03:29.06 | cyberfab007 | I think he may be flake |
03:29.46 | drmessano | iflux: LOL |
03:30.24 | cyberfab007 | Kidding , goign for a smoke , brb |
03:30.35 | drmessano | cyberfab007: i am the flake? Your company had one post in March about your "big ambitions, then nothing until 2 weeks ago, followed by spamming on every forum known to man. You guys get out much? |
03:30.59 | *** part/#asterisk SimplyZero (n=drewyate@pool-96-238-62-45.prvdri.fios.verizon.net) |
03:31.04 | drmessano | Lemme guess |
03:31.23 | drmessano | Ran out of quarters at Kinkos.. needed time to print the 20,000,000 shares? |
03:31.36 | raden_work | <PROTECTED> |
03:31.36 | raden_work | <PROTECTED> |
03:31.36 | raden_work | <PROTECTED> |
03:31.39 | iflux | drmessano: basically.. he made his savings throw.. |
03:32.01 | drmessano | D20000000? |
03:32.27 | drmessano | What I dont understand is |
03:32.57 | iflux | goodnight all |
03:33.12 | drmessano | If hes going to code/steal/beg for help on developing Joomla and Vtiger integration with Asterisk, why even use the Elastix GUI? |
03:33.19 | drmessano | Not welll thought out |
03:33.22 | drmessano | Night iflux |
03:34.21 | cyberfab007 | No that is paid for alreday |
03:34.23 | cyberfab007 | lol |
03:34.53 | zerko | anyone need a dedicated in the infomart? (dallas, tx) |
03:35.26 | cyberfab007 | you will see new release in about a week that will be complete except vitger,joomla, and a2billing intergration , |
03:35.31 | cyberfab007 | I need no help with this |
03:35.42 | cyberfab007 | What I want help on is another project |
03:35.53 | cyberfab007 | one that is just planing , |
03:36.51 | cyberfab007 | could Asterisk be the engine that is the basis for one online community that runs government? maybe! |
03:37.27 | raden_work | exten => _*67.,1,SET(CALLERID(number)=0000000000) |
03:37.27 | raden_work | exten => _*67.,1,Dial(LOCAL/${EXTEN:3}@outbound) |
03:37.31 | cyberfab007 | could asterisk lead the way in American Democracy 2.0? |
03:37.31 | raden_work | am i doing something wrong ? |
03:37.45 | drmessano | you will see new release in about a week that will be complete except vitger,joomla, and a2billing intergration , <-- Joomla, Vtiger are the entirety of what youve been spamming.. if those are not in there, then what the hell will be complete? |
03:38.12 | cyberfab007 | well buddy , no one code extensions like that overnight |
03:38.19 | cyberfab007 | you think I am GOD or somthing? |
03:38.59 | drmessano | If the integration of none of those items will be complete, what will this release be good for.. screenshots? |
03:39.09 | cyberfab007 | first lets get the extension manging the elastix services , than we will set up the billing and CRM intergration as I said that project is funded and on its way |
03:39.13 | raden_work | is there a way to disable caller id with vitelity ? |
03:39.21 | KyleK | damn i cant find the delcaration of ast_channel |
03:39.23 | cyberfab007 | Not here for that |
03:39.25 | yidiyuehan | hi, anybody knows why if I originate two calls through asterisk manager interface, the second exten won't be called unless the first call has been answered? |
03:39.38 | KyleK | raden_work: its only setting it to your 866 number? |
03:39.42 | drmessano | If the integration of none of those items will be complete, what will this release be good for.. screenshots? |
03:39.52 | netpro25_ | question about latency. Is tracert a good way to determine if I have a latency problem. Obviously I do since I am getting jitter. |
03:39.56 | raden_work | KyleK, no its always unavailable even when i dont touch it |
03:40.36 | drmessano | cyberfab007: what exactly will be in this next release if not the aforementioned features? |
03:40.54 | KyleK | odd |
03:42.12 | raden_work | how do i make asterisk try to reregister a peer if that makes sense |
03:42.21 | drmessano | I have this great idea.. its a search engine where instead of putting in keywords and it producing search results, you put in the search results and it produces the search terms. I'm not sure if it's even useful, but we already hired the sales and marketing teams, so who gives a shit |
03:42.27 | raden_work | vitel-outbound now unreachable : ( |
03:42.50 | cyberfab007 | Well a customer in joomla will be able to purches packeges with a number of extensions, view and edit the extensions in the packeges , view the CRD's on the account , purches and manage trunks , purches and manage 50+ dids from didx.net and Have the Joomla admin set all this up in joomla in less than a hour |
03:42.51 | zerko | netpro25_, do you see any delays in the traceroute? |
03:43.20 | netpro25_ | Okay so when I tracert my sip providers server |
03:43.23 | cyberfab007 | I will be wraping the ARI in this extension of course |
03:43.36 | cyberfab007 | for extension based management |
03:43.42 | netpro25_ | I get anywhere from 30 -90 ms |
03:43.57 | drmessano | Wy not just rewrite the ARI.. its crap, and I am sure the freePBX team would love a rewrite |
03:44.23 | cyberfab007 | ahh but it works and I wiill not re-invent the wheel investors dont like that |
03:44.31 | cyberfab007 | if it is there use it and move on , |
03:44.38 | cyberfab007 | ARI servers its purpose |
03:44.39 | netpro25_ | zerko: http://pastebin.com/m6bce59bb |
03:44.44 | drmessano | What happened to GPL democracy utopia? |
03:44.57 | drmessano | Oh, that shit only works if they're coding for you, right? |
03:45.15 | netpro25_ | drmessano: hah |
03:45.44 | netpro25_ | is this like a public board room or something |
03:45.55 | raden_work | KyleK, yeah caller id just came uo unavailable |
03:45.56 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
03:45.58 | netpro25_ | where are the sharks? |
03:46.30 | drmessano | Your model of a GPL democracy fails if the utopia only exists inside your ecosphere.. we call that a CULT, not a democracy |
03:47.04 | cyberfab007 | You obviously have no clue what I am hinting at |
03:47.29 | cyberfab007 | this is not a virtual world buddy get your head out of your ass , I am no virtual world freak |
03:47.53 | drmessano | Code for shares in this so-called company, I get it |
03:47.55 | cyberfab007 | I am talking a facebook dedicated to government powerd by asterisk for communications |
03:48.02 | drmessano | Paid in hours of work, at the end of the month |
03:48.39 | cyberfab007 | well if someone writes code the least least least I can do is give them shares , especially if paid positions are all filled up 10 of tehm |
03:48.42 | cyberfab007 | them |
03:49.12 | cyberfab007 | better shares than nothing , what other project is offering this? |
03:49.15 | drmessano | what is the estimated value of these shares? |
03:49.30 | cyberfab007 | well right now about 75 cents |
03:49.56 | cyberfab007 | I look to go on a market next april , maybe pinksheets |
03:49.58 | drmessano | 20 million shares, 10 million dollar investment.. thats 50 cents an hour to code for you.. Thats cheaper than india |
03:50.17 | cyberfab007 | its not about the money dude |
03:50.25 | drmessano | Of course not |
03:50.31 | drmessano | How could it be? |
03:50.56 | cyberfab007 | people who get paid by be as a job get 15 minum a hour and 30 max if your good |
03:51.11 | *** join/#asterisk brian (n=brian@unaffiliated/brian) |
03:51.17 | drmessano | and the others code for 50 cents an hour in their spare time? |
03:51.59 | cyberfab007 | well if they are helping in the community hopfully there code will do well and my stock 3 years from now will be at 20-30 per share that would be nice for everyone |
03:52.49 | cyberfab007 | it is gratitude to the community that I am doing this not because I want to attract people you have alot to learn my friend |
03:53.05 | drmessano | Hmm.. which would require the company to be valued at 40 million to 60 million? |
03:54.05 | cyberfab007 | Well I hope I do well , if I have people eventually writing code for me for free the LEAST I can do is give them a part of the company , better than all the other project out there |
03:54.51 | cyberfab007 | for instance I dont see kevin flemming or mark spencer giving us a part of Digium? |
03:55.03 | cyberfab007 | for writing all this code do I ? |
03:55.21 | cyberfab007 | What if they did , |
03:55.23 | drmessano | Yeah, no shit.. Digium owes me like $18.92 for all the debugging I have done for Asterisk, at 50 cents an hour |
03:55.48 | cyberfab007 | buddy read your shit right 50 shares an hour |
03:56.07 | cyberfab007 | which mean if you give me 20 hours a month that 1000 share |
03:56.33 | cyberfab007 | if my stock goes to 10 dollars two years form now that 10,000 dollars a month you just got |
03:56.53 | cyberfab007 | I hope it goes to 40 or 50 |
03:57.10 | raden_work | exten => _*67.,1,SET(CALLERID(number)=0000000000) |
03:57.10 | raden_work | exten => _*67.,1,Dial(LOCAL/${EXTEN:3}@to-callcentric) |
03:57.14 | cyberfab007 | based on dividen of course you want investment package I send out ? |
03:57.19 | cyberfab007 | ha you have no money |
03:57.23 | raden_work | tat wont even let me dial out :( |
03:57.47 | cyberfab007 | minum but in is 13,000 dollars , |
03:58.28 | KyleK | raden_work: 1 |
03:58.29 | KyleK | then 2 |
03:58.38 | drmessano | I did my math wrong.. For you to hit 20 to 30 dollars a share, that means you're looking a valuation of 400 million to 600 million |
03:58.39 | KyleK | exten => _*67.,2,Dial(LOCAL/${EXTEN:3}@to-callcentric) |
03:58.53 | drmessano | and $10 a share would be $200 million |
03:59.06 | raden_work | KyleK, omfg i thought i had n there been working on this 2 long lol thanks lemee try |
03:59.08 | cyberfab007 | or a dividend of about 40 - 60 cents a share |
03:59.20 | KyleK | raden_work: I occasionally do exten => 123,1,Noop() then exten => 123,n,etc |
03:59.39 | cyberfab007 | as I said I have the high market cap because I plan on giving out lots of stock |
03:59.48 | raden_work | KyleK, what does that do ? |
03:59.59 | KyleK | noothing operation |
04:00.04 | *** join/#asterisk rizwank (n=rizwank@pool-71-118-51-238.lsanca.dsl-w.verizon.net) |
04:00.21 | KyleK | http://en.wikipedia.org/wiki/Noop standard term |
04:00.46 | drmessano | Well, good luck.. I think your first order of business should be to find a developer, because is hans paperbag quit, and no one else is commiting any code, you're fucked |
04:00.54 | rizwank | Hi there. I'm looking for a way to have Asterisk take an incoming SIP call, and connect it to the PTSN (so the call effectively contains some extra data with a DID, and when Asterisk gets it, it completes the call via the PTSN.) Is there a name for this feature? |
04:01.09 | raden_work | KyleK, works with vitelity now but not callcentric |
04:02.10 | cyberfab007 | HA this week there will be lots of code commited , by 4 differnt developers , no worries |
04:02.28 | drmessano | Dude |
04:03.09 | drmessano | Im not the one that should be worried.. you got a 10 million dollar venture cap investment by a company that includes in their south african operations, the mining of blood diamonds |
04:03.13 | cyberfab007 | what you do sit here on IRC trying to demorilize people , ha you want pic of my 53 story penthouse over lookign the lake all because of asterisk ? Get a life buddy and stop trying to demorilize others |
04:03.14 | drmessano | YOU should be worried |
04:03.41 | *** join/#asterisk dysinger (n=tim@71-20-35-99.war.clearwire-wmx.net) |
04:03.52 | cyberfab007 | I have nothing to do with those peopel other than I got an IP address they used to have my friend |
04:04.53 | cyberfab007 | Let me roll a joint now , |
04:05.31 | geneticx | lol |
04:06.31 | cyberfab007 | so anyone in here interested in having there own online asterisk business for free? This is the extension I am making in joomla I need no help , only people to report bugs to my developers and test, but I am not here for that , I am here for asterisk and E Democract |
04:06.34 | drmessano | Interesting |
04:07.08 | drmessano | CEO of a multmilliondollar corporation talking about rolling joint on a logged IRC channel |
04:07.10 | cyberfab007 | What is E- Democracy? |
04:08.16 | cyberfab007 | Well thats the point socitey is wrong in wanting the fake professional face of a well seasoned politicant , my netwok accepts people for who they are , and in canada weed is accepted my friend |
04:08.43 | cyberfab007 | I am not a mulitmillion dollar coproatino |
04:08.47 | cyberfab007 | corporation |
04:09.02 | cyberfab007 | jsut a guy with a few project and some cash thats all , |
04:09.02 | raden_work | KyleK, thanks for eveything vitelity working at least I have to set my Call ID but thats ok :) |
04:09.08 | drmessano | You said you had a 10 million dollar investment.. did you smoke it already? |
04:09.51 | manxpower | At least tonight's drivel is different than most night's drivel. |
04:10.19 | manxpower | Starting your own business is far overrated. |
04:10.19 | cyberfab007 | I said a million that was last year |
04:10.34 | rizwank | hmm. |
04:10.40 | *** join/#asterisk dongs (n=blogger@l212047.ppp.asahi-net.or.jp) |
04:10.51 | dongs | hi, what setting in sip.conf controls "Expires: " header going out |
04:10.58 | cyberfab007 | now I do a reg D offering that lets me rais another million between 133 investors dude get your shit stright , I feel I am wasting my fingers |
04:11.08 | dongs | i mean i would just grep the sores for this,but y'know, in 2009 i was hoping I didn't have to do that. |
04:11.11 | manxpower | dongs: if it can be set, it would be listed in sip.conf.sample |
04:11.26 | cyberfab007 | Dongs that cant be set I dont thin k |
04:11.29 | dongs | ... |
04:11.38 | dongs | well on 1.4 its 3600 and on 1.6 its 120. |
04:11.57 | *** part/#asterisk grahamsaa (n=administ@cpe-74-67-180-93.rochester.res.rr.com) |
04:12.06 | dongs | 1.4 can register to my silly ip provider, 1.6 cannot. |
04:12.14 | cyberfab007 | dongs I have never used that before , |
04:12.16 | dongs | so i'm going comparing 1:1 and seeing the differences in headers. |
04:12.25 | manxpower | dongs: it might be a lot of work, but you should check UPGRADE*.txt and the changelog |
04:12.29 | cyberfab007 | usally I set the device to determain that in the device config |
04:15.23 | cyberfab007 | I think in asterisk 1.6 you may be able to define the expire time in sip.conf but do not quote me #:^) |
04:15.25 | *** join/#asterisk errotan (n=errotan@62.201.122.164) |
04:16.22 | cyberfab007 | http://lists.digium.com/pipermail/asterisk-dev/2003-September/001568.html |
04:16.29 | cyberfab007 | search expire on the page ok |
04:16.39 | dongs | 1.6 is missing "Event: registration" |
04:17.42 | cyberfab007 | twords the bottem of that link it talks abou that |
04:17.57 | cyberfab007 | We add a few extra features in each sip peer. |
04:17.57 | cyberfab007 | ; expire = the number of seconds that this registration should |
04:17.57 | cyberfab007 | ; indicate for expiry. Default is 500. |
04:18.42 | dongs | can this go in global. |
04:18.45 | cyberfab007 | Why is this so important , is this high security server or somthing ? |
04:18.45 | dongs | this has nothign todo with peer. |
04:19.15 | cyberfab007 | do you have 1000000 phone connectiing to this server and need to save resources? |
04:19.25 | cyberfab007 | or somthing |
04:19.34 | dongs | noidea, imguesingits some half-assed "lunix" solution and breaks at even slightest difference from what it expects. |
04:20.14 | cyberfab007 | so why bother? download asterisknow or elastix or somthing? |
04:20.22 | dongs | ... ? |
04:20.35 | dongs | i'mconnectingto provider's sip junk. |
04:20.46 | cyberfab007 | ohh use vitlity man |
04:21.00 | cyberfab007 | vitelity man they are best out there till i get my new site up ! |
04:21.15 | cyberfab007 | complete with elajoom LOL |
04:21.33 | dongs | ?????????????????????????????????????????????? |
04:22.04 | cyberfab007 | people are so worried about saving their half a cent with someone else , just pay good rate for good service with no hassels |
04:22.36 | dongs | looks like im grepping the sores after all. |
04:22.51 | cyberfab007 | yegh man |
04:22.56 | drmessano | half a cent is 33% in some cases |
04:23.12 | cyberfab007 | yegh if your doing 100,000 minuets |
04:23.16 | cyberfab007 | but your not |
04:23.24 | cyberfab007 | so worrie when you have the volume man , |
04:23.32 | cyberfab007 | not now |
04:23.51 | cyberfab007 | now you worrie on getting traffic |
04:23.55 | drmessano | Last time I checked, a half cent difference between one provider and the other was still a half cent at 1 minute or higher |
04:24.47 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
04:24.54 | cyberfab007 | ok , half a cent on 10000 minuets is 50 buckes man |
04:25.02 | cyberfab007 | are you doing 10,000 minuets |
04:25.20 | cyberfab007 | half a cent a minuet is 5 dollars on 1000 minuets |
04:25.47 | cyberfab007 | if you worrie bout that this early in the game your screwed trust me your worries need be else ware |
04:26.02 | *** join/#asterisk Tim_Toady (n=moi@adsl242-72.kln.forthnet.gr) |
04:26.08 | drmessano | So paying 30% more for anything is ok, because its stupid to worry about it? |
04:26.13 | dongs | why 1.6 doesn't print Event: registration? |
04:26.15 | cyberfab007 | no |
04:26.19 | dongs | in initial header |
04:26.26 | cyberfab007 | you have no pie , |
04:26.34 | cyberfab007 | 33% of no pie is nothing |
04:26.55 | cyberfab007 | when you have pie of 100,000 minuets a month than worrie about 33% till than forget it |
04:27.48 | drmessano | I'll tell that to my mother in law when she asks me why her phone cost her 60 a month instead of 40.. "Fuck you bitch, youre not doing 100,000 minutes here" |
04:28.01 | cyberfab007 | LOL |
04:28.16 | *** join/#asterisk grahamsaa (n=administ@cpe-74-67-180-93.rochester.res.rr.com) |
04:28.33 | cyberfab007 | I do like 750,000 a month , in long distance |
04:28.40 | cyberfab007 | minuts that is |
04:29.22 | cyberfab007 | maybe 60% is us the rest is international |
04:29.50 | cyberfab007 | it is about 4k a month in my pocket out of my canadian company |
04:29.55 | drmessano | I average about 5000 a month until around December.. then hit about 3,000,000 a month for December |
04:30.12 | drmessano | Santa gets a lot of calls |
04:30.20 | rizwank | Hi there. I'm looking for a way to have Asterisk take an incoming SIP call, and connect it to the PTSN (so the call effectively contains some extra data with a DID, and when Asterisk gets it, it completes the call via the PTSN.) Is there a name for this feature? |
04:30.30 | cyberfab007 | as I said get the pie first that worrie about how much your makign off it |
04:31.26 | drmessano | I thought your business was Elastix plugins |
04:31.29 | KyleK | rizwank: being a service provider? |
04:31.30 | manxpower | rizwank: Accepting a call and sending the call back out is a basic feature of Asterisk. I dunno about "extra data". |
04:32.15 | KyleK | rizwank: whats the application? are you trying to redirect rizwank@example.net to your cellphone? |
04:32.22 | rizwank | I effectively want to be able to deliver SIP calls via my land line. |
04:32.40 | rizwank | Not sure of the best way of doing that. |
04:32.42 | manxpower | rizwank: standard stuff. That's what a PBX does. |
04:32.45 | KyleK | "ip terminiation" |
04:32.51 | KyleK | brb |
04:32.53 | cyberfab007 | http://www.doingitwrong.com/wrong/20070515-003247.jpg |
04:33.36 | manxpower | rizwank: Asterisk is a toolkit for building a business phone system. It's not simple or easy. |
04:34.00 | rizwank | manx - wanted to make sure it was doable, and some terms for that feature set so I can do more research |
04:34.09 | drmessano | http://neoavatara.com/blog/wp-content/uploads/2009/04/l2_46730.jpg |
04:34.11 | cyberfab007 | no its not and anyone who has tried to write thier own dial plan will tell you that |
04:34.25 | rizwank | and I didn't know if Asterisk interfaced with analog modems or not. |
04:34.32 | manxpower | rizwank: It's such a basic features that it doesn't have a name. It's like asking for the name of the feature to describe a car moving. |
04:34.42 | cyberfab007 | LOL |
04:34.59 | manxpower | rizwank: it does not interface with analog modems. Access to analog phone lines requires a hardware card designed for that. |
04:35.01 | manxpower | ~boot |
04:35.02 | infobot | rumour has it, boot is what you get when you act like a DalNet user, or #debian-boot |
04:35.03 | manxpower | ~book |
04:35.04 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
04:35.07 | cyberfab007 | dude anlog modems is what we used before digium cards man |
04:35.14 | rizwank | =) |
04:35.15 | manxpower | rizwank: read the first part of the book. |
04:35.58 | rizwank | check. |
04:36.04 | rizwank | Thank you. |
04:36.18 | rizwank | trying to find out about delivering calling card calls via POTS |
04:36.33 | rizwank | so I wasn't show how the destination number was processed and dialed manually. |
04:36.38 | cyberfab007 | a2billing ma n |
04:36.41 | cyberfab007 | are you a noob |
04:36.41 | rizwank | didn't realize it was built in. |
04:36.44 | rizwank | yes. |
04:36.53 | cyberfab007 | it is like bread and water man |
04:37.02 | rizwank | we've got our own billing platform already... |
04:37.10 | rizwank | but I see your point =) |
04:37.17 | manxpower | The book will give yo a good inro |
04:37.44 | rizwank | thanks. FXO cards interface with POTS lines, so that's what we're looking at hardware wise... |
04:37.48 | rizwank | downloading it now. |
04:38.26 | manxpower | ~fxofxs |
04:38.26 | infobot | from memory, fxofxs is An FXO port (red Digium module) expects to receive dialtone and receive ring voltage. You can connect it to a PSTN line from the telco. An FXS port (green Digium module) expects to provide dialtone and provide ring voltage. You can connect a phone or a fax to it. |
04:39.10 | *** join/#asterisk jjshoe_ (i=jjshoe@75.85.173.90) |
04:39.11 | *** join/#asterisk d00gster (n=doughant@77.31.106.234) |
04:39.43 | cyberfab007 | Ok peoples , going to bed , good first nigh in the room here , wish I did not have to spend it arguing with Dr. Fag boy aka mulitple personality boy |
04:40.06 | cyberfab007 | see yall in the morning EST |
04:40.34 | jjshoe_ | if you're arguing with someone you're doing it wrong |
04:40.39 | jjshoe_ | use your client's ignore feature |
04:41.03 | manxpower | Joel: I wish my client had such a feature. Tonight was the first time in a while that I missed that feature. |
04:41.34 | cyberfab007 | jjshoe your right , but I give people much more often than I should the benifit of the doubt |
04:41.57 | cyberfab007 | everyone had their opinon and I shoudl deal with it |
04:42.03 | manxpower | But if nutjobs like cyberfab007 keep coming around I may just switch to a client that has that feature. |
04:42.42 | cyberfab007 | nutjobs sound like somthing your mom does |
04:43.15 | rizwank | thanks. |
04:44.40 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
04:44.59 | drmessano | Dr Fag Boy? Multiple Personality boy? |
04:45.18 | drmessano | Name calling.. you're a keeper |
04:45.41 | drmessano | I'll have you know, I have no personality and I don't smoke |
04:45.53 | drmessano | If you can't be smart, be accurate |
04:48.03 | cyberfab007 | member:iflux |
04:48.03 | cyberfab007 | cyber: yeah.. member:drmessano is actually 3 guys.. you never know which you're talking to at the time.. |
04:48.03 | cyberfab007 | member:ifluxx member:drmessano: are you bill, rich, or harry tonighy |
04:48.18 | cyberfab007 | Haha |
04:48.48 | drmessano | Apparently you're as dumb as I figured |
04:48.59 | cyberfab007 | so are you bill rich or harry tonight , |
04:49.03 | drmessano | iflux is my cousin |
04:49.18 | cyberfab007 | Yegh waiting for that confirm |
04:49.25 | drmessano | [23:30] <iflux> think I had him going |
04:49.30 | drmessano | :( |
04:49.42 | drmessano | Dont believe everything you read on IRC, n00b |
04:51.09 | cyberfab007 | well what ever , |
04:51.13 | drmessano | Time to hop in the Bentley and run to Taco Bell.. I will grab a Chalupa and throw it at your window on the way by, cyberfab007.. Still the 3rd basement window on the left, right behind your moms car? |
04:51.24 | cyberfab007 | you guys have a good night , |
04:51.44 | cyberfab007 | ha I live in a million dollars condo , with a view of the lake |
04:51.46 | drmessano | You too sweetheart |
04:52.08 | cyberfab007 | 53 story |
04:53.30 | *** join/#asterisk Zuchmir2 (n=Zuchmir-@ool-18bd3bfc.dyn.optonline.net) |
04:54.14 | cyberfab007 | well 53 floor 47 stories |
04:55.26 | drmessano | Palace Pier? |
04:55.29 | *** join/#asterisk frk2 (n=faraz@zivios/member/fkhan) |
04:55.50 | Joel | manxpower it helps if people arn't retarded and would just simply not engage the idiots |
04:56.12 | cyberfab007 | Montage |
04:56.17 | cyberfab007 | City place |
04:56.36 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-us/x-rahfpmsqbfuitiif) |
04:56.42 | cyberfab007 | all bought and paid for by Asterisk ! |
04:57.12 | cyberfab007 | http://www.cityplace.ca/montage_le/ |
04:58.27 | cyberfab007 | I have the south west penthouse , 2000 squear feet of niceness |
04:59.08 | Zuchmir2 | which ATA is descent that handles multiple SIP accounts (ideally also multiple FXS ports)? |
04:59.19 | drmessano | Too bad about the real estate market I guess |
04:59.55 | drmessano | All those listings for condos at the Montage for 200k - 500k |
05:00.09 | drmessano | Yawn |
05:00.15 | drmessano | Google harder |
05:01.27 | cyberfab007 | Yegh live in your trailer |
05:01.36 | cyberfab007 | goodnight yall |
05:02.43 | *** join/#asterisk TimRiker (n=timr@bzflag/projectlead/TimRiker) |
05:03.57 | cyberfab007 | I finally ignor that guy |
05:04.13 | cyberfab007 | should have done it sonner but had a few frinks tongiht |
05:04.15 | drmessano | scratches his crotch, burps, takes a swig of bud light and slaps the wall of his trailer.. |
05:04.51 | cyberfab007 | drinks , totaly distracted me from what I came her for tonight what a ass |
05:05.11 | Tim_Toady | lol, wtf |
05:05.37 | drmessano | Tim_Toady: You can code.. cyberfab007 can hook you up |
05:06.15 | cyberfab007 | Tim_Today ever have someone like that ? I dont chat much in IRN |
05:06.17 | cyberfab007 | IRC |
05:06.44 | Tim_Toady | like u? nah |
05:06.59 | drmessano | frinks his fanta |
05:07.16 | drmessano | sells that ad campaign |
05:07.46 | drmessano | My goal in life is to pressure coke into making Watermelon Fanta |
05:08.32 | drmessano | Then move next door to cyberfab007 |
05:08.42 | Tim_Toady | lol |
05:08.42 | drmessano | So we can be asterisk buddies |
05:08.46 | dongs | ok why am i getting ;rport added at end of Via: |
05:09.23 | dongs | nat=never |
05:09.32 | *** join/#asterisk maour (n=gnu@unaffiliated/maour) |
05:09.35 | drmessano | I honestly have no idea why Windows comes with any disk utilities at all |
05:09.41 | drmessano | They dont work |
05:09.48 | dongs | no, you just don';t know how to use htem |
05:09.51 | cyberfab007 | Tim what you mean like me , I let that guy totally drag me in , what a drunk Iwas tonight to take me off topic |
05:09.55 | drmessano | lol |
05:10.09 | drmessano | I'm well aware how to use them |
05:12.27 | dongs | why in the hell is default asterisk build option -g3??? |
05:12.52 | drmessano | M$ needs to hire the guy who wrote volrepair for netware and put him in charge of chkdsk |
05:12.54 | dongs | do you enjoy linking 40megs ofdebug info into 4 megs binaryor something? |
05:14.03 | dongs | anyone? |
05:14.15 | dongs | manxpower: what is the reason for default compile option for asterisk being -O6 -g3? |
05:14.32 | dongs | especially in tarballs designed for end-users? |
05:14.53 | KyleK | which tarball is this? |
05:15.00 | dongs | anything off asterisk.org. |
05:15.37 | cyberfab007 | http://support.sas.com/documentation/onlinedoc/sasc/doc/changes/z0090339.htm |
05:15.49 | dongs | what. |
05:16.44 | dongs | LOL got it to work after removing ;rport= shit |
05:17.06 | cyberfab007 | fucking with you lol |
05:17.13 | KyleK | hrm |
05:17.38 | cyberfab007 | it is old IBM mainframe options , confuses the shit out of people but it drove the answer out of ya hugh? |
05:18.01 | drmessano | He called you Hugh, frank |
05:18.04 | dongs | funny how someuseless option has made it in by default, justto support "buggy firmware" on some random uniden phone nobody uses. |
05:20.18 | dongs | cyberfab007: i'm not sure what your purpose in this chat is. |
05:20.33 | dongs | so far you've been nothing but an annoying self-righteous stuck up piece of opensores shit. |
05:20.43 | dongs | (but thats what a typical opensores chat is filled with, so im hardly surprised). |
05:21.44 | dongs | meanwhile im waiting for 'ld' to figure out how to link 40MiB*30 .o files linked with -g3 into a 40meg asterisk.bin because apparently turning off -g3 for release builds shipped to customers (lol, customers in opensores, rite) was too hard. |
05:22.09 | *** join/#asterisk miloux (n=KVIrc@milu.rit.se) |
05:23.05 | Tim_Toady | dongs ure compiling on a 486 or smth? |
05:23.11 | dongs | nope. |
05:25.10 | dongs | k blank ;rport= washte problem that was confusing the remote side. |
05:25.13 | cyberfab007 | dongs that sound like some complicated stuff , I am not familiar with to be honest |
05:26.05 | KyleK | hrm -g3 -O6 -O0 |
05:26.12 | KyleK | yea that is odd |
05:26.15 | drmessano | He's a coder, but doesn't understand compiler options.. move along |
05:27.19 | cyberfab007 | cyberfab007 is a bit offended by the use of "opensourses shit" |
05:27.51 | drmessano | or "opensores".. which ruins his allusion |
05:32.32 | *** join/#asterisk oej (n=olle@ns.webway.se) |
05:37.07 | Zuchmir2 | which ATA is descent that handles multiple SIP accounts (ideally also multiple FXS ports)? |
05:37.19 | drmessano | How many ports? |
05:37.26 | Zuchmir2 | 2 |
05:37.39 | drmessano | Linksys PAP2-T |
05:38.22 | Zuchmir2 | does that support multi accounts per FXS? |
05:38.48 | drmessano | That supports 1 account per each FXS.. it has 2 |
05:39.20 | [TK]D-Fender | Checkout time, later all |
05:39.42 | drmessano | I dont think you'll find a device that supports multiple accounts per port |
05:39.49 | Zuchmir2 | ok, i need 2 SIP accounts, 1 for incoming, 1 for outgoing, and both use the same 1 acct for out |
05:40.14 | drmessano | if youre using asterisk, you can do this in the dialplan |
05:40.23 | Zuchmir2 | i thought i read somewhere there's a device with multi accounts |
05:40.46 | drmessano | ATA's are not meant to be that smart.. make it work in Asterisk, which is easy |
05:40.54 | Zuchmir2 | i know the Prestige 2302R can do it |
05:41.18 | *** join/#asterisk lmsteffan (n=laurent@reef.ac-noumea.nc) |
05:41.21 | Zuchmir2 | but am having trouble with it |
05:41.21 | drmessano | Im guessing youre not even using asterisk |
05:41.51 | Zuchmir2 | i have an asterisk server, but it keeps losing reg, and CID doesnt display name |
05:42.40 | hesco | I'm writing a REST application which interacts with Asterisk. sometimes its methods are called by apache, sometimes by asterisk through agi scripts embedded in the dial plan. Asterisk is running as root. It seems completely incapable of unlinking call files from the staging queue putting a copy of them into the outbound spool. Can't fugure out why, though. |
05:45.11 | Zuchmir2 | hmm, the PAP2-T has GPL code, does that mean i would be able to mod this thing to add multi ports (or is this only partial GPL) |
05:46.18 | hesco | ps aux says asterisk is running as root. |
05:46.27 | drmessano | No, you can't mod it in that way |
05:46.43 | *** join/#asterisk thansen (n=thansen@76.27.110.194) |
05:46.54 | hesco | I assume that means that the agi scripts called from the dialplan will run as root as well. Does that make sense? |
05:47.27 | cyberfab007 | well I am gonna shoot some people on xbox live before I go to bed, I have lots of meetings tomorrow. Let me leave you people with one thing. OpenSource is our existence, never forget that almost like a god father was stallman. You have the dreamers and workers. Which are you? Which ever you are , we need both, as success-full as linux has become there are much larger things in the future. Asterisk could be |
05:47.28 | cyberfab007 | Engine which drives the future , I have been working on this design since I was about 17 years old about the time asterisk came around, an asterisk box in every living room that is free, opensource, user owned with encrypted data, has really been the goal since the beginning, Well imagine asterisk doing more than that, running AI in households via voice recognition, social networking, mass private telecom |
05:47.28 | cyberfab007 | networks over a neutral net, Self government. All of this requires the flexibility of the advanced communications technology that asterisk offer. For christ sake you can run advanced c and php scripts right from the asterisk dial plan those commands can be determined by voice recognition which can lead way to a whole variety of smart-house technology and make it an appliance in every home. Self government, h |
05:47.33 | cyberfab007 | Asterisk can already provide advanced communications , but can we make that communications smart? Combine it with social networking technology? Let people governthem selfs with the help of communicaitnos technology? Well my investors expect this of my company... |
05:47.41 | KyleK | omg spam |
05:47.46 | *** join/#asterisk |Cybex| (n=John@atwork-26.r-212.178.82.atwork.nl) |
05:47.46 | cyberfab007 | nope |
05:47.49 | cyberfab007 | worse |
05:48.00 | cyberfab007 | drunken ramble , from the heart ! |
05:48.10 | cyberfab007 | xbox time , you people think about that |
05:48.27 | *** join/#asterisk ming_zym (n=ming_zym@124.127.101.0) |
05:49.26 | cyberfab007 | cool link |
05:49.38 | drmessano | http://www.xbox360updates.com/uploaded_images/Xbox-360-Updates-729048.jpg <--- One can only hope |
05:49.44 | cyberfab007 | http://areyouadavinci.com/ |
05:49.53 | cyberfab007 | I bet 9/10 of you are |
05:51.57 | drmessano | I scored 100%.. I smell shenanigans.. I've always been a C+ or B- kinda guy |
05:52.20 | cyberfab007 | a stunning silence |
05:53.29 | *** join/#asterisk shinao1 (n=shinao1@41.219.193.248) |
05:53.47 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
05:54.15 | drmessano | Hmm.. the Davinci Method.. But if I already have the Davinci Code.. do I need this? |
05:54.19 | drmessano | Doubtful |
05:54.33 | KyleK | draw lots of pretty pictures? |
05:55.15 | drmessano | I'm gonna write a book called the Davinci Cheat Code |
05:56.02 | drmessano | U, U, D, D, L, R, L, R, B, A, B, A, Start <--- The secret to success in life |
05:57.08 | dongs | oh you're kidding. |
05:57.18 | dongs | did make clean just wipe my menuconfig settings. |
05:57.38 | dongs | what the fuck people. clean removes OBJECT CODE ONLY. |
05:57.44 | dongs | distclean = including configured settings. |
05:58.32 | dongs | lunix. its all about surprises. you don't know what's gonna fuck you over next. |
06:03.12 | KyleK | post some bugs |
06:03.18 | dongs | no point. |
06:03.25 | dongs | nobody reads bug reports. |
06:03.32 | dongs | it took them over 6 years to implement sip session timers. |
06:04.18 | dongs | infact ifi remember correctly (6 years and all), the bug i opened on that was closed with "wont fix, nobody fucking uses that shit". |
06:05.03 | drmessano | ebay is selling skype |
06:05.07 | drmessano | Yay |
06:05.14 | KyleK | how much |
06:05.48 | drmessano | Not officially announced yet, will be today |
06:06.12 | dongs | good riddance. |
06:06.55 | drmessano | Riddance? |
06:08.01 | drmessano | Skype will be much better off without ebay |
06:08.37 | KyleK | hopefully they sell it for less than 4 billion |
06:08.53 | KyleK | I'll be pissed off if marvel is worth less than skype :) |
06:08.57 | drmessano | Ebay has been shopping it out at 2 billion |
06:10.06 | drmessano | Who knows.. Guess I need to wait for Scott Fulton to proclaim Skype is dead over on Betanews around lunchtime |
06:10.31 | drmessano | Until then, i am out.. GOODBYE KIND PEOPLE |
06:10.39 | *** join/#asterisk TimToady_ (n=moi@77.49.29.202) |
06:10.41 | drmessano | FIGHT THE GOOD FIGHT |
06:11.01 | drmessano | I R NOT DRUNK BUT TEH OPEN SOURCE IS TEH OPEN DOMOCRASY |
06:11.19 | drmessano | U R SIP, I R SIP.. WE R ALL SIPS |
06:11.27 | drmessano | I SIP U, U SIP ME |
06:11.52 | KyleK | :) |
06:12.01 | TimToady_ | i missed something good eh? |
06:12.06 | KyleK | I just ignore the channel when teh crazies re in |
06:12.31 | drmessano | 01:47 |
06:12.34 | drmessano | ET |
06:12.45 | drmessano | Check the log on riker's site |
06:12.47 | drmessano | Good stuff |
06:12.52 | drmessano | Anyway |
06:13.10 | drmessano | SIP AWAY MAH GUD PEOPLE.. COMMUNISTS USE THE H323! |
06:13.21 | cyberfab007 | crazies are we , what do you do ? what is your life style like? |
06:13.22 | drmessano | DONUT BE COMUNISM |
06:13.46 | *** join/#asterisk grEvenX (n=even@apb99b.ip.ssc.net) |
06:14.17 | drmessano | (Hes all yours KyleK.. Just make sure if you cant get the hook from his mouth, you keep smashing his head with the stick.. he'll let go) |
06:14.17 | cyberfab007 | I live sex, drugs , music , code, engineering , lushness, I live this everyday |
06:14.42 | cyberfab007 | am I crazy , most say yes , am I sucessfull, most to to much , |
06:14.42 | drmessano | WUT R COMPILER FLAGS? |
06:14.50 | drmessano | \\\\GONE |
06:15.51 | cyberfab007 | I thought University was a joke , I was to busy making money I got by with my C |
06:20.11 | cyberfab007 | crazys hugh ? crazyones.org |
06:20.33 | *** join/#asterisk xrmx__ (n=rm@host197-226-dynamic.1-79-r.retail.telecomitalia.it) |
06:37.46 | hesco | ps aux says asterisk is running as root. |
06:37.47 | hesco | I assume that means that the agi scripts called from the dialplan will run as root as well. Does that make sense? |
06:38.07 | hesco | $> and $< seem to indicate that is the case. |
06:38.36 | hesco | However, my AGI script seems unable to remove a file in the queue. |
06:38.46 | hesco | Any ideas why that might be? |
06:39.50 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
06:42.37 | KyleK | whats the error its getting? |
06:43.49 | hesco | I'm not seeing any error messages, only a deleted file persisting in the pending queue directory |
06:44.06 | j_kroon | https://issues.asterisk.org/view.php?id=14577 - what's my options to get a fix for this? |
06:44.06 | KyleK | what lang is the agi |
06:44.33 | hesco | perl |
06:47.42 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
06:48.41 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
06:49.05 | hesco | I've been using print $FH to build the call file. File::Util->write_file to copy it over to the spool; and unlink to clean up behind myself. but the unlink has not been cooperating. |
06:49.58 | hesco | I just tried rename $queue/$call_file, $spool/$call_file. and that did not seem to work either. |
06:52.37 | KyleK | why File::Util->write_file instead of open write close? |
06:54.07 | KyleK | well i guess open; print $handle; close |
06:55.30 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
06:57.47 | KyleK | for rename is $queue on the same drive as $spool? |
06:58.03 | KyleK | well filesystem actually |
06:58.10 | hesco | same drive, different partition |
06:58.26 | hesco | latest attempt is: rename "$queue/$userid/$call_file", $spool/$call_file; |
06:58.34 | KyleK | move the queue |
06:59.02 | hesco | perl's `mv` is rename |
06:59.55 | KyleK | move is like this |
07:00.08 | Docteh | unless its on a different FS |
07:00.11 | Docteh | then it does this |
07:00.59 | *** join/#asterisk KyleK (n=Kyle@allspark.shadowmage.org) |
07:01.09 | KyleK | :) |
07:03.36 | *** join/#asterisk icyValk77 (n=icyValk7@gateway.ash.thebunker.net) |
07:04.23 | KyleK | but anyways if thats a callfile and you want it to work properly the queue and the spool directory have to be on the same file system |
07:04.39 | KyleK | or you'll want to copy to spool/.. and then rename it |
07:05.33 | hesco | the queue is at /tmp/app_name/queue/userid/ and the spool is at /var/spool/asterisk/outbound/ |
07:06.28 | *** join/#asterisk crazybyte (n=crzp@unaffiliated/crazypenguin/x-000001) |
07:06.52 | KyleK | is that the same filesystem? |
07:09.34 | hesco | yes, though distinct partitions |
07:10.22 | hesco | my /tmp is on /dev/hda2; while my /var is at: /dev/hda6 |
07:10.28 | KyleK | uhh |
07:10.53 | KyleK | filesystems sit in partitions which sit in drives |
07:11.16 | hesco | ok, so I guess they are different file systems |
07:11.31 | hesco | I though a single filesystem was rooted at /. |
07:11.36 | hesco | thought |
07:11.40 | *** join/#asterisk cjk (n=cjk@vodsl-9494.vo.lu) |
07:12.01 | KyleK | well in windows each drive is a file system |
07:12.21 | hesco | fstab says they are all reiserfs |
07:12.22 | KyleK | but you CAN mount into a drive such that C:\windows is E: |
07:12.30 | cjk | hi, does anyone know an app for nokia phones that is able to call a http url from the contacts. in order to develop click2dial for nokia |
07:12.38 | KyleK | hesco: so they're both like ntfs |
07:12.50 | hesco | how so? |
07:13.05 | KyleK | I'm throwing windows terminology at you |
07:13.17 | KyleK | I figure it might help you get it :) |
07:13.56 | hesco | reiserfs is an open source journaled filesystem, ntfs is proprietary so no one really knows what it is, or at least that is what I've been told. |
07:14.07 | KyleK | well yea |
07:14.22 | KyleK | well what are you better at |
07:14.26 | KyleK | Windows or Linux? |
07:14.27 | hesco | windows terminology is most likely to only confuse me |
07:14.42 | hesco | haven't owned a windows box since 2002 or so |
07:15.15 | hesco | though I launched a windows instance in AWS' EC2 last night to test an install of an app I have to support the end of this week |
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07:15.50 | KyleK | alrighty so i wont try throwing windows terminology at you |
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07:16.07 | hesco | I'm pretty easily confused at this hour as it is |
07:17.00 | hesco | so do you think this might work better if I moved my queue to the partition with /var on it? |
07:17.06 | KyleK | yea |
07:17.19 | hesco | ok, I'll try that then |
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07:28.20 | hesco | ok, now it is really broken. And I'm feeling the need for sleep. |
07:29.05 | KyleK | hopefully its not in production then ;) |
07:31.16 | hesco | no, on my test bed. |
07:31.45 | hesco | I just did an svn revert on the broken module and restored it to its previous state of brokenness |
07:32.21 | hesco | so it now generates the call, but fails to remove the call file from the queue. |
07:33.34 | hesco | I guess less broekn is better. |
07:33.38 | hesco | broken |
07:34.00 | hesco | ok, enough then, to bed with me. |
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08:54.52 | tzafrir_laptop | ZX81, this starfish-pbx appears interesting at first glance. Sadly they use a non OSI-compliant license for their software |
08:55.11 | ZX81 | yeah saw that |
08:55.36 | synthetic | sudo apt-cahce search headers |
08:55.36 | ZX81 | they say it's Open Source, but after reading the license I'd say it's pretty much a no-go |
08:55.44 | synthetic | err oops |
08:55.48 | ZX81 | :D |
08:55.56 | tzafrir_laptop | synthetic, apt-cache can be run as non-root |
08:55.57 | synthetic | damn pidgin doesnt scroll |
08:56.12 | synthetic | thought iw as helping someone |
08:56.19 | synthetic | my typing doesnt help people |
08:57.16 | ZX81 | man I wish Adium wouldn't read out the friggin time of a message if I'm online - I just want it to read me the text of messages so I can watch TV and be contactable at the same time :D |
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08:59.48 | ZX81 | I've update the news article to point out non compliance just in case people don't check first |
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09:08.08 | Gugge | anyone know if a feature like this one was ever integrated, https://issues.asterisk.org/view.php?id=7771 |
09:08.54 | Gugge | the patch seems to apply cleanly, and using SIPCHANINFO(ruri,field) works fine ... but i would rather use a unpatched asterisk :) |
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09:17.20 | KyleK | ~mysql |
09:17.21 | infobot | SQL (Structured Query Language) database server. URL: http://www.mysql.com/ |
09:17.27 | KyleK | ~cdrmysql |
09:17.36 | KyleK | eh i'll ask tomorrow |
09:17.49 | ZX81 | ~areski |
09:17.55 | ZX81 | ~cdrstats |
09:18.00 | ZX81 | meh |
09:18.01 | ZX81 | :) |
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09:29.55 | KyleK | oh its in addons |
09:30.14 | ZX81 | yeps |
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10:29.30 | garymc | Hi anyone know how to setup the polycom ip330? |
10:29.46 | garymc | just got 2 of them, one seems to have booted itself? and another is just sitting there |
10:30.24 | kaldemar | mine don't boot themselves unless i tell them to. |
10:30.48 | garymc | yeah weird hey |
10:31.02 | garymc | so i get the phone no instructions like default passwords etc |
10:32.48 | kaldemar | Polycom:456 iirc |
10:33.35 | kaldemar | http://www.polycom.com/support/voice/soundpoint_ip/soundpoint_ip330_320.html |
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10:37.35 | Valen | Heres a question, I have a number of users and they often play "musical desks" I was wondering if there was a "good" way for them to be able to login to a phone such that their extension number will follow them around the system? |
10:44.47 | garymc | kalemar: I cant seem to find the default passwords etc in those guides you gave me the link too |
10:47.29 | Sandheaver | garymc: the default passwd for the phone is 456. If you config via the phone's web interface, the user is Polycom (with a captial P) and the password is 456 |
10:47.38 | Sandheaver | that stuff is in the docs |
10:47.47 | kaldemar | you didn't do much lookin then. it's in the admin guide, and i told the default username and password a few lines above already. |
10:47.59 | garymc | well i didnt get any docs with it, only putting the parts together doc |
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10:48.13 | garymc | right ok i didnt know what that was |
10:48.15 | garymc | sorry |
10:48.30 | kaldemar | i have you the link to the guides. use them. |
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11:07.55 | garymc | i have to say the docs are not very clear :( |
11:08.02 | garymc | not to me anyways |
11:08.39 | garymc | also as soon as I go to dial extension 1000, the phone only lets me dial 10 and says the number is not in use |
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11:16.57 | AndyML | so, with our old nortel we recieved the CALLERID(name) from the telco, but now it never seems to make it into the system. Here are my dahdi configs. http://pastie.org/601524 - i even tried sendcalleridafter=1 but that seems to be for outbound. |
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11:36.21 | garymc | when im just setting up 2 polycom phones to test, so i can call extension to extension? Do i need to do this bootfile stuff? |
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11:40.21 | dongs | so uh |
11:40.22 | kaldemar | extensions and boot files have nothing to do with each other |
11:41.49 | garymc | right, well ive setup the phones, they got an extension name on them, but they are not registering with freepbx (which im asking about in #freepbx) also the time is just flashing, cant seem to get that to work |
11:42.29 | garymc | so im not sure if its my asterisk/freepbx setup not letting them work or the phones |
11:42.30 | AndyML | i've added a Wait(1) before it processes the call and now I'm getting the name in asterisk but it isn't getting sent properly to another device connected via PRI. |
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11:47.15 | Naikrovek | garymc: you have to tell the phones themselves which extension they are, and that's done via configuration file |
11:47.25 | Naikrovek | or via the web interface |
11:47.31 | garymc | yeah i done that |
11:47.52 | garymc | theres that many diff options though, not sure ive done all i need too |
11:48.15 | Naikrovek | also, you have to set up the Digitmap, so it knows what kind of dial pattern you're going to give it. |
11:48.35 | Naikrovek | if it's completing the call before you've finished dialing, that's the digitmap |
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11:49.07 | garymc | where can i find an explanation on that? |
11:49.13 | dongs | i got provider sip -> asterisk -> sip/pstn adapter x2 -> analog to ISDN converter, and connecting ISDN out from that to a keysystem. In 1.4, caller ID worked fine -> cid from provider was passed over to voip adapter, which in turn passed to ISDN. but now for some reason all callerid-aware devices I tried show the phone # of voip adapter, regardless of who's calling. what cahnged? |
11:49.18 | garymc | I can see my digitmap |
11:49.24 | Naikrovek | garymc: just use mine: 0T|011xxx.T|xxxxxxxxT|[2-9]xxxxxxxxxxT|[1-9]xxxxT|xxxT |
11:49.43 | Naikrovek | i need to fix mine, actually |
11:49.45 | Naikrovek | but that works |
11:49.45 | dongs | i tried doing the obvious of like Set(CALLERID(num)) or whatever, no change. |
11:49.54 | dongs | ${CALLERID(num)} shows correct number. |
11:50.52 | Naikrovek | dongs: then use that? I'm not really familiar with caller id |
11:51.10 | dongs | i am, its not passing it along to the sip/pstn adapter like it used to. |
11:51.15 | dongs | or it does, but wrong way from 1.4. |
11:52.03 | Naikrovek | you're using 1.6 now? |
11:52.05 | dongs | yeah. |
11:52.11 | Naikrovek | things are different in 1.6 |
11:52.17 | leifmadsen | Naikrovek: how so? :) |
11:52.22 | dongs | different to the pointof breaking? |
11:52.36 | Naikrovek | yes, dongs. that's why it's a different version number |
11:52.44 | Naikrovek | why do people never see that coming |
11:52.55 | dongs | um |
11:53.00 | dongs | caller ID is a basic function. |
11:53.09 | garymc | Naikrovek: giving that digitmap a try |
11:53.13 | Naikrovek | yes, but how it is handled changed from 1.4 to 1.6 |
11:53.23 | leifmadsen | dongs: you'd have to provide some sort of console output and relevant configuration output in a pastebin in order for people to give you anything other than vague answers |
11:53.43 | leifmadsen | Naikrovek: in which way? |
11:53.54 | Naikrovek | leifmadsen: in lots of ways, you know this |
11:54.10 | garymc | Naikrovec: that digitmap only lets me dial a single digit |
11:54.25 | Naikrovek | garymc: then google a better one. it works perfectly for me |
11:54.29 | Naikrovek | and i'm using a polycom phone too |
11:54.40 | leifmadsen | Naikrovek: I don't know this -- I'm curious to know what changed because I've obviously missed that part |
11:54.42 | Naikrovek | and you ahve to keep dialing or it'll assume you're done |
11:54.56 | leifmadsen | (in terms of CallerID specifically) |
11:55.40 | Naikrovek | leifmadsen: i don't know specifically, but people come in here every day with callerid issues and they have all upgraded from 1.4 to 1.6. I use trixbox so i'm not much help with that |
11:55.43 | dongs | well, i see what the problem is. |
11:55.44 | dongs | From: "realnumberthatscalling" <sip:siptopstnadapternumber@lanip>;tag=as1c8d2f42 |
11:55.55 | dongs | of course that'll never work. |
11:56.03 | leifmadsen | SIP? |
11:56.07 | dongs | yes. |
11:56.14 | leifmadsen | don't rely on the From headers -- use RPID |
11:56.25 | leifmadsen | trustrpid=yes, sendrpid=yes |
11:56.25 | dongs | sorry, tell that to the device manufacturer. |
11:56.34 | leifmadsen | get a better device manufacturer |
11:56.38 | dongs | nice try. |
11:56.39 | Naikrovek | dongs: get some things up on pastebin and wait for [tk]d-fender to show up. the VERY FIRST question he'll ask is for some pastebin links. he is smart and can help you |
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11:57.32 | leifmadsen | give a snarky answer, get one back |
11:59.10 | dongs | leifmadsen: i bet your answer is same reason broken code for adding a improperly-formatted ;rport= atthe end of every SIP invite was kept in release version of code intended for end-users, right? |
11:59.36 | dongs | 'work around bugs in <some uniden softphone nobody ever heard of' |
11:59.43 | leifmadsen | I'm just telling you what works for me on every device I've used -- I've not used every single device |
11:59.52 | dongs | (but break everything else meanwhile, i mean, nobody cares right. |
12:01.49 | Naikrovek | dongs: you're the one that upgraded without testing. i'm not saying that things should have changed, but you can't just upgrade and assume everything will continue to work exactly as before |
12:02.06 | Naikrovek | features come and go, some things get deprecated |
12:02.15 | dongs | i didnt upgrade anything. i retired the box and rewrote configurations implementing changes needed for 1.6 |
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12:02.41 | Naikrovek | there was a guy in here the other day complaining that 1.6 remove the ability to use | as a parameter separator in extensions.conf. he had to change everything |
12:03.13 | verywiseman | what hardware i need to run openbts with asterisk? |
12:03.21 | Naikrovek | idles |
12:06.13 | garymc | anyone tell me why my phones are not registering? |
12:08.09 | garymc | I check my phone status, it says not registetred. I look in Freepbx it says no phones Reged! HELP! |
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12:09.07 | leifmadsen | garymc: what information have you given that doesn't require wild guesses? |
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12:09.35 | garymc | ?? |
12:09.57 | leifmadsen | garymc: console output, sip debug, relevant configurations? |
12:10.25 | garymc | im using FREEPBX, not sure how to get thoses things |
12:10.43 | leifmadsen | It's like me telling my mechanic that my car isn't working, and not telling him what part is causing me issues |
12:10.58 | kaldemar | garymc: go ask in #freepbx then. |
12:11.00 | leifmadsen | garymc: ahh -- you may want to try #freepbx then |
12:11.08 | garymc | im in freepbx too |
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12:11.47 | garymc | must have to wait a couple of hours for people to come in or something |
12:11.47 | leifmadsen | most people here just use vanilla asterisk, thus can't help you with your GUI based system |
12:11.51 | garymc | Maybe youd know wy im getting Cronmanager encounted 1 errors |
12:11.54 | leifmadsen | if you can ssh into the box, you can get into the asterisk console via 'asterisk -r' and then run various console based commands to get output |
12:12.01 | garymc | Could not reload FOP server |
12:12.06 | leifmadsen | has no idea |
12:12.11 | garymc | ok |
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12:17.42 | leifmadsen | [TK]D-Fender: I don't want to meet your Mom! I just want... |
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12:19.31 | verywiseman | what hardware i need to run openbts with asterisk? |
12:19.34 | [TK]D-Fender | leifmadsen: ! ! ! |
12:19.41 | leifmadsen | \o/ |
12:20.08 | leifmadsen | verywiseman: not sure -- the openbts website seems to have the information for getting it configured -- I had never even heard of it till this morning |
12:20.18 | leifmadsen | there was even a youtube video |
12:20.51 | Chainsaw | verywiseman: Mostly, you need a spectrum license or a dummy load. |
12:21.50 | [TK]D-Fender | verywiseman: http://www.kestrelsp.com/OpenBTS.html <- go ask them |
12:22.13 | AndyML | has anyone solved the hum/buzz issue with polycom ip phones and plantronics headsets? |
12:22.40 | EiNSTeiN_ | verywiseman: I suggest you watch the talks that Harald Welte gave at 25C3 and HAR2009 |
12:22.48 | [TK]D-Fender | AndyML: I've seen this when your power supply is flakey |
12:22.54 | [TK]D-Fender | AndyML: NO OTHER TIME |
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12:23.59 | AndyML | [TK]D-Fender: so like, PoE probably solves it out of hand? I'm seeing it on every brand new IP 331 with new Polycom power supplies right out of the box all over this company (~45 phones) |
12:24.45 | [TK]D-Fender | AndyML: Well I don't know if you've been messing with gains, or if you're running an amp if you have it on the right position, etc |
12:25.21 | [TK]D-Fender | anyRunning 45 polycom phones on bricks instead of PoE was definitly a mistake :) |
12:25.44 | dongs | allright |
12:25.50 | dongs | lemmereblog my question for [TK]D-Fender |
12:26.02 | dongs | i got provider sip -> asterisk -> sip/pstn adapter x2 -> analog to ISDN converter, and connecting ISDN out from that to a keysystem. In 1.4, caller ID worked fine -> cid from provider was passed over to voip adapter, which in turn passed to ISDN. but now for some reason all callerid-aware devices I tried show the phone # of voip adapter, regardless of who's calling. what changed? |
12:26.44 | dongs | obviously i'd like the phone number of hte caller from incoming sip connection to be passed to pstn/isdn. |
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12:27.10 | AndyML | [TK]D-Fender: I've moved the headset tx gains down 3db to no effect, and we're using headsets plugged right into the polycom's, using their internal amplifier. the customer was given the option of replacing all their switches with PoE which they turned down. They don't have enough network drops to simply add-on. (Every phone's switch port is in use.) |
12:27.12 | dongs | ${CALLERID(num)} shows correctthing. |
12:27.26 | [TK]D-Fender | dongs: pastebin the complete call attempt with SIP debug and include your sip.conf |
12:27.51 | [TK]D-Fender | AndyML: they could replace their core switch |
12:27.52 | dongs | or you can just tell me what to look for , since i'm not spending 30 minutes cutting out my provider call info out of it |
12:28.04 | dongs | that'll save YOU 30 minutes oflooking through stuff. |
12:28.38 | AndyML | [TK]D-Fender: agreed. they decided against it. I will bring it up as a solution to this problem and ask again if they're willing to replace it. |
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12:29.35 | [TK]D-Fender | AndyML: I'd test it first. Isolate a phone as well and put it behind a UPS and see if the power conditioner clears up the hum |
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12:30.53 | dongs | [TK]D-Fender: so, what changed in cid handling that this is happening? |
12:31.34 | [TK]D-Fender | dongs: Nothing I can think of, and I'm not going to start randomly pointing fingers. Show the debug & configs |
12:32.02 | dongs | i insist neither are relevant to answer the question. |
12:32.19 | [TK]D-Fender | dongs: Bullshit. They guy who doesn't look doesn't find the problem. |
12:32.35 | dongs | [TK]D-Fender: i'mstill waiting for your advice on exactly WHAT to look for. |
12:32.49 | [TK]D-Fender | dongs: Insist all you want, I've worked with asterisk from before it hit 1.0 |
12:33.09 | dongs | [TK]D-Fender: glad to see in 2009 they finally added sip session timers. |
12:33.14 | [TK]D-Fender | [08:27]<[TK]D-Fender>dongs: pastebin the complete call attempt with SIP debug and include your sip.conf <- I already told you |
12:33.17 | dongs | you might remember me asking for htem in 2001. |
12:34.05 | [TK]D-Fender | brb |
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12:43.08 | dongs | lol. it was 'fromuser'. |
12:43.39 | *** join/#asterisk [TK]D-Fender (n=chatzill@216.191.106.162) |
12:43.43 | dongs | what fucking ridiculous documentation. first, change username to 2 possible (and insane) choices. then make it totally fucking unclear WHY that was done, and leave no explanation to use one or another. |
12:43.54 | dongs | [TK]D-Fender: problem was 'fromuser' in sip/pstn device config. |
12:44.29 | [TK]D-Fender | dongs: Yes, that will force the CID to that of fromuser unless you use sendrpid=yes and trustrpid=yes and they support those headers |
12:44.43 | [TK]D-Fender | dongs: Which I'd have pointed out the moment I'd have seen it |
12:44.46 | dongs | [TK]D-Fender: how about adding that to the documentation. |
12:44.59 | dongs | so that you wouldnt have to deal with "dumb" questions like these. |
12:45.09 | [TK]D-Fender | dongs: Cool it already |
12:45.09 | dongs | ;fromuser=yourusername ; Many SIP providers require this! |
12:45.12 | dongs | this is super useful! |
12:45.32 | [TK]D-Fender | dongs: Yes, and many providers also don't allow you to set your callerid. Should we document this as well? |
12:45.38 | dongs | i mean anyone converting old username= crap to newstuff wouldjustgo and change this, cuz u know?? most providers requirethis, let's add it and thenforget what it was for. |
12:45.43 | dongs | huh. |
12:45.57 | dongs | this is callerID from asterisk to sip/pstn adapter. |
12:46.01 | dongs | nothingtodo with my provider. |
12:46.07 | dongs | asterisk originates hte call. |
12:46.14 | [TK]D-Fender | dongs: and username is in place of just using the section header name etc |
12:46.29 | dongs | username = auth username. |
12:46.30 | [TK]D-Fender | dongs: SIP is SIP. Everything depends on what you're talking to |
12:46.37 | dongs | now you have defaultuser and fromuser. |
12:46.46 | dongs | how does that makesense without proper docs. |
12:47.18 | dongs | oh i have another rant. i had to edit sores of chan_sip and recompile removing that retarded ;rport= hack |
12:47.47 | dongs | because while it was *probably* possible to disable it by some combination of nat=whatever, it was easierto just take it out completely. |
12:47.59 | [TK]D-Fender | dongs: </monologue> |
12:48.08 | dongs | why something that only affects ONE random device was left in release code to be used by users? |
12:48.20 | dongs | (breaking random number of other devices in hte process) |
12:48.21 | [TK]D-Fender | dongs funny, we all seem to get by just fine |
12:48.43 | dongs | blank ;rport= crashed my provider's sip proxy with 400/bad request. |
12:48.44 | *** join/#asterisk afink (n=afink@204.26.87.226) |
12:49.24 | [TK]D-Fender | He's got a might weak proxy then :) |
12:49.25 | dongs | now you can whine about "get a better provider" all you want, and that would be completely irrelevant. |
12:49.34 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
12:49.35 | oej | dongs: Part of the problem was that everyone misunderstood "username=" and did not get that it was used together with defaultip= |
12:49.45 | oej | It was a poorly named configuration option from start |
12:49.55 | oej | username= has nothing to do with the name of the device at all. |
12:50.03 | [TK]D-Fender | oej: The man! |
12:50.15 | afink | morning everyone, I am trying to get streaming moh working and having a hard time getting it to work. I have been reading this and I have tried quite a few of the examples on here http://www.voip-info.org/wiki/view/Asterisk+config+musiconhold.conf and haven't got anything to work. Anyone have a working example they can show me? |
12:50.21 | dongs | oej: both defalutuser and fromuser are just as poor. wouldnt something making sense like "authuser" be better , especailly properly documented. |
12:50.35 | oej | Yes, that's something I would like to add. |
12:50.44 | oej | I hate config options with multiple uses |
12:50.50 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
12:50.55 | oej | fromuser is separate from username/defaultuser |
12:51.05 | oej | fromuser and fromdomain belongs together to form the From: header |
12:51.13 | kaldemar | multiple different comments for them in samples are just as fun. |
12:51.14 | *** join/#asterisk qdk (n=qdk@81.7.168.130) |
12:51.22 | [TK]D-Fender | \o/ |
12:51.22 | dongs | uhhuh |
12:51.37 | oej | so now we have "defaultuser" and "defaultip" on one side and "fromuser" and "fromdomain" on the other so you see which belongs together. |
12:51.43 | oej | We're still missing authuser I believe |
12:51.44 | kaldemar | the best one in sip.conf.sample is "Many SIP providers require this!" |
12:51.59 | oej | authuser is best set with the realm auth config |
12:52.03 | dongs | kaldemar: thats why i added it in my shit when i was moving the settings over! |
12:52.12 | dongs | i was like omg, most require, i must add it! |
12:52.20 | kaldemar | twice! |
12:53.16 | [TK]D-Fender | s/must/many. |
12:53.32 | [TK]D-Fender | Keep on writing between the lines... |
12:55.06 | afink | drr, wrong path to mpg123 |
12:58.30 | eliasp | hi |
12:59.02 | *** join/#asterisk dysinger_ (n=tim@71-20-35-99.war.clearwire-wmx.net) |
12:59.18 | eliasp | i have this extensions.conf: http://pastebin.com/ffadebdf incoming calls on the mISDN trunk are now rejected with this message: |
12:59.19 | eliasp | WARNING[13643] pbx.c: Channel 'mISDN/3-u0' sent into invalid extension '955327180' in context 'default', but no invalid handler |
12:59.31 | eliasp | has anyone a hint what i did wrong/were to look at? |
12:59.38 | eliasp | s/were/where/g |
12:59.58 | leifmadsen | exten = o,1, <-- definitely wrong |
13:00.29 | leifmadsen | eliasp: basically you're sending the call from MISDN into the default context, which doesn't contain a pattern match, or that number, which matches |
13:00.42 | leifmadsen | and you have no invalid handler (i extension) which means you get that message |
13:01.02 | leifmadsen | send it to a context that does match |
13:01.06 | eliasp | ah, ok... |
13:01.12 | [TK]D-Fender | eliasp: And you have no exten to match that number |
13:01.19 | [TK]D-Fender | eliasp: just like it says. |
13:01.27 | leifmadsen | in the dialplan you provided, nothing would match that |
13:01.37 | leifmadsen | don't confuse 's' with a catch all extension either, because it isn't |
13:01.59 | leifmadsen | actually nevermind -- I just looked again, and there are a couple contexts it would match on |
13:02.02 | eliasp | yes, and 's' is on the SIP trunk anyways... |
13:02.04 | leifmadsen | send it to one of those |
13:02.40 | leifmadsen | runs off for... a run |
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13:25.47 | kindyroot | hello, I am a new (happy?) user of asterisknow, I have everything working |
13:26.04 | kindyroot | but I don't know how to make a sip call from point A to point B |
13:26.10 | kindyroot | in my local network |
13:26.16 | kindyroot | can anyone help? |
13:27.37 | *** join/#asterisk MindTheGap (n=MindTheG@187.20.141.72) |
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13:29.06 | fiddur | kindyroot: try #asterisknow |
13:29.25 | [TK]D-Fender | kindyroot: #freepbx <- |
13:31.43 | eliasp | hmm i'm quite new to Asterisk + dialplan ... just trying to understand all this while reconfiguring an existing plain asterisk setup... i'm still confused by all these terms in the dialplan... 2 questions: what does DID mean? and where in a dialplan is the actual "send call with extension FOOO to context BAR" done? in a "exten =" line and the following command? |
13:32.16 | afink | ~book @ eliasp |
13:32.22 | afink | ~book |
13:32.23 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
13:32.32 | kindyroot | fiddur: thanks |
13:35.20 | *** join/#asterisk Zambezi (i=Zulu@80.67.9.2) |
13:37.08 | eliasp | afink: just reeding the freely available PDF version... looks nice.. let's see how far i come... |
13:37.36 | dongs | [Sep 2 01:42:22] NOTICE[874]: chan_sip.c:18520 handle_request_invite: Call from 'siptopstnadapter' to extension 'numberimcalling' rejected because extension not found. |
13:37.37 | dongs | now what. |
13:37.45 | dongs | they're both in same contexts. |
13:37.46 | dongs | etc. |
13:37.49 | dongs | what am i missing? (1.6) |
13:38.10 | Hatrix | hello, i could compile the debian package before, but today i get: |
13:38.10 | Hatrix | <PROTECTED> |
13:38.15 | kaldemar | siptopstnadapter's conext is something else |
13:38.20 | Hatrix | what's wrong? this is driving me nuts!!! |
13:38.52 | dongs | kaldemar: rly, i just checked and they're all in 'sip' context. |
13:39.42 | [TK]D-Fender | dongs: Clearly you don't have an exten to match that number in the context its looking in |
13:39.46 | kaldemar | how about "i don't believe you until you show it"? |
13:40.01 | dongs | [TK]D-Fender: printing WHAT context it looking in would be super. |
13:40.04 | [TK]D-Fender | kaldemar: You learn quickly Padawan :D |
13:40.05 | oej | dongs: run "sip show peer" on siptopstnadapter |
13:40.15 | oej | To check the context it receives calls in |
13:40.16 | dongs | kk |
13:40.17 | *** join/#asterisk Skeeter- (n=wil_c_wi@190-141.cgocable.ca) |
13:40.22 | dongs | its not reciving |
13:40.24 | dongs | originating |
13:40.26 | [TK]D-Fender | dongsYes, and like many years before it shows up when you pay attention to SIP DEBUG |
13:40.46 | kaldemar | [TK]D-Fender: quickly? lack of cluebat has made me this slow. |
13:40.55 | Skeeter- | anyone knows how to setup call routing with a distinctive ring?? |
13:40.57 | dongs | Context : sip |
13:41.17 | [TK]D-Fender | reaches for his ClueBat (tm) |
13:41.49 | [TK]D-Fender | dongs silngle lines like that are meaningless. You continue you insist you must be right. Well if you did everything perfectly it'd work. You aren't looking |
13:41.54 | Maliuta | [TK]D-Fender: the one with the big spikes and H.E. tips? |
13:42.14 | dongs | [TK]D-Fender: sip debug obviously does not show anything about searching for extensions. |
13:42.16 | [TK]D-Fender | MaliutaNo, straight ironwood :) |
13:42.26 | [TK]D-Fender | dongs: Yes it does. |
13:43.04 | [TK]D-Fender | Skeeter-: depends on the phone |
13:43.09 | dongs | Looking for numberimcalling in sip (domain mydomain) |
13:43.21 | dongs | did _ become some different meaningin 1.6? |
13:43.25 | [TK]D-Fender | dongs : I'm not seeing debug & configs... |
13:43.36 | [TK]D-Fender | dongs: No, and you continue to deflect |
13:43.37 | Skeeter- | Fender: wanna make it work for the whole system |
13:43.52 | Skeeter- | Fender: inbound distinctive ring |
13:43.59 | [TK]D-Fender | Skeeter-: Where is the distintive ring coming from? |
13:44.05 | [TK]D-Fender | Skeeter-: Inbound over what? |
13:44.45 | Skeeter- | Fender: 2 land lines, 3 phone number, 1 of the phone number is using a distinctive ring, and cannot be process |
13:44.59 | [TK]D-Fender | Skeeter-: what is it coming IN on? |
13:45.01 | *** join/#asterisk russellb_ (n=russell@asterisk/digium-open-source-team-lead/russellb) |
13:45.01 | *** mode/#asterisk [+o russellb_] by ChanServ |
13:45.24 | Skeeter- | Fender: can you be more specific |
13:46.13 | [TK]D-Fender | Skeeter-: What kind of friggen wire, plugged into what HARDWARE? |
13:47.02 | dongs | okay this is ridiculous. |
13:47.21 | dongs | extension IS in [sip] and i even got rid of _ and just replaced with 0 |
13:47.25 | dongs | still same shit, not found. |
13:47.35 | Skeeter- | Fender: regular land phoen wire, into the sangoma card, which is into the asterisk server |
13:47.37 | dongs | and itis lookingin sip context for it. |
13:47.53 | [TK]D-Fender | dongs: And you're still not showing the backup |
13:48.00 | Skeeter- | Fender: i want 2 phones lines, to work on 1 zap |
13:48.22 | dongs | [TK]D-Fender: what backup? |
13:48.28 | *** join/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
13:48.28 | *** mode/#asterisk [+o russellb] by ChanServ |
13:48.30 | [TK]D-Fender | Skeeter-: if its ANALOG, then go look on the WIKI for "distinctive ringing" |
13:48.37 | [TK]D-Fender | dongs: SIP Debug & dialplan |
13:49.11 | dongs | i did that. |
13:49.25 | dongs | already pasted relevant parts. |
13:49.25 | eliasp | what does ${EXTEN:0} or ${EXTEN:5} mean? i know what ${EXTEN} is, but i can't figure out, what the :$NUMBER means..... |
13:49.28 | *** join/#asterisk s14ck (n=s14ck@190-76-92-153.dyn.movilnet.com.ve) |
13:49.34 | dongs | eliasp: first n chars of extensioncut off. |
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13:49.57 | eliasp | dongs: aaah, this explains a lot.. thx! |
13:50.14 | dongs | [TK]D-Fender: Looking for numberimdialing in sip (domain mydomain) |
13:50.16 | [TK]D-Fender | eliasp: Chop off X digits |
13:50.17 | dongs | this is in sip debug. |
13:50.18 | dongs | getting 404. |
13:50.25 | [TK]D-Fender | dongs: PASTEBIN <- |
13:50.31 | dongs | ohfucksake. |
13:50.47 | [TK]D-Fender | dongs: You are still showing only worthless tiny bits and masking what little could be of any value at all in them |
13:51.07 | dongs | http://bcas.tv/paste/results/cxBInl92.html |
13:51.11 | dongs | happy now. |
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13:51.34 | dongs | or do you want to know what audio format the call was palced in? |
13:51.44 | [TK]D-Fender | dongs: I don't see the number or your fucking dialplan |
13:51.52 | [TK]D-Fender | dongs: You are showing precisely jack shit and being a whiny bitch about it. This is simple dialplan |
13:52.03 | dongs | precisely why I expect it to work. |
13:52.07 | [TK]D-Fender | dong and hasn't changed in a DECADE |
13:52.19 | beek | is surprised that [TK] hasn't exploded yet. |
13:52.31 | [TK]D-Fender | dongs: Now stop being a dumbass and PASTEBIN the actual call debug and your friggen dialplan |
13:52.41 | dongs | http://bcas.tv/paste/results/auLIGh53.html |
13:52.45 | beek | thinks it's getting close. |
13:52.46 | dongs | ^ |
13:52.57 | dongs | very simple. |
13:53.05 | eliasp | wee, got it working... i think i finally got it, how the dialplan works ;-) |
13:53.09 | eliasp | thx a lot guys |
13:53.24 | [TK]D-Fender | dongs: Again you're wasteing time masking everything |
13:53.36 | dongs | i didnt mask anything. |
13:53.46 | [TK]D-Fender | dongs: numberimcalling |
13:53.48 | [TK]D-Fender | O RLY |
13:53.51 | dongs | yes. its a number |
13:53.52 | dongs | fixed. |
13:54.02 | dongs | it had a _ before it |
13:54.11 | dongs | because my fucking sip/pstn adapter requires 0 beforeitdials out |
13:54.17 | dongs | so i replaced that with 0. |
13:54.20 | [TK]D-Fender | dongs: Stop masking things and show your actual code, I can tell you are hacking this shit up at every turn |
13:54.26 | [TK]D-Fender | dongs: Stop being a moron |
13:55.08 | kaldemar | _ doesn't match a digit or character. it tells asterisk that the extension is a pattern. |
13:55.17 | Maliuta | [TK]D-Fender: I think that's like telling it not to breathe ... assuming it's actually alive |
13:55.18 | Katty | :< |
13:55.32 | [TK]D-Fender | indeed doesn't match anything, only indicates that what follows is a pattern |
13:55.33 | Maliuta | waves at Katty |
13:55.52 | Katty | hi :< |
13:56.56 | Maliuta | Katty: 'sup? |
13:57.03 | Katty | stuck in toshiba meeting :< |
13:57.37 | beek | Katty: Are you being punished for something? ;-) |
13:57.42 | Maliuta | Katty: could be worse ... it could be Dell ;) |
13:57.45 | Katty | sometimes i wonder. |
13:58.04 | Katty | we don't sell asterisk based systems anymore |
13:58.36 | dongs | kaldemar: thx, works. |
13:59.09 | beek | Katty: that's unfortunate. |
13:59.24 | Katty | we also sell Samsung |
13:59.29 | beek | Samslug |
13:59.35 | Katty | we're whores. |
13:59.44 | Katty | :< |
13:59.47 | beek | Why no *? |
13:59.53 | Katty | no one wants it |
13:59.57 | Katty | they don't recognize the Brand |
14:00.37 | beek | That's depressing. |
14:00.41 | Katty | yes. |
14:00.58 | *** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
14:01.14 | beek | I actually just installed one at a site where the president of the company said "why are we replacing one proprietary piece of shit with another proprietary piece of shit." They have an Asterisk box now and are quite happy with it. |
14:02.19 | [TK]D-Fender | dongs: So, got something real to show for a change? Or have you found your mistake? |
14:03.24 | Katty | beek: well. i still use asterisk here for our phone system. we still have people that use it. i have one at home. |
14:03.42 | Katty | beek: just more stuff for the resume. |
14:03.51 | beek | Gotta look at the bright side... |
14:04.01 | Katty | nods |
14:04.14 | Maliuta | considers bed |
14:04.17 | [TK]D-Fender | beek: "WARNING : Do not look into the laser with your remaining good eye" |
14:04.34 | beek | Good advice! |
14:06.11 | Skeeter- | how can i see dring in the Asterisk CLI |
14:06.21 | Skeeter- | what verbose i should use |
14:06.40 | Katty | i always use 10 |
14:08.44 | Katty | this toshiba guy is all sales and no tech :< |
14:09.40 | *** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:4df0:d9ce:287a:92ea) |
14:10.17 | Skeeter- | still cant see it |
14:10.27 | Skeeter- | must be some : show .... command |
14:10.39 | [TK]D-Fender | Skeeter-: show us you've configured it right |
14:11.04 | beek | Katty: Have some fun with him and ask him some nice techie questions and make him squirm. |
14:11.13 | Katty | seriously considering it. |
14:11.45 | beek | You may as well enjoy yourself and it's some much better when it's at someone else's expense. |
14:12.03 | Katty | yeah but there are 3 sales reps in here with me |
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14:12.18 | Katty | it'd probably cause drama |
14:12.21 | beek | you really are being punished |
14:14.34 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
14:14.37 | Skeeter- | http://pastebin.com/m56696506 |
14:15.07 | Skeeter- | i just need to add the dring= |
14:15.21 | Skeeter- | with the 3 numbers, but i cant find those numbers in the CLI |
14:15.51 | [TK]D-Fender | Skeeter-: well you HAVEN"T done it. You certain won't see anything unless * is told to look for it |
14:16.02 | [TK]D-Fender | Skeeter-: you aren't going to see NUMBERS there |
14:16.24 | Katty | well. it could be worse. it oculd be 65F in here. |
14:16.45 | Skeeter- | Fender, the wiki says to look at the CLI to find the dring number, how am i suppose to find them then |
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14:17.49 | *** join/#asterisk cusco (n=trilili@2001:0:53aa:64c:1887:6a0:2ac0:762d) |
14:17.53 | cusco | hello folks |
14:18.55 | *** join/#asterisk maour (n=gnu@unaffiliated/maour) |
14:19.07 | Katty | hello |
14:19.09 | [TK]D-Fender | Skeeter-: core set debug 10 |
14:19.46 | cusco | we recently changed our service, clients calling us now have to press 1 if they have client nr and access code |
14:20.00 | cusco | several things happening: |
14:20.30 | cusco | when we press 1, sometimes jumps straight to asking access code, not asking client nr |
14:21.01 | cusco | sometimes when we press keys, they are not interpreted as a dftm? not taken by the system |
14:21.20 | dongs | well looks like after some cockups my 1.4->1.6 migration is ok. |
14:21.35 | Katty | cusco: have you considered hiring a consultant to have a look at the system for you? |
14:21.39 | dongs | nothanks to [TK]D-Fender and some thanks to kaldemar ^_^ |
14:21.51 | [TK]D-Fender | dongs: Yeah you got what you gave. |
14:22.04 | dongs | now i just gotta make a call longer than 5 mins to see if session timer shit actually works. |
14:22.15 | [TK]D-Fender | dongs: And we know the mistake was on your end, and no doubt something embarassingly simple |
14:22.28 | [TK]D-Fender | dongs: But glad you found it |
14:23.20 | dongs | [TK]D-Fender: ? i was assuming _ = 0 and thus it was looking for -1 sized extension and notfindingit. i forgotthat particular fixed-number extension was for a long-unused softphone that didn't require 0 before dialing. |
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14:23.40 | dongs | stupid? yes. |
14:23.46 | [TK]D-Fender | dongs: Yup, embarassingly simple.. |
14:23.49 | dongs | errors should have been more helpful? yes. |
14:23.57 | [TK]D-Fender | ~assume |
14:23.58 | infobot | About assumptions : http://www.moviewavs.com/php/sounds/?id=gog&media=WAVS&type=Movies&movie=Long_Kiss_Goodnight"e=assume.txt&file=assume.wav It makes an (ass) out of (u) and (me) |
14:24.01 | [TK]D-Fender | ^^^^ |
14:24.15 | dongs | having to turn on debug and sift through potentially hundreds of lines to track down a misnumbered extension = lol |
14:24.20 | dongs | but iguess you get what you pay for wiht opensource. |
14:24.33 | Katty | be happy there's a log to dig through (= |
14:24.37 | [TK]D-Fender | dongs: the error WAS extremely clear. You didn't have a match. It won't GUESS that you should have done something in one exten differently. It isn't PSYCHIC |
14:24.59 | [TK]D-Fender | dongs: it told you the number and the context. It is completely your own fault for not knowing dialplan 101 |
14:25.04 | dongs | [TK]D-Fender: im dialing 0123456 |
14:25.11 | dongs | its printing "cannot find extension 0123456' |
14:25.15 | [TK]D-Fender | dongs: And never has "_" meant "0" |
14:25.16 | dongs | not very helpful. |
14:25.41 | cusco | Katty: I guess something is wrong with dftm, I would liek somebody to point me in the right direction |
14:25.44 | *** join/#asterisk Take (i=take@nerd.fi) |
14:25.58 | [TK]D-Fender | dongs: It is helpful. means you should look to see if you have an exten to match. Only thing the basic debug didn't say was the context. Then again your exten was wrong regardless |
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14:26.19 | Take | Hello. |
14:26.29 | [TK]D-Fender | dongs: But you made a big fuss and masked shit left right and center and whined |
14:26.39 | Katty | cusco: alright. fair enough (= I would be patient tho. this room is filled with volunteers. |
14:26.52 | Take | I tried (again) to search via google if asterisk could be used as PoC PTT -server (used on Nokia phones atleast), but I didn't find an definitive answer. |
14:26.54 | Katty | cusco: it may take some time to get an answer. |
14:27.06 | Take | Does anyone know anything about the issue and if it can be done? |
14:27.19 | *** join/#asterisk manxpower (n=EWieling@69.73.94.162) |
14:27.57 | *** join/#asterisk The_Boy_Wonder (n=davidvos@asterisk/batman-developer/dvossel) |
14:28.05 | *** join/#asterisk friartuck (n=pmccary@66.162.90.56) |
14:28.31 | Katty | Take: i don't know. perhaps someone else does tho. |
14:29.10 | Take | Katty: there's several conversations available and everyone has heard that it can be done, but no references :-/ |
14:29.12 | Kobaz | http://www.reuters.com/article/newsOne/idUSTRE57U02B20090831 |
14:29.29 | Katty | Take: i would recommend being patient and waiting to see if anyone has an answer. |
14:29.44 | Take | Katty: I'm not going anywhere ;) |
14:29.46 | Skeeter- | FEnder: how can i see the debug log after that |
14:29.48 | Katty | (= |
14:29.49 | leifmadsen | I am! |
14:29.50 | *** part/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
14:29.56 | Katty | lol. |
14:30.59 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
14:31.00 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
14:31.00 | [TK]D-Fender | Skeeter-: that is CLI DEBUG |
14:31.06 | dongs | Kobaz: i just saw that on the news this evening. |
14:31.32 | dongs | good thing I got my hybrid before the chinks figured it out. |
14:31.47 | Katty | hi Deeewayne! |
14:32.05 | Deeewayne | hugs Katty |
14:32.07 | Deeewayne | hello :-) |
14:32.13 | Katty | hugs on Deeewayne |
14:32.58 | Deeewayne | rough morning.... I had to turn around halfway to work to rush to the dentist to hog tie my kids (3 & 5) on the dentist's bench |
14:33.00 | Skeeter- | Fender: then it didnt work |
14:33.06 | [TK]D-Fender | dongs: careful on your wording there... racial commentary like that can send you skidding out of here. |
14:33.16 | Katty | Deeewayne: eesha :< |
14:33.40 | Katty | Deeewayne: teeth cleaning? |
14:33.52 | Skeeter- | http://pastebin.com/m6a61d36b |
14:33.55 | Deeewayne | yes. no cavities :-) |
14:34.04 | Katty | excellent. those are never fun. |
14:35.23 | Katty | Deeewayne: your kids would probably like the new strawberry colgate. |
14:35.36 | Katty | they also carry watermellon now too (= |
14:35.57 | Deeewayne | they had bubblegum today |
14:36.04 | Deeewayne | bubblegum toothpaste, that is |
14:36.14 | Katty | trust me, the strawberry is much much better! |
14:36.16 | Katty | mm! |
14:37.10 | *** join/#asterisk AlexTO (n=aacm_ale@190.25.208.136) |
14:37.44 | cusco | how come DTMF's that arrive trough the primary channel, may be missinterpreted by Asterisk? |
14:38.38 | *** join/#asterisk Naikrovek (n=jjohnson@63-252-251-77.ip.mcleodusa.net) |
14:38.43 | Katty | hey Naikrovek (= |
14:38.58 | Naikrovek | hola |
14:39.30 | Katty | scowls. |
14:39.46 | Katty | i asked the toshiba rep a tech question, and one of my company's sales reps tried to answer me. |
14:39.49 | Katty | how insulting. |
14:39.57 | jaytee | "Wildfires are a normal part of nature, cleaning out underbrush from forest floors, allowing more room for the larger flame-retardant trees (like sequoias) to grow. These Los Angeles fires are no different, getting rid of undesirable two family houses and dense suburban areas so large celebrity mansions can take their place." |
14:40.17 | Katty | lol |
14:40.33 | beek | morning jaytee |
14:40.40 | jaytee | morning beek |
14:40.54 | beek | Katty: try again -- and this time ask your company's sales rep to politely STFU |
14:41.00 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
14:41.03 | Katty | that would cause drama. |
14:41.15 | Katty | and i have to work with these people all day. |
14:41.19 | beek | That would make a boring day interesting. |
14:41.21 | manxpower | Ask his boss when that person started working for Toshiba |
14:41.38 | beek | manxpower: nice touch |
14:41.39 | Katty | her boss, is sitting right in front of me. |
14:41.44 | Katty | this is a very small company. |
14:41.53 | Katty | the drama would come from both. |
14:41.56 | Katty | trust me. |
14:42.04 | manxpower | Katty: in that case I think poison is the solution of choice. |
14:42.11 | Katty | manxpower: i am forced to agree. |
14:42.24 | Katty | to tell you how bad it is, i had to make a public and a private facebook |
14:42.31 | Katty | because drama spawned from my political preferences. |
14:42.41 | manxpower | Katty: sounds like it's time to find a good place to work. |
14:42.50 | Katty | sighs. |
14:42.56 | Katty | i keep thinking the same thing. |
14:44.01 | *** join/#asterisk dmz (n=dmz@244.sub-75-209-248.myvzw.com) |
14:44.08 | cusco | lol @ "Female voice are known to once in a while trigger the recognition of a DTMF tone." |
14:44.09 | Katty | really it's not too bad. it's just one sales rep here that causes drama. |
14:44.28 | *** join/#asterisk wwalker (n=wwalker@72.249.1.66) |
14:44.38 | Katty | because if she feels she's been Wronged, she'll go tell everyone in the office and drag the owner into it. |
14:45.38 | wwalker | anyone have a reference to the actual speed one can expect from asterisk for originate requests through AMI? I seem to be getting an average of about 10 calls/sec with maybe 18 calls/sec peak. |
14:45.39 | Take | Katty: properly used laxatives after drama queen -acts work as a great lesson ;) |
14:45.42 | jaytee | if someone doesn't like my political preferences they are free to fuck themselves and die at any time, in the workplace or not. |
14:46.12 | *** join/#asterisk Jymm (i=jim@unaffiliated/jymmm) |
14:46.30 | Katty | jaytee: i agree. |
14:46.35 | *** join/#asterisk Ex_peter (n=Ex_peter@unaffiliated/expeter/x-019426) |
14:46.40 | Katty | jaytee: everyone here is right winged conservative. |
14:47.07 | Katty | jaytee: and i disagree on somethings, obviously, but i don't appreciate 5 different people trying to lecture me on my political views and how it may adversely affect the company's Reputation |
14:47.18 | jaytee | ugh, I hate those people. they all smell of death, decay and having drank the blood of small children |
14:48.02 | Katty | our customers don't need to know what my political preferences are. |
14:48.05 | Katty | nor do my co-workers. |
14:48.26 | Katty | so i just kicked everyone from work off my facebook, and put them on another one. if i'd just dumped them completely there'd be drama. |
14:48.34 | Katty | why did you dump $coworker? did they upset you? blahblahblah |
14:49.02 | Jymm | In X-lite, all unidentifed calls are going directly to VM, any suggestion? I can't seem to find the filter to allow them. |
14:49.51 | dongs | <Katty> our customers don't need to know what my political preferences are. < neither does anyone on irc. plz blog this crap either in privmsg to someone who cares (nobody) or off-line somewehre. |
14:50.23 | *** join/#asterisk k4tanaLINUX (n=k4tanaLI@190.196.70.189) |
14:50.51 | dongs | i thought in 2009 opensores channels would be past getting a hard-on the moment someone wiht a female /whois joins a channel. i guess not. |
14:51.00 | dongs | bbl |
14:51.12 | *** part/#asterisk dongs (n=blogger@l212047.ppp.asahi-net.or.jp) |
14:51.52 | Jymm | "dongs" is a femine nick, aint it? |
14:52.29 | *** join/#asterisk netpro25_ (n=mmanning@c-71-226-86-184.hsd1.fl.comcast.net) |
14:52.32 | Katty | i'm guessing they're having a bad day. |
14:53.07 | manxpower | A lot of nut cases have been joining this channel recently. |
14:53.14 | Katty | it's okay. |
14:53.20 | Katty | they just want help. |
14:53.23 | eppigy | ello |
14:53.23 | Jymm | Maxxed: Hey now, I resemble that! |
14:53.25 | eppigy | I am dave |
14:53.31 | Katty | hello, sir! |
14:53.32 | Katty | let's hug. |
14:53.32 | jaytee | TRABAJO! |
14:53.45 | Katty | hugs on eppigy |
14:53.48 | Jymm | manxpower: Hey now, I resemble that! |
14:53.58 | eppigy | huggles Katty |
14:54.05 | k4tanaLINUX | hi people ... somebody know how to convert mp3 or wav in g729 ? |
14:54.15 | eppigy | i passed my CCNA :D |
14:54.23 | jaytee | congrats! |
14:54.29 | thehar | caps meatpaws |
14:54.31 | thehar | clap |
14:54.33 | hesco | I'm having an issue with an AGI script described here: http://perlmonks.com/?node_id=792672 Any ideas? |
14:54.34 | Jymm | eppigy: totally useless cert |
14:54.49 | Jymm | eppigy: Kidding, congrats! |
14:55.10 | Jymm | eppigy: So, how many times did it take? |
14:56.56 | Jymm | My old job would pay up to 5 times, after that, you were on your own |
14:56.56 | hesco | k4tanaLINUX: there is a command line tool that handles most audio conversions, not sure if it does that one or not. sox and flac do many conversions. |
14:57.32 | manxpower | k4tanaLINUX: you will need commercial software to convert anything to/from g729. You can do it in Asterisk 1.6, but you will need a g729 license ($10/per channel) from digium. For just conversion, you should only need 1 channel. |
14:57.57 | Jymm | ~softphone |
14:57.58 | infobot | [~softphone] A soft-phone is a program that lets your computer use a compatible audio interface (usually a speaker&mic, or headset), and optionally a webcam, to place VoIP calls with a given protocol (SIP/IAX/H.323/MGCP/etc). For links enter ~zoiper , ~xlite , ~twinkle , ~bria , ~eyebeam , ~ekiga |
14:58.04 | *** join/#asterisk saint_ (n=templar@pdpc/supporter/base/saint) |
14:58.11 | Jymm | doh |
14:58.21 | k4tanaLINUX | ~asterisk |
14:58.22 | infobot | i heard asterisk is a free PBX, or #asterisk on irc.freenode.net, or http://www.asterisk.org, or just like a mini-mall |
14:58.25 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
14:58.30 | k4tanaLINUX | :) |
14:59.24 | Jymm | nobody uses xlite? |
14:59.31 | *** join/#asterisk Chinorro (n=Chino@19.226.117.91.dynamic.mundo-r.com) |
15:00.09 | k4tanaLINUX | ~a2billing |
15:00.27 | k4tanaLINUX | no description |
15:02.21 | k4tanaLINUX | 1500 for MOR ..... :( a little expensive for me |
15:02.32 | manxpower | ~manxpower |
15:02.32 | infobot | ManxPower has been using Asterisk in production since late 2001. Currently works at InterGlobe Communications, a CLEC based in NYC with service in NY, NJ, FL, and TX. http://www.nyigc.com |
15:02.52 | *** join/#asterisk momelod (n=smelo@CPE001f3a8fe859-CM0012c91df0bc.cpe.net.cable.rogers.com) |
15:03.17 | Jymm | manxpower: a lil self promotion there, huh? |
15:04.01 | *** join/#asterisk Naikrovek (n=jjohnson@63-252-251-77.ip.mcleodusa.net) |
15:04.07 | manxpower | some people seem to think I work for Digium |
15:04.25 | Jymm | manxpower: Ah |
15:04.39 | giovani | anyone have CNAM providers to recommend? |
15:04.53 | giovani | cnam.info hasn't been accepting customers for a while now |
15:05.13 | jaytee | it's the long hooded cloak and the sheleighly that confuse people |
15:05.49 | Jymm | an irish monk? |
15:05.58 | Skeeter- | http://www.voip-info.org/wiki/view/Asterisk+ZAP+channels#DetectingDistinctiveRingonIncomingCalls |
15:06.03 | Skeeter- | this is clearly not wokring |
15:06.09 | Skeeter- | cant find the numbers |
15:06.18 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
15:06.34 | *** join/#asterisk [TK]D-Fender (n=chatzill@216.191.106.162) |
15:06.43 | Jymm | Ebay dumps skype... http://money.cnn.com/2009/09/01/technology/ebay_skype/index.htm?postversion=2009090108 |
15:06.56 | Katty | hi fender. |
15:07.15 | momelod | greetings channel |
15:07.23 | [TK]D-Fender | Jymm: and plenty of people use X-Lite |
15:07.53 | momelod | how do i list my configured trunks from the CLI? |
15:08.02 | manxpower | Skype: The AOL of VoIP |
15:08.16 | [TK]D-Fender | momelod: SIP show peers |
15:08.44 | *** join/#asterisk Naikrovek (n=jjohnson@63.252.251.77) |
15:09.46 | manxpower | momelod: "zap show channels" |
15:09.52 | momelod | [TK]D-Fender: sorry i should have been more clear. im trying to setup a te122 card and i want to see if the channels are referred to as Zap/* or DAHDI/* |
15:10.49 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
15:10.49 | momelod | manxpower: right, but 'dahdi show channels' gives me the same output |
15:11.00 | [TK]D-Fender | momelod: either can work if you configured it as such |
15:11.02 | *** join/#asterisk xpot-mobile (n=james@mx0.synergyconsultant.net) |
15:11.12 | momelod | When i dial in, i see the channel in use is Zap/1-1 and these calls work. But when i dial out, i see DAHDI/g1 and i get an all circuits are busy message. Now when i edit the trunks, in the trunk identifier i put in g1 but how do i force Zap over DAHDI |
15:11.13 | manxpower | momelod: then the only way to know is to know what you have installed. zaptel or dahdi |
15:11.36 | [TK]D-Fender | momelod: that message is BS. |
15:11.41 | momelod | im using dahdi |
15:11.54 | momelod | but im trying to get it working with freepbx |
15:12.00 | [TK]D-Fender | momelod: Go look to see if chan_dahdi.so is even loaded, and pastebin "dahdi show channels" |
15:12.05 | momelod | and i set it to run in zap compatability mode |
15:12.47 | manxpower | momelod: I wish you the best of luck |
15:12.50 | manxpower | ~freepbx |
15:12.51 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
15:14.23 | momelod | [TK]D-Fender: http://pastebin.ca/1550385 |
15:14.52 | [TK]D-Fender | momelod: ok, looks fine so far |
15:15.10 | momelod | yeah, incoming is working.. just having problems with outgoing |
15:16.08 | [TK]D-Fender | momelod: And I don't see the problem |
15:16.16 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
15:16.52 | Naikrovek | someone say something |
15:16.57 | [TK]D-Fender | something |
15:16.59 | Naikrovek | testing my network |
15:17.00 | Jymm | woof |
15:17.01 | Naikrovek | perfect :) |
15:17.12 | eppigy | Jymm: 1 |
15:17.18 | [TK]D-Fender | ~areyouadog ? |
15:17.19 | infobot | Bark! Bark! |
15:17.22 | [TK]D-Fender | ~botsnack |
15:17.22 | infobot | [TK]D-Fender: :) |
15:17.27 | Naikrovek | finally got his Cisco router ACLs fixed up |
15:17.31 | [TK]D-Fender | infobot: Good Boy! |
15:17.31 | infobot | [TK]D-Fender: aw, gee |
15:17.36 | Katty | prods eppigy |
15:17.39 | momelod | [TK]D-Fender: that makes two of us :) |
15:17.40 | eppigy | :> |
15:17.52 | [TK]D-Fender | momelod: Yeah, You aren't SHOWING us. |
15:18.00 | eppigy | now I am ordering my ccnp stuffs |
15:18.07 | Katty | orly |
15:18.10 | eppigy | yesh |
15:18.19 | Katty | have you ordered a tree for your computer room yet? |
15:18.20 | eppigy | which will be a lot more interesting |
15:18.28 | eppigy | since it actually applies to what I do daily |
15:18.35 | eppigy | Katty: negative :[ |
15:19.01 | Jymm | eppigy: Since when is sitting on your butt playing WoW on a test? |
15:19.24 | *** join/#asterisk ArchGT (n=ArchGT@unaffiliated/archgt) |
15:19.36 | Jymm | eppigy: I'm just asking =) |
15:19.38 | Katty | Jymm: be nice to eppigy |
15:19.42 | Katty | Jymm: don't be rude. |
15:19.57 | eppigy | Jymm: lol |
15:20.00 | Katty | Jymm: life's too short to run around being a mean person. |
15:20.06 | eppigy | wow is a huge black hole |
15:20.10 | eppigy | for your time and money |
15:20.11 | Jymm | \ignore Katty |
15:20.16 | *** part/#asterisk manxpower (n=EWieling@69.73.94.162) |
15:20.56 | Jymm | eppigy: So, what did you think of the ccna test? what you expected? |
15:21.19 | eppigy | it was a lot easier than I thought it was going to be |
15:21.34 | eppigy | I would have gotten and almost perfect score |
15:21.44 | eppigy | but I made a mistake on a five part question |
15:21.47 | Jymm | eppigy: which study material did you use? |
15:22.13 | eppigy | http://www.amazon.com/Official-Certification-Library-640-802-Guide/dp/1587201836/ref=sr_1_2?ie=UTF8&s=books&qid=1251818527&sr=1-2 |
15:22.46 | eppigy | I also purchased some switches and routers |
15:22.48 | eppigy | for a home lab |
15:22.54 | eppigy | then our netadmin left |
15:23.00 | Jymm | eppigy: cisco's typical dry reading? |
15:23.01 | eppigy | and I took up the position |
15:23.10 | eppigy | Jymm: of course |
15:23.38 | eppigy | I mean if you could convey binary math and distance vector loop prevention measures |
15:23.43 | eppigy | in a light and colorful manner |
15:23.46 | eppigy | i'd love to see it |
15:25.27 | Jymm | Actually, I probably could. But authors dont make all that much money. |
15:25.44 | eppigy | yeah dude screw the arts |
15:25.58 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
15:27.09 | *** join/#asterisk Qwell (i=north@pdpc/sponsor/digium/Qwell) |
15:27.09 | *** mode/#asterisk [+o Qwell] by ChanServ |
15:27.17 | *** join/#asterisk Talirk81 (n=David@rrcs-67-78-39-22.sw.biz.rr.com) |
15:27.34 | Jymm | eppigy: (My tech writing usually focuses on a broad audience of all/no skill levels) |
15:28.17 | Talirk81 | I am running "AGI Script Executing Application: (VERBOSE) Options: (SET VARIABLE __SetCallerID_Name 'Level Call'" but when i use NoOP I see Executing [s@DialCallCenter:14] NoOp("SIP/vjzwsq-b7500468", "Level") in new stack why is it not capturing the full contents of the variable even though they are in quotes to prevent confusion by the system? |
15:28.38 | eppigy | I cannot write anything without it reading like hunter s thompson |
15:29.58 | Jymm | eppigy: LOL, so YOU'RE the one that wrote that manual! |
15:31.16 | *** join/#asterisk bmoraca (n=chatzill@66.242.174.254) |
15:32.16 | *** part/#asterisk Katty (n=asterisk@mail.copi-rite.com) |
15:34.13 | *** join/#asterisk ruben23 (n=RPL@122.55.48.243) |
15:34.17 | kindyroot | hello, I have asterisk not listening to port 5060, and I can't fix that |
15:35.01 | [TK]D-Fender | kindyroot: Show us |
15:35.52 | kindyroot | <[TK]D-Fender> http://pastebin.ca/1550382 |
15:36.23 | [TK]D-Fender | kindyroot: now DON'T filter it |
15:36.56 | kindyroot | ok |
15:37.29 | *** join/#asterisk hfb (n=hfb@pool-96-247-114-183.lsanca.dsl-w.verizon.net) |
15:38.03 | wwalker | anyone have a reference to the actual speed one can expect from asterisk for originate requests through AMI? I seem to be getting an average of about 10 calls/sec with maybe 18 calls/sec peak. |
15:38.32 | kindyroot | <[TK]D-Fender> http://pastebin.ca/1550411 |
15:39.21 | [TK]D-Fender | kindyroot: not with a BRAIN : netstat -an |
15:39.24 | [TK]D-Fender | now* |
15:41.15 | *** join/#asterisk oej (n=olle@ns.webway.se) |
15:47.09 | *** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:4df0:d9ce:287a:92ea) |
15:48.06 | kindyroot | <[TK]D-Fender> http://pastebin.ca/1550416 |
15:49.37 | Qwell | In case people hadn't seen it - http://www.reuters.com/article/pressRelease/idUS144421+01-Sep-2009+BW20090901 |
15:49.58 | [TK]D-Fender | kindyroot: Now show me that * is running |
15:51.06 | Jymm | Qwell: Did you see that Ebay is dumping skype and Marc Andressen is buying it? |
15:51.34 | Qwell | I hadn't seen that |
15:51.43 | mort_gib | how do I enable ilbc |
15:51.49 | Jymm | Ebay dumps skype... http://money.cnn.com/2009/09/01/technology/ebay_skype/index.htm?postversion=2009090108 |
15:51.53 | Qwell | Why is that name familiar? |
15:52.00 | Jymm | Qwell: Netscape |
15:52.05 | Qwell | right |
15:52.08 | Qwell | and something else |
15:52.39 | Jymm | Qwell: I forget his other post-netscape comany name, though I drove past the bldg the other day and I see an HP sign out front. |
15:52.59 | Jymm | doesn't mean much |
15:53.31 | rene- | wwalker: have you read about nir simionovich work? |
15:54.13 | [TK]D-Fender | mort_gib: There are libs to install that are in the docs... |
15:54.33 | mort_gib | Hi TK, just found it, is ilbc any good?? |
15:55.20 | [TK]D-Fender | mort_gib: I have no purpose for it |
15:55.54 | mort_gib | I have a VOIP provider saying "this is the way to go"... I'm testing them out... |
15:56.10 | kindyroot | <[TK]D-Fender> [root@localhost asterisk]# ps -A | grep asterisk |
15:56.11 | kindyroot | <PROTECTED> |
15:56.11 | kindyroot | <PROTECTED> |
15:56.58 | [TK]D-Fender | kindyroot: now connect to * cli and do "sip show peers" |
15:57.48 | kindyroot | as a sip client I have Ekiga, is that okay? |
15:58.03 | kindyroot | what should I do? |
15:59.44 | kindyroot | <[TK]D-Fender> is that what you mean by * cli? |
15:59.50 | raden_work | ok our fall over internet connection would have been handy this morning :( |
16:01.17 | [TK]D-Fender | kindyroot: If you don'tknow how to log into * CLI you have a lot of basic reading to do... |
16:01.24 | [TK]D-Fender | kindyroot: asterisk -rvvvvvvvvv |
16:01.42 | raden_work | [TK]D-Fender, morning |
16:01.47 | wwalker | rene-: no, will read up after lunch, thx |
16:02.12 | [TK]D-Fender | is off to lunch |
16:02.48 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.157) |
16:07.06 | kindyroot | <[TK]D-Fender> honestly that's the first time I hear about asterisk client :s |
16:07.13 | kindyroot | googling right now |
16:07.49 | kindyroot | [TK]D-Fender> ah you mean like FreePBX? |
16:08.23 | Talkradio | looking for suggestions on phones you guys like to use, i'm using polycom 330 and volume level is an issue for the old ladies working there |
16:08.42 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
16:08.48 | Jymm | Talkradio: buy new and improved ladies? |
16:09.01 | Talkradio | i wish i had that option lol |
16:09.47 | kaldemar | kindyroot: no, not FreePBX. FreePBX is a GUI to configure asterisk. you connect to asterisk CLI (command line interface) when asterisk is running. |
16:10.49 | kindyroot | kaldemar: haaaaaa! |
16:11.00 | kindyroot | this must be what I am missing |
16:11.15 | kindyroot | does that cli come with *now? |
16:11.57 | *** join/#asterisk GameGamer43 (n=GameGame@h69-129-142-83.mdsnwi.tisp.static.tds.net) |
16:11.57 | kaldemar | it comes with asterisk, so yes. |
16:12.13 | kaldemar | it's not a client software, it's a built-in CLI. |
16:12.29 | kindyroot | what's the name of the command ? |
16:12.34 | hardwire | sigh |
16:12.37 | hardwire | why do we have bosses? |
16:12.48 | kaldemar | kindyroot: < [TK]D-Fender> kindyroot: asterisk -rvvvvvvvvv |
16:13.10 | kaldemar | kindyroot: are you using asterisknow? |
16:13.27 | kindyroot | kaldemar: yes a fresh install of it |
16:14.21 | kindyroot | haaa I was dropped into the cli mini-shell |
16:14.37 | kaldemar | then go ask in #asterisknow. people here use plain asterisk, not GUI's. if someone tells you how to fix the problem from console, the GUI will just screw it up again sooner or later. |
16:15.53 | kindyroot | kaldemar: I can drop freepbx if I can solve my problem by hand |
16:16.07 | kindyroot | my purpose is to get things working and to learn |
16:16.38 | kindyroot | btw I have already been to asterisknow and they couldn't help |
16:18.23 | carrar | time to switch to Asterisk compiled from source then! |
16:19.04 | *** join/#asterisk jplank (n=GBove@cpe-098-024-169-236.carolina.res.rr.com) |
16:19.32 | kindyroot | carrar: I'd love to, I am just still kept back by fear |
16:19.44 | hardwire | anybody have a multi-associatable bluetooth headset? |
16:19.48 | kaldemar | kindyroot: what does command "sip show settings" show you? |
16:19.50 | hardwire | that's a word.. I checked. |
16:20.04 | kindyroot | carrar: even the fully configured distro didn't work for me |
16:20.12 | *** join/#asterisk hugorebelo (n=hugorebe@200-171-132-124.completo.com.br) |
16:20.37 | kindyroot | no such command |
16:21.07 | kaldemar | kindyroot: what about "module load chan_sip.so" ? |
16:21.17 | bpgoldsb | In my dialplan, I make lots of use of goto. However, goto doesn't support arguements. The community/docs say that using Gosub/Macro without returning is bad. What are my options for setting arguements for the context I'm going to? Currently I try to Set(VAR) before jumping, then inside the jumped-to context, check that VAR is set. Does anyone have a better idea? |
16:21.28 | *** join/#asterisk seanmh (n=johndoe@207.114.199.107) |
16:22.03 | kindyroot | it passed, means worked? |
16:22.12 | bpgoldsb | Another idea is to use the extension '_!' and key off ${EXTEN}, but that posses multiple problems. |
16:22.57 | kindyroot | kaldemar: what is this supposed to do? |
16:23.11 | Jymm | What's the purpose/benefit of having asterisk on a lil linksys router? I just don't "get it" |
16:23.16 | kaldemar | bpgoldsb: if i were you, i'd refactor my dialplan to use GoSub. |
16:23.37 | kaldemar | kindyroot: load the sip channel module, since it seems that it's not loaded. |
16:23.45 | kaldemar | kindyroot: now pastebin the output |
16:23.49 | raden_work | is there a way to set a unavailable message up with vitelity in case we loose registration |
16:24.03 | *** join/#asterisk qdk (n=qdk@195.242.194.41) |
16:24.18 | kindyroot | kaldemar: I am afraid there was no output, it just took it silently |
16:24.57 | *** join/#asterisk haryv (i=lanny@174.1.123.38) |
16:25.04 | kaldemar | kindyroot: "core set verbose 10" and try again |
16:25.10 | bpgoldsb | kaldemar, I was under the impression Gosub was meant to do something, and _return_. Is that not the case? |
16:25.27 | kaldemar | bpgoldsb: yes |
16:25.35 | kaldemar | it is the case. |
16:26.06 | kaldemar | if you want a mere goto and values to variables, keep on using Set and Goto. |
16:26.14 | *** join/#asterisk jkroon (n=jkroon@dsl-240-169-69.telkomadsl.co.za) |
16:27.11 | kindyroot | kaldemar: Verbosity was 9 and is now 10 |
16:27.24 | kindyroot | module load chan_sip.so |
16:27.35 | kindyroot | then again, took it silently |
16:29.37 | kaldemar | interesting. try "restart now" and the module command again. |
16:30.32 | garymc | im installing sox on my asterisk server now and I think i need mpg123 |
16:30.46 | garymc | do i just yum install them and hey presto? |
16:31.26 | *** join/#asterisk nohup_ (n=nohup@chef6.nohup.nl) |
16:31.30 | nohup_ | good afternoon |
16:31.31 | kindyroot | Disconnected from Asterisk server |
16:31.31 | kindyroot | Executing last minute cleanups |
16:31.41 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
16:32.43 | kindyroot | kaldemar: then I go through the cli again and load the module? |
16:32.50 | kaldemar | yes |
16:33.16 | nohup_ | i am thinking about 'upgrading' my power-slurping server to an intel atom pc, without any PCI slots. In my current server i have a ZAPATA PCI card, that won't work anymore.. so my question is if there would be any USB solutions i could use instead? |
16:33.54 | kindyroot | kaldemar: still goes silently |
16:34.01 | *** join/#asterisk sjobeck (n=Adium@64.122.41.37) |
16:34.06 | kindyroot | I think it's loaded now |
16:34.39 | kaldemar | what kind of solutions? analog? BRI? PRI? |
16:34.41 | kindyroot | but netstat still shows that It's not listening to the right port |
16:34.55 | kaldemar | kindyroot: why do you think it's loaded? |
16:34.58 | *** part/#asterisk sjobeck (n=Adium@64.122.41.37) |
16:35.03 | kindyroot | kaldemar: all software for the moment |
16:35.49 | kindyroot | kaldemar: because otherwise it would thrash garbage at my face, just a thought |
16:35.53 | nohup_ | kaldemar: sorry.. i have my landline (analog pstn) connected to my asterisk box |
16:36.55 | haryv | did asterisk ever resolve the line presence issue? |
16:37.41 | Qwell | haryv: you're gonna have to be a bit more specific... |
16:38.26 | kindyroot | kaldemar: what are my chances if I reinstall? |
16:38.27 | kaldemar | nohup_: you could use an ATA |
16:38.32 | haryv | Qwell, say a employee is on line 1 in the other room. It should show that line presence on all phone that this employee is on that line. |
16:38.50 | [TK]D-Fender | haryv: what is "line"? |
16:38.54 | nohup_ | kaldemar: you have those for imcomming pstn too? and they'recompatible with asterisk ? |
16:38.57 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
16:39.01 | [TK]D-Fender | haryv: And what sounds like stuff I've done already |
16:39.09 | kaldemar | nohup_: it converts analog to SIP for example |
16:39.11 | nohup_ | i have one of those linksys with two 'outgoing' pstn lines already...which works pretty good i might ass... |
16:39.21 | bpgoldsb | In Asterisk 1.6, is it still 7 levels of nesting before Asterisk explodes? |
16:39.35 | haryv | line1,line2,line3 on polycom phones. a local IP provider could never get it to work. |
16:39.56 | cusco | We also noticed sometimes extra DTFM comming trough DAHDI. Clinet types only, say a phone number and instead of 9 chars, we get 10 or 11 |
16:40.06 | [TK]D-Fender | haryv: then you aren't using the phone right |
16:40.19 | haryv | your not reading my message. |
16:41.00 | [TK]D-Fender | haryv: "a lcoa IP provider"? Who? Get what to work? What are they doing on their end? |
16:41.07 | haryv | A local ip provider did not get it to work properly. Mine was never setup that way. |
16:41.23 | kindyroot | well thank you <[TK]D-Fender> and <kaldemar> very much for your help, I think I will continue tomorrow, have a nice day/night |
16:41.28 | [TK]D-Fender | haryv: So what have you now done which still doesn't work? |
16:42.14 | haryv | TK, never configured it to work in that way. But does it even with any sip channel in use ? |
16:42.40 | nohup_ | kaldemar: what is the name of such a thing? (so i can go ebay for some stuff :) ) |
16:42.47 | [TK]D-Fender | haryv: Yes they support presence |
16:43.34 | haryv | I like to disprove them wrong but then again, thay are leaving the voip biz and entering into a different model |
16:43.51 | *** join/#asterisk frieze (n=frieze@pool-74-101-21-2.nycmny.fios.verizon.net) |
16:44.20 | frieze | anyone know if someone makes an SIP cordless phone with a PoE charger? Or failing that one with a charger that has a wall mount |
16:44.25 | [TK]D-Fender | haryv: Sounds like they never knew what hey wer doing |
16:45.40 | haryv | I dont know the companies history but did have a active working biz. Now it is just down to the owener and linux developer. |
16:46.14 | haryv | I think Shaw communications and telus killed off most voip providers |
16:46.17 | kaldemar | nohup_: for example linksys spaXXXX ATA |
16:46.20 | [TK]D-Fender | haryv: Sounding more accurate by the moment |
16:46.31 | haryv | knocked down there prices so low, the small companies could not compete |
16:47.11 | nohup_ | kaldemar: thank you! :) i'll be ebaying then :) |
16:47.48 | [TK]D-Fender | nohup_: Don't With your luck you'll end up buying a locked POS and waste more money on shipping |
16:48.07 | Jymm | LOL, that's cold man, cold. |
16:48.31 | nohup_ | [TK]D-Fender: yeah.. i noticesd some of them having a tag saying "UNLOCKED", so i'll be looking for only those :) |
16:48.53 | [TK]D-Fender | nohup_: An playing the odds on it all the while |
16:49.10 | Jymm | btw, is PAP2T still in production? I can't seem to find any at retailers. Seems like they might have been transitioned to Cisco Small Business |
16:49.28 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com) |
16:49.57 | Jymm | Well, PAP2T-NA specifically |
16:51.38 | Qwell | Jymm: they've been replaced, yes |
16:51.57 | Jymm | Qwell: oh noes! With what? |
16:52.03 | Qwell | chickens. |
16:52.15 | Qwell | no, SPA2102 I think |
16:53.05 | Jymm | I can't find that either it seems. Frys has ZERO non-branded VoIP gear - any brand. MicroCenter seems simular. |
16:53.20 | Jymm | same goes for costco online |
16:53.59 | [TK]D-Fender | Qwell: No, SPA-2102 was in production at the same time, and they are different |
16:54.39 | Jymm | Well, Staples Online has PAP2T - bastidges |
16:55.07 | [TK]D-Fender | staples = LOCKED |
16:55.15 | [TK]D-Fender | Retailers don't carry unlocked shit |
16:55.21 | Jymm | [TK]D-Fender: To whom? |
16:55.31 | [TK]D-Fender | Because they don't want to deal with idiot end users who have no clue what they're doing |
16:56.08 | Jymm | [TK]D-Fender: Who or what are they locked to/againest? |
16:56.34 | *** join/#asterisk dysinger (n=tim@71-20-35-99.war.clearwire-wmx.net) |
16:58.17 | *** join/#asterisk generalhan (n=asd@about/windows/staff/generalhan) |
16:58.37 | *** join/#asterisk xuser (n=xuser@unaffiliated/xuser) |
16:58.51 | bmoraca | whatever provider staples sells |
16:59.16 | bmoraca | www.voiplink.com sells unlocked PAP2Ts |
16:59.33 | Jymm | http://www.staples.com/Linksys-PAP2T-VoIP-Internet-Phone-Adapter-with-2-Ports/product_766549 |
16:59.47 | Jymm | No provider listed |
17:00.52 | *** join/#asterisk dymaxion (n=dymaxion@89.248.128.108) |
17:01.06 | [TK]D-Fender | Jymm: "Good luck" all I can say |
17:02.15 | Jymm | [TK]D-Fender: No, I wasn't doubting you, I just dont know and I'm asking. I know that PAP2T-NA *IS* unlocked, but when a provider isn't stated and no -NA, what would be locked? |
17:02.16 | thehar | we get our pap2ts from 800voipstore |
17:03.20 | Jymm | maybe I need the definition of "locked" in this respect. |
17:03.55 | Jymm | If I tried to hack a vonage box, sure I get it. |
17:04.11 | Zuchmir2 | just called staples: they say it's unlocked |
17:04.32 | garymc | anyone here know why i cant add music mp3 to my music on hold? its freepbx i know but nobody there knows why |
17:04.32 | dymaxion | Hi, is there anyone here usingi the Atcom IP04 for their asterisk PBX ? |
17:04.57 | Jymm | Zuchmir2: cool, ty |
17:05.39 | [TK]D-Fender | Zuchmir2: Yes, and Stapes are real techies... |
17:06.28 | *** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler) |
17:06.28 | *** mode/#asterisk [+o angler] by ChanServ |
17:06.46 | Jymm | [TK]D-Fender: Ok, what could it be locked to? |
17:08.48 | [TK]D-Fender | Jymm: Vonage Broadvoice, etc |
17:09.04 | Zuchmir2 | on the webpage they don't mention it being locked, the rep i spoke to spoke to a techie before responding |
17:09.26 | Zuchmir2 | wouldn't they want to specify that on their website? |
17:09.47 | [TK]D-Fender | Zuchmir2: because many are lazy with their descriptions |
17:09.57 | [TK]D-Fender | Zuchmir2: And many "techies" are morons. |
17:11.16 | *** join/#asterisk dysinger_ (n=tim@71.20.35.99) |
17:13.02 | Zuchmir2 | is the PAP2T better than HT-502? |
17:15.43 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
17:15.54 | *** join/#asterisk dysinger_ (n=tim@71-20-35-99.war.clearwire-wmx.net) |
17:16.12 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
17:16.26 | k4tanaLINUX | ~billing |
17:17.11 | [TK]D-Fender | ~gs |
17:17.12 | infobot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
17:18.44 | Zuchmir2 | quality of: build, fw, sound? or all of the above? |
17:19.31 | [TK]D-Fender | Zuchmir2: AOTA |
17:20.07 | Zuchmir2 | so the PAP2T is better |
17:23.53 | *** join/#asterisk outtolunc (n=me@c-98-248-96-110.hsd1.ca.comcast.net) |
17:27.36 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
17:29.14 | ariel_ | Afternoon folks |
17:30.25 | momelod | greetings |
17:35.51 | cusco | asterisk is taking 100% cpu |
17:35.56 | cusco | how can I find out why? |
17:36.09 | cusco | please |
17:37.34 | ariel_ | Maybe via tops you might find out which process is taking up your cpu. |
17:37.56 | cusco | I found |
17:38.05 | cusco | htop shows 2 processes taking huge amounts |
17:38.28 | *** join/#asterisk seanmh (n=johndoe@207.114.199.107) |
17:38.41 | dustybin | i think its time to get asterisk up and running |
17:39.23 | hardwire | do it do it now |
17:39.41 | dustybin | ok |
17:39.44 | cusco | all calls are having cuts |
17:40.02 | dustybin | there are so many compile optios that im lost in what i need / dont need |
17:40.06 | dustybin | *options |
17:40.07 | ariel_ | all calls are having cuts, oh due to 100% restart now |
17:40.20 | [TK]D-Fender | dustybin: Start with the defaults |
17:41.11 | dustybin | ok |
17:41.58 | dustybin | [TK]D-Fender: i will just do a ./configure && make && make install |
17:42.02 | cusco | asterisk machine got really slowwwwwwww |
17:42.07 | ariel_ | is a bit happy, he may have to take a trip to setup install/fix a system overseas.....Athens...I have not been there yet... |
17:42.14 | cusco | and now i coud reattach a cli ai could read LOTS of: |
17:42.15 | cusco | utils.c:1126 ast_carefulwrite: write() returned error: Broken pipe |
17:43.51 | dustybin | its compiling :D |
17:44.03 | dustybin | 1.6.1.5 |
17:44.41 | dustybin | im tempted to setup a softphone until i get a proper voip external |
17:44.44 | *** join/#asterisk Defraz (n=T0tal@corp.fuzecore.com) |
17:46.48 | [TK]D-Fender | dustybin: should do that regardless |
17:47.38 | *** join/#asterisk [jmc] (n=[jmc]@93-45-200-133.ip104.fastwebnet.it) |
17:47.39 | dustybin | [TK]D-Fender: are softphones used for testing? |
17:47.59 | [jmc] | hi guys :) |
17:48.01 | [TK]D-Fender | dustybin: They're used for whatever you use them for |
17:48.06 | dustybin | ace |
17:48.16 | dustybin | feels excited |
17:48.18 | *** join/#asterisk fofware (n=Client@host3.190-230-38.telecom.net.ar) |
17:49.25 | dustybin | my god, asterisk required no deps |
17:50.04 | russellb | just libncurses-dev |
17:50.04 | russellb | heh |
17:50.32 | Qwell | and gcc |
17:50.33 | Qwell | :p |
17:50.38 | dustybin | i just did: make samples |
17:50.45 | [jmc] | and libc6-dev |
17:50.49 | dustybin | there is another one, i missed it, something like make progdocs? |
17:50.59 | Qwell | dustybin: you don't need that |
17:51.13 | dustybin | ok |
17:51.23 | dustybin | well thats it, asterisk is installed |
17:51.32 | dustybin | now i need to find a decent softphone what runs on os x |
17:51.51 | Qwell | x-lite or whatever they call it these days |
17:51.58 | [jmc] | hmm |
17:52.20 | [jmc] | QuteCom? |
17:53.33 | [jmc] | hey does anybody know how a Linksys 3102 works? |
17:53.46 | [jmc] | there's something I don't get |
17:54.04 | ariel_ | 3102 there are allot of info on that unit it's a good fxo/fxs ata |
17:54.24 | [jmc] | yes |
17:54.32 | [jmc] | but I'm not expert with its configuration |
17:54.42 | [jmc] | and I'm doing something bad I think |
17:54.45 | dustybin | x-lite os x installed :D |
17:55.07 | [jmc] | I mean, I have an analog phone attached to the "Line" entrance, the FXS one |
17:55.21 | dustybin | now i need to buy a pay-as-you VOIP phone account |
17:55.23 | [jmc] | and the phone cable in the FXO entrance |
17:55.24 | Qwell | FXS isn't a line... |
17:55.46 | [jmc] | I said entrance |
17:55.51 | [jmc] | how do you call it? :D |
17:55.53 | Qwell | you also said "Line" |
17:56.04 | Qwell | does it say "Line" on the jack? |
17:56.11 | dustybin | im going to buy some VOIP credits from these people: http://www.voiptalk.org/products/IAX+PSTN+Call+Credit |
17:56.25 | [jmc] | err, "Phone" I'm sorry |
17:56.40 | [jmc] | "Line" is the other one, you're right |
17:57.12 | [jmc] | however, what I'm trying to get is that I want to call on the phone line from my Asterisk PBX |
17:57.28 | [jmc] | and have my phone working just like it did before |
17:57.33 | [jmc] | I receive calls perfectly |
17:57.45 | [jmc] | but I can't call anymore |
17:58.16 | [jmc] | when I dial something from the analog phone, it tries to call an extension of my PBX |
17:59.20 | cusco | is it possible to compile asterisk with proffiling enabled? |
18:00.33 | spck | hi guys |
18:00.43 | [jmc] | here's what the 3102 says on my syslog |
18:00.46 | [jmc] | http://www.nopaste.com/p/a69Yad8RU |
18:01.23 | [jmc] | obviously 192.168.0.2 is the IP of my Asterisk PBX |
18:02.05 | [TK]D-Fender | cusco: Profiling what? |
18:02.14 | [TK]D-Fender | [jmc]: What does ASTERISK say? |
18:02.28 | spck | i'm running a redundant setup with two asterisk boxes and redfone fonebridge between, currently i have an issue where one box will provision the spans and bring them up just fine, but the other box refuses too |
18:03.17 | [jmc] | [tk] I'll tell you in a moment, but the point is that asterisk should NOT say anything, I'd like it to dial on my PSTN line, not via VoIP |
18:03.24 | [TK]D-Fender | spck: almost noone uses TDMOE. |
18:03.40 | *** join/#asterisk SirColin (n=SirColin@my83-216-68-241.mynow.co.uk) |
18:03.46 | [TK]D-Fender | [jmc]: Oh... nothing to do with us... go check out www.voxilla.com 's forums |
18:03.48 | dustybin | i have signed up for a free acount with a VOIP provider, i now have a incoming number, a SIP ID and a password, is that all the details i need for when i configure asterisk? |
18:03.51 | *** part/#asterisk GameGamer43 (n=GameGame@h69-129-142-83.mdsnwi.tisp.static.tds.net) |
18:04.22 | [TK]D-Fender | dustybin: the host to send calls to/ auth calls from. Allowed codecs, etc |
18:04.29 | dustybin | ok |
18:04.57 | spck | i'm starting to realize why |
18:05.18 | dustybin | [TK]D-Fender: i have [SIP NUMBER]@voiptalk.org |
18:05.38 | [jmc] | [tk] I've been doing that for two days, and I can't get it to work, I'm not here to bother you all... as a last resort I tried to see if someone had my same experience. If you know a dedicated IRC channel I'll be happy to go bother them instead! :) |
18:06.16 | spck | is there anyway to force the spans to come up? |
18:07.05 | ariel_ | spck: red-fone has a good support department |
18:07.20 | *** join/#asterisk oej (n=olle@ns.webway.se) |
18:08.04 | *** join/#asterisk wayne_r (n=wayne_r@rrcs-24-173-187-234.sw.biz.rr.com) |
18:08.09 | spck | ariel: debateable |
18:08.19 | [jmc] | thanks, anyway |
18:08.24 | [TK]D-Fender | dustybin: Go look up some config samples for them |
18:09.17 | ariel_ | [jmc]: trick is the line (fxo) don't register that with the asterisk box. |
18:10.31 | dustybin | [TK]D-Fender: aye thanks |
18:11.03 | [jmc] | ok ariel_, thanks, I'll do some tests and see what happens ([tk]'s right, this is not really the right place to talk about it, I'm sorry) |
18:11.07 | [jmc] | ;) |
18:11.24 | dustybin | [TK]D-Fender: http://www.voiptalk.org/products/iaxconfig.html |
18:11.40 | hesco | if called from an AGI script, where does a print STDERR 'debug message'; go? |
18:12.03 | dustybin | extensions.conf v iax.conf |
18:12.15 | dustybin | likes the sound of iax.conf |
18:12.37 | *** join/#asterisk xuser_ (n=xuser@unaffiliated/xuser) |
18:13.16 | *** join/#asterisk cesar_CR (n=cesar@201.192.86.30) |
18:15.03 | dustybin | jesus lord, i have never seen so many .conf files in my life |
18:15.42 | dustybin | anybody would think asterisk is _serious_ stuff :P |
18:15.56 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
18:16.11 | wayne_r | I'm trying to spec out what I need for my first attempt at an asterisk installation. What I'd like to do is use four Linksys SPA3102s for a total of 4 FXO and FXS ports -- is it possible to route incoming calls on the FXO ports to asterisk and route them to the appropriate FXS port (on a different SPA3102)? |
18:16.32 | bmoraca | yes |
18:16.36 | bmoraca | they're all just sip peers |
18:16.50 | wayne_r | bmoraca: that's good news. :) |
18:17.02 | bmoraca | i personally would never do that...but it's possible |
18:17.03 | [TK]D-Fender | wayne_r: and thats a horrible thing to do for a business |
18:17.05 | cusco | [TK]D-Fender: so I can use gprof |
18:17.19 | [TK]D-Fender | wayne_r: I highly recommend your getting a decent FXO solution |
18:17.20 | cusco | to tell me where it may be spending cpu |
18:17.51 | wayne_r | [TK]D-Fender: why is the 3102 not a good solution? |
18:18.08 | [TK]D-Fender | SPA-3102 is ok for personal lo usage and failover, not primary business |
18:18.46 | [TK]D-Fender | wayne_r: Poorer EC, no fine grained interface control, more devices to configure, inability to pool channels, etc |
18:18.48 | wayne_r | hmmm well that's no fun |
18:19.30 | bmoraca | wayne_r: 4 of those is 2/3rds of the way to a proper FXO card anyway |
18:19.57 | Qwell | wayne_r: really aught to consider SIP phones too |
18:20.00 | wayne_r | the goal is running it in vmware and avoiding buying a new machine |
18:20.13 | *** join/#asterisk chandoo (n=chandoo@ool-4353b978.dyn.optonline.net) |
18:20.27 | wayne_r | is there something similiar that is better and would work without needing to be physically installed on a box? |
18:20.44 | bmoraca | i hope you're not trying to use something like VMWare Server or Workstation... |
18:20.56 | wayne_r | nah esxi |
18:20.59 | Qwell | bmoraca: player ;) |
18:21.59 | bmoraca | wayne_r: i've used those SPA3102s in that context, and it was a pain in the ass. i'd very much advise against it. use a sip provider and SIP phones if you absolutely have to run it in VMware. |
18:23.02 | bmoraca | they work well SIP to FXS, but the FXO to SIP is clunky at best |
18:23.10 | *** join/#asterisk moy (n=moy@bas1-unionville55-1177733667.dsl.bell.ca) |
18:23.42 | [TK]D-Fender | wayne_r: AudioCodes or Mediatrix analog gateway |
18:23.56 | wayne_r | bmoraca: fair enough |
18:24.02 | wayne_r | [TK]D-Fender: i'll have a look, thanks |
18:25.58 | bmoraca | it definitely works, but i'd prefer not to have to set that up again...hosted PBX customer had two analog lines at their location that they could not get rid of or move (they were extensions on a hospital's phone system) so i had little choice...but i definitely don't want to do it again |
18:30.44 | giovani | has anyone here used Metrostat for their CNAM service? |
18:35.45 | *** join/#asterisk bluOxigen (n=asad@static-host119-73-69-38.link.net.pk) |
18:36.30 | *** join/#asterisk De_Mon (i=de_mon@fl-69-34-134-91.dhcp.embarqhsd.net) |
18:39.17 | *** join/#asterisk dymaxion (n=dymaxion@89.248.128.108) |
18:41.37 | linagee | if i'm using a fixed SIP jitter buffer, how do I know where to set the buffer length? |
18:42.14 | garymc | [TK]D-Fender : http://pastebin.ca/1550604 |
18:42.21 | garymc | I sussed out what you was saying |
18:42.43 | [TK]D-Fender | garymc: except for "SHOW ME THE FAILURE" |
18:43.23 | carrar | No such command 'SHOW ME THE' (type 'help' for help) |
18:43.31 | garymc | yeah no such command |
18:43.34 | garymc | :P |
18:43.41 | *** part/#asterisk SirColin (n=SirColin@my83-216-68-241.mynow.co.uk) |
18:43.55 | carrar | hahah |
18:44.37 | carrar | Russel really needs to implement a command like that |
18:44.40 | carrar | +l |
18:44.49 | garymc | I dont know how to show the failure |
18:44.52 | garymc | :( |
18:44.58 | carrar | then nothing is broken? |
18:45.26 | garymc | but it aint working |
18:47.39 | garymc | does [TK]D-fender just blank you when hes had enough of you? |
18:47.39 | carrar | How do you know it' |
18:47.43 | carrar | s not working |
18:47.55 | garymc | Cos it says an error in the browser |
18:48.03 | garymc | also when i hold a call theres no music |
18:48.12 | garymc | :( |
18:48.22 | carrar | i don't see any error |
18:48.25 | garymc | Im thinking its a freepbx issue |
18:48.41 | carrar | Did you ask the people in the channel that deal with your setup? |
18:49.03 | garymc | I pressed reload config after it and i get loads of info so much so that the first lot is missing from the terminal window |
18:49.08 | garymc | Yes |
18:49.16 | garymc | Nobody seems able to help me |
18:49.18 | garymc | :( |
18:49.24 | garymc | its nearly my bedtime too |
18:49.32 | carrar | install asterisk from source |
18:49.39 | carrar | toss that freepbx |
18:50.39 | garymc | others seem to be doing ok with freepbx |
18:50.44 | garymc | just not me |
18:52.58 | carrar | But this is not a FreePBX channel |
18:53.08 | [TK]D-Fender | [14:48]<garymc>also when i hold a call theres no music <-- why am I not seeing a call with MoH failing? Why don't I see a dump of your MoH folder? |
18:53.36 | [TK]D-Fender | garymc: And why have you spilled into here? |
18:53.42 | carrar | heh |
18:53.51 | carrar | Need a bigger dam! |
18:54.01 | garymc | im usin chatzilla on pc easy to do by accident |
18:54.32 | carrar | yeah I sometimes accidently get stuck in the microsoft windows 95 channel |
18:54.37 | carrar | no idea how |
18:54.38 | [TK]D-Fender | garymc: I use Chatzilla as well. Doesn't pose any problems |
18:54.56 | garymc | yeah im quite new to irc |
18:55.00 | *** join/#asterisk bluOxigen (n=asad@static-host119-73-67-71.link.net.pk) |
18:55.06 | *** join/#asterisk wonderworld (n=w@ip-62-143-22-226.unitymediagroup.de) |
18:55.07 | garymc | im also using a mouse pad, pain in the arse |
18:55.17 | carrar | I use a desk |
18:56.19 | garymc | well i put a call on hold and nothing else appears in the cli |
18:56.22 | garymc | I like CLI |
18:56.59 | *** join/#asterisk IBC_jkenney (n=jkenney@ip65-44-169-66.z169-44-65.customer.algx.net) |
18:57.03 | carrar | What did you set your verbosity too? |
18:57.33 | bmoraca | yeah, you should have a message in CLI even for successful MOH |
18:59.06 | IBC_jkenney | hey i need some assistance if you all have the inclination. I have a problem with a Wildcard AEX2400 Board 1 when someone dials into the card and we have the card plugged into a bank of modems we get a RX TX line problem |
18:59.17 | IBC_jkenney | is there away to better set this card up for data |
18:59.23 | IBC_jkenney | we do not really do any speech on it |
18:59.56 | *** join/#asterisk murdock_ut (n=chatzill@mail.kimballequipment.com) |
19:00.00 | *** join/#asterisk jong2 (n=chatzill@65.100.10.89) |
19:00.22 | bmoraca | IBC_jkenney: no. and you're probably not going to get much support doing that. if you need a fax server or a dialin aggregator, get one. used AS5200s are pretty cheap on ebay. |
19:00.24 | IBC_jkenney | the inbound calls come from a PRI |
19:00.26 | *** join/#asterisk sasargen (n=chatzill@173-124-140-145.pools.spcsdns.net) |
19:01.05 | IBC_jkenney | bmoraca our inbound calls are coming in on a PRI and we are using the AEX2400 for pots line breakout |
19:01.18 | IBC_jkenney | there is no sip or "voip involved" |
19:01.28 | bmoraca | did i say there was? |
19:01.57 | bmoraca | either way, though, they're routing through Asterisk, which is not really equipped for what you're trying to do. that's why I recommended getting a real access server, such as an AS5200 |
19:02.15 | *** join/#asterisk oej (n=olle@ns.webway.se) |
19:02.31 | joako | I know this is off-topic as hell, but: does anyone know where I can contact to find a LOCAL Nokia service center? |
19:02.48 | carrar | GOOGLE! |
19:03.20 | carrar | http://tinyurl.com/nax7xj |
19:03.20 | rene- | hehe |
19:03.56 | joako | Hell I called Nokia, the rep told me there were local service centers but that they could not give out that info |
19:03.57 | carrar | You can play with the working |
19:04.02 | carrar | maybe change stuff |
19:04.03 | carrar | maybenot |
19:04.08 | carrar | wording |
19:04.13 | carrar | heh |
19:04.44 | joako | carrar: That took me to a page that says I need to enable JavaScript... why would i need a script in my coffee? I am confused.... :) |
19:05.20 | [TK]D-Fender | IBC_jkenney: Could have echo, clocking issues, gain, etc... all sorts |
19:05.41 | [TK]D-Fender | IBC_jkenney: is has always been recommended to keep data devices as far away from * as possible |
19:06.52 | bmoraca | IBC_jkenney: one possible solution is to go entirely Sangoma...as they do support FAX services from PRI to FXS with a timing sync cable between two cards with echo cancellers. as far as I know, no such similar claim exists for Digium cards. |
19:07.04 | *** join/#asterisk d00gster (n=doughant@77.30.9.36) |
19:07.09 | bmoraca | that is manufacturer supported, so you'd have some recourse when things don't work |
19:07.14 | bmoraca | but i still wouldn't recommend it |
19:07.35 | *** join/#asterisk qdk (n=qdk@0x573d8d8b.bynqu1.dynamic.dsl.tele.dk) |
19:07.44 | carrar | I use PRI -> Digium T1 card -> ADIC600 channel bank -> FAX |
19:07.46 | carrar | that works |
19:09.22 | *** join/#asterisk FlaPer87 (n=FlaPer87@unaffiliated/flaper87) |
19:09.38 | FlaPer87 | hey guys, is it possible to write C++ modules for asterisk? |
19:09.54 | bmoraca | it hasn't been made clear exactly what kind of modems are being dialed, so it's plausible that the person's attempting to terminate dialup calls |
19:10.20 | FlaPer87 | where can I find tutorials for creating Asterisk modules? |
19:10.29 | garymc | anyone know why the CLI is not showing my internal clal getting put on hold? |
19:10.35 | garymc | *call |
19:11.11 | bmoraca | garymc: core set verbose 10 |
19:11.21 | garymc | yep i did |
19:11.33 | joako | FlaPer87: take a look at app_skel.c it should be part of your Asterisk source code |
19:11.55 | FlaPer87 | joako ok, thanks |
19:12.04 | bmoraca | garymc: and you still haven't pastebinned an actual call. |
19:12.14 | [TK]D-Fender | FlaPer87: And there a few googleable tutorials out there as well |
19:12.18 | garymc | i have in another channel |
19:12.25 | [TK]D-Fender | bmoraca: I saw a (useless) sample |
19:12.26 | joako | garymc: FWIW I don't ever recall any output (well except perhaps SIP debug) that shows a call being placed on hold, but if you set your verbosity > 2 it should indicate when the music on hold is started |
19:12.36 | bmoraca | that doesn't particularly help me very much, now does it? |
19:12.49 | [TK]D-Fender | bmoraca: Didn't help me much either |
19:12.49 | Naikrovek | checks out starfish pbx |
19:13.02 | [TK]D-Fender | ~happyfunclownpbx |
19:13.15 | [TK]D-Fender | ~happyclownpbx |
19:13.16 | infobot | [HappyClownPBX] is currently in closed beta, approaching 12GB in size, uses Asterisk for it's core, it pwns, is also now compatible with the Diahatsumashiniriki Keyotason 200LP-A11 SIP phone |
19:13.19 | [TK]D-Fender | :) |
19:13.28 | *** join/#asterisk jtodd (i=uancftet@ns.fox-den.com) |
19:13.28 | *** mode/#asterisk [+o jtodd] by ChanServ |
19:13.58 | *** join/#asterisk MindTheGap (n=MindTheG@187.20.141.72) |
19:14.05 | bmoraca | Naikrovek: looks too much like a linksys interface. eck |
19:15.06 | garymc | ok |
19:15.21 | garymc | http://pastebin.ca/1550636 |
19:15.46 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
19:15.52 | *** join/#asterisk Chromis (n=millsu2@mail.serverplus.com) |
19:16.04 | garymc | see what you make of that one, as I made an internal call and put it on hold. Had them on Loud speaker and could hear myself coming through other phone. But no music when on hold. Red light was flashing on Ip330 phone |
19:16.21 | garymc | when was holding as supposed too |
19:16.36 | garymc | that was also a verbose setting of 10 or higher |
19:16.53 | bmoraca | garymc: for the love of god don't leave the freepbx "status" page up when you're doing these debugs...it's difficult enough already to debug it without that crap in the way |
19:17.19 | [TK]D-Fender | garymc: How did you put them on hold, and what device are you using? |
19:17.25 | garymc | whats that, its only a couple of lines, just thought id let you see my verbose setting etc |
19:17.32 | garymc | Polycom Ip330 |
19:17.34 | *** part/#asterisk korihor (n=korihor@190.77.83.180) |
19:17.37 | garymc | i pressed the hold button |
19:17.48 | garymc | I can also press hold then pick another line and phone them again |
19:17.59 | [TK]D-Fender | garymc: What's the other device? |
19:18.03 | *** join/#asterisk maour (n=gnu@unaffiliated/maour) |
19:18.03 | bpgoldsb | Wait, someone is actually making a product named happyclownpbx? |
19:18.05 | garymc | Polycom IP 330 |
19:18.11 | [TK]D-Fender | garymc: both? |
19:18.14 | garymc | yes |
19:18.22 | [TK]D-Fender | garymc: Something is very wrong |
19:18.30 | garymc | huuurraaahhh |
19:18.37 | spck | heh, vs starfishpbx? |
19:18.37 | garymc | I knew there was something weird going on |
19:18.47 | [TK]D-Fender | garymc: do another call. Place on hold, place 2nd call |
19:18.53 | garymc | ok |
19:20.20 | FlaPer87 | is it possible to compile an asterisk module without compilling the whole asterisk? |
19:20.29 | *** join/#asterisk rickross (n=rickross@supporter/active/rickross) |
19:21.22 | garymc | ok here we go, so much stuff i cant get to the beginning |
19:21.35 | garymc | http://pastebin.ca/1550642 |
19:21.39 | *** join/#asterisk jkroon (n=jkroon@dsl-240-169-69.telkomadsl.co.za) |
19:23.28 | [TK]D-Fender | garymc: while on hold do : sip show channels |
19:23.44 | garymc | the phones? |
19:23.44 | [TK]D-Fender | garymc: and prior do : core set debug 10 |
19:23.51 | [TK]D-Fender | garymc: and prior do : core set verbose 10 |
19:23.56 | *** join/#asterisk SirColin (n=SirColin@my83-216-68-241.mynow.co.uk) |
19:23.58 | garymc | it was verbose 10 |
19:24.05 | garymc | and the phones show channelas |
19:24.06 | [TK]D-Fender | garymc: Confirma dn do the other |
19:24.13 | garymc | ?? |
19:24.27 | [TK]D-Fender | garymc: Confirm and do the other |
19:24.42 | garymc | i set verbose to 10 |
19:24.47 | garymc | its already been done |
19:24.56 | garymc | and the phones show the channels |
19:25.11 | garymc | ??? if you mean something else i dont know what that is |
19:26.59 | Chromis | I keep getting errors reading: "pgsql_log: cdr_pgsql: Reason: ERROR: column "calldate" specified more than once". It has worked fine for over 2 weeks and just started doing this a few hours ago. |
19:27.22 | garymc | ok well i should be getting home to the missus now, been here for hours and hours. its 20:30 here so[TK]D-Fender you reckon you could have a think about it for me and i will grab you 2morow? |
19:28.22 | garymc | yeah as soon as i pres hold nothing happens |
19:28.25 | garymc | in the cli |
19:29.14 | Chromis | Asterisk is trying to insert duplicate row into the dtabase for some reason. "INSERT INTO cdr ("calldate","clid","src","dst","dcontext","channel","dstchannel","lastapp","lastdata","duration","billsec","disposition","amaflags","uniqueid","calldate","clid","src","dst","dcontext","channel","dstchannel","lastapp","lastdata","duration","billsec","disposition","amaflags","uniqueid") VALUES . . ." |
19:30.19 | rickross | is there an easy way to watch/log commands that asterisk receives via the manager interface on port 5038? |
19:30.28 | *** join/#asterisk netpro25_ (n=mmanning@64-238-176-253.ksg.apt.gru.net) |
19:30.36 | [TK]D-Fender | rickross: there is not AMI debug mode |
19:30.49 | [TK]D-Fender | rickross: Best I think you can do is spy on the port via wireshark, etc |
19:31.12 | garymc | tk? |
19:31.15 | rickross | [TK] - thanks |
19:31.15 | garymc | any thoughts? |
19:31.37 | [TK]D-Fender | garymc: I just asked you to show me 3 more things, and I've got nothing. |
19:31.39 | rickross | I guess maby ngrep would let me see this stuff? |
19:31.41 | [TK]D-Fender | garymc: there's a though |
19:31.43 | [TK]D-Fender | t |
19:31.51 | garymc | i showed you them |
19:31.53 | [TK]D-Fender | rickross: packets are packets |
19:32.07 | rickross | I don't know wireshark |
19:32.13 | [TK]D-Fender | garymc: no, you didn't |
19:32.19 | garymc | i thought i did |
19:32.29 | garymc | you asked if i set it to verbose 10 which i did |
19:32.32 | [TK]D-Fender | garymc: Sub-contract <- |
19:32.49 | garymc | thanks |
19:33.05 | [TK]D-Fender | garymc: I also asked for a "sip show channels" while ON HOLD, and enable CORE DEBUG |
19:33.28 | [TK]D-Fender | Swear to God some people need to learn to #*&$ing read |
19:33.45 | KyleK | odd |
19:33.54 | KyleK | smartypants asterisk addons ignores includedir |
19:36.19 | *** join/#asterisk dominic_ (n=chatzill@207.61.107.242) |
19:36.46 | *** part/#asterisk SirColin (n=SirColin@my83-216-68-241.mynow.co.uk) |
19:37.46 | dominic_ | anyone know why I would be getting this warning :[Aug 31 14:48:30] WARNING[22463] chan_sip.c: Asked to transmit frame type 64, while native formats is 0x4 (ulaw)(4) read/write = 0x4 (ulaw)(4)/0x4 (ulaw)(4) |
19:39.42 | dominic_ | looks like a codec issue... but 'frame type 64' ?? |
19:40.18 | Chromis | hesco: it looks like about a year ago you had the same problem I am having. Did you ever find a solution? |
19:40.29 | *** join/#asterisk MindTheGap (n=MindTheG@mail.lpj.com.br) |
19:40.50 | carrar | http://www.voip-info.org/wiki/view/Asterisk+codecs |
19:41.36 | dominic_ | ahh thx |
19:41.47 | *** join/#asterisk [TK]D-Fender (n=chatzill@216.191.106.162) |
19:42.10 | [TK]D-Fender | Stupid laptop |
19:42.14 | carrar | YEAH |
19:42.23 | carrar | No so SMRT |
19:42.57 | dustybin | what port will my x-lite phone use to auth my asterisk server? |
19:43.13 | KyleK | to auth? |
19:43.16 | carrar | UDP 5060? |
19:43.27 | dustybin | i think my firewall is blocking, so im going to telnet |
19:43.54 | dustybin | <PROTECTED> |
19:43.54 | dustybin | eeek |
19:43.54 | KyleK | UDP != TCP |
19:43.55 | Qwell | You can't telnet to UDP |
19:44.00 | dustybin | oh yes of course |
19:44.06 | lirakis | dust netcat |
19:44.12 | carrar | nmap -sU -p 5060 1.2.3.4 |
19:44.12 | lirakis | = friend |
19:44.45 | dustybin | i will check on this asterisk server using, netstat -natp | grep 5060 |
19:45.02 | dustybin | nothing open |
19:45.19 | carrar | *sigh* |
19:45.33 | lirakis | dustybin, drop the t |
19:45.37 | dustybin | ok |
19:45.41 | lirakis | dustybin, netstat -nap | grep 5060 |
19:45.53 | dustybin | yep, nothing open |
19:46.08 | dustybin | there are 100's of .confs |
19:46.12 | lirakis | dustybin, ... uhh... i think you are ahead of yourself here |
19:46.14 | lirakis | go read |
19:46.17 | lirakis | ~read |
19:46.18 | infobot | ACTION reads Lord of the rings |
19:46.24 | lirakis | err damn |
19:46.27 | lirakis | ~thebook |
19:46.28 | infobot | somebody said thebook was Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://astbook.asteriskdocs.org |
19:46.38 | dustybin | i bought the o-reilly book |
19:46.45 | dustybin | i will take a read i think |
19:46.47 | lirakis | dustybin, Thats Awsome !! ME TOO |
19:46.51 | lirakis | go read it |
19:46.52 | dustybin | asterisk wis not a 5 minute job.... |
19:46.55 | dustybin | its a 5 month job :D |
19:47.00 | dustybin | *year |
19:47.01 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
19:47.11 | lirakis | dustybin, after you read a bit - you will be fine |
19:47.14 | dustybin | ok |
19:47.22 | carrar | asterisk/linux/tcpip |
19:47.37 | *** join/#asterisk WHYS (n=drumm@137.28.94.209) |
19:48.22 | jong2 | if you have dial plan nightmare, try http://www.safisystems.com/ |
19:49.55 | [TK]D-Fender | dustybin: cat /etc/asterisk/modules.conf |
19:50.35 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
19:51.15 | dustybin | [TK]D-Fender: http://paste.debian.net/45490/ |
19:51.42 | [TK]D-Fender | dustybin: netstat -an|grep 5060 |
19:51.55 | [TK]D-Fender | dustybin: then at * cli : sip show peers |
19:52.21 | dustybin | ohhh it is open |
19:52.27 | dustybin | udp 0 0 192.168.1.10:5060 0.0.0.0:* |
19:52.33 | *** join/#asterisk oej_ (n=olle@ns.webway.se) |
19:53.21 | dustybin | im not sure how to execute the * cli command |
19:53.32 | [TK]D-Fender | dustybin: Fine, its listening. |
19:53.42 | [TK]D-Fender | dustybin: Congrats. |
19:53.45 | dustybin | :D |
19:53.47 | lirakis | lol |
19:53.50 | [TK]D-Fender | NEXT!@!@!!!@ (c) BKW |
19:54.06 | bmoraca | asterisk -rx 'command' |
19:54.51 | jong2 | gee gmail down.. |
19:55.12 | dustybin | http://paste.debian.net/45491/ |
19:56.11 | dominic_ | carrar: [Aug 31 14:48:30] WARNING[22463] chan_sip.c: Asked to transmit frame type 64, while native formats is 0x4 (ulaw)(4) read/write = 0x4 (ulaw)(4)/0x4 (ulaw)(4) |
19:56.39 | dominic_ | would mean that someone is asking for slin and I am providing ulaw? |
19:56.52 | *** join/#asterisk jcape (n=jcape@209.120.251.81) |
20:01.51 | KyleK | type 64 is sln? then yea it sounds like it |
20:02.46 | [TK]D-Fender | dustybin: OK, 1 SIP device configured, not yet registered and SIP is listening |
20:05.40 | dominic_ | I trying to find out who is asking for slin |
20:06.10 | dominic_ | I put disallow=all |
20:06.17 | dominic_ | allow=ulaw |
20:06.23 | dominic_ | everywhere |
20:06.46 | dominic_ | I googled allow=slin and only got bodybuilding stuff |
20:07.08 | KyleK | turn on sip debugging? |
20:08.05 | dominic_ | theres lots of traffic on the server |
20:08.12 | dominic_ | so I wont see the output |
20:08.15 | dustybin | nc -u 192.168.1.10 5060 <-- is this the correct command to check a udp port using netcat? |
20:08.35 | lirakis | dustybin, no |
20:08.51 | dustybin | nc -u -s 192.168.1.10 -p 5060 |
20:08.54 | lirakis | dustybin, save yourself and a lot of people here some headache ... go read the first few chapters please |
20:08.57 | *** join/#asterisk TimToady_ (n=moi@adsl209-7.kln.forthnet.gr) |
20:09.08 | dustybin | ok |
20:09.30 | dominic_ | yeah I will turn it on and try to see if I find something |
20:10.56 | kfife | DTMF doesn't break DISA dialtone? Exten => *67 invokes DISA() with null CLID. Connected to asterisk via private-loop PRI from a legacy PBX. Asterisk 1.6. Any ideas? Do I need to relax DTMF? |
20:11.10 | KyleK | is gmail down? |
20:12.14 | bmoraca | appears to be |
20:12.29 | dustybin | [192.168.1.10] 5060 (sip) open :D |
20:12.40 | dustybin | now reads book |
20:12.57 | dustybin | and sips some tea :D |
20:13.29 | kfife | The GMail IMAP interface seems to be online however. |
20:14.00 | dominic_ | thx everyone |
20:14.30 | timeshell_atwork | Hey, I got a new thing... |
20:14.31 | bmoraca | seems to be back up now |
20:15.14 | timeshell_atwork | I have a Polycom IP 601 that freezes up and reboots when call comes in on a specific analog trunk port that is forwarded to it using 1.6.0.14. |
20:15.28 | timeshell_atwork | This didnt happen before my upgrade from 1.6.0.9 |
20:15.46 | timeshell_atwork | Any ideas? |
20:15.52 | kfife | timeshell_atwork: is your polycom firmware up to date? |
20:16.02 | timeshell_atwork | kfife Yes |
20:16.20 | timeshell_atwork | Most recent boot and sip software for the 601 |
20:16.38 | timeshell_atwork | Upgraded it to 3.1.3 from 3.1.2 a month or so ago. |
20:17.01 | kfife | timeshell_atwork: Waiting for epiphany |
20:17.32 | [TK]D-Fender | kfife: epiphany is already installed, but Firefox is selected as default. Is that OK? |
20:18.30 | kfife | [TK]D-Fender: LOL. |
20:18.50 | [TK]D-Fender | is DONE for the day.... |
20:18.52 | [TK]D-Fender | BBL |
20:18.53 | kfife | [TK]D-Fender: Epiphany=webkit? |
20:19.41 | *** join/#asterisk kerframil (n=kerframi@gentoo/user/kerframil) |
20:20.22 | KyleK | ~dev |
20:20.23 | kfife | nope--I see now- mozilla based. |
20:21.03 | jong2 | no starfish channel? |
20:29.46 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
20:31.27 | *** join/#asterisk heit0050 (n=andy@mail2.heitkeconsulting.com) |
20:32.27 | *** join/#asterisk drclue (n=drclue@ip-65-49-160-6.wireless.dyn.beamspeed.net) |
20:34.48 | drclue | <PROTECTED> |
20:34.49 | drclue | Anything folks might want to put on a wish list? |
20:36.06 | drclue | The existing code and documentation I've posted as open source at http://code.google.com/p/fastagi-php-drclue/ |
20:36.27 | KyleK | FastAGI is AGI over sockets? |
20:36.36 | drclue | Pretty much |
20:37.05 | drclue | I also toss in a persistent AMI connection |
20:38.01 | drclue | The way I set it up my FastAGI.php runs as a daemon and loads ones script at dial time based on whatever you specify in the dial plan |
20:38.10 | *** join/#asterisk Caplain (i=shayne@caplain.loves.thraen.fbi.gov.silverelitez.org) |
20:38.55 | drclue | The FastAGI.php even has a command line option to generate the init.d script to start it at boot |
20:39.47 | KyleK | cool |
20:40.42 | *** join/#asterisk mnt_real (n=sinan@bas12-montrealak-1167975979.dsl.bell.ca) |
20:41.01 | drclue | I found that having the FastAGI.php load the scripts at dial time that PHP development/debugging cycle is pretty painless edit,save, dial |
20:43.03 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
20:43.03 | *** mode/#asterisk [+o denon] by ChanServ |
20:43.11 | *** join/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
20:43.19 | *** join/#asterisk rgavril (n=rgavril@89.120.4.153) |
20:43.32 | *** part/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
20:43.37 | drclue | I have a companion script called FastAMIevents.php , that does event monitoring but I have not really figured out what I want to do with it yet. I was sorta thinking of having it use shared memory and output XML and JSONized XML depending upon how it was called |
20:44.32 | *** join/#asterisk netpro25_ (n=mmanning@64-238-176-253.ksg.apt.gru.net) |
20:45.27 | netpro25_ | Can someone point me to some info regarding the # shortcut |
20:45.43 | netpro25_ | to transfer the call |
20:46.00 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
20:47.14 | drclue | netpro25_: what kind of phone are you trying to dial the "#" from? |
20:47.39 | netpro25_ | drclue: well I have a spa941 just wanna know how to go from desktop to cell if I gotta run |
20:49.01 | *** join/#asterisk GameGamer43 (n=GameGame@h69-129-142-83.mdsnwi.tisp.static.tds.net) |
20:50.04 | carrar | Are you listening for a trasfer option when you call out or receive a call? |
20:50.13 | netpro25_ | nm just figured out how to do it with the transfer button |
20:50.21 | netpro25_ | thats sweet |
20:54.20 | *** join/#asterisk tRSS (i=tRSS@140.192.34.2) |
20:54.53 | tRSS | quick question: can I have multiple periodic-announce messages for the same queue? |
20:59.21 | *** join/#asterisk ZX81 (n=Matt_Rid@121.74.14.197) |
21:00.21 | tRSS | quick question: can I have multiple periodic-announce messages for the same queue? |
21:00.56 | *** part/#asterisk drclue (n=drclue@ip-65-49-160-6.wireless.dyn.beamspeed.net) |
21:01.54 | netpro25_ | is it safe to have asterisk use the tmp folder for record cache |
21:02.19 | KyleK | I'd only worry about it if you have shared hosting on the same server |
21:02.27 | *** join/#asterisk IBC_jkenney (n=jkenney@ip65-44-169-66.z169-44-65.customer.algx.net) |
21:02.33 | netpro25_ | okay |
21:02.37 | *** join/#asterisk korcan (n=korcan@ip65-44-169-66.z169-44-65.customer.algx.net) |
21:02.48 | netpro25_ | well in that case where is a standard place to put it |
21:02.57 | netpro25_ | /var/lib/asterisk/tmp? |
21:03.51 | KyleK | I dunno actually |
21:03.59 | netpro25_ | well that sounds good to me |
21:04.00 | netpro25_ | heh |
21:10.45 | *** part/#asterisk ZX81 (n=Matt_Rid@121.74.14.197) |
21:12.47 | kfife | Digits 1,2 and 3 don't break DISA dialtone? Strange. DTMF is generated by legacy PBX, connected to Asterisk via private-loop PRI Any ideas? 1.6.0.13 |
21:13.14 | kfife | 4,5,6,7,8,9,0,#,* register properly |
21:17.51 | *** join/#asterisk voxter (n=voxter@76.77.95.2) |
21:19.31 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
21:20.31 | *** join/#asterisk GameGamer43 (n=GameGame@h69-129-142-83.mdsnwi.tisp.static.tds.net) |
21:22.46 | Naikrovek | all sip traffic is UDP, yes? |
21:23.12 | uqlev | RTP is UDP only |
21:23.17 | Naikrovek | okie dokie |
21:23.25 | Naikrovek | the SIP signalling is UDP as well? |
21:23.40 | netpro25_ | is it possible to include a whole directory in extensions.conf |
21:23.49 | netpro25_ | like #include "exts/*" |
21:24.12 | [TK]D-Fender | Naikrovek: SIP supports TCP & UDP. UDP is standard with *, TCP is 1.6 option |
21:24.41 | Nugget | SIP TCP is wonderful if you're stuck with ucky NAT. |
21:24.50 | Naikrovek | i'm setting up some ACLs on my router and whenever I enable the proper ACL, my asterisk system no longer hears any incoming sound from outside callers. even though my VOIP provider is given full UDP connectivity to my * server |
21:24.55 | Naikrovek | this is terribly frustrating. |
21:25.03 | Naikrovek | will add a TCP allow for them as well |
21:26.33 | dustybin | i have communication!!! |
21:26.42 | [TK]D-Fender | Naikrovek: TCP won't help your RTP issue |
21:26.51 | Naikrovek | this is starting to piss me off |
21:27.09 | Naikrovek | where could the traffic be coming from, if not from the trunk |
21:27.48 | kfife | What's the benefit of SIP over TCP? Just authentication or some other beneift? |
21:27.50 | dustybin | -- Registered SIP '1000' at 192.168.2.20 port 26342 |
21:27.52 | dustybin | :D |
21:28.13 | Naikrovek | when i have the ACL in place, and I dial from my cell phone, I can connect to my own IVR and hear the voice. but I can't dial an extension. My phone system hears nothing |
21:28.17 | netpro25_ | kfife: two different animals |
21:28.22 | netpro25_ | you mean UDP TCP |
21:28.25 | Naikrovek | wtf is up with that... |
21:28.31 | [TK]D-Fender | Naikrovek: kfife Bypasses need for NAT keep-alives |
21:28.43 | [TK]D-Fender | kfife: ^ |
21:28.53 | Naikrovek | ? |
21:29.01 | kfife | But transport is still over UDP no? |
21:29.20 | kfife | ...media transport I mean |
21:29.36 | kfife | TCP just for sip signalling--that's the idea, right? |
21:29.48 | Naikrovek | kfife: it's all UDP as far as I can tell |
21:30.00 | kfife | I was goign to say: otherwise katy bar the door! |
21:30.02 | Naikrovek | but [TK]D-Fender said that TCP is optional |
21:30.13 | kfife | There's latency, and then there's LATENCY |
21:30.23 | [TK]D-Fender | Naikrovek: No, I didn't |
21:30.26 | netpro25_ | kfife: TCP would help reduce packet loss and thus jittery sound |
21:30.41 | Naikrovek | [TK]D-Fender: what did you mean, then |
21:30.50 | kfife | netpro25_: I believe that's incorrect. |
21:30.57 | [TK]D-Fender | Naikrovek: Exactly what I said and nothign more. |
21:31.01 | Naikrovek | oh that you can use TCP SIP in asterisk 1.6? |
21:31.08 | [TK]D-Fender | netpro25_: Completely incorrect |
21:31.14 | netpro25_ | kfife: then I guess I dont fully understand |
21:31.43 | netpro25_ | I was thinking in the lines of UDP having to acknowledgement packet |
21:31.48 | kfife | If you had TCP guarante delivery, you'd end up with HUGE latency-seconds, minutes, etc as packets lined up to be delivered |
21:31.50 | *** join/#asterisk raden_work (n=tanning@69-179-99-17.stat.centurytel.net) |
21:31.55 | raden_work | there anything special i should have todo to make my procurve work with asterisk ? |
21:32.04 | netpro25_ | kfife: right slower |
21:32.12 | *** join/#asterisk manxpower (n=EWieling@69.73.94.162) |
21:32.13 | kfife | Ever done a VIDEO stream over TCP VPN? See what happens. |
21:32.14 | dustybin | the book says asterisk can also be used with analogue phone lines with the correct card, how cool is that? |
21:32.15 | [TK]D-Fender | Naikrovek: rarely do you realy have need of TCP for SIP. I can be helpful if you have multiple phones behind a remote NAT as the inbound is taken care of due to a persistent connection. |
21:32.21 | manxpower | ~answers |
21:32.22 | infobot | i heard answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
21:32.38 | Naikrovek | [TK]D-Fender: the phone server is not behind NAT; it's on a public IP, so I don't think NAT is the issue |
21:32.53 | [TK]D-Fender | Naikrovek: I also didn't say this was your issue. |
21:33.04 | netpro25_ | So how do sip providers like vitelity resolve nat issues, because I do not need to open ports for them |
21:33.04 | Naikrovek | [TK]D-Fender: i know, just ruling NAT out |
21:33.12 | [TK]D-Fender | Naikrovek: I never did get a solid description of it or SIP debug of a failed call |
21:33.18 | kfife | netpro25_: stun, ice, turn |
21:33.24 | [TK]D-Fender | Naikrovek: if * is behind NAT then you might have an issue |
21:33.34 | netpro25_ | kfife: all three or one of those |
21:33.43 | Naikrovek | [TK]D-Fender: it isn't |
21:33.44 | kfife | take your pick |
21:33.50 | kfife | netpro25_: take your pick. |
21:34.02 | kfife | netpro25_: any or all. |
21:34.03 | Naikrovek | [TK]D-Fender: fetching sip debug for call now. few moments, please |
21:34.05 | [TK]D-Fender | netpro25_: * needs ports forwarded to it otherwise no audio, and probably no calls |
21:34.28 | netpro25_ | [TK]D-Fender: yes, thats what I had to do |
21:34.41 | netpro25_ | otherwise I had no connection from my phone to server |
21:34.57 | raden_work | [TK]D-Fender, is there something keeping me from registering with vitelity or is there a network issue ? http://pastebin.com/d5e269648 |
21:35.45 | kfife | raden_work: I just dialed my vitelity DID, and forwarded the call to my mobile phone via vitelity trunk. All seems to work. |
21:35.45 | *** part/#asterisk GameGamer43 (n=GameGame@h69-129-142-83.mdsnwi.tisp.static.tds.net) |
21:36.04 | kfife | raden_work: try reloading your sip.conf |
21:36.12 | raden_work | did |
21:36.18 | raden_work | did a graceful restart |
21:36.21 | raden_work | as well |
21:36.24 | netpro25_ | seems like vitelity is the choice of a few people in here |
21:36.36 | netpro25_ | I like them, have had no probs |
21:36.47 | netpro25_ | anyone using them for business lines? |
21:36.58 | kfife | I reallly like their user interface for managing/adding/deleting etc. I also like their ticket system. |
21:37.00 | dustybin | i know you guys dont approve of guis, however, is there some kind of asterisk stat / simple gui system, so i can check things from html if im at work, etc |
21:37.15 | netpro25_ | kfife: yes and they respond so diligently and quickly |
21:37.19 | netpro25_ | very helpful |
21:37.25 | raden_work | well i have a route to them just for some reason seem to be having issues |
21:37.32 | kfife | raden_work: netpro25_: AMEN |
21:37.36 | Naikrovek | [TK]D-Fender: http://pastebin.com/d5716c0ae |
21:37.52 | kfife | At the moment they |
21:37.57 | raden_work | i like vitelity never said i didnt |
21:38.13 | netpro25_ | raden_work: did you forward 5060? |
21:38.20 | kfife | At the moment they're one of the only ITSP's that can set CNAM for your DID's. A mere $10 no MRC. |
21:38.23 | [TK]D-Fender | Naikrovek: Do not specify a specific IP and that PB is a complete waste |
21:38.27 | kfife | ^ VITELITY IS |
21:38.33 | Naikrovek | ugh |
21:38.36 | Naikrovek | k |
21:38.48 | Naikrovek | what is your pastebin preference |
21:38.57 | raden_work | netpro25_, yeah that all done on the router all i did was throw my hp procurve in and yeah every other provider registering |
21:39.08 | raden_work | vitel-outbound/tanning 64.2.142.17 5060 OK (83 ms) |
21:39.08 | raden_work | vitel-inbound/tanning 64.2.142.15 5060 UNREACHABLE |
21:39.08 | raden_work | callcentric/17772445766 204.11.192.31 5060 OK (46 ms) |
21:40.16 | *** join/#asterisk nightrid3r (n=kvirc@78-20-232-172.access.telenet.be) |
21:40.17 | netpro25_ | raden_work: strange |
21:40.25 | raden_work | netpro25_, i know |
21:40.32 | netpro25_ | raden_work: did you file a bug |
21:40.37 | raden_work | traceroute shows same route to both |
21:40.39 | netpro25_ | raden_work: err ticket |
21:40.45 | raden_work | not yet |
21:41.05 | netpro25_ | raden_work: as I said earlier they are very helpful and nice |
21:41.11 | raden_work | yeah they are |
21:41.24 | raden_work | call centric support better but i like vitelity overall better |
21:41.38 | Naikrovek | [TK]D-Fender: http://pastebin.ca/1550844 |
21:42.10 | netpro25_ | raden_work: ah they have unlimited at call_centric |
21:42.16 | raden_work | our ISP been down 5 hours in last 16 hours and 42 min so it could be something todo with that as well |
21:42.22 | kfife | I had a ticket in which Vitelity set my outgoing CLID to some number in New Jersey for certain terminations. Seems fixed now, but I'm sure it's because some cheap route in their 'Rate Deck' is is some half a$$ provider. Made me a bit nervous. Never had that problem with anyone else. A vexing probelm because you'd rarely be in a position to notice all the times it happens. |
21:42.32 | raden_work | netpro25_, there are advantages to both |
21:42.48 | netpro25_ | raden_work: like single channel with unlimited |
21:42.57 | raden_work | yeah :) |
21:43.13 | raden_work | or 3 channel unlimited inboung and 9 dollar a channel extra |
21:44.54 | netpro25_ | raden_work: ah personal unlimited |
21:44.57 | netpro25_ | $6 |
21:44.59 | [TK]D-Fender | Naikrovek: Looks ok, verify your WAN IP and confirm what you have forwarded |
21:45.00 | netpro25_ | thats good deal |
21:45.12 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
21:45.31 | Naikrovek | [TK]D-Fender: there's no NAT here, what do you mean forwarded |
21:45.53 | [TK]D-Fender | Naikrovek: Confirm your firewall settings. Also, yuo only reach an IVR. Still no audio direct? |
21:46.07 | *** join/#asterisk GameGamer43 (n=GameGame@h69-129-142-83.mdsnwi.tisp.static.tds.net) |
21:46.56 | Naikrovek | [TK]D-Fender: when i turn off the firewall it work, turn it on, and it doesn't work. I already know the firewall is the issue. I just don't know why. I can hear the IVR on my cell phone, but dialling numbers does nothing when the firewall is up. The DTMF or DMTF or FTDM or whatever it is doesn't get to my IVR |
21:47.21 | Naikrovek | turning the firewall on renders * deaf for me |
21:47.22 | raden_work | Naikrovek, you have your ports forwarded ? |
21:47.31 | Naikrovek | raden_work: no NAT |
21:47.34 | raden_work | Naikrovek, router / firewall model ? |
21:47.40 | Naikrovek | Cisco 2811 |
21:47.59 | raden_work | must be something with SPI |
21:48.11 | Naikrovek | ~SPI |
21:48.12 | infobot | i heard spi is serial peripheral interface. Software in the Public Interest |
21:48.12 | manxpower | Naikrovek: Audio uses UDP ports 10,000 - 20,000 by default. CISCOs use port 16384 - 32768 by default for audio. change /etc/rtp.conf to use those ports |
21:48.34 | Naikrovek | manxpower: the trunk is wide open to my provider. |
21:48.42 | manxpower | Naikrovek: apparently not. |
21:48.49 | manxpower | if it was you would not have this problem. |
21:48.50 | raden_work | Naikrovek, pastebin debug of call |
21:48.59 | Naikrovek | with or without firewall, TCP and UDP are wide open to the * server |
21:49.04 | raden_work | manxpower, are you the one that gave me contact for vitelity ? |
21:49.08 | manxpower | make sure you turn off the sip fixup in the router. |
21:49.08 | Naikrovek | according to the Cisco ACL they are |
21:49.16 | manxpower | Naikrovek: do you have canreinvite=no? |
21:49.21 | [TK]D-Fender | Naikrovek: Your firewall is blocking the wrong stuff then. You need to allow 5060 & 10000-20000 all UDP |
21:49.30 | manxpower | otherwise the phone will try talking directly to the provider |
21:50.27 | manxpower | Naikrovek: on your router issue the config command "fixup protocol sip 5060 " |
21:50.28 | Naikrovek | what EXACTLY does this ACL block between these two hosts? surely there must be some Cisco folk in here. |
21:50.30 | manxpower | then save the config |
21:50.33 | manxpower | ..er... |
21:50.33 | Naikrovek | <PROTECTED> |
21:50.33 | Naikrovek | <PROTECTED> |
21:50.34 | manxpower | wait! |
21:50.38 | manxpower | no fixup protocol sip 5060 |
21:50.38 | dustybin | before i go to bed, my softphone has a telephone number of 500, how can i get asterisk to call that number so it rings? |
21:51.17 | manxpower | Naikrovek: You understand that by default Asterisk gets out of the audio path and the phone and the provider talk directly, right? |
21:51.26 | Tapout | I'm getting service for a reverse-cell calling , do you guys think the communications are 'private'? In that, I should be able to give out my credit card numbers over phone...? |
21:51.41 | Naikrovek | manxpower: no |
21:51.49 | Naikrovek | how do recordings happen if that's the case |
21:51.56 | *** join/#asterisk alecdavis (n=sivad@202-78-149-14.cable.telstraclear.net) |
21:51.58 | Naikrovek | that must be what you mean by NOT default |
21:51.59 | KyleK | whats cdr(userfield) for? |
21:52.20 | Naikrovek | k let me try something else here then |
21:52.22 | netpro25_ | dustybin: google asterisk dial command |
21:52.24 | manxpower | Naikrovek: no, there are features that can be enabled that will prevent reinvites, but those are not DEFAULT. |
21:52.36 | dustybin | thanks |
21:53.10 | Tapout | anyone? are the services you buy from voip providers 'private' in that, I should be ok giving out my credit card number over the phone? |
21:53.32 | Tapout | I'm getting reverse cell phone calling .. so I get free north american calling on my cell, i wanna be able to give my credit card number over the phone without issues... what do you guys think? |
21:53.40 | manxpower | Tapout: your question is silly. Do you pay by credit card when you eat at a restaurant? Phones are more secure than that. |
21:53.52 | manxpower | Tapout: I don't even know what "reverse cell phone calling" is. |
21:54.19 | manxpower | Tapout: do you give your credit card number out over a land line? Easy to tap that too. |
21:54.22 | Naikrovek | jesus fucking christ i hate this |
21:54.38 | Naikrovek | my network is wide fucking open and phone calls work, or i secure it and calls break. what a choise |
21:54.39 | Naikrovek | choice |
21:55.11 | Tapout | manxpower, i'm paying a company so taht when I dial out on my cell, it rings busy.. i hang up, and it calls me back, and then I can dial out for free |
21:55.13 | manxpower | Naikrovek: no, listening to what I said about reinvites is also another choice. turn them off. happy Naikrovek |
21:55.14 | raden_work | [TK]D-Fender, any idea why i cannot register ? |
21:55.23 | Naikrovek | will never be happy |
21:55.27 | Naikrovek | so lose that idea now |
21:56.43 | Naikrovek | and |
21:56.53 | Naikrovek | i'm still talking to asterisk when audio stops working |
21:57.06 | manxpower | I guess it sucks to be you. |
21:57.09 | Naikrovek | so i haven't handed off to a phone yet |
21:57.26 | Naikrovek | the traffic has to be coming from somewhere other than the trunk |
21:57.37 | Naikrovek | my router isn't voip-aware so i don't think it woudl be that |
21:58.42 | *** join/#asterisk flujan (n=flujan@189.111.254.251) |
22:00.31 | Tapout | anyone doing at-home asterisks boxes in canada? I wonder if there is a voip provider that gives blocks of minutes cheap near calgary.. my voip is a bit shitty |
22:00.41 | raden_work | I have everything plugged into a hp procurve switch and then uplinked to netgear at moment could this be causing any problem with registration ? |
22:01.20 | *** join/#asterisk Defraz (n=Defraz@c72co-edge-router.fuzecore.com) |
22:01.32 | *** join/#asterisk shinao1 (n=shinao1@41.219.218.218) |
22:01.46 | KyleK | Tapout: who are you using right now? |
22:02.11 | Tapout | magic jack lol |
22:02.16 | dustybin | how odd, im missing the dial command |
22:02.21 | KyleK | ah |
22:02.26 | Tapout | for home, but i wanna setup my own box.. i'm paying telsair right now for cell phone, but i'd rather do it myself as this is awesome |
22:03.23 | KyleK | ah |
22:04.13 | KyleK | right now I'm using les.net, from BC it seems like his servers are colocated in BC too even though the company is manitoba based |
22:04.21 | Naikrovek | is there a way in linux to see where UDP packets are coming from |
22:04.36 | Naikrovek | netstat doesn't seem to be doing it |
22:04.36 | manxpower | dustybin: not odd at all. The CLI dial command is only available if you have the alsa headers and libraries installed when you build Asterisk |
22:04.45 | dustybin | eeek |
22:05.08 | KyleK | Naikrovek: tcpdump |
22:05.21 | dustybin | thanks for help this evening, time for bed |
22:05.34 | Tapout | KyleK, what are you paying a month? |
22:06.38 | Naikrovek | ah hah |
22:06.45 | Naikrovek | MFer |
22:06.51 | KyleK | I'm paying 1.5 cents a minute |
22:08.18 | Maliuta | KyleK: what if you just wanted it for inbound calls? |
22:08.45 | Maliuta | KyleK: I have been looking for a .ca DID (my parents live in Fort McMurray) |
22:09.04 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
22:09.35 | Naikrovek | okay i figured it out |
22:09.42 | Naikrovek | what in the holy piss is going on |
22:09.58 | Naikrovek | my incoming data is not coming from my trunk. funny thing |
22:10.09 | Naikrovek | will be calling about that tomorrow |
22:10.20 | Maliuta | Naikrovek: reinvite? |
22:10.24 | Naikrovek | no |
22:10.26 | Naikrovek | not reinvite |
22:10.37 | Naikrovek | not fixup protocol sip 5060 |
22:10.39 | Naikrovek | not NAT |
22:10.49 | Naikrovek | not anything anyone described fixed it |
22:11.06 | KyleK | Maliuta: I still have POTS service i'll look at what it is at les.net |
22:11.15 | KyleK | but they might not offer any dids in fort mcmurray |
22:11.24 | Naikrovek | the packets carrying audio INTO my PBX from my PSTN provider do not come from the trunk I have configured |
22:11.34 | Maliuta | Naikrovek: I know my provider use a bank of machines to handle stuff ... it pisses me off |
22:11.59 | Maliuta | KyleK: they have Edmonton .... which I figure is close enough (and cheaper than them calling my .au number) |
22:12.01 | Naikrovek | apparently mine does too |
22:12.26 | Naikrovek | when configuring the primary access list at my internet connection, I neglected to imagine that anything other than my effing trunk would be sending me voiip data |
22:12.27 | Naikrovek | silly me |
22:12.36 | KyleK | Maliuta: true dat |
22:12.37 | Naikrovek | so when I put the ACL in place, that data was blocked |
22:13.35 | Maliuta | Naikrovek: I had that too. I put a qualify on the sip trunk config and then used some whois foo to figure out what to do on the firewall |
22:13.58 | Naikrovek | i'm not opening up two /16 networks, which is what my voip provider has, apparently |
22:14.12 | Maliuta | ick |
22:14.15 | Naikrovek | i just opened the /24, and i'll be on the lookout for problems |
22:14.50 | Maliuta | I'd contact them and ask for info on exactly which servers/ip's you need to allow |
22:14.59 | Naikrovek | well that's fixed. only took three years to semi-secure that network. I can't BELIEVE we weren't hacked. about 8 machines were full open to the internet for years |
22:15.05 | Naikrovek | not even firewalls on the machines |
22:15.19 | Naikrovek | is going to reinstall everything anyway |
22:15.30 | Naikrovek | slaps the previous admin |
22:15.46 | Naikrovek | with a cricket bat |
22:15.49 | Naikrovek | sideways |
22:15.53 | Naikrovek | idiot |
22:15.59 | Naikrovek | shesus |
22:16.01 | Naikrovek | leaves |
22:16.37 | Maliuta | you don't want to use a cricket bat sideways ... you want the full face of the blade to deliver force in a more destructive manner |
22:16.51 | Naikrovek | sideways means more PSI |
22:16.56 | Naikrovek | pounds per square inch |
22:17.12 | *** join/#asterisk jcape (n=jcape@209.120.251.81) |
22:17.24 | Naikrovek | ... if pound is abbreviated "lb" why is "pounds per square inch" abbreviated "PSI" |
22:17.36 | Naikrovek | shouln't that be "lbSI" |
22:17.38 | Naikrovek | ugly. |
22:17.41 | Naikrovek | okay leaving now |
22:17.48 | Naikrovek | still mad even though i solved it, manxpower |
22:17.50 | Naikrovek | told ya :) |
22:19.53 | *** join/#asterisk TimRiker (n=timr@bzflag/projectlead/TimRiker) |
22:20.10 | KyleK | Maliuta: go with the flat rate DID |
22:20.22 | *** join/#asterisk LiNeTuX (n=LiNeTuX@rrcs-71-43-111-78.se.biz.rr.com) |
22:22.15 | *** join/#asterisk Tim_Toady (n=moi@adsl209-7.kln.forthnet.gr) [NETSPLIT VICTIM] |
22:22.43 | Maliuta | KyleK: will look into it, have to talk to the 'rents and see if it's worth it given their phone plan |
22:23.14 | KyleK | so they dont have broadband? |
22:23.25 | Maliuta | KyleK: I can already call all of .ca (including mobiles) for $0.08AU untimed |
22:23.52 | KyleK | haha including mobiles |
22:24.11 | Maliuta | KyleK: they have shaw cable, if I can get my shite together and take the right bits with me at xmas then it'll go to a * trunk |
22:24.15 | KyleK | mobile/not mobile is only noticable outside north america |
22:25.02 | Maliuta | which is most of the world :P |
22:25.14 | KyleK | true |
22:25.43 | Qwell | pfft, only by volume and by population |
22:25.54 | Qwell | s/volume/land mass/ |
22:26.29 | Maliuta | and every other meaningful measurement |
22:26.30 | KyleK | cdr_sqlite.so is deprecated? :( |
22:26.32 | Qwell | :D |
22:28.48 | KyleK | $0.08AU untimed as in 8 cents for the whole call? |
22:29.30 | Maliuta | KyleK: yeah |
22:29.39 | Tapout | KyleK, i don't see how you signup to les to use it at home... seems like you gotta pay X amount of dollars for colocation |
22:29.42 | Maliuta | KyleK: to a bunch of places |
22:30.06 | KyleK | Tapout: ? |
22:30.35 | *** join/#asterisk KingOfDos (n=irssi@ht1-ix.kingofdos.net) |
22:30.43 | KyleK | Tapout: i guess thier website is kind of hard to navigate |
22:30.44 | Tapout | KyleK, i want to setup voip at home, using my adsl .. thru les.net , how much initially should it take to use 'em? |
22:30.53 | Tapout | i'd guess, unless you know what you're looking for |
22:31.00 | Tapout | which, obviously i don't ;) |
22:31.07 | KyleK | :) just signup |
22:32.05 | KyleK | I threw in $20 via paypal, they sat on it for 14 days since I dont trust them with my CC info and I don't trust paypal with my banking account (unverified paypal account) :) but like $5 would do |
22:32.24 | bmoraca | KyleK: "mobile/not mobile is only noticable outside north america" <=== that's not true. mobile in the US is a different OCN than even the same providers' copper service...as such, they will have different negotiated rates. |
22:34.54 | KyleK | bmoraca: well i dont have access to any such negotiated rates, I'm just a sucker paying a flat rate :) |
22:36.42 | Maliuta | meh. like I care about $0.08AU to talk to mum for 2 hours |
22:36.52 | bmoraca | most people are far down the ratescale and their upstream providers have extracted enough markup from them that they don't have to account for it...but, for me, i know that calling AT&T/Cingular mobile costs roughly double what it costs to call AT&T copper...that is to say that it costs ~$0.005/min to call AT&T Wireless as opposed to ~$0.0026/min to call AT&T landlines |
22:37.41 | KyleK | got any numbers for canada? |
22:37.56 | Maliuta | bmoraca: how much volume do you have to do to get those rates? |
22:38.21 | bmoraca | no volume commitments...i collocate in the CLEC's data center |
22:38.38 | rene- | bmoraca: sweet |
22:38.55 | *** join/#asterisk SirColin (n=SirColin@my83-216-68-241.mynow.co.uk) |
22:39.29 | bmoraca | they negotiate rates with various providers and i get the benefit of THEIR volume committments |
22:40.04 | rene- | bmoraca: do you resell? |
22:40.06 | Maliuta | I guess that's the issue with collocating in my loungeroom in Brisbane |
22:40.19 | bmoraca | i charge $40/mo for hosted PBX trunks...which on average cost me about $3.50 in usage fees |
22:40.29 | rene- | bmoraca: sweet |
22:40.32 | bmoraca | rene-: yes. currently only to California |
22:40.35 | *** join/#asterisk dshap (n=samhel55@cpe-024-211-245-021.nc.res.rr.com) |
22:41.24 | bmoraca | regular trunks are $30/mo...but they either have to buy the equipment from me or lease it for another $75/mo...Adtran TA900s aren't cheap! |
22:42.44 | dshap | hey can someone help me out? all of a sudden my IVR menu is less responsive to DTMF |
22:42.52 | dshap | i have to press the button a couple times |
22:42.54 | dshap | or hold it down |
22:42.59 | dshap | to get it to switch to the extension that i want |
22:43.02 | *** join/#asterisk propellerhead (n=yogurt2u@190.210.1.37) |
22:43.05 | dshap | the weird thing is |
22:43.10 | dshap | when i run asterisk in the console |
22:43.14 | dshap | with output and everything |
22:43.15 | dshap | it works fine |
22:43.25 | dshap | it's only when i have it running as a daemon |
22:43.27 | dshap | any ideas? |
22:43.57 | KyleK | bmoraca: so they give you a good rate and you just need your own equipment to go from T1 to VoIP? |
22:46.54 | *** join/#asterisk voipmonk (n=voipmonk@65.14.229.26) |
22:48.26 | dshap | KyleK: you have any ideas about why running as daemon vs. console would make a difference? |
22:49.54 | KyleK | like -d? |
22:50.16 | KyleK | it shouldn't make a difference |
22:50.22 | KyleK | try running it in screen |
22:50.39 | dshap | when I do "asterisk &" |
22:50.43 | dshap | i'm havin weird issues |
22:50.45 | dshap | but when i do |
22:50.47 | dshap | "asterisk -cvvv" |
22:50.50 | dshap | i don't have those issus |
22:51.25 | dshap | i haven't made any changes to asterisk during the period that this started happening |
22:51.35 | dshap | however i have started using the linux box extensively more as a web server |
22:51.47 | KyleK | oh |
22:51.50 | dshap | do you think it gets more processing priority when it's a console? |
22:51.52 | KyleK | what linux |
22:51.52 | dshap | or something like that |
22:51.57 | KyleK | ubuntu? |
22:51.58 | dshap | CentOS 5.3 |
22:52.02 | dshap | running on a VERY old box |
22:52.11 | dshap | (surprisingly capable though, haven't had any issues what so ever until now) |
22:52.27 | KyleK | sounds like the computer is allocating more cpu power to tasks it considers interactive |
22:52.35 | KyleK | whats the current load |
22:52.40 | KyleK | (uptime) |
22:52.41 | dshap | how do i check |
22:53.34 | *** join/#asterisk FlaPer87 (n=FlaPer87@unaffiliated/flaper87) |
22:54.26 | FlaPer87 | hey guys, This is the vm_exec method of app_voicemail.c : http://pastebin.com/d79f3c03a don't understand when is recorded the message |
22:55.02 | FlaPer87 | can anybody tell me what part of the method execution creates the audio file and save the message? |
22:55.20 | KyleK | dshap: whats uptime say? |
22:55.59 | KyleK | <PROTECTED> |
22:56.05 | voipmonk | FlaPer87: you're looking at the wrong part of the code.. keep looking |
22:57.06 | FlaPer87 | voipmonk hmm, could you please help me to find what part does the thing? this part is executed by VoiceMail(123@anything) |
22:57.21 | FlaPer87 | It answers the call |
22:57.30 | voipmonk | what do you want to do when you find it? |
22:58.32 | *** join/#asterisk TimToady_ (n=moi@adsl16-233.kln.forthnet.gr) |
22:58.34 | dshap | KyleK: checking now |
22:58.51 | dshap | KyleK: 23 days |
22:59.03 | dshap | KyleK: seems about right...back when my dog accidentally unplugged the server |
22:59.17 | dshap | oh |
22:59.18 | dshap | aps.facebook.com/photodigger/logs.php |
22:59.20 | dshap | oops |
22:59.24 | dshap | 15:58:41 up 23 days, 5:59, 1 user, load average: 0.02, 0.02, 0.00 |
22:59.35 | FlaPer87 | voipmonk I need to process some vocal messages, So, The call will be answered and the user will say something, I need to save the things the user says in a file |
22:59.52 | dshap | what does that mean? 0.02, 0.02, 0.00 |
23:00.40 | KyleK | its how many processes are usually waiting to run |
23:00.51 | KyleK | first is average is over a minute, then 5 then 15 |
23:01.56 | *** join/#asterisk jjshoe (n=jjshoe@wsip-70-183-82-162.sd.sd.cox.net) |
23:02.19 | KyleK | also check top it displays cpu usage |
23:03.00 | dshap | none are over 1% cpu |
23:04.22 | dshap | how can i get a list of every program running on my server |
23:04.44 | dshap | is that it? |
23:04.45 | dshap | top? |
23:04.56 | dshap | what about running services? |
23:05.21 | KyleK | services are programs |
23:05.45 | dshap | what's the difference between ones you have to type "start" in front of though? |
23:05.57 | dshap | like u don't just type mysqld & |
23:06.00 | dshap | like i can for asterisk |
23:06.15 | KyleK | the init script picks up settings |
23:06.30 | dshap | ah so "start" means use an init script |
23:06.50 | KyleK | i believe so |
23:06.59 | KyleK | I've been ignorint centos more than i should |
23:07.07 | dshap | also how come sometimes when i start my FTP server which is "pure-ftpd", i type "pure-ftpd &" |
23:07.08 | dshap | and it works fine |
23:07.12 | dshap | but when i close the SSH terminal |
23:07.16 | dshap | it stops the server |
23:07.24 | KyleK | its still attached to the tty |
23:07.25 | dshap | i just want it to always be running |
23:07.35 | KyleK | nohup pure-ftpd |
23:07.45 | KyleK | or use an init script |
23:08.32 | dshap | and if i wanna kill it later i just use top to get the PID and then "kill [pid]" ? |
23:08.33 | Maliuta | the "&" simply backgrounds it in the current shell, it's not how to run a daemon proper |
23:08.48 | dshap | Maliuta: so I shouldn't be doing "asterisk &" ? |
23:09.02 | KyleK | uhm |
23:09.06 | Maliuta | if you use init scripts you can start and stop things at will |
23:09.07 | KyleK | "asterisk" |
23:09.13 | KyleK | backgrounds itself that i've noticed |
23:09.38 | Maliuta | I think someone needs to go take *nix admin 101 |
23:09.55 | KyleK | gotta start somewhere |
23:10.04 | netpro25_ | Anyone every used the call centric office unlimited seems they don't want you to use it with an IVR |
23:10.14 | FlaPer87 | where can I find docs about asterisk functions? |
23:10.15 | netpro25_ | can anyone confirm this? |
23:10.20 | dshap | Maliuta: i do, lol. u gota good book to recommend? |
23:10.23 | Maliuta | ~book |
23:10.24 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
23:10.32 | KyleK | Maliuta: thats not nix admin 101 |
23:10.33 | dshap | i meant a *nix book |
23:10.37 | dshap | i've read the * book |
23:10.37 | FlaPer87 | Maliuta thanks |
23:10.39 | netpro25_ | In other words can I not use call centric with asterisk |
23:10.44 | Maliuta | KyleK: that was for FlaPer87 |
23:10.45 | dshap | and i have a working * server that i've been programming |
23:10.47 | KyleK | oic |
23:10.48 | dshap | oh |
23:10.55 | KyleK | hrmm |
23:11.05 | dshap | KyleK: well it seems that this issue is pretty spontaneous |
23:11.05 | FlaPer87 | for example, what should this function do? ast_stream_and_wait |
23:11.09 | dshap | probably due to processor load |
23:11.10 | KyleK | dshap: check out o reily books |
23:11.18 | KyleK | FlaPer87: doesn't the code have comments on what it does? |
23:11.21 | dshap | KyleK: should i read something generic or CentOS specific? |
23:11.34 | KyleK | dshap: read the generic book first |
23:11.40 | FlaPer87 | KyleK not the one I'm reading |
23:11.45 | dshap | unix or linux? |
23:11.54 | FlaPer87 | I mean, I'm watching how they use it |
23:12.27 | KyleK | dshap: generic linux yes |
23:12.46 | Nugget | dshap: technically you're asking bash questions, which isn't specific to any unix or linux or posix os. |
23:13.05 | Nugget | I'm assuming you are running bash because that's what people run if they don't know what they're running. :) |
23:13.07 | KyleK | pshaw |
23:13.20 | KyleK | he just needs some background information on *nix administration |
23:13.28 | KyleK | then read the centos book |
23:14.40 | dshap | will do |
23:14.41 | dshap | thanks |
23:15.09 | dshap | but very quickly |
23:15.14 | dshap | what is the proper way to run a daemon in centos? |
23:15.19 | dshap | how should i be starting asterisk? |
23:15.27 | dshap | as opposed to my current method: "asterisk &" |
23:15.38 | KyleK | my asterisk backgrouns without the & |
23:15.43 | KyleK | so its no difference |
23:15.51 | netpro25_ | doent asterisk run in daemon mode by default? |
23:15.58 | netpro25_ | doesnt^ |
23:16.02 | KyleK | you could auto start it in /etc/rc.local or look around the dir for an init script |
23:16.17 | KyleK | netpro25_: I dont think it'll install a script into /etc/init.d/ by default |
23:16.28 | KyleK | I'm assuming hes smart and installed from source;) |
23:16.40 | KyleK | sqlite3-dev |
23:16.50 | netpro25_ | KyleK: when you compile from source? |
23:17.36 | netpro25_ | seemed to do so in ubuntu, unless it never removed the old when when I uninstalled the package |
23:17.52 | KyleK | aye |
23:18.23 | netpro25_ | KyleK: cant remember if I was talking to you earlier about callcentric |
23:18.28 | netpro25_ | or if it was someone else |
23:18.52 | KyleK | probably someone else |
23:19.11 | netpro25_ | do you use it? |
23:19.14 | KyleK | dshap: look in contrib/init.d the redhat one should work |
23:19.30 | KyleK | netpro25_: just les.net, I currently call only 2 people long distance |
23:19.42 | KyleK | been thinking about getting a voip phone for one of them :) |
23:19.56 | *** part/#asterisk SirColin (n=SirColin@my83-216-68-241.mynow.co.uk) |
23:20.44 | dshap | ok cool thanks! |
23:21.00 | netpro25_ | Yea CC has strange packages, the $9 office one says it cant be used for IVR |
23:23.41 | KyleK | they just intend people to use it for incoming calls to peoples |
23:23.57 | *** join/#asterisk voipmonk (n=voipmonk@65.14.229.26) |
23:24.18 | netpro25_ | yea with a ATA I am assuming |
23:24.21 | netpro25_ | not asterisk |
23:24.21 | KyleK | I'd be asking them lots of questions before choosing that package |
23:24.30 | KyleK | no you can still use asterisk |
23:24.34 | netpro25_ | KyleK: yea I sent an email |
23:25.02 | KyleK | IVR services probably means like you cant sell tickets but you can have a menu for extensions |
23:25.08 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
23:25.08 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
23:25.24 | *** join/#asterisk SirColin (n=SirColin@my83-216-68-241.mynow.co.uk) |
23:25.26 | netpro25_ | KyleK: what do you mean sell tickets |
23:25.53 | KyleK | if you're 1800buyporn they want you to get the $20 package |
23:26.01 | netpro25_ | ah |
23:26.03 | netpro25_ | lol |
23:28.13 | netpro25_ | Personal unlimited seems tempting also |
23:37.22 | netpro25_ | Anyway to use a regex on file includes? |
23:37.31 | netpro25_ | or just an asterisk |
23:37.45 | KyleK | for what? |
23:37.58 | netpro25_ | #include sips/*.conf |
23:38.20 | KyleK | oh in sip.conf |
23:38.24 | KyleK | I haven't tried even * |
23:38.31 | KyleK | so probably no for regex |
23:38.32 | [netman] | you can also include a directory, I think |
23:38.41 | netpro25_ | let me try that |
23:50.03 | bmoraca | why would you need to use regular expressions to include files? |
23:55.00 | Corydon76-dig | Because some people don't know the meaning of the word "overkill" |
23:55.22 | Corydon76-dig | file includes support globbing, though |
23:55.59 | manxpower | I always buy a quad core server with 8GB of RAM so I am SURE it will handle 8 analog calls. |
23:56.00 | manxpower | 8-) |
23:59.04 | linagee | can anyone go to this URL? http://www.voip-info.org/wiki/view/Asterisk+QoS |
23:59.22 | linagee | "Invalid Response error was encountered while trying to process the request:" |
23:59.58 | manxpower | LiNeTuX: I noticed it was broken the other day |