| 00:00.10 | p3nguin_ | I would like to see some evidence of what's going on. A debug as well as the context would be useful. |
| 00:02.27 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-170.cablep.bezeqint.net) |
| 00:03.10 | *** join/#asterisk jks (jks@193.189.93.254) |
| 00:03.38 | t_j | context http://freebsd.pastebin.com/m6814cc2 |
| 00:04.47 | t_j | call trace http://freebsd.pastebin.com/m2098137e |
| 00:04.57 | t_j | p3nguin_: ^^ |
| 00:12.39 | Kobaz | hmm. i think i found another bug |
| 00:12.40 | Kobaz | er |
| 00:13.34 | t_j | seems to be the name of the audio file that is the issue... |
| 00:14.13 | *** join/#asterisk idespinner (~idespinne@cpe-76-93-115-243.socal.res.rr.com) |
| 00:17.02 | p3nguin_ | Why does it say user disconnected after every time priority 7 runs? |
| 00:21.37 | t_j | the sound file did not exist :) |
| 00:22.46 | p3nguin_ | When I have a sound file that doesn't exist, it says File does not exist in any format; No such file or directory. |
| 00:23.46 | *** join/#asterisk Dibri (~gavit@190.98.19.139) |
| 00:24.20 | t_j | odd maybe a freebsd related issue |
| 00:24.22 | idespinner | p3nguin_, do you have a pastebin of your dialplan code |
| 00:24.34 | p3nguin_ | idespinner: No. Why would I? |
| 00:25.00 | idespinner | would help, i could take a quick look and see if i see anything there... |
| 00:25.32 | p3nguin_ | idespinner: What, exactly, do you plan to look at if I showed you my dialplan? |
| 00:25.40 | p3nguin_ | or look for, rather |
| 00:25.54 | idespinner | anything that would cause the channel to hangup |
| 00:26.10 | idespinner | only need the significant portion |
| 00:26.13 | t_j | that would be my dialplan code generated by freepbx |
| 00:26.14 | p3nguin_ | idespinner: What are you talking about? If I have a channel that hangs up, I typically know why. |
| 00:26.38 | t_j | debug: http://freebsd.pastebin.com/m2098137e dialplan: http://freebsd.pastebin.com/m6814cc2 |
| 00:26.58 | idespinner | hey if you know why, then i suppose you know... |
| 00:27.23 | t_j | idespinner: it seems the error was a missing/wrong sound file but the debug output didnt say no file.. |
| 00:27.31 | p3nguin_ | I'm not on the same page as you for some reason. |
| 00:30.27 | *** part/#asterisk jelly-bean (~jelly-bea@75-148-103-190-Utah.hfc.comcastbusiness.net) |
| 00:30.40 | *** join/#asterisk BCS-Satori (~BCS-Sator@75-148-21-113-WashingtonDC.hfc.comcastbusiness.net) |
| 00:31.05 | *** part/#asterisk codershop (~ryanoldha@75-148-103-190-Utah.hfc.comcastbusiness.net) |
| 00:31.10 | BCS-Satori | Where is the asterisk init service now in 1.6.2? I do not see it in /etc/init.d/ Or how do I restart asterisk service now? |
| 00:31.41 | kaldemar | BCS-Satori: did you install from source? |
| 00:31.51 | BCS-Satori | kaldemar: yes I did |
| 00:31.59 | BCS-Satori | crap! make config |
| 00:32.01 | BCS-Satori | i forgot |
| 00:32.08 | kaldemar | BCS-Satori: that's it. |
| 00:32.10 | idespinner | t_j, i'm looking at the line: -- Executing [8001@from-internal:11] GotoIf("SIP/7000-0000003a", "1?h") in new stack |
| 00:32.16 | BCS-Satori | Totally forgot lol; thanks |
| 00:32.28 | idespinner | on line 56 |
| 00:33.27 | t_j | idespinner: the lines reading "User disconnected" translate to the sound file is missing :( |
| 00:34.08 | kaldemar | the "User disconnected" is caused by app_read |
| 00:34.14 | p3nguin_ | That doesn't make sense to me. |
| 00:34.19 | p3nguin_ | Mine does not do that, but t_j's does. |
| 00:34.35 | idespinner | i suppose, but like p3nguin_ said... it usualy says " File does not exist in any format" |
| 00:35.02 | p3nguin_ | Maybe Read() fails in a different manner than Playback() and BackGround() do. |
| 00:35.21 | p3nguin_ | I should test Read() with a bogus file name. |
| 00:36.00 | *** join/#asterisk dzup (dzup@support.team.at.shellium.org) |
| 00:36.27 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 00:36.30 | dzup | hello, i need a php+mysql panel to do billing? |
| 00:36.46 | dzup | per minute x sip trunk and stuff |
| 00:37.17 | kaldemar | dzup: you can parse the call detail records how you wish. |
| 00:39.28 | p3nguin_ | t_j: Mine says user disconnected when I hangup the phone. My file does exist. |
| 00:40.19 | t_j | meh, it works now, i've moved on to lack of meetme support in 1.6.0 on FreeBSD.... |
| 00:40.50 | p3nguin_ | [Feb 17 18:40:29] WARNING[17827]: file.c:664 ast_openstream_full: File enter-conf-n-number does not exist in any format |
| 00:40.53 | p3nguin_ | [Feb 17 18:40:29] WARNING[17827]: file.c:991 ast_streamfile: Unable to open enter-conf-n-number (format 0x4 (ulaw)): No such file or directory |
| 00:41.01 | t_j | due to DAHDI support |
| 00:41.05 | p3nguin_ | There's the result of a bad file. |
| 00:41.11 | p3nguin_ | Install dahdi! |
| 00:41.29 | t_j | FreeBSD doesnt have dahdi :( |
| 00:41.33 | t_j | or not officially |
| 00:41.48 | t_j | i did find this: http://lists.digium.com/pipermail/asterisk-bsd/2010-January/003703.html |
| 00:42.19 | t_j | but i dont know if it has enough for meetme to work |
| 00:42.28 | Katty | who's up for a game of roulette |
| 00:42.30 | *** join/#asterisk simplydrew_ (~simplydre@pool-74-97-190-109.prvdri.fios.verizon.net) |
| 00:42.53 | Kobaz | russian or american? |
| 00:43.01 | Katty | chat. |
| 00:43.20 | p3nguin_ | FreeBSD DAHDI port can be checked out here: |
| 00:43.23 | p3nguin_ | http://svn.digium.com/svn/dahdi/freebsd/trunk/ |
| 00:43.33 | dzup | kaldemar: yes but i need some php+mysql web app that run out of the browser; my people on accounting is not linux people heh, if you know a panel in php+mysql already made? thanks |
| 00:43.37 | t_j | yes, but then how do I get meetme ? |
| 00:44.14 | p3nguin_ | Well, on Linux, you just have to load the dahdi_dummy kernel module to provide a timer. |
| 00:44.30 | t_j | no MeetMe needs the mixer not timer |
| 00:44.44 | p3nguin_ | I'm not familiar with that. |
| 00:44.57 | t_j | http://www.russellbryant.net/blog/2008/06/16/asterisk-16-now-with-a-new-timing-api/ |
| 00:47.10 | idespinner | t_j, thats interesting, but you still do need dahdi... |
| 00:47.54 | idespinner | t_j, if you cant get dahdi or zaptel, use this: http://www.voip-info.org/wiki/view/Asterisk+cmd+Conference |
| 00:48.30 | idespinner | i had a client require asterisk for windows one time(very peculiar circumstance) and the used that. |
| 00:48.50 | t_j | idespinner: is it compatiable as a droping for MeetMe? |
| 00:49.31 | idespinner | what do you mean by droping? |
| 00:50.00 | t_j | cat i s/MeetMe/Confrence/ in the dial plan or do i need to hack more? |
| 00:50.05 | t_j | s/cat/can/ |
| 00:50.49 | idespinner | t_j, err well, yea you do conference in the dialplan... i dont think im following you 100% though. |
| 00:51.01 | p3nguin_ | core show application Conference |
| 00:51.05 | p3nguin_ | Any help there? |
| 00:51.25 | p3nguin_ | I don't even have that app. |
| 00:51.27 | idespinner | its not in trunk by default... |
| 00:51.33 | idespinner | you have to compile it |
| 00:51.37 | t_j | idespinner: i mean in my existing code can i just switch calls to conference to MeetMe, or do I need to hack it more |
| 00:51.50 | idespinner | you will need to hack it a little |
| 00:51.52 | idespinner | but not much |
| 00:51.59 | idespinner | gtg |
| 00:52.04 | idespinner | afk for a bit |
| 00:52.44 | *** join/#asterisk lsenft (~levisenft@75-148-103-190-Utah.hfc.comcastbusiness.net) |
| 00:53.09 | *** join/#asterisk grey-monkey (~grey-monk@75-148-103-190-Utah.hfc.comcastbusiness.net) |
| 00:54.51 | Katty | Qwell: roulette? |
| 00:55.21 | Qwell | I'm nexting as fast I can! |
| 00:56.09 | Katty | oh |
| 00:56.12 | Katty | k, hold on |
| 00:56.15 | Katty | i'm on the wrong roulette game |
| 00:58.06 | t_j | how can I compile just meetme? |
| 00:58.47 | Katty | Qwell: k, ready set next!!! |
| 00:59.35 | Katty | i don't suppose you can make putty always be in the front |
| 00:59.52 | *** join/#asterisk Arsenick (~y@modemcable022.82-21-96.mc.videotron.ca) |
| 00:59.53 | Katty | short of full screen |
| 01:00.49 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 01:00.49 | Qwell | runs off |
| 01:00.50 | Katty | cause this is horribly irritating |
| 01:00.58 | Katty | come back! |
| 01:01.04 | Katty | i haven't found you yet. |
| 01:01.17 | Qwell | subtracts 1 from 31,000 |
| 01:01.20 | Qwell | that'll take a while :D |
| 01:01.21 | voipmonk | hrmm |
| 01:01.57 | Qwell | Katty: soooooon |
| 01:02.04 | Qwell | tuesday probably |
| 01:02.06 | Katty | what does your background look like? |
| 01:02.17 | Qwell | background... |
| 01:02.24 | Qwell | looks behind him |
| 01:02.25 | Katty | you know... |
| 01:02.27 | Katty | is there something specific |
| 01:02.42 | Qwell | Look for "Partner's camera is not found" |
| 01:02.49 | Katty | oh |
| 01:02.52 | Qwell | tuesday! |
| 01:03.02 | Qwell | ish |
| 01:03.18 | Katty | so your camera isn't connected? |
| 01:03.24 | Katty | i guess that means i should uncheck Cam Required |
| 01:03.30 | Qwell | it means I don't have one |
| 01:04.39 | *** join/#asterisk etnos (etnos@c-75-74-66-161.hsd1.fl.comcast.net) |
| 01:07.17 | Katty | well does anyone else wanna roulette? |
| 01:08.32 | voipmonk | ... |
| 01:08.41 | Katty | i'll take that as a no :< |
| 01:11.19 | Kobaz | puts 100 on black and red |
| 01:11.46 | voipmonk | 50 on black |
| 01:12.05 | Katty | wrong roulette! :P |
| 01:16.30 | *** join/#asterisk korihor (~korihor@190.205.251.97) |
| 01:16.48 | leifmadsen | I love roulette! |
| 01:16.52 | leifmadsen | my g/f on black! |
| 01:17.38 | dlynes | leifmadsen, your gf's black? |
| 01:17.48 | leifmadsen | not always |
| 01:18.16 | dlynes | oh...flavor of the week? |
| 01:18.42 | fifer | I'm definitly rusty here. I have a context that needs to call a macro when a particular extenion(s) is/are called |
| 01:19.08 | fifer | this line works fine when I dial 101: exten => 101,1,Macro() |
| 01:19.31 | leifmadsen | o.O |
| 01:19.35 | leifmadsen | which macro are you calling there? |
| 01:19.39 | dlynes | fifer, You are going to get to what your problem is, right? |
| 01:19.42 | fifer | but neither of these do: exten => s,1,Macro() or exten => 1XX,1,Macro() |
| 01:19.50 | leifmadsen | fifer: 1XX is literal there |
| 01:19.55 | leifmadsen | not a pattern match like you think it should be |
| 01:19.56 | fifer | Only one line in the context |
| 01:19.59 | dlynes | fifer, You want _1XX, not 1XX |
| 01:20.00 | fifer | Oh |
| 01:20.04 | fifer | ah |
| 01:20.07 | leifmadsen | dlynes: DON'T GIVE HIM THE ANSWER :) |
| 01:20.12 | leifmadsen | 's' is also not a catch all |
| 01:20.12 | fifer | how about s? |
| 01:20.13 | dlynes | fifer, '_' signifies a pattern |
| 01:20.13 | fifer | lol |
| 01:20.21 | leifmadsen | 's' would match on s |
| 01:20.28 | Katty | s is super |
| 01:20.44 | leifmadsen | if you had an analog line coming into the context, it would match on 's' |
| 01:20.47 | dlynes | fifer, 's' matches a drop with no extension specified |
| 01:21.01 | fifer | I always understood that if I want the lines to execute for everything comming in to that context regardless of pattern, you use s |
| 01:21.09 | dlynes | fifer, no |
| 01:21.12 | leifmadsen | you understood wrong |
| 01:21.15 | leifmadsen | 's' is not a catch all |
| 01:21.15 | fifer | definitly rusty |
| 01:21.23 | *** join/#asterisk albertoandrade (~albertoan@189.34.121.136) |
| 01:21.36 | fifer | ok, _1XX will work fine for me |
| 01:21.39 | leifmadsen | basically 's' is used by analog lines because they can't request an extension |
| 01:21.57 | leifmadsen | digital lines (PRI, SIP, etc..) can request a specific destination |
| 01:23.58 | fifer | Aha, got my code to work: It is rather convoluted: http://pastebin.ca/1800862 |
| 01:24.03 | fifer | but works for my weird usecase |
| 01:24.45 | fifer | ok, spoke too soon |
| 01:24.51 | Katty | your convoluted. |
| 01:24.56 | Katty | s/your/you're/ |
| 01:25.17 | fifer | calls the first macro everytime |
| 01:26.11 | fifer | anyone have experience with REGEX function? Trying to get this to work: http://pastebin.ca/1800862 |
| 01:27.32 | fifer | Though it may be the GotoIf |
| 01:28.04 | leifmadsen | example has quotes around the 2nd options I think |
| 01:28.21 | leifmadsen | you should break it out into a simple chunk and use NoOp() or Verbose() to see what you're actually getting back |
| 01:28.31 | leifmadsen | ok, I'm out |
| 01:29.22 | fifer | Good point, I keep forgetting about NoOp() and I don't think Verbose was around when I used to work with * all the time |
| 01:29.23 | fifer | thatnks |
| 01:31.52 | Katty | bye leif |
| 01:34.28 | *** join/#asterisk Tech_Travis (~Administr@cpe-76-168-191-127.socal.res.rr.com) |
| 01:37.57 | *** part/#asterisk korihor (~korihor@190.205.251.97) |
| 01:53.06 | *** join/#asterisk ElllisD (~JustaGuy@cpe-67-249-98-145.twcny.res.rr.com) |
| 01:53.55 | ElllisD | I'm fed up with my trixbox pro & am considering abandoning supported pbx entirely. is asterisknow all that different an experience? |
| 01:54.22 | voipmonk | you dont get the planned outages |
| 01:55.13 | ElllisD | as of now it seems the web interface on this box thats on a gigabit network is as slow as the tb forums- is it looping through theirs with that vpn i see go by at boot? |
| 01:56.14 | ElllisD | voipmonk: I remember you- you were going to help me get wanpipe working- I moved it from the virtual machine to a small form factor box instead |
| 01:56.36 | voipmonk | nice |
| 01:56.37 | ElllisD | voipmonk: thank you nonetheless |
| 01:56.41 | voipmonk | hows that working for you? :) |
| 01:56.45 | ElllisD | crap |
| 01:57.01 | ElllisD | the interface times out & i cant see why |
| 01:57.31 | ElllisD | I prolly ruined it with my tinkering in attempting to fix it.... :( |
| 01:57.36 | voipmonk | :) |
| 01:57.45 | voipmonk | you can rebuild |
| 01:58.15 | ElllisD | yup- hate re-uploading each & every hold music file...omg all f-in afternoon |
| 01:58.38 | ElllisD | resampled them all to be between 18-19.x meg |
| 01:59.53 | ElllisD | i had to get a power cable splitter for the sangoma card- a 2-way at BestBuy for a whopping $6 |
| 02:00.29 | ElllisD | are the same functions available in asterisknow? |
| 02:01.09 | ElllisD | looks like i have to add mp3 support :( |
| 02:02.10 | ElllisD | I don't have phones yet so maybe thats good if different ones are compatible |
| 02:03.37 | *** join/#asterisk etfonhomey (~etfonhome@74-131-159-160.dhcp.insightbb.com) |
| 02:03.40 | voipmonk | mp3 support can be added with asterisk-addons if its not there |
| 02:03.46 | voipmonk | so dont freak out :) |
| 02:04.13 | ElllisD | not freaked lol- i see theres a fax module- thats good for sure |
| 02:08.28 | ElllisD | waiting for cp.trixbox.com over & over aaah! |
| 02:08.46 | ElllisD | then i was freakin |
| 02:08.56 | ElllisD | now i have hope |
| 02:09.09 | ElllisD | cant get thru life on earth w/o hope |
| 02:09.13 | ElllisD | ty |
| 02:16.59 | *** join/#asterisk ruben23 (~AGENT@122.55.48.243) |
| 02:19.59 | norrec | hey, how do you add on support for ilbc? |
| 02:30.02 | *** join/#asterisk Mhaddog (~Mhaddog@adsl-11-171-127.mia.bellsouth.net) |
| 02:32.51 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 02:33.06 | carrar | ElllisD, cant get thru life on earth w/trixbox either |
| 02:33.56 | *** join/#asterisk ManxPower-work (~EWieling@40.sub-75-254-235.myvzw.com) |
| 02:34.19 | Katty | hi |
| 02:34.24 | carrar | HARRO |
| 02:34.32 | Katty | HARRO DER |
| 02:34.37 | Katty | DO YOU WNAT TO HUG |
| 02:34.42 | carrar | YES PLS |
| 02:34.49 | Katty | hugs carrar |
| 02:34.53 | carrar | hugs Katty |
| 02:35.06 | ElllisD | carrar: irritating thing- when it was virtualized, it'd drop the network connection 5 mins after boot. never found out why- the support I paid so dearly for didnt support that |
| 02:35.42 | Katty | funny how it works out |
| 02:35.55 | carrar | You should PAY ME |
| 02:35.56 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 02:36.06 | Katty | orly |
| 02:36.11 | Katty | what services do you offer |
| 02:36.12 | ElllisD | carrar: much respectz |
| 02:36.15 | carrar | just because I am ALIVE |
| 02:36.17 | ElllisD | moocho |
| 02:36.30 | Katty | how about you do my dishes |
| 02:36.35 | Katty | and some laundry |
| 02:36.36 | voipmonk | what kind of virtualization did you use? |
| 02:36.40 | carrar | I do dishes now |
| 02:36.52 | carrar | and I look at photos of Mt Fuji |
| 02:36.52 | carrar | http://www.fujigoko.tv/live/camera.cgi?n=13 |
| 02:36.56 | ElllisD | Katty: lemme just wiggle my nose a bit, here- ala- done! |
| 02:37.02 | carrar | right click and select view image |
| 02:37.08 | Katty | ElllisD: baroo? |
| 02:37.08 | carrar | on the pic |
| 02:37.10 | carrar | looks much better |
| 02:37.33 | adnc | anyone using kabeldeutschland voip with asterisk? |
| 02:37.39 | carrar | I'm not |
| 02:37.46 | Katty | sounds german. |
| 02:37.52 | carrar | yeah |
| 02:37.56 | adnc | Katty, yes it is. |
| 02:38.00 | Katty | i'm so smrt |
| 02:38.34 | adnc | I've a box from my operator and i did read the config inside where it says reg01.kabelphone.de, unfortunately this host is not reachable |
| 02:38.39 | carrar | Achtung!!! |
| 02:38.41 | adnc | so my asterisk doesnt register there |
| 02:38.53 | Katty | what does achtung mean |
| 02:38.58 | carrar | Attention |
| 02:38.59 | adnc | Katty, attention |
| 02:38.59 | Katty | also |
| 02:39.03 | Katty | oh |
| 02:39.03 | Katty | k |
| 02:39.04 | ElllisD | voipmonk: Using KVM in ProxmoxVE- Lenny-based bare-metal hypervisor... Real happy with it's recently supporting shared memory. 4 Windows boxes using 1 box's worth of RAM :) |
| 02:39.44 | Katty | i only know a few german words. |
| 02:39.47 | voipmonk | hrmm - happy but dropping network ? |
| 02:39.56 | voipmonk | maybe i misread it |
| 02:40.12 | ElllisD | no- just the TBPro iso would- none else would |
| 02:40.14 | *** join/#asterisk etfonhomey (~etfonhome@74-131-159-160.dhcp.insightbb.com) |
| 02:40.16 | Katty | das ist mir furzegal |
| 02:40.23 | Katty | dummes huhn |
| 02:40.34 | ElllisD | Centos 5.4 holds up fine |
| 02:40.39 | Katty | gottverdammt |
| 02:40.40 | Katty | etc |
| 02:40.54 | Katty | oh, and lech mich |
| 02:40.58 | Katty | dont' ask. |
| 02:41.09 | carrar | CentOS kiX0r5 |
| 02:41.26 | ElllisD | The TBPro iso |
| 02:41.29 | ElllisD | is 4.4 |
| 02:41.35 | ElllisD | 4.x |
| 02:41.45 | carrar | FreeBSD K1X0r5 also |
| 02:41.55 | carrar | and OpenBSD |
| 02:42.00 | Katty | and your mom |
| 02:42.03 | carrar | yeah!! |
| 02:42.20 | ElllisD | carrar: liked FreeNAS, but couldn't resist block-level checksumming on NexentaStor |
| 02:42.50 | ElllisD | Never seen any other bsds |
| 02:43.01 | ElllisD | have seen - never tried |
| 02:43.12 | carrar | DRDB |
| 02:43.20 | Katty | LFG BRD |
| 02:43.28 | ElllisD | ..seems waay complicated |
| 02:43.31 | carrar | NFLNBA |
| 02:44.14 | Katty | waits for bbq comment |
| 02:44.14 | carrar | Tomorrow will be 60 here, I can BBQ in 60 degrees |
| 02:44.15 | Katty | i hate you |
| 02:44.15 | Katty | also |
| 02:44.15 | carrar | heh |
| 02:44.15 | Katty | i will be over for dinner. |
| 02:44.22 | *** join/#asterisk Terminus (~justin@116.50.188.3) |
| 02:44.22 | carrar | Bring NUTS!! |
| 02:44.24 | Katty | packs a bib. |
| 02:44.27 | Katty | k |
| 02:44.54 | ElllisD | Some random person on the bus yesterday morning gave me a bag of peanuts. |
| 02:45.00 | ElllisD | Funny you should mention |
| 02:45.13 | ElllisD | how odd |
| 02:45.13 | Katty | oh well it's not random, actually |
| 02:45.21 | Katty | http://www.ustream.tv/channel/squirrel-critter-cam |
| 02:45.26 | Katty | that's what it's not random |
| 02:45.43 | Katty | just pick a random video below |
| 02:46.07 | Katty | http://www.ustream.tv/recorded/4639158 <- like that one |
| 02:46.11 | ElllisD | http://chatroulette.com/ |
| 02:46.17 | Katty | i was there earlier tonight |
| 02:46.28 | carrar | CRASH CAM! |
| 02:46.36 | Katty | *hee* |
| 02:46.39 | Katty | i should totally rename it |
| 02:54.54 | jaytee | what is chat roulette? |
| 02:55.04 | carrar | die chatting |
| 02:55.06 | ElllisD | some stupid viral thing |
| 02:55.47 | Katty | basically webcam chat |
| 02:55.53 | Katty | but it's random people |
| 02:56.02 | carrar | is that with clothes on? |
| 02:56.02 | Katty | the website was built by a 17 year old from russia |
| 02:56.08 | ElllisD | jaytee: you're randomly connected to someone by webcam |
| 02:56.11 | Katty | well i've been going to reddit roulette |
| 02:56.19 | Katty | so yes, fully clothed :P |
| 02:56.27 | Katty | i had a nice conversation with a girl from Flordia |
| 02:56.35 | Katty | she's in college to be an english teacher and has a schnauser |
| 02:56.50 | jaytee | Flordia? is that in Italy? |
| 02:56.59 | *** join/#asterisk OrNix (~ornix@l151-249-47.static.cn.ru) |
| 02:57.07 | Katty | err Florida |
| 02:57.30 | ElllisD | http://www.warwickinfo.net/florida/index.html |
| 02:57.42 | ElllisD | Florida's in New York, silly |
| 02:58.01 | ElllisD | I crack myself up |
| 02:58.41 | jaytee | a legend in his own mind |
| 02:59.09 | Katty | at least it's entertaining in there |
| 03:00.33 | carrar | .. and warm |
| 03:00.35 | Terminus | hello. question, for a new PBX, which version is suitable for deployment, 1.4 or 1.6? |
| 03:00.40 | p3nguin_ | 1.4 |
| 03:00.54 | carrar | they all are |
| 03:01.20 | voipmonk | depends on the experience of your asterisk integrator |
| 03:01.29 | adnc | my asterisk crashes when i call in, which verbosity level would be good to look? would core deliver me some usefull information? |
| 03:01.31 | carrar | 1.0 !! |
| 03:01.35 | Katty | it also depends on your favorite color. |
| 03:01.46 | carrar | Orange? |
| 03:01.54 | Terminus | p3nguin_: thanks. |
| 03:01.58 | Katty | your favorite color is orange? |
| 03:02.04 | carrar | no |
| 03:02.05 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 03:02.09 | carrar | thats Marks |
| 03:02.11 | Terminus | voipmonk: that would be me with zero experience. i'm just checking it out right now. XD |
| 03:02.12 | carrar | heh |
| 03:02.29 | Katty | well what's your favorite color then |
| 03:02.30 | etfonhomey | What is the main purpose of the qualify=yes command in sip.conf? Is it used to keep dymanic NAT translations active? And/or dymanic holes in firewalls open? |
| 03:02.46 | p3nguin_ | terminus: 1.4 is LTS, 1.6.x kinda flops around a lot and should soon begin to fade away. |
| 03:03.07 | p3nguin_ | terminus: I've been told that 1.8 will also likely end up as LTS. |
| 03:03.17 | Terminus | p3nguin_: ok, guess i should change the version installed on my test box then. |
| 03:03.17 | carrar | Blue & Black |
| 03:03.27 | carrar | and crome |
| 03:03.27 | Katty | pretty |
| 03:03.34 | Katty | sounds very masculine |
| 03:03.37 | carrar | http://pics.osburn.com/photo/42851/original |
| 03:03.40 | carrar | heh |
| 03:03.47 | p3nguin_ | etfonhomey: It's good for keepalives. |
| 03:04.05 | Katty | carrar: what do you do when it rains? |
| 03:04.15 | carrar | get wet |
| 03:04.51 | carrar | take my truck or car |
| 03:04.51 | p3nguin_ | I'd ride it. |
| 03:05.03 | carrar | or stay home since I work from home |
| 03:05.06 | p3nguin_ | Rain is a bitch, though. |
| 03:05.21 | Terminus | we're thinking about changing the existing PBX to voip and i don't want to be stuck with vendor lock-in. |
| 03:05.31 | etfonhomey | p3nguin, for a small office network (4 phones) with no NAT is it really needed? |
| 03:05.40 | p3nguin_ | etfonhomey: nope |
| 03:05.47 | Katty | i drive a black eclipse |
| 03:05.59 | carrar | I drive a 31 year old car |
| 03:06.03 | carrar | 10 MPG!! |
| 03:06.12 | p3nguin_ | etfonhomey: The only use it would be for you is that you can see the time it takes for the phone to respond... which is kinda useless on a LAN. |
| 03:06.23 | Katty | http://gtcarlot.com/gallery/photo.php?id=409263 <- this particular one isn't mine, but you get the idea |
| 03:06.42 | carrar | http://pics.osburn.com/photo/43401/original |
| 03:06.43 | carrar | heh |
| 03:06.48 | carrar | 23k original miles |
| 03:07.00 | Katty | wow |
| 03:07.08 | Katty | my dad drives an old oldsmobile caprice |
| 03:07.13 | carrar | car was free, I couldn't pass it up |
| 03:07.15 | Katty | he refuses to part with it for mysterious unknown reasons |
| 03:07.17 | etfonhomey | p3nguin_, the reason I ask is that my phones keep saying they're lagged even though they are plugged directly into the same switch as the Asterisk box. I get sub 1ms ping times at the times that Asterisk says a phone is lagged. |
| 03:07.18 | p3nguin_ | A Toronado?! |
| 03:07.29 | carrar | yeah |
| 03:07.31 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 03:07.37 | etfonhomey | Well, the phones don't say they're lagged. sip show peers does. |
| 03:07.40 | p3nguin_ | katty: Olds didn't make a Caprice. |
| 03:08.08 | p3nguin_ | Olds Cutlass, Chevy Caprice |
| 03:08.15 | Katty | whatever |
| 03:08.19 | Katty | it's a caprice |
| 03:08.33 | carrar | & dinner, bbl |
| 03:08.53 | Katty | and it's maroon, and very old |
| 03:08.58 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 03:09.49 | p3nguin_ | I have an old Cutlass, but I don't drive it anymore. Haven't driven it for several years, but I can't make myself scrap it. |
| 03:10.15 | *** join/#asterisk Aristotles (~your@203.44.139.44) |
| 03:22.29 | *** join/#asterisk Chinorro (~Chino@202.219.27.77.dynamic.mundo-r.com) |
| 03:28.59 | *** join/#asterisk Kumbang (~kumbang@rusnas.paume.itb.ac.id) |
| 03:29.00 | adnc | <PROTECTED> |
| 03:29.03 | adnc | i get this error |
| 03:29.13 | adnc | does someone see what mistake i'm doing? |
| 03:32.42 | ManxPower-work | that means there is no route to the destination. |
| 03:32.57 | ManxPower-work | Does the device show up with an IP when you do a "sip show peers" |
| 03:40.45 | *** join/#asterisk lordmortis (~lordmorti@CPE-121-215-49-65.lnse1.pie.bigpond.net.au) |
| 03:42.36 | *** join/#asterisk lordmortis (~lordmorti@CPE-121-215-49-65.lnse1.pie.bigpond.net.au) |
| 03:43.08 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 03:45.46 | *** join/#asterisk styelz (~yoohoo@m0o0.mooo.com) |
| 03:53.58 | sbrath | In the Dial() application I can specify SIP/1234&SIP/2345 does that have limitations, or does it just ring both sip channels? |
| 03:55.28 | ManxPower-work | sbrath: Yes. |
| 03:55.42 | ManxPower-work | It has limitations and will ring both sip channels |
| 03:56.01 | sbrath | I tried to Ring a SIP/1234&DAHDI/g1/1234 and that didn't work. |
| 03:56.31 | ManxPower-work | that would work, assuming you are not using analog FXO ports or CAS T-1 |
| 03:57.22 | sbrath | I thought it would work too, but the call never "rang" on the other side of the DAHDI/g1 side, just the SIP side .. |
| 03:57.40 | *** part/#asterisk SeriousMatters (~Sirius@87.115.170.210) |
| 04:00.30 | ManxPower-work | sbrath: Then you have something else wrong |
| 04:00.59 | ManxPower-work | what type of line is GAHDI/g1? |
| 04:01.06 | ManxPower-work | LOL DAHDI, even |
| 04:05.33 | *** join/#asterisk Deeewayne (~dwayne@c-71-207-214-190.hsd1.al.comcast.net) |
| 04:05.34 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 04:13.05 | carrar | POTS LIKE |
| 04:13.07 | carrar | LINE :) |
| 04:17.57 | *** join/#asterisk maxagaz (~maxagaz@soho2.i-xanadu.com) |
| 04:23.15 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
| 04:41.34 | ManxPower-work | ~answers |
| 04:41.35 | infobot | [answers] 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 |
| 04:47.44 | *** join/#asterisk s2krish (~chatzilla@113.199.170.254) |
| 04:48.16 | s2krish | hi there |
| 04:50.15 | *** join/#asterisk Dibri (~gavit@190.98.32.116) |
| 04:50.23 | s2krish | hello |
| 04:51.38 | rue_house | hi |
| 04:53.58 | s2krish | does asterisk run on Ubuntu? |
| 04:54.48 | voipmonk | yes |
| 04:54.52 | voipmonk | it runs on debian, too |
| 04:54.53 | voipmonk | :) |
| 04:55.46 | s2krish | great |
| 04:55.58 | s2krish | googling i find this manual, https://wiki.ubuntu.com/AsteriskRecipeOnDapper |
| 04:56.06 | s2krish | can you suggest good one? |
| 04:56.22 | *** join/#asterisk came0 (~came0@34.124.188.72.cfl.res.rr.com) |
| 05:08.38 | *** join/#asterisk Jak_o_Shadows (~Fake@121.209.35.170) |
| 05:08.55 | voipmonk | install it from scratch |
| 05:08.57 | voipmonk | :) |
| 05:09.01 | voipmonk | get the source |
| 05:09.02 | voipmonk | compile |
| 05:09.04 | voipmonk | and have at it |
| 05:09.10 | ChannelZ | easy peasy |
| 05:09.43 | s2krish | :-) |
| 05:09.59 | s2krish | how to run test? do I need to configure any driver? |
| 05:10.35 | ManxPower-work | s2krish: Asterisk is a TOOLKIT that gives you the tools to design and build a PBX, much like a compiler and libraries let you build an application |
| 05:10.39 | ManxPower-work | You should read the Asterisk book |
| 05:10.41 | ManxPower-work | ~book |
| 05:10.42 | 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 |
| 05:14.22 | s2krish | thanks |
| 05:14.26 | s2krish | for the ~book |
| 05:31.18 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 05:34.03 | drmessano | <PROTECTED> |
| 05:34.07 | drmessano | Now pls |
| 05:35.34 | rue_house | would someone please add to the default voicemail.conf that the program called for mail notification is passed the context for the first paramiter, the box number for hte 2nd paramiter and the number of voiemails as the 3rd paramiter? |
| 05:35.35 | ManxPower-work | We ate the last one earlier this evening |
| 05:35.43 | rue_house | or do I need to file it as a bug and patch? |
| 05:36.01 | ManxPower-work | rue_house: your best bet is to file at issues.digium.com |
| 05:36.23 | ManxPower-work | I'm sure a patch would be appreciated, but just filing the report is good too. |
| 05:36.28 | rue_house | low low odds of me ever doing that |
| 05:36.44 | rue_house | I dont think its documented anywhere |
| 05:37.28 | rue_house | I worked it out and got my front door MWI leds working again, so I'm happy |
| 05:38.17 | rue_house | hmm, need to add an extension for knowing if there is a power failure too |
| 05:39.53 | rue_house | oooh, thats why I needed that ups cable... hmmm |
| 05:40.26 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 05:40.56 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 05:46.31 | *** join/#asterisk ruben23 (~AGENT@122.55.48.243) |
| 05:48.01 | *** join/#asterisk sun28 (~light@sun28.ipfw.su) |
| 05:56.39 | *** join/#asterisk D-Boy (~Geek@unaffiliated/cain) |
| 05:57.30 | D-Boy | Hi, what's the minimal requirement for installing asterisk ? |
| 05:59.04 | rue_house | definitly a self driven problem solver |
| 05:59.18 | rue_house | a stark streak of DARE |
| 05:59.42 | rue_house | drive, like a 'never give up' kind |
| 05:59.53 | *** part/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 06:00.04 | rue_house | like that animal that wont let go of what it bites till it dies |
| 06:00.25 | rue_house | oh and you need a computer |
| 06:00.47 | rue_house | helps to have more than one, but its not strictly needed |
| 06:01.25 | D-Boy | rue_house : i'm interested in this point :) the computer ^^ how to evaluate the needs ? (cpu, ram, DD...) |
| 06:01.45 | rue_house | depends if you insist on running windows on it |
| 06:02.19 | D-Boy | rue_house : i'll go with debian, (or centos) |
| 06:02.31 | rue_house | if you want to use it with windows, a dual quad core machine with 16G of ram, solid state drives the most powerfull video card you can get |
| 06:02.42 | rue_house | oh, then a PII-500 should suit you fine |
| 06:03.01 | D-Boy | rue_house : for 20 channels ? |
| 06:03.06 | rue_house | sure |
| 06:03.21 | rue_house | if you like |
| 06:03.22 | D-Boy | rue_house : any suggests about the ram ? |
| 06:03.35 | ChannelZ | Get some |
| 06:03.53 | rue_house | give it a ~1Ghz machine, my * box has 256M of ram, it don't touch swap |
| 06:04.02 | rue_house | I'm using 3% drive space.... |
| 06:04.13 | rue_house | 1.4G |
| 06:04.24 | rue_house | used, so give it atleast a 3G drive |
| 06:04.46 | rue_house | I suggest software mirrored raid, cause having the system go down on ya sucks |
| 06:05.02 | D-Boy | rue_house : cool computer needs solved, next question : an spa400 will works with asterisk without the spa9000 ? |
| 06:05.15 | rue_house | DO NOT USE DIGIUM CARDS! |
| 06:05.30 | D-Boy | rue_house : i'm not ! |
| 06:05.32 | rue_house | ECHO ECHo ECho Echo ... |
| 06:05.45 | rue_house | oh sorry, thats sipura |
| 06:05.48 | rue_house | hmm |
| 06:05.59 | rue_house | whats the 9000? |
| 06:06.11 | rue_house | isn't the spa400 a 4 channel carrier card? |
| 06:06.15 | D-Boy | yes |
| 06:06.24 | rue_house | so whats the 9000 |
| 06:06.51 | rue_house | HWEC? |
| 06:07.17 | D-Boy | the rue_house : http://www.voip-info.org/wiki/view/Linksys+SPA9000 |
| 06:07.45 | rue_house | when he comes back, ask [tk... russelb is another good person to ask, qwell keeps to himself I have to go to bed! :) |
| 06:08.28 | D-Boy | okey :) |
| 06:08.33 | D-Boy | rue_house : good night ^^ |
| 06:11.50 | *** part/#asterisk voipmonk (~shido6@12.50.179.194) |
| 06:18.26 | *** join/#asterisk Rajmohan (~raj@122.164.185.8) |
| 06:33.16 | *** part/#asterisk Tech_Travis (~Administr@cpe-76-168-191-127.socal.res.rr.com) |
| 06:42.55 | *** join/#asterisk Kumbang (~kumbang@rusnas.paume.itb.ac.id) |
| 06:50.01 | *** join/#asterisk soman (~somnath@stargate.starnet.fi) |
| 07:02.14 | *** join/#asterisk maxagaz (~maxagaz@soho2.i-xanadu.com) |
| 07:05.52 | *** join/#asterisk lordmortis (~lordmorti@203-206-117-94.dyn.iinet.net.au) |
| 07:11.13 | *** join/#asterisk kotp (~vgoff@198.174.233.2) |
| 07:11.21 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 07:13.58 | *** join/#asterisk radcliff (~radcliff@h-63-22.A259.priv.bahnhof.se) |
| 07:20.03 | *** join/#asterisk e-jones (~jkastner@nat/redhat/x-wwkdclhaolrntsfp) |
| 07:20.12 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-gfeschrrnspidysa) |
| 07:21.18 | *** join/#asterisk benngard (~benngard@213.88.138.230) |
| 07:43.03 | kotp | What is a good resource for quickly learning Asterisk? |
| 07:43.50 | *** join/#asterisk ThoMe (tm@tm.muc.de) |
| 07:43.54 | ThoMe | hello! |
| 07:44.00 | *** join/#asterisk smooth_penguin (~smoove@59.99.105.4) |
| 07:44.03 | s2krish | hi |
| 07:44.15 | kotp | Hello |
| 07:44.19 | s2krish | get this book |
| 07:44.20 | s2krish | http://www.asteriskdocs.org/ |
| 07:44.47 | s2krish | The Future of Telephony 2nd Edition |
| 07:45.00 | s2krish | ThoMe: you can get free PDF download |
| 07:45.05 | ThoMe | me? |
| 07:45.30 | s2krish | kotp: sorry to kotp |
| 07:45.43 | kotp | Thanks... i will definitely check that out. Is that going to get me from asterisk (unknown for the most part) to Adhearsion (good it is with Ruby, that is a known) to Rails (probably not a difficult jump as Rails is known, asterisk and adhearsion not so much) |
| 07:46.12 | yang | ~book |
| 07:46.13 | 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 |
| 07:46.15 | s2krish | u'll get kick start and know how to get going |
| 07:48.10 | kotp | awesome... Thank you! |
| 07:48.47 | *** join/#asterisk sgimeno (~chatzilla@226.Red-80-33-64.staticIP.rima-tde.net) |
| 07:48.56 | kotp | such a friendly bot... first time I had seen that |
| 07:49.49 | kotp | yang: thanks for triggering |
| 07:50.00 | FSB_1 | What do I need to do to disallow people to call out? |
| 07:50.06 | kotp | s2krish: thank you as well |
| 07:50.46 | *** join/#asterisk basty (~basty@212.218.65.163) |
| 07:50.49 | basty | Good Morning |
| 07:52.41 | basty | I have a little Problem with my Asterisk 1.4.29. I used to update from 1.2.X to the latest 1.4.29. The migration went well - everything works fine, but I still have following problem. If an extension tries to setup a static call forwarding (on the phone) to another internal extension. The call is getting redirected - but the caller does not hear a ringing. Could anyone please give me a hand ? My SIP-Conf is on : http://pastebin.com/d4d82e8b0 |
| 07:54.51 | *** join/#asterisk icyValk77 (~icyValk77@gateway.ash.thebunker.net) |
| 08:01.33 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 08:05.30 | s2krish | how to add user or change user password? |
| 08:05.37 | carrar | useradd |
| 08:05.42 | carrar | passwd username |
| 08:05.58 | s2krish | oh that works like adding linux users |
| 08:06.06 | s2krish | carrar: thanks |
| 08:06.08 | carrar | no, that *IS* linux |
| 08:06.29 | s2krish | carrar: is apply same in asterisk |
| 08:06.45 | carrar | What are you talking about |
| 08:06.58 | s2krish | I am talking about adding user in Asterisk |
| 08:07.08 | carrar | adding a user? |
| 08:07.31 | s2krish | yup.. what if you want to allow to make call to authorized users only |
| 08:07.55 | carrar | I create SIP users |
| 08:08.00 | carrar | with passwords |
| 08:08.08 | carrar | SIP Devices |
| 08:08.36 | ThoMe | my asterisk said: |
| 08:08.36 | ThoMe | [Feb 18 09:07:53] NOTICE[19776]: chan_sip.c:8014 sip_reg_timeout: -- Registration for '089410748490002@193.108.19.205' timed out, trying again (Attempt #4) |
| 08:08.43 | s2krish | is there not I can create users with APIs, like I wan to add users from web based applicaiton |
| 08:08.58 | s2krish | i have call center application that going to build in PHP |
| 08:09.10 | kaldemar | s2krish: what kind of users are you talking about? |
| 08:09.17 | carrar | heh |
| 08:09.38 | s2krish | and when creating users in my PHP build application, I want to user to add in Asterisk |
| 08:09.49 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 08:09.53 | carrar | DEFINE: "user" |
| 08:10.08 | carrar | DEFINE: "user in asterisk" |
| 08:10.31 | s2krish | person who can call using connected softphone |
| 08:10.47 | carrar | So a SIP Device |
| 08:10.51 | kaldemar | i'd like to say that there no such concept as "asterisk user", but users.conf would bite that statement. however, that's the way to trouble. |
| 08:10.53 | carrar | not a "user" |
| 08:11.06 | kaldemar | he didn't say SIP *g* |
| 08:11.35 | carrar | softphones are typically SIP :) |
| 08:11.41 | s2krish | okay |
| 08:11.44 | carrar | BUT NOT always |
| 08:12.09 | s2krish | suppose u have 10 person in call center, how you track who has made call if there's no user system? |
| 08:12.11 | carrar | s2krish, I'm assuming you mean SIP Device |
| 08:12.26 | carrar | You trace SIP Devices |
| 08:12.28 | s2krish | is that by SIP Device ID? |
| 08:12.30 | carrar | track |
| 08:12.34 | carrar | yes |
| 08:13.14 | carrar | SIP Device name |
| 08:13.20 | carrar | like 200 |
| 08:13.24 | s2krish | okay so when I want keep record about employee A's call, i sud tap him with SIP Device name |
| 08:13.28 | carrar | or USER1024 |
| 08:13.35 | carrar | whatever ID you pick |
| 08:13.36 | kaldemar | carrar: sure, most are, but there are ones for every protocol and as you know, details matter in these things. :) |
| 08:14.26 | s2krish | is there any authentication in SIP kinds of password, so only person who know password can make call using device 101 |
| 08:14.35 | carrar | s2krish, you will probably send your CDR (call detail records) to a database (postgres is best :) and then create report on calls |
| 08:14.44 | carrar | or your queue log |
| 08:14.47 | carrar | whatever you want to report on |
| 08:14.58 | kaldemar | s2krish: yes there is. you'll find that and much more in the book you advertised here earlier. :) |
| 08:15.06 | carrar | yes multiple ways to authenticate SIP Devices |
| 08:15.06 | s2krish | my scenario is I want to restrict access ( or ability to call) |
| 08:15.13 | carrar | ~book |
| 08:15.14 | 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 |
| 08:15.18 | carrar | READ First |
| 08:15.18 | s2krish | so user with password can make call. |
| 08:15.21 | carrar | READ First |
| 08:15.35 | s2krish | :-) |
| 08:15.49 | carrar | I want to help you help yourself |
| 08:16.13 | carrar | It's possible |
| 08:16.18 | carrar | It's ALL possible |
| 08:16.41 | s2krish | carrar: thanks, i have downloaded that book n going on pages, thanks for suggesion. i gonna go mug on it again.. |
| 08:16.57 | carrar | mug away! |
| 08:18.07 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 08:19.16 | dzup | wow, am new in asterisk and i handle my install from scratch on a centos machine 2 weeks ego, now i have 45 extentions, i read some books already, but even that questions wasnt for me, thanks carrar i will read that too :) |
| 08:19.59 | dzup | i didnt know about that free book :) |
| 08:20.14 | benngard | ~phones |
| 08:20.14 | infobot | While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Linksys SPA-9XX, Snom, Cisco 79XX, everything else. Do not consider Grandstream phones. Ever. |
| 08:21.02 | carrar | Should be a auto msg for nicks that have never joined |
| 08:21.11 | carrar | heh |
| 08:23.12 | dzup | carrar: what am looking for right now is a webpanel written in php/mysql that handles billings to costumers by the minute and dial zones :( |
| 08:23.39 | carrar | dialzones? |
| 08:23.47 | carrar | You mean Rate Centers? |
| 08:24.25 | dzup | more like if i have some costumers in diferrent cities, to be able to bill then by the usage |
| 08:24.37 | carrar | I wrote all our billing in PSQL |
| 08:25.25 | carrar | You will need to know the rate per min for calls to eash Rate Center (RC) |
| 08:25.27 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 08:25.38 | carrar | rate or whatever increment of time |
| 08:25.55 | carrar | I assume you already have that db table? |
| 08:25.56 | dzup | hmm, i search some projects on sourceforge and i found one in java, i hate java 'cuz i have a modest computer and java runs very slow |
| 08:26.43 | FSB_1 | I have a hard time finding out how to disallow outbound calls and to tonly allow it from just one extension. |
| 08:27.02 | carrar | context control |
| 08:27.04 | FSB_1 | Does anyone have a link to a tutorial or maybe keywords to google for? |
| 08:27.22 | dzup | carrar: yes i have all that |
| 08:27.30 | dzup | carrar: in mysql |
| 08:27.44 | carrar | FSB, put phones that get outbound access in a context that has those permissions |
| 08:28.12 | carrar | or more appropriately, include that context |
| 08:28.45 | carrar | So then you can have the CDR |
| 08:28.53 | carrar | you have your rates for called RC |
| 08:29.07 | carrar | You just need to write some php |
| 08:29.14 | carrar | php is fun |
| 08:29.24 | carrar | and write some SQL queries |
| 08:29.28 | dzup | yeah i like php :) |
| 08:29.28 | carrar | maybe create some indexes |
| 08:29.28 | kotp | ~buybook |
| 08:29.29 | infobot | [~buybook] You can buy "Asterisk the Future of Telephony" at http://www.oreilly.com/catalog/9780596510480/ so go buy it SERIOUSLY |
| 08:30.46 | carrar | Letting your customers crunch your db for bill records might not be the best idea |
| 08:31.31 | carrar | but you probably have planned for that |
| 08:31.54 | *** join/#asterisk Great_Anta_Baka (~tensai@41.0.201.194) |
| 08:32.23 | carrar | FSB_1, read the book, it has exampels |
| 08:32.27 | carrar | ~book |
| 08:32.28 | 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 |
| 08:32.33 | dzup | really i need to control my billing, customers dont need to know |
| 08:32.48 | carrar | then just automate it once a month |
| 08:32.52 | carrar | no need to php for that |
| 08:33.00 | carrar | no need for php for that |
| 08:33.16 | FSB_1 | carrar: Oooh, thanks! |
| 08:35.08 | visz | dzup, check out mor billing |
| 08:35.10 | visz | http://www.voip-info.org/wiki/view/MOR |
| 08:35.32 | visz | it's written in ruby so modifying the code is loads of fun |
| 08:35.43 | ThoMe | have connection problems to a external registrar. internal no problem |
| 08:35.57 | ThoMe | http://paste.keks.be/9484/txt |
| 08:36.12 | *** join/#asterisk Da-Geek (~Da-Geek@62.189.17.99) |
| 08:36.27 | carrar | heh |
| 08:36.49 | dzup | visz: thanks i will, thanks carrar again, 2 more nice stuff to read :) |
| 08:37.14 | kaldemar | i did cdr parsing in bash once. now that was fun. |
| 08:37.21 | carrar | haha |
| 08:37.28 | ThoMe | [Feb 18 09:36:27] NOTICE[21181]: chan_sip.c:8014 sip_reg_timeout: -- Registration for '089410748490002@193.108.19.205' timed out, trying again (Attempt #1) |
| 08:37.31 | ThoMe | hmm |
| 08:38.56 | ChannelZ | It says what it means, means what it sayd. |
| 08:39.07 | ChannelZ | says even |
| 08:39.24 | carrar | but ChannelZ!!! Did you look at the SIP Trace!! |
| 08:39.43 | carrar | perhaps I should go to bed |
| 08:39.58 | *** join/#asterisk mikkel (~mikkel@84-238-113-66.u.parknet.dk) |
| 08:42.08 | *** join/#asterisk peacetreaty (~peacetrea@chello062178193036.3.15.vie.surfer.at) |
| 08:42.16 | peacetreaty | good morning |
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| 08:44.11 | *** join/#asterisk arapaho (~arapaho@pierre.infomaniak.ch) |
| 08:44.45 | peacetreaty | i have some problem with an asterisk pbx and isdn; everything seems to work but some callers don't come with an extension when they are calling to my pbx; but they are dialing the extension too; that's why fax is also not working from those persons |
| 08:46.51 | *** join/#asterisk soman (~somnath@stargate.starnet.fi) |
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| 09:28.46 | c0rnoTa | good day, all |
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| 09:36.37 | *** join/#asterisk hipitihop (~denis@203.132.229.187) |
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| 09:38.51 | hipitihop | I'm just getting my feet wet with first install and following the initial configuration according to the 2nd edition book.... cna someone pls clear up if the zaptel.conf is part of the standard or only if you have specific hardware ? |
| 09:40.16 | *** join/#asterisk icyValk77 (~icyValk77@cl-670.lon-02.gb.sixxs.net) |
| 09:40.32 | c0rnoTa | hipitihop: zaptel.conf only nedded i you have telephony cards |
| 09:41.36 | hipitihop | c0rnoTa, thought as much, I only have a sip account and an ATA on the network which has 2x FXS ports. |
| 09:42.09 | kaldemar | also, zaptel was renamed to DAHDI in 2008. the book was written before that. |
| 09:42.26 | hipitihop | kaldemar, thanks for the clarification |
| 09:44.35 | hipitihop | so do I define my FXS port on the ata in the extensions.conf ? |
| 09:45.42 | *** join/#asterisk c0rnoTa (~c0rnoTa@94.25.175.235) |
| 09:47.15 | kaldemar | hipitihop: the ATA is defined as a SIP device in sip.conf. extensions.conf has the dialplan, which tells asterisk what to do when it receives calls. you have to modify both. |
| 09:47.20 | kaldemar | ~book |
| 09:47.21 | 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 |
| 09:47.55 | kaldemar | ^ in case you haven't seen that, it's a good read with clear examples. |
| 09:49.16 | hipitihop | kaldemar, sorry, but that is the book I am trying to follow. so in my case it sounds like that both my sip account account details and the ATA will be defined in sip.conf ... sorry for the newbie questions |
| 09:50.35 | kaldemar | yes, both are defined in sip.conf. |
| 09:51.26 | *** join/#asterisk lordmortis (~lordmorti@203-206-117-94.dyn.iinet.net.au) |
| 09:52.14 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 10:01.30 | basty | anyone familar with using the "Pickup()" on Asterisk 1.4.29 ? I just setup this like : http://pastebin.com/d68ffd40e |
| 10:02.00 | basty | problem is, that it seems, that I cant pickup that channel "app_directed_pickup.c: No target channel found for " |
| 10:06.11 | kaldemar | basty: is someone calling 86@intern when that happens? |
| 10:06.55 | basty | kaldemar: yep..the extension 86 was rining and 87 tried to pickup that chan |
| 10:07.46 | basty | kaldemar: 86 was getting an external call and 87 tried to pickup |
| 10:08.12 | kaldemar | well, i can't see that in your paste |
| 10:09.52 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 10:11.51 | basty | kaldemar: hrm..can I use "Pickup(SIP/${nst})" also ? |
| 10:12.48 | *** join/#asterisk TimeRider (~steve@78.32.26.1) |
| 10:13.18 | kaldemar | basty: no. it uses an extension. do you have extension 86 in context intern that dials SIP/86? |
| 10:13.39 | basty | kaldemar: yep |
| 10:14.14 | kaldemar | feel free to show the whole calls. |
| 10:17.49 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 10:17.57 | basty | kaldemar: http://pastebin.com/d7a8dbe94 |
| 10:19.19 | kaldemar | the extension that could be picked up is 9617305@s2m, and you're trying on 100@intern. |
| 10:20.02 | kaldemar | SIP/100 is not an extension, it is a device. |
| 10:21.12 | basty | kaldemar: mh...so how could I make it work for internal and external? SIP 100 has an external like 9617305@s2m and internal like 100@intern |
| 10:21.24 | *** join/#asterisk icyValk77 (~icyValk77@gateway.ash.thebunker.net) |
| 10:23.39 | FSB_1 | carrar: Thank you for the link to the book. It really helps me with the understanding of asterisk. |
| 10:23.55 | kaldemar | basty: [s2m]: exten => 9617305,1,Goto(intern,100,1) |
| 10:25.50 | kaldemar | basty: you will do yourself a favor if you only have one extension per device that does the actual calling (i.e. executes the Dial app). |
| 10:26.45 | basty | kaldemar: well..actually its a test extension, because the acutally problem is, that I have a little hotline with some agents in it... |
| 10:26.47 | hipitihop | so if I run "sip show peers" or a specific device... this is guaranteed that * can see the device ? |
| 10:27.02 | *** join/#asterisk sky1975 (~Administr@ap328-f.u-aizu.ac.jp) |
| 10:27.09 | sky1975 | Hi |
| 10:27.11 | basty | kaldemar: there is a hotline number 9617300 that goes into a queue...behind the queue there are some agents with "add queue member" |
| 10:27.32 | kaldemar | hipitihop: no, its existance only tells that you have some configuration for it in sip.conf. |
| 10:28.32 | sky1975 | can anybody tell me hot to stop server generated ringing tone and let customer to hear original ringing tone |
| 10:29.39 | kaldemar | sky1975: remove option r from app Dial |
| 10:29.39 | hipitihop | kaldemar, I see, so even results from a "sip show peer mydeviceid" is no guarnateed it can actually see it |
| 10:30.14 | sky1975 | kaldemar I am a newbie. App Dial means? |
| 10:30.24 | sky1975 | where is it? |
| 10:30.33 | *** join/#asterisk techie (~root@unaffiliated/techie) |
| 10:30.37 | kaldemar | sky1975: the application in your dialplan. do you know what your dialplan is? |
| 10:30.53 | kaldemar | hipitihop: what do you mean by "see"? |
| 10:31.12 | sky1975 | I added them via a2billing |
| 10:31.27 | *** join/#asterisk fofware (~chatzilla@190.31.195.233) |
| 10:31.32 | sky1975 | If you tell me the file, I will be able to change it |
| 10:32.32 | kaldemar | hipitihop: if there are results, you have configured the peer, but it doesn't guarantee that the peer is working. if you defined the peer with host=dynamic and you see an IP address for it, then it would be registered. |
| 10:32.46 | kaldemar | sky1975: /etc/asterisk/extensions.conf by default. |
| 10:32.56 | *** join/#asterisk romain (~romain@peerfuse.org) |
| 10:33.07 | hipitihop | well.. I know my ATA is functional... but currently setup to talk directly via my sip provider... I setup peer with specific address as opposed to dynamic |
| 10:34.33 | sky1975 | kaldemar: thanks |
| 10:34.49 | adnc | when i say sip show registry i see my username followed by an at sign like this 0522191919@ is this normal? the rest doesnt have this |
| 10:34.50 | hipitihop | kaldemar, I guess I have to keep going through the book as I also need to configure sip.conf for my provider ... and then I guess I will re-configure my ATA to point at asterisk as opposed to my real provider |
| 10:35.10 | basty | kaldemar: i have setup "exten => _**8X.,1,Pickup(${EXTEN:3}@intern) and exten => _**8X.,2,Pickup(9617${EXTEN:3}@s2m)" - now it seems to work |
| 10:35.37 | sky1975 | One more thing. When we dial from x-lite, time conter goes from the time connected to Astersk. Can we change it so that it will show only when the destination is connected |
| 10:35.39 | *** join/#asterisk sun28 (~light@sun28.ipfw.su) |
| 10:36.57 | basty | now..I have another problem with my Asterisk 1.4.29. I used to update from 1.2.X to the latest 1.4.29. The migration went well - everything works fine, but I still have following problem. If an extension tries to setup a static call forwarding (on the phone) to another internal extension. The call is getting redirected - but the caller does not hear a ringing. Example: Extension 100 (setup a call forward to 101) Extension 101 Now - if I call 100 - the call i |
| 10:37.45 | kaldemar | basty: well that works, but now if you try to pick up an internal call with that, it won't. |
| 10:41.22 | basty | kaldemar: mhh..okay...i will see ;) |
| 10:41.34 | basty | kaldemar: do you have an idea about my other annyoing problem ? |
| 10:42.35 | kaldemar | sky1975: that may get fixed when you remove the r option. btw, you may not find the Dial() in extensions.conf because of a2billing. it may use an agi, a2billing.php that makes the call. you may have to modify it. i'm not that familiar with a2billing. |
| 10:44.03 | sky1975 | ok. thanks |
| 10:44.03 | sky1975 | I will try to do it via a2billing |
| 10:44.03 | *** join/#asterisk elwerene (~lalala@vpn-s-8d3a31ee.campus.uni-stuttgart.de) |
| 10:44.03 | kaldemar | basty: based solely on that, no. |
| 10:47.31 | FSB_1 | Can an extension belong to multiple contexts? |
| 10:47.48 | *** join/#asterisk Akiraa (~Akira@92.81.197.69) |
| 10:48.37 | *** join/#asterisk Dibri (~gavit@pop1.isgroup.sr) |
| 10:48.40 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 10:50.46 | elwerene | heyho, i am trying to build an asterisk as voip gateway for my student living community. I am having big problems in setting up the asterisk correctly so that incomming calls from voip ring on the dect phones at a siemens gigaset isdn pbx. |
| 10:50.57 | elwerene | My asterisk structure: voip <-> asterisk <-> 3*hfc-s card(dahdi) <-> siemens gigaset pbx |
| 10:51.10 | elwerene | i have no problems with outgoing calls(isdn -> asterisk -> voip). Has somebody the muse to help me? :) |
| 10:53.28 | basty | kaldemar: do you think it could be a problem on my sip settings ? http://pastebin.com/d46280f9c |
| 10:53.32 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 10:57.30 | kaldemar | basty: can't say without knowing what really happens during a call. |
| 10:58.49 | kaldemar | FSB_1: an extension is an exten defined in the dialplan. it can only belong to a single context. however, you can include a context in another with include => somecontext. |
| 10:59.09 | basty | kaldemar: Let me explain the situation. I have extension 100 and 101. Extension 100 setup a call forward to 101 on his phone. As soon as I try to call 100 - the call is getting forwared to 101 - the phone of 101 is ringing and you can accept...the voice is working. BUT the caller, that called 100 is not hearing a "Rining-Tone" while its ringing on the 101 extension. |
| 11:00.28 | kaldemar | basty: i undestand the situation. but a description of the situation doesn't tell what really happens during a call. a cli output with sip debug and the dialplan would tell. |
| 11:00.43 | basty | kaldemar: sec |
| 11:02.13 | FSB_1 | kaldemar: I tried setting up in sip.conf a context that has the name special which has include=>dialout and for dialin with my sip-provider., it also belongs to internal, where my internal sipaccount is at. |
| 11:02.20 | FSB_1 | But I get extension not found. |
| 11:03.41 | basty | kaldemar: 100 called 101 - 101 has a call forward to "5" where a Playback is running....100 cant hear the playback: http://pastebin.com/d5eaf5d95 |
| 11:03.57 | kaldemar | FSB_1: pastebin the configuration |
| 11:05.02 | basty | kaldemar: same would happen, if 101 would forward the call to a other extension....100 wouldnt hear the ringing. |
| 11:08.13 | *** join/#asterisk e-jones (~jkastner@nat/redhat/x-rzwctpoxeetslxqu) |
| 11:08.52 | kaldemar | basty: force asterisk to stay on the media path with canreinvite=no or directmedia=no depending on your version. |
| 11:09.02 | kaldemar | then try again. |
| 11:09.26 | *** join/#asterisk soman (~somnath@stargate.starnet.fi) |
| 11:10.57 | basty | kaldemar: the sip accounts do have both "canreinvite=no" http://pastebin.com/m72dfb491 |
| 11:11.30 | FSB_1 | kaldemar: extensions.conf ? |
| 11:12.39 | kaldemar | FSB_1: and sip.conf |
| 11:13.28 | FSB_1 | kaldemar: Got it working. |
| 11:13.37 | FSB_1 | I'm just starting out with this. |
| 11:13.55 | FSB_1 | I just understood that context in sip.conf belongs to context in dialplan. |
| 11:14.10 | FSB_1 | Now I am the only one allowed to make outbound calls. |
| 11:14.13 | FSB_1 | Pretty cool. |
| 11:15.02 | elwerene | how can i activate usefull debuging of dahdi? pri intense doesn't help me. To cryptic.. |
| 11:15.10 | *** join/#asterisk thazza (~thazza@124-254-81-140-static-dsl.ispone.net.au) |
| 11:15.36 | thazza | Hey all. Anyone here know much about the Asterisk manager system? |
| 11:15.50 | kaldemar | elwerene: core set verbose, core set debug, pri debug and pri intense debug is pretty much all you get. |
| 11:16.21 | kaldemar | basty: well, what's left is to show the sip debug i asked you earlier. |
| 11:16.22 | elwerene | damn. There are no errors at all :( |
| 11:16.28 | thazza | I am attempting to get either ADA or OutCall to work. |
| 11:16.37 | thazza | Can't get outbound calls happening. |
| 11:16.44 | kaldemar | elwerene: what's the issue? |
| 11:17.24 | elwerene | i am trying to call from voip to a dect phone at my isdn pbx, but it's not ringing |
| 11:17.45 | *** join/#asterisk diatonic (~diatonic@mail.clearwater-research.com) |
| 11:17.48 | elwerene | the isdn pbx is connected to a hfc-s card(dahdi) in bri net ptmp mode |
| 11:18.00 | *** join/#asterisk garymc (~chatzilla@host86-164-36-128.range86-164.btcentralplus.com) |
| 11:21.58 | basty | kaldemar: http://pastebin.com/d489a8e20 |
| 11:22.13 | elwerene | kaldemar: http://pastebin.com/d50a8e849 |
| 11:22.41 | *** join/#asterisk soman (~somnath@stargate.starnet.fi) |
| 11:25.39 | thazza | All good.. worked out what was wrong |
| 11:30.01 | kaldemar | basty: can you see the rtp packets going to the client? |
| 11:30.39 | basty | kaldemar: how can i check that ? |
| 11:31.43 | kaldemar | basty: rtp set debug on or for example tcpdump |
| 11:31.54 | basty | ah okay |
| 11:33.11 | kaldemar | elwerene: looks decent. what do you see when making a call? |
| 11:37.11 | elwerene | kaldemar: http://pastebin.com/d71e0d91c |
| 11:40.56 | kaldemar | elwerene: looks like your PBX is not at all reacting to what asterisk sends it. |
| 11:41.30 | elwerene | kaldemar: any ideas? Outgoing call are working without a problem. |
| 11:44.44 | kaldemar | elwerene: in your system.conf you have "span=2,1,0,ccs,ami". iirc, that means you'd take timing off the line. however you're in NT mode and afaik PBX's don't usually provide timing but they attempt to take it off the line. try to change that line to "span=2,0,0,ccs,ami". don't know if the HFC card can be used to provide timing though. |
| 11:47.53 | kaldemar | it's been years since i've played with those. |
| 11:49.17 | *** join/#asterisk fskrotzki (~fskrotzki@cpe-74-74-245-250.rochester.res.rr.com) |
| 11:49.46 | elwerene | they are cheap, that's their quality :) |
| 11:51.00 | *** join/#asterisk asteriskATmarmuD (~mundt@193.158.65.23) |
| 11:51.18 | *** join/#asterisk creativx (~creadurex@197.82-134-19.bkkb.no) |
| 11:52.27 | elwerene | i changed the timing preferences. Still the same issue |
| 11:52.57 | *** part/#asterisk asteriskATmarmuD (~mundt@193.158.65.23) |
| 11:55.58 | *** join/#asterisk _omer (~omer@119.153.174.116) |
| 11:56.41 | _omer | hello, I have installed Asterisk 1.4.29 and asterisk-addons .... now how do I know that my asterisk supports H323 ? can I check it with any CLI Command? |
| 11:58.18 | elwerene | _omer: core show codecs |
| 11:58.45 | _omer | but it is to check codecs .. isn't it? |
| 12:03.42 | elwerene | _omer module load chan_h323.so |
| 12:04.23 | *** join/#asterisk dlynes (~dlynes@bas6-hamilton14-1176462822.dsl.bell.ca) |
| 12:06.12 | _omer | ok let me check |
| 12:06.18 | *** join/#asterisk maxagaz (~maxagaz@soho2.i-xanadu.com) |
| 12:06.39 | _omer | <PROTECTED> |
| 12:07.13 | elwerene | _omer: No h323 i guess |
| 12:07.23 | _omer | when I do make menuselect in asterisk-addons ... then I can see [x] with H323 |
| 12:07.42 | _omer | and I have installed asterisk-addons... |
| 12:07.58 | _omer | doesn't it mean , h323 is installed? |
| 12:08.03 | elwerene | kaldemar: Another idea for my problem? I'm searching the net since 4 days.. |
| 12:08.45 | *** join/#asterisk asteriskATmarmuD (~mundt@193.158.65.23) |
| 12:09.09 | _omer | elwerene: [Feb 18 12:05:07] WARNING[21257]: loader.c:735 load_resource: Module 'chan_ooh323.so' already exists. |
| 12:09.14 | *** part/#asterisk asteriskATmarmuD (~mundt@193.158.65.23) |
| 12:09.26 | _omer | I am getting this result from module load chan_ooh323.so |
| 12:09.39 | elwerene | _omer: so you have the other h323 module |
| 12:09.42 | *** part/#asterisk thazza (~thazza@124-254-81-140-static-dsl.ispone.net.au) |
| 12:13.07 | _omer | but I dont see any command to check h323 channels ...etc |
| 12:15.02 | *** join/#asterisk Dovid (~annon@tony09-118-62.inter.net.il) |
| 12:17.12 | *** join/#asterisk hluesea (~hulusikah@88.247.127.66) |
| 12:19.06 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:19.07 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:21.21 | *** join/#asterisk michael-i (~michael-i@141.41.40.185) |
| 12:25.27 | *** join/#asterisk Victor_Yure (~victor@unaffiliated/victoryure/x-837844) |
| 12:26.09 | michael-i | Hi everyone. I'm wanting to write some dialplan logic which will read out the system's IP address. Has anyone already done this? My first step is to store a shell command output into a dialplan variable. Any pointers? |
| 12:26.21 | *** join/#asterisk oej (~olle@static-213-115-251-100.sme.bredbandsbolaget.se) |
| 12:28.09 | *** part/#asterisk sky1975 (~Administr@ap328-f.u-aizu.ac.jp) |
| 12:36.16 | *** join/#asterisk SeriousMatters (~Sirius@87.115.170.210) |
| 12:37.28 | *** part/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:37.37 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:37.37 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:38.08 | leifmadsen | michael-i: if using 1.6.x (I can't remember which version... possibly 1.6.0) you could save the output of a shell command via the SHELL() function |
| 12:39.15 | leifmadsen | michael-i: alternatively, it'd probably be a pretty simple AGI script |
| 12:39.27 | michael-i | leifmadsen: just what I was looking for. I was System()ing output into a file and using readfile. |
| 12:39.33 | *** join/#asterisk razu (~razu@razu.data.ee) |
| 12:39.47 | leifmadsen | michael-i: ya, that might be necessary on 1.4 as SHELL() doesn't exist there |
| 12:40.12 | michael-i | leifmadsen: This is just a band-aid for my next release...one of my systems has no serial console. I'm using 1.6.1 so this is perfect. |
| 12:41.03 | leifmadsen | michael-i: ifconfig eth0 |grep "inet addr" |awk '{print $2}' |awk -F: '{print $2}' |
| 12:41.25 | leifmadsen | Set(IPADDR_RESULT=${SHELL(ifconfig eth0 |grep "inet addr" |awk '{print $2}' |awk -F: '{print $2}')}) |
| 12:41.35 | *** join/#asterisk smooth_penguin (~smoove@59.99.105.4) |
| 12:41.35 | leifmadsen | you may need to escape characters or something in that (untested) |
| 12:42.11 | michael-i | google gave me this awk-fu: ifconfig eth0 | awk '/dr:/{gsub(/.*:/,"",$2);print$2}' |
| 12:42.50 | michael-i | better than I could ever do! Now I just need to feed that through cut to get the first, second, etc bits of the address and use say digits |
| 12:42.55 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 12:42.58 | michael-i | not perfect but it solves the problem |
| 12:43.15 | michael-i | leifmadsen: thanks for the info! |
| 12:43.45 | leifmadsen | michael-i: ya, the ifconfig grep stuff was googled as well :) |
| 12:44.19 | michael-i | leifmadsen: I have never been the shell string manipulation master....always resort to google |
| 12:44.41 | leifmadsen | michael-i: btw, if you replace 'eth0' with $IFACE it'll give you all the IPs on the box (separate lines, so that might not be useful in the dialplan, unless CUT() can separate on newlines) |
| 12:44.47 | *** join/#asterisk Poincare (~jefffnode@213.219.184.23) |
| 12:46.03 | michael-i | leifmadsen: perfect. This is to be used to accomplish the first login. These systems never have multiple IPs but may have a different default IF name. |
| 12:47.49 | Dovid | morning ev1 |
| 12:50.29 | *** join/#asterisk krynnotaur (~krynnotau@unaffiliated/krynnotaur) |
| 13:19.18 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 13:20.13 | *** join/#asterisk Ad-Hoc (~nimbus@194.219.187.115.dsl.dyn.forthnet.gr) |
| 13:28.40 | *** join/#asterisk cfh (~luca@host196-137-static.90-82-b.business.telecomitalia.it) |
| 13:29.18 | elwerene | Anyone an idea why my isdn pbx refuses to answer to call requests from asterisk? Calls from pbx to asterisk are working fine. |
| 13:32.06 | cfh | hi all, i m using asterisk 1.6.2 with : dahdi 2.2.0 + ISDN card b410 by digium . It works well :) but when it recived a call the callerid is without the first 0 (ex: real callerID 0212345678, cid in astersik 212345678). |
| 13:33.00 | cfh | but I prefer the real CID on asterisk |
| 13:33.07 | cfh | what can i do ? |
| 13:33.25 | [TK]D-Fender | cfh: Go add the 0 yourself |
| 13:33.31 | AJLove | Just for a basic sip connection (i want to access my asterisk from outside my local network what ports do i need to NAT/Firewall) 5060,4569,5036 ? anymore any less? |
| 13:33.57 | [TK]D-Fender | AJLove: 10000-20000 UDP typically as well. |
| 13:34.09 | [TK]D-Fender | AJLove: Depending on what protocols you're looking to run. |
| 13:34.41 | *** join/#asterisk soman (~somnath@stargate.starnet.fi) |
| 13:35.03 | AJLove | Do i need to portmap that large range? |
| 13:35.59 | [TK]D-Fender | AJLove: probably not, that is just what the sample RTP config defaults to. |
| 13:36.05 | cfh | <PROTECTED> |
| 13:36.26 | [TK]D-Fender | cfh: doesn't matter. this is a dialplan task. Add it yourself |
| 13:38.39 | cfh | [TK]D-Fender: ok , thanks. |
| 13:40.00 | *** part/#asterisk bzing2|gonehome (~dr105@dhcp-194-66-208-235.canterbury.ac.uk) |
| 13:42.31 | *** join/#asterisk soman (~somnath@stargate.starnet.fi) |
| 13:42.54 | *** join/#asterisk hyphenex (~Adium@60-241-253-37.static.tpgi.com.au) |
| 13:43.05 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 13:43.52 | hyphenex | G'Day gang, I've got a A400P with one FX0 port on it, but it complains about not having echo cancelation (of which I receive a lot). Was this card supposed to come with echo cancelation support? |
| 13:46.19 | *** join/#asterisk rossand (~aross@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 13:48.47 | ManxPower-work | hyphenex: contact the company that makes the card and ask |
| 13:49.08 | [TK]D-Fender | hyphenex: And where do we see this compaining you're talking about? |
| 13:49.43 | hyphenex | It was an e-bay store |
| 13:49.56 | [TK]D-Fender | hyphenex: So the store is the manufacturer? |
| 13:50.06 | ManxPower-work | That does seem odd. |
| 13:50.34 | hyphenex | Hmm, It was some random from e-bay |
| 13:50.41 | hyphenex | maybe I could try and contact OpenVox |
| 13:50.58 | [TK]D-Fender | hyphenex: Go show us something useful or contact the manufacturer |
| 13:52.17 | adnc | hello, i can see a lot of "Really destroying SIP dialog 'somevalues' Method: OPTIONS |
| 13:52.25 | adnc | what do those messages mean? |
| 13:52.30 | hyphenex | okies, I'll try and find the site I bought it from.. Does it help if I get this message? Unable to enable echo cancellation on channel 1 (No such device) |
| 13:52.59 | [TK]D-Fender | adnc: QuALIFY packets |
| 13:53.04 | *** part/#asterisk cfh (~luca@host196-137-static.90-82-b.business.telecomitalia.it) |
| 13:53.27 | adnc | [TK]D-Fender, qualify packets? so i don't need to worry about it? |
| 13:53.32 | [TK]D-Fender | hyphenex: maybe you should show us CONFIGs, CARD StAtuS dumps, etc... |
| 13:53.35 | ManxPower-work | adnc: That message means you are running with debug enabled. |
| 13:53.41 | hyphenex | http://www.voiplink.com/OpenVox_A400P01_1_FXO_p/openvox-a400p01.htm Yep, something like this |
| 13:53.57 | [TK]D-Fender | adnc: that's you setting "qualify=yes". |
| 13:54.00 | hyphenex | [TK]D-Fender: I don't know how to send the card status dumps, this is the first one of these I've tried to set up |
| 13:54.02 | hyphenex | sorry |
| 13:54.05 | *** join/#asterisk Xetrov` (~xetrov@unaffiliated/xetrov/x-827361) |
| 13:54.25 | [TK]D-Fender | hyphenex: Where are your configs? How did yuo even test this without getting some output you could have shared? |
| 13:54.46 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
| 13:55.01 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 13:55.09 | ManxPower-work | ~pb |
| 13:55.11 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 13:55.25 | adnc | i can't get kabeldeutschland running with asterisk. someone here who had success with this? |
| 13:55.51 | [TK]D-Fender | adnc: Never heard of them in here before. |
| 13:55.59 | ManxPower-work | adnc: I doubt the problems you have are specific to your ISP |
| 13:56.17 | hyphenex | [TK]D-Fender: I'll paste the config files, the tests were done by calling my mobile with asterisk -r -vvvv |
| 13:56.19 | adnc | ManxPower-work, i doubt too, but i can not figur out what to do |
| 13:56.44 | ManxPower-work | adnc: put copies of failed call CLI output on pastebin.ca |
| 13:56.51 | adnc | ok |
| 13:56.53 | [TK]D-Fender | adnc: Well you haven't shown us any broken calls with SIP debug or your configs.... are we supposed to guess? |
| 13:57.04 | adnc | [TK]D-Fender, sure you are right |
| 13:57.12 | adnc | ManxPower-work, what verbosity level should i use? |
| 13:57.25 | ManxPower-work | adnc: I always use 3 |
| 13:57.54 | *** join/#asterisk jaytee (~jforde@unaffiliated/jaytee) |
| 13:58.46 | hyphenex | http://paste2.org/p/675638 http://paste2.org/p/675640 and http://paste2.org/p/675643 :) [TK]D-Fender |
| 13:59.45 | adnc | first of all i do get ast_get_srv: SRV lookup for '_sip._udp.proxy.kabelphone.de' mapped to host prox01.kabelphone.de, port 5060 |
| 13:59.48 | ManxPower-work | hyphenex: well, you got at least some of the info |
| 13:59.51 | adnc | my client says 404 |
| 13:59.56 | [TK]D-Fender | hyphenex: those gains are psycho |
| 14:00.05 | ManxPower-work | adnc: srvlookup=no in sip.conf [general] AND TURN OFF THE |
| 14:00.08 | ManxPower-work | DAMN DEBUG\ |
| 14:00.11 | [TK]D-Fender | adnc: Client/ What client? |
| 14:00.18 | *** join/#asterisk af_ (~getsmart@88-149-230-64.dynamic.ngi.it) |
| 14:00.23 | adnc | twinkle |
| 14:00.29 | *** join/#asterisk mrbnet_ (~mrbnet@74-95-100-233-Minnesota.hfc.comcastbusiness.net) |
| 14:00.43 | [TK]D-Fender | adWhat does twinkle have to do with ASTERISK talking to your ITSP? |
| 14:01.03 | *** join/#asterisk cusco (~trilili@2001:0:53aa:64c:1445:2559:a077:acde) |
| 14:01.12 | adnc | [TK]D-Fender, as you know i need a client to make a call |
| 14:01.17 | cusco | hi |
| 14:01.19 | adnc | therefore i use twinkle |
| 14:01.26 | cusco | twinkle softphone users: |
| 14:01.29 | [TK]D-Fender | adnc: Show us the CALL with SIP DEBUG |
| 14:01.51 | cusco | there used to be this window jumping in the bottom of the screen alerting me of a incomming call |
| 14:02.02 | cusco | new version of twinkle doens't seem to have that |
| 14:02.05 | adnc | [TK]D-Fender, how can i get that debuged call? |
| 14:02.11 | cusco | adnc: would you know anything about that? |
| 14:02.20 | [TK]D-Fender | adnc: Good grief..... |
| 14:02.27 | ManxPower-work | adnc: "asterisk -rvvv" and copy the CLI output. |
| 14:02.38 | [TK]D-Fender | adnc: Go to ASTERisk cli, do "sip set debug on" and pastebint he ENTire call attempt |
| 14:02.51 | ManxPower-work | adnc: If you don't understand networking and linux you will fail at building a PBX |
| 14:03.48 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 14:05.06 | hyphenex | [TK]D-Fender: but that still wouldn't explain: Unable to enable echo cancellation on channel 1 (No such device) |
| 14:06.47 | *** join/#asterisk FloZix (~florian@vie38-3-88-170-254-128.fbx.proxad.net) |
| 14:07.34 | *** join/#asterisk coppice (~chatzilla@94.201.17.210.dyn.pacific.net.hk) |
| 14:07.35 | adnc | sip set debug on didn't work, i did sip set debug ip 192.168.193.21 |
| 14:07.42 | adnc | but this doesnt give me anything at all |
| 14:07.47 | adnc | on the CLI |
| 14:07.54 | adnc | when i try to make a call |
| 14:08.00 | Katty | good morning my asterisk does not work at all how to fix pls |
| 14:08.02 | *** join/#asterisk SuPrSluG (~SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 14:08.02 | adnc | anyway thank you all |
| 14:08.11 | adnc | i think i would need to ask to much |
| 14:08.23 | AJLove | What reason could cause the following: SIP to SIp woring fine, but when i call my VirtualNumber call gets connected but cant hear anything |
| 14:09.13 | *** join/#asterisk andres833 (~andres833@190.144.75.22) |
| 14:09.29 | ManxPower-work | AJLove: Your "virtual number" is not SIP? |
| 14:09.56 | [TK]D-Fender | hyphenex: What versions are you running? |
| 14:10.30 | [TK]D-Fender | adnc: "help sip" <- go see which syntax is right for your version |
| 14:11.06 | [TK]D-Fender | adnc: Do not target a specific IP |
| 14:11.08 | AJLove | Calling from Cell to the gateway to Zoiper works fine.... But calling from Cellphone to My Softphone on my HTC is problem... sorry for the wrong info at first... |
| 14:11.13 | adnc | [TK]D-Fender, thank you |
| 14:11.22 | adnc | i now have it, i will post it to pastebin |
| 14:11.44 | [TK]D-Fender | AJLove: Where the hell is 8 in that mess? |
| 14:11.46 | [TK]D-Fender | * |
| 14:12.28 | mrbnet_ | How can I confirm libpri is being loaded by asterisk? I have compiled on the system but the commands are not in the CLI |
| 14:12.47 | ManxPower-work | mbranca: then it's not loaded. |
| 14:12.54 | ManxPower-work | sorry, mrbnet_ |
| 14:13.28 | mrbnet_ | ManxPower-work: Where would I load it? |
| 14:13.58 | ManxPower-work | mrbnet_: it would load automatically when chan_dahdi.so is loaded |
| 14:14.08 | AJLove | Fender: I have a Gateway provider for the virtualnumber that my asterisk is connected to via SIP. The clients to my asterisk are SIP clients. |
| 14:14.36 | AJLove | so an incoming call on my virtual number comes from my provider gateway |
| 14:15.54 | adnc | http://pastebin.com/d51003a70 |
| 14:15.54 | ManxPower-work | did you forget to mention NAT? |
| 14:16.01 | adnc | thank you for help in advance |
| 14:16.14 | *** join/#asterisk minotaur01 (~minotaur0@24.215.3.50) |
| 14:17.27 | [TK]D-Fender | AJLove: I fail to see where ASTERISK is involved there. |
| 14:17.31 | mrbnet_ | ManxPower-work: And it looks like that isn't even being loaded. It is in the modules dir though. Shouldn't module load chan_dahdi.so load it? |
| 14:17.59 | ManxPower-work | mrbnet_: try "module load chan_dahdi.so", you should then see the error that tells you why it's not loading |
| 14:18.13 | [TK]D-Fender | adnc: Looking for 0160xxxxxxx in default (domain 192.168.193.22) SIP/2.0 404 Not Found |
| 14:18.23 | AJLove | Fender: All my SIP phones are connected to MY Asterisk... For outgoing calls my asterisk is connected to a SIP gateway with a virtual number. |
| 14:18.27 | [TK]D-Fender | adnc: You have a DIALPLAN error. this has NOTHING to do with your ITSP |
| 14:18.41 | adnc | ohh |
| 14:18.58 | [TK]D-Fender | adnc: Asterisk cannot find a match for "0160xxxxxxx" in [default]. go fix your extensions.conf |
| 14:19.06 | adnc | ok |
| 14:19.36 | ManxPower-work | Amazing how fast the problem is found when we get some CLI output. |
| 14:20.14 | mrbnet_ | ManxPower-work: just says unable to load |
| 14:20.18 | SuPrSluG | hi all. havng an issue with voicemail.conf not recognizing changes to fromstring=xxxxx . No matter what I change it to it send Asterisk PBX. serveremail works. Asterisk version 1.6.1.8. Any ideas? |
| 14:20.21 | adnc | ManxPower-work, i didn't know how to handle it. sure i' would have done this before if i would know |
| 14:20.41 | [TK]D-Fender | AJLove: I'm not seeing any configs or debug. Go get these if you want to get any help here.... |
| 14:21.33 | ManxPower-work | mrbnet_: pastebin the result of tying to load it. |
| 14:22.14 | adnc | http://pastebin.com/d641fa896 this is my extension.conf i'm not seeing mistakes |
| 14:22.42 | [TK]D-Fender | adnc: exten => _0.,3,Dial(SIP/${EXTEN}@3575560) <-- you have no STEP 1 |
| 14:22.51 | [TK]D-Fender | andYou can't just start at any # you want |
| 14:22.57 | [TK]D-Fender | adnc: You can't just start at any # you want |
| 14:22.59 | AJLove | Seems like its my Softphone on the HTC that is weird.... :S Anyone know of a good free SIP client for WM6 |
| 14:23.04 | adnc | thank you |
| 14:23.06 | mrbnet_ | ManxPower-work: Let me see if I can find some info worth pasting |
| 14:23.32 | ManxPower-work | mrbnet_: Is "unable to load" the EXACT AND FULL MESAGE? |
| 14:23.58 | adnc | thank you so much |
| 14:24.12 | [TK]D-Fender | adnc: You're welcome |
| 14:24.15 | adnc | [TK]D-Fender, that did solve the problem |
| 14:24.23 | adnc | ManxPower-work, thank you |
| 14:25.32 | [TK]D-Fender | adnc: Good, now you should have a better understanding that there were 2 calls going on there. Twinkle to *. Then if * accepts the call and matches what you dial if should hopefully hit that exten that tries to call out to your provider. that is a second call. When things go wrong, it can be any of these steps and you need to look at the complete attempt to see what point fails. |
| 14:25.57 | mrbnet_ | ManxPower-work: http://pastebin.com/m5fb3d25 |
| 14:26.00 | [TK]D-Fender | adnc: exten => 3575560,3,Dial(SIP/31) <-- I'd fix this too while you're at it |
| 14:26.05 | adnc | [TK]D-Fender, thank you |
| 14:26.20 | adnc | [TK]D-Fender, how you mean |
| 14:26.35 | [TK]D-Fender | adnc: this line ALSO only has a priority "3", no "1" |
| 14:26.38 | ManxPower-work | mrbnet_: I've never seen such a short error message in module load |
| 14:26.57 | [TK]D-Fender | adnc: these are step numbers. You do step 1, then 2, then 3, tec... you don't jsut start at 3 |
| 14:27.02 | ManxPower-work | mrbnet_: Did you install DAHDI/libPRI/Asterisk from source? |
| 14:27.03 | adnc | [TK]D-Fender, i changed it to 1 |
| 14:27.08 | [TK]D-Fender | adnc: good.. |
| 14:27.29 | adnc | [TK]D-Fender, i did just that i can use 1 and than n for the followings |
| 14:27.36 | mrbnet_ | ManxPower-work: Me either, yes I installed from source. chan_dahdi.so is in the modules directory as well. |
| 14:27.50 | ManxPower-work | mrbnet_: is this a production box? |
| 14:27.55 | mrbnet_ | No |
| 14:28.07 | [TK]D-Fender | adnc: But you didn't have a "" for this |
| 14:28.42 | Katty | curls up for a nap |
| 14:28.47 | ManxPower-work | mrbnet_: what does "module unload chan_dahdi.so" do? |
| 14:28.58 | adnc | [TK]D-Fender, a ""? |
| 14:29.10 | [TK]D-Fender | adnc: "1" |
| 14:29.14 | mrbnet_ | ManxPower-work: setting up another box to replace the production box. Is it possible it would not load because the T1 line is down? It load on the other box regardless. |
| 14:29.37 | ManxPower-work | mrbnet_: it would load even without anythingplugged into the port |
| 14:29.46 | [TK]D-Fender | mrbnet_: What card? Where do we see your configs? What software versions? |
| 14:30.08 | adnc | [TK]D-Fender, yes, i did place a 1 instead of 3, thats what you corrected, right? |
| 14:30.09 | mrbnet_ | ManxPower-work: That is what I thought. No result from "module unload chan_dahdi.so" |
| 14:30.20 | [TK]D-Fender | adnc: That's what i was referring to.... |
| 14:30.22 | ManxPower-work | mrbnet_: good. |
| 14:31.05 | ManxPower-work | mrbnet_: check /etc/asterisk/logger.conf make sure everything is printing for "console" Stop asterisk, start it as "asterisk -cvvv" try loading the module again |
| 14:31.26 | adnc | [TK]D-Fender, i understand, you did refer to the point that commands needs to be in sequential order |
| 14:31.29 | mrbnet_ | ManxPower-work: Sangoma A101, wanpipe driver 3.5.10, dahdi 2.2.1, asterisk 1.6.1.12 |
| 14:31.56 | adnc | numbered |
| 14:32.08 | [TK]D-Fender | adnc: Or at least generally start with 1 and jump if needed (Goto, etc) |
| 14:32.22 | adnc | cool, thank you |
| 14:32.24 | [TK]D-Fender | mrbnet_: Where is libpri in that? |
| 14:32.34 | [TK]D-Fender | mbranca: Where do we see your configs? |
| 14:33.03 | mrbnet_ | ManxPower-work: sorry, LibPri 1.4.10.2 |
| 14:36.38 | mrbnet_ | ManxPower-work: I changed the settings in logger for the console. Now I am getting some useful stuff. Looks like there are issues with compilation. "Not compiled with the same compile-time options as Asterisk" |
| 14:37.31 | *** join/#asterisk hluesea (~hulusikah@88.247.127.66) |
| 14:38.17 | adnc | i just do read that a provider i would like to use offers iax and sip (terrasip) would you recommend using iax or sip? |
| 14:38.52 | p3nguin_ | I like IAX2. It does trunking. |
| 14:39.00 | Katty | i like your mom. |
| 14:39.16 | p3nguin_ | You've met my mom? |
| 14:39.27 | Katty | i know everyone. |
| 14:39.29 | Katty | obviously. |
| 14:40.05 | *** join/#asterisk Znuff (~ibm86@2001:0:53aa:64c:281e:14d5:a65a:7c98) |
| 14:40.08 | adnc | Katty, whats the point with his mom? thats not nice |
| 14:40.12 | benngard | UDPTL packages = T.38, correct? |
| 14:40.51 | SeriousMatters | Maybe Katty likes p3nguin's mom's cooking? |
| 14:41.21 | coppice | benngard: yes. UDPTL is only used for T.38 |
| 14:41.25 | adnc | sure, thats possible, untill someone comes and says that he likes kattys ass |
| 14:41.37 | *** join/#asterisk superbeef (~lanej@74.84.194.4) |
| 14:44.03 | Katty | adnc: that comment is very inappropriate. |
| 14:45.02 | adnc | not more than involving other peoples family |
| 14:45.02 | SeriousMatters | there is an old saying that women with big ass are better with babies. |
| 14:45.02 | Katty | adnc: did it ever occur to you that maybe i know p3nguin_? |
| 14:45.20 | *** join/#asterisk Zagrophyte (~trosenbau@fw.swerdlin.net) |
| 14:45.22 | adnc | obviously not, since p3nguin_ asks you explicitly if you know her |
| 14:45.36 | Katty | adnc: are you sure of that? |
| 14:45.43 | Katty | adnc: think real hard about it. |
| 14:45.54 | *** join/#asterisk genuix (~genuix_@adslgva0816.worldcom.ch) |
| 14:46.09 | Xetrov` | i see blood! |
| 14:46.19 | adnc | no, i'm not, i'm only sure of what <p3nguin_> You've met my mom? said |
| 14:46.31 | p3nguin_ | adnc: Well, katty and I both live in the midwest, but my mother lives in Florida... so I wasn't aware that katty knew my mother. |
| 14:46.37 | benngard | then i dont understand why i dont jump to fax extension, i did enable faxdetect=yes on "my" extension |
| 14:46.42 | Katty | p3nguin_: that's not the point. |
| 14:46.51 | adnc | p3nguin_, well than |
| 14:46.53 | Katty | p3nguin_: the point is that adnc has no idea of the context |
| 14:47.09 | Katty | p3nguin_: and if adnc was aware of the context, he would understand the meaning of the comment. |
| 14:47.14 | Zagrophyte | This is slightly off-topic, but can anyone point me in the direction of tools to decode fax recordings from audio? One of our users has deleted a fax message, but we do have audio recordings. |
| 14:47.39 | adnc | the context doesn't matter, it is not fine bringing other peoples family into public like here. but if p3nguin_ is happy with this, than i wish you good luck with that case |
| 14:47.56 | p3nguin_ | Either way, you're both spent way too much time on my mom. |
| 14:47.58 | adnc | and ask for apollogize |
| 14:47.59 | p3nguin_ | ;) |
| 14:47.59 | Katty | adnc: context is everything. |
| 14:48.12 | Katty | adnc: you don't appologize to friends when you tease them. |
| 14:48.21 | Katty | adnc: you tease them more. |
| 14:48.47 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 14:49.19 | Katty | adnc: that's the whole point of the comment |
| 14:49.26 | Katty | adnc: p3nguin_ wouldn't feel loved if i didn't torment him every day |
| 14:49.41 | ariel_ | Morning folks |
| 14:49.45 | Katty | hugs ariel_ |
| 14:49.54 | ariel_ | hugs Katty |
| 14:50.12 | coppice | Zagrophyte: there's a sort of fax decode utility in the tests directory of spandsp, but its really intented for debug purposes rather than a user friendly decode |
| 14:50.15 | adnc | Katty, thats fine, if that is the way you you behave each other and everyone is happy, than there is no reason for me to spend more words o it |
| 14:50.19 | p3nguin_ | Hmm... I can't pass this up... |
| 14:50.21 | p3nguin_ | (0840.52) <SeriousMatters> Maybe Katty likes p3nguin's mom's cooking? |
| 14:50.25 | p3nguin_ | (0841.26) <adnc> sure, thats possible, untill someone comes and says that he likes kattys ass |
| 14:50.40 | p3nguin_ | Maybe I like katty's ass's cooking! |
| 14:50.41 | Katty | LOL |
| 14:50.45 | Zagrophyte | coppice: Yeah I found that on google, there's also some program called Fox III, but it is not available |
| 14:50.52 | Katty | bashhhhh |
| 14:51.22 | Katty | adnc: you just need to simmer. |
| 14:51.23 | Zagrophyte | coppice: got the source for spandsp, hopefully that works, wondering if there are any other tools |
| 14:51.28 | Katty | adnc: and have a redbull and vodka. |
| 14:51.45 | Katty | adnc: the world will not come to an end, i assure you |
| 14:51.55 | coppice | Zagrophyte: there are several decode packages if you pay, but some will only be sold to law enforcement agencies |
| 14:53.41 | Zagrophyte | coppice: yeah, weird though as the protocol's been around forever |
| 14:53.44 | ariel_ | I have an error in trying to load the dahdi drivers for the astribank. it's not loading the drivers: here is the pastebin: http://pastebin.ca/1801598 |
| 14:53.50 | Zagrophyte | gonna try fax_decode now |
| 14:53.51 | adnc | Katty, hold on, i need to find out what simmer is |
| 14:54.13 | p3nguin_ | it's slightly less active than a slow boil. |
| 14:55.18 | adnc | Katty, no, i just wanted to point that family, mom, vather are in a different position and there we all need to show more respekt, but if you know each othere than thats fine |
| 14:55.45 | Zagrophyte | did someone bust out a "yo momma" joke? :D |
| 14:56.00 | Katty | Zagrophyte: no |
| 14:56.05 | Katty | Zagrophyte: it was a Your Mom joke |
| 14:56.24 | tzafrir_laptop | ariel_, "/usr/share/dahdi/xpp_fxloader: line 237: astribank_is_starting" - harmless on its own. But makes me think you built this on a system with no libusb-dev(el) |
| 14:56.46 | *** part/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 14:56.48 | SeriousMatters | Katty, I like you mom too. |
| 14:56.54 | SeriousMatters | and adnc's |
| 14:57.04 | SeriousMatters | also Zagrophyte's |
| 14:57.14 | adnc | ;) |
| 14:57.16 | ariel_ | tzafrir_laptop: ok let me see if that is the issue t/y |
| 14:57.25 | SeriousMatters | because they gave birth to you guys, and i like you guys, so.... |
| 14:59.20 | ariel_ | tzafrir_laptop: apt-get install libusb-dev says it's is already the newest verison |
| 14:59.23 | ariel_ | version |
| 14:59.33 | Katty | SeriousMatters: my mom is awesome |
| 14:59.40 | p3nguin_ | seriousmatters: Hmm. Here they all are being smart asses, and you go and get all sentimental on us. |
| 15:00.16 | SeriousMatters | lol, i just found this conversation silly especially on #asterisk |
| 15:01.00 | p3nguin_ | Wait, what? This is #asterisk? |
| 15:01.14 | Zagrophyte | your face likes my mom |
| 15:01.16 | tzafrir_laptop | ariel_, what version is it? |
| 15:01.22 | Zagrophyte | :P |
| 15:01.31 | ariel_ | dahdi 2.2.1 combined |
| 15:01.33 | *** join/#asterisk cesar_CR (~cesar@201.192.86.30) |
| 15:01.48 | tzafrir_laptop | next thing on my mind: is /usr/sbin in PATH ? |
| 15:02.23 | *** join/#asterisk Akiraa (~Akiraaaa@79.112.11.73) |
| 15:03.12 | SuPrSluG | where does the fromstring in voicemail.conf get information? I hard coded a different From: Asterisk PBX in app_voicemail.c and it still sends the fromstring as Asterisk PBX. wtf? |
| 15:03.54 | ariel_ | tzafrir_laptop: ok got it working. I just used the two dahdi seperate the dahdi-linux and dahdi-tools.... I am going to setup a new bare test system I think there is something wrong with the combined |
| 15:04.53 | SuPrSluG | oops. missed one. |
| 15:05.05 | tzafrir_laptop | ariel_, btw: what distro is it? |
| 15:05.13 | ariel_ | debian 5 |
| 15:05.35 | tzafrir_laptop | my test packages are, as usual, under http://updates.xorcom.com/pkg-voip |
| 15:05.52 | p3nguin_ | suprslug: So you found the fromstring in voicemail.conf? |
| 15:06.21 | *** part/#asterisk hyphenex (~Adium@60-241-253-37.static.tpgi.com.au) |
| 15:06.31 | *** part/#asterisk Zagrophyte (~trosenbau@fw.swerdlin.net) |
| 15:06.32 | p3nguin_ | suprslug: 'Cause that's where they intended for the user to set that data. |
| 15:06.33 | SuPrSluG | no i'm hardcoding in app_voicemail.c |
| 15:06.41 | p3nguin_ | suprslug: Why? |
| 15:06.48 | ariel_ | tzafrir_laptop: ok t/y just link that site to my system. |
| 15:06.59 | p3nguin_ | suprslug: The fromstring setting in voicemail.conf does exactly that. |
| 15:07.00 | SuPrSluG | because when I change it it does nothing. |
| 15:07.15 | SuPrSluG | not working |
| 15:07.56 | ManxPower-work | SuPrSluG: do you understand the difference between the message and the SMTP envelope? |
| 15:08.05 | SuPrSluG | now it does :-D |
| 15:09.57 | SuPrSluG | the fromstring should set the from address in the email, right. serveremail works fine. when i change that i get the desired results. when changing fromstring= it always gives Asterisk PBX. |
| 15:10.39 | *** join/#asterisk Zagrophyte (~trosenbau@fw.swerdlin.net) |
| 15:11.49 | p3nguin_ | From: "The Asterisk PBX" <asterisk@asterisk.local.domain> |
| 15:12.02 | p3nguin_ | That's what the fromstring is intended to change. |
| 15:12.19 | SuPrSluG | Yes. But, it didn't. |
| 15:12.32 | *** join/#asterisk e4 (~e4@rrcs-76-79-59-194.west.biz.rr.com) |
| 15:13.14 | SuPrSluG | That's the nice thing about having the source. |
| 15:13.45 | p3nguin_ | If the source is capable of changing it, how could the conf not be able to? |
| 15:14.08 | [TK]D-Fender | SuPrSluG: OR.... you could simply have make a tiny shell-script to mod the address <- |
| 15:14.09 | p3nguin_ | suspects failure to reload app_voicemail |
| 15:14.28 | [TK]D-Fender | SuPrSluG: But enjoy your constant remerging of code to keep up with * releases :p |
| 15:14.48 | Zagrophyte | coppice: hah, I didn't even realize that you were the one that posted what I had found on google earlier, so I had come to the right place ;) |
| 15:14.55 | SuPrSluG | no, total restarts. not the issue. |
| 15:15.01 | p3nguin_ | And don't forget to submit your patches. |
| 15:15.25 | SuPrSluG | [TK]D-Fender: that certainly is a good reason not for the source edit |
| 15:15.58 | SuPrSluG | not a patch. remedial hack |
| 15:16.06 | p3nguin_ | aka patch |
| 15:16.58 | p3nguin_ | My concern is this, though: If the setting in the conf is not capable of changing the value while in operation, this is a bug and needs to be filed as such. |
| 15:17.09 | Katty | kills symantec |
| 15:17.14 | SuPrSluG | wouldn't be useful to anyone but me. Just changes the From to what i want in my case |
| 15:17.49 | p3nguin_ | A lot of people change that value in the conf. |
| 15:17.58 | p3nguin_ | A lot of people expect it to work. |
| 15:18.00 | SuPrSluG | p3nguin_, maybe I'll do that. |
| 15:18.03 | p3nguin_ | If it doesn't work, it's a bug. |
| 15:18.42 | SuPrSluG | could also be a mail program bug. |
| 15:19.09 | p3nguin_ | sendmail, postfix, exim, qmail? |
| 15:19.45 | Katty | is today a holiday or something? |
| 15:19.57 | SuPrSluG | sendmail and we all what a friendly config file that is :-) |
| 15:20.46 | p3nguin_ | I run sendmail on at least one server and I don't recall the config being much of a problem. |
| 15:21.14 | p3nguin_ | Change a few lines to make it relay mails from other hosts, and that's about it. |
| 15:22.14 | p3nguin_ | And it certainly didn't have any setting to lock in the from string of Asterisk to say "Asterisk PBX." |
| 15:22.51 | *** join/#asterisk thehar (thehar@thehar.xmission.com) |
| 15:25.57 | *** join/#asterisk andres833 (~andres833@190.144.75.22) |
| 15:26.59 | *** part/#asterisk benngard (~benngard@213.88.138.230) |
| 15:27.20 | *** join/#asterisk ReDNeQ (~ReDNeQ@70.114.229.58) |
| 15:29.16 | *** join/#asterisk Torrieri (~Torrieri@nelug/crew/torrieri) |
| 15:30.04 | *** join/#asterisk bittin (~quassel@c213-100-108-185.swipnet.se) |
| 15:30.25 | bittin | Somone know the standard port for asterisk? |
| 15:30.57 | p3nguin_ | Does not compute. |
| 15:30.57 | Chainsaw | There is no standard port for "Asterisk". |
| 15:31.03 | Chainsaw | What protocol are you using, bittin? |
| 15:31.09 | p3nguin_ | Protocols have ports. |
| 15:31.46 | bittin | Chainsaw: don't know iam just thinking off installing asterisk on my local client to play around with it |
| 15:31.56 | bittin | thinking off using it in Ekiga |
| 15:32.05 | p3nguin_ | Unless you decide on a protocol, there are no ports. |
| 15:32.12 | bittin | and then i wonder if theres some regular port the web interface is on |
| 15:32.20 | p3nguin_ | There is no web interface. |
| 15:32.25 | bittin | :o ok |
| 15:32.34 | bittin | i heard there was but i can be wrong |
| 15:32.40 | Katty | screams |
| 15:32.42 | Katty | rips hair out |
| 15:32.42 | p3nguin_ | But if you install one, you can run it on whatever port you want. |
| 15:32.46 | Katty | shreds curtains |
| 15:32.50 | bittin | ah ok |
| 15:33.02 | ManxPower-work | bittin: there are many GUIs for Asterisk. Asterisk itself does not have a GUI |
| 15:33.03 | *** join/#asterisk albertoandrade (~albertoan@189.58.9.244.dynamic.adsl.gvt.net.br) |
| 15:33.14 | bittin | what gui do you reccomend learning? |
| 15:33.20 | ManxPower-work | They all suck |
| 15:33.20 | p3nguin_ | vim |
| 15:33.21 | Katty | we don't |
| 15:33.22 | bittin | that are in apt-get |
| 15:33.31 | bittin | and that i later can compile for Gentoo |
| 15:33.32 | Katty | gui = problems |
| 15:33.34 | ManxPower-work | ~freepbx |
| 15:33.35 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 15:33.48 | ManxPower-work | FreePBX is one of the GUIs |
| 15:34.08 | p3nguin_ | If you don't already have an operating system on the computer and can afford using it as a dedicated asterisk machine, try AsteriskNOW if you need a GUI. |
| 15:34.49 | bittin | i don't have any dedicated Asterisk machine at the moment but thinking off getting one later |
| 15:35.13 | p3nguin_ | If you don't NEED a GUI, just install Asterisk and admin it like we all do. |
| 15:35.33 | Nugget | The best asterisk gui is gvim |
| 15:35.57 | Zagrophyte | Another general question: We have a ton of Polycom Soundpoint IP 550s, in what ways can they be repurposed? Originally they were in-house desktop phones. |
| 15:36.01 | romain | bittin: use xivo |
| 15:36.25 | p3nguin_ | zagrophyte: They can be repurposed as phones on other desktops. |
| 15:36.39 | Naikrovek | Zagrophyte: reuse them as phones or as some sort of PA system maybe |
| 15:36.39 | bittin | are the files in /etc/asterisk? |
| 15:36.40 | p3nguin_ | zagrophyte: or mounted on a wall. |
| 15:36.51 | p3nguin_ | bittin: usually |
| 15:36.58 | bittin | nice |
| 15:37.26 | *** join/#asterisk UQlev (~yuriy@nb11-125.static.cytanet.com.cy) |
| 15:38.33 | *** join/#asterisk d-k-t (~D@112.202.232.46) |
| 15:38.55 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 15:39.10 | *** join/#asterisk mort_gib (~mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 15:39.42 | *** join/#asterisk rubberneck (~chatzilla@ext-52.sagetelecom.net) |
| 15:39.43 | *** join/#asterisk nicknick_ (~administr@host213-123-201-13.in-addr.btopenworld.com) |
| 15:39.46 | Zagrophyte | p3nguin_: I've messed with smaller SIP PBX (3CX) on my home PC to try and set up a home SIP phone system, I spent so many hours on that, but I did get it "working" (I could call some test numbers of the server I was connecting to, but nothing out of that). |
| 15:40.19 | p3nguin_ | zagrophyte: Did you have any outside services? |
| 15:40.22 | Zagrophyte | But that's about the only other application I could find for it |
| 15:40.37 | [TK]D-Fender | Zagrophyte: "find for it"? find for what? |
| 15:41.12 | p3nguin_ | zagrophyte: You'll need to have either an ITSP to give you VoIP services or some type of PSTN connection so you can make calls on POTS. |
| 15:41.37 | ManxPower-work | Zagrophyte: There are millions of Pollycoms connected to many, many different SIP based pbxs at thousands and thousands and thousands of installations. |
| 15:41.41 | Zagrophyte | [TK]D-Fender: Well it's a mini linux PC, in a phone enclosure, would be interesting if it could do more than just be a SIP dumbphone |
| 15:41.56 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:41.56 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 15:42.05 | Zagrophyte | p3nguin_: Yeah, getting a way back to PSTN was where I hit a wall, I found a free provider but I couldn't get the configs right |
| 15:42.37 | [TK]D-Fender | Zagrophyte: .... Its a phone... but hey if you feel like going and wrting your own firmware pretty much froms cratch, then go for it. |
| 15:42.56 | Zagrophyte | [TK]D-Fender: It also supports custom applications |
| 15:44.12 | *** join/#asterisk lftsy (~lftsy@leonhart.leurent.eu) |
| 15:44.45 | Zagrophyte | One other application we've thought of is a VPN-backed "batphone" style system where we run a separate PBX for IT ;) |
| 15:45.15 | Katty | listening to non-technical people have a conversation about technology (in this case backups) is just hilarious |
| 15:45.30 | lftsy | Hello everybody, on Asterisk 1.2, I was able to use gosub and exten => i,1,Return . Now, on Asterisk 1.6, the "i" extention is not triggered anymore... Any idea why? |
| 15:45.36 | mort_gib | Zagrophyte: Asterisk-openVPN-Snom370 |
| 15:45.41 | ManxPower-work | Zagrophyte: "support" is nor really accurate, It has a dumb as dirt XML microbrowser and all "applications" have to run on a web server |
| 15:46.25 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 15:46.34 | Zagrophyte | The reason I'm looking for other uses is the management before me got suckered into spending lots of money on these things, (about 300% market), with a crap PBX, so we ended up falling back to Nortel BCM, and I'm trying to at least find a use for the 60 or so of these we have in boxes. |
| 15:46.40 | p3nguin_ | lftsy: 'i' is the "invalid" extension. I doubt they took that out in 1.6.x. |
| 15:47.00 | Zagrophyte | ManxPower-work: Yeah, it didn't look too promising |
| 15:47.07 | ManxPower-work | p3nguin_: more correctly it is the invalid extension for IVRs |
| 15:47.29 | Zagrophyte | mort_gib: thanks I'll look into that |
| 15:48.01 | visz | Zagrophyte, i have a bunch of used snom m3 cordless phones. i was thinking about using one as a media player remote, but that would be quite tricky to accomplish |
| 15:48.12 | lftsy | p3nguin_: I'm using Realtime dialplan, and now, the "i" extension Return is not triggered anymore |
| 15:48.35 | [TK]D-Fender | ifGosub didn't exist in 1.2 |
| 15:48.46 | [TK]D-Fender | lftsy: Gosub didn't exist in 1.2 |
| 15:49.03 | [TK]D-Fender | lftsy: Would be nice if you showed us your code & failed call... |
| 15:49.04 | [TK]D-Fender | ~pb |
| 15:49.05 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 15:49.07 | [TK]D-Fender | ^^^^^^^^^ |
| 15:50.09 | lftsy | [TK]D-Fender: OK, thanks I'm preparing it |
| 15:54.27 | *** join/#asterisk jpeeler (~jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:56.40 | lftsy | [TK]D-Fender: Here is the post http://pastebin.com/m5d8a9fb4 |
| 15:56.52 | lftsy | you will see the Asterisk 1.2 and 1.6 dialplan |
| 15:57.12 | lftsy | and the adapted dialplan in the MySQL database, with MySQL format |
| 15:58.50 | *** part/#asterisk Zagrophyte (~trosenbau@fw.swerdlin.net) |
| 15:59.50 | *** join/#asterisk Geminizer (~me@cpe-76-180-27-4.buffalo.res.rr.com) |
| 15:59.56 | lftsy | And of course, I have alsa in the mysql dialplan default,i,1,Return |
| 16:00.21 | Geminizer | Hello all. Are there different types of busy signals that a dialplan can create? |
| 16:00.56 | [TK]D-Fender | lftsy: 1.2 & 1.6 both using realtime? |
| 16:01.06 | lftsy | [TK]D-Fender: Yep |
| 16:01.23 | lftsy | I have just adapted the syntax |
| 16:01.29 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:02.41 | [TK]D-Fender | lftsy: Hrm... not sure why the behavior would be different. I supposed you could try a ChanIsAvail against it to see if its valid rather than trapping it after the fact |
| 16:03.24 | *** join/#asterisk elwerene (~lalala@vpn-s-8d3a31ee.campus.uni-stuttgart.de) |
| 16:06.04 | lftsy | [TK]D-Fender: i'm not sending it to a busy channel, it's just that when calling an extension that is not local, it's not in the database, so it comes back from the godub and go to another sip_trunk |
| 16:07.35 | Katty | lol "what would happen to my data if atom bombs went off all over the united states" |
| 16:07.39 | Katty | ^- an actual question |
| 16:08.48 | [TK]D-Fender | lftsy:I'm saying to use it to see if it matches, if so jump there, otherwise do something else... |
| 16:08.51 | lftsy | [TK]D-Fender: on Asterisk 1.2 it said, "Sent into invalid extension '0762211329' in context 'default'" wheread on asterisk 1.6, it says Spawn extension (default, 0860762211329, 6) exited non-zero |
| 16:09.27 | *** join/#asterisk bsilberman (~bsilberma@65.213.221.1) |
| 16:09.36 | bsilberman | hello people |
| 16:09.42 | lftsy | [TK]D-Fender: indeed it should work, but I would like to understand why it is not working anymore |
| 16:10.14 | bsilberman | does anyone here run 1.4.29 and have issues with voicemails being "recorded" but you can never check them? |
| 16:11.41 | Katty | bsilberman: why can't you check them |
| 16:11.44 | p3nguin_ | I run 1.4.29 and I do not have that problem. |
| 16:12.11 | lftsy | bsilberman: I tried to run voicemail 1.4.29 but I had segfault when reading voicemail (on CentOS), that's why I'm upgrading to asterisk 1.6 |
| 16:12.29 | lftsy | p3nguin_: which OS are you using pleade? |
| 16:12.40 | p3nguin_ | ArchLinux |
| 16:12.48 | lftsy | p3nguin_: Thanks |
| 16:12.59 | Katty | bsilberman: please do not send me private messages. |
| 16:13.11 | Katty | bsilberman: please type your reply here so that the entire channel can see it, and assist you. |
| 16:13.40 | bsilberman | btw, this is running on redhat 5.4 ... Red Hat Enterprise Linux Server release 5.4 (Tikanga) |
| 16:13.46 | leifmadsen | o.O |
| 16:13.54 | Katty | orly |
| 16:13.58 | Katty | hi blitz |
| 16:14.02 | Katty | hugs leifmadsen |
| 16:14.21 | leifmadsen | hugs back Katty |
| 16:14.33 | [TK]D-Fender | bsilberman: Show us. |
| 16:15.01 | *** join/#asterisk festr_ (~festr@77.93.216.244) |
| 16:15.29 | festr_ | hi |
| 16:15.47 | festr_ | I'm trying to hangup channel in agi script |
| 16:15.51 | Katty | ohaider |
| 16:15.56 | Katty | how're you today |
| 16:16.03 | festr_ | but I cannot find way how to pass hangupcause |
| 16:16.09 | festr_ | anyone any clue? |
| 16:16.10 | Katty | wow that good eh? |
| 16:16.12 | festr_ | (asterisk 1.4) |
| 16:16.43 | Katty | my pen just died |
| 16:16.44 | Katty | )= |
| 16:16.44 | lftsy | [TK]D-Fender: https://issues.asterisk.org/view.php?id=14411 looks to have my answer, strange, it's only working on asterisk 1.2 |
| 16:16.53 | adnc | exten => _0050.,1,Dial(SIP/${EXTEN}@sipprov1) and exten => _0.,1,Dial(SIP/${EXTEN}@sipprov2) this doesnt distinguish between national calls and calls to 0050 country. what am i missing? |
| 16:17.07 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 16:17.17 | festr_ | adnc: _0. is too hungry |
| 16:17.20 | *** part/#asterisk FloZix (~florian@vie38-3-88-170-254-128.fbx.proxad.net) |
| 16:17.22 | adnc | ahh |
| 16:17.29 | leifmadsen | 00501234 should match on the first, not the 2nd |
| 16:17.41 | festr_ | unfortunatelly it does not |
| 16:17.42 | leifmadsen | more specific should match, unless you're doing something funny with includes |
| 16:17.45 | leifmadsen | festr_: what version? |
| 16:17.48 | festr_ | 1.4 |
| 16:17.52 | leifmadsen | 1.4.? |
| 16:17.53 | adnc | leifmadsen, but it matches on sipprov2 |
| 16:17.58 | adnc | yes, it is 1.4 |
| 16:18.02 | leifmadsen | no.... |
| 16:18.05 | leifmadsen | 1.4.what |
| 16:18.20 | leifmadsen | 1.4.8 |
| 16:18.22 | leifmadsen | 1.4.28 |
| 16:18.24 | festr_ | leifmadsen: get the point :) i've saw some changes regarding this? |
| 16:18.24 | leifmadsen | ? |
| 16:18.31 | leifmadsen | I don't know... have you? |
| 16:18.33 | adnc | it is 1.4.14 |
| 16:18.39 | festr_ | leifmadsen: yes. |
| 16:18.40 | leifmadsen | yes, update that |
| 16:18.52 | festr_ | i've saw some changes in svn list |
| 16:18.55 | leifmadsen | right |
| 16:19.06 | [TK]D-Fender | bsilberman: PASTEBIN the call that leaves the VM, and the call that goes to CHECK it |
| 16:19.08 | adnc | leifmadsen, sorry, was that for me? |
| 16:19.10 | festr_ | but for a long time, _5. was winner over _55. |
| 16:19.19 | festr_ | adnc: yes |
| 16:19.19 | elwerene | Incoming calls from voip to isdn(hfc-s, zaphfc with dahdi) not working/handset not ringing. http://pastebin.com/md513604 http://pastebin.com/m471be0cd |
| 16:19.27 | leifmadsen | wait... I don't know who I'm talking to now :) |
| 16:19.31 | elwerene | Some new ideas? |
| 16:19.38 | leifmadsen | adnc: I was talking to you :) |
| 16:19.42 | festr_ | adnc: upgrade your version or revrite rules into _5XXXX. |
| 16:19.49 | festr_ | adnc: and it will work |
| 16:19.51 | *** join/#asterisk andres833 (~andres833@190.242.82.166) |
| 16:20.05 | festr_ | sorry . _0050. and _0XXXX. |
| 16:20.07 | adnc | festr_, i'm bound to the debian etch package unfortunately |
| 16:20.19 | leifmadsen | yuck |
| 16:20.20 | festr_ | adnc: so rewrite it to what I suggested |
| 16:20.25 | leifmadsen | heads off to do some work |
| 16:20.40 | adnc | festr_, _0XXXX wil match to four digits after 0, but what if the area code has more digits |
| 16:20.54 | festr_ | adnc: see "." at the end |
| 16:20.59 | adnc | ok |
| 16:21.35 | festr_ | adnc: but you should really upgrade to the latest 1.4 |
| 16:21.49 | festr_ | adnc: compilation is not so hard |
| 16:22.06 | adnc | i'll look for an other package |
| 16:22.16 | adnc | can i than leave the dialplan? |
| 16:22.27 | festr_ | yes there were some fixes regarding this |
| 16:22.43 | adnc | festr_, thank you |
| 16:22.57 | bmoraca_work | 01189998899991197253 <=== best phone number evar |
| 16:23.11 | festr_ | adnc: and i dont know if it was released or if it is in svn repo only |
| 16:23.21 | adnc | ;) |
| 16:23.22 | festr_ | adnc: but you can live with _5XXXX. |
| 16:23.23 | festr_ | :D |
| 16:23.26 | Chainsaw | bmoraca_work: You forgot the pause between the 5 and the 3 :D |
| 16:23.34 | bmoraca_work | lol |
| 16:23.42 | adnc | festr_, ok, i'll try _5XXX. for international call |
| 16:23.52 | festr_ | _0XXXX |
| 16:23.55 | festr_ | _0XXXX. |
| 16:24.04 | adnc | ohh, for that particular country |
| 16:24.09 | festr_ | the point is to have "." not before another "." in _0500. |
| 16:24.37 | festr_ | because _5. is matched asap on third number regardless there is another possibility |
| 16:24.46 | festr_ | sorry on second number |
| 16:25.09 | festr_ | ok and lets back to AGI. |
| 16:25.10 | festr_ | :) |
| 16:25.36 | festr_ | anyone know how to hangup channel in agi like in dialplan hangup(16) for example? |
| 16:26.25 | leifmadsen | isn't Hangup an AGI application? If not, I think there is a link to use any dialplan application you can |
| 16:27.38 | leifmadsen | festr_: agi show commands topic hangup |
| 16:27.54 | leifmadsen | (command available in at least 1.6.2) |
| 16:28.59 | festr_ | leifmadsen: I've no access to 1.6 :( |
| 16:29.05 | festr_ | leifmadsen: there is info about hangupcause? |
| 16:29.22 | leifmadsen | sorry, can't help you then as I haven't used 1.4 in quite a while and don't have a box running it right now to look |
| 16:29.45 | festr_ | no problem thanks |
| 16:30.06 | festr_ | leifmadsen: in asterisk 1.4 hangup command has only one parameter and it is channel. |
| 16:30.14 | leifmadsen | ok |
| 16:30.23 | leifmadsen | sounds like a true statement |
| 16:30.34 | festr_ | leifmadsen: 1.6.2 does have hcause param? |
| 16:30.44 | Kobaz | EXEC Hangup |
| 16:30.47 | Kobaz | if you want to use hangup from agi |
| 16:30.53 | festr_ | Kobaz: this does not work |
| 16:31.03 | Kobaz | what happens? |
| 16:31.32 | festr_ | EXEC Hangup 0 |
| 16:31.42 | festr_ | AGI Tx >> 200 result=-1 |
| 16:31.49 | festr_ | and it will do nothing |
| 16:32.13 | Kobaz | that's odd |
| 16:34.07 | festr_ | but, EXEC Hangup 17 |
| 16:34.12 | festr_ | followed by HANGUP works |
| 16:34.33 | festr_ | hangupcause is transformed to busy |
| 16:34.55 | adnc | is there a way to test dial pattern on the CLI to see which rule matches? |
| 16:35.56 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.157) |
| 16:38.37 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 16:43.24 | p3nguin_ | Darn, I can't decide which CPU cooler to get. http://www.newegg.com/Product/Product.aspx?Item=N82E16835118020 or http://www.newegg.com/Product/Product.aspx?Item=N82E16835118034 |
| 16:45.28 | Kobaz | the first one |
| 16:45.39 | jaytee | if it fits in the chassis |
| 16:45.45 | jaytee | it's a tall boy |
| 16:45.46 | Kobaz | uprights are better... since they will push air out the back of the case... and not down |
| 16:45.59 | rubberneck | adnc: dialplan show 123@context |
| 16:46.03 | Qwell | I love me some newegg. |
| 16:46.10 | jaytee | I prefer all copper heatsinks |
| 16:46.17 | Qwell | ordered something yesterday afternoon, and...it's out for delivery. |
| 16:46.22 | Qwell | looks at Katty |
| 16:46.28 | jaytee | nickel plated aluminum with copper cores aren't as efficient |
| 16:46.40 | p3nguin_ | I wanted to get the CNPS10X Extreme, but that thing is HUGE. |
| 16:47.12 | p3nguin_ | It weighs 920 grams. 2 POUNDS for you Americans. |
| 16:47.48 | p3nguin_ | I don't see how the board could support a 920g cooler hanging off it. |
| 16:47.51 | jaytee | if it was a Dell system I'd just say go with the cheapest one since the motherboard will blow a capacitor before the CPU melts down in my experience |
| 16:48.12 | p3nguin_ | My board has solid caps, so I hope that doesn't happen. |
| 16:48.15 | coppice | p3nguin_: its fine until you transport it :-) |
| 16:48.19 | Kobaz | p3nguin_: the modern cooler mounts are really beefy |
| 16:48.51 | p3nguin_ | I've been running the stock Intel cooler for two years. I've decided I need an upgrade. |
| 16:50.15 | *** join/#asterisk geneticx (~chatzilla@host-208-88-126-198.biznesshosting.net) |
| 16:50.54 | *** join/#asterisk andres833 (~andres833@190.144.75.22) |
| 16:51.20 | geneticx | Greetings. |
| 16:51.30 | geneticx | I come in peace. |
| 16:52.25 | Katty | looks at Qwell |
| 16:52.38 | Qwell | Katty: see above! |
| 16:52.39 | Katty | bmoraca_work: were you the one who gave me the mahi mahi recipe? |
| 16:52.40 | p3nguin_ | jaytee: Are you saying that you would not buy one of the copper and aluminum Zalmans? |
| 16:52.42 | Katty | Qwell: i did |
| 16:52.42 | Qwell | not Tuesday, maybe |
| 16:52.47 | Katty | Qwell: :> |
| 16:52.50 | Qwell | oh, I see what you did there. |
| 16:52.56 | bmoraca_work | Katty: yes |
| 16:53.02 | Katty | bmoraca_work: it's marinating as we speak |
| 16:53.12 | Katty | bmoraca_work: it smells divine too |
| 16:53.33 | bmoraca_work | cool |
| 16:53.46 | jaytee | p3nguin, I said what I said |
| 16:53.54 | *** part/#asterisk lsenft (~levisenft@75-148-103-190-Utah.hfc.comcastbusiness.net) |
| 16:53.57 | p3nguin_ | I DON'T KNOW WHAT THAT MEANS. |
| 16:54.21 | *** join/#asterisk FSB_1 (~LOLOLOL@unaffiliated/fsb1/x-429504) |
| 16:54.42 | jaytee | what part about nickel plated aluminum with a copper core vs a full copper heatsink is confusing you? |
| 16:55.01 | *** join/#asterisk ypahnu (~zveb@bl4-164-147.dsl.telepac.pt) |
| 16:55.15 | ypahnu | hi all, this is PHP software? |
| 16:55.19 | p3nguin_ | What if the aluminum wasn't nickel plated? |
| 16:56.15 | Kobaz | ypahnu: what? |
| 16:56.35 | Kobaz | p3nguin_: then you would just have aluminum>? |
| 16:56.40 | p3nguin_ | The CNPS8700 is listed as "pure copper," so maybe that would be better? |
| 16:57.07 | ypahnu | p3nguin_, is does asterisk use mySQL database? |
| 16:57.16 | Kobaz | ypahnu: it can |
| 16:57.23 | p3nguin_ | ypahnu: It is does not unless you tell it to. |
| 16:57.23 | Kobaz | ypahnu: and why are you asking him? |
| 16:57.40 | Katty | jaytee: Girdle String |
| 16:57.47 | Kobaz | my is does not maybe can will mysql |
| 16:57.55 | jaytee | Katty, huh? |
| 16:58.34 | ypahnu | what languange is asterisk programed ? |
| 16:58.42 | Kobaz | C |
| 16:59.01 | Katty | jaytee: youtube reference |
| 16:59.29 | adnc | unfortunately my dial plan doesnt't work this is the output of dialplan show http://pastebin.com/d9d69e4d |
| 16:59.36 | adnc | where do i do the mistake |
| 17:00.05 | p3nguin_ | I guess I'll get the second one, since it isn't nickel-plated aluminium. Hope it fits. http://www.zalman.co.kr/eng/product/Product_Read.asp?idx=278 |
| 17:00.06 | jaytee | p3nguin_, the nickel plate helps improve heat transfer but even then it's not as efficient as copper at conducting heat. I like the Ultra X-Wind for most 775 chips that aren't overclocked. If I'm gonna build an overclocked system then liquid cooling is the only way to go |
| 17:00.48 | jaytee | Thermaltake makes a good one that's all copper but it's a fat boy and the fan is a tad noiser since it spins at a much higher average RPM. |
| 17:01.01 | *** join/#asterisk Torrieri (~Torrieri@nelug/crew/torrieri) |
| 17:01.21 | jaytee | Katty, so I should go search youtube for Girdle String? |
| 17:01.53 | p3nguin_ | Do you know the specific model of Ultra X-Wind that you would use for LGA 775? |
| 17:02.38 | Geminizer | has anyone ever had the problem where they are getting constant "Remote UNIX connection disconnected" messages from the CLI ? |
| 17:02.46 | jaytee | p3nguin_, you might also try Arctic Silver thermal paste for the mounting. most ship with just a silicone paste that's less efficient in heat transfer bond between the cpu and heatsink. |
| 17:03.11 | Chainsaw | Geminizer: If you don't like those messages, stop calling asterisk -rx "blah" from your bash scripts. |
| 17:03.21 | jaytee | hehe |
| 17:04.51 | Geminizer | Chainsaw: ahh, ok... so that message doesn't mean functionality is broken anywhere then? |
| 17:05.05 | Chainsaw | Geminizer: No, it is part of normal operation. |
| 17:05.10 | Geminizer | ok... thanks |
| 17:05.55 | p3nguin_ | Hmm, kinda looks like the ULT33131 is the only one they make for the 775. |
| 17:06.04 | Katty | MARINATE FASTER |
| 17:06.10 | Katty | jaytee: no. |
| 17:06.18 | Katty | jaytee: it was a chrysler technical video |
| 17:06.28 | Katty | does anyone have the link to that crhsytal tech video from the other day? |
| 17:07.17 | Katty | http://www.youtube.com/watch?v=MXW0bx_Ooq4 <- here it is |
| 17:07.18 | Katty | yay for logs |
| 17:08.32 | Katty | lunar wane shaft. |
| 17:08.36 | Katty | *hee* |
| 17:09.29 | *** join/#asterisk mnick86 (~mnick86@188-195-95-211-dynip.superkabel.de) |
| 17:09.48 | *** join/#asterisk eject_ck (~eject_ck@haitek.cosmonova.net.ua) |
| 17:09.52 | eject_ck | Hi all |
| 17:09.58 | Katty | ohaider |
| 17:10.08 | p3nguin_ | I'm not so sure that the ULT33131 X-Wind supports Core 2 Quad processors. |
| 17:11.46 | Katty | i bet your mom does. |
| 17:12.01 | p3nguin_ | I doubt she even knows what a Core 2 Quad is. |
| 17:12.23 | Katty | probably not |
| 17:12.28 | Katty | but mine wouldn't either, so it's okay |
| 17:13.07 | *** join/#asterisk outtolunc (~me@c-98-248-96-110.hsd1.ca.comcast.net) |
| 17:13.19 | *** join/#asterisk t_j (~tj@tomjudge.vm.bytemark.co.uk) |
| 17:13.20 | p3nguin_ | I mean, she bought a Belkin home networking appliance, and didn't even bother asking me if there was something better. |
| 17:13.38 | t_j | for a wan trunk is SIP or IAX2 better? |
| 17:13.55 | p3nguin_ | There is no such thing as a SIP trunk, so that leaves IAX2. |
| 17:14.10 | t_j | this is such a thing as a sip trunk |
| 17:14.14 | t_j | so read up on it |
| 17:14.17 | p3nguin_ | ? |
| 17:14.28 | Katty | p3nguin_: he's confused. |
| 17:14.31 | p3nguin_ | I guess. |
| 17:14.34 | Katty | p3nguin_: he's calling it "Sip Trunk" |
| 17:14.37 | Katty | p3nguin_: it's not really what he means |
| 17:14.40 | p3nguin_ | He confused me, too. |
| 17:14.48 | Katty | you just stay confused. |
| 17:15.02 | p3nguin_ | Nah, I go in and out of confusion. |
| 17:15.13 | Katty | k |
| 17:15.21 | p3nguin_ | When things are clear-cut, I fully understand them. |
| 17:15.31 | Katty | pie is clearcut |
| 17:15.33 | t_j | its a trunk that uses SIP for the management protoco |
| 17:15.34 | Katty | i understand pie |
| 17:15.40 | Katty | mmm pie |
| 17:15.46 | p3nguin_ | When people go around trying to tell me there is such a thing as a SIP trunk, then I get confused. |
| 17:16.14 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 17:17.03 | Katty | t_j: there is no such thing as SIP Trunking |
| 17:17.13 | t_j | so what do you call a Trunk that uses SIP as the protocol? |
| 17:17.24 | creativx | our itsp calls it sip trunking |
| 17:17.27 | creativx | but its not |
| 17:17.30 | creativx | its just many sip connections |
| 17:17.54 | Katty | t_j: Sip trunking is a word that marketing weasels made up |
| 17:17.54 | t_j | siptrunk.org... |
| 17:18.23 | Katty | it's a sip peer or connection...or account |
| 17:18.34 | t_j | so why does *everyone* call it a trunk.... |
| 17:18.58 | Katty | Why? |
| 17:19.03 | Katty | Because sales people are stupid |
| 17:19.08 | Katty | and they can't grasp what they're being told. |
| 17:19.11 | Katty | Also, to make money. |
| 17:19.13 | *** join/#asterisk mnicholson (~mnicholso@nat/digium/x-hdepuylnoazngdej) |
| 17:19.14 | Katty | It sounds catchy |
| 17:19.22 | t_j | Level3 RCV AT&T Cisco Sprint Verizion, all the pbx software Asterisk-gui freepbx etc etc etc |
| 17:19.47 | Katty | I'm not here to argue with you. |
| 17:19.56 | Katty | You can take it or leave it |
| 17:20.30 | jaytee | Captain Kangaroo wasn't really a captain |
| 17:20.38 | Katty | LUNCHTIME |
| 17:20.40 | Katty | poofs |
| 17:20.47 | jaytee | yes, time for lunch |
| 17:20.50 | t_j | wow what a bunch of idiots |
| 17:21.10 | Katty | t_j: Really |
| 17:21.22 | Katty | t_j: well then you best move along |
| 17:21.25 | Katty | t_j: cause idiots can't help you |
| 17:21.31 | Katty | t_j: we're obviously not smart enough |
| 17:21.42 | *** join/#asterisk mnicholson (~mnicholso@nat/digium/x-qqsrvcyhnfleygtk) |
| 17:22.38 | *** join/#asterisk blebleble (~godie@s7.n194.n222.n216.static.myhostcenter.com) |
| 17:22.52 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 17:23.10 | t_j | Katty: so what is a trunk? in telephony terms... |
| 17:23.15 | rubberneck | I see the following in the CLI: Auto fallthrough, channel 'Local/<number is hidden>@outgoing-0b59;2' status is 'NOANSWER', how can i get that "NOANSER' variable? |
| 17:23.20 | p3nguin_ | ~trunk |
| 17:23.20 | infobot | somebody said trunk was a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. |
| 17:25.44 | t_j | so back to the point whats better for a BRI Trunk, SIP or IAX2 ? |
| 17:27.33 | *** join/#asterisk fink (~guest@static-162-84-93-164.fred.east.verizon.net) |
| 17:29.30 | *** join/#asterisk datagen24 (~datagen24@173-13-83-142-NewEngland.hfc.comcastbusiness.net) |
| 17:31.03 | *** join/#asterisk Kami|Busy (~not@office.supersoccer.co.uk) |
| 17:31.06 | Kami|Busy | hi |
| 17:31.21 | Kami|Busy | I was wondering if asterisk can be used to receive faxes |
| 17:31.39 | datagen24 | I am trying to figgure out a dial plan for a trunk we just added to our system in belgium and I have been unable to find any good tutorial or examples for anything but north america, at least in english. can someone walk me through setting up my first international dialing plan? |
| 17:32.00 | coppice | Kami|Busy: yes, of course it can |
| 17:32.17 | Kami|Busy | any links on how it can be setup ? |
| 17:32.33 | coppice | with 1.6 or 1.4? |
| 17:32.39 | ManxPower-work | ~trunk |
| 17:32.40 | infobot | [trunk] a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. |
| 17:34.15 | t_j | info bots definition is retarded |
| 17:34.24 | datagen24 | ok, I am using IAX2 to connect it..... not sure if it was the right terminology |
| 17:34.38 | t_j | in telecom a trunk is a collection of channels between switches |
| 17:35.37 | Kami|Busy | Coppice : Asterisk CVS-HEAD-01/04/05-09:54:06 -> is this the right info you need? |
| 17:36.37 | *** join/#asterisk smooth_penguin (~smoove@59.95.21.128) |
| 17:40.13 | *** join/#asterisk Wildy (~simba@mas4-gw.pleer.ru) |
| 17:42.59 | *** part/#asterisk t_j (~tj@tomjudge.vm.bytemark.co.uk) |
| 17:43.03 | leifmadsen | woh... 2005! |
| 17:43.34 | *** join/#asterisk benngard (~benngard@90-230-92-67-no148.tbcn.telia.com) |
| 17:43.50 | leifmadsen | ManxPower-work: should also add information about "Asterisk 'trunk' is the latest changes to Asterisk -- do not use 'trunk' unless you know what you're doing." |
| 17:43.53 | leifmadsen | or something like that :) |
| 17:44.02 | *** join/#asterisk gelpg (~chatzilla@qf.invitel.hu) |
| 17:44.26 | benngard | .9 |
| 17:44.29 | benngard | :) |
| 17:44.31 | coppice | 2005 is so old FAX was still an up to date medium :-) |
| 17:44.55 | benngard | or dont have a choice other than running trunk |
| 17:45.33 | gelpg | hi,I have an ISDN card with bri_cpe signalling |
| 17:46.12 | gelpg | if I receive a call from the range what my telco gives, the asterisk picks up the last two digit |
| 17:46.18 | coppice | t_j: actually text books define a trunk as a single shared communication channel |
| 17:47.14 | gelpg | if I'd like to dial out, it uses the first open line not the public phone number what is assigned to the extension |
| 17:47.49 | gelpg | how can I set the dedicated phone number to the each extensioons? |
| 17:48.33 | p3nguin_ | Set(CALLERID(num)) |
| 17:48.44 | coppice | Kami|Busy: 2005 is so old I don't even know what version of asterisk was the trunk one them. you might need something newer to take advantage of the fax facilities. With Asterisk 1.6, if you install spandsp before building Asterisk it will build FAX applications for you |
| 17:49.16 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 17:52.26 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 17:52.56 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 17:54.03 | Kami|Busy | Yea the box is old, thats why I had doubts. Will I need any special hardware in addtion to this? The card in there atm is a TE405 or something. |
| 17:54.12 | Kami|Busy | Makes me doubt if its capable of faxes. |
| 17:54.42 | *** join/#asterisk BCS-Satori (~BCS-Sator@75-148-21-113-WashingtonDC.hfc.comcastbusiness.net) |
| 17:55.17 | coppice | the card is OK |
| 17:55.54 | *** join/#asterisk sun28 (~light@sun28.ipfw.su) |
| 17:55.58 | Kami|Busy | ah, cool. Thanks for the info. |
| 17:56.39 | *** join/#asterisk MrTelephone (~test@h697179-171.picriverisp.net) |
| 17:56.48 | *** part/#asterisk datagen24 (~datagen24@173-13-83-142-NewEngland.hfc.comcastbusiness.net) |
| 17:56.54 | *** join/#asterisk smooth_penguin (~smoove@59.95.21.128) |
| 17:56.59 | MrTelephone | how do you get more information on the error saying DAHDI is not configured properly when you start asterisk? |
| 17:57.27 | benngard | hmm, when u answer() a call is there some variable that u can check for t.38 or udptl or so? |
| 17:58.31 | BCS-Satori | Does phoneprov.conf only work under Asterisk HTTP? or can I use apache? |
| 18:00.38 | Katty | bmoraca_work: Yum. |
| 18:00.40 | Katty | bmoraca_work: that's a keeper. |
| 18:02.29 | bmoraca_work | Katty: you made it? glad you liked it. it's really easy, too |
| 18:03.04 | Katty | bmoraca_work: yep (= i made rice and green peppers to go with it |
| 18:03.26 | bmoraca_work | nice |
| 18:03.59 | Katty | bmoraca_work: it's only recently that i've decided to eat more seafood for its omega3 |
| 18:04.11 | Katty | bmoraca_work: so fish and shrimp recipes are exactly what i need |
| 18:04.59 | bmoraca_work | i've started eating it because it's healthier for me than beef and it doesn't give me gout...so... although i've never been a big seafood person |
| 18:05.41 | bmoraca_work | Katty: i have a good recipe for a sweet and spicy peach shrimp, too, in a slow cooker that came out really good. and of course shrimp jambalaya is always good |
| 18:05.53 | bmoraca_work | if i remember, i'll bring both in for you tomorrow |
| 18:06.17 | Katty | :>>> |
| 18:06.19 | Katty | cheers |
| 18:06.20 | MrTelephone | has anyone had trouble with dahdi and asterisk 1.4.29? |
| 18:06.30 | thehar | has anyone used TransNexus? |
| 18:06.36 | Katty | has anyone seen my shoes? |
| 18:06.36 | gelpg | p3nguin: I did but still starts with the first number |
| 18:06.48 | *** join/#asterisk tkrn (~tkrn@216.196.213.246) |
| 18:07.02 | tkrn | how do i use exent => to transfer back to the IVR? |
| 18:07.25 | tkrn | exent => 101,4,????? |
| 18:07.37 | [TK]D-Fender | tkrn: Goto() <- |
| 18:07.39 | rubberneck | tkrn: yes |
| 18:07.54 | tkrn | ty! i was sitting in the trixbox channel and its a ghost town in there! |
| 18:07.56 | [TK]D-Fender | tkrn>exent => 101,4,????? <-- no "t" in "exten" |
| 18:08.05 | tkrn | opps you got fender |
| 18:08.38 | tkrn | so goto would referance the number of the IVR? |
| 18:09.37 | [TK]D-Fender | tkrn: "core show application goto" |
| 18:10.09 | tkrn | think i have it from here, thank you for the right places to look |
| 18:10.27 | p3nguin_ | 'no "t" in "exten"' |
| 18:10.28 | p3nguin_ | That's a good one. |
| 18:10.48 | [TK]D-Fender | p3nguin_: SHH! |
| 18:10.50 | [TK]D-Fender | ;) |
| 18:11.03 | MrTelephone | never seen any shoes |
| 18:11.21 | MrTelephone | give me a good pointer on how to compile dahdi and the dummy driver module only :) |
| 18:11.35 | *** join/#asterisk p4p4 (~P4p4@248.121.113.82.net.de.o2.com) |
| 18:12.13 | Katty | resists comment |
| 18:12.15 | Katty | struggles |
| 18:12.18 | Katty | twitches |
| 18:12.23 | *** join/#asterisk soman (~somnath@dsl-jklbrasgw1-fe19fb00-113.dhcp.inet.fi) |
| 18:12.26 | MrTelephone | I was checking my sangoma interface properties and see 96 carrier drops in the last 200 days. Should that be considered a problem? |
| 18:13.33 | hardwire | MrDroppedTelephone |
| 18:13.39 | b14ck | Sup everyone. |
| 18:13.44 | b14ck | I have a rather odd question to ask =p |
| 18:13.45 | Katty | b14ck: ohaider |
| 18:13.47 | MrTelephone | yeah no kitting |
| 18:13.51 | MrTelephone | kidding rather |
| 18:13.52 | Katty | no kitting |
| 18:13.57 | Katty | no KNITTING ALLOWED |
| 18:14.01 | MrTelephone | EH? |
| 18:14.06 | MrTelephone | go canada gold |
| 18:14.11 | b14ck | Would anyone be interested in doing some consulting work for me to help with some networking stuff? It isn't really Asterisk related. But I'm having a hard time finding someone to help with PIX stuff. |
| 18:14.13 | Katty | olympic knitting? |
| 18:14.26 | Katty | b14ck: i'm sure several folk would be. check out #asterisk-consultants |
| 18:14.29 | MrTelephone | olympic canadian mitts knitted in china? |
| 18:14.32 | b14ck | will do |
| 18:14.40 | MrTelephone | I'll take 4 pairs |
| 18:14.54 | Katty | you should outsource your job to China |
| 18:15.12 | Katty | you can pay a guy to do your work for you |
| 18:15.14 | MrTelephone | India is more technically advanced, they helped me format my DELL |
| 18:15.22 | Katty | OoOOooo so schnazzy |
| 18:15.49 | b14ck | I hate cisco pix |
| 18:15.50 | b14ck | :( |
| 18:15.58 | MrTelephone | They also told me windows crashed because the dvd drive had bad sectors on it and not the hard drive |
| 18:16.00 | Katty | i am sorry to hear that |
| 18:16.06 | Katty | would you like a cookie to ease your pain? |
| 18:16.22 | Katty | MrTelephone: pesky bad sectors. |
| 18:16.26 | *** join/#asterisk michael-i (~michael-i@141.41.40.185) |
| 18:16.28 | Katty | MrTelephone: those darn dvd drives!!!! |
| 18:16.42 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 18:16.48 | MrTelephone | recyclical redundancy failures. damn |
| 18:17.08 | Katty | the lunar wane shaft wasn't calibrated correctly |
| 18:17.15 | hardwire | wane? |
| 18:17.24 | hardwire | I'm with you on lunar and shaft.. but wane? |
| 18:17.33 | MrTelephone | dahdi seems to flip a coin to see if it will install the dummy driver or not making it hard to get the span configurations nailed down |
| 18:17.35 | Katty | http://www.youtube.com/watch?v=MXW0bx_Ooq4 <- hardwire |
| 18:17.37 | Katty | ^- watch |
| 18:17.49 | jaytee | I love it when I call Dell and get an Indian call center. "Hello, my name is 'Bob', thank you for calling Dell." "Hi, Bob, would you mind talking a little slower cuz your East Nebraska accent is really throwing me off." |
| 18:18.18 | Katty | I got a girl named Lauren this morning at Symantec. |
| 18:18.28 | MrTelephone | You ever get a north american accent and you feel bad because the local call center shut down and he had to move to India or get fired? |
| 18:18.34 | Katty | and i'm pretty sure that her name wasn't Lauren |
| 18:18.35 | [TK]D-Fender | Katty: BackupExec division? |
| 18:18.47 | jaytee | oh dear god, symantec sucks the dingle balls off a bear's arse |
| 18:18.52 | MrTelephone | What is the number to the asterisk call center |
| 18:18.52 | Katty | [TK]D-Fender: no |
| 18:18.57 | hardwire | prefabulated ambulite. |
| 18:19.03 | Katty | hardwire: ;> |
| 18:19.28 | MrTelephone | I'll email you my errors to jaytee@symantec.ca |
| 18:19.32 | hardwire | it appears your right about the calibration problem. |
| 18:19.38 | Katty | hardwire: mhmm |
| 18:19.47 | Katty | it's that inverse reactive current |
| 18:19.52 | Katty | and those cardinal gram meters |
| 18:19.56 | MrTelephone | stick a diode in there |
| 18:20.05 | Katty | "relative motion of conductors and fluxes" |
| 18:20.10 | Katty | ;P |
| 18:21.01 | Katty | i would say the relative motion of the flux is highly dependant on the condutor |
| 18:21.01 | jaytee | http://xkcd.com/123/ |
| 18:21.04 | gelpg | exten => 53,1,Set(CALLERID(num)=3914653), this syntax is corret? I'd like to set the caller id to an extension |
| 18:21.04 | Katty | and its rate of motion |
| 18:21.28 | michael-i | After moving my system from dahdi 2.2.0.2 to 2.2.1, a custom driver no longer wants to load its zone data. Kernel: Cannot start tones until tone zone is loaded. Can anyone point me in the right direction. Grepping for zone, etc...reveals nothing. |
| 18:21.37 | Katty | Magneto Reluctance!!! |
| 18:21.47 | p3nguin_ | gelpg: That sets the callerID when you call extension 53. |
| 18:21.55 | MrTelephone | when i read that electrons can form eddies on a copper conductor I dropped out of electric science |
| 18:22.06 | MrTelephone | thats too weird |
| 18:22.16 | bmoraca_work | b14ck: what's wrong with your pix? |
| 18:22.29 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 18:22.53 | gelpg | p3nguin: so maybe I'm lame, I need this number when I call from exten 53. How can I set up that? |
| 18:22.56 | Katty | the formula for magnetic reluctance is magnetomotive force dividied by magnetic flux |
| 18:23.07 | b14ck | bmoraca_work, want to join #asterisk-consultants? |
| 18:23.07 | Katty | in AMPERE TURNS |
| 18:23.22 | p3nguin_ | gelpg: You can't call FROM extensions. You call extensions from phones. |
| 18:23.45 | p3nguin_ | gelpg: So set the callerid option for the phone device in sip.conf if it is a sip phone. |
| 18:23.50 | *** join/#asterisk Da-Geek (~Da-Geek@87-194-2-213.bethere.co.uk) |
| 18:23.53 | Katty | so how do you get from magnetic flux following the path of lease magnetic reluctance |
| 18:24.00 | Katty | to the relative motion of conductors and flux |
| 18:24.11 | gelpg | p3nguin: no, it"s a dahdi device (analog phone) |
| 18:24.49 | p3nguin_ | gelpg: Does dahdi's config file support setting callerid for its channels? |
| 18:24.52 | MrTelephone | what kind of redundancy do you think is better? meshed asterisk dialplan between 2 boxes with dns srv clients, or heartbeat? |
| 18:25.10 | Katty | what is the motion relative to? |
| 18:25.17 | Katty | if it takes the path of least resistance |
| 18:25.40 | Katty | WHERE IS THE DISPLACEMENT |
| 18:25.51 | jaytee | I thought Magneto Reluctance was where Magneto just kept going back and forth on, "Should I kill Professor Xavier or not?" |
| 18:26.03 | Katty | well. |
| 18:26.07 | Katty | i guess that could be. |
| 18:26.13 | Katty | i assumed he meant magnetic reluctance tho |
| 18:26.36 | MrTelephone | how many redbulls did you drink at astricon |
| 18:26.40 | Katty | but even then, it's relative motion |
| 18:26.47 | [TK]D-Fender | gelpg: You don't call FROM an EXTENSION. You call from a DEVicE, and you set the callerid int he device's config |
| 18:26.49 | Katty | that means at minimum of two comparitve objects |
| 18:26.50 | bmoraca_work | MrTelephone: my opinion is that active-active (the former) is superior to active-standby (the latter) |
| 18:26.56 | Katty | jaytee flying on a plane, and me riding a bike |
| 18:27.01 | Katty | my relative motion to jaytee... |
| 18:27.08 | sun28 | moin \o/ |
| 18:27.21 | Katty | i guess there /could/ be a relative motion between a conductor and the flux |
| 18:27.22 | [TK]D-Fender | plus /o\ |
| 18:27.23 | Katty | but... |
| 18:27.38 | sun28 | [TK]D-Fender: ^_^ |
| 18:27.40 | Katty | yeah |
| 18:27.50 | p3nguin_ | gelpg: There has to be a config for setting the phones' authentication information. The callerid setting probably goes in there. I don't have any dahdi stuff, or I would probably be able to tell you exactly where/what needs work. |
| 18:28.03 | sun28 | ps xau | grep asterisk |
| 18:28.05 | sun28 | .... |
| 18:28.13 | sun28 | /o\ |
| 18:28.14 | p3nguin_ | ps -C asterisk |
| 18:28.27 | gelpg | p3nguin: thanks for the tip, I'm, looking for it |
| 18:28.38 | Kobaz | hmm |
| 18:28.43 | Katty | but he didn't say relative motion of conductor vs flux |
| 18:28.49 | Kobaz | okay so... i see StackPop() |
| 18:29.02 | Kobaz | is there a way to actually get the stack item that was popped off? |
| 18:29.12 | Kobaz | StackPop(): Removes last label on the stack, discarding it. |
| 18:29.16 | MrTelephone | who here can comment on the reliability of a cisco as5300 with t1 interface as opposed to sangoma/digium pci cards? |
| 18:29.34 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 18:29.42 | Katty | maybe he meant rotational motion :> |
| 18:29.44 | Kobaz | MrTelephone: the reliability of line cards is really a matter of the reliability of the machine it's hooked up to |
| 18:30.00 | MrTelephone | I find that the pci cards work ok but sometimes calls will cut into other streams briefly.. pci timing issues I'm guessing |
| 18:30.01 | bmoraca_work | MrTelephone: i use an AS5400 and it works fantastically. there were a couple little quirks about them, but they work very, very well |
| 18:30.02 | mnick86 | does anyone know this error message: dahdi_pri_error: Call Reference Length not supported: 0 ? |
| 18:30.20 | MrTelephone | bmoraca, what kind of quirks? |
| 18:30.24 | sun28 | p3nguin_: ps -C only in gnu ps :/ |
| 18:30.28 | Katty | bmoraca_work IS a quirk. |
| 18:30.38 | Kobaz | i wants to peek the stack! |
| 18:31.05 | sun28 | uses * on freebsd only |
| 18:31.05 | Katty | peek...the stack? |
| 18:31.08 | bmoraca_work | Katty: don't hate me 'cause i'm beautiful :P |
| 18:31.19 | Katty | bmoraca_work: what are you talkin about |
| 18:31.26 | bmoraca_work | MrTelephone: weird instances of dial-peer matching and things like that |
| 18:31.26 | Kobaz | Katty: yes... i want to see who called me |
| 18:31.27 | Katty | bmoraca_work: we are all luvs all the time |
| 18:31.39 | Katty | Kobaz: well that's rather difficult |
| 18:31.40 | p3nguin_ | sun28: I have procps version 3.2.8 |
| 18:31.43 | Katty | Kobaz: they could be miles away |
| 18:31.54 | Kobaz | Katty: so something like StackPop would be great... but I don't actually want to pop it... and i want to see what it is |
| 18:31.58 | Katty | Kobaz: and if your visions that good, i gotta take some more vitamins |
| 18:32.09 | MrTelephone | bmoraca, how is the echo cancellation? |
| 18:32.09 | Katty | Kobaz: why isnt' regular callerid working? |
| 18:32.14 | Kobaz | it's not callerid |
| 18:32.14 | bmoraca_work | MrTelephone: biggest quirk i found was that dial-peers were not being matched as they should have when a call came in that had hidden callerid. |
| 18:32.19 | Kobaz | i want to look at the dialplan stack |
| 18:32.23 | Katty | Kobaz: ooooooh |
| 18:32.24 | Katty | Kobaz: isee |
| 18:32.29 | Kobaz | who called, as in... what context jumped to my context |
| 18:32.35 | Kobaz | and from what line |
| 18:32.37 | MrTelephone | bmoraca, do you still log calls via asterisk logs or cisco logs? |
| 18:32.49 | *** join/#asterisk smooth_penguin (~smoove@59.95.21.128) |
| 18:32.49 | Katty | Kobaz: debug and log files? |
| 18:32.54 | Katty | smooth_penguin: :> |
| 18:32.56 | Kobaz | hah, no... this needs to be in dialplan |
| 18:32.57 | Katty | hugs smooth_penguin |
| 18:33.02 | bmoraca_work | MrTelephone: as far as quality goes, i have not had any problems at all. that said, i don't do my transcoding on the AS5400. I've heard that there are some issues with these devices and transcoding. |
| 18:33.03 | Katty | Kobaz: k |
| 18:33.03 | Kobaz | I think I'm going to write a StackPeek() |
| 18:33.10 | bmoraca_work | MrTelephone: i don' |
| 18:33.10 | Katty | superdeduper |
| 18:33.23 | smooth_penguin | hey Katty, sup |
| 18:33.26 | bmoraca_work | MrTelephone: i only log errors. CDRs are handled by Asterisk |
| 18:33.27 | Katty | hmm :/ i gots to go look at a winders server here shortly |
| 18:33.34 | Katty | smooth_penguin: kinda bummed. |
| 18:33.43 | bmoraca_work | MrTelephone: though, if you REALLY wanted to, you could have the AS5300 handle CDRs to a radius server. |
| 18:33.54 | smooth_penguin | heh |
| 18:34.04 | Katty | smooth_penguin: how're you? |
| 18:34.46 | smooth_penguin | Katty, am good taken off from work to finish up some stuff at home :p |
| 18:35.06 | Katty | smooth_penguin: i wanna take off work :< |
| 18:35.12 | Katty | pouts |
| 18:36.02 | *** join/#asterisk Lord-Rahl (~quassel@173-162-45-177-michigan.hfc.comcastbusiness.net) |
| 18:36.36 | smooth_penguin | heh, Katty the weekend is pretty far off :p |
| 18:37.28 | Katty | it so is |
| 18:37.37 | Katty | a whole day and 5 hours |
| 18:37.42 | Katty | UNACCEPTABUHLS |
| 18:39.48 | smooth_penguin | it gets longer when you think about it |
| 18:39.53 | smooth_penguin | find fun stuff to do |
| 18:40.40 | Katty | kay |
| 18:41.15 | smooth_penguin | like route people to tt-monkeys if you can :p |
| 18:41.33 | Lord-Rahl | I am running asterisk 1.4.28. the problem I am having is Queue's I have set context = support_out_queue , but when I press a key to drop to this context nothing happens. |
| 18:42.13 | Lord-Rahl | is there a bug in the queue with couse this |
| 18:42.25 | Lord-Rahl | causes* |
| 18:42.50 | carrar | There is a BUG and it's craweling up my ARM!!!! |
| 18:43.18 | Lord-Rahl | lol |
| 18:44.00 | Katty | i got somethin better than the monkeys |
| 18:44.15 | Katty | i've got rick astley |
| 18:45.13 | Katty | carrar: ohaider |
| 18:45.25 | smooth_penguin | Katty, that wont freak them out though |
| 18:45.44 | carrar | I |
| 18:45.44 | Katty | no need to freak them out |
| 18:45.50 | carrar | Just wanna tell you how I feel |
| 18:45.55 | MrTelephone | bmoraca, I'm not really sure if asterisk cdr will say the call is connected when it establishes a sip session with the as5300 before going out on the t1. For example, if the remote line is busy will the cdr still say "ANSWERED"? |
| 18:46.20 | ManxPower-work | MrTelephone: only if the gateway answers the call |
| 18:46.30 | Lord-Rahl | No one has had the queue problem before??? |
| 18:46.34 | *** join/#asterisk wikki (~wikki@doraemon.wikki.com) |
| 18:46.51 | carrar | Katty, do you rickroll yourself for fun? |
| 18:46.59 | ManxPower-work | Lord-Rahl: Generally the ones that try using queues end up jumping off a building |
| 18:47.05 | bmoraca_work | MrTelephone: no. the AS5300 will report the correct sip codes |
| 18:47.09 | ManxPower-work | (and no we don't push them off) |
| 18:47.10 | smooth_penguin | brb |
| 18:47.21 | Lord-Rahl | ManxPower-work: lol |
| 18:47.50 | Lord-Rahl | ManxPower-work: is there a better solution? |
| 18:48.12 | [TK]D-Fender | Lord-Rahl: Show us your failed call @ cli. Show us your configs. |
| 18:48.13 | MrTelephone | Can the as5300 operate a channel bank like an adit600? I guess all the communication between asterisk and the adit600 would be SIP.. |
| 18:48.26 | *** join/#asterisk smooth_penguin (~smoove@59.95.21.128) |
| 18:48.40 | ManxPower-work | MrTelephone: does the AS5300 have FXO or FXS ports? |
| 18:48.45 | carrar | MrTelephone, I use ADIT600 with T1's between the ADIT and Asterisk |
| 18:48.55 | carrar | they work AWESOME |
| 18:49.10 | MrTelephone | No, just a dual t1 card. My asterisk box has a dual port card. 1 port going to the telco and 1 port going to an adit 600 |
| 18:49.12 | Lord-Rahl | [TK]D-Fender: should I use pasted bin? |
| 18:49.22 | [TK]D-Fender | Lord-Rahl: Clearly |
| 18:49.30 | carrar | MrTelephone, thats good setup |
| 18:49.36 | ManxPower-work | MrTelephone: hard to be a channel bank when you have no analog ports |
| 18:50.08 | p3nguin_ | lord-rahl: I didn't experience that problem before I upgraded to 1.4.29. |
| 18:50.46 | carrar | MrTelephone, the ADIT600 doesn't do sip last time I checked |
| 18:51.06 | MrTelephone | ManxPower. I can't seem to grab the concept of FXO <-> ADIT 600 <-T1-> AS5300 <-SIP-> ASTERISK |
| 18:51.20 | ManxPower-work | Wow, people on #mysql post links to useless sites like Facebook just like here1 |
| 18:52.02 | Lord-Rahl | [TK]D-Fender: http://codepad.org/ElcB4sN8 is not much there. when i press a key nothing is register in cli i have verbose set to 35 |
| 18:52.07 | *** join/#asterisk korihor (~korihor@201.210.226.98) |
| 18:52.11 | bmoraca_work | MrTelephone: you'll need an external channel bank, but yes, the AS5300 can route to a channel bank |
| 18:52.14 | ManxPower-work | MrTelephone: think of it this way. Provider (FXO <-> ADIT 600 <-T1-> AS5300 ) <-Internet-> Asterisk |
| 18:52.25 | Lord-Rahl | p3nguin_: were you able to fix it |
| 18:52.34 | [TK]D-Fender | Lord-Rahl: show me ALL of them. that is a tiny piece of what i asked for |
| 18:53.49 | MrTelephone | ManxPower-work, I had it backwards I beleive. (Handsets <-> FXS <-> ADIT 600 <-T1-> AS5300 ) <-Internet-> Asterisk |
| 18:54.03 | Lord-Rahl | [TK]D-Fender: sorry updated |
| 18:54.12 | ManxPower-work | MrTelephone: this isn't rocket science. |
| 18:54.19 | *** join/#asterisk high-freq (~hfreq@99.188.122.87) |
| 18:54.33 | ManxPower-work | What I dont' understand is why you even have theAS5300 in the mix |
| 18:55.48 | bmoraca_work | he doesn't, but it may give him better audio quality or capacity depending on calling patterns, provider, and hardware |
| 18:55.54 | MrTelephone | Wanted to have a more reliable gateway other than a PC+T1 PCI card.. |
| 18:56.07 | [TK]D-Fender | Lord-Rahl: NEW LINK PLEASE |
| 18:56.29 | Lord-Rahl | [TK]D-Fender: http://codepad.org/O5N0Iu3u |
| 18:56.41 | [TK]D-Fender | MrTelephone: ADIT 600 <-T1-> AS5300 ) <--- tragic waste |
| 18:57.06 | [TK]D-Fender | MrTelephone: Sing Mediatrix 4124 gateway does it in 1 box with no BS. |
| 18:57.11 | [TK]D-Fender | single* |
| 18:57.44 | MrTelephone | I had to integrate with an already existing meridian system :( Nortel CICS <-> ADIT 600 <-> ASTERISK is the current setup |
| 18:57.51 | coppice | mediatrix might qualify as the world's most creative producer of bugs |
| 18:58.31 | Kobaz | Katty: i finished my StackPeek() |
| 18:58.39 | Kobaz | well, not totally |
| 18:58.42 | Kobaz | need to add some more stuff |
| 18:58.43 | [TK]D-Fender | Lord-Rahl: [support_out_queue] <-- this only has "s" in it, and links to other crap that only has "s" in in. no NUMBEREd extensions to dial |
| 18:58.52 | [TK]D-Fender | Lord-Rahl: that is a massacred GUI config |
| 18:59.00 | Kobaz | <PROTECTED> |
| 18:59.03 | Kobaz | [Feb 18 13:58:02] ERROR[22871]: app_stack.c:220 peek_exec: 261 trunkhandler_from-branch 8 0 |
| 18:59.18 | [TK]D-Fender | Lord-Rahl: [support_out_queue] <-- this appears TWICE even within the same config. |
| 19:00.09 | p3nguin_ | lord-rahl: I _didn't_ experience that problem before I upgraded to 1.4.29. |
| 19:00.14 | [TK]D-Fender | MrTelephone: Ditch the ADIT or reuse for other purposes and jsut shove a 4124 in front. |
| 19:00.40 | Lord-Rahl | [TK]D-Fender: I have the gui for user to do quick think like making users and stuff |
| 19:00.41 | [TK]D-Fender | MrTelephone: Lot cheaper than running a Cisco on topn and it frees up your ADIT for use elsewhere |
| 19:00.43 | michael-i | is it necessary to manually load dahdi_transcode and dahdi_echocan_*? |
| 19:00.55 | [TK]D-Fender | Lord-Rahl: Well there is nothing for your wueue callers to dial in that context. |
| 19:01.00 | bmoraca_work | MrTelephone: or rock an Adtran TA924 :P |
| 19:01.06 | *** join/#asterisk moy (~moy@74.12.129.100) |
| 19:01.45 | [TK]D-Fender | queue* |
| 19:01.50 | michael-i | I'm burnt for today fighting custom drivers on blackfin! Adios everyone. Time for some major supper. |
| 19:02.21 | *** join/#asterisk Alagar (~Administr@122.164.33.74) |
| 19:02.27 | *** join/#asterisk smooth_penguin (~smoove@59.95.21.128) |
| 19:03.03 | MrTelephone | I'll probably replace the meridian with polycoms before buying more channel bank equipment |
| 19:04.48 | jaytee | Kobaz, is that StackPeek a queue realtime stat viewer? |
| 19:04.54 | Kobaz | no |
| 19:05.00 | Kobaz | should it be? |
| 19:05.13 | Kobaz | i'll post my patch in a little bit |
| 19:05.58 | *** join/#asterisk luke-jr (~luke-jr@2002:62b3:1d4c:0:20e:a6ff:fec4:4e5d) |
| 19:06.07 | Lord-Rahl | [TK]D-Fender: Thank you, you may look at some think I missed. It working now |
| 19:07.14 | bmoraca_work | MrTelephone: get a Portico TVA and just reuse the nortel phones with asterisk directly! |
| 19:07.31 | [TK]D-Fender | EWW |
| 19:07.40 | [TK]D-Fender | bmoraca_work: Unstable last I heard... and pricy |
| 19:07.59 | bmoraca_work | don't know much about stability, but they ARE expensive |
| 19:08.07 | [TK]D-Fender | bmoraca_work: I'd only consider it if they had ancient wiring and building issues in redoing it |
| 19:08.22 | bmoraca_work | yep |
| 19:08.42 | bmoraca_work | lots of older buildings don't have ethernet to where they need all the phones, so it's a good option there. |
| 19:08.58 | carrar | If they have a PC then they have ethernet |
| 19:09.14 | kjs | whats the module for DAHDI called? |
| 19:09.23 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 19:09.36 | [TK]D-Fender | kjs: Which module? |
| 19:10.04 | kjs | the one so I can get meetme working, someone did tell me the other day, but I forgot to note it down. |
| 19:10.05 | [TK]D-Fender | bmoraca_work: I'd sooner run new ethernet around than spend the money of the TVA |
| 19:10.15 | [TK]D-Fender | kjs: dahdi_dummy |
| 19:10.22 | kjs | thats it, thanks :) |
| 19:11.46 | *** part/#asterisk Lord-Rahl (~quassel@173-162-45-177-michigan.hfc.comcastbusiness.net) |
| 19:11.55 | *** join/#asterisk Alagar (~Administr@122.164.33.74) |
| 19:12.10 | bmoraca_work | carrar: lots of places don't have PCs at all their locations where phones are. manufacturing plants, for instance. |
| 19:12.18 | carrar | WHAT |
| 19:12.18 | [TK]D-Fender | bmRun wire! |
| 19:12.21 | carrar | How is that POSSIBLE! |
| 19:12.46 | carrar | RUN WIRE RUN |
| 19:13.11 | carrar | Give everyone Polycom Spectralink wifi phones! |
| 19:13.19 | carrar | :) |
| 19:13.28 | bmoraca_work | carrar: business owners are cheap bastards and don't feel the need. also, running wire is often not a viable option. we've got some 100-year-old buildings with hardpan ceilings that it's just not feasible to run wire |
| 19:14.24 | carrar | run alone the floor wall |
| 19:14.27 | carrar | along |
| 19:14.39 | carrar | with sexy little covers |
| 19:14.42 | *** join/#asterisk mnt_real (~sinan@bas1-montreal43-1177754737.dsl.bell.ca) |
| 19:14.52 | bmoraca_work | ewww...raceway is narsty |
| 19:15.32 | carrar | and a 100 year old building isn't already narsty? |
| 19:15.33 | bmoraca_work | though we have done that at places. my father-in-law's business's building is all cinderblock. |
| 19:17.27 | [TK]D-Fender | bmoraca_work: THAt is the kind of scenario that would validate a TVA |
| 19:17.33 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 19:18.14 | Naikrovek | what's tva |
| 19:18.19 | Naikrovek | joins the conversation |
| 19:18.30 | ManxPower-work | Tennesee Valley Authority |
| 19:18.34 | bmoraca_work | Naikrovek: an ATA that emulates various digital PBXes (Nortels, mostly) |
| 19:18.34 | Naikrovek | ah |
| 19:18.40 | carrar | heh |
| 19:18.53 | *** join/#asterisk jeffrey (~jeffrey@unaffiliated/Jeffrey) |
| 19:19.03 | carrar | Manx is from the Valley for sure |
| 19:19.05 | bmoraca_work | Naikrovek: basically, it lets you use your current digital phones with a SIP PBX like Asterisk |
| 19:19.31 | Naikrovek | nice |
| 19:19.35 | Naikrovek | sounds very useful |
| 19:20.06 | bmoraca_work | Naikrovek: they sound more useful than they really are...they're very expensive and difficult to configure properly |
| 19:20.24 | Kobaz | the problem is... with the cost per port... it's sometimes almost as expensive as rewrireing and getting ip phones |
| 19:20.29 | Naikrovek | really |
| 19:20.31 | Naikrovek | huh |
| 19:20.54 | Kobaz | bmoraca_work: i've have several citel portico failures too |
| 19:20.57 | bmoraca_work | Naikrovek: it's almost always cheaper just to purchase all new phones. however, there's more to their usefulness than just that...they let you use current cabling, and infrastructure costs are almost always the highest |
| 19:21.05 | Kobaz | bmoraca_work: like, random ports just fry |
| 19:21.20 | Kobaz | half of a 24 port gateway went bad overnight |
| 19:21.26 | bmoraca_work | ouch |
| 19:21.57 | Kobaz | citel almost went bankrupt |
| 19:22.00 | Kobaz | they got bought out though |
| 19:22.03 | bmoraca_work | not surprised |
| 19:22.22 | Kobaz | they had something like 3 million in inventory they were sitting on |
| 19:22.38 | Kobaz | their stock was like in the pennies |
| 19:22.42 | *** join/#asterisk ruben23 (~AGENT@122.55.48.243) |
| 19:22.45 | bmoraca_work | i had a presentation from someone at Citel a couple years ago...i was intrigued to say the least, but the cost was way too high for most of my applications |
| 19:23.03 | Kobaz | but. not enough stock was on the open market for a hostile takeover... i think it was only 40% was public |
| 19:23.24 | Kobaz | if more stock was out, you could have bought the company for $300k in cash |
| 19:25.50 | coppice | citel had something that was almost, but quite, a good idea |
| 19:28.50 | Kobaz | there is another company similar called phybridge |
| 19:28.59 | Kobaz | doing 10mbit ethernet and poe over copper |
| 19:29.20 | Kobaz | but with the per-port cost... in most cases you might as well rewrite |
| 19:29.23 | Kobaz | rewire |
| 19:29.33 | [TK]D-Fender | bmoraca_work: A niche product for sure. But good that it is at least semi-reasonable |
| 19:31.49 | bmoraca_work | Kobaz: that UniPhyer product is probably just an ethernet-extender-type product...broadband over two pairs to provide internet. the interesting thing is being able to provide PoE...don't stick an analog phone on that line, it'll blow up! |
| 19:33.00 | bmoraca_work | i wonder how expensive they are, though... |
| 19:33.37 | Kobaz | heh |
| 19:33.50 | Kobaz | i got quotes from them a while ago it was like 150 bucks a port |
| 19:33.57 | bmoraca_work | ouch |
| 19:45.37 | ManxPower-work | that's cheap |
| 19:46.14 | *** join/#asterisk trelane` (trelane@funtoo/staff/trelane) |
| 19:49.12 | Kobaz | hmm |
| 19:49.25 | tkrn | is there a way to boost the volume of the playback of system recordings? |
| 19:49.38 | Kobaz | i think i need to add a StackPush() tooo |
| 19:49.57 | Kobaz | tkrn: record them with MixMonitor and bump up the volume |
| 19:50.10 | tkrn | ty kobaz |
| 19:50.11 | carrar | i think i need some chocolate and penutbutter icecream |
| 19:51.06 | *** join/#asterisk ruben23 (~AGENT@122.55.48.243) |
| 19:51.20 | bmoraca_work | i'm thinking about a subway sandwich...or maybe quiznos |
| 19:51.23 | bmoraca_work | yes |
| 19:51.29 | [TK]D-Fender | carrar: i've got a place that does that... its awesome |
| 19:52.51 | Katty | HELLO THERE ALL YOU WONDERFUL PEOPLE |
| 19:54.17 | carrar | You've got a PLACE!!!! |
| 19:54.29 | carrar | PICS!!! |
| 19:55.06 | Katty | baroo? |
| 19:55.16 | Katty | oh |
| 19:57.00 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 19:59.55 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 20:01.19 | *** join/#asterisk smooth_penguin (~smoove@59.95.21.128) |
| 20:04.48 | *** join/#asterisk benngard (~benngard@90-230-92-67-no148.tbcn.telia.com) |
| 20:05.09 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 20:05.24 | *** join/#asterisk javierg (~javier@200.49.23.142) |
| 20:05.35 | javierg | Hello all |
| 20:05.46 | javierg | I need some help with asterisk config |
| 20:06.19 | rubberneck | How can I destroy a global variable after it has been set? |
| 20:06.44 | javierg | how can i increment the pre-rings of asterisk? |
| 20:06.55 | Qwell | what? |
| 20:07.46 | javierg | how can i increment the pre-rings of asterisk? Because i have an slow fxo and when i dial a number there is one ring and a time space, about 10 seconds of silence |
| 20:08.15 | Qwell | I have no idea what a "pre-ring" is.. |
| 20:08.18 | Katty | how can i change the lunar wane shaft in my asterisk box? |
| 20:08.18 | javierg | my fxo takes some time after link the call |
| 20:08.24 | Qwell | and there's no such thing as a "slow fxo" |
| 20:08.32 | javierg | i now |
| 20:08.41 | javierg | my fxo is really a router |
| 20:08.45 | javierg | with fxo targets |
| 20:08.52 | javierg | cisco 1750 |
| 20:09.38 | *** part/#asterisk MrTelephone (~test@h697179-171.picriverisp.net) |
| 20:09.42 | javierg | when i dial a number i hear one ring and ten seconds later the link is created and the call begins |
| 20:10.10 | javierg | in that lapse of time i need to delete the silence or replace that silence by rings |
| 20:10.17 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 20:10.47 | bmoraca_work | i think that "need" might be a strong word in this case |
| 20:11.02 | Naikrovek | anyone know an immigration lawyer who I could speak with? i suspect some unlawful things going on at my employer |
| 20:11.27 | Naikrovek | suppose i could check the yellow pages |
| 20:11.29 | bmoraca_work | javierg: i suspect the problem is with the dial-peer timeouts in your 1750. fix those, and you'll probably fix your problem. |
| 20:11.43 | javierg | dial-peer? |
| 20:11.44 | bmoraca_work | Naikrovek: human trafficking? |
| 20:11.47 | Qwell | Naikrovek: you probably don't want an immigration lawyer, but the labor board |
| 20:11.58 | Katty | Naikrovek: you need to contact ICE |
| 20:12.05 | Katty | Naikrovek: 18663472423 |
| 20:12.17 | Qwell | Katty: why do you know that number readily? |
| 20:12.21 | Naikrovek | bmoraca_work: no; i think my boss is taking bribes to get people employed here (getting them in the country) rather than hiring US citizens |
| 20:12.24 | Naikrovek | who are qualified |
| 20:12.31 | Katty | Qwell: long story, don't ask |
| 20:12.41 | Qwell | yeah, you certainly want ICE then |
| 20:12.49 | Naikrovek | thank you |
| 20:13.00 | Qwell | but... |
| 20:13.01 | kotp | Katty... may as well put that in your dial plan! |
| 20:13.12 | Qwell | beware of the possible legal ramifications against you |
| 20:13.14 | Naikrovek | what's ICE stand for |
| 20:13.15 | Katty | who says it's not |
| 20:13.19 | kotp | add that question to infobot here. |
| 20:13.23 | Qwell | Immigrations and Customs Enforcement |
| 20:13.23 | Katty | immigration and customs enforcement |
| 20:13.29 | Naikrovek | ah okay |
| 20:13.38 | Qwell | Naikrovek: You *MAY* want somebody to call on your behalf. |
| 20:13.39 | bmoraca_work | work visa fraud? sweet! |
| 20:13.42 | Katty | Qwell: there isn't legal rammifications to reporting suspicious activity |
| 20:13.54 | Katty | Naikrovek: i would fake your outgoing callerid |
| 20:13.54 | javierg | bmoraca_work: this is an dial-perr configuration that i have: dial-peer voice 18819 pots |
| 20:13.54 | javierg | <PROTECTED> |
| 20:13.55 | javierg | <PROTECTED> |
| 20:13.55 | javierg | <PROTECTED> |
| 20:13.59 | Katty | Naikrovek: ;P |
| 20:14.00 | kotp | Unless those it is a falsification... then you may have problems. |
| 20:14.01 | Naikrovek | hehe |
| 20:14.23 | Katty | Naikrovek: you can report anonymously |
| 20:14.32 | Naikrovek | my fear is that everyone not american here will get deported |
| 20:14.33 | kotp | right... (;wink) |
| 20:14.39 | javierg | i use 1881 prefix for diferenciate the rates |
| 20:14.40 | Katty | Naikrovek: i doubt it |
| 20:14.47 | Naikrovek | most of them are very talented and are indeed obeying the law and all of that |
| 20:14.50 | Katty | Naikrovek: they have to have pretty solid evidence to do anything |
| 20:15.06 | Naikrovek | but one guy is just doing whatever the hell he wants and getting all sorts of his relatives over here |
| 20:15.23 | Katty | about once a month ICE comes to cape |
| 20:15.30 | javierg | bmoraka: can i help me? |
| 20:15.31 | Katty | and goes through half the resturants here |
| 20:15.49 | Katty | they let things settle down again |
| 20:15.51 | bmoraca_work | shit, they need to do that in California |
| 20:15.53 | Katty | and next month, they do it again |
| 20:16.00 | Katty | same with all the bars |
| 20:16.08 | Katty | once a month the cops go in and clear it all out |
| 20:16.08 | bmoraca_work | half the population would be taking a trip south |
| 20:16.29 | Katty | yep, the resturants lose a lot of staff |
| 20:16.42 | Katty | which is a shame cause i really like mexican food :P |
| 20:17.19 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 20:17.25 | javierg | what can i do for fix this silence time between the call? |
| 20:18.12 | javierg | it is there configuration for asterisk that gives me the posibility to add more rings before calling? |
| 20:18.30 | javierg | or before linking? |
| 20:19.54 | p3nguin_ | Answer() |
| 20:20.27 | p3nguin_ | "core show application Answer" |
| 20:20.39 | Katty | musiccccccccc on holldddd |
| 20:20.48 | javierg | is there an expert on asterisk here? |
| 20:20.51 | Katty | no |
| 20:20.57 | Katty | sorry |
| 20:20.59 | Katty | please come again |
| 20:21.09 | Katty | </sarcasm> |
| 20:21.13 | Katty | ^- just in case you couldn't tell |
| 20:22.21 | javierg | katty... i need some help... can u help me? |
| 20:22.30 | Kobaz | help the computer |
| 20:22.34 | Kobaz | i'm a computer |
| 20:22.36 | Katty | javierg: yeah, just let me get my therapist clipboard and a white coat |
| 20:22.40 | Kobaz | stop all the downloading! |
| 20:23.30 | javierg | katty: very funny |
| 20:23.45 | carrar | more funny pls |
| 20:23.53 | leifmadsen | it's not friday... BACK TO WORK! |
| 20:24.04 | carrar | it's ALMOST |
| 20:24.11 | Kobaz | almolst? |
| 20:24.13 | p3nguin_ | ~ask |
| 20:24.14 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 20:24.19 | p3nguin_ | javierg: ^ |
| 20:26.08 | *** join/#asterisk ELBunce (~erik@kde/developer/bunce) |
| 20:26.49 | javierg | ok |
| 20:29.13 | raden_work | there a way to make SIP/101 ring 5 seconds then have SIP/102, SIP/103,SIP/104 start ringing for the next 15 seconds with SIP/101 |
| 20:29.31 | p3nguin_ | Two Dial() lines. |
| 20:30.30 | ManxPower-work | raden_work: people will tell you "dial twice" but that's not the best solution for many people |
| 20:30.44 | javierg | When i call through my fxo router, there is some time that my router takes for dial the number and make the link the call, arround 10 seconds. What can i do for fix this? Can i put false rings in that time? The router is a cisco 1750 |
| 20:31.04 | p3nguin_ | I was thinking something like: Dial(SIP/101,5) then Dial(SIP/101&SIP/102&SIP/103&SIP/104,15) |
| 20:31.34 | raden_work | Interesting, I will have to think on it |
| 20:31.36 | ManxPower-work | javierg: you don't mind if callers hear a ringing tone, and then a busy tone if they are calling a busy number? |
| 20:32.16 | ManxPower-work | 'cause that's exactly what you are asking for. |
| 20:32.17 | *** join/#asterisk smooth_penguin (~smoove@59.95.21.128) |
| 20:33.03 | carrar | raden_work, have your original dial branch out to serverall extensions with waits in front of them before ringing the later phones |
| 20:33.05 | bmoraca_work | raden_work: use your dial, but also build a local channel that will wait 5 seocnds before dialing the other three extensions. this will make it so that 101 doesn't get rung "twice" per each call. |
| 20:33.25 | javierg | ManxPower-work: you right... so the problem is in the router... but, what is the problem? |
| 20:33.41 | raden_work | bmoraca_work, carrar , thats exactly what i want todo |
| 20:33.48 | carrar | DO IT!! |
| 20:34.02 | carrar | Live the DREAM! |
| 20:34.16 | raden_work | Have to play with it a lil bit |
| 20:34.27 | bmoraca_work | javierg: it's a timeout issue, like I said. the default timeout for dial-peers is like 16 seconds or something. but, without a complete listing of all the dialpeers and a debug of the troubled call, i can't really help. |
| 20:34.40 | p3nguin_ | javierg: If the router is delaying the connection, how would you add ringing? |
| 20:34.57 | raden_work | javierg, I doubt its your 1750 |
| 20:35.07 | ManxPower-work | javierg: the problem is that it takes TIME to send those DTMF digits. |
| 20:35.30 | bmoraca_work | javierg: also, you have to remember that we're all here of our own accord and we may or may not have time or the will to answer your questions. many of us peek in here from time to time to answer questions and sometimes we don't feel up to diagnosing systemic issues |
| 20:35.56 | ManxPower-work | Assuming a 300ms per DTMF, that would be about 3.3 seconds of delay for a 11-digit call. that's not counting overhead like dialtone detection. |
| 20:36.44 | ManxPower-work | Assuming a 500ms DTMF, that's about 5 seconds of delay |
| 20:37.54 | javierg | mmm |
| 20:38.25 | javierg | an example of dial-peer is: dial-peer voice 17129 pots |
| 20:38.25 | javierg | <PROTECTED> |
| 20:38.25 | javierg | <PROTECTED> |
| 20:38.25 | javierg | <PROTECTED> |
| 20:39.19 | bmoraca_work | javierg: that doesn't help. we need the failed call posted on pastebin. |
| 20:40.59 | javierg | ok |
| 20:41.11 | bmoraca_work | ~pb |
| 20:41.12 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 20:41.49 | *** join/#asterisk smooth_penguin (~smoove@59.95.21.128) |
| 20:44.37 | javierg | here is my pb... please see it http://pastebin.com/ddbbf2d2 |
| 20:45.11 | javierg | thats my router configuration |
| 20:45.20 | bmoraca_work | i need your failed call log from Asterisk's CLI. |
| 20:46.40 | carrar | Should use MD5 for your router passwords |
| 20:46.45 | javierg | bmoraca_work: the call do not fail, remember that the problem is the delay generated by the router |
| 20:46.53 | carrar | and enable password-encryption |
| 20:47.13 | bmoraca_work | javierg: i'm aware of that. i still need the call log, though. |
| 20:47.42 | rubberneck | and change username xxxxxx privilege 15 password 0 xxxxxx, from 'password' to 'secret' |
| 20:48.01 | carrar | and close ip your VTY from the world |
| 20:48.03 | carrar | up |
| 20:48.15 | javierg | the passwords are modified obiously |
| 20:48.26 | javierg | i dont go to revelated |
| 20:48.30 | javierg | =) |
| 20:48.31 | carrar | your router is open to world |
| 20:48.34 | rubberneck | obviously but you should be using secrets instead of passwords. |
| 20:48.46 | javierg | of course |
| 20:48.52 | carrar | thats bad |
| 20:49.01 | bmoraca_work | carrar: no telnet password specified, so it won't accept telnet connections |
| 20:49.01 | Nugget | telnet is eeeeeeevil! |
| 20:49.12 | carrar | Connected to 23.49.200.volcan.bsr1000.papnet.cl. |
| 20:49.13 | carrar | Escape character is '^]'. |
| 20:49.13 | carrar | User Access Verification |
| 20:49.13 | carrar | Username: |
| 20:49.26 | carrar | You are using username |
| 20:49.49 | carrar | don't need a VTY password when using usernames |
| 20:49.49 | bmoraca_work | ahh, he's got aaa enabled. it'll default to local user database |
| 20:49.50 | javierg | yes |
| 20:50.30 | carrar | China has probably already logged into your router by now :) |
| 20:50.31 | p3nguin_ | And I was going to take over his network. |
| 20:50.47 | kotp | maybe even twice, huh carrar? |
| 20:50.50 | javierg | please friends... i need to fix my problem... i go to post in pb the asterisk log |
| 20:50.56 | carrar | maybe |
| 20:51.07 | javierg | jajaja |
| 20:51.09 | javierg | naah |
| 20:51.30 | javierg | take it easy buddys |
| 20:56.12 | javierg | ok |
| 20:56.45 | javierg | this is the log from CLI asterisk: http://pastebin.com/d1a299968 |
| 20:58.50 | javierg | Some idea? |
| 20:59.43 | mrbnet_ | Anyone setup mysql-cdr on debian lenny? Addons is asking asking for mysqlclient which is installed. Any other depends? |
| 21:00.35 | *** join/#asterisk Tigerplug292 (Tigerplug2@92.251.135.4.threembb.ie) |
| 21:00.40 | Tigerplug292 | hi guys |
| 21:00.49 | javierg | hey! |
| 21:01.36 | *** join/#asterisk Tigerplug292 (Tigerplug2@92.251.135.4.threembb.ie) |
| 21:02.06 | bmoraca_work | javierg: why are you using number translations? |
| 21:02.22 | javierg | because im using celulinks |
| 21:02.38 | Tigerplug292 | Hey all, I have a quick question... trying to justify Asterisk as a replacement for our old Nortel PBX. Im not a communications engineer so I understand its frustrating. I have googled and found some examples but I still have questions. |
| 21:02.44 | javierg | meaby the translation is the problem |
| 21:02.44 | dzup | javierg: wher to get 'a2billing' ? |
| 21:02.50 | bmoraca_work | javierg: what is "celulinks"? |
| 21:02.53 | dzup | s/whe/where/g |
| 21:02.58 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 21:03.52 | javierg | a celulink is a mobil cell phone implementation for calling to gsm networks |
| 21:05.28 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 21:05.39 | Katty | i wonder how much a pound of bacon cooks down into, weight wise. |
| 21:05.54 | Katty | 1lb of super lean ground beef cooks down to about 12 oz |
| 21:06.13 | Katty | and Pork Belly is a lot fattier. |
| 21:06.17 | Katty | more fat. |
| 21:06.20 | Katty | etc. |
| 21:07.11 | javierg | When i call through the router the ring tipe is different |
| 21:08.07 | *** join/#asterisk Tigerplug292 (~chatzilla@92.251.233.90.threembb.ie) |
| 21:08.16 | Tigerplug292 | user mode and to get things started you need to login using administrator credentials. For this, click on 'switch' link on the top left corner. A dialog box appears where you need to provide username as 'maint' and password as 'passw |
| 21:08.24 | Tigerplug292 | sorry! |
| 21:08.42 | ManxPower-work | Tigerplug292: Try asking on the correct channel. |
| 21:09.01 | Tigerplug292 | FFS |
| 21:09.06 | ManxPower-work | AST |
| 21:09.12 | *** join/#asterisk Tech_Travis (~tech_trav@208.179.137.131) |
| 21:09.29 | Tigerplug292 | Didn't know that was on my clipboard |
| 21:10.09 | carrar | Katty, please test |
| 21:10.11 | *** join/#asterisk Tech_Travis (~tech_trav@208.179.137.131) |
| 21:10.14 | carrar | then send me the bacon |
| 21:10.29 | Tigerplug292 | anyway, I've been tasked with deciding on a replacement for some ANCIENT nortel system at our office. About 100 - 150 extensions. We would have to upgrade our phones etc so I guess I'm wondering is asterisk worth a go. |
| 21:10.40 | Tigerplug292 | I've been looking at some case studies but still a little hazy on this |
| 21:11.01 | Tigerplug292 | and I really dont want to use CallManager |
| 21:11.01 | Tigerplug292 | because of $ |
| 21:11.13 | carrar | Tigerplug292, Asterisk from source |
| 21:11.19 | carrar | proceed now to install |
| 21:11.59 | Tigerplug292 | yes...I wont bore you with technical questions. But from what I read it is possible to integrate this with legacy pbx system that we have.... Im wondering is this a bad way to go though. |
| 21:12.04 | Tigerplug292 | I understand it depends on business needs etc. |
| 21:12.21 | carrar | put Asterisk between the old pbx and the phone company |
| 21:12.41 | ManxPower-work | Tigerplug292: how many simultanious calls to the PSTN do you want to support? do you want to run your calls over the unreliable internet or over a reliable (but more expensive) direct telco connection. |
| 21:13.25 | Tigerplug292 | carrar: for someone that hasn't worked with any voice technologies except basic CME (I'm mainly windows - don't laugh!) is this a "flaky" solution? |
| 21:13.33 | Tigerplug292 | we will have max 10 outgoing voice calls |
| 21:13.38 | carrar | Tigerplug292, not at all |
| 21:13.49 | ManxPower-work | Tigerplug292: where are you located? |
| 21:13.50 | Tigerplug292 | and we dont have carrier select lines so we are locked to one carrier |
| 21:13.54 | Tigerplug292 | because of the building cabling etc. |
| 21:13.56 | Tigerplug292 | Im in Ireland |
| 21:14.15 | Tigerplug292 | I've considered 3CX also but have doubts about how stable it is |
| 21:14.56 | carrar | Asterisk can easliy handle 200 extensions |
| 21:15.39 | Tigerplug292 | talking of getting a box to run calls destined for "cell" out through another system (provided but cell provider) that I'm not familiar with... |
| 21:16.04 | Tigerplug292 | Okay so I need to go google hardware requirements for such a system I guess |
| 21:16.22 | Tigerplug292 | problem is I don't want to get stuck on this... thats my only doubt because I haven't done it before |
| 21:16.37 | ManxPower-work | Tigerplug292: oh you won't have a good idea of the hardware requirements until you know everything else. |
| 21:16.42 | carrar | Well you are gonna have to learn a lot, but thats the fun of it all |
| 21:17.10 | Tigerplug292 | I don't mind that... I'm use to alot of study (mainly Cisco R&S)... all this though... really new to me |
| 21:17.28 | ManxPower-work | Tigerplug292: Remember, Asterisk isn't really a PBX. It is more of a PBX toolkit that lets you build a PBX, like a compiler and libraries let you build an application |
| 21:17.32 | kotp | ~book |
| 21:17.32 | 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 |
| 21:17.54 | Tigerplug292 | kotp: downloading.. thanks |
| 21:18.28 | kotp | yw |
| 21:18.55 | kotp | The favor was afforded me this morning, so I figured I saw you saying you have a lot of studying... |
| 21:19.03 | kotp | returning the favor... pass it forward! |
| 21:19.32 | Tigerplug292 | Would freepbx with asterisk be considered a bad solution in enterprise. I guess I want to have things in such a way that if someone new walked into the job they won't have to do as much study as me for basic admin |
| 21:20.02 | *** join/#asterisk xmitter (~xmitter@72-62-113-107.pools.spcsdns.net) |
| 21:20.08 | Tigerplug292 | I have messed with asterisk in the past but only on vmware..... I figure just having config files with no GUI could be a problem when presenting to management |
| 21:20.32 | *** join/#asterisk devyll (~paul@213.233.93.221) |
| 21:21.03 | devyll | how can I recieve calls from Skype into my asterisk box ? |
| 21:21.28 | ManxPower-work | ~freepbx |
| 21:21.29 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 21:21.32 | devyll | I just want to be able to receive calls. I'm not interested in placing outgoing calls to skype or to land lines via skype |
| 21:22.15 | [TK]D-Fender | ~skypeforasterisk |
| 21:22.16 | infobot | skypeforasterisk is probably a Digium-made channel driver that allows Asterisk to connect to the Skype network using a Skype-supported connection method. Unlike all other solutions it does not require X11, kernel modules, or a Windows machine. See http://www.digium.com/skype for details |
| 21:22.18 | [TK]D-Fender | ^^^^^^^ |
| 21:22.26 | Tigerplug292 | ManxPower-work: okay... can you give me an example of a production system you have with asterisk? I'd appreciate it |
| 21:22.58 | ManxPower-work | Tigerplug292: A production system I have with Asterisk? |
| 21:23.22 | [TK]D-Fender | Tigerplug292: What does that even mean? |
| 21:23.40 | Tigerplug292 | yes, if you have one .... not specifics. I guess capacity reliability etc. |
| 21:23.53 | carrar | Tigerplug292, look at SwitchVox if you want someone to take it over without learning |
| 21:23.54 | adnc | what is the difference if there is a number behind EXTEN like ${EXTEN:4} ? |
| 21:24.03 | carrar | http://www.switchvox.com/catalog/smb_bundles.php |
| 21:24.21 | p3nguin_ | adnc: That would strip off the first 4 numbers in the ${EXTEN} number. |
| 21:24.28 | adnc | ahh |
| 21:24.31 | adnc | thanks |
| 21:24.32 | ManxPower-work | Tigerplug292: I just manage PBXs I don't how many calls they take. |
| 21:24.39 | Tigerplug292 | and... I'm assuming centOS is the best way to go |
| 21:24.40 | carrar | hahaha |
| 21:24.50 | Tigerplug292 | ManxPower-work: okay - thanks anyway |
| 21:24.52 | carrar | whatever OS you like |
| 21:24.58 | carrar | centos works great however |
| 21:25.13 | Tigerplug292 | carrar: see last time I tried with Ubuntu and failed miserably! |
| 21:25.24 | carrar | what failed? |
| 21:25.49 | carrar | Your patience? |
| 21:25.55 | Tigerplug292 | carrar: my attempts to do anything useful with it. But at the time it was only for fun. |
| 21:25.58 | [TK]D-Fender | CORRECT |
| 21:26.21 | Tigerplug292 | carrar: no, not my patience. I didn't have any specific requirement at the time so I had no structured approach if you like. |
| 21:26.27 | carrar | well you'll need to have patience coming new into unix from windows |
| 21:26.46 | Tigerplug292 | carrar: but now is time to consider a new phone system and I'd like to give asterisk a look |
| 21:27.03 | carrar | look at switchvox also |
| 21:27.06 | Tigerplug292 | carrar: well I work with Ubuntu alot for system monitoring / management |
| 21:27.10 | carrar | it's asterisk and digium supported |
| 21:27.22 | Tigerplug292 | cool, will check it out now |
| 21:27.34 | carrar | or if you want to take unix on asterisk from source |
| 21:27.43 | carrar | AFS!! |
| 21:27.49 | ManxPower-work | Switchvox gives you none of the advantages of running Asterisk and yet gives you all the drawbacks of using a GUI. |
| 21:27.57 | ManxPower-work | IT'S THE PERFECT SOLUTION |
| 21:28.11 | carrar | Switchvox works great for a lot of clients |
| 21:28.33 | carrar | just depends on the needs of the client |
| 21:29.00 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 21:29.03 | [TK]D-Fender | carrar: Each solution tends to work great for a the people using them. |
| 21:29.07 | carrar | heh |
| 21:29.12 | carrar | that is amazing how that works |
| 21:29.20 | [TK]D-Fender | carrar: indeed |
| 21:29.34 | Tigerplug292 | ManxPower-work: so most of your management is editing config files etc.. e.g. if you need to make changes. Do you find this quicker? |
| 21:29.38 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 21:29.39 | [TK]D-Fender | checkout time... later all |
| 21:29.52 | carrar | Tigerplug292, command line is always faster! |
| 21:30.10 | bmoraca_work | Tigerplug292: the idea is that once you set it up, you shouldn't really have to make many changes to it. |
| 21:30.12 | carrar | and write scripts if it isn't |
| 21:30.14 | Tigerplug292 | carrar: if you know what your doing of course :-) |
| 21:30.22 | Tigerplug292 | bmoraca_work: well thats what I'd like! |
| 21:30.34 | ManxPower-work | Tigerplug292: do you think "joe sip.conf" and editing a few lines is faster than stumbling around a GUI? |
| 21:30.39 | kotp | and if you only have to make changes because of users coming and out, then you can use awk, sed or sisters. |
| 21:30.51 | bmoraca_work | Tigerplug292: that pretty much invalidates the "speed" increase of using a GUI. |
| 21:31.08 | Tigerplug292 | I don't even know what sort of hardware I should be looking for to interface with the old system. I know I need a card of some sort but I haven't a clue. Reading the asterisk site now |
| 21:31.19 | kotp | cli is almost always more powerful than deciding with mouse click to do on a limited menu... |
| 21:31.22 | carrar | read the asterisk book |
| 21:31.34 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 21:31.35 | kotp | Which book? |
| 21:31.38 | kotp | This book? |
| 21:31.39 | kotp | ~ |
| 21:31.40 | Tigerplug292 | point taken, coming from a cisco background I know how fast command line can be, I'm just curious about asterisk thats all |
| 21:31.44 | kotp | ~book |
| 21:31.44 | 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 |
| 21:31.58 | bmoraca_work | Tigerplug292: i'll build you a system for $150K...it'll do whatever you want it to do! |
| 21:31.58 | *** join/#asterisk lanning (~lanning@208.87.235.224) |
| 21:32.04 | Tigerplug292 | okay, I'll go read it and then come back in a couple of days with my questions :-P |
| 21:32.17 | carrar | Tigerplug292, asterisk cli has command completion too :) |
| 21:32.21 | Tigerplug292 | bmoraca_work: when can you start? |
| 21:32.21 | Tigerplug292 | ;) |
| 21:32.30 | kotp | Most of all Tigerplug292 have fun. |
| 21:32.36 | Tigerplug292 | carrar: now thats the sort of stuff I like! |
| 21:33.15 | Tigerplug292 | kotp: I intend on doing so... just so much stuff at the moment I'm studying. strapped for time :-( |
| 21:33.29 | ManxPower-work | Tigerplug292: then you will fail. |
| 21:33.41 | ManxPower-work | Learning Asterisk is not a casual thing. It takes time and commitment and money. |
| 21:34.03 | Tigerplug292 | ManxPower-work: I don't believe so. I'm strapped for time per day/week etc... It just takes a little longer (i.e commitment) |
| 21:34.11 | carrar | ... and patience |
| 21:34.29 | Tigerplug292 | ManxPower-work: well... I have no money so I need to convince the company to get me some stuff for testing. |
| 21:34.48 | bmoraca_work | the ability to learn asterisk is relative...if you're SMRT, you can learn it casually/easily. if you're a dumb dumb, you won't learn it regardless of how much time you put into it...just look at p3nguin_ :P |
| 21:35.31 | p3nguin_ | is not amused |
| 21:35.34 | Tigerplug292 | as for phones... if I was to go SIP instead of integrating with existing system is there a simple phone that any of you guys recommend? I've looked into cisco (not happening!) and grandstream phones.... |
| 21:35.40 | Tigerplug292 | lol |
| 21:35.52 | carrar | Polycom |
| 21:35.56 | bmoraca_work | was only kidding |
| 21:35.59 | ManxPower-work | ~phones |
| 21:36.00 | infobot | While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Linksys SPA-9XX, Snom, Cisco 79XX, everything else. Do not consider Grandstream phones. Ever. |
| 21:36.19 | Tigerplug292 | just something to give to the sales guys... the less features the better... they break everything |
| 21:36.40 | raden_work | Is there a reason that Mysql tables need to be myisam and not innodb |
| 21:37.03 | Tigerplug292 | I've used Linksys SPA-9XX in the past with 3CX I believe and they feel so cheap... I understand they are cheap but they feel so.... fake |
| 21:37.21 | bmoraca_work | EEEEWWWWWWWW...3cx |
| 21:37.46 | ManxPower-work | Tigerplug292: I'm sure we are happy for you, but I doubt most people here have ever even heard of 3CX |
| 21:39.00 | bmoraca_work | raden_work: http://tag1consulting.com/MySQL_Engines_MyISAM_vs_InnoDB |
| 21:39.12 | Tigerplug292 | ManxPower-work: well.. I think its worth a look because it is community driven etc. But I think its not worth using because its on Windows. Its good to be aware of whats out there I guess (not that I can talk) |
| 21:39.54 | raden_work | bmoraca_work, I understand the differences Im just wondering why myisam is always used when Innodb seems more logical |
| 21:40.11 | carrar | raden, use postgres |
| 21:40.15 | bmoraca_work | raden_work: couldn't tell you, honestly. |
| 21:40.34 | ManxPower-work | Tigerplug292: I think the issue is that apparently they think running a PBX on windows is a good idea. |
| 21:40.43 | ManxPower-work | It's not. It never will be. |
| 21:40.50 | Tigerplug292 | bmoraca_work: lol |
| 21:41.18 | Tigerplug292 | ManxPower-work: even I know that :_ |
| 21:41.26 | Tigerplug292 | :) |
| 21:42.17 | raden_work | why does everyone in mysql keep telling me how horrid realtime is :(* |
| 21:42.22 | *** join/#asterisk smooth_penguin (~smoove@59.95.21.128) |
| 21:42.23 | Tigerplug292 | hardware suggestions that I can purchase myself to mess about with this at home? |
| 21:42.37 | *** join/#asterisk Dibri (~gavit@190.98.32.116) |
| 21:42.53 | ManxPower-work | Tigerplug292: ANY x86 System. |
| 21:43.17 | ManxPower-work | you're not running the federal reserve, you are going to be doing a couple of phone calls. Hell, a linksys router can do that.\ |
| 21:43.59 | carrar | hell!! |
| 21:47.28 | raden_work | ManxPower-work, so i presume you have realtime experiance ? |
| 21:48.15 | ManxPower-work | raden_work: I have experience over the course of 5 years watching poor sods here trying to use Realtime |
| 21:48.39 | raden_work | ManxPower-work, Ill get it to work properly :) and Ill document it :) |
| 21:49.06 | raden_work | ManxPower-work, you are correct more i read more challeneges I have to overcome |
| 21:49.57 | *** join/#asterisk titter` (~titter@c-98-208-158-125.hsd1.fl.comcast.net) |
| 21:55.57 | *** join/#asterisk grey-monkey (~grey-monk@75-148-103-190-Utah.hfc.comcastbusiness.net) |
| 21:57.39 | sbrath | what's the syntax for passing in 2 extensions to stdexten macro " exten = 4357,1,Macro(stdexten,4107,4107&4117)" ?? |
| 21:57.57 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:58.09 | sbrath | ARG1 is VMbox, that is right, but ARG2 is Extesions(s) |
| 22:01.00 | *** join/#asterisk war9407 (war@liquidswords.org) |
| 22:04.31 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 22:10.47 | raden_work | bmoraca_work, how on earth can i make it so calls from a phone on server a to a call on server b would route between the 2 asterisk servers ? |
| 22:12.42 | bmoraca_work | raden_work: few ways to do that... |
| 22:13.47 | bmoraca_work | raden_work: you can use DUNDi or you can look up the number in your database to determine whether or not it exists within your network and then send it to whatever peer it should be associated with |
| 22:15.52 | bmoraca_work | raden_work: I do the latter any time any of my on-net customers dial a phone number. |
| 22:16.05 | bmoraca_work | DUNDI basically does the same thing, however. |
| 22:16.34 | *** part/#asterisk jaybinks (~jaybinks@ppp219-21.static.internode.on.net) |
| 22:20.21 | leifmadsen | DUNDI is awesome :) |
| 22:21.12 | bmoraca_work | leifmadsen: when did regexten get added in for DUNDI? i don't recall seeing that when I last did my dundi testing about 6 months ago... |
| 22:21.16 | leifmadsen | sbrath: well there isn't really any format for that... you could create your Macro() to be clever by generating the string for you with ARG3+n |
| 22:21.21 | Katty | watches 20000 leagues under the sea |
| 22:21.36 | leifmadsen | bmoraca_work: regexten? I can't remember... can you elaborate what that does? |
| 22:21.37 | *** join/#asterisk xmitter (~xmitter@c-24-21-213-242.hsd1.or.comcast.net) |
| 22:21.48 | Katty | i wish someone would host an Ocean Cam |
| 22:21.58 | Katty | or maybe Reef Cam |
| 22:22.09 | leifmadsen | Katty: actually... |
| 22:22.55 | Katty | leifmadsen: oh? |
| 22:23.10 | leifmadsen | Katty: I'm trying to find it, but a University in Vancouver is doing just that |
| 22:23.26 | Katty | :>>>> |
| 22:23.36 | leifmadsen | Katty: I think it's this: http://www.venus.uvic.ca/discover-venus/what-is-venus |
| 22:25.10 | bmoraca_work | leifmadsen: it works with regcontext in that it places the NoOp extensions in a specific context, but it allows you to have multiple extensions per peer automatically created in that context |
| 22:25.25 | leifmadsen | oh ya -- regexten has been around a long time |
| 22:25.29 | Katty | leifmadsen: yeah buti can't watch it |
| 22:25.31 | leifmadsen | like since 1.4 I think |
| 22:25.36 | Katty | leifmadsen: :< |
| 22:25.39 | leifmadsen | Katty: there is a way to do it -- you have to sign up somewhere |
| 22:25.47 | leifmadsen | it's been a while since I've looked at it |
| 22:25.57 | leifmadsen | (I agree, it wasn't the most intuitive website) |
| 22:26.16 | javierg | exit |
| 22:26.22 | Katty | hmm |
| 22:27.01 | leifmadsen | ok, dinner! |
| 22:27.10 | Katty | ooooh |
| 22:27.20 | Katty | 20k leagues under the sea has a narwhall :> |
| 22:28.26 | *** join/#asterisk came0 (~came0@74-95-115-29-Colorado.hfc.comcastbusiness.net) |
| 22:28.31 | Katty | and the girl who's in it, also played in Dune |
| 22:29.12 | *** join/#asterisk fifer (~fifer@67.208.108.228) |
| 22:29.34 | came0 | hm I just pluged my polycom 335 in at my hotel room and every number is ringing busy! i think these buggers blocked the port! any idea how to get around it? |
| 22:31.09 | javierg | quit |
| 22:32.03 | bmoraca_work | came0: you probably cannot. |
| 22:32.26 | came0 | bmoraca_work: stfu! seriously? I cant jsut use a different port? |
| 22:32.43 | bmoraca_work | came0: they're probably blocking it at layer 7 which is port agnostic |
| 22:34.21 | came0 | well.... w.... t.... f..... |
| 22:34.37 | came0 | how can I check if its blocked? |
| 22:34.42 | came0 | it looks like it registers fine |
| 22:34.44 | came0 | hm |
| 22:35.45 | Katty | came0: do you have a laptop |
| 22:35.51 | came0 | yeah |
| 22:35.56 | Katty | put a software phone on it |
| 22:36.03 | Katty | you should be able to see what the software is doing |
| 22:36.06 | came0 | its pluged into the other jack on the phone |
| 22:36.06 | came0 | and it connects fine |
| 22:36.07 | Katty | or not doing |
| 22:36.17 | came0 | ah ok |
| 22:36.20 | came0 | let me try my x-lite |
| 22:36.33 | *** join/#asterisk cidu (~PissInMyA@whthyt253-70.northwestel.net) |
| 22:39.57 | cidu | ok, so i was trying to get asterisk to send voicemail notifications straight to a smsc via smpp, but couldnt find anything of intrest there, so i moved on, and got kannel (oss smsc) up and running, kannel can accept sms's via a http request, so what i want tpo do is get asterisk to send out voicemail notifications (pager) as a http string rather than an email...... any ideas? im guessing im going to have to go digging through the voicemai |
| 22:39.57 | cidu | l app, but i thought id check here incase one of you guys knew of something already working out there |
| 22:40.11 | *** part/#asterisk grey-monkey (~grey-monk@75-148-103-190-Utah.hfc.comcastbusiness.net) |
| 22:41.02 | cidu | or possibly with mailcmd , but havent gotten too far there either |
| 22:41.20 | came0 | x-lite is ringing!!!! |
| 22:42.15 | came0 | so can my polycom just be blocked? |
| 22:42.19 | came0 | but not the softphone? |
| 22:42.23 | came0 | seems funny |
| 22:42.30 | came0 | Katty:^ |
| 22:42.51 | cidu | i wasnt here for your question, your getting a softphone to ring with the same account credentials that wont ring on your polycom? |
| 22:43.00 | came0 | yes |
| 22:43.01 | came0 | at a hotel |
| 22:43.25 | came0 | a nice hotel...i'm going to be mad if these buggers blocked the ports on purpose |
| 22:43.28 | bmoraca_work | came0: on someone else's network, they can do whatever they want. |
| 22:44.05 | came0 | sure they can and I can go down the street to a different hotel as well... so |
| 22:44.44 | cidu | well, why would the softphone be talking on different ports and your sipphone? |
| 22:45.29 | cidu | and=than |
| 22:45.29 | *** join/#asterisk raden_work (~jon@69-179-99-17.stat.centurytel.net) |
| 22:45.43 | came0 | i think it may be a DNS issues |
| 22:45.44 | came0 | issue |
| 22:45.57 | came0 | i'm going to try different settings... ip addresses and whatnot |
| 22:46.02 | cidu | so use ip's? |
| 22:46.22 | cidu | or point your softphone to an public dns server |
| 22:46.27 | cidu | err, your polycom |
| 22:46.30 | came0 | idk i havent even looked whats in there yet.... is it typically ips? |
| 22:46.37 | came0 | yeah yeah... good idea |
| 22:46.39 | cidu | googles public dns is easiest to remember, 8.8.8.8 |
| 22:46.53 | came0 | so google can know a little bit more about me :P |
| 22:46.55 | rubberneck | Im trying to dial out to a SIP trunk provider with the dial() app, is there a way in the dialplan to detect if a fax answers the outbound call? I dont want to send a fax, I want to know I reached a fax machine and remove it from our # to dial list. |
| 22:48.02 | p3nguin_ | Verizon's are easy to remember, too: 4.2.2.1-6 |
| 22:48.11 | cidu | those are public? |
| 22:48.15 | p3nguin_ | yes |
| 22:48.21 | cidu | nice, another to add to the collection :) |
| 22:48.21 | bmoraca_work | 8.8.8.8 and 8.8.4.4 are google's |
| 22:48.38 | bmoraca_work | i've found that 4.2.2.1's caching is a bit wonky sometimes |
| 22:48.52 | bmoraca_work | i just use my own DNS servers |
| 22:49.14 | p3nguin_ | OpenDNS's addresses aren't too terribly bad to remember, either. |
| 22:49.26 | cidu | 208.67.222.222 and 220.220 |
| 22:49.27 | cidu | yeah |
| 22:49.33 | cidu | i stuck those in my brain a while back |
| 22:49.34 | p3nguin_ | Although they do some odd filtering. |
| 22:49.39 | cidu | aye |
| 22:50.01 | came0 | i switched to google from opendns awhile ago and it seems to work better |
| 22:50.12 | came0 | also i dont like opendns hijacking my 404 |
| 22:50.41 | Qwell | came0: they aren't hijacking 404s |
| 22:50.50 | cidu | i waffle back and forth, but yes...so anybody got an idea how to make asterisk pass voicemail pager notification as an http string instead of an email? |
| 22:51.09 | came0 | Qwell: we'll i see their search page when a page doesnt exist... ? |
| 22:51.11 | Qwell | came0: what they're hijacking is "invalid" domains |
| 22:51.21 | came0 | ah yes you are right |
| 22:51.30 | Qwell | very very different. one is far more malicious |
| 22:52.54 | bmoraca_work | we use a purgatory DNS server that redirects ANY domain name lookup to a page that reminds the customer to PAY THEIR DAMN BILL |
| 22:53.22 | cidu | or, on an alternative track, is there a smpp plug for asterisk? google turned up nothing on that front |
| 22:54.35 | bmoraca_work | cidu: i've never heard anything good about anybody ever trying to use SMS from asterisk. i think the general concensus is to use an external gateway webpage that has an API you can call from an AGI or other script |
| 22:55.43 | cidu | ahh, okie, i just keyed into another line of thought here, if i can mod the mailcmd, and the pager mail structure i can push to my kannel gateway via http, if it works ill post on it somewhere.... |
| 22:57.24 | bmoraca_work | there's a setting that will let you call a script when the system receives a voicemail...probably easiest to do that |
| 22:57.46 | *** part/#asterisk RobH (~robh@cpe-173-169-30-118.tampabay.res.rr.com) |
| 22:57.57 | *** join/#asterisk RobH (~robh@cpe-173-169-30-118.tampabay.res.rr.com) |
| 22:58.06 | bmoraca_work | cidu: externnotify |
| 22:58.09 | cidu | hmm, you happen to know if its possible to have different mailcmd setting for pager vs email notifications? |
| 22:58.13 | cidu | ohh, thanks :) |
| 23:02.09 | *** join/#asterisk javierg (~javier@200.49.23.142) |
| 23:02.53 | came0 | damn these things take forever to reboot |
| 23:03.37 | *** join/#asterisk javierg (~javier@200.49.23.142) |
| 23:04.01 | javierg | Can i hold on the asterisk rings in case the destination is far away? |
| 23:13.05 | *** join/#asterisk Kailtor (~Kailtor@206.205.228.222) |
| 23:14.45 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 23:22.48 | nickaugust | ok so I changed the DNS servers to openDNS and it still wont register... i dont understand why the sotphone can register but the polycom cant/ |
| 23:22.49 | nickaugust | ? |
| 23:23.02 | nickaugust | i even switched the phone to use the port that the softphone connected on |
| 23:23.50 | nickaugust | my time and date keeps flashing on my phone does that mean it cant register? |
| 23:27.09 | leifmadsen | no, that means the phone can't connect the NTP server |
| 23:29.18 | bmoraca_work | nickaugust: i've noticed that polycom phones sometimes don't pick up the gateway through DHCP. try checking that. |
| 23:29.49 | nickaugust | bmoraca_work: we'll im getting a local IP address... how can I check the gateway? |
| 23:29.56 | nickaugust | just set it manually? |
| 23:30.01 | nickaugust | to what my laptop gets |
| 23:30.03 | nickaugust | ? |
| 23:30.09 | bmoraca_work | in the ethernet status menu, it will tell you what the phone is getting. |
| 23:30.28 | nickaugust | sorry to be typing on so many lines.. i hate when people do that.. im not sure why ive been doing it now.. just frustrated i guess |
| 23:30.52 | nickaugust | bmoraca_work: Ok i'll try that when the phone reboots... thanks |
| 23:31.38 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 23:33.11 | idespinner | nickaugust, what phone firmware are you using? |
| 23:33.16 | *** join/#asterisk Deeewayne (~dwayne@75.76.254.162) |
| 23:33.16 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 23:33.42 | nickaugust | idespinner: Rev: 3.2.1.0078 |
| 23:33.47 | nickaugust | is that the firmware version? |
| 23:35.44 | *** join/#asterisk hfb (~hfb@pool-96-247-114-78.lsanca.dsl-w.verizon.net) |
| 23:39.50 | *** join/#asterisk xmitter (~xmitter@c-24-21-213-242.hsd1.or.comcast.net) |
| 23:49.31 | Katty | GUESS WHAT |
| 23:50.19 | bmoraca_work | CHICKEN BUTT! |
| 23:50.24 | nickaugust | ok i'm just realizing that its probably the frickin web interface that the hotel makes you login to! |
| 23:50.25 | Katty | :< |
| 23:50.36 | nickaugust | the phone obvioulsy hasnt agreed to the terms of service or whatever |
| 23:50.53 | Katty | use a software client |
| 23:50.55 | bmoraca_work | nickaugust: depends on how that page works, but it's a good suspision |
| 23:51.05 | nickaugust | Katty: the softphone works |
| 23:51.09 | nickaugust | GRRRRRR!!! |
| 23:51.10 | Katty | cheers. |
| 23:51.13 | Katty | problem solved |
| 23:51.27 | nickaugust | maybe I can just switch to the laptops ip address? |
| 23:51.37 | nickaugust | or do you think its by MAC |
| 23:51.49 | Katty | one way to find out |
| 23:51.59 | nickaugust | ah... well atleast it makes sense now! |
| 23:52.00 | bmoraca_work | nickaugust: you realize that the hotel makes a LOT of money from the phone calls guests make out of their rooms, right? they've probably done a lot to make sure that guests can't plug their own phones in |
| 23:52.01 | nickaugust | let me try that |
| 23:52.19 | Katty | bmoraca_work: yeah but hotel people aren't IT folk ;) |
| 23:52.30 | Katty | bmoraca_work: which means they forgot one very important detail. |
| 23:52.32 | bmoraca_work | Katty: depends on the hotel and the size of the hotel. |
| 23:52.41 | Katty | bmoraca_work: an IT person on a mission.... |
| 23:52.50 | Katty | bmoraca_work: is a very crafty person indeed |
| 23:53.14 | nickaugust | bmoraca_work: yes I understand that... but i need to take my phone system with me for business... and im not about to let this fricken hotel hinder my ability to travel and still run my business |
| 23:53.26 | Katty | ^- IT person on a mission |
| 23:53.40 | nickaugust | Katty: :) |
| 23:54.33 | Katty | i would already have the phone setup and booted. |
| 23:54.46 | Katty | then use the laptop to make the change and plugin in the phone like ASAP |
| 23:55.23 | nickaugust | hopefully its just a DCHP lease... but i doubt it |
| 23:55.31 | Kailtor | Anyone here willing to help me with a FreePBX question real quick? I just need help in getting FreePBX to see my AEX2400 board...drivers are loaded already |
| 23:55.37 | Katty | we dpm |
| 23:55.43 | Katty | we don't support freepbx here. |
| 23:55.48 | Katty | you might try #freepbx |
| 23:55.53 | Kailtor | yeah they're asleep |
| 23:56.03 | bmoraca_work | Kailtor: freepbx doesn't know or care about DAHDI. you need to configure your board by itself. |
| 23:56.08 | Katty | how does that change the fact we don't support freepbx. |
| 23:56.09 | Kailtor | all 82 of them |
| 23:56.20 | Kailtor | welp i thought maybe someone would have some insight :) |
| 23:56.37 | Kailtor | okay so i need to setup config files on my own and not through freepbx? |
| 23:57.01 | Katty | bmoraca_work: this is why the freepbx users keep showing up in the wrong channel. |
| 23:57.11 | *** topic/#asterisk by leifmadsen -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.4, 1.6.1.16, 1.6.0.24 (2010/02/18), 1.4.29.1 (2010/02/18), *-Addons 1.6.2.0 (2009/12/18), 1.6.1.2, 1.6.0.4, 1.4.10 (2009/12/02), dahdi-linux 2.2.1 + dahdi-tools 2.2.1 (2010/01/20), Libpri 1.4.10.2 (2009/10/20) -=- Related channels: #asterisknow,#switchvox,#freepbx,#asterisk-dev,#asterisk-bugs |
| 23:57.25 | Katty | it's because no one reads the topic anymore. |
| 23:57.29 | Katty | isn't it. |
| 23:57.37 | bmoraca_work | everyone always yells at me when i offer a basic answer about freepbx, lol |
| 23:57.42 | carrar | I READ DER TOPIKS |
| 23:57.46 | Katty | bmoraca_work: yes, and for good reason :P |
| 23:57.51 | nickaugust | Ok... i'm going to spoof the mac address of my phone with my laptop and then login to the network and switch over to the phone.... |
| 23:57.52 | Katty | bmoraca_work: we are tired of them going to the wrong channel :P |
| 23:57.56 | Kailtor | look i read the topic, i am just trying to get some help |
| 23:58.02 | Katty | ^- IT person on a mission |
| 23:58.03 | Kailtor | no one is talking in the other channel but people are chatting here |
| 23:58.05 | leifmadsen | Asterisk 1.2.40, 1.4.29.1, 1.6.0.24, 1.6.1.16, and 1.6.2.4 are now available. For more information about this release, please see the announcement at: http://www.asterisk.org/node/49906 |
| 23:58.06 | Kailtor | no need to be rude my friends |
| 23:58.08 | carrar | only the part of the topic that doesn't run off the screen cause it's soo large |
| 23:58.23 | Katty | in soviet russia |
| 23:58.27 | *** join/#asterisk bmoraca (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 23:58.29 | Katty | DER TOPIKS READS YOU |
| 23:58.32 | carrar | heh |
| 23:58.35 | nickaugust | Kailtor: I would have just asked the question before mentioning freepbx... are you sure its freepbx related even? |
| 23:58.39 | bmoraca | blarg |
| 23:59.04 | bmoraca | Kailtor: there are litterally hundreds of walkthroughs online about configuring dahdi and freepbx...seriously... |
| 23:59.23 | Kailtor | hundreds! man my googlefu must suck then huh |
| 23:59.26 | leifmadsen | hundreds? literally? I want a list! |