| 00:01.15 | dandate2 | i just have a strange feeling that some ukranian is now hustling my ip:socks port to craigslist and email spammers world wide |
| 00:02.53 | drmessano | You really have no idea what you're doing, do you? |
| 00:03.51 | dandate2 | bah i didn't make the website |
| 00:04.05 | dandate2 | someone in a forum just told me it was hacked so i looked at it and bypassed the security exception |
| 00:05.06 | dandate2 | but now thinking mabye i should reformat hd for doing so? heh |
| 00:05.59 | *** join/#asterisk sjobeck (n=Adium@pool-98-108-149-215.ptldor.fios.verizon.net) |
| 00:10.06 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 00:10.44 | *** part/#asterisk sjobeck (n=Adium@pool-98-108-149-215.ptldor.fios.verizon.net) |
| 00:17.00 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
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| 00:26.55 | *** join/#asterisk JT (n=j@unaffiliated/jt) |
| 00:27.38 | manxpower | sounds to me like you need to be on a different channel |
| 00:30.12 | *** join/#asterisk coppice (n=chatzill@26.168.17.210.dyn.pacific.net.hk) |
| 00:34.29 | *** join/#asterisk ltd_wk (i=z@patwk.transact.net.au) |
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| 00:55.35 | *** join/#asterisk matt_d (n=matt@70.134.79.103) |
| 00:55.45 | matt_d | Hello everyone. |
| 00:56.52 | matt_d | I haven't developed on Asterisk for a while and am back at it. Is ChanSpy broken by design? It works for listening to a channel, but when I add the 'w' tag or w~hispering it doesn't work. no audo -- er r audio doesnt go to the spied on channel. |
| 00:57.07 | matt_d | I have searched google all over to find similiar problem, but dont see one. |
| 00:59.12 | *** join/#asterisk matt_d (n=matt@70.134.79.103) |
| 01:02.28 | dandate2 | yeah chanspy is hella broke |
| 01:02.34 | dandate2 | i had the same problem |
| 01:02.41 | dandate2 | if u want that to work u need to go with elastix or something |
| 01:02.57 | matt_d | are you serious? crap.. hehe im on a time frame for this project to be done! |
| 01:03.42 | matt_d | elasstix eh? hmm.. easty to learn? |
| 01:07.54 | matt_d | or any ideas on how to do something similiar? |
| 01:12.37 | *** join/#asterisk SkywaIker (i=pirch@113.53.160.68) |
| 01:15.44 | matt_d | Elastix is Asterisk, did tehey make a better chanspy app? |
| 01:23.18 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 01:24.40 | *** join/#asterisk levity (i=canuck@unaffiliated/canuck) |
| 01:34.11 | manxpower | matt_d: What specific version of Asterisk are you having problems with? |
| 01:37.30 | matt_d | manxpower: 1.4.26 |
| 01:37.38 | matt_d | sorry about the delay to your question. :) |
| 01:37.58 | *** join/#asterisk TimRiker (n=timr@bzflag/projectlead/TimRiker) |
| 01:39.57 | *** join/#asterisk drclue (n=drclue@ip-65-49-160-6.wireless.dyn.beamspeed.net) |
| 01:41.42 | *** join/#asterisk voxter (n=voxter@216.53.198.130) |
| 01:42.21 | manxpower | Have you tried the 1.6 branch? |
| 01:43.22 | matt_d | manxpower no, i have not. that is an option though. will i seek success there? :) |
| 01:43.35 | matt_d | or is it pretty wide excepted that chanspy is broken? |
| 01:45.12 | matt_d | hehe my jaw dropped when i couldnt get this function to work. i have to have it done by friday. hehe |
| 01:49.29 | drclue | I was playing with channel spy last week on my 1.6 and it seemed to work. Not something I was focusing on , but rather something I was playing with in passing |
| 01:50.02 | *** join/#asterisk sToRm_ (n=lol@c-71-234-68-58.hsd1.ct.comcast.net) |
| 01:50.22 | drclue | Starting with a blank hard drive , I've only been playing with asterisk for a month |
| 01:50.54 | sToRm_ | hi, can anyone help me install app_flite on freebsd? it won't install and spits out errors. i've googled and looked at tutorials for linux, but there seems to be little help for freebsd users |
| 01:51.19 | matt_d | what are the errors? |
| 01:51.24 | *** join/#asterisk Kumbang (n=dsp@167.205.24.69) |
| 01:51.30 | sToRm_ | would you like me to pastebin? |
| 01:51.54 | sToRm_ | http://pastebin.com/m35989865 |
| 01:52.01 | matt_d | in a private message pls |
| 01:52.08 | matt_d | thats fine too :) |
| 01:52.11 | sToRm_ | lol |
| 01:52.13 | sToRm_ | :p |
| 01:53.04 | matt_d | Missing dependency. What is on line 50 of the make file? |
| 01:53.21 | sToRm_ | ifeq ($(ASTERISKVERSION),1.2) |
| 01:53.21 | sToRm_ | <PROTECTED> |
| 01:54.20 | drclue | Of course , what brings me here today is what is probably a simple question. I've scribbled up a Fast AGI server with a persistent AMI connection and I'm wondering how to from within the FastAGI server to check the outcome of a Dial command? I don't mean the 200 result=0 but rather Busy,noanswer , answer etc. |
| 01:54.38 | drclue | matt_d: I was able to spy on my other extensions |
| 01:55.08 | matt_d | drclue: chan spy works, but i cant whisper. thats the only thing that doesn't work on this version. well if 1.6 worked i will load up a VM with 1.6 :) |
| 01:55.39 | matt_d | sToRm_ : it looks like there is a missing operator in your make command. did you read the INSTALL or INTSLALL.txt or README.txt ? |
| 01:55.57 | matt_d | sorry for the spelling mistakes. the terminal im on doesnt hav e a working backspace key. |
| 01:55.57 | sToRm_ | i read README, there is no INSTALL |
| 01:56.05 | sToRm_ | it says make;make install |
| 01:56.24 | sToRm_ | i commented out those two ifeq blocks, because i'm not running 1.2 or 1.0 |
| 01:56.31 | sToRm_ | now it's compiling but coming out with more errors |
| 01:56.35 | matt_d | sToRm_ : does that error come up on make or make install? |
| 01:56.39 | sToRm_ | make |
| 01:57.20 | drclue | matt_d: I would need to play with the chan spy thing a bit more. I'm only a month into learning Asterisk and have really been focusing on building my FastAGI/AMI tools. |
| 02:00.12 | matt_d | drclue : im thinking the ownly returns are SUCCESS, FAILURE and HANGUP |
| 02:00.12 | dandate2 | matt_d if whisper works in 1.6 please report back |
| 02:00.21 | matt_d | dandate2 : will do. im loading up a M (damn backspace key) VMWare image right now to try 1.6 |
| 02:00.49 | drclue | Right now I need to figure out how to check the outcome of a Dial command. Any ideas? If I can resolve this item , I can go play with the channel spy thing for a few days |
| 02:00.49 | drclue | as I'm way ahead of schedule on this project |
| 02:01.45 | matt_d | drclue: well then I am ALL YOURS! |
| 02:01.47 | matt_d | hehe |
| 02:02.45 | *** join/#asterisk dmz (n=dmz@64.203.233.195.dyn-cm-pool35.hargray.net) |
| 02:04.15 | matt_d | drclue: the DIALSTATUS var should have BUSy, NO ANSWER, ANSWER, CANCEL, DONTCALL, etc .. |
| 02:04.58 | *** join/#asterisk [netman] (n=netman@112.Red-83-38-221.dynamicIP.rima-tde.net) |
| 02:05.00 | drclue | Should I pluck DIALSTATUS from AMI getvar? |
| 02:06.48 | *** join/#asterisk OrNix (n=ornix@78.40.81.34) |
| 02:06.49 | drclue | Well there it is |
| 02:07.03 | matt_d | yes |
| 02:09.01 | drclue | Cool , now I have a plugin for my FastAGI server that weighs in at 7 lines of code , answers a call,plays a message spoofs the callerid/callername, does a hunt and reports the result. |
| 02:09.43 | matt_d | cool |
| 02:09.45 | drclue | So lets see what we can do with chan spy |
| 02:09.47 | matt_d | loves asterisk |
| 02:10.02 | matt_d | well.. not the can -- chanspy vproblem :) |
| 02:11.02 | drclue | I love asterisk , but hate the dial plan idea. Everything I'm working on tries to skip as much of the dial plan stuff as possible and avoid reloads |
| 02:11.07 | *** join/#asterisk OrNix (n=ornix@78.40.81.34) |
| 02:11.38 | matt_d | good idea. |
| 02:11.57 | *** join/#asterisk OrNix (n=ornix@78.40.81.34) |
| 02:12.10 | drclue | OK, lets see if we can point me at some chanspy docs |
| 02:12.52 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
| 02:12.56 | matt_d | i'm sure you know about it, but voip-info.org si a great wiki for asterisk: http://www.voip-info.org/wiki/view/Asterisk+cmd+ChanSpy |
| 02:13.30 | sToRm_ | i can live with TTS, but now i have another question. i'm trying to use the ${CALLERIDNUM} variable to read back the caller's telephone number (essentially making an ANAC), but it's empty |
| 02:13.42 | drclue | So you want ExtenSpy |
| 02:14.29 | matt_d | drclue: no, ChanSpy . This is whats happening. User calls in and plugs in phone number 1 and phone number 2. A call file is then generated with the spy group number. |
| 02:14.57 | matt_d | drclue: the user is then placed in spy mode with shisper looking for that spy group. call file calls the first phone number then the second. i need to be able to whisper to the first phone number. |
| 02:15.16 | matt_d | drclue: i had it working on one call.. but then about ten calls after that it was delayed about 10 seconds. now, no voice is going though chan spy. |
| 02:15.44 | matt_d | Strogg |
| 02:16.01 | matt_d | woops.. i mean STORM_: its CALLERID(num) |
| 02:17.03 | matt_d | CALLERIDNUM was taken iout in 1.4 i think |
| 02:17.31 | *** join/#asterisk Kumbang (n=krwlng@125.163.83.153) |
| 02:17.46 | sToRm_ | i'll give it a shot |
| 02:18.21 | drclue | I use EXEC SET CALLERID(num)=805-555-1212 |
| 02:18.36 | drclue | and SET CALLERID(name)=SomeName |
| 02:18.47 | sToRm_ | \o/ |
| 02:18.49 | drclue | EXEC SET CALLERID(name)=SomeName I mean |
| 02:18.50 | sToRm_ | it works, thank you |
| 02:19.16 | matt_d | sToRm_ any time |
| 02:19.49 | drclue | matt_d : are you recording those spied on channels too? |
| 02:20.11 | matt_d | drclue: no |
| 02:20.15 | matt_d | drclue: i saw that bug too |
| 02:20.27 | drclue | Just checking :) |
| 02:21.00 | matt_d | drclue : yeah no prob :) i appreciate the help.. im installing a new VMWare system still .. i just had to find my ub server cd . |
| 02:23.18 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 02:24.35 | matt_d | i never thought of this. does the zaptel timer thing need to be installed for chanspy to work right? |
| 02:24.38 | matt_d | i wouldn't think so. |
| 02:25.35 | russellb | nope |
| 02:25.58 | drclue | Well, I'm not installing my Digium card until Monday , so right now I'm just using ztdummy and at least the basic chanspy works |
| 02:26.24 | matt_d | i'm teling you man. this has got me so frustrated. i took a break and went out on my bike. i was jamming down the street at 170 MPH becuase i was so freaking mad! hehe |
| 02:26.34 | matt_d | ten i knew it was time to slow down and come back home ... |
| 02:27.08 | drclue | matt_d : Is there a particular reason you think ExtSpy would not work for you? It looks like it would be easier to use |
| 02:28.20 | matt_d | hmm... i thought that was just for zap or something. i t looks here you can spy on group as well. |
| 02:28.23 | matt_d | let me try that real quick! |
| 02:29.14 | matt_d | i though it ws just for extension only. i hope this works :) |
| 02:30.08 | matt_d | k ... here we go |
| 02:31.42 | drclue | IT might work , might not. I've only been doing asterisk for about a month |
| 02:31.51 | matt_d | no joy. |
| 02:32.07 | matt_d | i can hear everything, but can't whisper to the extention im spying on |
| 02:32.15 | matt_d | same as chanspy |
| 02:33.07 | matt_d | that was not on 1.6 im still installing that |
| 02:33.18 | drclue | Well, that is some sort of good sign. It would seem you can use ExtSpy , and that at least the problem is not specific to one or the other |
| 02:33.59 | matt_d | i'm only spying though the group id becuase the channel number is unknown |
| 02:34.09 | matt_d | wonder if thats why.. that shouldn't be the case though becuase I can hear everything .. |
| 02:36.27 | drclue | I know when I vector things into FastAGI from the dialplan asterisk coughs up the channel ID and all that good stuff |
| 02:37.20 | matt_d | heres the thing. the user logins in and enters the two phone numbers. it then generates a call file which calls out those two numbers. thats why the user context cant know about the new channels |
| 02:37.24 | matt_d | or is there a better way? |
| 02:37.27 | matt_d | such as AGI? |
| 02:37.47 | drclue | This sounds like a job for FastAGI |
| 02:37.57 | *** part/#asterisk levity (i=canuck@unaffiliated/canuck) |
| 02:38.14 | matt_d | I haven't used FastCGI .. i will try it out right now ); |
| 02:40.33 | drclue | While I've only been at this Asterisk thing for a month , I've already found that the most powerful ground I can stand on is a FastAGI server with a persistent AMI connection |
| 02:40.33 | drclue | which allows me to pick and choose my way through the mine field. So far I've really not seen any one of Asterisk's modes being all there , so I sorta rolled them into one class |
| 02:41.13 | *** join/#asterisk bulba29 (n=bulba29@96.245.85.180) |
| 02:41.27 | bulba29 | Hello...I am trying to register ny vitelity trunks vi the asterisk-gui |
| 02:41.49 | bulba29 | no can do...i can't find a lick of documentation on the asterisk-gui....is that by design? |
| 02:42.00 | *** join/#asterisk EiNSTeiN_ (n=einstein@unaffiliated/einstein/x-615171) |
| 02:42.32 | drclue | bulba29 : Lots of Asterisk GUIs out there , and yes documentation really sucks |
| 02:42.45 | matt_d | bulba29 its unsupported.. i don't know anything about asterisk gui. |
| 02:43.01 | bulba29 | ok, so i guess just stick with nano and my conf files |
| 02:43.08 | bulba29 | because i had that working fin |
| 02:43.14 | bulba29 | e |
| 02:43.30 | bulba29 | i didnt realize asterisk-gui was so disjoined |
| 02:43.57 | *** join/#asterisk [T]ank (n=chwall@24.10.218.148) |
| 02:43.58 | drclue | bulba29 : I sorta like the PIAF distro with the FreePBX web GUI. Sorta klunky , but it works and I was able to setup my asterisk without looking at dial plans |
| 02:44.14 | drclue | or looking at docs |
| 02:44.25 | matt_d | drclue: is it possible not to have to create a call file? becuase if you use the dial command ... hmm.. i lost my train of though :) |
| 02:44.26 | bulba29 | yeah, i had trixbox running my office for a year and a half |
| 02:44.38 | bulba29 | so i got comfortable with freepbx |
| 02:44.42 | [T]ank | is anyone here using sip through the "astaro" firewall? |
| 02:44.55 | bulba29 | but i just decided i wanted to slim it down and simplify and trixbox seemed to be getting bloated to me |
| 02:45.08 | drclue | I'm using SIP WiFi phones with NAT |
| 02:45.41 | [T]ank | drclue: was that directed at my question? |
| 02:46.37 | drclue | [T]ank: I think so |
| 02:47.23 | [T]ank | how did you enable your rtp traffic? |
| 02:48.05 | drclue | Look up iptables NAT and SIP |
| 02:48.30 | [T]ank | i know how to do it... just trying to fogure out how to do it in Astaro |
| 02:49.03 | *** join/#asterisk bbryant (n=brett@c-68-59-22-114.hsd1.sc.comcast.net) |
| 02:50.23 | drclue | I don't know Astaro , but figured bringing up NAT would send you in the right direction. |
| 02:59.46 | *** join/#asterisk BeeBuu (n=beebuu@219.135.42.80) |
| 02:59.52 | BeeBuu | hi,all |
| 02:59.59 | matt_d | Hi BeeBuu |
| 03:00.15 | BeeBuu | new to asterisk 1.6, how to config my E1 cardï¼ |
| 03:00.31 | BeeBuu | matt_d: nice to meet you. |
| 03:00.37 | matt_d | BeeBuu , im installing 1.6 right now myeself. but it should be the same. |
| 03:00.39 | matt_d | don't u think |
| 03:01.09 | BeeBuu | is there a zaptel.conf too? |
| 03:01.58 | *** join/#asterisk _guitarman_ (n=guitarma@d209-121-157-169.bchsia.telus.net) |
| 03:02.02 | matt_d | BeeBuu: http://www.voip-info.org/wiki/view/Asterisk+Data+Configuration |
| 03:02.50 | matt_d | BeeBuu: and http://www.voip-info.org/wiki/view/Asterisk+config+zaptel.conf |
| 03:04.28 | BeeBuu | matt_d: thanks |
| 03:04.57 | BeeBuu | matt_d: are you using asterisk 1.6? |
| 03:05.11 | matt_d | BeeBuu: no, still installing it right now. |
| 03:05.15 | matt_d | im using an older version |
| 03:05.54 | BeeBuu | /etc/zaptel.conf Becomes /etc/dahdi/system.conf after May 19th 2008. |
| 03:06.36 | BeeBuu | matt_d: any way,thanks for you help . |
| 03:08.10 | matt_d | BeeBuu: http://www.voip-info.org/wiki/view/DAHDI |
| 03:17.28 | box2 | are there any IAX hardphones that are worth paying for? |
| 03:18.24 | box2 | only requirement is downloading config from something better than tftp |
| 03:22.09 | matt_d | man i hope cahnspy works in 1.6 |
| 03:24.04 | matt_d | ls |
| 03:28.34 | drclue | matt_d :Monday , I'll get the Digium card installed and add a couple of land lines into the mix so I can have something to spy on. I have two WiFi phones here in my old hippie school bus. The server is sitting about 150 miles away in Santa Barbra, so I have to wait for my slave with the hands to show up for work and stick the card in. |
| 03:31.42 | matt_d | drclue , hehe cool. your server is out my way :) |
| 03:34.15 | drclue | I myself am located near Brawley, Ca |
| 03:35.06 | matt_d | cool. i live in LA. |
| 03:35.09 | matt_d | and San Jose |
| 03:35.30 | drclue | II used to live in Sunnyvale during the dot com boom |
| 03:35.43 | BeeBuu | ~book |
| 03:35.44 | 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 |
| 03:36.19 | *** join/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 03:36.21 | matt_d | drclue: its still a very nice city |
| 03:36.44 | *** part/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 03:36.48 | matt_d | SJ is my primary home.. i was out riding my cycle today it was burning hot! |
| 03:36.52 | matt_d | went up though sunnyvale and back down |
| 03:37.35 | BeeBuu | is there a book about asterisk 1.6? |
| 03:37.48 | drclue | Down here it stays around 110-125 all summer |
| 03:38.28 | matt_d | oh thats right u r near brawley :) |
| 03:38.38 | matt_d | i used to live in LanCaster ... i hated that heat! |
| 03:38.44 | matt_d | it would be like that all the time |
| 03:39.27 | drclue | I run two air conditioners here in the bus when it gets really hot. |
| 03:40.07 | matt_d | u were not kidding about the bus then : |
| 03:40.08 | matt_d | :) |
| 03:41.13 | drclue | Heck no. I have wireless internet , solar power , 1200 Amp hours of batteries , all my phones are voip. |
| 03:41.40 | matt_d | do u just travel oall over? |
| 03:42.24 | drclue | I pretty much stay put , but when it's time to move I just turn the key. The air conditioners run on Nacho Power |
| 03:43.37 | drclue | The bus is a 1967 Superior Coach school bus on a Ford 500 Chassis |
| 03:44.06 | matt_d | Nacho Power ... oil? :) |
| 03:45.01 | drclue | Nacho = Not Yours . In my case , a lonely light pole |
| 03:46.35 | drclue | Nacho Water , Nacho land , Nacho Building to park the bus in |
| 03:46.59 | matt_d | hehe |
| 03:47.10 | drclue | Too far away from the city for Nacho internet, but I'm working on it |
| 03:47.28 | matt_d | what kind of wireless internet do u have hooked up? sat or cell ? |
| 03:48.08 | drclue | Terrestrial radio link internet provider. When I'm up in the mountains I use a sat |
| 03:48.46 | drclue | Just enough bandwidth to watch HULU and make a phone call |
| 03:48.54 | matt_d | love HULU! |
| 03:49.30 | drclue | Until D*sh network went Nag3 I watched nacho dish for years |
| 03:50.32 | matt_d | okay here we go. testing the spying with 1.6 |
| 03:51.04 | drclue | Now I'm eying an abandoned car dealership wit ha big 6 foot sat dish so I can watch c-band/Ku-band for FREE and legal |
| 03:51.34 | matt_d | hmmm... ChanSpy works .. but whiser is delayed by 10 seconds .. |
| 03:51.37 | matt_d | going to try extenspy now |
| 03:52.22 | matt_d | woops.. that was extenspy that was delayed |
| 03:52.25 | matt_d | going to try cahnspy now |
| 03:52.52 | drclue | Well , if it has a ten second delay , whispering will suck working or not |
| 03:57.34 | matt_d | that is my dream camping rip |
| 03:57.36 | matt_d | rip = trip |
| 03:59.27 | matt_d | okay weird.. chanspan and extenspy work alike |
| 03:59.38 | drclue | I figured they would |
| 03:59.38 | matt_d | both have 15 (not 10) second delay. is this by design? |
| 03:59.45 | drclue | Nope |
| 03:59.56 | matt_d | <PROTECTED> |
| 04:00.10 | drclue | What good would a whisper be heard 10,15,20,30 seconds later |
| 04:00.15 | matt_d | i know its not wnetwork bog down or cpu bog down |
| 04:00.55 | drclue | Seems like maybe life is thread locked |
| 04:01.21 | matt_d | i am going to try biridging |
| 04:01.32 | matt_d | bridigin (darn backspace key). but i bet all audio will pass though to everyone |
| 04:01.35 | matt_d | worth a try |
| 04:01.41 | drclue | Issuing those commands from another thread might make it work better |
| 04:02.54 | dandate2 | matt_d what kind of router do u have |
| 04:03.14 | matt_d | dandate2: here at home a Linksys router |
| 04:03.15 | matt_d | NAT |
| 04:03.35 | matt_d | and its calling out to 2 real phones though my IAX provider. |
| 04:03.44 | matt_d | think NAT is loosing packets? |
| 04:03.55 | dandate2 | hmm u might need soemthing voip quality |
| 04:03.56 | drclue | Linksys/NAT here too |
| 04:04.06 | matt_d | i can hear just fine |
| 04:04.06 | dandate2 | these home routers i dno about |
| 04:04.10 | matt_d | no delay what so ever |
| 04:04.12 | *** join/#asterisk geneticx (n=geneticx@adsl-146-67-165.mia.bellsouth.net) |
| 04:04.17 | matt_d | its just when i try to talk (whisper) to the spied on channel |
| 04:04.50 | dandate2 | i have an SMC comcast business class router and i get the same crap where it takes like 10 seconds delay to dial out when using a phone plugged into a linksys voip router but not through softphone |
| 04:05.11 | dandate2 | so i dno if for me its comcast or just linksys in general |
| 04:05.20 | dandate2 | but it seems like we got something in common |
| 04:05.23 | matt_d | i can hear just fine without delay .. from the real phone. i can call out to the public telco and have a non delayed converstion |
| 04:05.30 | matt_d | its just the channel spying :) |
| 04:05.50 | drclue | I use linksys/NAT and WiFi phones , and no delays. This is strictly an asterisk thing |
| 04:06.05 | dandate2 | asterisk delaying 10 seconds huh] |
| 04:06.09 | dandate2 | are u using a pentium 4 lol |
| 04:06.18 | dandate2 | that would make sense |
| 04:06.21 | matt_d | hehe |
| 04:06.35 | matt_d | its been doing it on two machines. i have one phsysical box with asterisk (delayed) |
| 04:06.41 | matt_d | and the one im owrking now in VMWare |
| 04:06.56 | matt_d | but I have a fou r processor mac .. so there is no question about processing speed |
| 04:07.17 | drclue | I would as I mentioned try issuing those commands from outside the dialplan in a separate process/thread and see if it's threadlock |
| 04:07.44 | matt_d | i dont hink i can issuechanspy outsie the dialplan |
| 04:07.46 | matt_d | unless agi |
| 04:07.48 | matt_d | i will try |
| 04:08.20 | *** join/#asterisk DarkRift (n=dark@bas1-sthubert21-1096590347.dsl.bell.ca) |
| 04:08.31 | drclue | FastAGI (not just AGI) in conjunction with AMI is what I'm talking about |
| 04:09.16 | drclue | That will break it up into three or four threads and hopefully allow it to smooth out |
| 04:09.16 | *** join/#asterisk ArchGT (n=ArchGT@unaffiliated/archgt) |
| 04:09.40 | matt_d | okay |
| 04:10.00 | drclue | As is I would not be surprised if those delays tended to get longer and longer |
| 04:11.23 | matt_d | i was going to try bridging. but bridign stops at the command it doesnt contnue with the dialing plan |
| 04:11.25 | matt_d | errr... |
| 04:15.59 | drclue | I'm trying a google with ChanSpy whisper latency since "delay" keeps tripping on parameters |
| 04:17.50 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.229.58) |
| 04:18.30 | *** join/#asterisk troubled (n=troubled@unaffiliated/troubled) |
| 04:24.03 | sToRm_ | hi, i know there is ${CALLERID(num)}, but is there such a thing as ${CALLERID(ANI)} ? or how would i be able to determine the ANI of an incoming call? |
| 04:25.44 | matt_d | i dont nkow if it spoofs ANI |
| 04:26.46 | sToRm_ | i don't want to spoof ANI, just read the ANI of an incoming call |
| 04:27.17 | sToRm_ | ultimately doing a readback |
| 04:28.54 | drclue | I could have sworn I read something today about ANI on the /www.voip-info.org WiKi while I was spoofing regular callerid |
| 04:29.26 | drclue | Their WiKi does have a search box |
| 04:30.09 | sToRm_ | thank you, i'm well-aware |
| 04:30.42 | sToRm_ | i spoofcard'd to my PBX, and it read back BOTH (num) and (ANI), but neither were the spoofed CID |
| 04:31.15 | drclue | matt_d: are all sides of the connection using the same codec? |
| 04:31.31 | matt_d | drclue: good question.. let me check |
| 04:32.36 | matt_d | okay my outbound to the telco is ulaw |
| 04:32.49 | matt_d | and im using a softphone for the "user" pone |
| 04:32.54 | matt_d | i will for that to use ulaw and test it out |
| 04:35.02 | matt_d | dude i go t all happy becuase the whisper wasnt delay. then i realized i still had it to try bridging... damn.. |
| 04:35.05 | matt_d | going to retry now |
| 04:35.22 | drclue | I was just reading this somewhat old item and i made me think of the question |
| 04:35.22 | drclue | http://www.b2bua.org/wiki/AsteriskCodecNegotiationPatch |
| 04:40.18 | drclue | I also saw a piece about transcode_via_sln=no in the file asterisk.conf and locking everyone to ulaw |
| 04:40.48 | matt_d | how odo i force ulaw? |
| 04:40.53 | matt_d | for the outgoing iax |
| 04:41.21 | Kobaz | [Aug 10 00:40:32] WARNING[4670]: pbx.c:3080 pbx_extension_helper: No application 'LOCK' |
| 04:41.49 | Kobaz | I have context foo { bar => { LOCK(abc); } } |
| 04:42.05 | Kobaz | its not an application it's a function |
| 04:42.20 | Kobaz | i guess i can do Set(LOCK()..) |
| 04:43.13 | Kobaz | NoOp(LOCK(abc)); works |
| 04:44.16 | Kobaz | oh wait.. no... duh any args to noop are just printed |
| 04:45.22 | Kobaz | unless they are evaulated |
| 04:45.34 | Kobaz | NoOp(${LOCK(abc)}); |
| 04:45.38 | Kobaz | i think that should do it |
| 04:47.32 | matt_d | drclue: i think its all ulaw and still same result. im going to try and call in from landline and see if the softphone is the reason why |
| 04:49.30 | drclue | matt_g :softphones always seem to add a little delay , but 10 seconds does not sound right at all. BTW , here's an interesting little snippet I'm going to hold onto |
| 04:49.31 | drclue | exten => _520XXX,1,ChanSpy(Agent/${EXTEN:3},q) |
| 04:51.10 | matt_d | and that works great! its just the whispering part :) hehe |
| 04:52.36 | *** join/#asterisk joelsolanki (i=joelsola@202.160.161.94) |
| 04:52.46 | joelsolanki | Good evening / morning gusy |
| 04:52.47 | joelsolanki | guys |
| 04:53.21 | joelsolanki | i have a T1 connected to my asterisk. i wanted to know the highest concurrent calls connected on asterisk so that i can know if i need to increase my T1. |
| 04:53.35 | joelsolanki | is there any way to count those ? |
| 04:53.49 | matt_d | it depends on processing power |
| 04:53.51 | matt_d | and memory |
| 04:54.29 | joelsolanki | no i dont want to test that. i want know how much highest concurrent call it went |
| 04:55.13 | matt_d | u mean how many calls its handling at the moment? |
| 04:55.23 | kaldemar | use GROUP functions to set a group for all calls and save the counts somewhere |
| 04:56.01 | joelsolanki | core show channels shows me concurrent calls connect at moment. but i know what what was the highest concurrent call it handled. |
| 04:56.09 | joelsolanki | oh GROUP functions |
| 04:58.04 | joelsolanki | isnt there a way ? |
| 05:00.36 | *** join/#asterisk keebler (n=Christop@adsl-75-17-124-183.dsl.rcsntx.sbcglobal.net) |
| 05:00.46 | keebler | IS Asterisk GPL'd? |
| 05:00.50 | keebler | Or GNU? |
| 05:01.07 | keebler | Or, what opensource license is it under? |
| 05:01.15 | matt_d | GNU |
| 05:01.24 | keebler | Awesome. |
| 05:01.29 | [TK]D-Fender | keebler: http://www.asterisk.org/about |
| 05:01.31 | matt_d | GNU General Public License to be exact |
| 05:02.03 | keebler | [TK]D-Fender: Thanks. Didn't feel like starting X. |
| 05:02.04 | keebler | :) |
| 05:02.12 | [TK]D-Fender | GNU GPL *2* to be exact |
| 05:02.31 | [TK]D-Fender | keebler: thats why there is LINKS |
| 05:03.45 | keebler | Hey, let me stick my my excuse for laziness. |
| 05:03.46 | keebler | :) |
| 05:04.45 | *** join/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 05:05.00 | *** part/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 05:06.12 | linagee | are grandstream's FXS to ethernet products as crappy as their hard phones? (is there a better one?_ |
| 05:06.29 | [TK]D-Fender | linagee: Linksys <- |
| 05:08.04 | drclue | matt_g: here is another interesting item. Ignore for the moment that it is not whispering. |
| 05:08.04 | drclue | exten => _*29XXXX,1,Answer |
| 05:08.04 | drclue | exten => _*29XXXX,n,set(calltime=${STRFTIME(${EPOCH},,%C%y%m%d%H%M%S)}) |
| 05:08.04 | drclue | exten => _*29XXXX,n,MixMonitor(/var/spool/asterisk/monitor/X${calltime}X${CALLERID(num)}X${EXTEN:3}X.wav) |
| 05:08.04 | drclue | exten => _*29XXXX,n,Chanspy(SIP/${EXTEN:3}|q) |
| 05:08.04 | drclue | http://www.voip-info.org/wiki/view/MixMonitor |
| 05:08.59 | matt_d | wonder if there is an easy way to get the voice from the spier to the sped on channel |
| 05:09.00 | matt_d | that way |
| 05:11.56 | matt_d | wat is that zap dummy lib called? |
| 05:12.10 | drclue | ztdummy ? |
| 05:12.35 | drclue | AFAIK it uses the USB port for timing |
| 05:12.35 | matt_d | tathats it |
| 05:12.58 | *** join/#asterisk Borai (n=DYN@S0106001c109e98db.no.shawcable.net) |
| 05:13.12 | Borai | hello everyone, |
| 05:13.23 | drclue | The thing about the above fragment that had caught my eye had something to do with the thought of when the audio mixing occured |
| 05:14.04 | matt_d | drclue: i conifrmed that my softwa |
| 05:14.09 | matt_d | errr that my softphone is not causing delay |
| 05:14.13 | matt_d | hi Borai |
| 05:14.49 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 05:15.30 | drclue | Something about it mixing on the inside and not the outside of some layer of codecing. Lots of stuff to read and my eyes are starting to fall out. Longs for the days of a simple captain crunch whistle |
| 05:16.15 | Borai | I have a server that is running proxmox and in the proxmox environment i have installed trixbox with a KVM/QEMU, the system is totally voip only but the IVR promts are choppy when i run dahdi_test the timings are Best: 99.997 -- Worst: 95.149 -- Average: 99.058749, Difference: 100.001099 |
| 05:16.18 | matt_d | hehe |
| 05:16.41 | Borai | and the kernel modules are installed on the host |
| 05:17.24 | Borai | when i let the test run for a long time it goes down to 50 |
| 05:21.01 | Kobaz | voip under vm's generally doesn't work very well |
| 05:21.31 | Borai | hmm |
| 05:21.42 | Borai | ok |
| 05:22.00 | [TK]D-Fender | Stealing time-slices away from an app that streams packets at 20ms intervals = dumb |
| 05:23.27 | Borai | ok i guess then im gonna go back to centos and setup asterisk manually |
| 05:28.18 | Borai | Pentium IV 3,2 GHz, 2048MB RAM, 2x250 GB (S-ATAII) HDD, would that run asterisk with 10 users - 2,3 simul. calls at max.? |
| 05:31.18 | kaldemar | yes |
| 05:33.22 | drclue | Well , enough reading on chanspy with whisper , time to go watch some hulu |
| 05:33.53 | kaldemar | Borai: http://www.voip-info.org/wiki/view/Asterisk+dimensioning |
| 05:35.02 | [TK]D-Fender | Borai: More than 25 times over |
| 05:39.34 | [TK]D-Fender | checkout time, later all |
| 05:43.09 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 05:44.19 | *** join/#asterisk GrnFrog (n=GrnFrog@pool-173-78-9-181.tampfl.fios.verizon.net) |
| 05:44.52 | *** join/#asterisk fiddur (n=fiddur@192.121.104.122) |
| 05:50.12 | GrnFrog | Looking for insight getting res_snmp working in Asterisk 1.4 - http://pastebin.com/mff20f9e |
| 05:50.46 | GrnFrog | seems configured correctly but Asterisk MIBS not responding to snmpwalk |
| 05:58.46 | *** join/#asterisk JanisB (i=rootz@unaffiliated/janisb) |
| 05:59.08 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 06:01.26 | box2 | i thought you said Asterisk AIDS |
| 06:01.48 | box2 | facepalms |
| 06:03.20 | *** join/#asterisk Octothorpe (n=Octothor@pdpc/supporter/professional/octothorpe) |
| 06:09.44 | *** join/#asterisk bbryant1 (n=brett@c-68-59-22-114.hsd1.sc.comcast.net) |
| 06:09.47 | *** join/#asterisk scardinal (n=supreme@0905ds1-rdo.0.fullrate.dk) |
| 06:19.28 | *** join/#asterisk ming_zym (n=ming_zym@202.165.107.100) |
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| 06:24.46 | *** join/#asterisk Kumbang (n=punch@125.163.83.153) |
| 06:26.23 | matt_d | im done for the niht |
| 06:26.27 | matt_d | guys nite |
| 06:34.30 | *** join/#asterisk xrmx__ (n=rm@87.13.48.2) |
| 06:41.47 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-8244b6ed4c5d6c4f) |
| 06:43.18 | *** join/#asterisk oej (n=olle@ns.webway.se) |
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| 06:46.10 | *** part/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 06:57.35 | *** join/#asterisk toasterisk (n=zhulizho@58.251.230.1) |
| 07:04.04 | toasterisk | hello to all of you |
| 07:04.41 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
| 07:05.55 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 07:06.50 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 07:07.18 | *** join/#asterisk icyValk77 (n=icyValk7@gateway.ash.thebunker.net) |
| 07:10.13 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-b9dbfc7ec1ddf0a9) |
| 07:15.45 | *** join/#asterisk DuPe (i=blahsss@S0106001217be899c.ed.shawcable.net) |
| 07:16.34 | DuPe | whee.. i love playing with fun stuff that makes my brain hurts. good thing i got coffee |
| 07:26.30 | *** join/#asterisk oej_ (n=olle@ns.webway.se) |
| 07:27.56 | DuPe | if anyone is available for some paid consulting work ;) drop me a line... trying to reconfigure 3 asterisk boxes and could use some help |
| 07:29.16 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 07:32.00 | toasterisk | what is your problem? |
| 07:32.09 | toasterisk | just connecting 3 boxes? |
| 07:32.15 | *** join/#asterisk Tim_Toady (n=moi@adsl52-231.kln.forthnet.gr) |
| 07:33.40 | DuPe | i got shoved into a project.....well i'm rebuilding them from scratch with asterisknow, and making that happen, then i need to connect them together, havent done that before :P |
| 07:35.38 | toasterisk | there are some links for two asterisk connection |
| 07:35.47 | toasterisk | it should be good to you |
| 07:36.02 | DuPe | whew ;) |
| 07:36.57 | toasterisk | http://www.voip-info.org/wiki/view/Asterisk+Connect+2+servers |
| 07:37.59 | DuPe | awesome, that does look good. i hadnt gotten that far yet, still fighting with some dahdi quirks but thats good to know |
| 07:38.34 | toasterisk | it should be no problem. you can use sip or iax to connect 2 boxes |
| 07:39.50 | toasterisk | from dahdi? how do you connect two asterisks |
| 07:40.14 | toasterisk | unless you use fxs and fxo for pstn and net and cpe for pri |
| 07:40.42 | DuPe | have pstn and sip providers |
| 07:41.55 | DuPe | as |
| 07:41.59 | DuPe | er |
| 07:46.44 | *** join/#asterisk alunca (n=alun@c-24-130-216-30.hsd1.ca.comcast.net) |
| 07:46.54 | alunca | hello |
| 07:49.23 | DuPe | i love my coffee pot mmm |
| 07:55.35 | *** join/#asterisk JanisB (i=rootz@unaffiliated/janisb) |
| 07:55.45 | alunca | should i buy a voip phone or ata adapter? |
| 07:56.38 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
| 08:08.32 | *** join/#asterisk iksik (i=xk@livedata.pl) |
| 08:15.27 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 08:19.18 | *** join/#asterisk Fisknisse (n=fd@cpe90-146-29-51.liwest.at) |
| 08:19.48 | Fisknisse | Anyone know where to download a gsm codec so i can listen to recordings from asterisk. Running windows. |
| 08:20.30 | Fisknisse | I found a codec a coupple of years ago and now i reinstalled my computer :( |
| 08:20.40 | Fisknisse | Its impossible to find it again... |
| 08:21.38 | DuPe | quicktime can play them |
| 08:22.10 | Fisknisse | I could play them in winamp.. it was really great to not have to install quicktime. |
| 08:22.26 | DuPe | theres a winamp plugin too i recall seeing |
| 08:22.56 | DuPe | google says... http://mlkj.net/gsm/ |
| 08:24.54 | Fisknisse | It works like dream. Thank you verry much! |
| 08:29.59 | *** join/#asterisk Gnarfy (n=mirc@213.144.244.115) |
| 08:32.03 | *** part/#asterisk toasterisk (n=zhulizho@58.251.230.1) |
| 08:47.20 | *** join/#asterisk xrmx__ (n=rm@host204-254-dynamic.15-87-r.retail.telecomitalia.it) |
| 08:49.27 | *** join/#asterisk Gnoll (n=Gnoll@ppp-175-15.98-62.inwind.it) |
| 08:49.28 | Gnoll | hi |
| 08:50.21 | Gnoll | there's some italian in this room? |
| 08:53.59 | *** join/#asterisk af_ (n=getsmart@88-149-230-210.dynamic.ngi.it) |
| 08:54.35 | *** join/#asterisk Pouet78 (i=56d6b0a6@gateway/web/freenode/x-8c226634d5c3dd8e) |
| 08:54.41 | Pouet78 | Hi! |
| 08:54.53 | box2 | hello |
| 08:54.59 | box2 | and welcome to your daily internets |
| 08:55.06 | Pouet78 | I try to use asterisk with MGCP. |
| 08:56.27 | Pouet78 | I try to "copy" a working network |
| 08:58.00 | Pouet78 | on the network I have this sequence : |
| 08:58.34 | Pouet78 | box > server RSIP |
| 08:58.37 | Pouet78 | < OK |
| 08:58.49 | Gnoll | |
| 08:58.49 | Gnoll | What comes from this problem to you: WARNING[9619]: chan_sip.c:18389 reload_config: Section 'number-out' lacks type ? |
| 08:58.51 | Pouet78 | box < Server AUEP |
| 08:58.57 | Pouet78 | > OK |
| 08:59.09 | Pouet78 | <DLCX |
| 08:59.25 | Pouet78 | > OK |
| 08:59.28 | Pouet78 | < RQNT |
| 08:59.31 | Pouet78 | > OK |
| 08:59.32 | Pouet78 | < RQNT |
| 08:59.33 | Pouet78 | > OK |
| 08:59.55 | Pouet78 | but with asterisk I only have the first 2 packets. |
| 09:00.35 | Pouet78 | Is it a normal behavior? |
| 09:00.44 | Pouet78 | Can I change it? |
| 09:01.24 | *** join/#asterisk SALstar (i=ondrejj@work.salstar.sk) |
| 09:04.50 | *** join/#asterisk alunca (n=alun@c-98-210-114-183.hsd1.ca.comcast.net) |
| 09:05.19 | alunca | I installed AsteriskNow; how can I change screen to 1280x1024? thanks! |
| 09:05.47 | *** join/#asterisk TommyBotten (n=tommy@217-14-12-26-dhcp-osl.bbse.no) |
| 09:12.50 | SALstar | When using SendFax application using outgoing directory, how I can get variables like FAXSTATUS, FAXERROR or FAXPAGES? |
| 09:15.07 | oej_ | Gnoll: You have a section without type=peer/friend/user |
| 09:15.23 | Gnoll | oej i have section type=friend |
| 09:16.04 | oej | in [number-out] ? Then the message doesn't make sense |
| 09:17.06 | Gnoll | if he had the sense I was not here: D |
| 09:17.49 | Gnoll | I understand the problem, i check ... |
| 09:24.21 | alunca | is there anyway to change the resolution of the AsteriskNow 1.5? |
| 09:33.27 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-64d8f9d2600f53f1) |
| 09:35.19 | *** join/#asterisk nightrid3r (n=kvirc@78-20-232-172.access.telenet.be) |
| 09:39.06 | *** join/#asterisk arthax0r (i=arthax0r@gateway/shell/blinkenshell.org/x-3dcf557d8e59b7b3) |
| 09:39.28 | Gnoll | i have a new problem, i set musiconhold(default) on incoming call, the debug file report this error: == Auto fallthrough, channel 'SIP/83.211.2.218-b7b45ae8' status is 'UNKNOWN' |
| 09:58.24 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 09:58.25 | dymaxion | hi there, how powerful hardware woudl I need to power an Trixbox based PBX for 10 users? Would I need something like a most basic Nehalem Xeon 5502 suffice? 6Gb RAM ? |
| 09:59.15 | Chainsaw | dymaxion: That seems ample, unless you're planning to host conferences with all 10 users at the same time. |
| 09:59.40 | Chainsaw | dymaxion: Don't cheapen out on your FXO/FXS cards, if any, and you'll be fine. |
| 10:00.10 | dymaxion | we are planning to have only VOIP via a sip trunk provider... no PSTN hardlines |
| 10:00.44 | JD | dymaxion: we used to run the phones for 25+ users on a P3 with less than 512Mb |
| 10:00.44 | Chainsaw | *nod* No conference bridging planned? |
| 10:00.56 | Chainsaw | JD: It's always good to design some margin into a system though. |
| 10:01.09 | dymaxion | we have 2 branches one in UK one in HK, do we build 2 PBXs... conferences between two offices & possibly external... dialin... if everyone at home possibly 10 ppl dialing in |
| 10:01.34 | dymaxion | "dial-in" i mean conect over internet |
| 10:01.49 | Chainsaw | dymaxion: Doing conferences right generally means you need a telephony adapter just for clocking. |
| 10:01.52 | JD | Chainsaw: there's some margin and then there's massively overspeccing :) |
| 10:02.08 | JD | my point was that you don't need a huge amount of processing power |
| 10:02.10 | Chainsaw | dymaxion: (Even if everything else is SIP) |
| 10:02.31 | Chainsaw | JD: Until you get to conferences where one or more participants are on an oddball codec. |
| 10:02.38 | dymaxion | for clocking? ("£*$& ??) even if all on SIP... oh ? |
| 10:02.54 | dymaxion | when you have codec/transcoding then need CPU i read.. |
| 10:03.05 | dymaxion | we also need video conferencing |
| 10:03.17 | Chainsaw | dymaxion: Indeed, for clocking. It may have gone away as a requirement in the Asterisk 1.6 series at some point. |
| 10:03.31 | Chainsaw | dymaxion: But for 1.2 or 1.4 you definitely need clocking from a zaptel/DAHDI card. |
| 10:03.43 | dymaxion | when you say clocking .. in what context of the word?? don't rmmeber reading about that... dont understand req. |
| 10:03.57 | Chainsaw | dymaxion: But video codec support is strictly 1.6 last I checked, so your clocking requirement may vanish as a result of that. |
| 10:03.58 | viraptor | Chainsaw: doesn't zt_dummy with HPET kernel solve it? zt_test gets me something ~99.99% min in that setup |
| 10:04.24 | Chainsaw | viraptor: If your hardware is suitable and the driver compiles, yes. |
| 10:05.34 | dymaxion | Chainsaw, thanks. Do I need one PBX per office and setup routing between the two.... we need the two office to appear as if they were one.. is that posisble with asterix? |
| 10:06.07 | Chainsaw | dymaxion: I would install two PBXes (to minimise latency towards the local phones) and bridge them together using IAX. |
| 10:06.10 | viraptor | it's weird... other pbxes work perfectly fine with >40 people on pthread timing :/ |
| 10:06.21 | dymaxion | looks like a lot more user support / helpful people here than on the #sipx camp.... is there any reason I should choose sipx over asterisk given my requiements? |
| 10:06.42 | Chainsaw | I've not worked with sipx, couldn't say. |
| 10:07.36 | dymaxion | fair nuff.. seems like asterix should have all my bases covered... and presumably I should pick a VOIP provider that has direct presence in that geographical region |
| 10:08.54 | Chainsaw | Generally best if you want termination, yes. |
| 10:09.13 | Chainsaw | (i.e. a phone number reachable from the regular phone system) |
| 10:13.27 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 10:15.47 | dymaxion | Chainsaw, thanks v much. also can't find a decent softphone for linux, somethign like http://www.sipx.com/_IMAGES/products/softphone.gif Ekiga is always the one that crops up, but I find doesnt fit the bill... |
| 10:16.03 | Chainsaw | dymaxion: Ekiga is in fact the best one I know. |
| 10:16.09 | Chainsaw | dymaxion: Did you try version 3? |
| 10:16.58 | viraptor | dymaxion: twinkle or ekiga |
| 10:17.00 | dymaxion | I'm using 3.2.5, but what somethign that looks more like a telephone! that can easily transfer / on-hold, etc calls. Suppose I can live with ekiga |
| 10:17.50 | *** join/#asterisk sercik (n=ciccio@host48-111-dynamic.53-79-r.retail.telecomitalia.it) |
| 10:17.54 | sercik | hi! |
| 10:18.19 | sercik | someone can point me to a quickstart for very noob?? |
| 10:21.01 | DuPe | sercik: http://www.voip-info.org is the best place to start |
| 10:21.16 | sercik | hi! |
| 10:21.18 | sercik | dupe |
| 10:21.27 | sercik | i'm very new :) |
| 10:21.48 | sercik | whick hardware is needed to create a postation with 4 client? |
| 10:22.20 | sercik | i need a pc ok? a linux distribution the voip phone and then? |
| 10:22.33 | sercik | asterisk indeed |
| 10:22.56 | sercik | i need some special hardware into pc? |
| 10:23.59 | mbrevda | depends on the type of lines your using |
| 10:24.02 | mbrevda | ~book |
| 10:24.03 | 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 |
| 10:24.45 | sercik | a normal pstn |
| 10:28.16 | mbrevda | sercik: I would recomend that you read the book ^^ |
| 10:28.36 | sercik | you are right i know |
| 10:28.40 | sercik | is important to read |
| 10:29.00 | sercik | i hope that i can understand engligh correctly |
| 10:32.19 | *** join/#asterisk stix (n=stix@exchange2003.corporate.billetkontoret.dk) |
| 10:33.05 | stix | I remember, that "The book" is available online in pdf somewhere - but where? |
| 10:36.58 | dymaxion | is it possible to encrypt all sip & video traffic using asterisk and softphones? |
| 10:39.31 | Gnoll | bye |
| 10:39.32 | dymaxion | is ZPhone the best option for this? |
| 10:40.36 | *** join/#asterisk Had (n=chatzill@82-45-194-9.cable.ubr02.hari.blueyonder.co.uk) |
| 10:40.50 | Had | hello there |
| 10:41.36 | Had | i'm just trying to setup linksys SPA2102 with T.38 |
| 10:42.12 | Had | my provider support T.38 with their SIP link |
| 10:42.25 | Had | they use Cisco AS5300 gateway |
| 10:43.06 | Had | can somebody help me with this? |
| 10:48.05 | *** join/#asterisk grEvenX (n=even@apb9hb.ip.ssc.net) |
| 10:53.33 | *** join/#asterisk cidu (n=hagbard@whthyt253-29.northwestel.net) |
| 10:56.14 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 11:05.14 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 11:07.34 | *** part/#asterisk awkfu (n=awkfu@66.162.90.57) |
| 11:13.03 | *** join/#asterisk odenkos (n=odenkos@85-135-151-103.adsl.slovanet.sk) |
| 11:17.42 | *** join/#asterisk oberon (n=oberon@89-138-172-78.bb.netvision.net.il) |
| 11:17.46 | oberon | hi |
| 11:17.56 | oberon | I'm new to asterisk |
| 11:18.19 | oberon | very neww actually, I just installed version 1.6.0.10 |
| 11:18.38 | oberon | here is what I need/want to do: |
| 11:19.02 | oberon | my box has a telephony provider server it can connect to using SIP |
| 11:19.16 | oberon | I have the IP+username+password |
| 11:19.37 | oberon | the box itself isnt suppose to accept calls |
| 11:19.47 | box2 | ~book |
| 11:19.48 | 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 |
| 11:20.09 | oberon | what I want to do it tell it to dial to number A, then to dial to number B and then connect the 2 calls |
| 11:20.14 | oberon | is it possible ? |
| 11:20.29 | manxpower | oberon: Yes! .call files. Read the book |
| 11:20.44 | oberon | it's a 600 pages book |
| 11:20.53 | manxpower | oberon: and that is why I said ".call files" |
| 11:21.14 | DuPe | its good bathroom reading material |
| 11:21.14 | manxpower | But I can tell you that most of us are not going to explain .call files for the 5,000th time. |
| 11:21.39 | oberon | ok .. |
| 11:21.53 | oberon | I liked the "most of us" phrase :-) |
| 11:23.39 | odenkos | liked? |
| 11:23.40 | odenkos | :D |
| 11:23.49 | *** join/#asterisk zeeesh (n=zeeesh@203.215.179.43) |
| 11:25.23 | Had | is there anything about t.38 in that book? |
| 11:26.49 | zeeesh | getting error 'ERROR[5259]: res_jabber.c:1889 aji_client_initialize: JABBER ERROR: No Connection'? |
| 11:27.16 | manxpower | Had: I doubt it. T.38 was not supported in Asterisk except in "pass-thru" mode until at least 1.6 and the book has not been updated for 1.6 yet. Remember, there are extensive docs in the Asterisk source directory in this super secret directory called......"doc/" |
| 11:28.45 | Had | all i need is the passthru, i have spa2102 and sip provide with t.38 support |
| 11:29.38 | Had | I followed a few guides and tested it with couple of asterisk versions (1.4... and 1.6...) and still no joy... |
| 11:29.48 | manxpower | Had: Don't expect it to be easy, well documented, or even compatible between the SPA and the provider. It's a very new protocol and there are significant interop issues between implementations. |
| 11:30.12 | manxpower | Where do you enable T.38 in the SPA? |
| 11:30.35 | manxpower | ~mailinglist |
| 11:30.36 | infobot | [~mailinglist] The Asterisk mailing lists can be found at http://lists.digium.com , http://www.asteriskguru.com/archives Search the archives by adding "site:lists.digium.com" to your Google search. |
| 11:30.36 | Had | well SPA is linksys which is cisco and my provider use cisco gateways... |
| 11:30.49 | Had | it is in line setting |
| 11:31.03 | Had | admin login - advance - voip - line 1 |
| 11:31.30 | manxpower | That's so cute. Thinking two devices with T.38 support from the same company will work togather. |
| 11:32.01 | Had | well I also have zyxel ATA with t.38 support... |
| 11:32.12 | manxpower | Had: Check the mailinglist archives. T.38 comes up there fairly often. |
| 11:33.12 | Had | I do know those sites, spent my last couple of weeks browsing through them... |
| 11:33.58 | manxpower | Had: then you know more about it than anyone on this channel. 8-( |
| 11:34.28 | Had | :( |
| 11:34.37 | oberon | how do I make a call manually in the CLI using the SIP account I setup ? |
| 11:34.48 | oberon | .. just to see that it works |
| 11:35.03 | box2 | i don't know of a CLI softphone |
| 11:35.03 | cidu | hrmmm, i cant seem to get my sangoma card to work properly on a new box, works fine on the old one....grrr, lost |
| 11:35.34 | oberon | I dont have a softphone |
| 11:35.41 | *** join/#asterisk unasi7 (n=unasi7@80-218-32-110.dclient.hispeed.ch) |
| 11:35.53 | oberon | I wanna make a call manually |
| 11:36.05 | oberon | and then make another call manually |
| 11:36.06 | manxpower | oberon: Get a softphone. You'll never figure out .call files until you use Asterisk a little. |
| 11:36.12 | oberon | .. and then connect the 2 calls |
| 11:36.17 | oberon | .. in the CLI |
| 11:36.21 | manxpower | cidu: Sangoma support is VERY good. |
| 11:37.03 | oberon | manxpower, the server is configured only to make outgoing cals, I believe a softphone wont be able to connect to it |
| 11:37.12 | oberon | *calls |
| 11:37.15 | manxpower | oberon: I cannot help you further. |
| 11:42.49 | cidu | yes |
| 11:43.07 | cidu | but they dont work at any of the times hen i can bring down the system to test the new box :( |
| 11:43.36 | cidu | and i agree, they are very good, just cant have the whole cell system down during the day ...meh |
| 11:43.54 | manxpower | cidu: That makes it much harder to fix the problem. 8-( Your best bet is to hope there is someone on this channel that uses Sangoma. |
| 11:44.18 | cidu | aye |
| 11:44.26 | manxpower | <-- is a big fan of Sangoma, but have not used their cards in about a year. |
| 11:44.36 | cidu | what have you been useing? |
| 11:45.00 | manxpower | Whatever the client uses. |
| 11:46.07 | cidu | keep getting dchannel alrm on the far side no matter what i change, hrmm |
| 11:46.08 | manxpower | Most of the issues I've helped customers with in the past couple of years are not PSTN related, just dialplan, .call file, SIP, etc issues. |
| 11:46.14 | cidu | ahhh |
| 11:46.15 | cidu | yeah |
| 11:46.45 | cidu | i know its not the cable because it works just fine with same cables with card in other computer |
| 11:46.46 | manxpower | <-- does NOT work for Digium |
| 11:51.16 | *** part/#asterisk manxpower (n=EWieling@69.73.94.162) |
| 11:54.59 | *** join/#asterisk yziquel (i=53acc979@gateway/web/freenode/x-6e7c52cccd0358fd) |
| 11:58.02 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 11:58.16 | *** join/#asterisk seanbright (n=sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 12:00.50 | SALstar | I have some feature enhancement requests. :) |
| 12:01.01 | *** part/#asterisk SALstar (i=ondrejj@work.salstar.sk) |
| 12:01.39 | box2 | well, those requests don't seem unreasonable at all |
| 12:01.54 | *** join/#asterisk thisismyname66 (n=quassel@rkom.r-kom.de) |
| 12:03.22 | thisismyname66 | hi there |
| 12:03.45 | thisismyname66 | anyone up here? |
| 12:04.00 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 12:07.21 | DuPe | well hrmmm does anyone wanna make some $? :) im fighting with errorstrying to get a softphone to dial a 2nd asterisk box connecting via iax |
| 12:07.45 | *** join/#asterisk thisismyname (n=quassel@rkom.r-kom.de) |
| 12:07.59 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:08.10 | DuPe | bashes my head against the wall |
| 12:08.37 | thisismyname | hmm... people bash their head against walls... but noone is talking.. |
| 12:08.38 | *** join/#asterisk beek (n=klinebl@pdpc/supporter/professional/beek) |
| 12:08.56 | [TK]D-Fender | sits quietly amused... |
| 12:09.03 | DuPe | ya that too |
| 12:09.17 | thisismyname | u got a little problem? |
| 12:09.20 | thisismyname | :) |
| 12:09.24 | thisismyname | that u cant fix |
| 12:09.53 | DuPe | yeah :< and being sleepy doesnt help |
| 12:10.56 | thisismyname | but the thing with walls and ur head is better? |
| 12:11.37 | [TK]D-Fender | ~osmosis |
| 12:11.38 | infobot | [~osmosis] Osmosis is the act of beating yourself on the head repeatedly with THE BOOK, until some measure of absorption has occured ... or at least until your unconsciousness restores peace to the channel ... |
| 12:11.41 | [TK]D-Fender | ^^^^^^^^^^^^ |
| 12:12.00 | *** join/#asterisk aethelrick (n=richard@mail.mexuar.net) |
| 12:12.05 | thisismyname | :) |
| 12:13.45 | aethelrick | hi all, I have some code that issues a Redirect command through AMI (which all works). My question is, can I set some custom channel variables (using AMI) on the channel being redirected? |
| 12:14.39 | *** join/#asterisk vlt (n=dm@suez.activ-job.com) |
| 12:17.37 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 12:19.53 | thisismyname | anyone here with some knowledge about Capi an the cologne E! card? |
| 12:19.54 | [TK]D-Fender | Yeah, thre's patience for you... |
| 12:19.56 | thisismyname | E1 |
| 12:20.15 | thisismyname | or.... only CAPI-knowledge |
| 12:20.19 | thisismyname | ;) |
| 12:20.55 | fiddur | If I were to professionally go with softphones and headsets; what headsets would I then chose? |
| 12:25.01 | *** join/#asterisk jeff_phillips (n=jeff_phi@209-142-149-133.stat.centurytel.net) |
| 12:25.05 | jeff_phillips | good morning |
| 12:25.59 | jeff_phillips | picking back up where I left off on this timeclock app last wednesday -- was stuck in another building thurs & fri. |
| 12:26.11 | jeff_phillips | I got it so I can dial the extension, but it doesn't seem to be connecting to the mysql database |
| 12:26.13 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 12:27.06 | ariel_ | Morning |
| 12:27.18 | jeff_phillips | It's coming back with null as a connid. http://pastebin.ca/1523845 |
| 12:30.03 | [TK]D-Fender | fiddur: Hideous choice... |
| 12:31.50 | [TK]D-Fender | jeff_phillips: I don't see that NoOp being executed |
| 12:32.11 | [TK]D-Fender | jeff_phillips: Nor do I actually see it in the context part of your PB |
| 12:32.39 | [TK]D-Fender | jeff_phillips: and wher you did seem to bolt it in is using "|" where you should be using "," |
| 12:32.42 | fiddur | [TK]D-Fender: I think I have a hard time understanding why a small phone like snom 300 running linux would be SO much better than my oversized quadcore workstation... |
| 12:32.53 | ariel_ | jeff_phillips: is that post you did actually working? they way it's post. first line is not corret there both in the smae lne also error is not in the context |
| 12:33.11 | [TK]D-Fender | jeff_phillips: Actually I see ti now.. looks like a LF fail <- |
| 12:34.13 | [TK]D-Fender | fiddur: Because putzing around with some shitty little app that steals focus away from apps I'm working on would drive me crazy. then add the fact you have to hope the sound card doesn't suck <- This happens. |
| 12:34.38 | box2 | fiddur: and because your computer is volatile compared to a phone |
| 12:34.49 | box2 | hard drive crash, no phone |
| 12:35.05 | jeff_phillips | [TK |
| 12:35.10 | [TK]D-Fender | jeff_phillips: You are "autofallthrough" because your lack of LF means you don't even have a "2" priority |
| 12:35.27 | jeff_phillips | [TK]D-Fender: sorry I didn't write the code, don't know why | was used. |
| 12:35.32 | box2 | uninstall your codecs by accident or install things that ruin your sound configuration |
| 12:35.40 | [TK]D-Fender | jeff_phillips: Apparently didn't look at it either.. |
| 12:36.45 | [TK]D-Fender | fiddur: And the price of a good headset quickly approaches the cost of a decent entry-level phone |
| 12:37.44 | fiddur | [TK]D-Fender: Ok, some valid points... This would mainly be for people in the office not using phones that often.. |
| 12:38.01 | *** join/#asterisk jkroon (n=jkroon@dsl-240-162-164.telkomadsl.co.za) |
| 12:39.27 | jeff_phillips | I'm not seeing where the |'s even came from. |
| 12:39.59 | jkroon | if you find out, please let me know. |
| 12:40.38 | jeff_phillips | i see ,'s in the conf files, not |'s |
| 12:40.50 | [TK]D-Fender | jeff_phillips: parsed that way actually because of your FL fail <- |
| 12:40.56 | [TK]D-Fender | LF* |
| 12:40.57 | jkroon | anyway, the d flag to Queue(), the only thing I can find is that it's something to do with minimum delay (data-quality (modem) call) from the description, but I seem to be unable to find an actual explanation of how exactly it's handled differently. |
| 12:41.10 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 12:41.22 | jkroon | jeff_phillips, I suspect the |s comes from older (<1.2) versions of asterisk. |
| 12:41.34 | [TK]D-Fender | jkroon: IIRC something to do with more direct rebridging of the calls removing App_queue from the mix |
| 12:41.56 | jkroon | would you want to enable it for normal voice calls? |
| 12:42.02 | jkroon | does it make sense? |
| 12:42.12 | [TK]D-Fender | jeff_phillips: I've said it 3 times... seriously.. go caffeinate. |
| 12:42.13 | jeff_phillips | jkroom: i'm using 1.4.20 |
| 12:42.22 | jkroon | jeff_phillips, use ,s. |
| 12:42.31 | [TK]D-Fender | jeff_phillips: LINE FIFTEEN |
| 12:42.44 | [TK]D-Fender | *ARGH* |
| 12:43.07 | jkroon | lol @ fender. at least when you bliksem me with a lead pipe in some direction i usually go look :p. |
| 12:43.08 | jeff_phillips | LF=line fifteen? what? |
| 12:43.22 | [TK]D-Fender | jeff_phillips: LINE FEED |
| 12:43.42 | [TK]D-Fender | jeff_phillips: You put 2 extens WITHOUT HITTING FRIGGEN ENTER |
| 12:43.58 | *** join/#asterisk Shazaum (n=aaaaaaaa@unaffiliated/shazaum) |
| 12:44.03 | [TK]D-Fender | jeff_phillips: module load res_coffee.so |
| 12:44.06 | jeff_phillips | oh, i didn't do that. matthew did |
| 12:44.12 | jeff_phillips | thanks though |
| 12:44.39 | box2 | soda soda |
| 12:45.59 | jkroon | does the coffee thing too. |
| 12:46.16 | thisismyname | goes and gets a coffee, too |
| 12:46.24 | jeff_phillips | maybe i should... i just never cared for the taste. i'm an orange juice guy |
| 12:46.38 | jkroon | then go get some vitamin C |
| 12:46.50 | jeff_phillips | thanks now it answers and asks if I want english or spanish and then hangs up. |
| 12:46.56 | jeff_phillips | progress. :) |
| 12:50.51 | thisismyname | so, anyone here familiar witbh CAPI?? |
| 12:51.19 | jkroon | thisismyname, more so than I'd like to be, which doesn't say much about my knowledge either. |
| 12:51.50 | *** join/#asterisk malaiwah (n=mbelleau@host-64-47-115-5.masergy.com) |
| 12:52.07 | thisismyname | i just cant find out how to configure my /etc/isdn/capi.conf |
| 12:52.17 | thisismyname | with a junghanns.NET single E1 |
| 12:52.27 | thisismyname | cologne HFC-E1 is the chipset.. |
| 12:53.12 | jkroon | tries to recall where he's using CAPI. |
| 12:53.37 | jkroon | oh yes, Diva card ... using some custom drivers that happen to present a CAPI capable interface. |
| 12:53.58 | jkroon | cologne ... that sounds more like a misdn supported card? |
| 12:54.46 | *** join/#asterisk coppice (n=chatzill@26.168.17.210.dyn.pacific.net.hk) |
| 12:58.40 | jeff_phillips | hmm, now it doesn't seem to care whether I press any DTMF keys at its prompts. http://www.pastebin.ca/1523857 |
| 12:59.01 | jeff_phillips | just repeats the message twice asking english or spanish, then immediately hangs up regardless of pressing anything |
| 13:05.27 | jeff_phillips | i'm beginning to think this timecard app i found really sucks |
| 13:06.24 | [TK]D-Fender | jeff_phillips: Written by an incompetant schmuck who clearly doesnt' know how to make an IVR, and debugged by someone of similar background... |
| 13:07.23 | jeff_phillips | lol. |
| 13:08.54 | jkroon | jeff_phillips, what do you need it to do? |
| 13:09.07 | jkroon | and why not cook something that does exactly what you need? |
| 13:09.46 | [TK]D-Fender | jeff_phillips: that is the most craptastic pile of dialplan I've seen in a while... |
| 13:09.46 | *** join/#asterisk shido6 (n=shido6@74-132-202-71.dhcp.insightbb.com) |
| 13:10.03 | jeff_phillips | jkroon: Well I had found this page http://www.asterisktimeclock.org/ and thought I'd give it a try. |
| 13:10.30 | [TK]D-Fender | jeff_phillips: Mis-ordered, using pattern indicators where none is needed, not doing anything right as far as IVR functionality is concerned. |
| 13:11.15 | jkroon | jeff_phillips, anything that doesn't tell me on the homepage what the purpose of the thing is doesn't deserve a second look. |
| 13:11.54 | jeff_phillips | it's suppose to allow you to clock in and out through an IVR, with the back end run by phpTimeclock -- a web based timeclock app |
| 13:11.57 | jkroon | project info is a tad more useful. yes, decent goal if you want to be a draconian on your employees. |
| 13:12.26 | jkroon | hmm, only other use is to enable secretaries/receptionists to know who's in/out. |
| 13:12.34 | jkroon | so actually maybe not a bad idea :p |
| 13:12.56 | jeff_phillips | well the issue is the employees are working at two different buildings now and going back and forth as needed. Time clock is only in one building. |
| 13:13.15 | jeff_phillips | we were just punching in and then driving to the other building until some secretary decided to throw fits about people actually being paid for their time to get there |
| 13:13.22 | *** join/#asterisk dwayne (n=dwayne@76.29.245.9) |
| 13:14.00 | jeff_phillips | i find that if I hitch a ride with her husband, I get paid for the drive time though, but if I drive myself and spend my own gas, I don't get paid for the time nor the gas. Hmm. |
| 13:14.51 | ariel_ | jeff_phillips: it's a good Idea, just need to bring up the code to 1.4 or 1.6 level. |
| 13:17.35 | jeff_phillips | i'm not seeing how it even worked at all for the guy who wrote it |
| 13:18.04 | ariel_ | jeff_phillips: then it's time to re-write |
| 13:18.11 | jeff_phillips | i guess so |
| 13:18.12 | jkroon | jeff_phillips, usually the person that writes it has some assumptions about the surrounding system which he doesn't make clear. |
| 13:18.24 | jkroon | it can't be that difficult to do. |
| 13:18.45 | *** join/#asterisk Shazaum (n=aaaaaaaa@unaffiliated/shazaum) |
| 13:18.45 | jeff_phillips | well first off I'm gonna nix out all the spanish stuff. |
| 13:19.03 | jeff_phillips | ... watch next week we'll hire a spanish speeking employee just to spite me. |
| 13:19.10 | Chainsaw | :D |
| 13:19.21 | ariel_ | jeff_phillips: what you posted it's missing allot of clean up and i t sections it looks more like work in progress then actual ready code. |
| 13:19.46 | *** join/#asterisk TimRiker (n=timr@bzflag/projectlead/TimRiker) |
| 13:19.59 | jeff_phillips | ariel_: I kind of noticed that about his website too |
| 13:20.27 | ariel_ | jeff_phillips: then I feel it's time for you to hire someone to finish this up for you then. |
| 13:21.22 | jkroon | jeff_phillips, if the rest of you are english then he'll just have to adapt. |
| 13:21.24 | box2 | box2 sounds like a professional guy, you should hire him |
| 13:22.20 | jkroon | yes, different website for every nitty project. |
| 13:22.48 | jeff_phillips | lol, well you're probably right. but at the same time I should probably learn more by tackling it on my own. If it doesn't work or becomes a back burner project, that's okay. I'll just ride to the other building with the secretary's husband so I get paid for my time. :) |
| 13:23.47 | ariel_ | jeff_phillips: time sheets, and emails also works just talk to there hr |
| 13:23.59 | jeff_phillips | we don't have an HR |
| 13:24.11 | ariel_ | yes you do |
| 13:24.45 | ariel_ | even when I was a one man shop I had HR, me, myself and I....hehehe there is always someone incharge of payroll |
| 13:24.49 | jeff_phillips | small family owned business. owner's daughter in law thinks she is in charge of everything, is really just a secretary. owner is too much of a pansie to put his foot down on anything |
| 13:25.31 | ariel_ | jeff_phillips: it's called communications with them.... |
| 13:25.33 | box2 | he can flyyyyyy! |
| 13:25.56 | jeff_phillips | indeed. something we seriously lack around here |
| 13:26.03 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:26.03 | jeff_phillips | i just like the free trips to mcdonalds now |
| 13:26.17 | jkroon | jeff_phillips, you found your HR person. go speak with HIM. until he acts, or points you to the right person :p. |
| 13:26.28 | jeff_phillips | he wont |
| 13:26.54 | jkroon | hmm, sounds like the reason(s) i no longer work for a boss. |
| 13:27.14 | ariel_ | bbl going to the lab need to test sip packetization=60 command that we found in the code... |
| 13:27.21 | jeff_phillips | the owner's reaction to his daughter in law bitching about people getting paid to drive to the warehouse was to respond by telling everyone to start at their shift start time at the warehouse. So his son who is "in charge" over there got irritated that he now had to show up at his shift start time which makes him wake up 30 minutes earlier. |
| 13:28.00 | jeff_phillips | so the result is now he lets us all go to mcdonalds for breakfast because he feels the change cut into the time he would be spending eating breakfast at home |
| 13:28.03 | ariel_ | sound like a management issue |
| 13:28.17 | jeff_phillips | of course it's a mismanagement issue. this company consists entirely of those. |
| 13:28.35 | box2 | management from someone who can fly |
| 13:28.48 | box2 | that should be a loadable module |
| 13:29.08 | jeff_phillips | they have me dowing tooling inventory in the warehouse two days a week now because apparently the other people they hired can't read and write numbers properly... |
| 13:29.15 | jeff_phillips | (its sadly true) |
| 13:29.43 | jeff_phillips | Well friday they had 5 people hassling each other about finding tooling for a perticular part we got an order on -- nobody asked me, as I knew what was there and what wasn't. |
| 13:30.07 | jeff_phillips | after 2 hours of those fools wasting everyone's time they ask me, I say it isn't here, ask the production manager, and he has it sitting right there in his work area because it's a part that is commonly re-ordered |
| 13:30.23 | jeff_phillips | so yeah, nobody asked the production manager if he even needed this thing that they were all looking for to try to get for him |
| 13:30.33 | *** join/#asterisk ruben23 (n=RPL@122.55.48.243) |
| 13:30.35 | box2 | void decision( string issue ) { if(true) flyAway(); } |
| 13:30.40 | ruben23 | hi |
| 13:30.54 | jeff_phillips | box2: I've seriously been thinking about doing just that |
| 13:31.02 | box2 | lol |
| 13:35.23 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
| 13:39.34 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 13:41.08 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:42.52 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 13:42.52 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:46.24 | *** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 13:46.58 | *** join/#asterisk Faustov (i=user@gentoo/user/faustov) |
| 13:48.00 | Faustov | hi, any idea why an incoming pots call redirected to a local ivr (sangoma, dahdi, *-1.6) ends with the following result: the ivr gets executed (can see that in the cli), however the caller can hear indication as if no one has answered, no ivr |
| 13:48.51 | DuPe | codec issue/filenames is my first thought but i'm no expert |
| 13:49.18 | [TK]D-Fender | Faustov: PASTEBIN the failed call and it might help to know what protocol is used.. |
| 13:49.21 | Faustov | i've set it to alaw and I'm from europe... also the same ivr works from any other voip call |
| 13:49.29 | Faustov | hmm |
| 13:50.03 | Faustov | [TK]D-Fender: problem is over the "failed" call there are no warnings or errors in the console with a fairly high verbosity level (15) |
| 13:50.16 | Faustov | i'll try to get one in a moment... |
| 13:51.03 | *** join/#asterisk s14ck (n=s14ck@ccscliente156.ifxnetworks.net.ve) |
| 13:52.29 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 13:53.18 | *** join/#asterisk afink (n=chatzill@204.26.87.226) |
| 13:53.52 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 13:55.39 | jeff_phillips | well that's nice. I skipped the english/spanish prompt and now it goes through all the motions of letting you clock in or out, but it doesn't actually put anything in the database |
| 13:55.39 | *** join/#asterisk xuser (n=xuser@unaffiliated/xuser) |
| 13:56.11 | *** join/#asterisk dajhorn (n=dajhorn@206.16.96.160) |
| 13:56.34 | oberon | what softphone would you recommend using on linux ? |
| 13:57.17 | viraptor | oberon: twinkle, ekiga |
| 13:59.19 | *** join/#asterisk ccesario (n=ccesario@189-19-6-236.dsl.telesp.net.br) |
| 14:00.58 | box2 | i'm a fan of linphone myself |
| 14:01.02 | jkroon | ok, AgentCallbackLogin was removed in asterisk 1.6 ... replacement mechanism? |
| 14:01.58 | *** part/#asterisk jkroon (n=jkroon@dsl-240-162-164.telkomadsl.co.za) |
| 14:02.56 | [TK]D-Fender | Yup, ANother patient person... |
| 14:03.21 | *** join/#asterisk TimRiker (n=timr@bzflag/projectlead/TimRiker) |
| 14:04.03 | [TK]D-Fender | TimRiker: ! ! ! |
| 14:05.19 | Faustov | [TK]D-Fender: http://pastebin.com/m63e7538f |
| 14:05.23 | *** join/#asterisk jkroon (n=jkroon@dsl-240-162-164.telkomadsl.co.za) |
| 14:05.41 | [TK]D-Fender | jkroon: "core show applications like queue" <- |
| 14:06.04 | *** join/#asterisk spikie (n=spikie@ven69-2-82-241-121-171.fbx.proxad.net) |
| 14:06.55 | [TK]D-Fender | Faustov: What kind of link? |
| 14:06.58 | jkroon | [TK]D-Fender, thanks. makes sense, AddQueueMember ... slightly more work, but it'll get the job done I guess. |
| 14:07.18 | Faustov | [TK]D-Fender: analog via sangoma A200d |
| 14:08.02 | [TK]D-Fender | Faustov: What zone for lines? |
| 14:08.26 | vlt | Hello. Can I use ztdummy w/o kernel module rtc? |
| 14:08.48 | jkroon | makes penalties harder too ... |
| 14:09.27 | *** join/#asterisk JerJer (n=PhatJ@asterisk/original-h323-guy/JerJer) |
| 14:09.31 | Faustov | [TK]D-Fender: Europe, if that's what you mean |
| 14:09.42 | Faustov | all lines here should use the alaw |
| 14:09.54 | [TK]D-Fender | Faustov: I'd be sure your indications and zones are set right for it... |
| 14:10.03 | [TK]D-Fender | Faustov: And analog doesn't have a codec |
| 14:10.07 | JerJer | is there a way to make FastAGI send all channel variables it knows about when it initiates the request ? |
| 14:10.20 | Faustov | WAIT |
| 14:10.21 | *** join/#asterisk Katty (n=asterisk@mail.copi-rite.com) |
| 14:10.24 | Katty | :> |
| 14:10.29 | [TK]D-Fender | Katty: Mew |
| 14:10.30 | Faustov | stupid wancfg |
| 14:10.34 | Katty | [TK]D-Fender: herroes. |
| 14:10.34 | JerJer | i set custom variables, both regular and CDR |
| 14:10.45 | Faustov | it overwrote my configs |
| 14:10.45 | Strogg | hrmm I just discovered I can't dialout on my asterisk box.. hrmm |
| 14:10.49 | [TK]D-Fender | JerJer: Send? Send Where? |
| 14:11.05 | Faustov | [TK]D-Fender: i got loadzone=us and defaultzone=us |
| 14:11.13 | Faustov | would that be the case? |
| 14:11.16 | Strogg | when I start asterisk in the foreground, I get this error... "asterisk: symbol lookup error: /usr/lib/asterisk/modules/chan_sip.so: undefined symbol: ast_pickup_ext" Anyone know what I need to do to fix that? |
| 14:11.17 | [TK]D-Fender | JerJer: FastAGI jsut calls AGI, and its up to the AGI to request whatever vars it wants |
| 14:11.27 | JerJer | [TK]D-Fender: when [Fast]AGI starts it sends the 'standard' variables via STDIN |
| 14:11.27 | [TK]D-Fender | Faustov: Doesn't sound right to me... |
| 14:11.38 | Strogg | or rather, when I start it in the foreground and try dialing out. asterisk dies with that error. |
| 14:11.50 | JerJer | [TK]D-Fender: was hoping to do push rather than pull |
| 14:11.51 | Katty | GOOD MORNING ALL YOU WONDERFUL PEOPLE! |
| 14:11.55 | [TK]D-Fender | JerJer: It sends minimal Call leve (* standard), and that is all |
| 14:12.14 | JerJer | suckage :( |
| 14:12.15 | [TK]D-Fender | JerJer: "vi app_fastagi.c" <- |
| 14:12.34 | JerJer | there isn't budget for development in this project |
| 14:13.09 | Strogg | Ah! I found it.. nevermind me. heh |
| 14:13.09 | *** join/#asterisk Rico29 (n=Rico@lns-bzn-48f-81-56-220-28.adsl.proxad.net) |
| 14:13.21 | Katty | there's never a budget for anything, is there. |
| 14:13.32 | Katty | next we'll be budgeting off smiles and hugs. |
| 14:13.33 | Faustov | [TK]D-Fender: I can't find what other zones can I set - should those generally match the indications? |
| 14:13.47 | [TK]D-Fender | Faustov: Probably... |
| 14:14.07 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 14:17.15 | *** join/#asterisk FinboySlick (n=shark@207.134.8.4) |
| 14:18.04 | Katty | i has a query. |
| 14:18.18 | Katty | let's say you have an asterisk server ona public ip, like ya do, with port 5060 and rpt ports open. |
| 14:18.31 | FinboySlick | I just had a 'strange' but normal request from my boss. He'd like our receptionist to be able to know who's busy before transfering. Is this something that relates more to the phone, or is it something that asterisk ought to be taking care of? |
| 14:18.45 | Katty | and you have a polycom, at a house, that you want to use, but you don't want to leave port 5060 open to the entire world. now let's also assume that you can't get a static IP at your house. is it possible do some sort of proxy server setup? |
| 14:18.54 | [TK]D-Fender | FinboySlick: * doesn't take care of things. You control what you do |
| 14:19.10 | synthetic | the phone and asterisk has ability to light lamps if the eprson is busy |
| 14:19.20 | [TK]D-Fender | Katty: Remote phones don't need forwarding. |
| 14:19.44 | [TK]D-Fender | Katty: Generally no need for a proxy either |
| 14:19.59 | *** join/#asterisk Maximo (n=maximo@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 14:20.01 | [TK]D-Fender | Katty: And if you feel particularly paranoid you can always run the phone on a different port |
| 14:20.04 | Katty | [TK]D-Fender: i am not simply going to forward port 5060 at the firewall |
| 14:20.15 | Katty | [TK]D-Fender: i want to allow it from a single IP |
| 14:20.20 | [TK]D-Fender | Katty: You don't forward for remote phones period. |
| 14:20.36 | Katty | [TK]D-Fender: i must not understand. if i only allow port 5060 access from certain IP address....that does not parse. |
| 14:20.47 | FinboySlick | [TK]D-Fender: Ah sorry, what would be the more common way for self-empowered people to handle this type of situation? Some sort of switchboard phone? |
| 14:20.51 | Katty | [TK]D-Fender: the firewall must see that the remote IP is allowed to go through the firewall |
| 14:20.57 | Katty | [TK]D-Fender: else fire wall says SOD OFF |
| 14:21.17 | [TK]D-Fender | Katty: Make a separate peer. And the address is dynamic anyway as you jsut said |
| 14:21.39 | [TK]D-Fender | FinboySlick: Depends how you want to process things |
| 14:21.46 | *** join/#asterisk idi (n=idi@78.33.22.254) |
| 14:22.24 | [TK]D-Fender | FinboySlick: Phone with presence support. sit in * CLI and monitor. Make some nice status view. Use FOP. CTV camera spying on your users, etc |
| 14:22.25 | FinboySlick | [TK]D-Fender: I'm imagining the alternative PC/web-based switchboard application? |
| 14:22.32 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 14:23.43 | [TK]D-Fender | FinboySlick: Is that a QUESTION? |
| 14:25.10 | idi | anyone able to offer me a little insight as to why sip service appears to crash randomly on incoming sip call, with no errors in logs or debug? |
| 14:25.48 | Faustov | [TK]D-Fender: changed default indication to pl and set also pl in loadzone and defaultzone, restarted asterisk and wanrouter and still nothing... |
| 14:26.01 | JerJer | [TK]D-Fender: is it possible to pull CDR variables via AGI ? |
| 14:26.16 | JerJer | like CDR(duration) |
| 14:26.35 | [TK]D-Fender | JerJer: AGI = dialplan in a language of your choice.... |
| 14:26.37 | JerJer | most likely via DeadAGI, if that helps |
| 14:26.37 | FinboySlick | [TK]D-Fender: Ok, FOP looks like it might be shat I need. |
| 14:26.51 | JerJer | [TK]D-Fender: i very much know what it is |
| 14:26.55 | [TK]D-Fender | Yes... FOP was certainly "shat" |
| 14:27.04 | [TK]D-Fender | JerJer: So the answer is clearly "yes" |
| 14:27.12 | JerJer | ok - how do i get CDR(duration) ? |
| 14:27.17 | JerJer | via agi |
| 14:27.26 | JerJer | get variable returns nothing |
| 14:27.44 | JerJer | get full variable returns a literal 'CDR(duration)' |
| 14:28.01 | [TK]D-Fender | JerJer: because tis a FUNCTION, not a variable. get a little more creative |
| 14:28.20 | *** join/#asterisk jsjc (n=j@219-90-165-225.ip.adam.com.au) |
| 14:28.22 | JerJer | ok so h exten time |
| 14:28.26 | jsjc | hello! |
| 14:29.15 | jsjc | I am looking to get into asterisk and I am building a new machine for the office so I was thinking should I use PCI cards or will be better to get gateways? |
| 14:29.27 | *** join/#asterisk c0rnoTa (n=c0rnoTa@80.251.113.56) |
| 14:29.33 | Faustov | span sections in chan_dahdi are not meant for analog, right? |
| 14:29.38 | [TK]D-Fender | jsjc: Describe your needs |
| 14:29.49 | [TK]D-Fender | Faustov: they are not. |
| 14:30.31 | Faustov | thanks |
| 14:30.39 | Faustov | looking at a system.conf sample file... |
| 14:31.45 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 14:31.45 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:31.50 | jsjc | [TK]D-Fender want a system that can be upgradeable, I am building a little computer (begining looking for a small thing but now needing PCI....). At the moment around 2FXO and 6FXS but this is to start playing with in a future most likely will get bigger |
| 14:32.16 | jsjc | will have some VOIP lines as well |
| 14:32.17 | *** join/#asterisk lowtek (n=anonymou@mail.heavylogic.com) |
| 14:32.39 | [TK]D-Fender | jsjc: what kind of line expansion are your predicting? |
| 14:32.53 | *** join/#asterisk TSM (n=the_soft@fw-lon1.wenn.com) |
| 14:32.58 | TSM | is it possable to setup a trunk, so that once the call is finished, it can update an entry in the DB with how many seconds the call was? |
| 14:33.14 | [TK]D-Fender | TSM: Its called CDR and * does this already |
| 14:33.54 | idi | any ideas as to why sip service appears to restart randomly on incoming sip call, with no errors in logs or debug? |
| 14:34.10 | TSM | [TK]D-Fender: i know that, i need to create an app that can check how many mins have been used in a certian period to decide if i can use a particular route |
| 14:34.17 | jsjc | [TK]D-Fender what you mean but what kind of line expansion?? |
| 14:34.42 | [TK]D-Fender | jsjc: You're starting with 2 lines. What would your expansion expectations be? |
| 14:34.53 | TSM | [TK]D-Fender: i can use the CDR data, but i dont know how to pipe out information to an external app and then make decisions on its return data |
| 14:35.01 | [TK]D-Fender | TSM: AGI <- |
| 14:35.09 | *** join/#asterisk TimRiker (n=timr@bzflag/projectlead/TimRiker) |
| 14:36.04 | jsjc | [TK]D-Fender at the moment looking to a budget start till I get a good understanding of asterisk. And "by the look" of it all this computer cards are ridiculously expensive for the type of electronics they have... |
| 14:37.22 | c0rnoTa | Hello, All! I can't hold it inside me, sorry. One of my customer bought Planet ATA. Connect it with FAX, set codec alaw -1, ulaw -2, gsm -3 etc in web interface of this device (there is no option to disable other codecs), I set alaw in peer option in sip.conf and disable any t38 udptl transfer |
| 14:37.38 | TSM | [TK]D-Fender: will AGI work directly in the extentions.conf? i thought it was just a remote access into asterisk |
| 14:37.43 | jsjc | [TK]D-Fender might end up having lines but up to a maximum of 6 I will say but there will be up to 10-15 phones |
| 14:38.09 | c0rnoTa | but when fax receive a call, it's couldn't receive a fax message |
| 14:38.22 | [TK]D-Fender | TSM: It is not and you don't appear to have done any reading on it. there are several chapters in The BOOK on this. Go read. |
| 14:38.22 | [TK]D-Fender | ~book |
| 14:38.23 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 14:38.24 | [TK]D-Fender | ^^^^^^^^^ |
| 14:38.51 | spikie | Hi I've got a question about IAX2 protocol? |
| 14:38.52 | [TK]D-Fender | jsjc: How long till your predict 4? And what about 6? |
| 14:39.03 | [TK]D-Fender | spikie: Do you? |
| 14:39.10 | TSM | i just realised that after i posted my reply, AGI().. |
| 14:39.29 | idi | i guess noone has any ideas to my issue then ;) |
| 14:39.32 | c0rnoTa | in sip debug messages for this peer I saw that planet accept only one codec |
| 14:39.46 | jsjc | [TK]D-Fender might be around a year... |
| 14:39.58 | [TK]D-Fender | jsjc: 1 year for 4? |
| 14:40.16 | spikie | yes. In fact during a call i don't understand why asterisk send a TXREL after that spam the iaxclient with IXREQ but I haven't ask any transfert...... |
| 14:40.50 | [TK]D-Fender | idi: You have nothing to show and apparently aren't volunteering even the most basic information. |
| 14:41.03 | jsjc | [TK]D-Fender i heard that gateways are cheaper than PCI cards, and of course if I need to build my computer/server/pbx with no pci I can get cheaper system as well without PCIs |
| 14:41.25 | jsjc | [TK]D-Fender yes around a year for 4 and year and a half for 6 |
| 14:41.42 | idi | erm well generally was waiting for a response? what info you needed etc? |
| 14:42.38 | [TK]D-Fender | idi: You offered nothing to respond to. What VERSION maybe. What are you interacting with? What OS? 32/64 bit? You know... something USEFUL. |
| 14:43.14 | [TK]D-Fender | jsjc: By default I'd suggest a sangoma B600d, but that caps at 4 lines. A400d would be my next choice. |
| 14:43.23 | idi | wow nice attitude. a simple response of "maybe, what version" etc would be nice |
| 14:43.42 | TSM | personaly i prefer the A series sangoma stuff, cant remember if the B series has lifetime warranty |
| 14:44.04 | [TK]D-Fender | idi: If you wanted help one would think you wouldn't make us fish for answers from you. |
| 14:44.10 | jsjc | [TK]D-Fender but what about the FXS?? |
| 14:44.20 | *** join/#asterisk TimRiker (n=timr@bzflag/projectlead/TimRiker) |
| 14:44.37 | DuPe | argh okie... can someone do me a favour and look at http://pastebin.ca/1523952 regarding an iax problem that 2 machines cant talk to each other? i've been staring at this simple problem for so long my head is foggy |
| 14:44.39 | *** join/#asterisk jmacz (n=mcorb@190.144.75.22) |
| 14:44.44 | [TK]D-Fender | jsjc: that should be done by external gateways... Linksys SPA are my normal recommendation |
| 14:44.50 | TSM | jsjc: look on the Sangoma website, their A400 cards have modules for FXO or FXS |
| 14:45.05 | idi | well its * v1.6.0.10 linux 32 bit, sip only install with freepbx 2.5.1.5 |
| 14:45.08 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com) |
| 14:45.17 | *** join/#asterisk luminblade (n=luminbla@rrcs-71-42-115-245.sw.biz.rr.com) |
| 14:45.22 | idi | and to be honest 1st time here, no info as to "approved" question format |
| 14:45.27 | idi | so dont need the attitude here |
| 14:45.36 | TSM | jsjc: ive got complete documentation on how the linksys SPA and sipua cards can be remote configed |
| 14:46.08 | TSM | idi: there are lots of people looking here, just post with your question and someone will reply, dont wait, just jump in |
| 14:46.09 | [TK]D-Fender | idi: And what are you interacting with? what do you get in SIP debug prior to crash? |
| 14:46.25 | *** join/#asterisk afink (n=chatzill@204.26.87.226) |
| 14:46.49 | jsjc | TSM but if I will end up having 10-15 phones then that means I will need 10-15 FXS ports if I am not wrong right?... (sorry this newbie questions as I said I am looking into get the system to start learning) |
| 14:47.02 | [TK]D-Fender | jsjc: I recommend against PCI FXS unless absolutely necessary |
| 14:47.14 | TSM | jsjc: their A600 card i think scales up to 24 ports on a single card |
| 14:47.20 | idi | TSM, maybe so but someone who not been here before, everyone does things differently and i didnt want to just "jump i" |
| 14:47.35 | [TK]D-Fender | jsjc: How many phons to start, how quick to expand? Have you considered getting proper independent SIP phones? |
| 14:47.41 | TSM | idi: now you know, thats how it works here :) |
| 14:48.04 | TSM | jsjc: what are you trying to convert from? |
| 14:48.11 | [TK]D-Fender | idi: People are more inclined to help when you make it easy for them. |
| 14:48.51 | idi | yeah and a simple "what is your setup/acknowledement" helps for new ppl, thats all im saying |
| 14:48.55 | idi | anyhooos |
| 14:49.09 | jsjc | TSM I am trying to convert from having 2 lines and 6 phones and fax connected without any pbx system to getting everything controlled by a pbx and introducing VOIP lines and skype gatway |
| 14:49.10 | TSM | what is the problem? |
| 14:49.11 | luminblade | any pointers or references on how to debug audio problems using g729 (as a pass-thru)? i've installed the open-source drivers, as well as licensed digium g729 licenses, and neither works (again this is pass-thru, so they aren't really being used). |
| 14:49.13 | box2 | jmp 80 |
| 14:49.18 | idi | how you mean "interacting with"? |
| 14:49.27 | TSM | jsjc: ooo fax, can be problematic |
| 14:49.49 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 14:49.52 | [TK]D-Fender | DuPe: http://www.voip-info.org/wiki/view/Asterisk+No+authority+found |
| 14:50.07 | jsjc | TSM I heard that Faxes can give some issues here and there but its not impossible right? |
| 14:50.12 | TSM | jsjc: ive not had much experience with fax, but you need to look at ATAs that have t38, SPA-8000 will do it |
| 14:50.19 | lowtek | luminblade: pastebin your sip.conf and extensions.conf relevant sections ... |
| 14:50.20 | [TK]D-Fender | luminblade: What kind of "problems"? |
| 14:50.49 | TSM | jsjc: sangoma guarantee that their FXO/FXS cards will do fax properly |
| 14:51.08 | [TK]D-Fender | idi: You said it crashes on SIP communications. communications what what service/device? |
| 14:51.18 | DuPe | <PROTECTED> |
| 14:51.33 | DuPe | i got pepsi though... not as good as coffee but will have to do |
| 14:51.37 | idi | right, crashes on incoming sip call from upstream ITSP |
| 14:52.10 | luminblade | lowtek: no audio, i get sip 183/180 messages, and occastionally i can hear someone if they answer, my guess is g729a vs. g729r8 or something, but i cannot really tell. |
| 14:52.14 | TSM | idi: do you need *1.6 ? did you compile it yourself or install from disto? |
| 14:52.18 | idi | there does not appear to be any sip debug data when its enabled |
| 14:52.20 | [TK]D-Fender | DuPe: Look how the names alternate in the sample... |
| 14:52.39 | idi | tsm: its a self complile |
| 14:53.04 | TSM | idi: any reason, have you tried a disto first? |
| 14:53.05 | idi | sip debug stuff is fine when its running, but there is no debug output when the crash happens |
| 14:53.07 | [TK]D-Fender | idi: verify that with a local device, and pastebin the complete CLI output of that test follwoed by an inbound call attempt |
| 14:53.09 | [TK]D-Fender | ~pb |
| 14:53.10 | 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 |
| 14:53.12 | [TK]D-Fender | ^^^^^^^^^^^^^^^ |
| 14:53.35 | TSM | ~infobot |
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| 14:53.45 | TSM | how can i get all the infobot info? |
| 14:53.47 | idi | i did use distro but wanted newer version as debian on 1.4 |
| 14:54.09 | idi | so clean install, fresh complile of 1.6.0.10 |
| 14:54.26 | [TK]D-Fender | TSM: You can't |
| 14:54.33 | TSM | :( |
| 14:56.25 | idi | [TK]D-Fender, problem is that its totally random, it works fine for hours then will just restart sip |
| 14:56.39 | idi | but only on incoming itsp sip call |
| 14:56.44 | [TK]D-Fender | idi: Can you clarify "restart SIP"? |
| 14:57.21 | idi | um best way i can put it, basically the incoming sip call fails and all internal registered phones loose registration |
| 14:57.29 | idi | it comes back up after 20-30 secons |
| 14:57.34 | idi | *seconds |
| 14:57.49 | idi | and is fine until next time |
| 14:57.59 | jsjc | I am so confused dont know what to buy this world its much bigger than what i thought... big price difference, lot of different solutions that every shop calls THE BEST... any place in the internet with a good man of asterisk and PBXing? |
| 14:57.59 | [TK]D-Fender | idi: But this is not a complete * restart? |
| 14:58.27 | idi | there is nothing in the logs to state an actual sip restart no, there is just nothing in the logs when it happens |
| 14:58.36 | [TK]D-Fender | jsjc: Nothing that gives scalability advice for your kind of situation... |
| 14:58.41 | idi | that is on vvv and sip debug |
| 14:58.46 | TSM | jsjc: everyone has their own pref of equipment, some is better than others but again diffrent experience |
| 14:58.50 | *** join/#asterisk bhodder (n=blake@142.166.111.222) |
| 14:59.02 | box2 | mine is better than all of yours |
| 14:59.10 | [TK]D-Fender | jsjc: You can already see a range of different devices that could be suitable, its a question of the most cost-effective in your case. |
| 14:59.43 | bhodder | Hi, has anyone successfully set up a customer controlled call forwarding dialplan? |
| 15:00.15 | box2 | customer controlled, yuck |
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| 15:00.30 | [TK]D-Fender | bhodder: Sure |
| 15:00.55 | bhodder | I have put an extension in and set of commands but it ask for the number but never seems to set it |
| 15:01.11 | [TK]D-Fender | bhodder: ok/fine/sure... |
| 15:01.12 | jsjc | so most of the people has PCI cards to control all the FXO/FXS ports?? because i have seen gateways are more cost effective when larger size |
| 15:01.15 | JerJer | [TK]D-Fender: the h exten gets fired before the CDR data is closed out |
| 15:01.25 | JerJer | thus billsec, duration, and end are not populated |
| 15:01.34 | [TK]D-Fender | jsjc: I always recommend SIP gateways for FXS unless faxing is required |
| 15:02.01 | TSM | jsjc: depends what you want, A600 cards will scale up to over 120 ports on 5xPCI slots but using only 1 PCI/PCIe interface |
| 15:02.03 | [TK]D-Fender | JerJer: So the information you seek does not even exist anywhere at the point of your call... |
| 15:02.04 | JerJer | so how creative do I gotta get? This project has a requirement not to use the cdr modules |
| 15:02.05 | luminblade | lowtek: [TK]D-Fender: here's a pastebin of my sip.conf and extensions.conf (relavent parts), my call is being controleld by adhearsion, which is executing a 'dial' command with a ring timeout of 45 seconds, no other parameters.... http://pastebin.com/d350dbcfe |
| 15:02.23 | idi | [TK]D-Fender, sry just reread ur last msg, no not a complete * restart, only sip is affected |
| 15:02.34 | JerJer | [TK]D-Fender: or i have to create a cdr module that will post to their damn SOAP api |
| 15:02.59 | [TK]D-Fender | luminblade: You have not described the actual problem, and please pastebin a failed call at verbose 10, SIP DEBUG enabled <- |
| 15:03.01 | [TK]D-Fender | ~pb |
| 15:03.01 | 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 |
| 15:03.02 | [TK]D-Fender | ^^^^^^^^ |
| 15:03.22 | [TK]D-Fender | idi: Ok, wanted to be sure.... this is indeed very strange |
| 15:04.31 | JerJer | [TK]D-Fender: the information exists cuz i can query it via the CLI... its simply not coming down this fast agi nightmare |
| 15:04.32 | idi | [TK]D-Fender indeed it is, its as tho it doesnt like summat from the itsp on initial connection, but why so random i dont get |
| 15:04.47 | idi | unfortunatly i dont have an alternate itsp to test with either |
| 15:04.52 | jsjc | So thinking what this solution looks like...: 1xPCI with my FXO and 1 FXS for the famouse FAX and then gateway for rest of FXS? that will make me need just a PCI! yey. Now I found this brand called openvox that is pretty cheap compared with rest... (too crappy?) |
| 15:05.00 | bhodder | http://pastebin.com/m21680e13 here is a pb of what i have but it does not seem to work like it appears it would |
| 15:05.10 | [TK]D-Fender | JerJer: What about in regular dialplan? |
| 15:05.18 | JerJer | i guess i can hack together a cdr manager disaster |
| 15:05.32 | [TK]D-Fender | idi: Setup ekiga.net and test with them. |
| 15:05.34 | JerJer | exten => h,n,Set(cdr_billsec=${CDR(billsec)}) |
| 15:05.35 | [TK]D-Fender | (free) |
| 15:05.42 | JerJer | "cdr_billsec=0") in new stack |
| 15:05.57 | TSM | openvox: ok, makesure you get the hardware echo cancel stuff, bit more but better, still i think sangoma stuff is better, A series cards have lifetime warranty |
| 15:05.58 | idi | actually, i lie, do have an alternate for incoming |
| 15:05.58 | JerJer | h is fired before CDR is closed out |
| 15:06.17 | idi | back to making calls to get it to do it again |
| 15:06.45 | JerJer | unless i am setting it wrong |
| 15:06.46 | TSM | jsjc: openvox, ok, makesure you get the hardware echo cancel stuff, bit more but better, still i think sangoma stuff is better, A series cards have lifetime warranty |
| 15:06.48 | [TK]D-Fender | bhodder: that is FreePBX crap... |
| 15:06.48 | jsjc | or even having instead of a gateway for FXS every new phone bought been a SIP one... |
| 15:06.58 | JerJer | i think that's the right way |
| 15:07.00 | bhodder | oh ok |
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| 15:07.33 | TSM | jsjc: for the phones, yes get SIP phones, much easier, poly 330, abt £85+VAT GBP, duno about other countries |
| 15:07.35 | [TK]D-Fender | bhodder: And that dialplan means precisely NOTHING when you consider that what it sets needs to be processed by the dialplan you use to place your calls to devices <- |
| 15:08.10 | luminblade | here's a pastebin of a 'failed' call. the problem is really there is no audio during the ringing (from the 183/180 messages until answer): http://pastebin.com/d49756473 |
| 15:08.26 | bhodder | ya true |
| 15:08.32 | [TK]D-Fender | bhodder: Like your car is dead on the side of the side of the road and you're saying you put gas in your car. It is however in a jung in the BACK SEAT, and not in the tank. |
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| 15:08.44 | jsjc | SIP phones makes it easier then... mhm one more option to consider heheh |
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| 15:09.39 | bhodder | ok |
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| 15:11.12 | bhodder | Can you point me in the direction of correctly doing this |
| 15:11.20 | JerJer | go left |
| 15:11.38 | [TK]D-Fender | bhodder: what you showed is the FRONT end of things... you need to actually loko at what is set before placing your dial <- |
| 15:12.32 | jsjc | SIP phones need power supply tho... there is a massive array of options. I am amazed... |
| 15:13.35 | [TK]D-Fender | jsjc: Thats the advantage of this. |
| 15:13.48 | [TK]D-Fender | jsjc: A million ways to skin a cat |
| 15:13.50 | TSM | jsjc: yes, some come with supplys, the poly 330 dont, the poly 550 do, or you can buy supplies seperatly, can also get small POE switches |
| 15:14.02 | beek | [TK]D-Fender: Why would you want to skin a cat? |
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| 15:14.26 | coppice | jsjc: SIP phones can be powered through PoE, eliminating a mass of wall warts |
| 15:14.28 | box2 | mmmm cat |
| 15:14.31 | [TK]D-Fender | beek: require re-casing for the Raw-Cat Lawn Chair! |
| 15:14.48 | beek | :D |
| 15:15.16 | [TK]D-Fender | grabs his, takes aim at bhodder and LOLcat's the bejeebus out of him. |
| 15:16.01 | jsjc | [TK]D-Fender as soon as i start to undewrstand this world I think I am going to love it!! that means lot of cool things can be done... |
| 15:16.15 | brad_mssw | Is dahdi_dummy required for meetme timers in asterisk 1.6 if using a sangoma card? |
| 15:16.19 | [TK]D-Fender | jsjc: Yup, the ceiling is comfortably high... |
| 15:16.25 | [TK]D-Fender | brad_mssw: No |
| 15:16.50 | bhodder | If that is simply the front end what has to be done before using this then? |
| 15:16.53 | brad_mssw | [TK]D-Fender: how about without a card at all? |
| 15:17.39 | [TK]D-Fender | brad_mssw: DUH <- |
| 15:17.42 | brad_mssw | (I'd assume without the sangoma the dummy would be necessary) |
| 15:17.46 | brad_mssw | ok, thanks ... |
| 15:17.56 | [TK]D-Fender | bhodder: Before you dial YOU have to check their status |
| 15:17.57 | brad_mssw | couldn't find any reference saying the sangoma provided timing support |
| 15:18.44 | [TK]D-Fender | bhodder: Its your job to code the dialplan that allows them to set these values, and its your job to code the dialplan that CHECKS them when deciding what to do when they dial. |
| 15:19.13 | [TK]D-Fender | brad_mssw: Its a DAHDI device and does everything that one should. |
| 15:19.21 | brad_mssw | [TK]D-Fender: thanks |
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| 15:34.05 | jsjc | Grandstream GXW-4008 8 Port FXS IP Analog Gateway has t.38 support... will be ok for fax then? |
| 15:34.55 | jsjc | pretty cheap solution for FXS... |
| 15:35.07 | box2 | ~gs |
| 15:35.08 | infobot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 15:35.52 | coppice | the grandstream ATAs are not bad |
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| 15:38.00 | jsjc | box2 infobot thanks for advice |
| 15:38.10 | jsjc | openbox cheap as well how is that? |
| 15:38.16 | *** join/#asterisk Dr-Linux|home (n=Nothing@119.63.130.34) |
| 15:38.24 | jsjc | openvox A1200P is what i am thinking |
| 15:38.36 | Dr-Linux|home | can i register one SIP phone on multiple asterisk servers? |
| 15:38.47 | Qwell | Dr-Linux|home: only if the phone has multiple identities |
| 15:38.55 | Qwell | ~cheap |
| 15:38.55 | infobot | methinks cheap is a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 15:39.15 | Dr-Linux|home | Qwell: the phone as online one line. |
| 15:39.20 | [TK]D-Fender | jsjc: OpenVox has been hard to contact and get support for if things go wrong, but YMMV |
| 15:39.27 | [TK]D-Fender | jsjc: I would never take that risk |
| 15:39.46 | [TK]D-Fender | Dr-Linux|home: then you've nswered your own question |
| 15:40.46 | Dr-Linux|home | well, i want this extensions should recieve one call at a time but it can be registered on 3 asterisk servers |
| 15:41.38 | Dr-Linux|home | i've no phone problems but i'm planning redundancy |
| 15:41.49 | ariel_ | Dr-Linux|home: you can setup calls from all the asterisk to the one that has the phone just make the rules |
| 15:42.09 | jsjc | hehe ok I see I see... |
| 15:42.56 | [TK]D-Fender | Dr-Linux|home: Go read your phone's manual |
| 15:43.17 | Dr-Linux|home | ariel_: I can setup but i want "this phone be registered on all these 3 servers" becasue everything will be decided through the sip connection state |
| 15:44.12 | Dr-Linux|home | [TK]D-Fender: i'm not talking about my specific phone but i'm talking about Asterisk |
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| 15:44.59 | [TK]D-Fender | Dr-Linux|home: What does * have to dow ith your phone being able to register to 3 separate servers? |
| 15:45.21 | [TK]D-Fender | Dr-Linux|home: your phone does what your phone does, and * has absolutely nothing to do with it |
| 15:45.39 | Dr-Linux|home | example, if i've ext 222, it is registered on all 3 * servers, when i do show sip peer on any server it shows "OK" |
| 15:45.42 | Dr-Linux|home | hhmmm... |
| 15:45.42 | Dr-Linux|home | ok |
| 15:45.59 | Dr-Linux|home | makes sense |
| 15:46.26 | ariel_ | Dr-Linux|home: look at HAV type of setups, like with hartbeat or SER, Enswitch has a nice setup for this |
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| 15:48.48 | Dr-Linux|home | Sip phone -----> Asterisk1 ------> Asterisk 2 : is it possible that SIp phone can register with Asterisk 2? not directly |
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| 15:50.20 | [TK]D-Fender | Dr-Linux|home: Read your phone's MANUAL <----- |
| 15:50.28 | rene- | hello, how can i reliably tell weather a caller has hangup when i am playing a message? i am doing automated outbound calls, play a message and transfer to queue right after the message ends, however i get lots dead airs, |
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| 15:51.04 | jsjc | Then whats the opinion in Digium vs OpenVox? |
| 15:51.06 | [TK]D-Fender | rene-: if you get dead air, * already can't tell they've disconnected otherwise dialplan execution would have already halted <- |
| 15:51.10 | Dr-Linux|home | phone is cisco 7960 |
| 15:51.26 | [TK]D-Fender | jsjc: Clearly digium. Mind you I have my own preference for Sangoma. |
| 15:51.51 | rene- | i am thinking of just set a wait(3) before joining the callers to a queue, since most of those dead airs are callers that abandon the queue within 0-3 secs but i dont know if playing the audio with something like backgrounddetect would make a difference |
| 15:52.34 | [TK]D-Fender | rene-: the caller isn't supposed to be making any noise. that is not a plausible solution. |
| 15:52.39 | [TK]D-Fender | rene-: Fix your CDS |
| 15:53.05 | rene- | D-Fender: caller detect supervision? |
| 15:53.10 | [TK]D-Fender | rene-: Yes |
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| 15:53.55 | rene- | D-Fender: what about the wait option before enqueueing the other party, it would be a bit annoying to people that havent hanged up but it at least would give asterisk time to detect the other party hanging up |
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| 15:54.36 | [TK]D-Fender | rene-: You'll get hit one way or the other. a Wait won't change your CDS situation at all. |
| 15:54.48 | [TK]D-Fender | rennIt won't make * notice any better or faster |
| 15:56.09 | rene- | D-Fender: hmm i used to think CDS was only an issue with analog telephony, i am running this off a PRI |
| 15:56.24 | [TK]D-Fender | rene-: then its the far end... |
| 15:56.31 | rene- | very likely |
| 15:56.32 | [TK]D-Fender | rene-: At which point you're FUBAR'd |
| 15:56.33 | Faustov | when using dahdi, is zaptel.conf or zapata.conf still needed? |
| 15:56.44 | [TK]D-Fender | Faustov: No |
| 15:56.51 | [TK]D-Fender | Faustov: read the docs in the tarball |
| 15:57.51 | Faustov | I have, I've tried every possible option and I'm a bit clueless now |
| 15:58.25 | [TK]D-Fender | Faustov: Including the giant docs in * and DAHDI tarballs, checked the sample configs for the clearly new bits, etc? |
| 15:59.26 | Faustov | [TK]D-Fender: I'm mostly referring to the analog parts in dahdi so it is not that huge |
| 15:59.54 | [TK]D-Fender | Faustov: the conf files are the same for digital as well... just like always |
| 15:59.59 | Faustov | [TK]D-Fender: I noticed another thing btw, when redirecting the incoming analog calls to a sip phone, the situation is the same |
| 16:01.02 | Faustov | and with people making calls all the time it is almost impossible for trial and error, damn it |
| 16:01.05 | *** join/#asterisk nauticalthinker (n=mratliff@74.5.197.138) |
| 16:01.10 | [TK]D-Fender | Faustov: And you can't be on a PRI... you said you're running an A200d <------- |
| 16:01.25 | [TK]D-Fender | Faustov: Which DOES require CDS |
| 16:01.40 | [TK]D-Fender | Faustov: Sorry, bad aim |
| 16:01.44 | [TK]D-Fender | Scratch that |
| 16:01.56 | [TK]D-Fender | rene-: You're still up a creek ;) |
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| 16:02.38 | Faustov | i wonder if my provider might be not supporting koolstart |
| 16:02.48 | Faustov | but according to the docs it is quite populr |
| 16:03.06 | Faustov | i'll try groundstart once all the mofos end their calls... |
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| 16:05.55 | [TK]D-Fender | Faustov: Kewlstart should affect the telco ACK-ing your pickup.... |
| 16:06.06 | [TK]D-Fender | Faustov: Check with Sangoma support ASAP |
| 16:06.23 | ricko73 | Is dial tone detection before dialing on a Zap channel available in 1.4 or is that a 1.6.x feature? |
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| 16:11.45 | nauticalthinker | I was able to get both Asterisk and an F9600 system to talk to each other. Had to play around with different configs on the cable. |
| 16:11.56 | nauticalthinker | both are configured for fxs |
| 16:12.07 | *** join/#asterisk mog (n=mog@c-68-62-169-247.hsd1.al.comcast.net) |
| 16:12.07 | *** mode/#asterisk [+o mog] by ChanServ |
| 16:12.45 | nauticalthinker | my problem is that when an ext is dialed from the f9600 side to reach asterisk, I'm not sure how to handle it on the incoming context |
| 16:13.09 | [TK]D-Fender | nauticalthinker: Can't both be fXS... |
| 16:13.18 | nauticalthinker | ah |
| 16:13.26 | nauticalthinker | ...that would explain things possibly |
| 16:13.40 | nauticalthinker | should asterisk be an fxo or does it matter? |
| 16:13.53 | [TK]D-Fender | nauticalthinker: Depends what the F9000 is using. |
| 16:14.05 | [TK]D-Fender | nauticalthinker: Your hardware has to match. |
| 16:14.12 | [TK]D-Fender | nauticalthinker: And yes it matters |
| 16:14.22 | nauticalthinker | it's set for fxs |
| 16:14.54 | [TK]D-Fender | nauticalthinker: So the F9000 is expecting you to plug in PHONES? |
| 16:15.00 | ariel_ | wonders what is set as, as you either have fxs ports or fxo's you plug one to the other not to same |
| 16:15.03 | nauticalthinker | how should I hand the incoming context on that channel? What I want is for any sip user to be reached if that is dialed from F9600 |
| 16:15.17 | nauticalthinker | it does contain phones...yes |
| 16:15.18 | [TK]D-Fender | nauticalthinker: What kind of jack? What card? |
| 16:15.21 | Katty | ATTENTION |
| 16:15.23 | Katty | IT IS LUNCH TIME |
| 16:15.24 | nauticalthinker | I'm bridging the two |
| 16:15.42 | nauticalthinker | it's a pilot project and both need to act as if it's all one system |
| 16:15.58 | ariel_ | Katty: yes it is, it's hot pockets for today.. |
| 16:16.00 | nauticalthinker | Te121 on the Asteisk server |
| 16:16.07 | Katty | ariel_: :< |
| 16:16.13 | Katty | ariel_: come over for alfredo. |
| 16:16.23 | [TK]D-Fender | nauticalthinker: Why on earth are you using FxS/FXO signalling on a digital trunk? |
| 16:17.35 | nauticalthinker | maybe because I'm still a newbie |
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| 16:17.39 | *** mode/#asterisk [+oooo putnopvut Qwell file russellb] by irc.freenode.net |
| 16:17.39 | nauticalthinker | how should it be handled in your opinion? |
| 16:17.46 | [TK]D-Fender | nauticalthinker: PRI |
| 16:17.50 | bhodder | Ok, Im stuck I can not find any documentation that is working for setting up call forwarding can someone point me in the right direction |
| 16:18.15 | [TK]D-Fender | bhodder: the is no documentation unless someone posted some complete sample of their own way out there |
| 16:18.16 | nauticalthinker | are you saying that t1 to t1 want work then? |
| 16:18.34 | Qwell | puts the life rafts away |
| 16:18.40 | [TK]D-Fender | bhodder: You seemt o fail to grasp that YOU have to check those values you set before shoosing what to do in your extens. |
| 16:19.03 | [TK]D-Fender | nauticalthinker: PRI is a SIGNALING used over T1 <- |
| 16:19.45 | nauticalthinker | okay...I'll change fxsls to bchan=1-23 dchan=24? |
| 16:19.52 | nauticalthinker | need to change the other side to match? |
| 16:20.02 | [TK]D-Fender | nauticalthinker: Of course. |
| 16:20.07 | bhodder | ok, I understand that I have to check, but does that mean for every extension I have to check whether or not the extension is being forwarded |
| 16:20.22 | [TK]D-Fender | bhodder: for every one you care to. |
| 16:20.24 | Katty | do i want: Tortillini Marinara, Three Cheese Chicken Bake, Creamy Ham Fettuchini, or Baked Mostaccoli |
| 16:20.24 | *** join/#asterisk hohum (n=dcorbe@206.71.169.115) |
| 16:20.54 | [TK]D-Fender | Katty: ... YES |
| 16:21.02 | Katty | i was afraid you were gonna say that |
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| 16:22.11 | idi | [TK]D-Fender thx for help, will be back to bug you more later ;) |
| 16:22.32 | *** part/#asterisk idi (n=idi@78.33.22.254) |
| 16:23.08 | dandre | hello, |
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| 16:23.22 | Katty | tortillini marinara wins!!! |
| 16:23.28 | Katty | http://www.tasteofhome.com/Recipes/Tortellini-Marinara <- lunch! |
| 16:23.51 | jsjc | tortellini does not go good iwth marinara |
| 16:24.00 | Katty | yes it does. |
| 16:24.00 | jsjc | make some fetuchinni marinara... |
| 16:24.14 | Katty | cheese tortellini goes with everything. |
| 16:24.16 | [TK]D-Fender | Katty: That sounds so tragiically white :p |
| 16:24.22 | Katty | [TK]D-Fender: your face is. |
| 16:24.37 | Katty | [TK]D-Fender: <3 |
| 16:24.37 | jsjc | wow that recipe its not marinara! |
| 16:24.42 | Katty | ctrl ad |
| 16:24.47 | [TK]D-Fender | Katty: I tanned out a bit on the kayak on Saturday :) |
| 16:24.48 | jsjc | looks like bolognese more than marinara in the pic hehe |
| 16:24.58 | dandre | I have an analog phone connected tto a tdm800 fxs port. I can't get the callerid informations displayed on the phone. If I test this phone on ana analog line, the callerid are displayed. |
| 16:25.10 | jsjc | and marinara with sausage!?!? |
| 16:25.11 | dandre | what should I do? |
| 16:25.42 | [TK]D-Fender | dandConfigure it properly |
| 16:25.45 | jsjc | Sorry I am a hospitality worker/chef/cook so about PBX cannot help much as of yet but cooking any questions I more likely will have an answer |
| 16:25.46 | jsjc | hehe' |
| 16:26.04 | dandre | I don't see how? |
| 16:26.12 | dandre | zapata.conf? |
| 16:26.31 | [TK]D-Fender | dandre: You don't even know your card's config files? GEEZ |
| 16:28.23 | Strogg | hrmmmm allrighty. I think the next thing I'll have to setup is call forwarding to internal extensions |
| 16:28.25 | dandre | the card itself conf file? |
| 16:28.54 | *** join/#asterisk many (n=many@dslb-088-067-168-196.pools.arcor-ip.net) |
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| 16:30.08 | [TK]D-Fender | dandre: SAD. |
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| 16:30.30 | [TK]D-Fender | dandre: channel setup = zapata.conf / chan_dahdi.conf depending which you've installed |
| 16:30.40 | *** part/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 16:31.26 | dandre | yes I know |
| 16:31.41 | dandre | I am using zapata |
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| 16:59.38 | [TK]D-Fender | BRB |
| 16:59.41 | laggo | how do i install zaptel headers? |
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| 17:01.26 | *** join/#asterisk dustybin (n=dustybin@thinkdebian.org) |
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| 17:01.35 | dustybin | anybody here use one of these: |
| 17:01.36 | dustybin | http://www.voipon.co.uk/images/polycom_ip430_netcom_m5390_bundle_big.jpg |
| 17:02.02 | dustybin | (only the phone) |
| 17:02.11 | Qwell | dustybin: sure, tons of people |
| 17:02.25 | dustybin | are they any good? i am considering buying one |
| 17:02.28 | dustybin | http://www.voxhub.com/voip/pages/images/products/polycomIP430_large.jpg |
| 17:03.03 | Qwell | yes |
| 17:03.34 | dustybin | ace |
| 17:03.57 | dustybin | is it worth getting a video phone? |
| 17:04.12 | dustybin | i imagine video tariffs to be more expensive? |
| 17:04.33 | *** join/#asterisk Imo (n=Imo@brln-4dbafda8.pool.einsundeins.de) |
| 17:05.21 | Imo | hello i have installed asterisk on centos with this docu http://codeghar.wordpress.com/2009/03/08/how-to-install-asterisk-on-centos-package-installation/ |
| 17:05.37 | Imo | but when i'm run asterisk i get this error Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 17:05.53 | Imo | and i cant use service asterisk start or stop or restart |
| 17:05.54 | Qwell | does /var/run/asterisk/asterisk.ctl exist? |
| 17:06.11 | Qwell | ~asterisk-non-root |
| 17:06.11 | infobot | [~asterisk-non-root] Running * as non-root is covered in chapter 13 of Asterisk:The Future of Telephony 2nd Edition (~book) and an article written by Leif Madsen at http://www.taug.ca/node/115 |
| 17:06.24 | Imo | yes |
| 17:06.27 | kaldemar | Imo: you're not root, i assume? |
| 17:06.37 | Imo | no i'm root |
| 17:06.44 | Imo | and i installed asterisk in root |
| 17:06.51 | Imo | i login with root |
| 17:06.56 | Imo | and i installed asterisk |
| 17:07.27 | Qwell | Imo: read what the bot said |
| 17:07.45 | dustybin | if one used a wireless network for VOIP, am i right in thinking that the wireless bandwidth could easily be eaten away |
| 17:07.52 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com) |
| 17:08.09 | dustybin | ie. imagine i was downloading stuff on my laptop, and i also took a VOIP call via wireless |
| 17:08.18 | dustybin | could this cause problems? |
| 17:08.30 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 17:08.36 | jaytee | what'd you say? you're breaking up...I didn't catch that |
| 17:08.42 | dustybin | LOL |
| 17:08.52 | dustybin | 'shes breaking up jim' |
| 17:09.47 | [TK]D-Fender | I'll gier'er all I got and a weeee bite moooooooooorrrrrrreee! |
| 17:09.47 | Imo | i get this error |
| 17:09.47 | Imo | This account is currently not available. |
| 17:09.47 | Imo | when i make it step by step , |
| 17:09.47 | *** join/#asterisk snapple42 (n=snapple4@h216-18-80-131.gtconnect.net) |
| 17:10.14 | Imo | can i install asterisk on another way ? |
| 17:10.18 | Imo | i wont only run asterisk |
| 17:10.29 | tzafrir_laptop | dustybin, what kernel version do you use? |
| 17:10.41 | dustybin | Linux wizbox 2.6.26-2-686 #1 SMP Sun Jun 21 04:57:38 UTC 2009 i686 GNU/Linux |
| 17:10.42 | tzafrir_laptop | If recent enough, look into ionice |
| 17:11.15 | dustybin | what is ionice? |
| 17:11.45 | tzafrir_laptop | like nice/renice for I/O |
| 17:11.48 | Imo | why i get the erros ? |
| 17:12.07 | Imo | i dont understand that ? i have installed asterisk on antoher centos with the same commands. |
| 17:12.17 | dustybin | tell hell with it, i dont need wireless for VOIP |
| 17:13.30 | exsync | . |
| 17:13.36 | [TK]D-Fender | [13:09]<Imo>This account is currently not available. <- Huh? What gives you this exactly? |
| 17:14.13 | Imo | i get this url |
| 17:14.13 | Imo | http://www.taug.ca/node/115 |
| 17:14.26 | Imo | and i make it step by step and by su asterisk i get this error |
| 17:14.49 | Imo | and i dont understand why its so difficult, normaly i can install this very easy |
| 17:15.01 | [TK]D-Fender | Imo: PASteBIN the complete attempt including that error |
| 17:15.03 | [TK]D-Fender | ~pb |
| 17:15.04 | 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 |
| 17:15.05 | [TK]D-Fender | ^^^^^^^^6 |
| 17:15.42 | Imo | there is nothing to pastbin |
| 17:16.17 | [TK]D-Fender | Imo: Yes there is.. there is CLI output leading that that ERROR MESSAGE |
| 17:16.24 | [TK]D-Fender | to* |
| 17:16.33 | Imo | ?? |
| 17:16.55 | Imo | ypu mean this or what Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?)? |
| 17:16.59 | [TK]D-Fender | Imo: You get that error on your scrren. I'm sure there is stuff that would outputtd BEFORE that error as well. |
| 17:17.11 | [TK]D-Fender | [13:13]<[TK]D-Fender>[13:09] <Imo> This account is currently not available. <- Huh? What gives you this exactly? <--------- |
| 17:17.29 | Imo | sorry i dont understand what you mean |
| 17:17.32 | *** join/#asterisk jicksta_ (n=jicksta@c-98-210-108-28.hsd1.ca.comcast.net) |
| 17:17.49 | Imo | please forget it |
| 17:18.05 | [TK]D-Fender | Imo: You said you got an error message saying "This account is currently not available.". WHERE? |
| 17:18.10 | [TK]D-Fender | Imo: Doing what? |
| 17:18.23 | Qwell | [TK]D-Fender: when he ran `su asterisk`, I'm guessing |
| 17:19.04 | Imo | su asterisk |
| 17:19.19 | [TK]D-Fender | Imo: Guess you didn't creat it <- |
| 17:19.28 | Imo | i create this |
| 17:19.59 | [TK]D-Fender | Imo: "su asterisk -" <- |
| 17:20.20 | Imo | yes |
| 17:20.23 | ariel_ | "su - asterisk" |
| 17:20.24 | *** join/#asterisk hugorebelo (n=hugorebe@200-171-132-124.completo.com.br) |
| 17:20.31 | [TK]D-Fender | Imo: and? |
| 17:20.38 | Imo | the same |
| 17:20.49 | [TK]D-Fender | Imo: Show me you created the user/ |
| 17:20.55 | Imo | i now |
| 17:20.59 | [TK]D-Fender | Imo: And pastebin the failed attempt to change users |
| 17:21.00 | Imo | adduser asterisk |
| 17:21.19 | Imo | adduser: Benutzer asterisk vorhanden |
| 17:21.55 | [TK]D-Fender | Imo: ..pastebin the failed attempt and backup that it exists |
| 17:22.13 | Imo | what is the failed attempt ? |
| 17:22.24 | *** join/#asterisk hfb (n=hfb@pool-96-247-114-183.lsanca.dsl-w.verizon.net) |
| 17:22.30 | dustybin | I have made my final decision, i am going to buy this |
| 17:22.31 | dustybin | http://www.polycom.eu/products/voice/desktop_solutions/soundpoint/desk_phones/soundpoint_ip450.html |
| 17:22.33 | Imo | i dont understand what you mean sorry |
| 17:23.37 | Imo | is there another way to install asteriks ? now i reinstall my server |
| 17:23.43 | [TK]D-Fender | Imo: What is there to not understand? PASTEBIN the &^#$ING CLOI OUTPUT FROM LINUX FOR YOUR ATTEMPT TO CHANGE USERS. |
| 17:24.16 | [TK]D-Fender | dustybin: Good phone... what kind of call volume? |
| 17:24.38 | Imo | dduser: Benutzer asterisk vorhanden |
| 17:24.41 | Imo | this is only what i get |
| 17:24.46 | Imo | not more |
| 17:24.48 | [TK]D-Fender | Imo: .....PASTEBIN |
| 17:24.50 | [TK]D-Fender | ~pb |
| 17:24.50 | 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 |
| 17:24.52 | [TK]D-Fender | ^^^^^^^^ |
| 17:24.59 | Imo | you want kidding me ? |
| 17:25.18 | [TK]D-Fender | Imo: And I told you several times to show me that the user is properly configured. with a SHELL specified as well |
| 17:25.20 | Imo | i should pastbin thos " dduser: Benutzer asterisk vorhanden" ? |
| 17:25.32 | Imo | mom please |
| 17:25.38 | dustybin | [TK]D-Fender: call volume, what do you mean? |
| 17:25.41 | Imo | i reinstall the server and try again to install asterisk |
| 17:25.55 | [TK]D-Fender | dustybin: How many calls in a day, at a time, headset required, etc |
| 17:26.09 | [TK]D-Fender | Imo: Complete waste.. if you just reinstall what will be different? |
| 17:26.11 | dustybin | [TK]D-Fender: not much, steady |
| 17:26.12 | Imo | can i install only asterisk |
| 17:26.20 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com) |
| 17:26.22 | Imo | i dont know |
| 17:26.58 | Imo | <PROTECTED> |
| 17:27.10 | [TK]D-Fender | dustybin: How steady? Mostly inbound? Have to juggle a lot of calls around at a time? |
| 17:27.14 | Imo | and i dont installed any kernel or devel etc. |
| 17:27.34 | [TK]D-Fender | Imo: And you aren't showing me what I've asked for several times now. |
| 17:27.38 | dustybin | [TK]D-Fender: no way, the max will be 2 calls at once |
| 17:27.38 | Imo | but i get a new server and must install asterisk again and now i get this problems |
| 17:27.45 | dustybin | [TK]D-Fender: and that will be rare |
| 17:27.56 | [TK]D-Fender | dustybin: Headset requirements? |
| 17:28.15 | Imo | sorry than i dont understand what you mean |
| 17:28.18 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com) |
| 17:28.34 | [TK]D-Fender | Imo: [13:23]<[TK]D-Fender>Imo: What is there to not understand? PASTEBIN the &^#$ING CLOI OUTPUT FROM LINUX FOR YOUR ATTEMPT TO CHANGE USERS |
| 17:28.36 | dustybin | [TK]D-Fender: I havent looked into headsets, maybe in the future |
| 17:28.41 | [TK]D-Fender | Imo: Please ask again when you can read. |
| 17:28.42 | Katty | lunch is served! |
| 17:28.47 | Imo | ok |
| 17:28.56 | dustybin | [TK]D-Fender: can i use a headset with that phone? |
| 17:29.03 | [TK]D-Fender | ~wmmfpb |
| 17:29.04 | infobot | [~wmmfpb] WHERE'S MY MOTHER-^%#ING PASTEBIN?!? |
| 17:29.29 | Imo | what tha ? &^#$ING CLOI OUTPUT FROM LINUX FOR YOUR ATTEMPT TO CHANGE USERS |
| 17:29.42 | Imo | i dont now what you want |
| 17:29.44 | [TK]D-Fender | loves that one... and hates that it actually represents a necessary question. |
| 17:30.08 | [TK]D-Fender | Imo: Linux CLI output for your attempt to change users <- What part is not clear? |
| 17:30.33 | [TK]D-Fender | Imo: show me the user config files that prove the user exists and has a shell defined. |
| 17:31.00 | Imo | please give me the command and i will give you the output in pastbin |
| 17:31.03 | ariel_ | wow linux 101 class. |
| 17:31.15 | ariel_ | su - asterisk |
| 17:31.25 | Imo | AHH WHAT I GIVE YOU THIS |
| 17:31.38 | Imo | adduser: Benutzer asterisk vorhanden |
| 17:32.31 | [TK]D-Fender | Imo: I don't want jsut the error message, I want to see the coomand as you issue it. Do not give me another worthless story. now pastebin ALL OF IT |
| 17:32.49 | Imo | ok |
| 17:32.57 | Imo | moment please |
| 17:33.34 | jplank | fender is going to go postal one day, and I hope my logs catch it |
| 17:33.57 | ariel_ | rofl |
| 17:34.11 | jeff_phillips | jplank: do you want me to pm you next time I set him off so you can make sure your log is active? lol |
| 17:34.20 | Katty | you're so imo. |
| 17:34.35 | Imo | ? |
| 17:34.36 | [TK]D-Fender | wishes his lawn was emo... so it would cut itself... |
| 17:34.47 | jplank | lol ^^^ |
| 17:34.50 | Katty | you know they make something for that. |
| 17:34.59 | ariel_ | HOA house.... they cut my lawn... |
| 17:35.01 | Katty | it's called roundup. |
| 17:35.08 | Katty | just spray the whole yard down. |
| 17:35.22 | jeff_phillips | i've thought about paving mine, but it would be expensive |
| 17:35.39 | jplank | ariel, I'm in a HOA also, and I'm lucky if the weeds next door don't hit 6 feet |
| 17:35.54 | Katty | jeff_phillips: it's all about total cost of ownership |
| 17:36.18 | Katty | jeff_phillips: sit down and figure out how much it costs to pave, and maintain the pavement, as opposed to maintenance of your lawn. |
| 17:36.25 | Katty | jeff_phillips: you may find that in 3 years, you recoup your cost. |
| 17:36.31 | ariel_ | we voted to have the lawn co do all yards.. no matter what. front and back... saves having to fight with the others |
| 17:36.32 | jeff_phillips | good point |
| 17:36.48 | Katty | but it wouldn't be nearly as pretty. |
| 17:36.52 | Katty | and the bunnies wouldn't like you. |
| 17:36.58 | WindowsUser | hrm |
| 17:37.01 | jeff_phillips | could just let the village do it for $600 per mowing. |
| 17:37.10 | jeff_phillips | or the neighbor kid for $10 |
| 17:37.18 | WindowsUser | instead of spending money on corn maybe the US govt should spend money on lawns |
| 17:37.27 | Katty | i don't trust the neighbors with our lawn |
| 17:37.37 | Katty | they'd probably mow right over my flowers. |
| 17:37.38 | ariel_ | were paying 650 per cut for 72 homes... it's really not bad |
| 17:37.47 | box2 | i don't trust the neighbors with being my neighbors |
| 17:38.04 | Katty | box2: well, we have a german shepherd. the neighbors don't pose a threat. |
| 17:38.11 | WindowsUser | i don't trust the neighbors to be human |
| 17:38.14 | WindowsUser | ha! i win |
| 17:38.18 | Katty | box2: they don't come on our property anymore |
| 17:38.22 | Katty | WindowsUser: you do win!!! |
| 17:38.25 | Katty | WindowsUser: congrats! |
| 17:38.28 | *** join/#asterisk [netman] (n=netman@202.Red-88-23-83.staticIP.rima-tde.net) |
| 17:38.30 | Katty | WindowsUser: a lifetime supply of ricearoni!!!! |
| 17:38.34 | WindowsUser | woo! |
| 17:38.50 | box2 | i heart rice-a-roni |
| 17:39.10 | Imo | here i pastbin alll my setup http://pastebin.com/d7ea6415f |
| 17:39.36 | Imo | i make it on a fresh centos ;) |
| 17:39.51 | Katty | with a side of rice a roni? |
| 17:39.57 | *** join/#asterisk pimpwell (n=domin8@ool-ad03dcac.dyn.optonline.net) |
| 17:40.05 | Katty | sounds awfully carby. |
| 17:40.44 | box2 | heh! |
| 17:40.54 | Imo | [TK]D-Fender: its thats ok ? |
| 17:40.59 | [TK]D-Fender | <PROTECTED> |
| 17:41.49 | Imo | ? |
| 17:41.54 | jeff_phillips | that's a lot of rice a roni |
| 17:42.28 | kaldemar | Imo: you have to start asterisk to be able to connect it |
| 17:42.29 | [TK]D-Fender | Imo: And Nothing in there shows me that you STARTED ASTERISK |
| 17:42.43 | [TK]D-Fender | Imo: And I don't see your attempt to change users like you were trying. |
| 17:42.49 | Imo | i cant start asterisk |
| 17:43.01 | [TK]D-Fender | Imo: "asterisk -r" DOES NOT START ASTERISK |
| 17:43.09 | Imo | when im start i get this error and i cant stop that and i must restart my server |
| 17:43.14 | [TK]D-Fender | Imo: It connects to an ALREADY RUNNING instance of * |
| 17:43.53 | Imo | here |
| 17:43.55 | Imo | <PROTECTED> |
| 17:43.56 | Imo | http://pastebin.com/d3fde3412 |
| 17:44.20 | Imo | i now |
| 17:44.26 | Imo | i must type asterisk start |
| 17:44.34 | Imo | or service asterisk start |
| 17:44.50 | Imo | but i get this http://pastebin.com/d3fde3412 |
| 17:45.16 | bhodder | Ok, I've tried to get the dialing plan to check the status of the extension and to get it to callforward from there once the user has set the number to forward to but it still will not forward the cal |
| 17:45.52 | [TK]D-Fender | Imo: Because you had a problem with DAHDI most likely which is causing * to crash in circles <- |
| 17:46.31 | *** join/#asterisk generalhan (n=asd@about/windows/staff/generalhan) |
| 17:46.52 | Imo | i dont think tha |
| 17:46.53 | Imo | t |
| 17:47.05 | Imo | on my other server i dont installed dhadi too and it works |
| 17:47.36 | [TK]D-Fender | Imo: PASteBIN the output of "asterisk -gvvvvvvvvvc" |
| 17:48.20 | Imo | http://pastebin.com/d2f3f08c4 |
| 17:48.49 | Qwell | broken package. |
| 17:48.55 | [TK]D-Fender | Imo: [codec_speex.so]Aug 10 19:47:44 WARNING[5162]: loader.c:326 __load_resource: /usr/lib/asterisk/modules/codec_speex.so: undefined symbol: speex_preprocess_ctl -- Aug 10 19:47:44 WARNING[5162]: loader.c:555 load_modules: Loading module codec_speex.so failed! |
| 17:49.46 | [TK]D-Fender | Imo: echo "noload => codec_speex.so" > /etc/asterisk/modules.conf |
| 17:49.51 | [TK]D-Fender | Imo: Do this ^^^ |
| 17:50.01 | bhodder | If I get the extension to be set to call forward to a number what do i use to check the status of the extension before it dials it |
| 17:50.02 | Imo | nothing |
| 17:50.24 | [TK]D-Fender | Imo: SHOW ME |
| 17:50.39 | [TK]D-Fender | Imo: And you should have tried to start it manually again |
| 17:50.45 | [TK]D-Fender | 13:47]<[TK]D-Fender>Imo: PASteBIN the output of "asterisk -gvvvvvvvvvc" |
| 17:51.02 | [TK]D-Fender | bhodder: "core show application gotoif" <- |
| 17:51.49 | Imo | http://pastebin.com/d5a49cbbc |
| 17:52.05 | jaytee | god, you're such a masochist! I'm in awe :-) |
| 17:53.18 | [TK]D-Fender | jaytee: Yeah, I "oppsed" on the echo too :) |
| 17:54.56 | *** join/#asterisk proute (n=AnthonyC@ARouen-153-1-70-63.w90-17.abo.wanadoo.fr) |
| 17:55.05 | proute | hello all |
| 17:55.12 | box2 | -vvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvv |
| 17:55.31 | proute | I use * 1.4.25.1 with dahdi (the last release) with Bri card B410P |
| 17:55.33 | box2 | <asterisk> oh man i think i ate something bad for lunch |
| 17:55.44 | box2 | TOO MANY -v |
| 17:56.19 | proute | sometime for an incoming call via ma B410P, the call works, sometimes, the incoming call not work and my SPAN is in red alert.... |
| 17:56.23 | proute | why? |
| 17:57.07 | proute | I have the same problem for an outgoing call :( |
| 17:57.10 | jaytee | [TK]D-Fender, was it missing an extra > to append or was it the quotes? |
| 17:59.14 | proute | in my cli, I have Primary D-Channel on span 1 down and up.... |
| 17:59.36 | [TK]D-Fender | jaytee: > |
| 18:00.14 | jaytee | [TK]D-Fender, that's what I thought after you pointed it out. I didn't catch it when you first typed it. |
| 18:00.15 | *** join/#asterisk cusco_ (n=tralala@2001:0:53aa:64c:1c35:38e5:2ac0:762d) |
| 18:00.16 | cusco_ | hi |
| 18:00.31 | cusco_ | what is the latest stable 1.6? |
| 18:00.37 | jaytee | 1.6.1 |
| 18:00.54 | cusco_ | there was an error in the addons for that, on the mysql addon |
| 18:01.01 | cusco_ | right? |
| 18:02.02 | jaytee | dunno |
| 18:02.02 | cusco_ | is svn usable? |
| 18:02.12 | cusco_ | or maybe there is not a public svn trunk |
| 18:02.13 | jaytee | don't use it myself, might ask in #asterisk-dev |
| 18:05.24 | eppigy | SUBVERSION |
| 18:06.32 | jaytee | TRABAJO |
| 18:06.50 | b14ck | there is no theory of evolution, just a list of animals chuck norris allows to live |
| 18:06.52 | b14ck | D: |
| 18:07.58 | jaytee | Chuck Norris never does laundry, his clothes are so afraid of making him angry they wash, dry and fold themselves. |
| 18:08.16 | cusco_ | thats ok, I will install the latest stable |
| 18:08.16 | cusco_ | thanks |
| 18:08.33 | jaytee | cusco_, which version of addon-ons do you have? |
| 18:08.40 | [TK]D-Fender | Imo: You're up & running |
| 18:08.47 | [TK]D-Fender | Imo: Yuo can connect via "asterisk -r" now |
| 18:08.50 | b14ck | Chuck Norris can win a game of connect four with only 3 moves. |
| 18:09.51 | beek | Given Chuck's religious views, I find the comment about evolution hilarious. |
| 18:10.32 | [TK]D-Fender | Imo: I reinstalled the RPM for your *, and added the "noload => codec_speex.so" to your modules.conf and started the service after verifying that everything was OK |
| 18:11.48 | bhodder | I get it to test but it does not see that it is set for call forwarding or the seting the call forwarding is not working |
| 18:12.00 | Imo | [TK]D-Fender: thanks but can you pastbin all the commands ? when i have the problem again i dont want ask again ;) |
| 18:13.19 | b14ck | lol |
| 18:13.34 | b14ck | "Chuck Norris once ate a whole cake before his friends could tell him there was a stripper in it." rofl! |
| 18:14.25 | [TK]D-Fender | Imo: yum uninstall astersk |
| 18:14.29 | [TK]D-Fender | Imo: yum install astersk |
| 18:14.55 | [TK]D-Fender | Imo: echo "noload => codec_speex.so" >> /etc/asterisk/modules.conf |
| 18:14.59 | [TK]D-Fender | Imo: Thats it |
| 18:15.03 | Imo | ohh |
| 18:15.07 | Imo | very thanks ;) |
| 18:15.24 | b14ck | Hey, have any of you done any work with audiohook.h? |
| 18:16.01 | b14ck | I was looking for a way to modify audio on a channel realtime and came across it, but don't see any examples or samples of it anywhere. Was just curious if anyone had played around with it. |
| 18:17.32 | *** join/#asterisk moy (n=moy@CPE001cdfec4cee-CM00080dab8485.cpe.net.cable.rogers.com) |
| 18:19.46 | *** join/#asterisk rene- (n=renemend@200.34.66.137) |
| 18:20.03 | rene- | hello, how can i get a list of rboc/non rboc area codes? |
| 18:20.32 | rene- | sorry to be OT |
| 18:21.35 | *** join/#asterisk TSM2 (n=the_soft@87-194-32-212.bethere.co.uk) |
| 18:23.00 | proute | When I try a call to my B410P (dahdi...) my SPAN go to RED.... |
| 18:23.01 | proute | why? |
| 18:24.06 | *** join/#asterisk thansen (n=thansen@76.27.110.194) |
| 18:24.52 | bhodder | ok, the gotoif checks it fine but the nuber appears not to get set as it always ends up being an empty number when it tries to dial it |
| 18:26.21 | bhodder | this is what I am using to check the extension status :exten => 315,1,GotoIf($[${CFIM}=""]?3:2) |
| 18:26.39 | *** join/#asterisk xpot-mobile (n=james@71-213-48-149.slkc.qwest.net) |
| 18:26.41 | bhodder | is that correct or should that be something else? |
| 18:27.26 | [TK]D-Fender | bhodder: How would we know> So far you are just comparing some random channel variable. We don't see if or where it ever gets set |
| 18:28.09 | [TK]D-Fender | Imo: ps -A|grep aster |
| 18:28.19 | [TK]D-Fender | Imo: then kill any process # you see |
| 18:28.34 | [TK]D-Fender | Imo: And only start * MANUALLY like I told you before : asterisk -gvvvvvvvc |
| 18:29.03 | *** join/#asterisk dmz (n=dmz@64.203.233.195.dyn-cm-pool35.hargray.net) |
| 18:30.06 | WindowsUser | reoccuring problems? |
| 18:30.16 | bhodder | here is a pb of what is being done to set it |
| 18:30.18 | bhodder | http://pastebin.com/m3a77acd8 |
| 18:30.49 | *** join/#asterisk errotan (n=errotan@62.201.123.54) |
| 18:31.01 | [TK]D-Fender | bhodder: You are setting an AstDB key-paitr, not a CHANNEL-VARiABLE |
| 18:31.32 | [TK]D-Fender | bhodder: ${CFIM/CALLERID(num)} <- also quite invalid |
| 18:32.33 | [TK]D-Fender | Imo: PASTEBIN YOUR OTHER SERVER'S MODULES.CONF |
| 18:32.40 | bhodder | ok, the method that is used to set the CFIM is that valid |
| 18:33.12 | [TK]D-Fender | bhodder: there is no setting. when you dial 315 its a NEW CALL. that variable does not exist |
| 18:33.32 | [TK]D-Fender | bhodder: Do don't seem to have the slightest clue what the exten that SETS values for you is doing |
| 18:33.59 | [TK]D-Fender | bhodder: You put the # into AstDB, but you aren't PULLING the value from AstDB when it comes time to check it. |
| 18:34.47 | bhodder | ok, how do I pull the value from AstDB to check it |
| 18:35.14 | [TK]D-Fender | bhodder: go read the basics on "asterisk functions" on the WIKI |
| 18:35.16 | [TK]D-Fender | ~wikis |
| 18:35.17 | infobot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 18:36.04 | [TK]D-Fender | bhodder: And while we're on a reading spree go read the "channelvariables" doc in your doc/ folder from your source tarball (might be in a /tex sub-folder) |
| 18:36.21 | *** join/#asterisk dupe (i=blahsss@S0106001217be899c.ed.shawcable.net) |
| 18:36.24 | bhodder | ok |
| 18:36.26 | bhodder | thanks |
| 18:36.37 | TSM2 | im looking in the asterisk docs, could not see if AstDB is a persistant DB? |
| 18:37.48 | [TK]D-Fender | TSM2: it is |
| 18:38.57 | TSM2 | good good |
| 18:39.14 | *** join/#asterisk Thummy (n=swamplan@66-191-62-219.static.stpt.wi.charter.com) |
| 18:40.25 | *** join/#asterisk Dovid (n=annon@213.8.121.90) |
| 18:41.04 | *** join/#asterisk h00k (n=anthony@unaffiliated/h00k) |
| 18:43.30 | ariel_ | So far I am getting more used to 1.6, seems it's going to workout after all. |
| 18:46.39 | *** join/#asterisk xpot-mobile (n=james@173-14-225-193-Utah.hfc.comcastbusiness.net) |
| 18:47.19 | *** join/#asterisk bmoraca (n=chatzill@66.242.174.254) |
| 18:53.07 | *** join/#asterisk sekil (n=Ognjen@80.93.247.26) |
| 18:59.24 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 19:03.24 | *** join/#asterisk jicksta (n=jicksta@c-98-210-108-28.hsd1.ca.comcast.net) |
| 19:05.47 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 19:12.11 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 19:13.09 | TSM2 | i am trying to work out the >database get command, can someone give me a quick example that works |
| 19:18.13 | [TK]D-Fender | TSM2: "help database" |
| 19:18.42 | eppigy | ZING |
| 19:19.46 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 19:20.42 | Katty | ZONG! |
| 19:21.35 | *** join/#asterisk KazaLite (n=KazaLite@94-193-98-124.zone7.bethere.co.uk) |
| 19:26.23 | h00k | Is the best place to ask for AsteriskNOW questions going to be in #asterisknow (20 users) or in here? |
| 19:26.45 | WindowsUser | the former |
| 19:27.45 | [TK]D-Fender | h00k: Depends on the question |
| 19:28.38 | h00k | [TK]D-Fender: mostly an initial setup (Using AsteriskNOW server), and linksys spa962 phones. |
| 19:28.56 | h00k | [TK]D-Fender: Thummy is with me and we're seeking assistance. |
| 19:29.33 | dupe | im back, and still having a braindead issue with iax transfers. i'm tired and willing to toss some $$ to someone who can just tell me why it wont work. http://pastebin.ca/1524214 is the latest attempt |
| 19:29.44 | dupe | i need to get more than 3 hours of sleep sigh |
| 19:30.03 | JT | use sip? ;) |
| 19:30.31 | h00k | JT: as per Thummy in #asterisknow: I am new to the asteriskNow world. I have set up Cisco systems in the past and am looking at AsteriskNow for testing. We are testing with linksys spa962 phones and just internal calls. when setting up the phones we do not get dial tones. when pressing the line button, the phone gives a failed, Not reached warning. is there a guide to getting these phone working with asterisk Now |
| 19:30.56 | [TK]D-Fender | dupe: http://pastebin.ca/1524247 |
| 19:31.22 | [TK]D-Fender | h00k: tNo dialtone = not registered |
| 19:31.44 | [TK]D-Fender | h00k: enable SIP debug and look for registration attempts |
| 19:33.47 | Thummy | [TK]D-Fender: Where would one find this? is that an extension? |
| 19:34.28 | [TK]D-Fender | Thummy: In GUI terms, yes they probably call it "extension setup". But I said look at * CLI <-------- |
| 19:34.41 | [TK]D-Fender | Thummy:To see if the phone is even trying to contact yours erver |
| 19:34.44 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 19:34.55 | TSM2 | [TK]D-Fender: yes i have tried that but when i try ">database get cidname 2200" i get nothing |
| 19:35.01 | *** join/#asterisk VoipForces (n=kvirc@mail.net-forces.com) |
| 19:35.24 | [TK]D-Fender | TSM2: PASTEBIN, along with "database show" |
| 19:35.29 | VoipForces | Anyone has recomendation on analog handfree outdoor doorphone that works with mediatrix devices? |
| 19:35.48 | dupe | [TK]D-Fender: still registration refused.... sigh |
| 19:36.36 | [TK]D-Fender | dupe: You have IP's defined, you don't NEED registrations |
| 19:36.51 | [TK]D-Fender | Aww crap |
| 19:37.17 | TSM2 | VoipForces: yup, try http://www.2n.cz/products/door-lift-phones/door-entry-systems.html |
| 19:37.33 | TSM2 | VoipForces: they do voip versions and analogue version which are far cheeper |
| 19:37.35 | [TK]D-Fender | dupe: http://pastebin.ca/1524259 |
| 19:38.02 | Imo | now i try to install asterisk with yum but i get this error http://pastebin.com/d112c0efa |
| 19:38.28 | TSM2 | [TK]D-Fender: http://pastebin.ca/1524264 |
| 19:38.37 | VoipForces | TSM2: cool. i'll check it out. prob is that mediatrix send out 18v but they barly do 20mA on the current loop so with the impedance of the doorphone the voltage drops and deviices like Bogen ADP1 fail. |
| 19:39.03 | [TK]D-Fender | TSM2: Now try giving me what I asked for <- |
| 19:39.42 | jeff_phillips | VoipForces: Why don't you just throw a small ATA device in for that phone? |
| 19:40.21 | [TK]D-Fender | jeff_phillips: What do you think the Mediatrix is? |
| 19:40.36 | jeff_phillips | I had a problem with the Audiocodes MP-124 gateway not putting out enough current to drive our loud bell ringer, so I threw an old SPA-2000 in for that extension and it worked fine |
| 19:40.52 | [TK]D-Fender | :/ |
| 19:41.10 | VoipForces | jeff_phillips: I doubt that a small ata will provide the necessary voltage and current loop |
| 19:41.10 | VoipForces | TSM2: Nice device but way too big and complex. What I need is something like this http://www.camelectronics.com/boadandoph.html |
| 19:41.39 | TSM2 | VoipForces: they do diffrent versions without the pin pad |
| 19:41.40 | VoipForces | jeff_phillips: AUdiocodes suck. |
| 19:41.40 | jeff_phillips | Well you might be suprised, I've found that sometimes the expensive $ gear doesn't always do what you want as easily as the cheap stuff. |
| 19:41.46 | jeff_phillips | yeah they suck |
| 19:41.48 | jeff_phillips | i got it off ebay |
| 19:42.16 | VoipForces | jeff_phillips: hmmm might worth the try. |
| 19:44.00 | TSM2 | [TK]D-Fender: http://pastebin.com/m46b7efc5 dont want to post all of it |
| 19:44.23 | Imo | [TK]D-Fender: can you say me what can i do ? |
| 19:44.29 | [TK]D-Fender | TSMApparently you haven't learned what the FAMILY part of your tree is <- |
| 19:44.40 | *** join/#asterisk wcselby (n=wcselby@216.110.88.254) |
| 19:44.57 | [TK]D-Fender | TSM2: did you think the AMPUSER part wasn't important? |
| 19:45.19 | TSM2 | i thought that AMPUSER is the family |
| 19:45.33 | TSM2 | ahh now i get it |
| 19:45.38 | [TK]D-Fender | TSM2: It is, and nowhere in your GET do you specify it. |
| 19:45.41 | wcselby | is anyone available to help me figure out why my polycom IP 7000 SIP phone constantly loses connection from *, whereas the other 150 phones I have (different make / models) work fine? |
| 19:45.42 | TSM2 | database get AMPUSER/2200 ciduser |
| 19:45.53 | *** join/#asterisk alecdavis (n=sivad@202-78-149-14.cable.telstraclear.net) |
| 19:46.21 | [TK]D-Fender | TSM2: Getting warmer.... |
| 19:47.07 | TSM2 | i did it and it worked |
| 19:47.16 | *** join/#asterisk sjobeck (n=Adium@c-71-193-223-71.hsd1.or.comcast.net) |
| 19:47.41 | TSM2 | [TK]D-Fender: is there somthing missing? |
| 19:49.30 | wcselby | my polycom Soundstation IP 7000 conference phone registers to my * server for about 45 seconds after it first boots, you can make or receive calls with two-way audio during that time, but after 45 seconds if you have an open call, you go to one-way or sometimes no-audio, and if there isn't a call going, you lose the connection to the server entirely (unable to make or receive calls) |
| 19:49.59 | VoipForces | wcselby: firmware issue? |
| 19:50.10 | wcselby | i've tried multiple firmwares, including the latest |
| 19:50.15 | wcselby | that was my first thought as well |
| 19:50.19 | TSM2 | wcselby: is you * on the same net as the server, or is it going through a firewall/router? |
| 19:50.25 | *** part/#asterisk sjobeck (n=Adium@c-71-193-223-71.hsd1.or.comcast.net) |
| 19:50.38 | VoipForces | wcselby: are you doing the proviosionning via tftp? |
| 19:50.38 | TSM2 | wcselby: i ment polycom & asterisk |
| 19:50.47 | wcselby | i've got a sip debug trace along with sip.conf file posted on pastebin.com; TSM2 - it's on the same network, no NAT |
| 19:50.52 | wcselby | provisioning via FTP |
| 19:51.27 | TSM2 | wcselby: have you been changing the sip.conf? |
| 19:51.41 | TSM2 | wcselby: i have not had any problems with my poly 330 & 550 |
| 19:51.42 | wcselby | I have tried changes to the sip.conf file, yes |
| 19:51.44 | VoipForces | wcselby: i would try manual provisionning to see if it does the same |
| 19:52.01 | wcselby | i have probably 75 working polycom 601's on the network, along with another 60-70 cisco 7960s |
| 19:52.15 | *** join/#asterisk dmz (n=dmz@64.203.233.195.dyn-cm-pool35.hargray.net) |
| 19:52.17 | wcselby | the manual provisioning does the same thing |
| 19:52.27 | Thummy | [TK]D-Fender:I'm still trying to find how to enable this, but thanks for the tip |
| 19:52.29 | wcselby | well, i don't have a sip debug trace from a manual prov |
| 19:52.43 | wcselby | here's the sip debug, if you're interested - http://pastebin.com/m2be2930f |
| 19:53.21 | VoipForces | wcselby: when you are not able to make/receive calls, is it still registeres to asterisk (sip show peers) |
| 19:53.31 | wcselby | yes, it is registered in sip show peers |
| 19:53.39 | TSM2 | wcselby: have you looked at the app logs from the phone? |
| 19:53.50 | hudony | hi, I have a question regarding chanspy() + manager, Should I ask now? |
| 19:54.05 | VoipForces | hudony: dont ask to ask, just ask |
| 19:54.33 | hudony | Ok, i was wondering if you guys had a ticket or position system etc.. whatever... :) |
| 19:54.35 | hudony | ok |
| 19:54.53 | wcselby | i've looked at the app logs from the phone, not seeing anything I think is relevant (but I could be wrong) |
| 19:54.59 | TSM2 | hudony: no, just put question and it may get answered |
| 19:55.25 | [TK]D-Fender | Thummy: I gave you a PB... http://pastebin.ca/1524259 |
| 19:55.38 | TSM2 | wcselby: seems like a NAT issue, but as you said there is no NAT happening, what version app is the phone running? |
| 19:55.49 | [TK]D-Fender | hudony: No, just ask, provide as much clear detail as possible |
| 19:55.58 | wcselby | Version=3.1.1.0191 25-Nov-08 13:49 |
| 19:57.09 | hudony | I need to let a call center supervisor listen in real-time calls form his agents. I'm using the manager via ajax request. I plan to use the "originate" manager function with chanSpy as the application parameter. I want to the superviser be able to listen to call not using his phone but via the internet (mp3, wav etc) |
| 19:57.17 | hudony | Is is possible to do so? |
| 19:57.19 | *** join/#asterisk t_ (i=tom@freenode/staff/tomaw) |
| 19:57.29 | wcselby | I have also tried a few other versions as well, don't have the exact numbers on me, but they were 3.1.1base; 3.1.3RevC Combined, and 3.0.2 Rev C |
| 19:58.24 | TSM2 | wcselby: when you uploaded the latest firmware, did you remember to put the sip.conf in the FTP dir and make sure its unmodified and set in the <MAC>.conf |
| 19:58.29 | [TK]D-Fender | wcselby: Polycom's should not be subscribing to * VM. * VM is push-notify-d according to your peer entry |
| 19:58.38 | VoipForces | wcselby: The call-limit=0 bugs me, have you tries with call-limit=1 |
| 20:00.07 | VoipForces | hudony: don't think so. chanspy bridges you as a listener-only to the requester channel |
| 20:00.08 | wcselby | TSM2 - I put the sip.cfg from the firmware release .zip file for each specific release in with each firmware (i hope this makes sense); [TK]D-Fender - I can make a change, this was a copy of a workign polycom 601 config; VoipForces - I can try this. What does this setting do? |
| 20:00.33 | VoipForces | wcselby: limit the number of concurent calls for that extension... |
| 20:00.35 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 20:00.57 | wcselby | lol - okay, so it was the obvious answer.... |
| 20:01.02 | VoipForces | hudony: you could have the supervison connect to the saterisk server using a softphone |
| 20:01.24 | wcselby | booting my phone now with the changed call-limit setting |
| 20:01.40 | VoipForces | wcselby: you did reload asterisk ? |
| 20:01.47 | wcselby | sip reload |
| 20:01.54 | VoipForces | wcselby: ok |
| 20:02.07 | hudony | Oh, I see |
| 20:02.17 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 20:02.40 | wcselby | btw, since this is a production server, it's still running v 1.2.27 (was the version on the server when they contracted me....we're building a new server running 1.4.26). |
| 20:02.51 | hudony | What about : when supervisor wants to listen in real time, u record the call then stream it with a sec or 2 interval ? |
| 20:03.33 | VoipForces | hudony: asterisk will do the recording in 2 files (in and out) and in wav. I don't think you can stream wav... |
| 20:03.35 | [TK]D-Fender | hudony: AMI Record + boss calls ChanSpy |
| 20:03.44 | *** join/#asterisk T` (n=total@pdpc/supporter/student/T) |
| 20:03.52 | T` | hi.. anyone here using asterisk with google voice? |
| 20:03.57 | jeff_phillips | hmm, the guy who wrote the timecard ivr that was broken says he'd charge $45 to fix it |
| 20:04.03 | *** part/#asterisk alecdavis (n=sivad@202-78-149-14.cable.telstraclear.net) |
| 20:04.05 | VoipForces | hudony: Of when the supervisor wants to listen-in, the asterisk box calls him on his cell. |
| 20:05.00 | wcselby | sip show peers shows my extension as unreachable again |
| 20:05.13 | hudony | OK, let me sort things out |
| 20:05.17 | T` | is there a way to make use of google voice without running asterisk locally? |
| 20:05.21 | VoipForces | wcselby: hmm can you ping the ip of that phone |
| 20:05.35 | T` | i have a netbook at my parents place.. and woudl like to setup Google voice so we can talk for free |
| 20:05.59 | wcselby | VoipForces - hmmmmm....now I can't, at least not from the * server |
| 20:06.01 | VoipForces | T`: Why go through asterisk for that, just to googlevoice to googlevoice |
| 20:06.05 | wcselby | I can ping it from my dhcp server |
| 20:06.11 | hudony | Fender : Can you be more precise please? |
| 20:06.14 | TSM2 | wcselby: yr not the first one to come up with problems with the IP 7000, someone else had a couple of them just die |
| 20:06.16 | ariel_ | tzafrir_laptop: you around? |
| 20:06.28 | VoipForces | wcselby: strange... faulty wire, faulty switch port? |
| 20:06.29 | hudony | Forces : you are saying that when superviser wants to listen in real-time, asterisk call him and bridge him to the conversation? |
| 20:06.47 | T` | VoipForces, well google voice doesn't allow connecting with a SIP phone like ekiga does it? |
| 20:06.48 | wcselby | now - lots of changs recently on the network - recently upgraded everything to a juniper network, which takes away CDP |
| 20:07.08 | wcselby | each phone plugs into a port that has a specific voice vlan on it |
| 20:07.13 | VoipForces | hudony: Kinda instead of calling a deskphone with the chanspy, just call the cell phone number. |
| 20:07.28 | TSM2 | wcselby: vlan easy to set on poly phones from DHCP if required |
| 20:07.47 | wcselby | TSM2 - how's that? |
| 20:07.57 | VoipForces | wcselby: Keep it simple. try that phone on a test server with no vlan or anything. |
| 20:08.12 | TSM2 | yup try that first |
| 20:08.27 | VoipForces | wcselby: True I did it on Aastra, butnot polycom. I don't really like polycom |
| 20:08.48 | wcselby | VoipForces - good idea. OH - meant to mention this - a softphone registers just fine using the 2570 user I had defined in sip.conf (I know, not really called user) |
| 20:09.32 | VoipForces | wcselby: you could easylly setup a vmware for this. I use that all the time to devbelop/test |
| 20:10.15 | VoipForces | T`: Dunnu, never tried it yet. |
| 20:10.20 | wcselby | VoipForces - good idea. I already have a virtualbox setup on my laptop with asterisk running, I should probably connect using that... |
| 20:10.23 | TSM2 | wcselby: http://pastebin.com/m522b455c |
| 20:10.25 | wcselby | thanks for the ideas guys |
| 20:10.51 | hudony | Forces : ok I dont know exactly which command to use to achieve that but I guess I can figure it out by myself. However, I'm wondering, why do you keep saying "cell phone" and not his desktop phone ? (sorry if this question sounds weird) |
| 20:10.54 | *** part/#asterisk h00k (n=anthony@unaffiliated/h00k) |
| 20:11.01 | TSM2 | wcselby: once the poly has got the VLAN info it will then DHCP again to get the new IP from the other VLAN |
| 20:11.41 | wcselby | TSM2 - is this like vlan tagging (cdp / lldp) ? |
| 20:12.10 | VoipForces | hudony: well, you said " I want to the superviser be able to listen to call not using his phone" so I presumed you did not want to use the deskphone and wanted an other maybe remote mean |
| 20:12.13 | wcselby | i.e can I trunk both data and voice onto a port, use this dhcpd config you just posted, and have the phone use the proper vlan? |
| 20:12.36 | hudony | Forces : ah ok, no, I was talking about his "normal phone" :) |
| 20:13.03 | wcselby | since we can't do that currently with the juniper switches (since they use lldp vlan tagging and the polycoms / 7960's don't support lldp, only cdp) ? |
| 20:13.06 | TSM2 | wcselby: yup, the phone then tags everything with the vlan you specified, the DHCP server on the other VLAN will assign the phone an IP, if you daisy chain a computer to the PC port on the phone it will just passthru any data from the PC without putting it in the same vlan as the phone |
| 20:13.21 | wcselby | ahhhhhhhh |
| 20:13.26 | VoipForces | hudony: Ok, then yes you could have his deskphone (sip) ring and when he picks up he is in chanspy. you can use a combination of originate, redirect for that |
| 20:14.02 | wcselby | they used to do that with the cisco switches they had, setup with cdp. when they put the junipers in, no one could figure out how to trunk both ports without using the tagging. i never wouldh ave thought of it this way... |
| 20:14.18 | TSM2 | wcselby: the phone touches nothing comming from the PC port, all it does is make sure the PC does not mess up with the QOS of the phone, ie the computer connected to the phone cant saturate the lan link and make the phone not work |
| 20:14.47 | wcselby | i'm sorry, i meant to say trunk both vlans |
| 20:15.04 | hudony | Forces : ok, I guess I'll go that way. Of course, I think that your softphone option would be more efficient since everything would be web-based and not "hybrid" (web/physical phone) but I have like no idea how to achieve that kind of setup (installing and configuring softphone etc.) and I have not much time left |
| 20:15.28 | TSM2 | wcselby: polys have good setup, the conf files are massive bunch of settings to mess things up |
| 20:15.46 | wcselby | tsm2 - believe me I know :) |
| 20:16.12 | *** join/#asterisk bbryant (n=brett@c-68-59-22-114.hsd1.sc.comcast.net) |
| 20:16.15 | wcselby | i'll probably be coming back here over the next few days asking questions...you guys are really helpful |
| 20:16.24 | VoipForces | hudony: use zoiper, it's easy to setup. just ccreate a sip or iax exteions in asterisk and point zoiper to it. |
| 20:16.37 | wcselby | i've got a lot of this stuff figured out, but there's still a lot left to learn apparently... :) |
| 20:16.39 | VoipForces | hudony: + zoiper runs on linux, windoze and mac. |
| 20:16.49 | hudony | ok |
| 20:16.52 | hudony | :) |
| 20:16.58 | hudony | This is really my first choice |
| 20:17.24 | hudony | Ok, I'll have a look at it. Thanks for your advices! |
| 20:17.30 | VoipForces | hudony: no prob and welcome to the asterisk world ! |
| 20:18.00 | *** join/#asterisk BlackHawk (n=marco@p5B07E4A0.dip.t-dialin.net) |
| 20:18.28 | TSM2 | wcselby: ive still got loads to learn, cant do asterisk stuff 100% as it needs to fit in with my other IT Manager work |
| 20:20.18 | BlackHawk | hello, could you please help me with a problem with asterisk? when I try to run it by executing 'asterisk -r' i get the error "unable to connect to remote server (does /var/run/asterisk/asterisk.ctl exist?)" and this file exists (although it is empty). any idea how to solve that? |
| 20:20.47 | VoipForces | BlackHawk: asterisk -r is to connevct to a running asterisk process console |
| 20:21.06 | VoipForces | BlackHawk: is that an asterisk distro or you have compiled it? |
| 20:21.11 | BlackHawk | i typed 'asterisk' before, but don't know whether it runs or not |
| 20:21.26 | BlackHawk | I installed it through yum |
| 20:21.52 | TSM2 | BlackHawk: service asterisk start |
| 20:21.52 | BlackHawk | the distro is centos 5.3 |
| 20:22.21 | BlackHawk | automatically restarting asterisk. asterisk died with code 1 ... |
| 20:22.25 | [TK]D-Fender | [16:15]<TSM2>wcselby: polys have good setup, the conf files are massive bunch of settings to mess things up <-- thats like saying they are so awesomely powerful they suck :) |
| 20:22.25 | wcselby | does he need to start dahdi / zaptel first? |
| 20:22.39 | BlackHawk | that's what it says, when I type your suggestion |
| 20:22.57 | wcselby | service dahdi start or service zaptel start <-- BlackHawk |
| 20:23.03 | tzafrir_laptop | ariel_, yes |
| 20:23.04 | wcselby | then try the service asterisk start |
| 20:23.24 | VoipForces | BlackHawk: what are the last like of /var/log/asterisk/full |
| 20:23.48 | BlackHawk | wcselby: it says 'unknown service' to both zaptel and dahdi |
| 20:23.55 | wcselby | ahhh |
| 20:24.46 | TSM2 | [TK]D-Fender: this is true as somepeople edit the sip.conf instead of creating a seperate override file and only put the settings they require, ive got a reasonable setup and keeps it simple |
| 20:24.48 | wcselby | i'd go with VoipForces - what's in the log file |
| 20:25.24 | BlackHawk | oh sorry, didn't read his suggestion :-$ |
| 20:25.52 | VoipForces | BlackHawk: bad configuration |
| 20:25.53 | *** part/#asterisk jeff_phillips (n=jeff_phi@209-142-149-133.stat.centurytel.net) |
| 20:25.55 | VoipForces | BlackHawk: Installing via yum might be a pain. you better download and compile from source, you will need dahdi, asterisk, libpri, asterisk-addons |
| 20:25.56 | BlackHawk | ok, 'full' doesn't exist |
| 20:26.03 | VoipForces | BlackHawk: It this is your first attempt at asterisk, then go and download AsteriskNow (the freepbx version) |
| 20:26.17 | ariel_ | tzafrir_laptop: any links for asteriskNOW to configure the astribank 2 |
| 20:26.23 | VoipForces | BlackHawk: what do you have as far as log files in /var/log/asterisk |
| 20:26.47 | tzafrir_laptop | ariel_, well, you basically need dahdi 2.2.0 |
| 20:26.49 | BlackHawk | VoipForces: actually it is my first time, but I'm just helping a friend, who isn't that good at english to talk in it ;) |
| 20:27.09 | ariel_ | tzafrir_laptop: got that |
| 20:27.40 | VoipForces | BlackHawk: This is for a production business server or for home hoby? |
| 20:27.46 | [TK]D-Fender | ok, checkout time, later all... |
| 20:27.50 | BlackHawk | in /var/log/asterisk there is cdr-csv, cdr-custom, event_log, messages and queue_log |
| 20:28.05 | VoipForces | VoipForces: Cause I would really suggest that you start with something like AsteriskNow |
| 20:28.17 | wcselby | i'll be back alter, thanks guys :) |
| 20:28.34 | BlackHawk | it's kind of a bussiness ;) |
| 20:28.39 | BlackHawk | -s |
| 20:28.43 | VoipForces | BlackHawk: ok, you need to edit /etc/asterisk/logger.conf and enable the full debugging then restart asterisk and see the last lines in /var/log/asterisk/full |
| 20:29.12 | BlackHawk | VoipForces: my friend rejects using asterisknow, because he uses a root server and says there is no asterisknow available for such things |
| 20:29.37 | *** join/#asterisk h00k (n=anthony@unaffiliated/h00k) |
| 20:29.47 | VoipForces | BlackHawk: a root server??? what do you mean? AsteriskNow is based on CentOS |
| 20:30.30 | BlackHawk | well, that's just what he said ... I don't know what he meant ... |
| 20:32.28 | VoipForces | BlackHawk: well, asterisk is not just a software, it's a communication framework, and just installing from yum will do nothing. you will have to work on most if not all config files just to get asterisk started, then you will have to get through the configuration for the dialplan and extensions. |
| 20:32.41 | VoipForces | BlackHawk: What will be your primary use for asterisk? |
| 20:33.00 | BlackHawk | VoipForces: ok, it's my fault, he meant this root server is accessable in the web |
| 20:33.29 | *** join/#asterisk timeshell (n=chatzill@gw.lusi.on.ca) |
| 20:33.39 | BlackHawk | ok, so he should compiel it from source, right? |
| 20:33.53 | VoipForces | BlackHawk: Still don't get it. If you do not put your server on the internet unprotected, this is not an issue. Even them you can set apache to properly secure it. |
| 20:34.35 | VoipForces | BlackHawk: Check this howto http://www.freepbx.org/support/documentation/installation/install-process-for-centos-5-1 |
| 20:34.51 | BlackHawk | ok :) thank you! I'll try |
| 20:35.08 | VoipForces | BlackHawk: It will show you how to install asterisk + freepbx (which is to my opinion the best gui to manage asterisk) on a centos server |
| 20:36.05 | BlackHawk | so he tried asterisk several times on a debian-system and had no problems there, but now he has to use centos for a groupware |
| 20:36.20 | BlackHawk | (this groupware is already ready to run) |
| 20:36.36 | BlackHawk | just asterisk doesn't work as it should :/ |
| 20:38.12 | VoipForces | BlackHawk: Actually I had much better luck so far on centos than on debian. But again I never use yum. I compile from source. |
| 20:39.15 | BlackHawk | ok, I'll tell my friend that and hope he will accept that advice :) |
| 20:40.57 | *** part/#asterisk oftis (n=nicok@dslb-094-217-064-041.pools.arcor-ip.net) |
| 20:41.30 | Katty | :> |
| 20:42.29 | *** join/#asterisk dmz (n=dmz@64.203.233.195.dyn-cm-pool35.hargray.net) |
| 20:42.56 | VoipForces | Katty: quite heu |
| 20:44.58 | jaytee | what a day! |
| 20:45.20 | iksik | hey |
| 20:45.26 | iksik | any asterisk gui users here? |
| 20:45.59 | Katty | heu? |
| 20:46.01 | Nugget | gvim is the closest I come to a gui. |
| 20:46.12 | BlackHawk | VoipForces: thanks a lot for your help :) he'll try compiling it |
| 20:46.15 | VoipForces | i'm here for an other 10-15 minutes |
| 20:46.16 | Katty | jaytee: you can say that again. |
| 20:46.22 | Katty | jaytee: i'm quite preturbed at the office manager. |
| 20:46.25 | TSM2 | iksik: yup, you mean freepbx |
| 20:46.59 | jaytee | Katty, preturbed? you mean perturbed don't you? or is this some kind of perturbance in advance? |
| 20:47.13 | iksik | TSM2 it this one http://www.asterisknow.org/install-related is a freepbx, ok, I mean freepbx |
| 20:47.26 | Katty | jaytee: you know what i mean, don't confuse me with the facts. |
| 20:47.34 | jaytee | :-) |
| 20:47.45 | Katty | jaytee: i will get over it, i'm sure. |
| 20:47.55 | jaytee | "This too shall pass!" |
| 20:47.56 | Katty | jaytee: policies just tend to be highly inconsistent. |
| 20:48.01 | *** join/#asterisk jasonwoot (n=some@69.73.89.233) |
| 20:48.03 | TSM2 | iksik: whats the problem |
| 20:48.13 | iksik | Registration from '6010 <sip:6010@my.host.here>' failed for 'IP.IP.IP.IP' - No matching peer found |
| 20:48.15 | TSM2 | iksik: there is a #freepbx channel |
| 20:48.15 | Katty | jaytee: getting ANYTHING in writing is like ...pulling teeth. |
| 20:48.19 | Katty | jaytee: aligator teeth |
| 20:48.24 | *** join/#asterisk jtodd (i=zntl6kyv@ns.fox-den.com) |
| 20:48.24 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 20:48.50 | *** join/#asterisk haryv (i=lanny@S010600a0c93f6f7e.vs.shawcable.net) |
| 20:49.06 | *** part/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 20:49.12 | VoipForces | iksik: ok, do you have the sip peer created |
| 20:49.22 | iksik | yes |
| 20:49.33 | jaytee | Katty, could be worse.....could be like pulling teeth from a giant sandworm on Arrakis. |
| 20:49.34 | iksik | TSM2 http://blog.tmcnet.com/blog/tom-keating/images/asterisk-gui-20-conference-bridge.png - it's freepbx / |
| 20:49.35 | iksik | ? |
| 20:49.58 | VoipForces | iksik: do you see if if you do sip show peers |
| 20:50.04 | iksik | nope |
| 20:50.08 | iksik | and that is weird |
| 20:50.10 | Katty | jaytee: GREAT MAKER |
| 20:50.13 | Katty | jaytee: THE SPICE MUST FLOW |
| 20:50.27 | VoipForces | iksik: do you have it in /etc/asterisk/sip_additional.conf |
| 20:50.30 | haryv | Is there a way for ast to dial a busy number over and over ? |
| 20:50.34 | jaytee | I just got through reading the prequel Houses trilogy |
| 20:50.42 | iksik | VoipForces I have it in /etc/asterisk/users.conf |
| 20:50.52 | Katty | jaytee: :> |
| 20:50.54 | VoipForces | iksik: no that is for the queues |
| 20:51.00 | Katty | jaytee: you have much to look forward too. |
| 20:51.05 | iksik | VoipForces and there is no sip_additional.conf file |
| 20:51.10 | Katty | jaytee: especially the bene gezerit |
| 20:51.11 | jaytee | Katty, how so? |
| 20:51.17 | Katty | jaytee: and the sardukar |
| 20:51.23 | Katty | and the twins! |
| 20:51.30 | VoipForces | iksik: sip.conf |
| 20:51.49 | iksik | VoipForces I see it when I type: sip show objects |
| 20:52.00 | *** join/#asterisk spck (n=spck@unioncab.com) |
| 20:52.08 | jaytee | Katty, I've already read Dune, Dune Messiah, Children of Dune over 20 years ago. |
| 20:52.14 | spck | afternoon guys |
| 20:52.27 | Katty | jaytee: oh :< |
| 20:52.32 | *** join/#asterisk Anth8708 (n=Anth8708@client105.jdcc.edu) |
| 20:52.36 | Katty | jaytee: does that mean you no longer look forward to reading it again? |
| 20:52.48 | jaytee | just reread it about a month ago |
| 20:52.55 | VoipForces | iksik: sip show objects??? |
| 20:53.00 | iksik | yea :| |
| 20:53.07 | Katty | oh :< |
| 20:53.14 | jaytee | after I finished the other trilogy prequel, The Butlerian Jihad, The Machine Crusade and The Battle of Corrin |
| 20:53.15 | Katty | well what will you read now? :< |
| 20:53.28 | iksik | VoipForces, but I think I should see it in: sip show users - right? |
| 20:53.31 | Katty | you could read xanth novels. |
| 20:53.34 | jaytee | well, I never read Dune: Chapterhouse or Heretics of Dune |
| 20:53.36 | Katty | those are always fun. |
| 20:53.37 | VoipForces | iksik: ok, just saw your screen dump, this is not freepbx, this is the lame digium asterisk-gui |
| 20:53.39 | jaytee | so I'll read those |
| 20:53.42 | Katty | mkay. |
| 20:53.47 | citywok | when the queue_log rolls over at night, what happens to scripts that are attempting to Tail it? will they still be tailing the new log or will the tail get broken |
| 20:53.59 | VoipForces | Katty: That one uses asterisk realtime engine, which I have never used and will never use |
| 20:54.12 | spck | anyone ever notice any differences between parking a call with a phone vs the management interface? |
| 20:54.16 | iksik | mabe lame, but it has more usabillity then freepbx can have ever ;P |
| 20:54.53 | jaytee | freepbx uses asterisk realtime engine? |
| 20:55.11 | Katty | VoipForces: do what? what on earth are you talking about? |
| 20:55.23 | Katty | VoipForces: X(an)^th? |
| 20:55.26 | VoipForces | jaytee: no digium asterisk-gui does |
| 20:55.40 | jaytee | ah |
| 20:55.53 | dustybin | is there a such thing as a irssi >> asterisk script |
| 20:55.57 | VoipForces | iksik: no way. freepbx has MUCH more that asterisk-gui |
| 20:56.00 | Katty | wonders if she's having multiple conversations with people and not realizing it. |
| 20:56.26 | VoipForces | Katty: Sorry clicked on the wrong user |
| 20:57.25 | Katty | ohisee :> |
| 20:58.52 | jaytee | Katty, so is scifi your favorite genre for reading? |
| 20:59.18 | iksik | VoipForces, has MUCH more what? |
| 20:59.24 | iksik | oh damn, to late |
| 20:59.45 | jaytee | more features |
| 21:00.02 | iksik | I was talking about USABILLITY, not about features :| |
| 21:00.29 | jasonwoot | is tollfreeforwarding.com worth their weight in used pinball machine parts? |
| 21:00.33 | dustybin | can one use asterisk using terminal only? |
| 21:01.03 | dustybin | to hell with it, why dont i just install it dog |
| 21:01.04 | dustybin | doh |
| 21:01.04 | beek | jasonwoot: "their weight in used pinball machine parts" could be fairly valuable in my estimation... |
| 21:01.28 | beek | dustybin: You mean someone uses Asterisk in something other than text mode? |
| 21:01.56 | dustybin | i want text mode only |
| 21:02.03 | dustybin | GUIs are not good |
| 21:02.08 | beek | The way god intended Asterisk to be run... |
| 21:02.15 | dustybin | ace |
| 21:05.03 | jaytee | quittin time, back later |
| 21:05.28 | dustybin | should i compile asterisk myself, or use a ready-built debian package? |
| 21:05.36 | citywok | compile it dustybin |
| 21:05.43 | dustybin | any reason why? |
| 21:05.50 | citywok | i spent 2 hours fighting mine before spending 5 minutes to compile it from scratch, which fixed all my issues |
| 21:05.58 | dustybin | I see |
| 21:06.12 | dustybin | what distro? |
| 21:06.14 | citywok | 1.6 clean imaged server installed broken with the deb |
| 21:06.20 | citywok | debian lenny |
| 21:06.42 | dustybin | right ok! |
| 21:06.56 | *** part/#asterisk FinboySlick (n=shark@207.134.8.4) |
| 21:07.10 | dustybin | citywok: there are a _lot_ of debian packages, are you sure you picked the correct one? |
| 21:07.13 | dustybin | http://paste.debian.net/43828/plain/43828 |
| 21:07.27 | dustybin | when i see that many variations of a package, it does make me think that this should be compiled manually |
| 21:07.35 | citywok | it's tough to screw up apt-get install asterisk |
| 21:07.47 | *** join/#asterisk TSM2 (n=the_soft@87-194-32-212.bethere.co.uk) |
| 21:08.01 | beek | dustybin: Just compile it... it's so much easier in the end. |
| 21:08.07 | dustybin | yar i will |
| 21:08.18 | citywok | yea, it literally took less than 5 minutes to download, compile, install the entire thing |
| 21:08.18 | eppigy | TRABAJO |
| 21:08.27 | dustybin | asterisk-mp3 wtf |
| 21:08.34 | dustybin | wtf has mp3 got to do wth asterisk |
| 21:08.34 | citywok | though i'm running pure sip so i didnt have to wait for zaptel/dahdi to compile |
| 21:08.42 | citywok | dustybin: music on hold |
| 21:08.50 | dustybin | oh ace :D |
| 21:09.18 | exsync | what's the fail over destination "CONGESTION" do? |
| 21:09.19 | dustybin | i will compile the latest stable version |
| 21:10.03 | dustybin | the only packages i compile are mythtv, asterisk and zoneminder |
| 21:10.21 | citywok | i'd compile windows7 mce but well, oh yea, you dont need to :P |
| 21:10.43 | dustybin | i tried compiling windows 7, however it crashed |
| 21:11.12 | dustybin | WINDOWS 7 ................. [FAIL] |
| 21:11.28 | dustybin | who the hell uses ISDN in 2009 ? |
| 21:11.42 | exsync | dude, 45kbps, dont hate |
| 21:11.50 | exsync | channel bonded baby |
| 21:11.58 | exsync | er, 64kbps |
| 21:14.11 | dustybin | why does one need mysql support for asterisk? |
| 21:15.54 | TSM2 | dustybin: most people PRI & T1 |
| 21:16.23 | TSM2 | dustybin: correct def for single ISDN is BRI |
| 21:16.30 | dustybin | aye ok |
| 21:16.36 | TSM2 | dustybin: BRI is hardly used anymore |
| 21:16.41 | dustybin | one last question, what langauge is asterisk written in? |
| 21:16.50 | TSM2 | dustybin: english :) |
| 21:16.52 | dustybin | LOL |
| 21:16.53 | TSM2 | dustybin: duno |
| 21:16.59 | dustybin | falls off chair |
| 21:17.17 | TSM2 | dustybin: prolly C or C++, somthing fast like that |
| 21:17.21 | dustybin | TSM2: how can you use a bit of software without knowing what langage it was written in? |
| 21:17.32 | Katty | eppigy: HUGJO |
| 21:17.40 | TSM2 | dustybin: i think its C as it uses GCC to compile i think |
| 21:17.44 | Katty | eppigy: what are you making me for dinner. |
| 21:17.44 | dustybin | ok |
| 21:18.05 | TSM2 | dustybin: ive not had to compile asterisk yet, i dont write C/C++ |
| 21:18.47 | TSM2 | dustybin: i learn well when i need to do work in a language i have not used, so far in the company im in ive not had to do C, mainly perl/php/VB |
| 21:19.12 | eppigy | Katty: whatever you want :> |
| 21:19.52 | Katty | good answer. good answer. |
| 21:20.01 | Katty | but srsly, now. |
| 21:20.05 | Katty | for reals. |
| 21:34.21 | *** join/#asterisk manxpower (n=EWieling@69.73.94.162) |
| 21:37.02 | *** join/#asterisk puzzled (n=foobar@puzzled.xs4all.nl) |
| 21:40.02 | ruben23 | hi |
| 21:40.17 | ruben23 | any idea how do i setup |
| 21:40.27 | ruben23 | ip kall on my softphoenes |
| 21:40.47 | *** join/#asterisk davidandgoliath (n=david@S0106001d60d4e488.vn.shawcable.net) |
| 21:42.49 | davidandgoliath | anyone want to give some voip advice to a morepawn? :p |
| 21:43.23 | davidandgoliath | I own a red stapler if it makes a difference. |
| 21:43.26 | *** part/#asterisk hesco (n=hesco@24.99.160.121) |
| 21:44.50 | exsync | what's the question, i'm learning myself |
| 21:44.58 | davidandgoliath | Actually, few errands first. Will pester you geniuses in a bit! :) |
| 21:45.12 | davidandgoliath | Was going to ask some q's about infrastructure recommendations and such, vm / phones |
| 21:45.15 | exsync | yeah, you'd better run |
| 21:45.19 | davidandgoliath | :D |
| 21:54.21 | manxpower | ~ask |
| 21:54.21 | 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. |
| 21:55.17 | *** join/#asterisk ACK-NAK (n=Miranda@home.chicagoventure.com) |
| 21:57.18 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 21:57.40 | *** part/#asterisk beek (n=klinebl@pdpc/supporter/professional/beek) |
| 21:58.57 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 22:02.37 | ACK-NAK | Our PRI is not given CNAM from the telco. Instead we look it up ourselves. We populate CALLERID(name), and THEN place a call via a second PRI span to a legacy PBX. Problem: The PBX isn't getting the CALLERID(name) string. Name still shows as the number. Ideas? |
| 22:03.10 | ACK-NAK | cli> pri debug span 2: shows |
| 22:03.22 | *** join/#asterisk propellerhead (n=yogurt2u@host109.190-31-68.telecom.net.ar) |
| 22:04.20 | ACK-NAK | the looked-up CNAM string right there in the Q.931 messages |
| 22:05.29 | ACK-NAK | Does the caller ID name need to be assigned to any special variable or 'optional' callerid parameter in order to be available to a normal NI2 client pbx? |
| 22:06.09 | manxpower | there are two ways to send Caller*ID Name on PRI. Also you don't have any extra quotes in the CLID do you? |
| 22:06.21 | manxpower | Don't ask me about the difference in the two ways, I have no idea. |
| 22:07.55 | *** join/#asterisk russellb_ (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 22:07.55 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 22:07.55 | ACK-NAK | manxpower: the q.931 message says > Display (len=10) Charset: 31 [ SMITH,DICK ] |
| 22:08.04 | ACK-NAK | no quotes |
| 22:08.27 | manxpower | ACK-NAK: can you ask your PBX vendor to accept the CLID Name "the other way"? 8- |
| 22:09.33 | ACK-NAK | I'll ask him. |
| 22:09.37 | ACK-NAK | THanks |
| 22:10.11 | manxpower | ACK-NAK: if you can't get it working by tomorrow evening let me know, I might be able to look up the correct terms for the two types of CLID |
| 22:11.31 | ACK-NAK | manxpower: Thanks a lot. I really appreciate it. Have a great evening! |
| 22:13.00 | *** join/#asterisk NovceGuru (i=novcegur@server1.jsreedinc.com) |
| 22:14.45 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
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| 22:27.26 | citywok | AgentCallbackLogin was removed from asterisk1.6 -- what is supposed to be used in its place? i can't seem to find anything voip-info says Removed 1.6, but doesnt give an alternate solution |
| 22:28.04 | manxpower | citywok: That information should be in the UPGRADE*.txt files included in the Asterisk source. |
| 22:28.49 | citywok | perfect, ty |
| 22:33.06 | citywok | hrmm, doesnt say anything about agentcallbacklogin in UPGRADE-1.6.txt and if i look in the docs folder and read queues it says read more in queues-with-callback-members.txt which doesn't exist. upon googling that file it appears to be a txt file that was in the 1.4 source tree. probably references agentcallbacklogin lol |
| 22:35.39 | citywok | ahhhh according to more googling "Deprecation of AgentCallbackLogin in favor of a dialplan-based solution " |
| 22:37.07 | manxpower | Oh, I knew *that*. I just don't know how you would do that in the dialplan. 8-) I'm not a big fan of queues, and I've not needed to use them. Very simple queues can be emulated in the dialplan. |
| 22:38.11 | citywok | most of my queues are pretty basic, and i went back and forth for a bit trying to decide which way to go. realistically it may be easier to just dial(SIP/1&SIP/2&SIP/3&SIP/4&SIP/5) -> voicemail |
| 22:38.36 | manxpower | chan_local can be VERY helpful. |
| 22:38.56 | manxpower | also checking the value of DIALSTATUS after each dial. |
| 22:39.11 | citywok | do you hvae a way to emulate RRMEMORY of queues? |
| 22:40.13 | manxpower | I assume you want Round Robin with Memory. |
| 22:40.32 | citywok | yea, precisely -- rrmemory is how it's defined in a queue |
| 22:40.43 | manxpower | That would be complicated to do in the dialplan, but not impossible. |
| 22:40.51 | manxpower | let me think a moment |
| 22:40.53 | citywok | probably astdb it is my only idea |
| 22:41.51 | manxpower | You could use astdb or global variables, locked using MacroExclusive to prevent race conditions |
| 22:43.31 | manxpower | My customers did not require memory. Basically it was ring the first line on the operator phone, if that line is busy, go to the next one, of that line was no answer, then send to the backup operator, if no answer there send to their supervisor so their supervisor gets mad and deals with the issue of the operators not answering the phone. If all else fails, send the call to voicemail. |
| 22:43.38 | citywok | currently to get agentcallbacklogin i'm using queues and assigning each phone directly to the queues which works in my environment, but i've been exploring whether or not this was necessarily the best solution |
| 22:43.49 | *** join/#asterisk ingenius (n=alektro@host215.200-45-165.telecom.net.ar) |
| 22:44.09 | haryv | when the bos complaints about fw and ast running on seperate boxes ..she wants both to reduce the anuall power consumption. |
| 22:44.13 | manxpower | citywok: are people answering queue calls supposed to answer them any time they are at their desk? |
| 22:44.13 | citywok | the more i look at it, the more i think that it probably is the most elegant solution |
| 22:44.40 | citywok | manxpower: yes, if they are not at their desk they have logged in to break, which closes out their phone for them |
| 22:44.54 | haryv | Im not a master in ipchains but if anyone here has it and squid running on there asterisk box let me know :) |
| 22:45.12 | manxpower | I wonder if you could just have the queue move to the next if their line is "busy", then have them DND when they are away from desk |
| 22:45.22 | citywok | haryv: what are you trying to do? i've actually done that before, it's not too terribly difficult and shouldn't take anything fancy |
| 22:45.42 | haryv | squid/ipchains running on ast box |
| 22:45.54 | citywok | yea, thats the idea. we tossed around DND/close out the phone entirely and they were okay to either solution |
| 22:46.17 | haryv | so it worked for you and no security issues? |
| 22:46.36 | citywok | iptables -P INPUT DROP; iptables -A INPUT -s <internallan> -j ACCEPT |
| 22:47.19 | manxpower | you never want to blanket disable ICMP |
| 22:47.28 | T` | anyone here use ipkall? |
| 22:48.29 | citywok | manxpower: yes, -p icmp -j accept is generally in there, that was a glorified example of how to make asterisk not complain about being on a firewalled machine :P |
| 22:49.41 | citywok | i've blocked it a few times in transparent firewall situations where i didnt want anybody to know there was a box in the middle, and had it 99.9% locked down except for a couple ip addresses that could get to the management ip on the bridge |
| 22:51.14 | manxpower | blocking stuff like packet-too-big causes major lag and packet loss when talking to a host behind a connection with a smaller MRU than you are at. |
| 22:51.17 | citywok | i think i'll just stick to using call queues & control when agents are/aren't at their desks through their interfaces to make sure it doesnt ring their phone when they arent around. i prefer to completely log them out of their phone so that the queue will exit when it is empty. would the queue exit out if all the members were on DND? if so, that would be fine as well |
| 22:52.13 | citywok | i've never played with a firewall in an environment where i had an MTU that wasn't the standard (1500 otoh?) |
| 22:52.45 | *** join/#asterisk timeshell (n=chatzill@206.248.136.108) |
| 22:53.15 | manxpower | citywok: lots of sites have an mtu smaller than 1500. |
| 22:53.32 | manxpower | in that case they will almost be inaccessable if you block icmp-packet-too-big |
| 22:53.41 | manxpower | It's a classic network admin newbie mistake. |
| 22:54.00 | *** join/#asterisk LiNeTuX (n=LiNeTuX@119.153.205.68.cfl.res.rr.com) |
| 22:54.03 | haryv | what not backing up? |
| 22:54.13 | manxpower | haryv: blocking all ICMP |
| 22:54.15 | bmoraca | citywok: you've never worked over a VPN or DSL line? those all have MTUs less than 1500. 1500 is ethernet, but a lot of layer 2 media have smaller frame sizes. |
| 22:54.17 | haryv | ahh |
| 22:54.56 | citywok | bmoraca: i work with connections from T1 -> DS3 -> 100mbit eth handoff |
| 22:55.46 | citywok | i use openvpn a bit to connect work from home users to our office using DD-WRT boxes (suppper slick setup), and haven't had any issues |
| 22:56.15 | *** part/#asterisk grey-monkey (n=ericshel@75-148-103-190-Utah.hfc.comcastbusiness.net) |
| 22:58.28 | citywok | oh i see, it's the opposite direction of what i was thinking. i've always got related,established -j ACCEPT rules which would keep that from being a problem, even if you left icmp disabled |
| 23:13.32 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
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| 23:42.37 | *** join/#asterisk drclue (n=drclue@ip-65-49-160-6.wireless.dyn.beamspeed.net) |
| 23:43.44 | drclue | Howdy all. A strange thing happened today in the course of installing my new Digium card. After installing the card and running the update scripts and such |
| 23:43.45 | drclue | my SIP phones still work , except when I Dial an extension from FastAGI. In that instance I get one way audio, yet if the SIP phones call directly audio is fine |
| 23:43.45 | drclue | All was working fine before the card install and the updates, so not sure why the connection behavior changed. Seems to have something to do with native bridging |
| 23:43.45 | drclue | Any thoughts on this? |
| 23:46.00 | manxpower | drclue: what is that actual Dial statement your AGI is doing? |
| 23:47.01 | drclue | The Dial statement is the same as it was yesterday , but it's Dial SIP/2000|20|m |
| 23:48.29 | manxpower | are any of the phones on a different network AND behind NAT? |
| 23:49.39 | drclue | The phones are both on a different network and run through NAT , but if I use the regular extension numbers to dial them they work just fine, and it all worked fine this morning |
| 23:50.28 | manxpower | Try putting canreinvite=no in [general] in sip.conf |
| 23:52.23 | drclue | I did put canreinvite=no individually for the two extensions , and that did not seem to work , but I could try setting it in general if you think that would work any better than having set it off individually. Again these were working this morning with canreinvite=yes |
| 23:52.58 | manxpower | I can't comment on "it worked before". |
| 23:53.05 | manxpower | Did you upgrade Asterisk? |
| 23:53.58 | *** join/#asterisk puzzled (n=foobar@puzzled.xs4all.nl) |
| 23:54.18 | drclue | I did do the update-source etc etc , but other then that it is the same asterisk 1.6 I started with |
| 23:56.45 | drclue | canreinvite=no in sip.conf [general] has no effect on the FastAGI dialed call |
| 23:57.01 | drclue | Still one way audio |
| 23:57.10 | manxpower | I'm not familiar with "update-source" but if you changed the code, then something changed. |
| 23:57.27 | LiNeTuX | manxpower: update-source = PIAF |
| 23:57.32 | manxpower | is there an "unupdate-source" script? |
| 23:57.47 | manxpower | Pain In The Ass F??? |
| 23:57.47 | drclue | The PIAF distro has some utility scripts for getting updates |
| 23:57.59 | manxpower | Sorry, I can't help with distro stuff. |
| 23:58.43 | LiNeTuX | drclue: hop over to #pbxinaflash - I might be able to give you some insight |
| 23:58.48 | drclue | The phones still register and can dial each others extensions without any audio issues. It's just when the native bridging occurs that the one way audio happens |
| 23:59.32 | drclue | LiNeTux - OK , I'll pop a window open for #pbxinaflash |
| 23:59.46 | manxpower | drclue: I've seen bugs that only happen every 100th call. Or only when someone does two transfers in a row. Don't expects bugs to be logical in their symptoms. |
