| 00:01.15 | ariel_ | pussfeller, exten => s,1,AbsoluteTimeout(120) exten => s,2,NoOp |
| 00:01.47 | ariel_ | after that menu or segment you should put a T extension |
| 00:02.32 | pussfeller | thanks |
| 00:02.55 | pussfeller | i do that immediately after the voicemail command? |
| 00:03.19 | pussfeller | or before, |
| 00:03.25 | ariel_ | before |
| 00:03.58 | ariel_ | you set your timeout before you do something. |
| 00:03.58 | *** join/#asterisk tld (~tld@80.203.70.227) |
| 00:05.19 | pussfeller | ok |
| 00:05.35 | pussfeller | sothen the T references the absolute timeout you set |
| 00:07.10 | *** join/#asterisk abatista (~Ariel@dsl-20-177.cofs.net) |
| 00:09.27 | nords | any ideas on how i can get more debug information on figuring out why the channel gets hung up. I have the verbosity already set to 23 |
| 00:09.53 | *** join/#asterisk newbien (~e@147.241.33.65.cfl.res.rr.com) |
| 00:10.14 | *** join/#asterisk ortel (~ortel@c-24-15-69-152.hsd1.il.comcast.net) |
| 00:10.46 | ortel | hello, would anyone know of a good howto file for the tdm400 from digium? |
| 00:11.46 | ariel_ | nords, iax2 debug |
| 00:12.44 | nords | that gives me the iax2 transport debug, but will that identify why it is hanging up. I thought this would be more of an asterisk issue |
| 00:14.44 | ariel_ | it's a start. your channels are iax2 do you have notransfer=yes? |
| 00:15.36 | *** join/#asterisk gaggaman (~ga@host-82-135-28-39.customer.m-online.net) |
| 00:16.02 | gaggaman | Hi all! |
| 00:16.06 | gaggaman | anybody here with some experience on ISDN NT-P2MP ? |
| 00:16.53 | gaggaman | I have a ISDN phone wired up with one of the ISDN cards and setup an extension (with amportal). |
| 00:17.04 | gaggaman | now I can dial a number on the isdn phone and then pick up the isdn phone. everything works |
| 00:17.12 | gaggaman | but: |
| 00:17.22 | gaggaman | if I first pick up and dial afterwards, I end up nowhere. |
| 00:17.44 | gaggaman | <PROTECTED> |
| 00:17.44 | gaggaman | <PROTECTED> |
| 00:17.44 | gaggaman | <PROTECTED> |
| 00:17.44 | gaggaman | <PROTECTED> |
| 00:17.45 | gaggaman | <PROTECTED> |
| 00:17.45 | gaggaman | <PROTECTED> |
| 00:17.53 | gaggaman | any ideas? |
| 00:19.12 | *** join/#asterisk Sedorox (brandon@sedorox.staff.smartserv) |
| 00:19.18 | gaggaman | I am using zaptel / bristuff / zaphfc |
| 00:22.10 | nords | ariel_, i didn't explicty set that option. When i look at the iax2 debug info there isn't anything in there out of the ordinary |
| 00:27.39 | MikeJ[Laptop] | gaggaman, use pastebin please |
| 00:27.42 | MikeJ[Laptop] | ~pastebin |
| 00:27.44 | jbot | rumour has it, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 00:27.55 | *** join/#asterisk easimon (~easimon@localhorst.kawo2.RWTH-Aachen.DE) |
| 00:29.38 | *** join/#asterisk Amino_ (~amino@host217-44-61-133.range217-44.btcentralplus.com) |
| 00:29.54 | MikeJ[Laptop] | gaggaman, it looks like your phone implements immediate if you pick it up before dialing. You are going to either need to turn that off, or do somthing in exten s to wait for a dial string |
| 00:31.45 | gaggaman | sorry for flooding. |
| 00:32.09 | gaggaman | seems I can't switch that of on ISDN phones. |
| 00:32.44 | gaggaman | if I dial after picking up, it still does not dial with dtmf tones. |
| 00:33.22 | gaggaman | if I press a number, I again get "Executing Macro("Zap/8-1", "hangupcall") in new stack" ... |
| 00:37.55 | MikeJ[Laptop] | ok, so do somthing in your dialplan on s to snag those dtmf and do somthing with them |
| 00:39.17 | gaggaman | eh, no, i *don't* get dtmf! |
| 00:39.24 | *** join/#asterisk pcm (~pcm@user-69-73-0-22.knology.net) |
| 00:40.05 | gaggaman | so s doesn't seem to be an option. |
| 00:40.36 | MikeJ[Laptop] | yeah, cuz s is doing somthing |
| 00:40.41 | MikeJ[Laptop] | pastebin your config |
| 00:41.00 | MikeJ[Laptop] | extensions.comf |
| 00:41.01 | MikeJ[Laptop] | conf |
| 00:42.00 | gaggaman | I made the .conf with amportal. thus, it's quite long :-) |
| 00:43.11 | MikeJ[Laptop] | well... hmmmm |
| 00:43.15 | MikeJ[Laptop] | lets see it |
| 00:46.49 | rabelais | does iax.cc do number portability transfers? I have a broadvoice number, 310-598 that I want to transfer to iax.cc |
| 00:47.57 | Mavvie | I think you should ask that question to the people from iax.cc |
| 00:49.42 | tzanger | I'm so torn |
| 00:49.51 | tzanger | I hate debian's politics but they ship postgresql by default... :-) |
| 00:50.52 | Nugget | so conflicted. :) |
| 00:51.12 | tzafrir_laptop | tzanger, what's the problems with Debian's politics? |
| 00:51.22 | gaggaman | ok |
| 00:51.28 | gaggaman | here: http://pastebin.ca/11713 |
| 00:51.28 | tzafrir_laptop | Actually, I see even more politics elsewhere with Asterisk |
| 00:51.40 | Nugget | debian are gpl fanatic ossholes. |
| 00:51.41 | tzanger | GNU/oh GNU/nothing, GNU/I just GNU/get GNU/sick of GNU/their GNU/pandering to GNU/GNU |
| 00:51.43 | Nugget | but postgresql is good. |
| 00:51.54 | tzanger | ossholes? HAAAAAAAAAAAAAAAAHAHAHAHAHAHHA |
| 00:52.38 | *** join/#asterisk Newbie___ (~me@60.48.43.123) |
| 00:52.44 | gaggaman | 24 is the extension of the isdn phone that makes the trouble. |
| 00:52.46 | tzafrir_laptop | tzanger, you mean debian gnu/freebsd? |
| 00:52.51 | tzafrir_laptop | ;-) |
| 00:52.55 | tzanger | :-) |
| 00:53.03 | gaggaman | it is on zap/g2 |
| 00:53.08 | tzanger | hmm |
| 00:53.21 | tzanger | how do I query apt for a package? I want to install mysql but not sure how |
| 00:53.32 | tzafrir_laptop | apt-cache show packagename |
| 00:53.32 | JerJer | apt-get install mysql |
| 00:53.36 | MikeJ[Laptop] | gaggaman, what context does that phone hit? |
| 00:53.40 | tzanger | Package mysql is not available, but is referred to by another package. |
| 00:53.40 | tzanger | This may mean that the package is missing, has been obsoleted, or |
| 00:53.40 | tzanger | is only available from another source |
| 00:53.40 | tzafrir_laptop | apt-cache policy packagename |
| 00:53.43 | tzanger | I get that back |
| 00:53.47 | gaggaman | mom... |
| 00:53.51 | tzafrir_laptop | apt-cache -n search packagename |
| 00:53.52 | JerJer | apt-get install mysql-server |
| 00:54.12 | tzanger | there we go |
| 00:54.13 | tzanger | thanks |
| 00:54.29 | gaggaman | MikeJ: the context is "from-internal" |
| 00:55.16 | MikeJ[Laptop] | ok |
| 00:55.56 | MikeJ[Laptop] | can you make a specific phone use a diff context in amp? |
| 00:56.27 | MikeJ[Laptop] | exten => s,1,Macro(hangupcall) |
| 00:56.49 | *** join/#asterisk tsp (~tyler@S01060080c825173c.vc.shawcable.net) |
| 00:57.02 | tsp | how do I play a sound file to a caller I'm calling on the console? |
| 00:57.12 | file | I have seen the latest episode of Enterprise my friends |
| 00:57.18 | file | and I have one thing to say, "crap" |
| 00:57.40 | denon | eh, I thought that show has been lame for several years now? |
| 00:57.43 | denon | havent seen it in a long time |
| 00:57.43 | tsp | how do I play a sound file to a caller I'm calling on the console? |
| 00:57.49 | file | denon: the ending was cheap |
| 00:57.53 | Mavvie | tsp: you're repeating yourself. |
| 00:57.58 | tzafrir_laptop | anyway, I've just written a man page for astgenkey, as lintian told me that this was a binary without a man page |
| 00:58.05 | gaggaman | MikeJ yes I can specify the context of a specific phone. |
| 00:59.02 | MikeJ[Laptop] | ok, so make your own context.. you can use somthing like this |
| 00:59.03 | MikeJ[Laptop] | http://www.voip-info.org/wiki-Asterisk+cmd+WaitExten |
| 00:59.16 | MikeJ[Laptop] | but dump it using a goto into from-internal |
| 00:59.26 | MikeJ[Laptop] | instead of dial |
| 00:59.42 | MikeJ[Laptop] | that should do it |
| 01:00.45 | gaggaman | ok, let me try! |
| 01:03.42 | *** part/#asterisk tsp (~tyler@S01060080c825173c.vc.shawcable.net) |
| 01:07.20 | *** part/#asterisk NatRH (~Nat@dargo.trilug.org) |
| 01:08.06 | gaggaman | MikeJ: Hey, looks good! I get something like "now were going to dial 21". |
| 01:08.55 | |Vulture| | Anyone here ever install a door intercom device? it has a line in and out... when I plug FXS into line in, it make the device always off the hook |
| 01:09.06 | gaggaman | now i just have to bring it back into the amportal dialplan with the dialed number. |
| 01:12.58 | *** join/#asterisk easimon (~easimon@localhorst.kawo2.RWTH-Aachen.DE) |
| 01:13.43 | *** join/#asterisk newmedian (~crowlther@Quebec-HSE-ppp230300.qc.sympatico.ca) |
| 01:15.07 | gaggaman | to name it, the only thing I didn't just copy-and-paste (the goto) doesn't work :-) |
| 01:15.42 | gaggaman | exten => _X.,1,Goto(from-internal,s,1) |
| 01:16.45 | gaggaman | no? |
| 01:19.11 | *** join/#asterisk easimon_ (~easimon@localhorst.kawo2.RWTH-Aachen.DE) |
| 01:21.02 | *** join/#asterisk s9510278 (~bigdaddy2@203.210.152.185) |
| 01:24.11 | *** part/#asterisk s9510278 (~bigdaddy2@203.210.152.185) |
| 01:27.14 | newmedian | I've got someone with an Option 11 system in a non VOIP setup, with 20-30 Meridian phones (e.g. M2008). If I move him to an * setup, can(/should) the phones be re-used (via Channel Bank I assume) to *, or are they basically feature locked to the Nortel PBX? |
| 01:28.25 | Chuji | newmedian |
| 01:28.28 | Chuji | http://www.voip-info.org/tiki-index.php?page=Digital%20Telephone%20Adapt |
| 01:28.28 | Chuji | ers |
| 01:28.38 | Chuji | http://www.voip-info.org/tiki-index.php?page=Digital%20Telephone%20Adapters <--- that one |
| 01:31.32 | newmedian | Chuji: thanks; Any real-world comments about doing that? |
| 01:32.32 | gaggaman | MikeJ: ok, now it works! Thanks a lot! |
| 01:33.11 | Chuji | Hey Drum, you get moved? |
| 01:33.49 | drumkilla | yep! I'm at my desk right now |
| 01:34.31 | tzanger | wb drumkilla |
| 01:34.36 | drumkilla | thank ya |
| 01:35.12 | drumkilla | wasted my whole afternoon trying to get cable/internet access -- it took 4 hours |
| 01:35.18 | drumkilla | so I'm back at work catching up |
| 01:35.32 | Chuji | ~google site: lists.digium.com "Convertnig from Norstar" |
| 01:35.53 | Chuji | newmedian : Look that that first link |
| 01:36.33 | newmedian | Chuji: thank you kindly. |
| 01:38.18 | MikeJ[Laptop] | hey drumkilla |
| 01:40.18 | *** join/#asterisk techIII (~techii@69-161-115-199.lkwnny.adelphia.net) |
| 01:42.16 | *** join/#asterisk ManxPower (~eric@dsl-209-16-67-160.datasync.com) |
| 01:42.32 | ManxPower | Who here think the Intel EE Pro cards are good? |
| 01:44.12 | WilliamK | EE? |
| 01:44.21 | ManxPower | Whoever it is, I want to know what kind of sick mind control drugs they fed you. |
| 01:44.29 | ManxPower | WilliamK: EtherExpress |
| 01:44.31 | WilliamK | and yeah I like them |
| 01:44.55 | ManxPower | The one built into my motherboard refuses to work in 100Mbpz/full with my linksys switch |
| 01:45.45 | ManxPower | Intel motherboard, eepro network card (built in) chipset. |
| 01:45.52 | WilliamK | linksys...somehow doesn't surprise me, I've had 75% luck |
| 01:45.53 | ManxPower | %50 packet loss |
| 01:46.07 | ManxPower | WilliamK: Works fine with the 3com cards I normally use |
| 01:46.27 | WilliamK | yeah, ran into the same problem with linksys to motorola surfboard 5100 |
| 01:46.33 | Sedorox | I had a older linksys 5 port switch that went bad.. dropped everything like %75 of the time... after a while |
| 01:46.35 | MikeJ[Laptop] | ManxPower, if it is set to autodetect, force it to 100/full, if it is forced, switch it to autodetect |
| 01:46.58 | ManxPower | MikeJ[Laptop]: There is no option in the bios to do that. |
| 01:47.07 | gaggaman | Manx: But force *both* sides to 100/full. |
| 01:47.13 | ManxPower | MikeJ[Laptop]: When I force it using ethtool it still doesn't work. |
| 01:47.23 | ManxPower | gaggaman: You can't force Linksys modes. |
| 01:47.35 | ManxPower | But the lights look correct. |
| 01:47.48 | ManxPower | I disabled the builtin nic and put a 3com card into the system. |
| 01:47.51 | ManxPower | worked first time |
| 01:47.57 | ManxPower | </vent> |
| 01:48.20 | gaggaman | the switch is unmanaged? |
| 01:48.27 | ManxPower | gaggaman: Yes. |
| 01:48.35 | ManxPower | I'm not aware of any managed linksys switches |
| 01:48.54 | WilliamK | there are a few out there |
| 01:49.00 | ManxPower | I should have just done 5 runs back to my cisco switch in the computer closet 8-) |
| 01:51.18 | gaggaman | manx: btw, how do you know it is in 100/half ? |
| 01:51.25 | techIII | err, what exactly is a pbx? |
| 01:52.06 | ManxPower | gaggaman: using ethtool on linux and the lights on the Linksys |
| 01:53.43 | gaggaman | can't trust ethtool with Intel Ether Express. Had one 1GB with ethtool insisting it is in 100 Mbit. But it was definitely in 1GBit. |
| 01:53.54 | techIII | nvm |
| 01:54.44 | denon | ManxPower: linksys does make a managed switch |
| 01:54.48 | denon | 24pt as mem serves |
| 01:54.56 | denon | no way would I use it .. but .. |
| 01:54.58 | newmedian | There. Are. Four. Lights! |
| 01:55.10 | techIII | how would I address another asterisk instalation over the internet (as a regular user) through dns, or a central directory? |
| 01:55.34 | WilliamK | gaggaman, I even have that issue with the marvel's on intel boards w/1GB ports |
| 01:57.00 | gaggaman | <PROTECTED> |
| 02:02.38 | techIII | is there any system to obtain an address to use with asterisk (kind of like a directory service) |
| 02:02.42 | Chuji | Heh, I have Dell managed switches. I'd use about anything |
| 02:02.59 | Chuji | ~enum |
| 02:03.02 | Chuji | ~dundi |
| 02:03.21 | Chuji | Lame ass bot |
| 02:03.55 | Chuji | jbot: dundi is http://www.dundi.com |
| 02:03.56 | jbot | Chuji: okay |
| 02:04.48 | Chuji | jbot: enum is http://www.voip-info.org/wiki-Enum |
| 02:04.49 | jbot | okay, Chuji |
| 02:04.56 | Sedorox | linksys's higher end switches look really nice... then again.. look and functionality are two different things |
| 02:05.09 | *** part/#asterisk techIII (~techii@69-161-115-199.lkwnny.adelphia.net) |
| 02:06.09 | *** join/#asterisk NewSole (dave@i216-58-44-245.avalonworks.net) |
| 02:06.23 | *** join/#asterisk lilneon (~tj_r3@cuscon13125.tstt.net.tt) |
| 02:06.32 | lilneon | hey good night everyon |
| 02:06.33 | lilneon | e |
| 02:09.49 | lilneon | ne one here? |
| 02:10.43 | lilneon | hey guys i am looking for an itsp who supplies toll free nos directly to my asterisk box |
| 02:10.50 | lilneon | know ne ? |
| 02:11.04 | Chuji | iax.cc |
| 02:12.22 | *** join/#asterisk xkor (~kor@d209-89-221-44.abhsia.telus.net) |
| 02:13.04 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@pcp02795302pcs.roylok01.mi.comcast.net) |
| 02:15.03 | *** join/#asterisk Sedorox (brandon@sedorox.staff.smartserv) |
| 02:15.30 | lilneon | iax.cc?? |
| 02:15.31 | xkor | My TDM400P's FXS module seems to be a bit weird: when incoming calls come in, and I answer them through the FXO module, when talking I can hear my voice, and it is fairly distorted. After about 10 seconds of this, there is a subtle click on the line and everything starts working perfectly. Has anyone else experienced this? |
| 02:15.59 | *** join/#asterisk MikeJ_ (~ircatjerr@pcp02795302pcs.roylok01.mi.comcast.net) |
| 02:16.22 | lilneon | no i want incoming.. i knew there were a few places allowing you to get a toll free number.. |
| 02:16.34 | lilneon | which is redirected to your asterisk box.. |
| 02:16.53 | lilneon | so.. PSTN->tollfreeno.->asterisk |
| 02:16.57 | lilneon | like a calling card |
| 02:17.01 | lilneon | kinda setup? |
| 02:18.38 | Newbie___ | MikeJ_: hi |
| 02:18.49 | Newbie___ | any one knows a decent billing software ? |
| 02:20.01 | newmedian | (Chuji:) Any opinions as to what users of a two-line hands-free Meridian phone would feel most comfortable with in a VOIP phone? (Make/model) |
| 02:22.33 | Legend | newmedian: a sayson/aastra,if they would get off their asses and release a stable firmware |
| 02:24.45 | newmedian | Legend: thanks; And if they didn't? Second choice would be? (Would have to replace 20-30 phones, so cost is a factor as well, but audio quality/reliability can't be bad) |
| 02:30.10 | *** join/#asterisk harryvv (~leonardo@S010600a0c93f6f7e.vs.shawcable.net) |
| 02:37.49 | Silik0n | Newbie___ for VoIP? |
| 02:40.36 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
| 02:40.36 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 02:41.59 | Newbie___ | Silik0n: yes |
| 02:51.31 | newmedian | Friday night and the traffic is low. Everyone out watching Unleashed or something? |
| 02:53.15 | Silik0n | Newbie___ you find one lemme know heh |
| 02:54.10 | Newbie___ | Silik0n: sure |
| 03:02.17 | alt | hey guys. anyone have any thoughts on Asterisk's H.323 support? stability? scalability? |
| 03:02.49 | Silik0n | alt: more like lackthereof |
| 03:02.53 | alt | I see :) |
| 03:03.54 | alt | Silik0n: so your opinion is that I should probably not bother with it? |
| 03:03.56 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@pcp02795302pcs.roylok01.mi.comcast.net) |
| 03:04.33 | Silik0n | well that all depends on waht you want to do with it |
| 03:05.19 | alt | mostly I was thinking of using it to test out our H.323 routes. |
| 03:06.52 | Silik0n | there's a possibility of a new stack coming but * is mainly used for SIP and IAX |
| 03:07.25 | alt | we use it only on SIP. I didn't even have the IAX stack loaded until I decided to play with it :P |
| 03:08.12 | Silik0n | heh |
| 03:08.25 | Silik0n | i use it for both... mainly SIP but use IAX trunking |
| 03:09.26 | *** join/#asterisk |Vulture| (~V@199.227.253.212) |
| 03:09.36 | |Vulture| | wow the things we do when were bored... |
| 03:09.54 | newmedian | Yes, but the photos were entertaining. |
| 03:09.55 | alt | we haven't bothered to look at IAX. None of our peers use it. |
| 03:10.06 | alt | |Vulture|: did you ever get the $1000 for that? |
| 03:10.29 | |Vulture| | alt: for what? |
| 03:11.02 | alt | 20:09 < |Vulture|> wow the things we do when were bored... |
| 03:11.02 | alt | 20:09 < newmedian> Yes, but the photos were entertaining. |
| 03:11.06 | alt | 20:10 < alt> |Vulture|: did you ever get the $1000 for that? |
| 03:11.11 | alt | (the photos) |
| 03:11.12 | alt | ;-) |
| 03:11.19 | |Vulture| | :P |
| 03:11.34 | |Vulture| | no my friends and I made an IVR for a joke sex line |
| 03:11.43 | |Vulture| | with like funny names for each of my roomies |
| 03:12.02 | alt | that's a good one |
| 03:12.20 | |Vulture| | 9042240771 |
| 03:12.32 | |Vulture| | hahaha yea its running off an unused DID from my PRI LOL |
| 03:14.36 | alt | hee hee hee |
| 03:14.50 | alt | I'll have to get Lana at the office to do prompts for us. |
| 03:15.24 | alt | it'd probably go over well as one of our sales people is a nudist :) |
| 03:15.46 | |Vulture| | hahaha |
| 03:15.51 | |Vulture| | it was a fun thing to do |
| 03:16.07 | |Vulture| | make names for everyone like Saucy Shannon, and Hot Sauce Hugh |
| 03:16.21 | Silik0n | just use real pr0n star names |
| 03:16.38 | Silik0n | heyya file |
| 03:16.46 | file | hi hi |
| 03:16.47 | file | waddup |
| 03:16.58 | alt | actually, Lana may not have the best voice for doing those prompts... I'll get "Boom Boom" to do it :) |
| 03:17.05 | file | drumkilla: clock radios are for silly people |
| 03:17.08 | Silik0n | looking around for rating software and trying to decide if I should write my own |
| 03:17.10 | |Vulture| | Boom Boom... wow |
| 03:17.27 | alt | |Vulture|: that's her unofficial nickname... she doesn't know she has it :P |
| 03:17.38 | |Vulture| | alt: we had a girl named "Lovely Butts" and a guy named "Dick Super" oh and "Mike Googe" |
| 03:17.50 | Silik0n | alt: she have "good genetics" "upstairs" |
| 03:17.58 | alt | Silik0n: hence the name ;-) |
| 03:18.01 | Silik0n | hah |
| 03:18.21 | newmedian | "Is my voicemail Hot or Not?" |
| 03:18.24 | Silik0n | yeah I worked with a girl at a fortune 500 who's nickname was simply Juggs |
| 03:18.37 | |Vulture| | haha |
| 03:18.49 | |Vulture| | Silik0n: wow... |
| 03:19.31 | Silik0n | she was a netowrk engineer on my ops team and she would leave me VMs "hey this is juggs call me back about ....: |
| 03:19.56 | alt | O_o |
| 03:20.19 | |Vulture| | who doesn't |
| 03:20.25 | alt | feminazis |
| 03:20.48 | Silik0n | she was a trip to work with... while working late in the datacenter one nite she spilled it about how she liked to get "f'd in the a" |
| 03:20.48 | alt | but then, they don't like anything except watching men's balls get crushed using a walnut cracker. |
| 03:20.52 | *** join/#asterisk TheEmperor (TheEmperor@218.111.51.8) |
| 03:21.06 | |Vulture| | no one wants them anyways... guys want to sleep with girls... not guys with female anatomy |
| 03:21.51 | Silik0n | back to packing for me... |
| 03:21.51 | TheEmperor | hello |
| 03:21.52 | |Vulture| | I like to stay away from the ass thank you very much |
| 03:21.56 | alt | only problem is she'd talk and never follow through :\ |
| 03:21.58 | Silik0n | (moving sux ass) |
| 03:22.18 | |Vulture| | alt: that leads to sexual frustration |
| 03:22.22 | |Vulture| | sup TheEmperor |
| 03:22.24 | TheEmperor | does anyone where I can find information about setting up iptables in linux for QOS? |
| 03:22.25 | TheEmperor | :) |
| 03:22.34 | alt | |Vulture|: yes... yes it does |
| 03:23.12 | TheEmperor | gotta make sure that voice calls (or sip ports/iax ports) get priority over other things.. |
| 03:23.14 | *** part/#asterisk lilneon (~tj_r3@cuscon13125.tstt.net.tt) |
| 03:23.17 | newmedian | Speaking of which, the trailer for the new Zorro film looks not bad. |
| 03:23.23 | newl | TheEmperor: google for tc and packet marking |
| 03:23.47 | TheEmperor | ok newl, thanks |
| 03:23.52 | |Vulture| | newmedian: speaking of sex... or speaking of QoS? |
| 03:23.53 | |Vulture| | :P |
| 03:24.06 | Sedorox | both? |
| 03:24.07 | Sedorox | :p |
| 03:24.36 | |Vulture| | okay time to go home... enough work today |
| 03:24.38 | newmedian | Well, having one's fortune in a rears. |
| 03:28.10 | TheEmperor | does anyone have any examples of how to configure iptables for iax and sip ports? |
| 03:30.33 | drumkilla | http://www.voip-info.org/wiki-Asterisk+firewall+rules |
| 03:32.59 | TheEmperor | ah thanks |
| 03:35.07 | drumkilla | np |
| 03:38.53 | TheEmperor | how often do people reboot their asterisk server? |
| 03:39.16 | wisdom | I haven't rebooted the one in my office for well over 500 days |
| 03:39.21 | wisdom | needed to patch a few things ;-) |
| 03:39.37 | sudhir492 | Teh Empreror: It depeneds. One of my asterisk servers lasted for over 8 months without rebooting |
| 03:39.41 | TheEmperor | i find that after awhile cpu gets used up by asterisk |
| 03:39.48 | TheEmperor | and it helps by rebooting |
| 03:40.01 | TheEmperor | any thoughts on this? |
| 03:40.19 | sudhir492 | But other just crash on their own |
| 03:40.22 | wisdom | could be anything ... especially if the box is being used for other things than * |
| 03:40.31 | newmedian | ~astbot |
| 03:40.34 | TheEmperor | these * boxes are just for * |
| 03:40.44 | sudhir492 | cpu gets used up most of the times by mpg server |
| 03:40.54 | sudhir492 | I mean mpg123 |
| 03:40.59 | TheEmperor | is it bad to reboot once a day? |
| 03:41.23 | wisdom | Better to reboot in a controlled fashion than crash in a not-so-controlled fashion ;-) |
| 03:41.35 | TheEmperor | sounds right |
| 03:41.36 | *** part/#asterisk Nuxi (~nuxi@cust-sdsl-204-250-82-202.bzn-co-i1000-01.bridgeband.net) |
| 03:41.38 | sudhir492 | A system that has to be rebooted once a day is definitely not trustworthy and I would not use that in a business env |
| 03:41.54 | TheEmperor | what is the alternative to using mpg123? for music on hold? |
| 03:42.08 | wisdom | actually use an external music source ? |
| 03:42.16 | sudhir492 | read the wiki. you can play wav file in raw |
| 03:42.30 | wisdom | like an ipod with a bunch of mpegs being introduced by a sound card ;-0 |
| 03:42.38 | wisdom | ;-) |
| 03:42.38 | TheEmperor | playing wav files without mpg123? |
| 03:42.53 | sudhir492 | yes, you can just cat them |
| 03:43.11 | sudhir492 | mpg123 btw plays .mp3 files |
| 03:43.11 | TheEmperor | cat the mpgs? |
| 03:43.17 | TheEmperor | oh |
| 03:43.19 | sudhir492 | cat the .wav files |
| 03:43.29 | TheEmperor | so convert the mpg files back to .wav files? |
| 03:43.34 | sudhir492 | yep |
| 03:43.46 | TheEmperor | then i wouldn need to run mpg123? |
| 03:43.56 | sudhir492 | or use the original .wav file which you converted to .mp3 |
| 03:44.00 | sudhir492 | :-) |
| 03:44.46 | TheEmperor | then i would need to reconfigure musiconhold.conf right? |
| 03:46.02 | newmedian | I just tripped over "astbot" http://www.mixdown.ca/~andrew/astbot/ |
| 03:48.41 | sudhir492 | what is the largest number of extensions on people's asterisk here |
| 03:50.29 | *** join/#asterisk iq|laptop (~iq@63-230-44-112.omah.qwest.net) |
| 03:53.35 | TheEmperor | drumkilla:do you have that link in the wiki which shows how to prioritize bandwidth for sip and iax ports? |
| 03:53.46 | drumkilla | google, i guess |
| 03:54.05 | TheEmperor | k |
| 03:54.14 | sudhir492 | TheEmperor: The wiki has good step by step instruction on how to use .wav file and cat to for music on hold |
| 03:55.11 | TheEmperor | sudhir492:ok, i'll go check it out thanks |
| 03:55.41 | *** part/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
| 03:55.46 | Silik0n | . |
| 03:56.35 | sudhir492 | using this approach also reduces cpu load caused by mpg123 |
| 03:57.12 | TheEmperor | sudhir492:you wouldnt happen 2 have the link do you? :) |
| 04:02.37 | sudhir492 | TheEmperor: Try http://www.voip-info.org/tiki-index.php?page=Asterisk%20mpg123%20faking%20it |
| 04:09.53 | newl | TheEmperor: This is what you're looking for. Modify to suit for the ports or protocols you're interested in. Be sure to read up on iptables marking as well. http://pastebin.ca/11719 |
| 04:12.03 | TheEmperor | hey thanks guys :) |
| 04:12.34 | tzafrir_laptop | how big is that wav file you use, BTW? |
| 04:13.28 | TheEmperor | right now i am using mpg123, but plan to change it to .wav, not sure how big it will be.. |
| 04:20.28 | *** join/#asterisk meppl (mephisto@p54AAE973.dip.t-dialin.net) |
| 04:21.06 | *** join/#asterisk gambolputty (~gambolput@cblmdm69-45-216-83.buckeye-express.com) |
| 04:24.15 | *** join/#asterisk jdg (~jdg@CA03F39B.adsl.mana.pf) |
| 04:24.34 | *** join/#asterisk MikeJ_ (~ircatjerr@pcp02795302pcs.roylok01.mi.comcast.net) |
| 04:24.58 | *** join/#asterisk ManxPower (~eric@dsl-209-16-67-160.datasync.com) |
| 04:25.46 | *** join/#asterisk firestrm (firestrm@S010600047577bccd.gv.shawcable.net) |
| 04:27.28 | firestrm | im finally free of the iconnect stupidity.. |
| 04:28.45 | *** join/#asterisk Corydon76-home (mauve@pcp08665860pcs.500ash01.tn.comcast.net) |
| 04:29.43 | TheEmperor | newl:which parts do I modify? |
| 04:30.12 | newl | take a look at the section for ssh |
| 04:32.31 | TheEmperor | ah |
| 04:32.48 | TheEmperor | line 42? |
| 04:32.48 | newl | Something like this can help too: iptables -A PREROUTING -t mangle -i $EXTIF -p tcp --sport 22 -j TOS --set-tos Minimize-Delay |
| 04:33.09 | newl | yes, 42 |
| 04:33.29 | TheEmperor | just change the port 22 to 5060 and 4569? |
| 04:33.33 | TheEmperor | would that be right? |
| 04:34.35 | *** join/#asterisk zm4ne4 (~nomoregoo@c-67-176-11-72.hsd1.co.comcast.net) |
| 04:34.55 | *** join/#asterisk newmember (~newmember@S010600a0c93dce87.cg.shawcable.net) |
| 04:35.15 | newl | there is no port numbers listed there. Best to read up on tc. There's some great info there. |
| 04:35.21 | TheEmperor | ok |
| 04:35.46 | *** part/#asterisk jdg (~jdg@CA03F39B.adsl.mana.pf) |
| 04:36.22 | newl | you won't have to understand all of it but getting the basics will give you the understanding you need to implement things correctly. |
| 04:42.11 | zm4ne4 | anybody here seen a problem where asterisk will disconnect a call after 30 minutes? |
| 04:45.00 | alt | blah... I can't get pwlib to compile on FC3 :( |
| 04:45.06 | PTG123 | thats because you have the unregistered version of asterisk, you n eed to register it :) |
| 04:45.12 | alt | ha ha ha |
| 04:52.24 | *** join/#asterisk simon_ca (sedgett@S01060004e23873e1.vc.shawcable.net) |
| 04:53.50 | *** join/#asterisk Sato1 (~rauleli@sato1.wizardteam.com) |
| 04:54.18 | tzafrir_laptop | newl, not that I'm able to help you with that platform, but more details will bring you better help |
| 04:54.34 | tzafrir_laptop | pwlib version? errors? |
| 04:55.29 | tzafrir_laptop | FWIW, I can get it to build on my registered debian version |
| 04:55.42 | *** join/#asterisk iamnotbob (~nolan@216.199.159.79) |
| 04:56.10 | tzafrir_laptop | And it depends on about half the world, including aalib |
| 04:56.58 | simon_ca | trying to change blidxfer key to ## so I can enter # on IVR calls (inbound on x100p) - added blindxfer => *4 to features.conf - didn't work. Any thoughts? |
| 04:57.22 | zm4ne4 | log file shows Bridge stops because we're zombie or need a soft hangup and then the call is disconnected |
| 04:57.58 | zm4ne4 | It is like clockwork.. the calls only last for 30 minutes |
| 04:58.37 | *** join/#asterisk m0f0x (m0f0x@m0f0x.user) |
| 05:03.31 | `Sauron | Moof! |
| 05:03.48 | *** join/#asterisk loud (~ariel@gw.honeypot.no) |
| 05:10.16 | *** join/#asterisk gordo (~fungo@201009088159.user.veloxzone.com.br) |
| 05:10.38 | *** part/#asterisk gordo (~fungo@201009088159.user.veloxzone.com.br) |
| 05:12.23 | Newbie___ | where do i go to purchase G729 ? |
| 05:12.30 | *** join/#asterisk boch (~boch@201.254.110.249) |
| 05:12.34 | JerJer | Digium |
| 05:12.35 | Sato1 | at ditium |
| 05:13.21 | Newbie___ | does it come with how to manual ? |
| 05:13.41 | loud | its a three step process. |
| 05:15.14 | JerJer | and requires a stool sample |
| 05:15.52 | *** join/#asterisk cjk (~cjk@80.92.75.120) |
| 05:16.14 | Newbie___ | stupid me, still cant find in digium.com |
| 05:17.34 | iamnotbob | Does anyone know how to make MPG123 w/ asterisk & Sip? |
| 05:17.58 | JerJer | type make mpg321 in the asterisk source dir |
| 05:18.27 | JerJer | oops |
| 05:18.30 | JerJer | make mpg123 |
| 05:18.53 | iamnotbob | that's what i thought, and still can't get it to work, I get errors. |
| 05:19.22 | JerJer | what errors? |
| 05:21.02 | iamnotbob | I will get something like warning 6126: res_musiconhold.c:370 moh1_exec: unable to start the moh,,,, for 500 seconds on channel sip/1003-d986 |
| 05:22.23 | JerJer | fun |
| 05:22.58 | iamnotbob | a pain it has been... |
| 05:23.00 | *** join/#asterisk Johnsie (~john@acs-24-154-32-12.zoominternet.net) |
| 05:24.35 | *** join/#asterisk Pete_Largo (~Pete_Larg@adsl-65-71-225-121.dsl.rcsntx.swbell.net) |
| 05:24.56 | iamnotbob | I thought that I replaced the mpg321 with mpg123 and did all the make files right and the extensions.conf & musiconhold.conf correctly. |
| 05:25.49 | *** join/#asterisk jayk_ (jayk@vapid.reprehensible.net) |
| 05:29.26 | nords | i am running an agi script to put some people into a conference room. Do you think the agi script should exit after the person goes into the meetme room? |
| 05:30.30 | nords | BTW for those that weren't here earlier this is a carry over from not being able to use the "redirect" command |
| 05:30.52 | nords | i think i have narrowed it down to the agi script not functioning correctly |
| 05:32.35 | MikeJ[Laptop] | JerJer, what happend to the nufone cvs mirror? |
| 05:33.05 | PTG123 | man i need someone in england to record my favorite shows now that the stupid sites have been shut down :( |
| 05:33.27 | Qwell | PTG123: there is always piratebay :p |
| 05:33.42 | PTG123 | whats that? |
| 05:33.45 | Qwell | .org |
| 05:34.10 | PTG123 | how do i make it in english? |
| 05:34.17 | Qwell | heh, you don't |
| 05:34.32 | Qwell | find the listbox |
| 05:34.44 | PTG123 | heh what listbox? |
| 05:34.51 | Qwell | category |
| 05:35.32 | Qwell | alot of the words are similar |
| 05:35.51 | JerJer | MikeJ[Laptop]: nobody bothered to tell me to support cvsup and bkw was being a bitch - so i pulled it |
| 05:36.10 | PTG123 | hmm |
| 05:36.15 | PTG123 | hustle is an old episode on there |
| 05:36.16 | PTG123 | bastards |
| 05:36.58 | MikeJ[Laptop] | support cvsup |
| 05:36.59 | MikeJ[Laptop] | ? |
| 05:37.29 | PTG123 | qwell:L when i pick a show |
| 05:37.32 | PTG123 | it shows no seeds and no peers |
| 05:37.36 | PTG123 | in my client |
| 05:37.39 | PTG123 | is it just slow or something? |
| 05:37.50 | Qwell | yes |
| 05:37.59 | Qwell | it times out alot, but let it go, it should retry |
| 05:38.45 | PTG123 | doesn't seem as good as btefnet |
| 05:38.48 | PTG123 | but i guess it will have to do :) |
| 05:39.02 | PTG123 | anyone here in england, i'll send you money to record hustle :) |
| 05:50.30 | *** join/#asterisk outtolunc (~me@adsl-69-110-50-162.dsl.pltn13.pacbell.net) |
| 05:54.39 | *** join/#asterisk docelmo (~me@116-39.202-68.tampabay.res.rr.com) |
| 05:56.25 | *** join/#asterisk wreck-gar (~generic@67.62.68.43) |
| 05:56.25 | *** join/#asterisk t0pCop (t0pCop@221.128.88.110) |
| 05:56.46 | docelmo | any idea what causes this: http://pastebin.ca/11720 |
| 05:57.32 | docelmo | Im trying to compile oh323 on the current head which doesnt work cause a file called channels_pvt.h is missing in asterisk/include/asterisk/ |
| 05:57.47 | docelmo | when I use the stable branch I get that.. I cant win for loosing |
| 05:59.24 | wreck-gar | does one have to create the /dev/dsp for the XP100 to show up as a channel in asterisk? |
| 05:59.41 | docelmo | no.. |
| 05:59.55 | docelmo | dsp is for the soundcard. Your XP100 will show up as a ZAP device |
| 05:59.59 | Qwell | x100p |
| 06:00.07 | wreck-gar | i do a show zap channels in the CLI |
| 06:00.17 | wreck-gar | and all it shows is pesudo channel... |
| 06:00.25 | Qwell | config it |
| 06:00.28 | wreck-gar | yeah |
| 06:00.36 | Qwell | see what ztcfg says |
| 06:00.36 | PTG123 | i hate the mpaa |
| 06:00.41 | Qwell | PTG123: don't we all |
| 06:00.52 | wreck-gar | fxsks=1 #my POS clone card |
| 06:00.52 | wreck-gar | loadzone = us |
| 06:01.16 | wreck-gar | ztcfg Zaptel Configuration |
| 06:01.16 | wreck-gar | ====================== |
| 06:01.16 | wreck-gar | Channel map: |
| 06:01.16 | wreck-gar | Channel 01: FXS Kewlstart (Default) (Slaves: 01) |
| 06:01.16 | wreck-gar | 1 channels configured. |
| 06:01.32 | Qwell | ~pastebin |
| 06:01.33 | jbot | [pastebin] a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 06:01.37 | wreck-gar | sorry. |
| 06:02.45 | docelmo | PTG123, I was almost sued 2x by the MPAA |
| 06:02.50 | docelmo | I host www.uk-torrents.com |
| 06:02.52 | PTG123 | for what? |
| 06:02.55 | docelmo | or used to |
| 06:02.59 | PTG123 | docelm0, your in the uk? :) |
| 06:03.02 | docelmo | Nope |
| 06:03.09 | newmedian | I almost won the lottery 2x. Except for all those missing numbers. |
| 06:03.10 | PTG123 | docelm0, damn, i want hustle! :) |
| 06:03.32 | docelmo | I dont host them anymore MPAA shut em down |
| 06:03.39 | docelmo | I lost 500 a month in revenue.. :( |
| 06:04.04 | wreck-gar | at least the FCC struck down the broadcast flag |
| 06:04.04 | docelmo | Any developers in the room? |
| 06:05.04 | docelmo | ok you dont count.. |
| 06:05.29 | *** join/#asterisk ferbo (~chris@pool-71-105-69-156.lsanca.dsl-w.verizon.net) |
| 06:05.33 | PTG123 | docelm0, what do you need a developer for? |
| 06:05.35 | docelmo | Im trying to figure out what they are calling the CallerID function in asterisk now. it used to be CID now its something else |
| 06:05.36 | outtolunc | maybe if you just asked the question <G> |
| 06:05.43 | docelmo | little help with oh323 |
| 06:05.50 | docelmo | PTG can you check out oh323_call |
| 06:05.53 | docelmo | err hold |
| 06:06.02 | docelmo | http://pastebin.ca/11720 |
| 06:06.03 | PTG123 | heh, i don't develop in head, my stuff is in stable :) |
| 06:06.21 | docelmo | I dont care either way. I need to resove this issue |
| 06:06.41 | docelmo | and channels_pvt.h isnt available in head. Only stable |
| 06:06.50 | outtolunc | why not look in /usr/src/asterisk/include/asterisk/callerid.h |
| 06:07.15 | Qwell | Don't try to fix a server at 5pm on a Friday |
| 06:07.15 | docelmo | maybe you do count.. :p |
| 06:07.16 | outtolunc | note: struct ast_callerid cid; |
| 06:07.23 | Qwell | at least, not a Windows server |
| 06:07.25 | outtolunc | in ast_channel now |
| 06:07.29 | PTG123 | qwell: why it worked for me tonight :) |
| 06:07.39 | PTG123 | i fixed the server, and 3 asterisk bugs |
| 06:07.39 | Qwell | PTG123: I tried moving a win2k3 install to new hardware... |
| 06:07.45 | PTG123 | ah |
| 06:07.47 | Qwell | it didn't work too well. ;] |
| 06:07.51 | PTG123 | well rephrase to "fix a windows server" |
| 06:08.06 | Qwell | I did |
| 06:08.25 | docelmo | hmm.. Im looking at the oh323.c code and find c->cid.xxx so cid is there.. sigh |
| 06:08.28 | PTG123 | qwell: isohunt.com |
| 06:08.33 | Qwell | ? |
| 06:08.38 | PTG123 | good site |
| 06:08.41 | PTG123 | why h323 docel? |
| 06:09.10 | JerJer | and then why oh323? |
| 06:09.20 | outtolunc | the issue is that chan_oh323 probably hasn't been updated |
| 06:09.40 | JerJer | probably? most likely |
| 06:09.55 | outtolunc | i was being nice <G> |
| 06:09.57 | PTG123 | why would anyone need to use h323, is sip firmware not available for your device? |
| 06:10.00 | wreck-gar | http://pastebin.ca/11721 |
| 06:10.12 | wreck-gar | i know this might sound stupid |
| 06:10.32 | docelmo | Im working on something for a friend of mine in equador and keeps begging for h323 so I am trying to find him a solution |
| 06:10.41 | wreck-gar | but does the phone line have to be plugged in to the x100p before asterisk will regocnize it as a channel? |
| 06:11.04 | PTG123 | but you still didn't answer the question :) |
| 06:11.05 | JerJer | wreck-gar: no |
| 06:11.31 | wreck-gar | then i am stumped.... |
| 06:11.35 | docelmo | I know chan_oh323 is outdated.. I just dont know how to update it. I can read C cant develop in it.. |
| 06:11.51 | wreck-gar | my asterisk won't show my x100p channel |
| 06:11.52 | docelmo | And my gear is all sip.. He says all people down there use h323 of some sort.. |
| 06:11.57 | JerJer | then run chan_h323 using cvs -head code |
| 06:12.10 | docelmo | will not compile.. |
| 06:12.18 | JerJer | then u are not running cvs -head |
| 06:12.56 | wreck-gar | i put my /etc/zapata.conf and /etc/asterisk/zaptel.conf up on paste bin. -> http://pastebin.ca/11721 |
| 06:13.22 | wreck-gar | and when i dmesg is said it found the card. |
| 06:13.23 | JerJer | um |
| 06:13.33 | JerJer | there is no channel => 1 in zapata.conf |
| 06:13.52 | wreck-gar | well apparently i've been to the wrong sites then... |
| 06:13.54 | *** join/#asterisk tengulre (~tengulre@61.185.238.166) |
| 06:14.06 | JerJer | see zapata.conf.sample |
| 06:14.19 | wreck-gar | k |
| 06:14.21 | wreck-gar | thanx |
| 06:14.26 | docelmo | uhh nope.. running stable |
| 06:14.50 | JerJer | docelmo: i just did a make clean install on my -head development tree and it compiled perfectly fine - including chan_h323 |
| 06:14.53 | PTG123 | jerJer likes the bleeding edge :) |
| 06:15.50 | JerJer | no i like code that works |
| 06:15.58 | JerJer | -stable does not work |
| 06:15.59 | tengulre | Hi,all |
| 06:16.17 | ferbo | might anyone know if chan_bluetooth might be receiving incoming call support? |
| 06:16.32 | JerJer | as soon as you code it |
| 06:16.47 | ferbo | I see, ok :) |
| 06:17.09 | Newbie___ | hi, is there any reference on how to install g729 on a redhat ? |
| 06:17.27 | outtolunc | CVS-HEAD-05/10/05-10:16:33 |
| 06:18.20 | PTG123 | hah works better for me :) |
| 06:18.34 | PTG123 | Newbie___, its not very hard |
| 06:18.53 | docelmo | newb download g729 drop in modules and register license |
| 06:19.16 | Newbie___ | i did that |
| 06:19.28 | docelmo | KILL asterisk and start it |
| 06:19.33 | wasim | Newbie___: but make sure you don't change any of your NICs afterwards |
| 06:22.14 | *** join/#asterisk gres (~serg@81.222.48.242) |
| 06:22.21 | JerJer | or if you do clone the mac address :P |
| 06:22.38 | Newbie___ | no clone, |
| 06:22.58 | tzafrir_laptop | as in 'ifconfig hwadd 11:22:33:44:55:66 ? |
| 06:23.58 | tzafrir_laptop | ifconfig hw ether 11:22:33:44:55:66 , actually |
| 06:25.13 | PTG123 | heh shhhh, don't ell tehm if you find someone with a license and clone their mac they can pirate the license |
| 06:25.19 | Newbie___ | i just bought for 1 channel but plan 29 more. i can upgrade if i need more . right? |
| 06:25.37 | PTG123 | yes newbie |
| 06:25.40 | Newbie___ | PTG123: i found a site that does free download in intel.com |
| 06:26.31 | wreck-gar | so my zapata.conf is fine. |
| 06:27.01 | wreck-gar | i think you mistook my ztcfg output as part of my zapata.conf.... |
| 06:27.42 | *** join/#asterisk joaovianna (naturalvoi@node-40247a6a.ewr.onnet.us.uu.net) |
| 06:28.02 | JerJer | zapata.conf has to have a channel keyword |
| 06:28.08 | JerJer | i do not see one in your pastebin |
| 06:28.09 | PTG123 | I can't believe Star Trek is dead |
| 06:28.37 | PTG123 | jerJer: your putting up with a suprisingly large amount of stupid questions lately |
| 06:29.17 | wreck-gar | i don't see it in the zapata.conf.sample... |
| 06:29.21 | docelmo | cvs checkout asterisk |
| 06:29.25 | docelmo | this will get me head right? |
| 06:29.29 | wreck-gar | the channel keyword at least |
| 06:29.42 | wreck-gar | its in the zaptel.conf |
| 06:30.00 | PTG123 | i don't know if that will get it for you, but a ferrari usually does the trick. |
| 06:30.06 | JerJer | ;callerid="Green Phone"<(256) 428-6121> |
| 06:30.06 | JerJer | ;channel => 1 |
| 06:30.06 | JerJer | ;callerid="Black Phone"<(256) 428-6122> |
| 06:30.06 | JerJer | ;channel => 2 |
| 06:30.11 | JerJer | look again |
| 06:30.16 | joaovianna | I'm having problems to run Perl AGI in my *. Anyone can help me ? |
| 06:30.24 | wreck-gar | so thats for incoming lines as well then? |
| 06:30.32 | JerJer | joaovianna: only if you ask a specific question |
| 06:30.46 | JerJer | wreck-gar: its for any channel |
| 06:30.54 | JerJer | wreck-gar: do yo have a license to operate a computer? |
| 06:30.55 | wreck-gar | ok |
| 06:31.37 | wreck-gar | no, last i checked you didn't need one |
| 06:31.46 | Qwell | unfortunately |
| 06:33.09 | docelmo | if checkout just asterisk will that give me head or stable? |
| 06:33.15 | wreck-gar | i thougth the channel and callerid stuff was just that, if an incoming call with that number matches |
| 06:33.17 | joaovianna | Thanks JerJer... When I run from console perl agi-test.agi I have error: Can't locate Asterisk/AGI.pm in @INC . I re-installed perl5.6.8 and asterisk-perl 3 times and I'm still having this error. From * my agi returns 0 but nothing happens... Any clue ? |
| 06:33.25 | wreck-gar | it sends it to that channel internally. |
| 06:34.04 | Qwell | wreck-gar: read a howto... |
| 06:34.05 | *** join/#asterisk [Lamer] (Lamer@221.128.88.110) |
| 06:34.30 | JerJer | joaovianna: you have to install Asterisk::AGI |
| 06:34.35 | outtolunc | he needs to read the whole thing, as in www.digium.com/handbook-draft.pdf |
| 06:34.54 | wreck-gar | Qwell: you want to see all the how-to's i've read before i decided to come here? |
| 06:35.02 | wreck-gar | i googled the shit out of this |
| 06:35.11 | wreck-gar | this was my last resort so lay off |
| 06:35.18 | Qwell | Did you read "learning English in 3 easy steps"? |
| 06:35.23 | joaovianna | JerJer: Asterisk::AGI comes with asterisk-perl ??? |
| 06:35.28 | JerJer | how about simply reading zapata.conf.sample ? |
| 06:35.40 | JerJer | howtos are overrated |
| 06:35.57 | wreck-gar | i did Jerjer |
| 06:36.13 | wreck-gar | i guess i just don't understand what goes with what. |
| 06:36.47 | outtolunc | did you read the handbook-draft.pdf? |
| 06:37.02 | wreck-gar | yes, but i am going over it again |
| 06:37.18 | outtolunc | k |
| 06:37.23 | [Lamer] | can anyone tell me if there is an additional setting to allow video mode among SIP phones? |
| 06:38.33 | outtolunc | ;videosupport=yes ; Turn on support for SIP video |
| 06:39.01 | [Lamer] | outtolunc: for each SIP extension in sip.conf? |
| 06:39.03 | newmedian | Channel is angsty tonight |
| 06:39.08 | outtolunc | obviously, your sip client will have to have builtin support as well |
| 06:39.24 | JerJer | then a codec |
| 06:39.30 | JerJer | allow=h623 |
| 06:39.41 | outtolunc | if you notice, it's in the [general] section |
| 06:39.55 | [Lamer] | outtolunc: I'm using one of the Exten softphones, eyeBeam |
| 06:40.09 | outtolunc | cool |
| 06:40.19 | outtolunc | this that and what jerjer said |
| 06:40.24 | JerJer | a mac user |
| 06:41.06 | [Lamer] | jerjer: should allow=all work ? |
| 06:41.23 | JerJer | evil |
| 06:42.10 | [Lamer] | JerJer: Huh, that dangerous? |
| 06:46.19 | wreck-gar | ok so now i know what to add, is there a specific place to put the Zap/1-1 line |
| 06:46.54 | wreck-gar | or specific section, there is nothing in the handbook on where it should go, or does it not matter |
| 06:49.12 | newmedian | wreck-gar: If you've got a spare machine, you may want to start this way: http://asteriskathome.sourceforge.net/ |
| 06:49.44 | outtolunc | wreck, after config'n zaptel and zapata it 'creates' a Zap device, that channel=1 makes it 'Zap/1' |
| 06:50.16 | wreck-gar | i did the zaptel config already. |
| 06:50.25 | wreck-gar | i think its all set now |
| 06:50.26 | wreck-gar | thanx |
| 06:54.40 | *** join/#asterisk kryme (~PK@63.151.31.248) |
| 06:55.09 | wreck-gar | newmedian: i have it up and running, i just can't get it to register my x100p as a channel |
| 06:56.49 | firestrm | :) |
| 06:57.12 | wreck-gar | hey firestrm when something decent comes out in the way of documentation i'll gladly go buy it and read it |
| 06:57.22 | wreck-gar | but until then i'm stuck asking questions |
| 06:58.12 | firestrm | wreck-gar, you'll get no argument from me.. I wish that some ppl here would realize that we all start off asking stupid questions. |
| 06:58.16 | wreck-gar | i am |
| 06:58.44 | wreck-gar | and i think its telling me to add the line Zap/1 to my zapata.conf |
| 06:58.46 | wreck-gar | which i did |
| 06:58.54 | outtolunc | NO |
| 06:59.11 | wreck-gar | ok |
| 06:59.37 | wreck-gar | then what is it telling me to do? |
| 06:59.42 | outtolunc | remember the 'channel=1' that jerjer and i mentioned |
| 06:59.49 | wreck-gar | ya |
| 06:59.54 | outtolunc | in zapata.conf |
| 07:00.00 | tzafrir_laptop | wreck-gar, if this is how you feel, then have you started correcting and adding some pages to voip-info.org ? |
| 07:00.27 | wreck-gar | tzafrir_laptop: i will be glad to |
| 07:00.37 | wreck-gar | i've already started making side notes |
| 07:00.55 | tzafrir_laptop | wreck-gar, zapata.conf basically defines Zap channels. The dialplan (extensions.conf) tells the PBX how to use them |
| 07:01.07 | firestrm | i wonder if it wouldnt be better for the advancement of asterisk to split the channel into #asterisknewbe and #asteriskL337antisocials |
| 07:01.34 | outtolunc | well i said that if zaptel and zapata are setup that channel=1 (since Zap device is created) the channel=1 makes it whole as you will now have a Zap with 1 channel (ie.. Zap/1) you DO NOT manually enter that |
| 07:02.13 | wreck-gar | ok |
| 07:02.23 | tzafrir_laptop | firestrm, split? no. But would you bother answring questions on a newbies channel? |
| 07:02.51 | firestrm | tzafrir_laptop, yes |
| 07:03.42 | tzafrir_laptop | Well, there's currently no such channel. You can start one and register it if you want |
| 07:03.48 | firestrm | tzafrir_laptop, i do actually like helping others.. thats how projects like asterisk are promoted, one install at a time.. |
| 07:04.33 | firestrm | tzafrir_laptop, and i believe there are no stupid questions, just alot of inqusitive idiots :P |
| 07:05.02 | Mavvie | firestrm: wrong assumption |
| 07:05.09 | Mavvie | about the questions that is. |
| 07:05.23 | Mavvie | stay a little bit longer on this channel and you'll see it for yourself. |
| 07:05.51 | wreck-gar | thanx!!!! it works now |
| 07:06.22 | wreck-gar | it was just that one line, so i guess you guys can go ahead and flame me |
| 07:06.55 | firestrm | i do think that people do need to do need to do some homework, but belittling them for asking, for the most part are well intioned questions, is uncalled for, and only shows one's own ignorance.. |
| 07:07.00 | outtolunc | you got past it.. |
| 07:07.27 | Mavvie | firestrm: aaaaah... *pout* but it is so much fun! |
| 07:07.32 | Qwell | there is a difference between needing help, and unwillingness to learn |
| 07:07.36 | outtolunc | firestorm, are you going to direct that at a particular person? |
| 07:08.06 | Mavvie | Qwell: needing help, unwillingness to learn and not ready yet. Three of them. |
| 07:08.13 | firestrm | no, im not.. its not needed, actions speak for themselves.. |
| 07:08.33 | Mavvie | somebody who comes here and has problems with untarring the asterisk source is "not ready yet" for example. |
| 07:08.50 | outtolunc | yes they do, so when are you going to start answering questions to help people? |
| 07:09.08 | Qwell | Mavvie: I'd argue that the third is a subset of the second |
| 07:09.10 | wreck-gar | you mean saturday morning don't you |
| 07:09.18 | newmedian | yes, but I'm lagged. |
| 07:09.30 | wreck-gar | ahh |
| 07:09.37 | newmedian | ;) |
| 07:11.29 | firestrm | outtolunc, im mostly musing, that it seems that some of the more advanced people on theis channel, seem to be annoyed by newby's and dont give them much of a chance, perhaps its for the better good to move them to a different channel.. |
| 07:11.56 | firestrm | outtolunc, the newby's that is.. |
| 07:12.13 | outtolunc | hehe (backspaces) <G> |
| 07:12.36 | firestrm | eg.. #asteriskImNotWorthey |
| 07:13.42 | outtolunc | as long as i've been doing this (that is a *really* long time), there has always been the reactionary commentary by those that 'have figured it out on thier own' to those that 'seemingly didnt' |
| 07:13.44 | newmedian | An * says what? |
| 07:14.17 | outtolunc | and i seriously doubt it will completely go away |
| 07:14.34 | PTG123 | if you can't take a little grumpy people, use windows |
| 07:15.08 | firestrm | outtolunc: the gift of figuring it out, doesnt belong to everyone, some cant "figure it out". Just be glad if you have it, and use it to help others if you can.. |
| 07:15.36 | tzafrir_laptop | first of all, installing * does not need to be *that* difficult. Take a look at apache and friends , for instance |
| 07:15.56 | newmedian | Isn't it really three levels? The newbie, the intermediate and the crusty developers? |
| 07:15.57 | outtolunc | i agree, but there are times that 'those that cant' just figure (and say) they deserve an answer NOW <G> |
| 07:16.00 | firestrm | tzafrir_laptop: i agree.. |
| 07:16.11 | tzafrir_laptop | A complicated piece of software. And if you want to stack php, mysql, and others: even more. |
| 07:16.38 | tzafrir_laptop | Yet, you usually don't notice it, because most of the required work is now automated. |
| 07:16.42 | firestrm | outtolunc: i agree, if you cant figure it out, you must be respectful of those who can help you.. |
| 07:17.13 | outtolunc | or at least state that you read A,B,C and still don't get it |
| 07:17.20 | firestrm | yes.. |
| 07:17.32 | outtolunc | that 'shows' the desire and willingness |
| 07:17.41 | outtolunc | that's all it *usually takees |
| 07:17.43 | outtolunc | er -e |
| 07:17.51 | Qwell | http://www.catb.org/~esr/faqs/smart-questions.html |
| 07:17.54 | firestrm | many times thats the case with me.. i read A, which conflicts with B and C is going off somewhere totally wierd.. |
| 07:18.11 | tzafrir_laptop | in order to ask smart questions, you need to know enough. |
| 07:18.17 | Mavvie | it would be much easier if telephony wasn't so difficult. |
| 07:18.21 | kryme | Sometimes you read A, it doesn't make sense, and you can't even find B and C. :P |
| 07:18.28 | firestrm | lol |
| 07:18.29 | firestrm | yes |
| 07:18.36 | Qwell | the above link outlines all of this |
| 07:18.38 | outtolunc | i *know asterisk fairly well, i beat myself silly yesterday (because of a bad copy/paste) <G> |
| 07:18.42 | PTG123 | if you want to use asterisk, you better know C :) |
| 07:18.47 | tzafrir_laptop | In order to parse correctly the results of a google search you need to know enough jargon to be able to ignore irrelevant results |
| 07:18.56 | tzafrir_laptop | PTG123, why? |
| 07:19.15 | PTG123 | so you can figure out stuff easier, and fix things when they break :) |
| 07:19.37 | PTG123 | if i wasn't a developer, i would have given up on asterisk long ago |
| 07:19.53 | firestrm | outtolunc: sometimes you also just need another pair of eyeballs lokking at the problem.. i cant tell you how many times ive had some stupid assembler app broken because of a missplaced typo.. |
| 07:20.04 | outtolunc | hehe |
| 07:20.06 | kryme | I just scratched my head and cried a lot until I found Asterisk@Home. :P |
| 07:20.42 | tzafrir_laptop | PTG123, OTOH, if you're a C programmer the dialplan may give you BASIC nightmares and scare you away |
| 07:20.44 | outtolunc | it's really fun when you cross compile code from 3 diff languages <G> |
| 07:21.08 | firestrm | kryme, i tried @home, i guess im too much of a low level kind of guy, it gave me a headache, and i couldnt figure out where everything was, so i wound up hacking the configs by hand anyways.. |
| 07:21.21 | Mavvie | tzafrir_laptop: it reminded me about the time I had to program PLCs. |
| 07:21.38 | kryme | I just wanted something easy to setup and try, and it made it that simple, so I was quite happy. |
| 07:21.39 | firestrm | outtolunc, try, 15k lines of verilog code, plus test bench.. |
| 07:21.57 | tzafrir_laptop | firestrm, any easy interface for asterisk's dialplan is bound to limit you |
| 07:21.58 | firestrm | outtolunc: im still seeing a counsoler about that one.. |
| 07:23.03 | firestrm | tzafrir_laptop, i spend a day on it, could get it to pick the phone, even had voice prompts, but for the life of me i couldt figure out how to set up extensions.. |
| 07:23.08 | outtolunc | never dealt with verilog code, but just the last couple weeks i'd say i consumed about 30k lines of code in 3 diff languages, just to piece together something simple |
| 07:23.56 | firestrm | outtolunc: verilog, is kinda like c, except all allong your writing another program (test bench) that proves the first one works as expected.. |
| 07:24.29 | firestrm | outtolunc: its used for avionics alot because of that.. |
| 07:24.45 | outtolunc | never heard that (i've only been written code for the last 28 years though <G>) |
| 07:25.11 | outtolunc | never got into avionics |
| 07:25.40 | firestrm | outtolunc: pay= good, but not fun.. you software MUST work every time.. |
| 07:26.11 | outtolunc | i understand the requirements... i've got a few friends that did nasa work |
| 07:26.16 | wreck-gar | don't they use it for hardware layout too |
| 07:26.27 | wreck-gar | ? |
| 07:26.53 | firestrm | yes, verilog and vhdl is used alot for chip design.. (ASIC and FPGA's) |
| 07:27.05 | Mavvie | that explains why it is in ports/cad |
| 07:27.24 | wreck-gar | yeah i was gonna have a class in that before i dropped out of the e-school |
| 07:27.41 | wreck-gar | which was a big mistake |
| 07:27.45 | Qwell | You dropped out of e-school? |
| 07:27.50 | firestrm | its actually a cool language, in one program you can define the microprocessor hardware and the software that runs on it.. |
| 07:28.17 | wreck-gar | yeah |
| 07:28.26 | wreck-gar | i got feed up with the jack ass teachers |
| 07:28.37 | wreck-gar | so i'm gettin my applied math degree |
| 07:28.43 | wreck-gar | then goin else where for e-school |
| 07:28.52 | firestrm | lol |
| 07:29.19 | wreck-gar | apparently alot of kids that started my year in the EE program got dick and are still at the school |
| 07:29.24 | wreck-gar | ~6 years later.... |
| 07:29.56 | wreck-gar | had 6 in my applied algebra class that i thought had graduated. |
| 07:30.14 | wreck-gar | but apparently i had thought wrong |
| 07:30.49 | wreck-gar | is there a place i can go read about all this signaling junk? |
| 07:30.55 | Qwell | google |
| 07:30.57 | wreck-gar | like loopstart and kewlstart and all that |
| 07:31.05 | firestrm | google |
| 07:31.27 | wreck-gar | ok |
| 07:31.33 | outtolunc | well i'd do a google for 'telecom glossary' |
| 07:31.51 | tzafrir_laptop | wreck-gar, voip-info.org and wikipedia tend to provide information about many unknown terms |
| 07:34.14 | firestrm | well ppl, its been a slice, but im off early in the morning to go out ATV'n so must sleep.. gnite.. |
| 07:34.39 | outtolunc | <- boogie on teh bayou |
| 07:34.41 | outtolunc | er the |
| 07:34.58 | firestrm | teh works :) |
| 07:35.15 | outtolunc | yeah should be fun |
| 07:35.47 | outtolunc | downtown campbell, for any bay area folks that may be hangin |
| 07:36.11 | firestrm | its promised to be.. nothing like ATV's, Booze and Guns to make for an interesting day :) |
| 07:36.49 | outtolunc | wilber you are supposed to shoot it, not run it over <G> |
| 07:37.09 | firestrm | no run over it first, easier to shoot |
| 07:38.00 | firestrm | of better, get it drunk, then wound it, then run it over.. |
| 07:38.20 | outtolunc | any * Q's before i head out? |
| 07:38.39 | firestrm | outtolunc: why do men have nipples? |
| 07:38.56 | wreck-gar | yeah where can i get the heltiva (sp) font so i can do make progdocs |
| 07:39.16 | wreck-gar | and not have it complain |
| 07:39.20 | wreck-gar | J/K!! |
| 07:39.50 | outtolunc | <PROTECTED> |
| 07:39.50 | outtolunc | <PROTECTED> |
| 07:40.21 | wreck-gar | yeah you need graphviz too |
| 07:40.21 | outtolunc | as for the nipple Q, umm i plead the 5th |
| 07:40.27 | firestrm | rotfl |
| 07:40.38 | wreck-gar | for the dot program |
| 07:40.49 | firestrm | good night.. :) |
| 07:40.57 | outtolunc | gnight |
| 07:41.21 | outtolunc | hmm way back when, i don't remember needing anything else for progdocs |
| 07:41.38 | wreck-gar | maybe the doxygen i got was retarted then.... |
| 07:41.42 | outtolunc | i had to mod the code tho <G> |
| 07:41.53 | wreck-gar | but i def had to go d/l graphviz |
| 07:42.12 | wreck-gar | i think i might have been able to get way with downloading the webdot program |
| 07:42.39 | wreck-gar | but i didn't feel like downloading it and compiling it to find out it didn't have what i needed. |
| 07:43.58 | outtolunc | mod /usr/src/asterisk/contrib/asterisk-ng-doxygen |
| 07:44.05 | outtolunc | HAVE_DOT = YES |
| 07:44.10 | outtolunc | change to NO |
| 07:44.16 | outtolunc | that's ALL you had to do |
| 07:45.00 | wreck-gar | i'm not quite that bright with unix/linux |
| 07:45.07 | wreck-gar | i use my mac most of the time |
| 07:45.14 | outtolunc | np |
| 07:45.22 | outtolunc | maybe it will help someone else |
| 07:45.23 | wreck-gar | and am happy being ignorant of most unix like things |
| 07:45.32 | wreck-gar | =0) |
| 07:46.10 | outtolunc | anything else? |
| 07:46.30 | wreck-gar | no, but thank you very much for helping me |
| 07:46.35 | outtolunc | oh what the hell.. afk drink (i'll stay a bit) |
| 07:46.43 | outtolunc | np, glad to help |
| 07:48.09 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 07:48.31 | wreck-gar | and being patient. |
| 07:48.41 | wreck-gar | with my slow ass-self |
| 07:50.53 | outtolunc | believe me, i'm used to worse <G> |
| 07:51.17 | *** join/#asterisk JunK-Y (~junky@modemcable174.107-81-70.mc.videotron.ca) |
| 07:51.44 | outtolunc | meaning i'm surrounded (day job) with blithering idiots <G> |
| 07:52.47 | wreck-gar | whats the day job |
| 07:53.11 | outtolunc | is mgr for a company that has 7 locations in NoCal |
| 07:53.33 | wreck-gar | nice |
| 07:53.35 | outtolunc | by myself |
| 07:54.20 | outtolunc | everytime i hire a new guy, he never lasts, the office managers can't deal with them |
| 07:54.37 | outtolunc | <- lone dweeb |
| 07:55.14 | wreck-gar | tech work? |
| 07:55.25 | outtolunc | i do everything |
| 07:55.53 | outtolunc | including tech work on various gear |
| 07:56.10 | wreck-gar | cool gear? |
| 07:56.22 | outtolunc | i don't think so <G> |
| 07:56.39 | outtolunc | but seems few deal with things like i do |
| 07:56.59 | wreck-gar | ahh |
| 07:57.14 | outtolunc | monster 96 port analog switches (pbx) |
| 07:57.43 | wreck-gar | wow |
| 07:57.44 | outtolunc | muxes and shit |
| 07:58.01 | outtolunc | it's old stuff |
| 07:58.02 | wreck-gar | i got thrownin to diagin an NEC electra Elite IPK |
| 07:58.24 | wreck-gar | you ever seen an osicom boz? |
| 07:58.26 | wreck-gar | box |
| 07:58.34 | outtolunc | nope |
| 07:58.34 | wreck-gar | i think i spelled it right |
| 07:58.56 | outtolunc | my background is old dialogic PEB stuff |
| 07:59.05 | wreck-gar | i think they are used to link offices over a leased line |
| 07:59.24 | wreck-gar | dialogic PEB?? |
| 07:59.40 | outtolunc | yeah, old old old ... stuff |
| 07:59.50 | outtolunc | pre dsp |
| 07:59.56 | wreck-gar | this osicom box was too man |
| 08:00.14 | wreck-gar | i googled it and it came up with like one link |
| 08:00.50 | outtolunc | our boxes in a t1 config run a DTI124, 2 AMX81's, D41B's |
| 08:01.00 | *** join/#asterisk [cc]smart (~smart@gw.ptr-62-65-149-158.customer.ch.netstream.com) |
| 08:01.47 | wreck-gar | lol |
| 08:01.57 | wreck-gar | so apparently intel owns that company now |
| 08:02.15 | outtolunc | yes |
| 08:02.19 | newmedian | D41ESC and earlier re: Dialogic, right? Nostalgia. |
| 08:02.32 | outtolunc | intel grabbed dialogic what 5ish years ago |
| 08:02.49 | wreck-gar | so you do a decent amount of telecom stuff then i guess |
| 08:02.50 | outtolunc | yeah d41D was the first dsp board |
| 08:02.51 | newmedian | Did you do any VOS work? |
| 08:03.12 | outtolunc | oh yeah |
| 08:03.33 | newmedian | kids these days *thumps desk* they have it simple ;) |
| 08:03.42 | wreck-gar | can you tell me what kind of freakin connection comes in over an AMP50 connector? |
| 08:04.14 | outtolunc | yeah stuff all that gear in an expansion chassis and connect to a 386/486 and rock <G> |
| 08:04.39 | outtolunc | comes in over? |
| 08:04.53 | outtolunc | it's 'simply' a connector |
| 08:05.21 | wreck-gar | o |
| 08:05.24 | outtolunc | hell i send rs232 over amp conns <G> |
| 08:05.40 | wreck-gar | the company i daiged the NEC for had them |
| 08:05.55 | wreck-gar | there system is more or less slowly dying |
| 08:05.56 | newmedian | Now you're reminding me of my old LA-34 Decwriter. |
| 08:06.01 | outtolunc | they are a 'common' telecom connector |
| 08:06.09 | wreck-gar | O |
| 08:06.20 | wreck-gar | *their |
| 08:06.34 | wreck-gar | does digium sell a card that can connect to that type of thing? |
| 08:06.51 | outtolunc | not directly |
| 08:07.05 | outtolunc | you need a t1 card, then use a chanbank |
| 08:07.16 | outtolunc | the chanbank will have a 50 conn |
| 08:07.21 | wreck-gar | ahh |
| 08:07.23 | outtolunc | (usually) |
| 08:07.23 | newmedian | outtolunc: any particular channel bank you'd recommend? |
| 08:07.42 | outtolunc | adtran makes great gear |
| 08:07.59 | outtolunc | i just sold a csu ace on ebay |
| 08:08.26 | wreck-gar | could i just take that amp 50 and figure out the individual lines on it, and kinda kludge on the RJ-11? |
| 08:08.30 | outtolunc | (and no i'm not telling you the actually make/model of the channel bank i use, until i have enough stock <G>) |
| 08:08.49 | newmedian | heh |
| 08:08.50 | outtolunc | ask me in 6 weeks <G> |
| 08:08.56 | wreck-gar | isn't all of adtrans gear fairly expensive anyways. |
| 08:09.14 | outtolunc | all chanbanks are expensive |
| 08:09.22 | wreck-gar | true |
| 08:09.26 | outtolunc | there are some that you 'can' get for cheap on ebay |
| 08:09.45 | wreck-gar | cheap from ebay usually equals shade |
| 08:09.48 | wreck-gar | shady |
| 08:10.03 | outtolunc | if there are some here that don't know.. kram used to work for adtran |
| 08:10.29 | outtolunc | so you can't go wrong with them |
| 08:11.08 | *** join/#asterisk revolt (~revolt@218.111.51.8) |
| 08:11.53 | newmedian | Chuji pointed me to http://www.voip-info.org/tiki-index.php?page=Digital%20Telephone%20Adapters earlier, when I was asking " I've got someone with an Option 11 system in a non VOIP setup, with 20-30 Meridian phones (e.g. M2008). If I move him to an * setup, can(/should) the phones be re-used (via Channel Bank I assume) to *, or are they basically feature locked to the Nortel PBX?" |
| 08:12.02 | wreck-gar | man i wish some place delivered food a 4 in the mornign |
| 08:12.34 | newmedian | After reading http://lists.digium.com/pipermail/asterisk-users/2005-April/100388.html thread I am somewhat cautious (about price & functionality). |
| 08:13.09 | outtolunc | never used that stuff myself, so i can't say either way |
| 08:13.11 | newmedian | Essentially if someone has an Option 11, the most "economical" thing to do is scrap it all? and go with VOIP phones and an Asterisk box. |
| 08:13.56 | wreck-gar | any of you guys mess with asterisk's data capabilites? |
| 08:14.15 | wreck-gar | like when you jack in the t-1 having it section off part of it has voice and the rest as data |
| 08:14.18 | outtolunc | the way i look at it, a te4xxp and a channel bank or 2 is the way to got for a good sized setup |
| 08:14.26 | newmedian | It's hard to tell someone to scrap an investment, but the alternative of trying to keep those Meridian phones going looks very expensive. |
| 08:15.13 | outtolunc | what is the current annual support costs for a meridian system <G> |
| 08:15.26 | newmedian | Yeah, well. You got it. |
| 08:15.28 | outtolunc | hehe |
| 08:15.59 | newmedian | Not to mention charging $$$$$ for all the options & upgrades and software packs that ultimately.... deliver part of what Asterisk delivers. |
| 08:15.59 | outtolunc | it's a mentality |
| 08:16.53 | outtolunc | if they are not willing to give up the 4 hour support and annual payments for the privledge of being held hostage.. then you won't get asterisk in |
| 08:17.25 | outtolunc | 'its that simple' (tm) |
| 08:17.51 | newmedian | Agreed; And the service is usually ridiculous. I've had to deal with corrupt Option 11 boxes before. :( nightmare. Particularly in 24x7 scenarios where they won't even let you reboot. |
| 08:18.59 | outtolunc | noted: an asterisk box (for a larger company) requires someone in the know also, but asterisk is growing as so is the support base |
| 08:19.18 | outtolunc | and you aren't 'locked' to a feature at a price |
| 08:19.35 | outtolunc | (unless you want something like ss7 or 729 <G> |
| 08:19.39 | outtolunc | ) |
| 08:20.13 | outtolunc | stock pbx features are there |
| 08:20.23 | newmedian | If you were providing termination, say PSTN->VOIP or VOIP->PSTN, is there any "ideal" configuration of (voice) line? e.g. PRI NI-2 (and a 10 meg ethernet burstable link for the Internet side)? |
| 08:20.52 | outtolunc | well you are talking 2 different things |
| 08:21.00 | outtolunc | they are separate |
| 08:21.07 | outtolunc | the pstn side, is just that |
| 08:21.16 | outtolunc | the voip side is just that |
| 08:21.41 | outtolunc | each has there own requirements |
| 08:22.12 | outtolunc | on the pstn side, you 'should' get a pri |
| 08:22.29 | newmedian | It's 4:23 here, so I'm not quite focused ;) |
| 08:22.42 | *** join/#asterisk implicit (~implicit@ip68-7-149-247.sd.sd.cox.net) |
| 08:22.44 | syle | how do people offer local calling free is what i want to know hehe |
| 08:23.04 | outtolunc | on the voip side get whatever bandwidth (and the proper switches) to make sure you can handle the voice quality for the codec you are using for the quantity of calls you are making |
| 08:23.17 | wreck-gar | your a minute off newmedian |
| 08:23.21 | wreck-gar | =P |
| 08:23.44 | newmedian | I need to sync, I guess. :) |
| 08:24.07 | wreck-gar | lol |
| 08:24.17 | *** join/#asterisk Singod (~ud@netblock-66-159-209-133.dslextreme.com) |
| 08:24.26 | outtolunc | the first thing i'd suggest is to go some codec tests on the voip side |
| 08:24.57 | outtolunc | the pstn side is fairly static, meaning you get what you need |
| 08:25.49 | newmedian | But nothing in particular about the signalling, if one had a choice, re: the PRI/T1? |
| 08:26.48 | *** join/#asterisk jeffgus (~jeffgus@alpha.zimage.com) |
| 08:28.44 | outtolunc | newmedian: the diff between t1/pri is that t1 the signalling is *inband* whereas a PRI is out of band |
| 08:28.51 | outtolunc | that means |
| 08:29.28 | JunK-Y | dchan |
| 08:29.40 | newmedian | In a simplistic fashion, DID >> PRI -> TE110P -> Asterisk -> Internet (and vice-versa); Or for SOHO DID >> PRI -> TE410P -> Channel Bank and Asterisk -> Internet |
| 08:29.41 | outtolunc | with a t1, you get about a total of 6 'causes' that happen to the line, whereas a pri its more like 26 |
| 08:30.26 | outtolunc | and with the t1, you have to 'determine those causes' WITHIN the audio of the channel |
| 08:30.54 | newmedian | outtolunc: That's a clear explanation. :) |
| 08:31.07 | outtolunc | i hoped so |
| 08:32.15 | newmedian | Technologically, would it be a better choice to get a TE410P(/TE405P) even if you're not using a channel bank and you've only got 1 PRI? (i.e. differences in firmware, circuit, etc.)? |
| 08:32.25 | newmedian | versus getting a TE110P I mean |
| 08:32.35 | outtolunc | stop |
| 08:32.50 | wreck-gar | they are the samething, just the Te410 has 4 connections |
| 08:33.02 | wreck-gar | vs the 1 on the 110P |
| 08:33.04 | outtolunc | the difference between getting a single or quad is NEED |
| 08:33.24 | newmedian | thanks for the qualification. :) |
| 08:33.50 | outtolunc | if all your traffic is VOIP inbound, and you only want 1 t1/pri outbound to pstn a single is fine |
| 08:33.53 | outtolunc | BUT |
| 08:34.29 | outtolunc | if you want to have 2 t1/pri to pstn and a channel bank for some local fxs channels |
| 08:34.35 | outtolunc | then get the quad |
| 08:34.38 | wreck-gar | does one really need a channel back if u have a stright T1 coming in? |
| 08:34.50 | newmedian | One thing I have found, after getting back into Telephony development, is that with Asterisk in particular all of the documentation & information seems fragmented, torn up and spread across the winds. Large repositories (of sometimes outdated information) but all of the relevant nuggets you have to get a bag and start combing the hills picking up the good bits until you've got a complete set to spread out in front of you. |
| 08:35.12 | outtolunc | then channel bank is ONLY useful if you want 'real phones' at the user/rep station |
| 08:35.18 | outtolunc | OR |
| 08:35.19 | wreck-gar | ahhh |
| 08:35.35 | outtolunc | if you want to SPLIT the input of the t1/pri |
| 08:35.44 | wreck-gar | split it for??? |
| 08:35.48 | wreck-gar | voice/data? |
| 08:35.49 | *** join/#asterisk RoyK (~roy@179.80-203-29.nextgentel.com) |
| 08:35.52 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
| 08:36.05 | outtolunc | say 10 channels pstn, rest for internet connect |
| 08:36.16 | wreck-gar | according to the docs |
| 08:36.25 | outtolunc | so yes |
| 08:36.29 | wreck-gar | those TEx cards |
| 08:36.35 | wreck-gar | with those TE cards |
| 08:36.47 | wreck-gar | it can handle passing data through it |
| 08:36.49 | outtolunc | royk, was that you with the 1xxxxxxxx asterisk mem usage? |
| 08:37.10 | outtolunc | (message in the ML) |
| 08:37.41 | wreck-gar | . For example, the card can support 12 channels dedicated to voice and 12 to data while passing all traffic through to the Asterisk PBX, which reliably routes the channels to their designated locations. This eliminates the need for an external router. |
| 08:38.02 | outtolunc | yes it does |
| 08:38.23 | outtolunc | but *usally* the reason they want the 'breakout' is for fxs devices |
| 08:38.34 | outtolunc | hence the need for a channel bank |
| 08:38.40 | wreck-gar | but thats internally right? |
| 08:38.41 | JunK-Y | RoyK: see http://bugs.digium.com/view.php?id=4223 this is the only one i know so far (except something related to .call files, probably pbx_spool). |
| 08:38.46 | outtolunc | yes |
| 08:39.12 | outtolunc | remember, a channel bank 'can' be fxs or fxo or a combination of both |
| 08:39.23 | wreck-gar | that i didn't know |
| 08:39.28 | wreck-gar | i thought it was one or the other |
| 08:39.35 | wreck-gar | but not both at the same time |
| 08:39.40 | outtolunc | they usually have 'modules' |
| 08:39.47 | outtolunc | and whatever the port size |
| 08:39.57 | wreck-gar | i c |
| 08:39.59 | outtolunc | 2,4,8,12,24 |
| 08:40.03 | JunK-Y | wreck-gar: just buy different modules (some for FXO, the rest for FXS). |
| 08:40.12 | RoyK | JunK-Y: that's not the one |
| 08:40.19 | wreck-gar | thats a diff card Junk-Y |
| 08:40.39 | wreck-gar | you are talkin about the wildcat series of cards |
| 08:40.55 | JunK-Y | wreck-gar: for adtran yes. you have a module fxo on one side, you've the fxs on the other. |
| 08:41.03 | wreck-gar | i mean the TFM cards |
| 08:41.15 | wreck-gar | ohh sorry |
| 08:41.17 | outtolunc | TFM? |
| 08:41.18 | newmedian | I'm burning' out so I'll say ciao, but thanks outtolunc for the useful info. g'night all. |
| 08:41.21 | wreck-gar | damn i can't type |
| 08:41.23 | JunK-Y | he means tdm. |
| 08:41.29 | outtolunc | <G> |
| 08:41.29 | JunK-Y | :) |
| 08:41.44 | wreck-gar | closing in on 5 AM |
| 08:41.46 | JunK-Y | RoyK: in which files are the mem leak? |
| 08:42.00 | outtolunc | i'm about toast, it's 1:40 locally |
| 08:42.03 | wreck-gar | typin starts to fall apart |
| 08:42.15 | JunK-Y | outtolunc: come on its 4:40 here :) |
| 08:42.17 | wreck-gar | 4:40 on the east coast man |
| 08:42.21 | outtolunc | <- tis old remember <G> |
| 08:42.34 | outtolunc | hell i'll be 40 this year |
| 08:42.36 | wreck-gar | and i got up at 7 in the morning |
| 08:42.47 | wreck-gar | lol |
| 08:43.54 | *** join/#asterisk jeffik (jefik@69.158.19.117) |
| 08:43.57 | wreck-gar | i think i might talk my friend in to buyin a TE110P |
| 08:44.09 | *** join/#asterisk Koshatul (~evangelio@ip157.net65.ipnetworks.net.au) |
| 08:45.43 | RoyK | JunK-Y: how am I supposed to know? |
| 08:46.20 | RoyK | JunK-Y: the box is in production and all I see is continous amount of used memory |
| 08:46.42 | *** join/#asterisk cmk (~cmk_@p54A3EE4E.dip.t-dialin.net) |
| 08:46.44 | JunK-Y | gdb? |
| 08:46.47 | JunK-Y | head ? |
| 08:49.32 | outtolunc | gnight folks |
| 08:50.36 | RoyK | JunK-Y: 1.0.7 with sipfriends and some AGI |
| 08:50.38 | RoyK | that§s it |
| 08:51.13 | JunK-Y | at beginning, agi isnt so good. and i cant talk about stable, dont know that branch. |
| 08:51.26 | JunK-Y | im running head on prod machines. |
| 08:52.39 | wreck-gar | how is that working out for you JunK-Y |
| 08:53.04 | JunK-Y | head? that's great for me. |
| 08:54.12 | RoyK | anyone that knows a decent asterisk coder that can help me sort this out? |
| 08:54.41 | JunK-Y | RoyK: submit a bug with more infos? |
| 08:57.18 | wreck-gar | later on people. thanks to everyone that gave me a hand with my silly little problem |
| 09:05.09 | *** join/#asterisk OzJames79 (~opera@CPE20320889-1842-1.gex.ncable.net.au) |
| 09:08.18 | *** part/#asterisk OzJames79 (~opera@CPE20320889-1842-1.gex.ncable.net.au) |
| 09:10.17 | *** join/#asterisk slamb (~slamb@adsl-69-230-8-158.dsl.pltn13.pacbell.net) |
| 09:24.22 | *** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 09:26.19 | *** join/#asterisk Ridgeback (~Ridgeback@ppp218-189.lns1.adl2.internode.on.net) |
| 09:26.29 | Ridgeback | hey hey |
| 09:26.38 | Zeeek | ho ho |
| 09:27.10 | Ridgeback | quick question should I fire off PrivacyManager as priority s,1 or should I do an s,1,Answer then s,2,PrivacyManager? |
| 09:28.04 | Mavvie | does privacymanager answer the call itself? |
| 09:28.13 | Ridgeback | i dont know |
| 09:28.20 | Mavvie | well, only one way to find out then! |
| 09:29.37 | JunK-Y | <PROTECTED> |
| 09:29.37 | JunK-Y | channel. |
| 09:30.06 | JunK-Y | If no Caller*ID is sent, PrivacyManager answers the |
| 09:30.06 | JunK-Y | channel and |
| 09:30.07 | Ridgeback | but do i need to have it s,1,Answer first? |
| 09:30.11 | Ridgeback | oh ok |
| 09:30.19 | Ridgeback | no need for answer then |
| 09:30.20 | JunK-Y | shit, 5h30 already |
| 09:30.21 | JunK-Y | see ya |
| 09:30.50 | revolt | does anyone know where I can download modprobe for redhat? |
| 09:31.02 | revolt | i don't have any hardware cards installed |
| 09:31.10 | *** join/#asterisk Falstaf (falstaf@diana.pervo.nu) |
| 09:31.14 | Mavvie | revolt: doesn't that come included with the OS? |
| 09:31.21 | Ridgeback | huh? should be part of the basic install |
| 09:31.30 | revolt | i dunno.. |
| 09:31.38 | Ridgeback | try /sbin/modprobe |
| 09:31.43 | revolt | ah |
| 09:31.59 | revolt | found it :) |
| 09:32.33 | revolt | thanks Ridgeback |
| 09:32.39 | Ridgeback | no prob |
| 09:38.09 | *** join/#asterisk TheEmperor (TheEmperor@218.111.51.8) |
| 09:40.18 | *** join/#asterisk RestLessGemini (~umairbari@202.142.189.86) |
| 09:41.59 | *** join/#asterisk revolt_ (~revolt@218.111.51.8) |
| 09:45.05 | Newbie___ | guy, if i want to use g729. that mean i would use disallow=all allow=g729 ? |
| 09:45.15 | Ridgeback | yep! |
| 09:45.38 | Newbie___ | can i do that in sip.conf and iax.conf ? |
| 09:46.19 | Ridgeback | okay, sip.conf you leave as allow-ulaw for your phone to phone connectivity. then in iax.conf disallow=all, allow=g729, allow=ulaw |
| 09:46.48 | Ridgeback | dont want to waste g729 licenses on internal comms! what a waste! |
| 09:47.01 | Newbie___ | got it, tks |
| 09:47.21 | Newbie___ | sure hope g729 has a better voice quality |
| 09:47.41 | Ridgeback | its nice, not as good as ulaw |
| 09:48.41 | Newbie___ | testing on 2 licenses for now |
| 09:54.40 | slamb | Newbie___: Its quality is not going to be very good, since it's 8 kbit/s |
| 09:55.06 | slamb | Newbie___: I'm struggling with Asterisk voice quality now...I made some records of different bitrates... http://www.slamb.org/insane/ |
| 09:55.58 | slamb | Newbie___: Err, by 8 kbit/s, I mean 8 kHz. |
| 09:56.27 | Ridgeback | 8khz sampling rate is the same as ulaw |
| 09:56.29 | slamb | Newbie___: anyway, my conclusion is that 32 kHz speex sounds good. I wasn't happy with even uncompressed 16 kHz. |
| 09:56.37 | slamb | Ridgeback: yeah. |
| 09:57.09 | Ridgeback | i have pretty good success with g729 |
| 09:57.54 | Newbie___ | our calls tends to have delay, but the ping result said is <50ms. was told by the other side to use 729 |
| 09:57.57 | slamb | Ridgeback: well, if it sounds good to you...but if you listen to the samples I made at different frequencies, you'll hear the difference between the 8, 16, and 32. the 32 sounds much better. |
| 09:58.49 | Ridgeback | well yeah, higher sampling rates ouwld soundbetter |
| 09:59.11 | Newbie___ | higher sampling rates takes up more bandwidth, no ? |
| 09:59.16 | Ridgeback | my polycom nor my grandscreams supprt speex, other wide i would give it a shot |
| 09:59.19 | Ridgeback | correct |
| 09:59.51 | slamb | Newbie___: For the same compression, yes. But the 32kHz speex will use less bandwidth than the 8kHz ulaw. |
| 09:59.58 | Ridgeback | at the moment i have a call up with g729 and jnettop shows 10.1Kbytes/sec or about 80 kbits per second |
| 10:00.03 | Newbie___ | i am short of bandwidth here, i guess in my scenario, i would choose 8 vs 16 or 32 |
| 10:00.07 | Ridgeback | im sorry gogoed that up |
| 10:00.20 | Ridgeback | that was a ulaw call at 10.Kbytes/s |
| 10:00.43 | Ridgeback | my g729 calls get around 3.1kBytes/s or 24kbits/s |
| 10:01.10 | Ridgeback | im happy wth the g729 quality. it clips the highs a bit... no big deal to me. |
| 10:01.24 | slamb | Ridgeback: ulaw will be 8 kHz * 16 bits/sample = 128 Kbit/s = 16 |
| 10:01.29 | Ridgeback | atleast i can now surf pr0n while calling the 976 numbers ;) |
| 10:01.32 | slamb | 16 kbyte/sec |
| 10:01.45 | slamb | plus protocol overhead |
| 10:01.56 | Ridgeback | slamb, umm according to my research tat should be 8 bit samples no? |
| 10:02.06 | slamb | really? hmm, let me check. |
| 10:02.09 | Newbie___ | well, like i always tell my boss. u want good quality, use the normal phone and dial |
| 10:02.11 | Ridgeback | becuase my net meter shouldn't lie ;) |
| 10:02.34 | slamb | Yeah, I'd hope not. |
| 10:02.38 | [Lamer] | Hi there, I had videosupport=yes and allow=h623 in [general] of the sip.conf but cannot have the video mode between two SIP phones |
| 10:02.46 | [Lamer] | whatelse do I need? |
| 10:02.57 | Ridgeback | slamb, ulaw is th same as DS0 in telco DS1 signalling which is 8000 samples/sec * 8 bits of sampling = 64000 bits/s |
| 10:03.13 | Ridgeback | 24 DS0 at 64000 bits equals 1.544mbit T1 |
| 10:03.21 | Ridgeback | math is so fun! |
| 10:03.23 | Ridgeback | not! |
| 10:03.25 | Ridgeback | lol |
| 10:04.00 | slamb | okay. is everything else using 8-bit samples, too, then? |
| 10:04.02 | Ridgeback | [Lamer], ummm H323 is a bitch. is the Asterisk server behind a NAT firewall |
| 10:04.08 | Ridgeback | slamb, i belive so yes. |
| 10:04.31 | slamb | Hmm. That sucks. |
| 10:04.53 | Ridgeback | slamb, yeah it does, probably has to do with the hardware that does the actual recording i.e. your handset that is the limit |
| 10:05.14 | Ridgeback | your asterisk box doesnt do sampling it only transcodes |
| 10:05.29 | slamb | In my case, I've been playing with softphones. So the recording happens at 44.1kHz, 16-bit. That's what my PowerBook's audio hardware does. |
| 10:05.53 | slamb | or 48kHz, if I use the iSight microphone. |
| 10:05.56 | *** join/#asterisk coppice (~chatzilla@43.198.17.210.dyn.pacific.net.hk) |
| 10:05.57 | Ridgeback | slamb, ahhhh hmmm thats interesting. then it must downsample to 8 in the codec |
| 10:06.20 | Ridgeback | yeah imagine sending a CD quality sample across a phone channel! |
| 10:06.58 | coppice | at one time that was one of the goals of ISDN |
| 10:07.12 | Ridgeback | 44100 * 16 bits = 705000 bits per second! or 88200 bytes/sec! |
| 10:07.30 | coppice | before compression, of course :-) |
| 10:07.34 | Ridgeback | lol |
| 10:08.07 | Ridgeback | 705kbit link would be needed for one call at CD quality! imagine that! |
| 10:08.26 | slamb | Ridgeback: See, I've been spoiled by skype. They're apparently using 32 kHz iLBC. Not sure now if it's 8-bit or 16-bit. I want to get audio quality that matches that. |
| 10:09.08 | coppice | skype uses 16k samples/second. You can easily tell and f from an s, which normal phone lines don't permit |
| 10:09.22 | Ridgeback | iLBC isnt all that sneezy. your mic and hardware isn't perfomring at that level. its a standard 8 bit compressor also. g729 is closest match. ulaw is the best becuase it ins't compressing anything |
| 10:09.56 | coppice | ulaw is compressed, just not very much |
| 10:09.57 | slamb | Ridgeback: but ulaw is only at the 8 kHz. So even though it's not compressing, it still is lower quality. |
| 10:10.20 | Ridgeback | coppice, really? whats the compression algorith. casue the RFC shows no compression |
| 10:10.21 | coppice | skype is far better than ulaw |
| 10:10.54 | coppice | ulaw is defined in an ITU spec, not and RFC. ulaw compresses 12/13 bit words to 8bit words |
| 10:11.07 | Ridgeback | nah, i think its more of a marketing hype, how can compressing your ausi with a lossful compression equate to better quality than something not compression at all? |
| 10:11.22 | coppice | bandwidth |
| 10:11.48 | Ridgeback | coppice, but thats not compression, actraully.. gzip, bzip etcc are compression alogirthims, not bit for bit tranlsations |
| 10:12.02 | coppice | ulaw cuts off at about 3.5kHz. skype at about 7.5kHz |
| 10:12.08 | slamb | Ridgeback: because you're starting with much, much more data. |
| 10:12.16 | coppice | that *is* compression |
| 10:12.19 | tzafrir_laptop | when you don't compress you'll have to drop packets |
| 10:12.21 | slamb | It's not marketing hype. Use it. You'll see. |
| 10:12.25 | *** join/#asterisk r0d3nt (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 10:12.53 | Ridgeback | slamb, i tried skype. i didn;'t think it was all that much different than anormal ulaw call |
| 10:13.27 | coppice | it depends on connection quality, and the hardware you use. it should be far better than ulaw |
| 10:14.08 | Stereo | Ridgeback: skype sounds better than phones, and certainly better than ulaw when you have enough bandwidth |
| 10:14.15 | slamb | btw, the word you're looking for is "lossless" when describing gzip and bzip2 compression. where as jpeg, mp3, speex, etc. are all "lossy". |
| 10:14.31 | Ridgeback | right thats true |
| 10:14.32 | coppice | even ulaw is lossy |
| 10:14.57 | Zeeek | inlaws are lossy, too |
| 10:15.08 | Zeeek | disallow=inlaw |
| 10:16.41 | coppice | its sad that most VoIP is still locked into 1900 style 3.5kHz bandwidth calls. |
| 10:17.21 | [Lamer] | Ridgeback: Sorry, i was away and no, my * is not behind a firewall, one of the SIP phones is |
| 10:17.48 | Ridgeback | voice doesnt extend pass that really. would need higer waulity codecs for music though |
| 10:18.01 | Ridgeback | s/wuality/quality |
| 10:19.08 | slamb | Ridgeback: It definitely does extend past it. I'm tempted to set up a double-blind experiment on my little test files at 8kHz, 16kHz, 32kHz, and 48kHz. I don't know enough people out here to do it, though. |
| 10:19.28 | Ridgeback | what i mean is the range for human voice is only about 4khz |
| 10:19.29 | [Lamer] | Ridgeback: my case is about not having video mode between two SIP phones |
| 10:19.43 | Ridgeback | unless your mariah carey singing 15 octaves above everyone else ;) |
| 10:19.53 | slamb | Then why can I distinguish between recordings of a voice at those sampling rates? |
| 10:19.54 | coppice | wrong. at 3.5kHz you can't tell between unvoiced sounds - e.g. f and s. You need about 7-8kHz for that. The bandwidth of voice is actually wider than most music. The difference between 10kHz and 15kHz bandwidth is more obvious with speech than with music. |
| 10:20.17 | [Lamer] | Ridgeback: while the sip.conf has the settings videosupport=yes and allow=h623 |
| 10:20.17 | Ridgeback | slamb, beacause of the change of damage with the compression, your going from one level of quality ot another |
| 10:20.28 | Newbie___ | huy, is it possible to use ATA to send a fax to * , * pick up a POTs and send out the fax ? |
| 10:20.28 | slamb | Ridgeback: these are uncompressed files |
| 10:20.55 | Ridgeback | coppice, really wheres some thing that shows voice extends pass that range? |
| 10:21.04 | slamb | my samples are at http://www.slamb.org/insane/ |
| 10:21.23 | Ridgeback | i have never heard,even a high performance opera singer can barely hit 15khz. |
| 10:21.30 | coppice | Ridgeback: you have a weird attitude to compression. it is a good thing, not a bad thing. for any chosen bit rate it lets you get much better results. |
| 10:21.52 | slamb | Ridgeback: your sampling rate should be twice the frequency you're trying to represent. hold on, let me find a wikipedia article for you. |
| 10:22.02 | Ridgeback | coppice, im not bashing compression, hell i use g729. |
| 10:22.29 | slamb | http://en.wikipedia.org/wiki/Nyquist-Shannon_sampling_theorem |
| 10:22.47 | coppice | Ridgeback: there is plenty of material on the internet about this, if you search. lots of examples to hear, too. I used to work in voice compression (e.g half rate GSM, which is a *really* nice codec, that somehow got marginalised). |
| 10:23.10 | coppice | slamb: they forgot Whittaker and the Russian guy :-) |
| 10:23.25 | slamb | coppice: fix it. ;) |
| 10:23.48 | coppice | i can't really. i forgot the russian guy's name :-) |
| 10:23.54 | slamb | Oh, actually, it is in the full article: "It is also known as Whittaker-Nyquist-Kotelnikov-Shannon sampling theorem or just simply the sampling theorem." |
| 10:24.09 | coppice | OK. remembered it again! |
| 10:24.17 | Zeeek | coppice what is the relation between gsm as a codec and gsm phones? The obvious gsm phones use gsm? Gsm was developed for them? |
| 10:24.48 | coppice | very few GSM calls use the thing asterisk users call the GSM codec. |
| 10:25.05 | coppice | 06.10 is the proper name for that codec. it was the original GSM codec |
| 10:25.16 | *** join/#asterisk proti (~seb@calypso.frankengul.org) |
| 10:25.35 | proti | Morning |
| 10:26.00 | Ridgeback | slamb, im familiar with Nyquist, but the problem I see is rarely do any of the modern codec conform to his thoeries? I wuld guess they don't |
| 10:26.54 | coppice | then came half-rate and enhanced full rate. those are the commonest in use today. half rate is really good for a 5.95K codec. It scales down well to 4.2K, which is the iDEN codec. EFR is better than 06.10, but for a modern coec its mediocre. AMR is the new generation, common to GSM and UMTS |
| 10:27.18 | coppice | Nyquist didn't have any theories. That was the other three :-) |
| 10:27.24 | dwmw2_gone | Ridgeback: it's not about codecs. It's about sample rates. |
| 10:27.31 | coppice | Nothing can disobey their theory |
| 10:27.59 | slamb | Ridgeback: No codec is going to do better than uncompressed audio at the same sampling rate. Only a lossless codec can even do as well. |
| 10:28.01 | Ridgeback | right, what codec offer sampling rates for the quality we all want? |
| 10:28.10 | proti | I would like to know if asterisk is able to register automatically after a Internet connection outage to the VOIP provider. |
| 10:28.15 | coppice | Nyquist understood sampling intuitively, but couldn't rationalise a theory around it. the other 3 did |
| 10:28.23 | slamb | Ridgeback: As far as I can tell, speex is the only one that offers the 32 kHz sampling rate I want. |
| 10:28.38 | coppice | Ridgeback: skype uses a reasonable balance |
| 10:28.38 | Ridgeback | slamb, but no phone offers speex right ? |
| 10:28.56 | slamb | Ridgeback: Not sure yet. I've only looked at softphones so far. |
| 10:29.09 | jeffik | Ridgeback/slamb: is there any need for gsm? |
| 10:29.43 | coppice | slamb: more importantly, speex is one of the few scalable codecs. the extra bandwidth is easily stripped away on route if need be (such as PSTN interworking |
| 10:29.57 | slamb | jeffik: What do you mean by need for? |
| 10:30.01 | slamb | coppice: cool |
| 10:30.06 | Ridgeback | i would reallyl ike to have speex on my polycoms..... |
| 10:30.47 | jeffik | well is there an advantage to using gsm codec, i use x-lite and hae it turned off, only using ulaw |
| 10:30.50 | jeffik | 711 |
| 10:31.00 | coppice | well, when the marketing droid tell you nobody has asked for something, they are often being honest. ask |
| 10:31.35 | coppice | jffik: 06.10 is popular with free stuff specifically because it is unencumbered |
| 10:31.38 | Ridgeback | dammit, does voipjet block 1-800 toll free numbers? |
| 10:31.50 | coppice | it is also one of the lest compute heavy codecs |
| 10:31.55 | slamb | jeffik: It's low-bandwidth and commonly available. It's 8 kHz only, so I don't like the quality. |
| 10:33.48 | tzafrir_laptop | Say, what exactly are the gc flags -O4 and -O6 used in the makefile of zaptel? Do they actually do something different than -O2? |
| 10:34.26 | *** join/#asterisk nemisus (~nemisus@203-217-80-42.dyn.iinet.net.au) |
| 10:34.47 | slamb | coppice: good point about asking. I should ask my voip provider (sixtel) if there's any possibility of speex support. I want to be using this codec all the way through. |
| 10:35.25 | *** join/#asterisk newbien (~e@147.241.33.65.cfl.res.rr.com) |
| 10:36.33 | Ridgeback | check this out! http://focus.ti.com/pdfs/vf/bband/dsl_lbr.pdf |
| 10:37.15 | jeffik | thaks for explainatoin |
| 10:37.27 | gres | Hi all. I Can't find official or unofficial asterisk forum. Can anybody help me? |
| 10:37.38 | Ridgeback | apparently the the mean opion score shows shows g711 is the best quality (subjectively) of course. but all the codecs were very close, i guess it shows codec to codec, most people dont really care |
| 10:37.50 | Ridgeback | except for us geeks |
| 10:37.52 | Ridgeback | ;) |
| 10:38.02 | gres | Where i can find logs for this channel? |
| 10:39.31 | tzafrir_laptop | gres, this is an unofficial asterisk forum |
| 10:40.19 | tzafrir_laptop | Also: |
| 10:40.21 | tzafrir_laptop | ~docs |
| 10:40.22 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 10:40.39 | tzafrir_laptop | I wonder: |
| 10:40.41 | tzafrir_laptop | ~list |
| 10:40.42 | jbot | one warez list being sent |
| 10:40.56 | gres | I have unusial trouble... |
| 10:40.56 | tzafrir_laptop | ~mailing-list |
| 10:40.57 | jbot | hmm... mailing-list is see http://www.debian.org/MailingLists/subscribe |
| 10:41.07 | slamb | Ridgeback: G.711 = ulaw, I think. It's the best at that sampling rate. I assert that 32 kHz or 16 kHz speex would have won, had it been in the comparison. |
| 10:41.12 | Ridgeback | g711 wins again :) http://www.sevana.fi/audio_speech_codecs_quality_analysis.html |
| 10:41.28 | Ridgeback | slamb, that would be cool to add those inthe mix, i agree |
| 10:41.32 | tzafrir_laptop | who has been poisning poor jbot with debian stuff!? |
| 10:41.41 | gres | I ask question on this channel, but nobody help me. :( |
| 10:41.52 | Ridgeback | whats wrong gres |
| 10:42.02 | gres | mmm.. |
| 10:42.12 | gres | Hi all. I have * with te100p. I have extention for reception, where people, ringing from pstn phone, input local number. Most pstn phone work well. But when i call from pstn phone panasonic 2363 and iput local nuber, * do not see anything. |
| 10:42.46 | gres | how can i debug dtmf? |
| 10:42.51 | coppice | Ridgeback: if you compare G.711 with things worse than G.711 it tends to win :-) |
| 10:43.04 | Ridgeback | gres, asterisk -rvvvvvvvvv |
| 10:43.14 | Ridgeback | coppice, true |
| 10:43.37 | Ridgeback | coppice, in the end though it is really subjective though |
| 10:43.51 | gres | No, i have not seen anything interesting in console. |
| 10:43.59 | Ridgeback | hmmm |
| 10:44.22 | coppice | Ridgeback: not in this case. its purely the result of restricting the test to 8kHz sampling codecs. nothing more |
| 10:44.52 | Ridgeback | coppice, the mean score was based on human listeners, so again it would be subjective |
| 10:45.13 | gres | In that office many other phone and asterisk understand them. |
| 10:45.29 | coppice | The 8kHz limit is so huge, it swamps any subjective issues in this case |
| 10:45.59 | Zeeek | gres maybe there is no audio at all reaching asterisk? |
| 10:46.26 | Zeeek | or not enough |
| 10:46.48 | coppice | practically any codec sampling at 16kHz will thrash G.711 in listener tests |
| 10:46.54 | Ridgeback | i belive the limit to 8kbit codec was due to several factors at the time of the test. we dont know perhaps they only had a need for 8 bit codecs, or ther was no hardware that supported it, or no software setup for it. also industry tends to ignore the free stuff like speex |
| 10:47.06 | gres | Bat everything work well from other phones!! |
| 10:47.22 | gres | but :) |
| 10:47.26 | coppice | what's with this 8bit codecs thing? what do you mean by that? |
| 10:47.34 | Ridgeback | coppice, ok okay perhaps they do. but how much bandwidth do they use? and whose ears will say it's better? |
| 10:48.04 | Ridgeback | coppice, you stated earlier about the test limited to 8khz codecs |
| 10:48.39 | coppice | G.722 is a really crappy codec, originally designed to replace G.711. It runs at the same bit rate. Crappy or not, it trashes G.711 |
| 10:48.50 | coppice | what have 8kHz got to do with 8bit? |
| 10:50.26 | Ridgeback | coppice, i think you maybe confusing yourself with 8 bit and 8khz stuff. i may have type 8 bit earlier, but meant 8 khz, in repsone to your post. i could paste your message again if needed be |
| 10:51.42 | Ridgeback | so how come if there are all these kickass codec, no one implements them? is it money, bandwidth, architecture or what? |
| 10:52.23 | Ridgeback | my expesive polycom only supports g729, ulaw and some other codec, i cant rememebr.. i touwl be nice to expand a bit dont you think? |
| 10:52.37 | coppice | its the 8kHz lock in thing. the PSTN is locked into that for various technical reasons. VoIP is locked into it through short-sightednes |
| 10:52.43 | *** join/#asterisk Sn0p (~aggrw@cm-84.118.197.101.chello.no) |
| 10:53.10 | Ridgeback | hmmm |
| 10:53.15 | coppice | if people don't break the mould, skype will do exceedingly well |
| 10:53.25 | Ridgeback | perhaps with asterisk we can change that? |
| 10:53.40 | slamb | Ridgeback: here's hoping. |
| 10:53.53 | coppice | but * needs fixing. 8kHz is embedded all over the place :-) |
| 10:53.58 | Ridgeback | hey did you guys see Mark Spencer from Digium on Call for Help today!! was cool! |
| 10:54.02 | Ridgeback | bugger |
| 10:57.24 | slamb | coppice: What's all over the place? I see that the file loading stuff - like format_wav - assumes 8k. I also see the stuff like IAX_RATE_xxKHZ. Is that not used yet? |
| 10:58.26 | coppice | the SDP stuff is hard coded to say 8000. there is no rate negotiation when choosing codecs. lots of little bits in lots of places need fixing |
| 10:58.27 | Ridgeback | so what about sRTP or IAX encryption? i saw on the dev list a loooong time ago one could do encrypt=yes in iax.conf has anyone tried this stuff yet? |
| 10:58.52 | coppice | Ridgeback: it isn't implemented yet |
| 10:59.00 | Ridgeback | crap |
| 10:59.14 | coppice | sRTP isn't in, and encryption for IAX has not yet been designed |
| 10:59.27 | Ridgeback | would like that stuff working.... |
| 10:59.40 | slamb | coppice: Does it just need an implementation, or is there no protocol spec for rate negotiation either? |
| 10:59.59 | Ridgeback | coppice, really? I'll have to dig up that message, i remember him saying to give it a try encrypt=yes... |
| 11:00.41 | Ridgeback | http://lists.digium.com/pipermail/asterisk-security/2005-February/000031.html |
| 11:00.52 | Ridgeback | this guy ran iax crypto... i wonder if it actually worked |
| 11:01.19 | Ridgeback | http://lists.digium.com/pipermail/asterisk-security/2005-January/000011.html |
| 11:01.20 | *** join/#asterisk dieck (~dieck@saengerschafter.de) |
| 11:01.45 | Ridgeback | here Mark did build some code for iax crypto, dont know if it works though |
| 11:02.27 | dieck | hi there, I got a little problem with an Asterisk box calling another asterisk box(2) via IAX2, and (2) has a registered sip phone (grandstream bt101). (2) produces a lot of: chan_sip.c:1829 sip_write: Asked to transmit frame type 512, while native formats is 4 (read/write = 4/4) while transferring the iax to sip call |
| 11:02.35 | dieck | I googled but I cant |
| 11:02.46 | dieck | ups, but I can't figure out how to solve that error |
| 11:03.03 | Ridgeback | dieck, are you using the cvs version? |
| 11:03.18 | dieck | Ridgeback: i think it's the stable version on both sides |
| 11:03.51 | dieck | run via asterisk and console via asterisk -vvvrdc |
| 11:03.53 | Ridgeback | dieck, hmmm... i use the cvs version in 7 switches here and similar configs as yours, no problems at all. |
| 11:04.32 | dieck | maybe it's not an error than but kind of a debug message that says asterisk is recoding the signal from type 512 to native 4? |
| 11:05.00 | *** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
| 11:05.22 | dieck | hm, no, should be a warning: in chan_sip.c: ast_log(LOG_WARNING, "Asked to transmit frame type %d, while native formats is %d (read/write = %d/%d)\n", |
| 11:09.33 | coppice | slamb: what protocol spec would be needed? |
| 11:11.47 | slamb | coppice: to negotiate rates? I haven't looked at IAX2, but it seems like ideally it'd say "I can speak speex at these rates, GSM at this rate, what do you want?" The codecs and rates seem to be tied together. |
| 11:13.56 | *** join/#asterisk kryme (~PK@63.151.31.248) |
| 11:15.10 | *** join/#asterisk coppice (~chatzilla@153.166.17.210.dyn.pacific.net.hk) |
| 11:17.18 | slamb | coppice: Actually, now, I am looking at the Internet draft. It looks like the sampling rate is never sent at all. |
| 11:17.44 | coppice | slamb: which spec? |
| 11:17.52 | slamb | http://www.cornfed.com/iax.pdf |
| 11:19.43 | *** join/#asterisk oden (~oden@194-237-146-22.customer.telia.com) |
| 11:20.46 | slamb | oh, I do see the IAX_IE_SAMPLINGRATE stuff in iax-parser.c, though. |
| 11:20.52 | coppice | slamb: I think it doesn't send the rate, but it sends the codec type, which infers the rate. it means everyone down the line probably has to have a translation table, which is not ideal |
| 11:22.51 | slamb | but what if the codec supports multiple rates? Say I want to prefer iLBC if they support it at 16 kHz (which I believe requires a license), but I want to prefer speex if they only support 8 kHz iLBC. |
| 11:24.38 | slamb | Maybe I'm wrong about iLBC, but hopefully you see what I'm getting at. |
| 11:24.43 | coppice | different rates usually means different codecs, so it isn't a probably. because speex has low bandwidth nested within high bandwidth it might not fit that model |
| 11:27.55 | newbien | anyone get linphone 1.0.x to work with ast* ? |
| 11:30.26 | newbien | whats the worst and best hardphones? |
| 11:31.40 | coppice | anything below 80 cents is pretty near the worst :-) |
| 11:34.19 | *** join/#asterisk fafnir (~hello@tdds-gw.Moscow.gldn.net) |
| 11:35.00 | *** join/#asterisk Gh0sty (~Ghosty@ip-81-11-198-207.dsl.scarlet.be) |
| 11:35.05 | newbien | any hardphone that is a prize in a breakfast cereal box is probably junk ;) |
| 11:36.53 | coppice | I think they would have to be lower than 80 cents for that. A typical desktop phone is about $1 FOB, but its pretty hard to strip a low end phone much lower than the average |
| 11:41.27 | newbien | are the analog to digital phone converters reliable? |
| 11:41.27 | slamb | coppice: time to sleep. thanks for the info. |
| 11:42.00 | fafnir | newbien: what are you relating to? |
| 11:43.31 | newbien | fafnir: black boxes that allow you to use your analog phone as a voip phone |
| 11:43.57 | Zeeek | newbien all depends on exactly what usage you intend |
| 11:44.12 | Zeeek | office, home, hobby? |
| 11:44.14 | fafnir | what cereal box? |
| 11:44.17 | fafnir | and protocol? |
| 11:44.19 | Zeeek | Wheaties |
| 11:44.30 | *** join/#asterisk Blackvel (~blackvel@dsl-084-057-124-186.arcor-ip.net) |
| 11:44.36 | fafnir | iax? sip? |
| 11:44.42 | Blackvel | what is a good proggy to sample a .wav file down to 16bit? |
| 11:44.44 | Zeeek | trix |
| 11:44.53 | dieck | Blackvel: man sox |
| 11:44.55 | Blackvel | May 14 13:14:03 WARNING[31201]: format_wav.c:169 check_header: Can only handle 16bits per sample: 1 |
| 11:45.21 | Blackvel | ah cool |
| 11:45.50 | dieck | Blackvel: sox is for sound what "convert" is for graphics: just choose input and output and let sox do the magic :) |
| 11:47.01 | Blackvel | is it even a good idea to play .wav files with GET DATA (AGI)? or should the file be .gsm? |
| 11:47.01 | Blackvel | dieck: sometimes I just know too less about linux and proggies :) |
| 11:47.22 | Zeeek | anyone tried voipbuster with * ? |
| 11:47.48 | dieck | Blackvel: depends |
| 11:48.16 | Blackvel | how is BV today with asterisk? |
| 11:48.27 | Blackvel | tried yesterday a call to UK and sound quality with that bad |
| 11:48.33 | dieck | Blackvel: if the call is in gsm, having the sounds to play in gsm need no further convertion. If the call is not in gsm, the sound has to be converted, if it's wav or gsm does not matter |
| 11:48.58 | Blackvel | dieck: ulaw/alaw |
| 11:49.51 | dieck | Blackvel: convert the sounds to the format you expect the most calls in |
| 11:50.20 | *** join/#asterisk TheEmperor (TheEmperor@218.111.51.8) |
| 11:50.37 | Blackvel | ulaw |
| 11:50.39 | in-side | hi |
| 11:51.10 | Blackvel | just want to get around what can only handle 16bits msg |
| 11:51.10 | Blackvel | :) |
| 11:51.26 | Blackvel | just want to get around the "can only handle 16bits msg" |
| 11:51.30 | TheEmperor | anyone know how i can reduce the load on my asterisk server? |
| 11:51.37 | Blackvel | load? |
| 11:51.44 | coppice | switch it off |
| 11:52.00 | TheEmperor | asterisk is eating up 97% of the cpu |
| 11:52.12 | Blackvel | coppice: check codec conversion |
| 11:52.25 | Blackvel | how many users, what cpu? |
| 11:52.39 | in-side | does asterisk is supported under mac ? |
| 11:52.42 | shido6 | thats a lot |
| 11:52.48 | shido6 | in-side you can run it on a mac |
| 11:52.53 | TheEmperor | pentium 2.4 ghz, 512mb ram, 5 users |
| 11:52.53 | coppice | what is "check codec conversion" supposed to mean? |
| 11:53.10 | Blackvel | coppice: if asterisk has to do codec conversion for calls |
| 11:53.14 | in-side | ho w can i turn off codec conversion under asterisk ? |
| 11:53.27 | shido6 | dont have it go in one codec and out another |
| 11:53.38 | in-side | I don't want asterisk to convert anything just pass it trought always |
| 11:53.52 | shido6 | then force it to come in one codec |
| 11:53.56 | shido6 | and out the same |
| 11:53.58 | TheEmperor | perhaps it's safe_asterisk? |
| 11:54.09 | TheEmperor | if i run it another way, would that be better? |
| 11:54.19 | in-side | shido6: how is that? |
| 11:54.20 | shido6 | safe_asterisk should run if you want it to respawn if it dies |
| 11:54.22 | dieck | if I call a grandstream BT101 via Sip it always shows "t ri" in the display - how can I set a number there? |
| 11:54.35 | Blackvel | dieck: what is the "magic option" for sox to convert a .wav into 16bit? |
| 11:54.42 | TheEmperor | shido6:but the load on my * server is 97% just from asterisk.. |
| 11:54.51 | shido6 | what the heck are you running it on |
| 11:54.57 | shido6 | and how many active channels? |
| 11:55.09 | shido6 | I've never see it go past 40% |
| 11:55.17 | shido6 | but I have prescotts and xeons |
| 11:55.17 | dieck | Blackvel: man sox |
| 11:55.31 | TheEmperor | right now there are 0 active channels |
| 11:55.36 | in-side | TheEmperor: wich operating system and wich version of asterisk ? |
| 11:55.45 | shido6 | are you running it on a PII? |
| 11:55.50 | TheEmperor | redhat 9 and asterisk cvs head |
| 11:55.54 | TheEmperor | p4 |
| 11:55.57 | shido6 | err |
| 11:56.01 | in-side | damn man.. |
| 11:56.02 | shido6 | you've got problems |
| 11:56.12 | shido6 | p4 celeron? |
| 11:56.13 | in-side | why are using a cvs head? |
| 11:56.27 | in-side | why you don't use a stable version instead ? |
| 11:56.36 | TheEmperor | sorry, i mean't cvs stable |
| 11:56.42 | shido6 | i use head |
| 11:56.43 | in-side | ah ok.. |
| 11:56.53 | TheEmperor | no its not a pentium celeron |
| 11:56.58 | in-side | TheEmperor: that seems to much man |
| 11:57.02 | TheEmperor | redhat no good? |
| 11:57.02 | in-side | I have a amd 2400+ |
| 11:57.08 | TheEmperor | would fedora be better? |
| 11:57.09 | shido6 | redhat is good |
| 11:57.12 | shido6 | fedora would be worse |
| 11:57.13 | in-side | ant it won t pass 11% |
| 11:57.23 | TheEmperor | so what should i do? |
| 11:57.28 | in-side | TheEmperor: I can't speak about linux... |
| 11:57.37 | shido6 | what else is runnin on that box? |
| 11:57.39 | in-side | TheEmperor: I just hate then all |
| 11:57.48 | TheEmperor | sendmail |
| 11:57.52 | TheEmperor | mpg123 |
| 11:57.57 | TheEmperor | snort |
| 11:58.24 | in-side | it is using 97% all the times? |
| 11:58.39 | TheEmperor | pretty much.. and it's asterisk that's using 97% |
| 11:58.41 | Zeeek | FWIW, I run a PIII-800 and while calling the directory app, I see 2% CPU |
| 11:58.50 | TheEmperor | that's what i don't understand |
| 11:59.01 | in-side | TheEmperor: try to bind it to only the interface you want |
| 11:59.12 | in-side | too see if it changes anything |
| 11:59.20 | TheEmperor | how do i do that? |
| 11:59.54 | in-side | whell I 'm not the best person to help you but try to see in sip.conf |
| 11:59.58 | in-side | in listen lines |
| 12:00.04 | in-side | of your configuration |
| 12:00.12 | TheEmperor | ok |
| 12:00.34 | in-side | to to try to catch out the problem |
| 12:00.44 | Zeeek | TheEmperor bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) |
| 12:00.49 | in-side | try to disable any modules or stuff your really not want to |
| 12:00.50 | Zeeek | change that to the address |
| 12:01.05 | in-side | yes it is bindaddr sorry |
| 12:01.10 | TheEmperor | ok |
| 12:01.11 | in-side | I was confusing it with ser |
| 12:02.18 | TheEmperor | bindaddr is already at 0.0.0.0 |
| 12:02.31 | Zeeek | do you have more than one eth card? |
| 12:02.35 | TheEmperor | only one |
| 12:02.42 | TheEmperor | load average is 1.00 |
| 12:02.49 | Zeeek | in that case, it won't matter- check what modules are loading |
| 12:03.05 | Zeeek | stop astersik and see if that changes to 0 ? |
| 12:03.11 | Zeeek | astersick |
| 12:03.19 | Zeeek | join #astersick |
| 12:05.38 | Blackvel | dieck: is there any website for sox? I fail to find the options for resampling to 16bit |
| 12:05.42 | *** join/#asterisk Shoragan (~shoragan@d072.apm.etc.tu-bs.de) |
| 12:07.40 | Zeeek | data size The precision the data is stored in. Most popular are 8-bit bytes or |
| 12:07.40 | Zeeek | <PROTECTED> |
| 12:08.57 | Zeeek | Blackvel : data size The precision the data is stored in. Most popular are 8-bit bytes orso, |
| 12:09.06 | Zeeek | oops |
| 12:09.28 | Zeeek | so... look at the flags -b/-w/-l/-d |
| 12:09.39 | Blackvel | thanks, will try |
| 12:09.48 | Zeeek | don't try, just look and read it |
| 12:10.10 | Blackvel | too stupid |
| 12:10.17 | Blackvel | therefore I have to say I try |
| 12:10.20 | Blackvel | :) |
| 12:10.28 | Zeeek | no, not stupid, lazy brain |
| 12:10.59 | Blackvel | maybe it is English which confuses me too |
| 12:11.05 | Zeeek | what country are you in so I can pull out some stupid nationalist jokes? |
| 12:11.11 | Blackvel | haha |
| 12:11.12 | Blackvel | plz not |
| 12:11.17 | Blackvel | being a German is enough |
| 12:11.20 | Zeeek | Poland? Belgium? |
| 12:11.23 | Zeeek | ah, German |
| 12:11.30 | Zeeek | let me see.... |
| 12:11.40 | Zeeek | How many Germans does it take... |
| 12:11.56 | Zeeek | to make a Strudel ? |
| 12:12.04 | Blackvel | haha |
| 12:12.07 | Blackvel | hmm tell me :) |
| 12:12.21 | Zeeek | none. Strudel is Austrian |
| 12:12.58 | Blackvel | I cant laugh... |
| 12:13.05 | dieck | Blackvel: ich bin sicher die sox-manpage gibt's irgendwo auch auf deutsch |
| 12:13.06 | Blackvel | too tired from coding and fixing java bug |
| 12:13.10 | Zeeek | Blackvel anyway, you have the magic answer now, sox -w blah |
| 12:13.11 | Blackvel | s |
| 12:13.24 | dieck | Blackvel: google may help :) |
| 12:13.50 | Blackvel | right, my lazy brain has now to find out how to specifiy that format option for the outfile. I am still confused by man sox :) tried google either. had no luck |
| 12:13.59 | Blackvel | you can not expect that from a java coder |
| 12:14.03 | Blackvel | :) |
| 12:14.05 | Zeeek | Here ist ein good formulae : |
| 12:14.38 | Zeeek | sox eingangfilename -w -r 8000 -c 1 ausgangfilename |
| 12:14.48 | coppice | I barely expect the ability to add from a Java programmer (ducks) :-) |
| 12:14.54 | Zeeek | ich tinke que ça peut marcher une fois |
| 12:15.11 | Zeeek | inchallah |
| 12:16.01 | Zeeek | Blackvel mein German sucks, but you could try that line |
| 12:16.56 | Blackvel | zeeek: better than my english :) |
| 12:17.17 | Zeeek | if it works, put it in a file called "wav16": |
| 12:17.29 | Blackvel | coppice: to add from a java programmer? |
| 12:17.30 | Zeeek | sox $1 -w -r 8000 -c 1 $2 |
| 12:17.32 | Blackvel | what does that mean? |
| 12:17.48 | Zeeek | chmod +x wav16 |
| 12:18.05 | Zeeek | then you type ./wav16 infile.wav outfile.wav |
| 12:18.21 | Zeeek | and never again wonder |
| 12:18.55 | Blackvel | seems to work |
| 12:18.58 | Zeeek | or should that be wunder |
| 12:19.03 | Blackvel | I can not believe how easy it is |
| 12:19.08 | Zeeek | what d'you mean, seems! |
| 12:19.12 | Blackvel | seems man confused me too much |
| 12:19.29 | Zeeek | man was written to confuse |
| 12:19.37 | Zeeek | that's the whole point of man |
| 12:19.44 | Zeeek | man is corrupt |
| 12:19.58 | Blackvel | really, most times I have problems to find the correct arguments for commands |
| 12:20.07 | Zeeek | man was after all created in the image of man |
| 12:20.55 | *** join/#asterisk meppl (mephisto@p54AAE973.dip.t-dialin.net) |
| 12:21.44 | Zeeek | How many Java programmers does it take to... |
| 12:21.56 | Zeeek | change a lightbulb |
| 12:22.16 | Blackvel | what is a lightbulb? |
| 12:22.22 | Zeeek | indeed |
| 12:22.49 | Zeeek | das Bilt |
| 12:23.01 | Zeeek | das Mädchen |
| 12:23.13 | Zeeek | ich habbe kein camera |
| 12:23.14 | coppice | a lightbulb is an object, so a java programmer will only change it after massive documentation updates |
| 12:23.23 | Zeeek | ~lightbulb |
| 12:23.26 | jbot | Web-based tool for organising your list of things to do.. URL: http://www.alexium.com/lightbulb/ |
| 12:24.02 | *** join/#asterisk gambolputty (~gambolput@cblmdm69-45-216-83.buckeye-express.com) |
| 12:24.21 | Zeeek | surely there is already an inherited change method? |
| 12:25.03 | Blackvel | where is the difference between java and any oo oriented language then? |
| 12:25.08 | Blackvel | you are kidding after java, but it also hits c++ |
| 12:25.19 | Blackvel | about java |
| 12:25.31 | Blackvel | in fact any language which is not lazy c |
| 12:25.32 | Blackvel | :) |
| 12:25.40 | Zeeek | all of those languages suffer from the same probelm. They are no good for old people like me who can't remember the long names |
| 12:25.51 | Blackvel | hehe |
| 12:25.58 | Blackvel | true |
| 12:26.05 | Blackvel | but I am young |
| 12:26.09 | shido6 | I HATE HTML |
| 12:26.11 | Zeeek | then it's ok |
| 12:26.18 | Blackvel | I don't want to remember pointers |
| 12:26.20 | Blackvel | :) |
| 12:26.22 | *** join/#asterisk prh (~paul@212.13.203.80) |
| 12:26.25 | Zeeek | I hate BAD HTML |
| 12:26.36 | coppice | what we need is a genuine C++. an true incremental move beyond C |
| 12:26.37 | Blackvel | I don't like html programming neither |
| 12:29.22 | *** join/#asterisk prh (~paul@212.13.203.80) |
| 12:33.59 | tzanger | I forget, who's the company that makes good (but expensive) telephony products for the office... they have a 4-port FXS and a 24-port FXS-SIP product as well |
| 12:34.10 | tzanger | it's not a channel bank |
| 12:34.36 | tzanger | mediatrix |
| 12:34.37 | tzanger | there we go |
| 12:36.03 | shido6 | Shido-TroniX |
| 12:36.30 | gbdrbobAFK | what options can be passed to the monitor app from within queues.conf - is it only the format or can I usee all the options such as filename amd "m" for mixng |
| 12:48.01 | jontow | has anyone done any work with the channel API? |
| 12:48.09 | jontow | enough to get any documentation, at least? |
| 12:56.35 | *** join/#asterisk taspegao (jorgeraide@c-69-180-117-144.hsd1.fl.comcast.net) |
| 12:56.40 | taspegao | hello |
| 12:56.45 | taspegao | alguien me peude ayudar? |
| 12:57.09 | coppice | working out how a simple channel works is simple - look at a simple channel driver. working out the whole picture for something like chan_zap is a whole different matter |
| 12:58.42 | taspegao | hello |
| 12:59.50 | taspegao | i have been with some sipuras and my asterisk |
| 13:00.01 | taspegao | i have been problems |
| 13:00.47 | taspegao | works over minutes fine but after 5 minutes only can call but can't receive calls |
| 13:00.55 | taspegao | is strange |
| 13:01.09 | gbdrbobAFK | hello, who knows anything about the monitor app? |
| 13:01.17 | jontow | im not aiming for chan_zap |
| 13:01.27 | gbdrbobAFK | does the b option work reliably with call queues? |
| 13:01.27 | jontow | that one.. is way over my head for now (and quite a while in the future) |
| 13:03.28 | Zeeek | taspegao see if the phone is still registered |
| 13:03.35 | taspegao | yes |
| 13:03.47 | taspegao | is stranger |
| 13:04.22 | taspegao | if i turn off the power and conect again works fine by 5 minutes more |
| 13:04.25 | taspegao | jejej strange |
| 13:04.43 | taspegao | i have 5 sipuras with the same problem |
| 13:05.10 | jontow | using qualify=... in sip.conf? |
| 13:05.53 | taspegao | qualify=no |
| 13:06.05 | jontow | ok.. eh, why not? :) |
| 13:06.37 | taspegao | well i can to do teh test with qualify |
| 13:06.47 | meppl | guten morgen |
| 13:06.57 | taspegao | but i have other sipuras with the same cfg and works fine |
| 13:07.08 | taspegao | but i am going to do teh test now |
| 13:07.12 | taspegao | if the posibility |
| 13:07.13 | Zeeek | taspegao you can maybe shorten the register period on the sipuras as well |
| 13:07.37 | Zeeek | taspegao are all the phones in the same place? |
| 13:07.45 | taspegao | no |
| 13:07.52 | taspegao | but |
| 13:08.01 | Zeeek | are they all in the same NAT/firewall situation? |
| 13:08.28 | jontow | ah, if you have others in that configuration.. it does sound more likely to be unique to that one |
| 13:08.29 | Zeeek | are any on fixed ip? |
| 13:08.37 | taspegao | when i have this kind the problem i do the test here or other palces and nothing |
| 13:08.52 | taspegao | yes i know that |
| 13:09.00 | Zeeek | you have to find the difference between the phones' situation. If none, you have a bad phone |
| 13:09.05 | taspegao | but is not the router |
| 13:09.29 | Zeeek | same firmware versions? |
| 13:09.40 | taspegao | listen the sipura have the tone alltime |
| 13:10.06 | Zeeek | can you answer some of the questions? |
| 13:10.18 | taspegao | only is five minutes more a less it can to call but it can't to receive |
| 13:10.42 | taspegao | when i see the call over asterisk |
| 13:10.53 | taspegao | the call is fine |
| 13:11.53 | taspegao | yes |
| 13:11.57 | taspegao | i did the update |
| 13:12.17 | taspegao | all sipuras have the same firware |
| 13:12.30 | Zeeek | what is the difference between this phone and the other phones? Are they all behind the same kind of router? |
| 13:12.38 | Zeeek | are they all on dynamic ip? |
| 13:12.51 | Zeeek | have you tried another phone from the same location? |
| 13:13.04 | Zeeek | have you tried that phone somewhere else? |
| 13:13.08 | taspegao | phone |
| 13:13.23 | taspegao | uhm |
| 13:13.37 | Zeeek | take a deep breath |
| 13:13.42 | Zeeek | count to 10 |
| 13:13.44 | taspegao | jeje |
| 13:13.48 | taspegao | yes i need it |
| 13:14.02 | Zeeek | then try to explain what the fuck you have already tried |
| 13:14.12 | Zeeek | because this is like pulling teeth |
| 13:14.52 | taspegao | men i have the same cfg in all my sipuras extension |
| 13:15.04 | taspegao | and some works fine and other have this problem |
| 13:15.12 | taspegao | in the same place |
| 13:15.28 | taspegao | with thge same router |
| 13:15.29 | taspegao | the |
| 13:15.47 | *** join/#asterisk zyke (~zakforeve@84.45.132.117) |
| 13:15.48 | jontow | something is different |
| 13:15.50 | taspegao | now i have one that have the problem |
| 13:15.59 | jontow | no matter how many times you assume everything to be the same |
| 13:16.24 | jontow | whether its a buggy device or not; thats potentially a question.. but i would ride more on SOME random small bit being very much different between a working vs. non-working phone |
| 13:16.46 | Zeeek | is it possible to try a "bad" phone in another location or to try another phone that works elsewhere in your location? |
| 13:17.00 | Zeeek | you have to find out whether it is network related or phone related |
| 13:17.08 | taspegao | ok |
| 13:17.14 | Zeeek | you understand? |
| 13:17.14 | jontow | (does it do the same from that location with a soft-phone of comparable function?) |
| 13:17.25 | taspegao | yes |
| 13:17.25 | Zeeek | good point and cheap to test :) |
| 13:17.30 | zyke | hi guys .. do i need to load res_config_odbc.so in the modules.conf to enable asterisk realtime? |
| 13:17.32 | taspegao | je ok |
| 13:17.37 | jontow | ;) |
| 13:17.58 | Zeeek | taspegao so you will now try a softphone? |
| 13:18.15 | taspegao | yes |
| 13:18.20 | Zeeek | good! |
| 13:18.27 | Zeeek | we're coming along |
| 13:18.40 | taspegao | you know what |
| 13:18.47 | Zeeek | softphones are great for that |
| 13:18.56 | taspegao | yesterday i did the test the x-pro over dial-up and fuck |
| 13:18.58 | taspegao | wow |
| 13:19.00 | taspegao | bad |
| 13:19.07 | Zeeek | impressive, huh? |
| 13:19.26 | taspegao | but some friend tall me that i need tu used the fireflay with iax |
| 13:19.33 | taspegao | firefly |
| 13:19.34 | taspegao | sorry |
| 13:19.47 | shido6 | iaxcom works |
| 13:19.54 | taspegao | my english is very...very...baddd |
| 13:19.54 | Zeeek | iaxphone too |
| 13:19.55 | taspegao | jeje |
| 13:20.17 | taspegao | well i hopre so that works fine for dial-up |
| 13:20.27 | Zeeek | but you need to do a test with a siphone |
| 13:20.33 | jontow | dialup indeed |
| 13:20.38 | taspegao | i did |
| 13:20.54 | taspegao | with sip and g729 |
| 13:21.00 | jontow | dialup is bad stuff; i've been messing with an IAX2 link over dialup to a T1 .. using ilbc, gsm, and speex |
| 13:21.10 | jontow | but its only a 26400bps dialup ;) |
| 13:21.11 | taspegao | aja |
| 13:21.21 | taspegao | yes i am going to do teh test |
| 13:21.30 | jontow | i can hear pre-recorded audio; but any spoken word lags or doesn't ever get across the link |
| 13:21.41 | jontow | and its scratchy as hell no matter what you do |
| 13:22.00 | taspegao | jeje but i want to send the softphone to one friend that only have dial-up |
| 13:22.09 | jontow | i say.. good luck getting it to work |
| 13:22.15 | taspegao | jejeje |
| 13:22.23 | taspegao | ok |
| 13:22.33 | jontow | i haven't seen it possible to make a real call oer it |
| 13:22.41 | jontow | over |
| 13:23.00 | taspegao | well my my friend is living in cuba |
| 13:23.11 | taspegao | jejejeonly dial-up |
| 13:23.13 | taspegao | only |
| 13:23.16 | syle | cuba woohoo |
| 13:23.20 | taspegao | and bad dial-up |
| 13:23.21 | syle | lets go smoke some crack |
| 13:23.31 | taspegao | 33kb/sec |
| 13:23.38 | taspegao | jejjee |
| 13:23.40 | jontow | 33kbit you mean ;) |
| 13:23.56 | jontow | 33600; beats mine like crazy if he actually gets that kind of bandwidth on it |
| 13:23.58 | taspegao | mission imposible |
| 13:23.59 | taspegao | jeje |
| 13:24.29 | zyke | hi guys .. do i need to load res_config_odbc.so in the modules.conf to enable asterisk realtime? |
| 13:24.37 | taspegao | well i think that if i used iax with gsm,ilbc or g729 should works |
| 13:24.53 | taspegao | " " |
| 13:24.58 | taspegao | jeje |
| 13:25.38 | dieck | hm, I found out that I have to forward ports 5060 and 5004 to a sip ip phone behind a dsl router - but is that tcp or udp? haven't found any information about the protocol, just the ports. I'm forwarding both by now, but i want to reduce to the neccessary minimum :) |
| 13:25.45 | Zeeek | taspegao skype will work a lot better |
| 13:26.06 | syle | zyke: http://www.voip-info.org/wiki-Asterisk+RealTime |
| 13:26.19 | Zeeek | 5004 isn't nec to forward |
| 13:26.29 | zyke | syle: it doesn't say .. if you need to load the module in the modules.conf |
| 13:26.30 | Zeeek | 5060 both will do |
| 13:27.32 | taspegao | but skype works with iax? |
| 13:27.35 | taspegao | let me check |
| 13:27.53 | Zeeek | screw iax. You want to talk to someone on dialup, the simplest way is skype |
| 13:27.58 | ariel_ | dieck, udp |
| 13:28.03 | Zeeek | forget "voip" |
| 13:28.18 | taspegao | ok |
| 13:28.37 | Zeeek | worst case, you can buy an accessory to use skype with a phone |
| 13:28.42 | syle | zyke: idk haven;t play with it yet, but i would use mysql not the odbc module |
| 13:28.58 | taspegao | i have the usbphone |
| 13:28.59 | Zeeek | it really is too bad there's no way to hook skype up to our asterisk boxes |
| 13:29.06 | Zeeek | but that's the way it is |
| 13:29.09 | syle | zyke: i been putting it off since i heard you can;t support NAt with realtime for some strange reason |
| 13:29.11 | taspegao | jejee is better for softphone |
| 13:29.11 | taspegao | ;) |
| 13:29.37 | Zeeek | I would never inflict SIP or IAX on a dialup user |
| 13:29.50 | Zeeek | skype shines in that one instance |
| 13:30.16 | Zeeek | and has better conferencing and taxt chat and encryption built in |
| 13:30.38 | Zeeek | no trouble to install |
| 13:31.33 | Mavvie | Zeeek: also has almost nothing you can configure. |
| 13:35.00 | Zeeek | Again, back to the goal: talking to a dialup user. Configure? Works or not. |
| 13:35.15 | *** join/#asterisk drumkilla (~russell@user-24-214-77-225.knology.net) |
| 13:35.15 | *** mode/#asterisk [+o drumkilla] by ChanServ |
| 13:35.33 | Zeeek | X-Lite has a zillion things to configure - you wanna explain that to your non-tech friends? |
| 13:36.00 | Zeeek | I did manage to get a bunch of non-tech to use firefly |
| 13:36.09 | Zeeek | but I had to write a manual with screen shots to do it |
| 13:36.27 | Mavvie | Zeeek: I can't let skype choose my VoIP server. |
| 13:36.27 | Zeeek | screqw voip |
| 13:36.27 | Mavvie | Zeeek: I can't let skype choose my LDAP server. |
| 13:36.33 | shido6 | actually.. i have a manual finished |
| 13:36.35 | Zeeek | I am talking about a result, not the means |
| 13:36.38 | shido6 | it will be online by tomorrow |
| 13:36.48 | Zeeek | shido6 for firefly ? |
| 13:36.49 | shido6 | with screenshots |
| 13:36.51 | shido6 | for xlite |
| 13:36.59 | Zeeek | there are many of those online now |
| 13:37.01 | shido6 | i'll do firefly, too if you want |
| 13:37.17 | Zeeek | besides my manuals contained our server settings |
| 13:37.42 | Zeeek | I guess what you need is a web page where you fill in the server settings then generate the screens! |
| 13:37.56 | Zeeek | great idea! |
| 13:38.26 | Zeeek | then you send the user a link http://blah/heh?u=123345 |
| 13:39.00 | Mavvie | what you need (for a corperate / ISP environment with their own VoIP server) is a "click on a website" which loads the configuration of xlite/firefly/whatever onto the computer. |
| 13:39.01 | Zeeek | and they see: username 12345 password:belch69hahaa server:domaon.com |
| 13:39.17 | Zeeek | Mavvie that's a good one! |
| 13:39.29 | Zeeek | so you'd only need the format of the config file of the client |
| 13:39.40 | Zeeek | I like that idea |
| 13:39.47 | Mavvie | Zeeek: windows registry? |
| 13:39.59 | Zeeek | is X-Lite in reg? |
| 13:40.09 | Zeeek | pretty sure sjphone is not |
| 13:40.14 | Mavvie | I don't know, but I wouldn't be surprised. |
| 13:41.01 | Zeeek | yes it is and with a shitload of ... shit |
| 13:42.58 | Zeeek | so, long story short, it would be possible to slap the minimum info onto the registry to get a new user started with your server as proxy#1 |
| 13:45.16 | *** join/#asterisk NewSole2 (dave@i216-58-44-245.avalonworks.net) |
| 13:47.44 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) |
| 13:51.21 | Mavvie | firefly is also an registry file |
| 13:54.56 | Mavvie | thanks for the idea zeeek :-) |
| 13:55.15 | Zeeek | it was yours |
| 13:55.23 | Zeeek | except for the web page :) |
| 13:55.48 | Zeeek | of course executing the regedit addition from a web page won't be easy |
| 13:56.05 | Zeeek | prolly have to generate an executable or something |
| 14:00.03 | Mavvie | wonder what the correct mime-type for a .reg file is :-) |
| 14:01.04 | [Lamer] | Hi, to use the video mode what settings do I use in *? |
| 14:01.45 | [Lamer] | videosupport=yes and allow=h623 don't seem to work |
| 14:01.54 | Zeeek | Mavvie http://msdn.microsoft.com/workshop/networking/moniker/overview/appendix_a.asp |
| 14:03.52 | [Lamer] | anyone using video mode with SIP phones at all? |
| 14:09.27 | *** join/#asterisk doolph (doolph@200.46.148.35) |
| 14:09.29 | doolph | hello |
| 14:11.20 | Zeeek | helo |
| 14:16.21 | Zeeek | If you're border, here's an interesting article/editorial about voip: http://loxos.blogspot.com/2005/04/business-voip-and-five-9s-problem.html |
| 14:16.34 | Zeeek | bored, not border :) |
| 14:17.58 | Mavvie | Header("Content-type: application/x-msdownload"); |
| 14:17.58 | Mavvie | Header("Content-Disposition: attachment; filename=\"firefly.reg\""); |
| 14:17.59 | Mavvie | readfile("firefly.reg"); |
| 14:18.02 | Mavvie | works like a charm. |
| 14:19.16 | [Lamer] | Zeeek: ever use a SIP phone+ video mode with *? |
| 14:20.06 | [Lamer] | or Mavvie |
| 14:20.26 | Mavvie | [Lamer]: what makes you think I do? |
| 14:21.09 | [Lamer] | Mavvie: it's just a question, sorry to confuse you |
| 14:21.21 | Mavvie | [Lamer]: that's okay. just wondering why you think I do. |
| 14:21.44 | Mavvie | [Lamer]: I mean, you asked the question and I clearly didn't respond to it, must be obvious that I don't know the answer. |
| 14:21.59 | Mavvie | [Lamer]: but still you ask me, against better judgement. |
| 14:22.12 | [Lamer] | Mavvie: I just had my * up and running today and trying to find related articles on video mode problem shootings |
| 14:22.49 | [Lamer] | Mavvie: I decided to ask after shouting out for about 8 or more hrs |
| 14:23.25 | [Lamer] | Mavvie: there must be something wrong with my question so everyone seems to ignore |
| 14:23.45 | Mavvie | maybe it's your body odour... let me smell your armpit. |
| 14:23.47 | Mavvie | nope, it's not that. |
| 14:24.17 | jontow | lamer; i think the problem is, not too many people invest in the equipment to do video-based SIP |
| 14:24.26 | jontow | so there isn't really a lot of open experience with it |
| 14:24.35 | jontow | i know i wouldn't even know where to begin implementing it |
| 14:24.43 | Mavvie | [Lamer]: I give you a hint: asterisk-users |
| 14:24.43 | [Lamer] | jontow: i see |
| 14:25.08 | jontow | actually.. yeah.. Mavvie is right |
| 14:25.09 | [Lamer] | Mavvie: Ok |
| 14:25.14 | jontow | first search the mailing list archives |
| 14:25.20 | jontow | if you can't find the info there; subscribe and post |
| 14:25.52 | [Lamer] | jontow: I'm searching through google now |
| 14:26.07 | jontow | but if you post; make sure you get enough information in it so that people will actually know what you are talking about and trying to do; otherwise its likely to get ignored |
| 14:26.11 | doolph | how can I tune up my voice quality |
| 14:26.14 | [Lamer] | jontow: like I did for * installation today |
| 14:26.21 | doolph | the sound hear like far away |
| 14:26.23 | jontow | the more specific the question; the more specific the answers if they are known ;) |
| 14:26.31 | nemisus | is there a FC3 or RHEL4 rpm for asterisk? |
| 14:26.32 | jontow | doolph; zaptel or native VoIP? |
| 14:26.52 | [Lamer] | nemisus: me? FC3 |
| 14:27.03 | doolph | zaptel |
| 14:27.14 | nemisus | [Lamer] got a url? |
| 14:27.14 | doolph | no |
| 14:27.17 | doolph | i mean |
| 14:27.17 | jontow | doolph; have you played with txgain/rxgain in zapata.conf ? |
| 14:27.29 | doolph | zaptel for pstn, but i am using a sipura like fxs |
| 14:27.45 | jontow | ok; maybe there is an option in the sipura for gain? |
| 14:27.50 | [Lamer] | nemisus: you mean to access my * server? |
| 14:28.00 | doolph | I tried codec g711 but g729 seems to be better |
| 14:28.05 | doolph | which is better |
| 14:28.09 | doolph | g723 or what |
| 14:28.13 | doolph | or 711 |
| 14:28.20 | jontow | lamer; he's asking if you have an RPM (redhat archive) of asterisk for FC3 or RHEL4 ;) |
| 14:28.29 | [Lamer] | nemisus: yes, I'm running it with a real IP and set up a couple of SRV records for my domain but haven't tried it out |
| 14:29.00 | [Lamer] | jontow: right |
| 14:29.05 | doolph | which is the best asterisg management |
| 14:29.27 | [Lamer] | jontow: I am not sure then I installed using cvs head stuffs |
| 14:29.31 | nemisus | [Lamer] I'm after a asterisk rpm for FC3 or RHEL4? you got? |
| 14:30.04 | jontow | doolph; i like ulaw .. G.711U; but its heavy on bandwidth.. the management... not really many good public solutions; best bet is to learn what your scope of management is and try to add that to an existing one that doesn't quite do it; or write something simple from scratch |
| 14:30.18 | jontow | nemisus; why not just build from cvs? |
| 14:30.22 | coppice | doolph: what do you consider better? better sound, lower bit rate, etc? |
| 14:30.31 | jontow | nemisus; the process is quite well documented and will save you repeating the question so many times :) |
| 14:30.45 | doolph | coppice better sound quality |
| 14:30.58 | [Lamer] | nemisus: I followed the http://www.voip-info.org/tiki-index.php?page=Asterisk%20Fedora%20Core%203 |
| 14:31.02 | doolph | i was at lan, bitrate shouldnt be a problem |
| 14:31.11 | [Lamer] | nemisus: to install on FC3 |
| 14:31.16 | nemisus | jontow, cos I've got a SLA for this box and an RPM would save time |
| 14:31.16 | coppice | G.711 is far getter than G.729 or G.723.1 for sound quality. no real comparison |
| 14:31.26 | jontow | how? |
| 14:31.32 | jontow | if its fast hardware |
| 14:31.46 | jontow | i can download and compile asterisk from CVS in less time than you've spent repeating the question? |
| 14:32.06 | nemisus | jontow, does it matter, if you don't have an FC3 rpm then say so |
| 14:32.33 | jontow | it does if you have an SLA; an RPM wouldn't save you much for time.. no i don't have one, i use freebsd, netbsd, and gentoo |
| 14:32.59 | nemisus | ok glad we worked that out |
| 14:33.25 | doolph | ok |
| 14:33.33 | doolph | I will reinstall the asterisk from scratch again |
| 14:34.09 | *** join/#asterisk easimon_ (~easimon@localhorst.kawo2.RWTH-Aachen.DE) |
| 14:34.18 | jontow | nemisus; just type this at your prompt: export CVSROOT=":pserver:anoncvs@cvs.digium.com:/usr/cvsroot";touch ~/.cvspass;cvs login;cvs co -rv1-0 asterisk libpri zaptel;cd zaptel;make;sudo make install;cd ../libpri;make;sudo make install;cd ../asterisk;make;sudo make install;sudo make samples |
| 14:34.39 | [Lamer] | nemisus: I installed using cvs and does it make any diferrence to the video mode for SIP connections? |
| 14:34.49 | jontow | or spend another 15-30mins looking for an RPM :) |
| 14:35.21 | doolph | jontow |
| 14:35.27 | *** join/#asterisk dca_ (~yruh8ing@c-67-166-37-218.hsd1.co.comcast.net) |
| 14:35.30 | [Lamer] | jontow: your discussion isn't really creative |
| 14:35.55 | [Lamer] | jontow: stop that buudy |
| 14:35.57 | nemisus | jontow I'm not installing CVS on an enterprise box... jontow, I know to do all this.. i choice not to, to minimise risk |
| 14:35.59 | jontow | it isn't meant for creativity |
| 14:36.20 | doolph | anyone got installed AMP on Redhat enterprise |
| 14:36.35 | nemisus | doolph will do by tomorrow |
| 14:36.50 | doolph | I tried but it fails |
| 14:36.52 | [Lamer] | jontow: I think nemisus knows what 's good for him |
| 14:38.38 | [Lamer] | I don't have the rpm version either so I used cvs |
| 14:39.05 | *** part/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 14:47.14 | *** join/#asterisk l-fy (~diana@l-fy.developer.yate) |
| 14:47.15 | l-fy | hello |
| 14:47.33 | *** part/#asterisk l-fy (~diana@l-fy.developer.yate) |
| 14:49.56 | *** join/#asterisk my007ms (~arkuser@217.139.240.35) |
| 14:53.11 | my007ms | hi all |
| 14:56.40 | [Lamer] | my007ms: hi, how are you doing? |
| 14:57.39 | *** join/#asterisk taspegao (jorgeraide@c-69-180-117-144.hsd1.fl.comcast.net) |
| 14:58.05 | taspegao | i can't to uses firefly |
| 14:58.08 | taspegao | :( |
| 14:58.26 | taspegao | some cand send me the cfg in asterisk? |
| 14:58.42 | taspegao | because i am doing something wrong |
| 14:58.43 | taspegao | :( |
| 14:59.12 | [Lamer] | taspegao: which cfg? |
| 14:59.38 | taspegao | configuration |
| 14:59.39 | [Lamer] | taspegao: you can "make samples" to get the samples file |
| 14:59.40 | taspegao | sorry |
| 14:59.45 | taspegao | yes |
| 14:59.46 | taspegao | please |
| 14:59.53 | taspegao | but more specific |
| 14:59.59 | taspegao | i need the cfg in iax.conf |
| 15:00.04 | taspegao | and extensions.conf |
| 15:00.14 | taspegao | becasuse i doesn't work |
| 15:00.28 | taspegao | i need the samples |
| 15:00.30 | [Lamer] | taspegao: http://www.voip-info.org can help you out |
| 15:01.16 | taspegao | i saw some samples |
| 15:01.19 | taspegao | but nothing |
| 15:01.23 | [Lamer] | or cd /usr/src/asterisk; make samples |
| 15:02.04 | [Lamer] | it will replace all your existing *.conf ones |
| 15:02.15 | MikeJ[Laptop] | taspegao, you don't need to hit enter between every word... you can actually type out a thought THEN hit enter ;) |
| 15:02.22 | [Lamer] | in /etc/asterisk for example |
| 15:02.28 | *** join/#asterisk Deryl (pgpkeys@static-141-149-128-140.buff.east.verizon.net) |
| 15:04.33 | doolph | sip is more used than iax eh |
| 15:04.50 | zyke | how do you see realtime sip users in the CLI? sip show peers doesn't seem to show realtime users |
| 15:08.13 | *** join/#asterisk darwin35 (~darwin35@24.3.226.147) |
| 15:08.21 | darwin35 | ManX you around |
| 15:10.35 | MikeJ[Laptop] | no, I ate him |
| 15:11.40 | file | good god little kids are in the house |
| 15:11.41 | file | shoot me now |
| 15:13.26 | *** join/#asterisk TheEmperor (TheEmperor@218.111.51.8) |
| 15:14.25 | MikeJ[Laptop] | are they yours?? :) |
| 15:14.38 | *** join/#asterisk phillytech (phillytech@adsl-207-245-115-100.cust.oldcity.dca.net) |
| 15:15.50 | darwin35 | Mike do you know abut the a command for vm |
| 15:16.20 | phillytech | <<< willing to compensate for help on sending CALLERID to serial port ttyS0 |
| 15:16.37 | Moc | phillytech, that not that hard to do |
| 15:16.45 | Moc | you can do C ? |
| 15:17.04 | phillytech | No, I can't even doe bash or sh |
| 15:17.15 | Moc | lol then it might not be that easy.. |
| 15:17.47 | darwin35 | why to ttys0 ? |
| 15:17.59 | Moc | probably for VM system |
| 15:18.01 | phillytech | I tried in extensions s,1,System(/bin/echo "{CALLERID}" > /dev/ttyS0 and get a 0 result |
| 15:18.02 | Moc | external one |
| 15:18.09 | darwin35 | ahh |
| 15:18.25 | darwin35 | why use external vm ? |
| 15:18.29 | phillytech | need to integrate emergency cellular callboxes to pan tilt zoom street cameras |
| 15:18.35 | ManxPower | phillytech: Let me think about what you need for a few mins. I'm only on my first cup of coffee. |
| 15:19.05 | Moc | phillytech, that would be the easy way to di it.. but might have escape problems |
| 15:19.06 | jontow | phillytech; SMDI? |
| 15:19.11 | Moc | and missing $ |
| 15:19.34 | phillytech | oh, I have $ in it, just forgot in here |
| 15:19.45 | phillytech | SMDI ? |
| 15:19.47 | ManxPower | phillytech: Yes, Moc is right. it's ${CALLERID} |
| 15:20.12 | ManxPower | phillytech: SMDI is some telco protocol for talking between serial devices. |
| 15:20.22 | jontow | yeah, its a messaging protocol by bellsouth iirc.. used to signal line changes, ie. stuttered dialtone for waiting voicemail |
| 15:20.25 | ManxPower | phillytech: Do you need callerid NAME or just Callerid NUMBER? |
| 15:20.45 | phillytech | just number |
| 15:20.48 | phillytech | exten => s,1,System(echo "#${CALLERID}." > /dev/ttyS0) |
| 15:21.01 | ManxPower | phillytech: then use ${CALLERIDNUM} |
| 15:21.12 | ManxPower | no escape issues then |
| 15:21.23 | phillytech | I use the # and . charachters for other automation |
| 15:21.32 | phillytech | oh ill try that |
| 15:22.01 | Moc | Im feeling soo good, having time to do C code ;) |
| 15:22.16 | my007ms | hi all can someone tell me where to find doc abut AMP i install it and it's fine great |
| 15:22.22 | Moc | you dont know how relaxing is a black console window with C code !!! |
| 15:22.24 | my007ms | but where is the manual |
| 15:22.37 | ManxPower | Moc: You're sick. |
| 15:22.41 | jontow | :D |
| 15:22.49 | ManxPower | Moc: That's like a blank page when you're trying to write a paper. |
| 15:23.28 | Moc | ? I actually have code already in... |
| 15:23.36 | phillytech | the result i get in asterisk -r is - Executing System("Zap/1-1", "echo "#2153388415." > /dev/ttyS0") in new stack |
| 15:23.38 | ManxPower | Moc: Ah. OK. |
| 15:23.41 | Moc | my meetme strip down version |
| 15:23.56 | phillytech | oh oopps now you know my phone # :) |
| 15:23.57 | ManxPower | phillytech: That's looks right. |
| 15:24.05 | TheEmperor | can someone tell me if this is correct? Dial,(IAX2/{ARG1}|60|tT) |
| 15:24.17 | ManxPower | phillytech: You'll need to handle locking of the serial port and stuff like that too. |
| 15:24.18 | phillytech | yeah but nothing on ttyS0, maybe permissions? |
| 15:24.27 | jontow | permissions are a very likely case |
| 15:24.31 | ManxPower | phillytech: As long as you are running asterisk as root. |
| 15:24.42 | ManxPower | phillytech: Get it working OUTSIDE of Asterisk first. |
| 15:24.46 | phillytech | Im using A@home |
| 15:24.54 | ManxPower | phillytech: HAHAHA! |
| 15:25.07 | jontow | hmm |
| 15:25.12 | jontow | i reinforce my statement now |
| 15:25.13 | Moc | A@Home is such a bad thing in alot of way.. |
| 15:25.14 | phillytech | I can type echo blah > /dev/ttyS0 and that works |
| 15:25.19 | jontow | permissions are a very likely case. |
| 15:25.22 | ManxPower | phillytech: System(echo "Hello World." > /tmp/bob) |
| 15:25.34 | ManxPower | then look at the permissions of /tmp/bob (specifically the user/group |
| 15:25.43 | phillytech | Manx, that doesnt work either |
| 15:25.45 | Moc | people using asterisk without understanding it.. |
| 15:25.50 | ManxPower | I think Asterisk@Home runs as non-root. |
| 15:26.14 | file | SO, I blocked my door with my Dell workstation so the little kids in the house couldn't open my door |
| 15:26.24 | Moc | A@Home for Ethernal dummies ;) |
| 15:27.08 | ManxPower | file: the owners of the kids didn't like you blocking the door with a shotgun? |
| 15:27.10 | phillytech | so maybe I should build just an asterisk box by itself |
| 15:27.24 | Moc | phillytech, yes.. |
| 15:27.39 | file | ManxPower: correct |
| 15:27.41 | ManxPower | phillytech: If you want to do ANYTHING even slightly odd, build your own box |
| 15:27.47 | ManxPower | file: Well that sucks. |
| 15:27.47 | Moc | non-root usage is a good thing.. but if you use something build up by someone else, you will never know your pbx.. |
| 15:27.48 | file | they thought my finger my slip and fire a round at some innocent |
| 15:27.52 | phillytech | since all it will do is passivley pull CID and no other function |
| 15:28.41 | Moc | and asterisk usage need a knowleage user when in production, because you need to know how to protect yourself when building your cfg, or some phreaker will do alot of LD calls |
| 15:28.47 | phillytech | Thnaks guys, I gues trying to get a@h to run as root would not be wize? |
| 15:29.09 | file | alot being more then one! |
| 15:29.11 | jontow | hehe |
| 15:29.11 | Moc | asterisk default setup is as root... but it better to run as non-root for security reason |
| 15:29.14 | ManxPower | phillytech: Most people run Asterisk as root. Asterisk@Home is NOT the same as Asterisk |
| 15:29.31 | pussfeller | there somthing that takes the csv and spits out a pretty webpage |
| 15:29.35 | ManxPower | As Moc says it's a good idea to run Asterisk as non-root. It's just not exactly trivial to do. |
| 15:29.54 | phillytech | It has helped me learn the config stuff |
| 15:30.13 | *** join/#asterisk NewSole (dave@i216-58-44-245.avalonworks.net) |
| 15:30.14 | ManxPower | phillytech: you can always run an suid-root version of "echo" LOL! |
| 15:30.26 | phillytech | just never had time to build a raw asterisk, when I saw an iso, then had time for that |
| 15:31.02 | file | I had a dream lastnight I was flying to Cluecon, but I was a month too early |
| 15:31.17 | Moc | file, I was told chicago is a nice city |
| 15:31.47 | WilliamK | Moc, unless you're planning to go over by Hasley from what I understand |
| 15:32.07 | file | Moc: we should have fun. |
| 15:32.20 | Moc | I think we will... |
| 15:32.53 | phillytech | Im an electronic security sytems admin, and most of us lack proper IT skills for the newer IP systems, even the manufacturers of such seem to lack |
| 15:33.23 | file | Moc: did you see our presentation times were put up? |
| 15:33.38 | Moc | yes, Im on the last day |
| 15:33.41 | file | indeed |
| 15:34.05 | MikeJ[Laptop] | come to cluecon. www.cluecon.com |
| 15:34.08 | MikeJ[Laptop] | :) |
| 15:34.18 | file | yes indeed, everyone should come |
| 15:34.36 | MikeJ[Laptop] | bring the kids, file loves em |
| 15:34.52 | file | Junky can do daycare *G* |
| 15:34.56 | MikeJ[Laptop] | hehe |
| 15:35.11 | my007ms | pleas some one help i can used config file but can not used AMP where i can add contex in this AMP |
| 15:35.24 | MikeJ[Laptop] | welcome to french canadian daycare with MOC and Junky, how may we help you :) |
| 15:35.49 | file | yup lol |
| 15:35.50 | Moc | lol I'll try to articulate alittle more |
| 15:36.01 | MikeJ[Laptop] | ? |
| 15:36.06 | file | Moc: I'm doing my presentation in Keynote ;) |
| 15:36.11 | my007ms | do i need to make Trunks in AMP to used Extension |
| 15:36.18 | my007ms | ? |
| 15:36.26 | Moc | I actually need to show how to develope a C App |
| 15:36.37 | file | my007ms: we aren't an AMP support channel, heck most of us never used it 'nor like it |
| 15:36.42 | MikeJ[Laptop] | make is somthing cool |
| 15:36.52 | file | Moc: write one while talking :) |
| 15:36.54 | MikeJ[Laptop] | file, I've "used" it for a day or so |
| 15:37.06 | file | MikeJ[Laptop]: and how was "it" |
| 15:37.19 | MikeJ[Laptop] | basically it's like a code converter written in basic |
| 15:37.20 | Moc | I'll have to find some simple evil stupid app to do |
| 15:37.24 | file | LOL |
| 15:37.29 | MikeJ[Laptop] | it works, but makes really ugly dialplans |
| 15:37.33 | file | yeah |
| 15:37.36 | my007ms | so how i can used asterisk with db |
| 15:37.43 | file | I looked at it's dialplan once, then I said... oh noooooooo |
| 15:37.49 | MikeJ[Laptop] | but once you get used to what it calls everything, it's not bad |
| 15:37.56 | ManxPower | I should publish my /ignore list. |
| 15:38.00 | MikeJ[Laptop] | hehe |
| 15:38.03 | my007ms | whaere i can add contex in asterisk db i what taple |
| 15:38.04 | my007ms | ? |
| 15:38.42 | MikeJ[Laptop] | my007ms, there is an additional file called somthing like extension_custom.conf that you can add custome confs in amp |
| 15:38.47 | MikeJ[Laptop] | it is not db driven |
| 15:38.57 | MikeJ[Laptop] | you will have to read the readmes |
| 15:39.18 | ManxPower | I see a lot of people spending enormous amounts of time trying to help people in my /ignore list. |
| 15:39.22 | Moc | I have a webinar to prepare for a Lawyer IT Group... |
| 15:39.32 | file | my fortune for today: Prosperity and love are in reach |
| 15:40.04 | MikeJ[Laptop] | awww, our file is in LOVE :) |
| 15:40.13 | MikeJ[Laptop] | who is it? |
| 15:40.20 | file | not a clue |
| 15:40.28 | MikeJ[Laptop] | get a clue |
| 15:40.39 | file | at Cluecon! |
| 15:40.50 | MikeJ[Laptop] | www.cluecon.com |
| 15:41.22 | MikeJ[Laptop] | has anyone seen the crazy conference bot that the guys running cluecon made? |
| 15:41.26 | MikeJ[Laptop] | it's really cool |
| 15:41.37 | MikeJ[Laptop] | you can see it in the #996 channel |
| 15:41.50 | Moc | very annoying thought ;) |
| 15:41.55 | MikeJ[Laptop] | hehe |
| 15:42.05 | phillytech | Manx, just want to extend my gratitude for your help, I was hesitant to even ask, thats why I offered compensation. If I come up with a working solution, I might be compensated with a bonus that I would be willing to sahre with those that assist |
| 15:42.08 | MikeJ[Laptop] | the bot, or the people with permissions to use it? |
| 15:42.36 | ManxPower | phillytech: The fact that yu TRIED stuff first was why I spent the time trying to help. |
| 15:42.48 | Moc | phillytech, just try to help others if you can ;) |
| 15:42.50 | MikeJ[Laptop] | try=good |
| 15:42.55 | ManxPower | phillytech: you can thank me by helping me to find a job in Europe. Do a /whois ManxPower or /msg ManxPower for more information. |
| 15:43.33 | phillytech | I dont think I have any connections in Europe :( |
| 15:43.49 | ManxPower | phillytech: 8-( I figured it could not hurt to ask. |
| 15:44.01 | MikeJ[Laptop] | ManxPower is trying to make a Eruopean connection;) |
| 15:44.11 | phillytech | I work at a large University, where I have access to large bandwidth |
| 15:44.29 | my007ms | what is Trunks used for |
| 15:44.30 | MikeJ[Laptop] | ManxPower, maybe if you come to cluecon you might meet somone |
| 15:44.31 | my007ms | ? |
| 15:44.39 | file | fyi people, the hotel for Cluecon does have free high speed internet access |
| 15:44.44 | file | so let's all be sure to max it out while there, mmmkthx |
| 15:44.48 | Moc | woohoo... |
| 15:44.48 | MikeJ[Laptop] | and a pool |
| 15:44.52 | MikeJ[Laptop] | I think |
| 15:44.53 | ManxPower | MikeJ[Laptop]: Perhaps I'll meet someone if I go to VON Europe and Astricon Europe. |
| 15:44.56 | file | yes there's a pool |
| 15:45.08 | *** join/#asterisk joaovianna (naturalvoi@node-40247a6a.ewr.onnet.us.uu.net) |
| 15:45.10 | ManxPower | file: if someone buys me a plane ticket, I'll be there. |
| 15:45.11 | Moc | damn it gona be fun.. |
| 15:45.27 | MikeJ[Laptop] | druve |
| 15:45.29 | MikeJ[Laptop] | drive |
| 15:45.34 | phillytech | I could only introduce you to key people involved in their VOIP over Internet2 and centrex elimination prohect |
| 15:45.45 | ManxPower | MikeJ[Laptop]: If someone buys me a car, I'll be there. |
| 15:45.50 | Moc | yea drive... Im be driving from MTL to chicago with a friend |
| 15:45.50 | MikeJ[Laptop] | hehe |
| 15:45.54 | phillytech | at the University where I work, but it's in Philly |
| 15:45.56 | MikeJ[Laptop] | bus?? |
| 15:46.03 | file | I'm flying alllllll the way |
| 15:46.17 | Moc | it just 8 hours drive |
| 15:46.18 | MikeJ[Laptop] | walk? |
| 15:46.29 | MikeJ[Laptop] | :D |
| 15:46.36 | Moc | so 4 hours of driving each |
| 15:46.44 | MikeJ[Laptop] | ManxPower, where are you? |
| 15:46.54 | ManxPower | MikeJ[Laptop]: near New Orleans. |
| 15:47.05 | MikeJ[Laptop] | train :) |
| 15:47.15 | jontow | moc; you normally in montreal? |
| 15:47.22 | phillytech | <<<< am basically an overgrown ameture script kiddie :) |
| 15:47.27 | ManxPower | I really don't have any interest in going to Cluecon after spending a month in Europe. |
| 15:47.42 | MikeJ[Laptop] | fine, be that way.... |
| 15:47.52 | PTG123 | morning all :) |
| 15:47.55 | MikeJ[Laptop] | morning |
| 15:47.59 | file | hi |
| 15:48.22 | my007ms | in the extensions table there is application what typ in this something like this Dial or Dial(SIP/1000,60,Tt) |
| 15:48.38 | ManxPower | I'd suggest having Cluecon in New Orleans, but New Orleans in Aug is...well...not fit for people. |
| 15:49.03 | file | thus why you suggested it? lol |
| 15:49.47 | my007ms | can some one help to understand who asterisk read from mysql |
| 15:50.23 | file | excellenr |
| 15:50.25 | PTG123 | i suggest having one someplace worth actually going to :) |
| 15:50.44 | my007ms | i can not add context in mysql db where i can add this |
| 15:50.46 | my007ms | ? |
| 15:52.33 | ManxPower | PTG123: Amsterdam! |
| 15:52.50 | jontow | my007ms: Error: invalid syntax. ... and furthermore: please check the number, and dial again; etc. |
| 15:52.51 | jontow | :) |
| 15:52.59 | PTG123 | hah |
| 15:53.05 | PTG123 | i don't wanna take that long of a flight :) |
| 15:54.14 | MikeJ[Laptop] | my007ms, you are using AMP correct? |
| 15:55.10 | my007ms | coz i need to make asterisk work with mysql |
| 15:55.25 | MikeJ[Laptop] | amp does not make asterisk work with mysql |
| 15:55.41 | Sato1 | realtime? |
| 15:55.44 | MikeJ[Laptop] | amp generates text config files off of a db. That is all amp, not astersik |
| 15:55.55 | JerJer | there is absolutely no reason to make asterisk depend on a database |
| 15:56.07 | my007ms | no there is need |
| 15:56.07 | JerJer | realtime is not the answer |
| 15:56.08 | zyke | how do you see realtime sip users in the CLI? sip show peers doesn't seem to show realtime users |
| 15:56.17 | MikeJ[Laptop] | you don't |
| 15:56.24 | JerJer | zyke: dont' run realtime |
| 15:56.29 | MikeJ[Laptop] | zyke, there is a bug for that in the bugtracker |
| 15:56.34 | my007ms | coz there are to many nomber and mysql make it easy to update |
| 15:56.53 | zyke | jerjer: why not.. |
| 15:57.01 | Sato1 | must of devices (sip/iax) on my asterisk are friends, and realtime does not allow friend types |
| 15:57.04 | MikeJ[Laptop] | JerJer is a realtime hater :) |
| 15:57.06 | zyke | i'm using realtime using odbc |
| 15:57.21 | JerJer | i'm dying for someone to tell me which Class 4/5 telco switch depends on a database to operate |
| 15:58.08 | MikeJ[Laptop] | jerjer, how do big telco switches find out where to send calls in this day of LNP? |
| 15:58.09 | Moc | anyone from voipsupply in here ? |
| 15:58.15 | Sato1 | the point of the realtime is to not restart asterisk everytime a device is added, so, that would be a good reason for depending on a db i guess |
| 15:58.20 | JerJer | MikeJ[Laptop]: OCN |
| 15:58.22 | Sato1 | but still, not really reliable |
| 15:58.34 | my007ms | then what is mean for DID Routes? |
| 15:58.39 | JerJer | Sato1: you do not restart asterisk every time there is a device added - find a clue |
| 15:58.42 | zyke | jerjer: what would u use if you have a production box with large number of users and would u be reloadin * evertime you add a new user? |
| 15:58.50 | JerJer | not realtime |
| 15:59.15 | Sato1 | JerJer, adding extra devices |
| 15:59.30 | JerJer | Sato1: ...does not require RESTARTING of asterisk |
| 15:59.32 | JerJer | period |
| 15:59.40 | JerJer | new paragraph |
| 15:59.45 | JerJer | indent |
| 15:59.58 | JerJer | do not force asterisk to depend on the database to operate - then you have Cisco Call Manager |
| 16:00.10 | JerJer | memory is cheap |
| 16:00.30 | Sato1 | i mean, if i have 3 sipphones, but i add 10 contexts in sip.conf, i wouldnt need to restart asterisk in a little while (unless i get more than 10 sip phones, hehehe) |
| 16:00.50 | JerJer | um one more time for the cheap seats |
| 16:00.57 | JerJer | YOU DO NOT RESTART ASTERISK |
| 16:01.03 | MikeJ[Laptop] | Sato1, reload sip |
| 16:01.24 | Sato1 | wel.. that was a sample, iax does need to restart, not reload |
| 16:01.24 | jontow | (again for those cheap seats ;)) |
| 16:01.36 | JerJer | do i need to break out the clue bat? |
| 16:01.40 | MikeJ[Laptop] | ~troutslap everyone |
| 16:01.41 | jbot | ACTION slaps everyone around with a large trout |
| 16:01.43 | zyke | MikeJ: what happens if you want to add iax user? u can't do iax reload |
| 16:01.45 | Sato1 | that was the only think i came up for that situation |
| 16:01.45 | JerJer | even for iax YOU DO NOT RESTART ASTERISK |
| 16:01.51 | *** join/#asterisk mmlj4 (~looseduk@ip68-14-124-25.no.no.cox.net) |
| 16:01.52 | jontow | jerjer; when implementing the chan_skinny or chan_h323 stuff, did you run across docs of the channel API or have you been working with knowledge you've gained on your own? |
| 16:02.02 | Sato1 | what other way would be good to dont restart asterisk on adding sip/iax devices? |
| 16:02.07 | Sato1 | besides reload on sip |
| 16:02.17 | JerJer | you never restart asterisk |
| 16:02.18 | MikeJ[Laptop] | reload |
| 16:02.30 | MikeJ[Laptop] | reload!=restart |
| 16:02.35 | jontow | (just curious as i'd like to start playing with it a bit and wanted to know if there are any pre-existing not-very-distributed documentation on such things) |
| 16:02.39 | JerJer | yes listen to MikeJ[Laptop] |
| 16:02.53 | JerJer | jontow: i simply studied chan_sip |
| 16:02.58 | MikeJ[Laptop] | WOW.. .did I just catch that right :) |
| 16:03.00 | jontow | ok :) |
| 16:03.02 | Sato1 | i got the point between reload!=restart, but making something clear... |
| 16:03.09 | jontow | sato1; reload chan_iax2.so |
| 16:03.15 | MikeJ[Laptop] | JerJer, that's an ugly ugly learning tool |
| 16:03.18 | jontow | sato1; also reload chan_sip.so, or 'sip reload' |
| 16:03.20 | Sato1 | reload iax2 too? |
| 16:03.24 | JerJer | and asked questions to Mark - but he was less busy back then |
| 16:03.31 | jontow | aha |
| 16:03.44 | Sato1 | hmm... dont know where i readed that not always reload really "reloads" all configurations |
| 16:03.47 | JerJer | now there is a whole -dev channel and mailing list |
| 16:03.49 | zyke | reloading does not slow down * ? |
| 16:03.52 | MikeJ[Laptop] | there are simpler chaneldrivers around to look at the basics as well |
| 16:04.10 | MikeJ[Laptop] | Sato1, did you try it? |
| 16:04.16 | jontow | yeah, i was looking briefly at the included chan_nbs.c |
| 16:04.22 | jontow | much smaller source |
| 16:04.25 | MikeJ[Laptop] | jerjer, how many users do you have? |
| 16:04.27 | Sato1 | i would now.. i got only an asterisk at office and at home |
| 16:04.41 | JerJer | MikeJ[Laptop]: all of them |
| 16:04.42 | Sato1 | and a gateway one in a colocate for friends |
| 16:04.45 | MikeJ[Laptop] | nice |
| 16:04.52 | jontow | haha |
| 16:05.03 | JerJer | you are going to have to be more specific if you want a real answer |
| 16:05.11 | JerJer | Nufone is not my only voip play here |
| 16:05.17 | MikeJ[Laptop] | my point being jerjer runs plenty of users without a db |
| 16:05.31 | Sato1 | so.. that would be a nice thing to do in my little TCL/TK gui |
| 16:05.37 | JerJer | we have a db - we simply don't force asterisk to DEPEND on it to operate |
| 16:05.38 | MikeJ[Laptop] | I know others who have many thousands of users on text config files |
| 16:05.39 | Sato1 | not restarting... reload |
| 16:05.59 | jontow | yeah, im not so convinced on the realtime stuff either.. only reason i see a database as being useful is generating configuration from in a web-based or similar environment; the rest.. meh, maybe CDR records for reporting purposes. |
| 16:06.04 | Sato1 | hehe, i recall the phrase, "dont fold the maps.. roll them", from "Twister" the movie |
| 16:06.04 | MikeJ[Laptop] | jerjer... hear what I mean, not what I say |
| 16:06.38 | JerJer | in fact I have an entire provisioning system around asterisk |
| 16:07.40 | MikeJ[Laptop] | but you crank it out into text configs right? |
| 16:08.00 | JerJer | did i say that? |
| 16:08.24 | MikeJ[Laptop] | no, hense the ? at the end. |
| 16:08.44 | JerJer | there ya go |
| 16:08.49 | jontow | he turns it into goo, flips it twice with the lucky spatula, makes it land just so and then does (are you paying attention sato1?): "reload chan_iax2.so" |
| 16:08.58 | jontow | *cough* |
| 16:08.59 | jontow | :) |
| 16:09.20 | Moc | finally... conf_run have 88 line of code now .. |
| 16:09.26 | MikeJ[Laptop] | JerJer, do you really enjoy playing symantec games? |
| 16:09.42 | zyke | how do you see realtime registered users in the CLI? |
| 16:09.48 | MikeJ[Laptop] | Moc, I won't be happt untill it's 10 :) |
| 16:09.52 | jontow | i used to work with a jeremy mcnamara btw.. however, he is not jerjer, and did not write chan_skinny or anything related; was a cook in the restaurant i used to work in :) |
| 16:10.04 | jontow | i found that odd coincidence when i first stepped into the asterisk world |
| 16:10.05 | JerJer | its called i am not going to freely provide others, who are most likely at some level a competitor to me, with all of the details to duplicate what I have done |
| 16:10.08 | MikeJ[Laptop] | zyke, I told you too look in mantis |
| 16:10.22 | MikeJ[Laptop] | well that's fair enough |
| 16:10.23 | *** join/#asterisk Wi_Fi (~OUT@c-24-127-12-85.hsd1.ca.comcast.net) |
| 16:10.26 | tzanger | JerJer: why not? it's the american way |
| 16:10.38 | MikeJ[Laptop] | say that, it makes you sound less like a jerk |
| 16:10.47 | JerJer | but i am a jerk |
| 16:10.54 | zyke | MikeJ: what's mantis? |
| 16:11.10 | MikeJ[Laptop] | well, clearly. |
| 16:11.15 | jontow | http://bugs.digium.com/ |
| 16:11.21 | jontow | ask no further about this 'mantis' you speak of ;) |
| 16:11.34 | *** join/#asterisk jeffik (~jeffik@CPE00080da2f8fc-CM0011aec8aa0a.cpe.net.cable.rogers.com) |
| 16:12.48 | JerJer | jontow: i went to a university that had another jeremy mcnamara in the same major as me |
| 16:13.04 | JerJer | talk about crazy odds |
| 16:13.22 | jontow | hahah |
| 16:13.24 | jontow | weird :) |
| 16:13.27 | Moc | hehe |
| 16:13.43 | jontow | guess it's a more common name than i thought |
| 16:13.46 | JerJer | then i guess there is some jeremy mcnamara out there that went crazy and shot up a high school somewhere |
| 16:14.11 | Moc | yea, that the excuse .. |
| 16:14.11 | JerJer | some fuckers in this channel brought that up a while ago |
| 16:14.19 | JerJer | there is no way i was in high school in like 1998 |
| 16:14.19 | Moc | it not me, it the other one.. |
| 16:14.34 | Moc | I was.. oh oh |
| 16:14.56 | jontow | i was :/ |
| 16:14.56 | Moc | ok back to... work |
| 16:15.03 | jontow | damn age nonsense |
| 16:15.09 | jontow | i'd just started using freebsd in january of 1998 |
| 16:15.18 | Juggie | hmmmmmmm |
| 16:15.24 | Juggie | when did i first install linux |
| 16:15.27 | Moc | poor you |
| 16:15.31 | Juggie | 2.2.x |
| 16:15.39 | Juggie | redhat 5 |
| 16:15.43 | Juggie | what year was that |
| 16:15.52 | Moc | first linux is redhat 4 |
| 16:16.00 | Moc | or debian, or mandrake.. dont remember.. |
| 16:16.00 | jontow | i went through all of the versions up to 3.5 on a 486sx/25mhz (overclocked to 40mhz!), 12MB of RAM, and dual 210MB drives |
| 16:16.03 | jontow | ;) and no working cdrom |
| 16:16.18 | JerJer | i installed linux for the first time when Slackware first came out - can't immedately even remember when that was - pretty long ago |
| 16:16.19 | Juggie | i have a 2x cdrom that still works |
| 16:16.20 | Juggie | somewhere |
| 16:16.27 | Juggie | a creative labs one |
| 16:16.39 | Juggie | connected to your sound card, wasnt ide |
| 16:16.57 | Moc | lol oh yea remember my CDROM drive like that... I hated it |
| 16:17.04 | Moc | costed 1200$ for a 2x CDROM |
| 16:17.05 | JerJer | i found my Yamaha CDR101 WORM drive the other day - when I decided to clean my barn |
| 16:17.05 | jontow | made a good router too, until one of the drives died in ~2001 .. that machine was amazing; would easily push 250kb/sec constantly over my cablemodem which did only slightly over that until about 2002 when they upped it to a full 3mbit |
| 16:17.10 | Juggie | used an ide cable tho.... |
| 16:17.39 | jontow | i had a pair of 'em.. mitsumi and matsushita drives connected to Aztech gold soundcards rebranded by packard bell |
| 16:17.41 | Juggie | i still have a p133 running a router at my parents house |
| 16:17.41 | jontow | (yuck.) |
| 16:17.45 | Juggie | i think i have redhat 9 on it |
| 16:17.56 | jontow | juggie :( me too; my apartment's router is a p133 |
| 16:18.06 | *** join/#asterisk Rick_Hunter (~rhunter@06-064.008.popsite.net) |
| 16:18.14 | Juggie | i'm gonna take it out next time i'm home |
| 16:18.26 | Juggie | and put in a wifi router |
| 16:18.26 | JerJer | swap in a WRAP board - they rule |
| 16:18.50 | jontow | neat.. i only route dialup at the apartment so im not so worried about swapping mine out for better hardware ;) |
| 16:18.53 | *** join/#asterisk eivindtr (~eivindtr@062016241059.customer.alfanett.no) |
| 16:19.00 | Juggie | dialup? |
| 16:19.01 | Juggie | eugh |
| 16:19.07 | jontow | thats the sound i make too |
| 16:19.13 | Juggie | why dialup |
| 16:19.27 | Juggie | damn russia scored |
| 16:19.30 | Juggie | 4-3 for canada now |
| 16:19.32 | jontow | because i live far far far out in nature's beautiful landscape.. you know that place waaaaay over on the horizon.. thats pretty close ;) |
| 16:19.42 | JerJer | GOOOOOOOAL |
| 16:19.51 | jontow | but seriously.. the only choices i have at my apartment are dialup and satellite-based expensive bandwidth with much latency |
| 16:20.06 | jontow | the dialup still offers better interactive (ssh) use, which is how i do just about *everything* |
| 16:20.08 | JerJer | jontow: sounds like you need to start a wireless isp |
| 16:20.12 | Juggie | you should talk your landlord into getting something installed |
| 16:20.20 | jontow | my dad is my landlord |
| 16:20.27 | jontow | ;) free rent, i pay utilities |
| 16:20.30 | Juggie | hah |
| 16:20.30 | jontow | thats why it is worth it to me |
| 16:20.40 | JerJer | hell then start a CLEC |
| 16:20.58 | *** join/#asterisk jwitte (~jwitte_su@firefly.alpha-lab.net) |
| 16:21.03 | jontow | its 3 minutes from at least 7 beautiful waterfalls and a bunch of nice shady picnic/camping spots :) |
| 16:21.06 | Juggie | jontow, what country are you in |
| 16:21.10 | jontow | upstate NY, USA |
| 16:21.17 | Juggie | ahhh |
| 16:21.21 | Juggie | close to me |
| 16:21.23 | Juggie | i'm in ottawa |
| 16:21.27 | JerJer | hell - stick a wrt outside |
| 16:21.33 | JerJer | you might find 'free' b/w |
| 16:21.37 | jontow | the company i work for has just started a CLEC.. they won't touch my area with a 20mile pole |
| 16:21.39 | jontow | not a chance |
| 16:21.47 | jontow | i officially have the only wireless hardware within 10 miles ;) |
| 16:22.01 | jontow | unofficially; perhaps.. but still |
| 16:22.03 | Juggie | i dont think i'd live without high speed |
| 16:22.06 | JerJer | sounds like the perfect market to start a wireless isp then |
| 16:22.13 | jontow | it is; if you can find bandwidth |
| 16:22.25 | Juggie | i got a cable modem in like fuck.... 94/95 |
| 16:22.26 | JerJer | wireless distrubution |
| 16:22.28 | Juggie | 10 years ago |
| 16:22.32 | JerJer | light up a 150 foot tower |
| 16:22.38 | JerJer | 5ghz backhaul |
| 16:22.39 | jontow | this is literally more backwoods-ish than most people have seen.. closest source of bandwidth is literally 30 miles away |
| 16:22.41 | JerJer | 2.4ghz broadcast |
| 16:23.05 | jontow | problem is.. there isn't even enough customers to cover cost |
| 16:23.17 | Juggie | well get the hell out ;) |
| 16:23.19 | jontow | thats why the telco isn't even going to bother running DSL out that far (oh yeah.. 20miles from the telco) |
| 16:23.27 | JerJer | you would be surprised |
| 16:23.29 | Moc | down 80 line now hehe |
| 16:23.43 | Juggie | jontow, is their cell phone coverate? |
| 16:23.45 | Juggie | *coverage |
| 16:23.46 | jontow | no |
| 16:23.51 | Juggie | nothing? |
| 16:23.56 | jontow | *maybe* one bar on a good windy day |
| 16:24.03 | Juggie | hah. |
| 16:24.17 | JerJer | someone with a couple mil could come in there and own that market |
| 16:24.18 | Juggie | well i guess 1x/gprs is out |
| 16:24.20 | jontow | just enough to send and receive text messages if you stand on the bumper of my jeep and hold your arm up with the cellphone in it |
| 16:24.28 | jontow | (seriously.) |
| 16:24.29 | *** join/#asterisk iq|laptop (~iq@63-230-46-28.omah.qwest.net) |
| 16:24.42 | Juggie | why are you still there |
| 16:24.45 | Juggie | i could never live there :P |
| 16:25.01 | jontow | because its temporary, i can actually save money.. my housing bills total about $100/month |
| 16:25.01 | JerJer | 20 customers is break even |
| 16:25.11 | JerJer | i'm sure you can find 20 people in a 10 mile radius |
| 16:25.13 | jontow | there is also no money in this area ;) |
| 16:25.22 | jontow | you couldn't charge 20 people enough.. nobody'd pay it |
| 16:25.31 | JerJer | perfect |
| 16:25.32 | jontow | they all bitch about dialup |
| 16:25.39 | JerJer | write a grant |
| 16:25.47 | jontow | but when an alternative comes around, they won't pay the cost to get it installed |
| 16:26.01 | JerJer | rural broadband initiative |
| 16:26.06 | Juggie | the goverment has a big broadband push |
| 16:26.08 | jontow | (ie. $100/month for the first 3 months, then drops down to $50/month after that) |
| 16:26.09 | Juggie | write a application |
| 16:26.28 | jontow | without a seriously huge grant, the area won't get broadband for the next 10 years or more |
| 16:26.43 | jontow | anyone familiar with the monopolizing ways of TDS telco? |
| 16:26.46 | JerJer | i could do it with only a million or two |
| 16:26.54 | JerJer | which is trivial |
| 16:27.04 | JerJer | go around them |
| 16:27.14 | Juggie | the goverment spends millions easy |
| 16:27.24 | Juggie | hell, we bought 1.25million in cisco gear last month |
| 16:27.26 | JerJer | daily |
| 16:27.34 | jontow | thats the local company that does all of our phone service, all internet service, and is the only source of bandwidth until you haul $longwaysaway from verizon |
| 16:27.46 | *** join/#asterisk ta[i]nted (~tainted@adsl-69-108-107-206.dsl.irvnca.pacbell.net) |
| 16:27.49 | jontow | i've wanted to do wireless for a while |
| 16:27.54 | jontow | but .. the state will probably not like that much at all |
| 16:27.55 | Juggie | they are still a bitch to the goverment |
| 16:28.04 | jontow | the town is mostly a state-owned wildlife park |
| 16:28.05 | JerJer | sounds like the perfect market for the rural broadband initiative |
| 16:28.14 | Juggie | all the more reason to do wifi |
| 16:28.31 | jontow | so getting enough land to put up a 150ft tower safely without crossing any borders or anything like that is .. not easy :) |
| 16:28.43 | JerJer | one of my wisp projects was given $125,000 to cover a college town that already had cable and DSL |
| 16:28.58 | jontow | my dad does own a reaaal nice hill (tallest around) that'd probably facilitate such a tower though |
| 16:29.22 | jontow | center of the hamlet; and would reach most spots around.. though im in the mostly lightly populated area |
| 16:29.31 | jontow | closest town has 79 residents at last census poll |
| 16:29.38 | jontow | and i think they all responded.. |
| 16:29.38 | Juggie | hahh |
| 16:29.40 | jayk_ | i have two Cisco 7960 phones routed through asterisk..whenever i call voicemail, i can't use the touch keys, they are ignored by the aterisk server |
| 16:29.43 | jayk_ | anybody know why? |
| 16:29.59 | jontow | jayk; skinny or SIP ? |
| 16:30.00 | JerJer | define touch keys |
| 16:30.21 | JerJer | dtmfmode=rfc2833 either way |
| 16:30.23 | jayk_ | the key pad, 0-9, * and # |
| 16:30.25 | jayk_ | ok |
| 16:30.31 | jayk_ | i have dtmfmode=relax atm |
| 16:30.35 | Juggie | jayk_, you have to write code in your dialplan |
| 16:30.38 | JerJer | bleh |
| 16:30.49 | Juggie | to trap * and spawn the voicemail management |
| 16:30.50 | jontow | i've yet to touch dtmfmode=relax |
| 16:31.11 | jontow | i've mostly had to mess with dtmfmode=inband though |
| 16:31.12 | Juggie | i believe its a |
| 16:32.23 | Juggie | exten => a,1,VoiceMailMain(${ARG1}@default) |
| 16:32.34 | Juggie | see, when someone presses a inside VoiceMail |
| 16:32.37 | Juggie | it will exit |
| 16:32.38 | *** part/#asterisk Johnsie (~john@acs-24-154-32-12.zoominternet.net) |
| 16:32.39 | Juggie | and that traps it |
| 16:32.49 | Juggie | er, * inside VoiceMail |
| 16:32.56 | jayk_ | k |
| 16:33.26 | Juggie | if you press * you should see it looking for a in the context |
| 16:33.28 | jontow | i need a bigger desk.. and louder speakers.. |
| 16:36.53 | PyroSteve | Hey guys |
| 16:37.05 | PyroSteve | I just spotted a Cisco Phone in Oceans Eleven |
| 16:37.12 | PyroSteve | probably not an IP phone |
| 16:37.15 | jontow | i hear the damned things in my sleep |
| 16:37.21 | jayk_ | k |
| 16:37.24 | jayk_ | that worked, thanks. :) |
| 16:37.25 | jontow | everywhere i go i hear the default 7960 ringtone |
| 16:37.29 | jontow | oh, how it bothers me |
| 16:37.33 | Juggie | PyroSteve, cisco has never made anything that wasnt an ip phone |
| 16:37.42 | PyroSteve | oh ok |
| 16:37.48 | Juggie | they are in 24 too |
| 16:38.11 | PyroSteve | Maybe they were H323 |
| 16:38.25 | Juggie | i dont think cisco has ever done h323 |
| 16:38.30 | PyroSteve | Wow |
| 16:38.33 | Juggie | only sccp/skinny/mgcp/sip |
| 16:38.41 | PyroSteve | Oh ok |
| 16:39.13 | Juggie | 2minutes left and its 4-3 canada |
| 16:39.15 | Juggie | yay |
| 16:39.54 | PyroSteve | does anyone in here use ChanSpy ? |
| 16:39.55 | PyroSteve | Is it stable ? |
| 16:40.59 | Juggie | 1minute! |
| 16:41.02 | Juggie | whee |
| 16:43.09 | *** join/#asterisk Johnsie (~john@acs-24-154-32-12.zoominternet.net) |
| 16:46.29 | Moc | PyroSteve, yes it working fine for me |
| 16:46.32 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 16:47.26 | PyroSteve | Moc: I tried to download it a while back and couldn't find it. I looked in the asterisk source, the wiki, and on the digium ftp site ... |
| 16:47.42 | PyroSteve | Can you give me a link ? |
| 16:48.47 | doolph | hi |
| 16:49.28 | Zeeek | too many nachos |
| 17:01.27 | *** join/#asterisk likwid-- (likwid@nc-65-40-228-65.dyn.sprint-hsd.net) |
| 17:11.44 | *** join/#asterisk kapejod (~kapejod@e178002108.adsl.alicedsl.de) |
| 17:11.55 | Moc | isnt in CVS head now ? |
| 17:18.06 | *** join/#asterisk santiago (~santiago@63.245.86.227) |
| 17:27.23 | *** join/#asterisk Tond (Tond@Toronto-HSE-ppp3646966.sympatico.ca) |
| 17:28.07 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 17:31.38 | *** join/#asterisk cjk (~cjk@80.92.64.103) |
| 17:32.56 | cjk | hi, monitoring of sip is easy, using sipsak. but do you guys know a way to monitore an iax only system |
| 17:33.12 | Zeeek | ethereal |
| 17:33.32 | kapejod | signalling or audio monitoring? |
| 17:34.20 | Zeeek | mutant-X sucks; Somebody say something! |
| 17:34.47 | Zeeek | I wood but I'm board |
| 17:40.18 | doolph | May 14 12:38:58 VERBOSE[10758]: == Parsing '/etc/asterisk/res_mysql.conf': May 14 12:38:58 VERBOSE[10758]: == Parsing '/etc/asterisk/res_mysql.conf': Not found (No such file or directory) |
| 17:40.24 | doolph | any idea |
| 17:40.38 | PTG123 | anyone know a registration site that suggests domains for yoiu? |
| 17:40.40 | Qwell | create it |
| 17:40.41 | Zeeek | yes, there is no res_mysql.conf ? |
| 17:40.56 | Zeeek | PTG123 many do. Try whois.sc |
| 17:41.02 | mmlj4 | directnic.com |
| 17:41.02 | jontow | i don't think i understand the ast_queue_control function in channels.c :/ |
| 17:41.31 | jontow | godaddy.com does |
| 17:41.44 | mmlj4 | PTG123: i hear hguehgiduejfwex.net is free |
| 17:41.46 | doolph | Zeeek should i create it? |
| 17:42.14 | mmlj4 | (how's that for a suggestion?) |
| 17:42.18 | jontow | mmlj4; and you know, remember.. that sister site hguehgiduejfwey.net is up for grabs too |
| 17:42.20 | Zeeek | ptg123.com is available |
| 17:42.40 | mmlj4 | Zeeek: not for long |
| 17:43.44 | Zeeek | doolph I just googled on that and the third line is |
| 17:43.46 | Zeeek | http://www.voip-info.org/wiki-Asterisk+RealTime |
| 17:43.53 | Zeeek | where it sates what to do |
| 17:43.58 | Zeeek | why not take a look ? |
| 17:44.48 | doolph | Zeeek I was following a Manual, and when i tried to start the services it gave me that message |
| 17:45.11 | Zeeek | and you immediately googled for that |
| 17:45.31 | Zeeek | the answer is there on that page |
| 17:46.16 | doolph | but i dont have taht file |
| 17:46.45 | *** join/#asterisk Carp1 (carp_xigon@ip-204-97-151-156.modem.logical.net) |
| 17:47.01 | Zeeek | did you read the page above? |
| 17:47.07 | Zeeek | http://www.voip-info.org/wiki-Asterisk+RealTime |
| 17:47.44 | doolph | yes |
| 17:47.48 | doolph | it says |
| 17:47.51 | doolph | Copy asterisk-addons/configs/res_mysql.conf.sample to /etc/asterisk/res_mysql.conf |
| 17:48.08 | Zeeek | did you make samples? |
| 17:48.12 | doolph | [root@www default]# ls /usr/src/asterisk-addons/configs/ |
| 17:48.13 | doolph | cdr_mysql.conf.sample CVS |
| 17:48.27 | doolph | yes |
| 17:48.37 | doolph | can you check on your version |
| 17:48.49 | Zeeek | I don't use it |
| 17:49.42 | Zeeek | but what is this? http://asterisk.gnuinter.net/files/digium/asterisk-addons/configs/ |
| 17:50.23 | Zeeek | looking at that, it seems you could just touch the file and leave it empty |
| 17:50.43 | Zeeek | since everything is commented out anyway (uses defaults) |
| 17:50.58 | Zeeek | someone who uses realtime can confirm this? |
| 17:51.10 | doolph | i think that they changed the name |
| 17:51.49 | doolph | res_mysql.conf == cdr_mysql.conf.sample |
| 17:51.50 | doolph | i think |
| 17:52.01 | jontow | nah, not really |
| 17:52.07 | doolph | then |
| 17:52.28 | Zeeek | why not copy the file and see if that works? |
| 17:52.30 | doolph | why it keep ask me res_mysql.conf when it doesnt exists |
| 17:52.38 | doolph | yes that is what i am doing |
| 17:52.40 | gbdrbob | whats the situation re fixing bugs in the stable branch - I've found a bug in res_monitor that is fixed in cvs head but not in stable - the bugfix was merged to cvs on 21/01/05 and there have been two updates to the stable branch since then, but neither has carried the bugfix over... |
| 17:52.43 | Zeeek | or create a blank one? |
| 17:58.20 | cjk | kapejod: sorry for not answering, i mean monitoring, alerting reacting when asterisk is down or not accepting any new channels |
| 17:59.23 | Juggie | gbdrbob. you can contact drumkilla, or post a patch for the latest cvs-stable and let drumkilla check it and merge it |
| 17:59.30 | Juggie | as long as its just a fix and not a feature, it should be fine |
| 18:00.44 | doolph | omg |
| 18:00.52 | doolph | now I cant start asterisk |
| 18:03.32 | doolph | mmm |
| 18:03.41 | doolph | the ssh is off now |
| 18:03.49 | jontow | you broke it :( |
| 18:04.05 | doolph | argh |
| 18:04.10 | doolph | what happen |
| 18:04.25 | doolph | ah cool it is back now |
| 18:04.32 | doolph | it scared me off |
| 18:04.33 | kapejod | cjk: just send some calls from another * (or iax2) into an extension which will send back a dtmf sequence |
| 18:06.26 | doolph | mmm |
| 18:06.32 | doolph | how do i disable realtime |
| 18:07.35 | *** join/#asterisk jj505 (~jeffj@h-68-167-224-76.phndaz91.dynamic.covad.net) |
| 18:09.01 | *** join/#asterisk mog_home (~mogorman@user-24-236-84-48.knology.net) |
| 18:11.00 | ariel_ | doolph, I guess you can do a noload on the realtime app. But I don't have any system with cvs head so I don't know which one it is. |
| 18:11.18 | ariel_ | but here is info on the realtime from the wiki: http://www.voip-info.org/tiki-index.php?page=Asterisk%20RealTime |
| 18:11.47 | ariel_ | and also for testing your asterisk start you can do instead of safe_asterisk asterisk -vvvvvgc |
| 18:12.03 | mes | is anyone familiar with chan_phone? |
| 18:16.49 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@bzq-218-62-72.cablep.bezeqint.net) |
| 18:17.31 | *** join/#asterisk newmedian (~crowlther@Quebec-HSE-ppp230300.qc.sympatico.ca) |
| 18:19.30 | PoWeRKiLL | hi |
| 18:19.46 | cjk | kapejod: ok thats not the best metho |
| 18:20.37 | *** join/#asterisk Gunnar (~gunnar@34.80-203-209.nextgentel.com) |
| 18:21.47 | kapejod | cjk: well, you then know that it is working |
| 18:23.02 | *** part/#asterisk HeppyCat (~unknown@cpe-24-164-217-41.jam.res.rr.com) |
| 18:26.25 | newmedian | Realtime vs http://svn.asteriskdocs.org/res_data/ast_data/ ? different? same? |
| 18:26.43 | *** part/#asterisk santiago (~santiago@63.245.86.227) |
| 18:26.44 | Juggie | different |
| 18:26.46 | Juggie | much |
| 18:26.59 | newmedian | Juggie: opinion on which to use? |
| 18:29.29 | newmedian | ok, I get that. :) |
| 18:30.05 | Juggie | depends on if you want to use stable |
| 18:30.08 | Juggie | or head |
| 18:30.25 | ariel_ | I don't see a real need to have the info in a database instead of the extensions.conf |
| 18:30.30 | *** join/#asterisk pimpwell (~pimpwell@ool-44c6ab45.dyn.optonline.net) |
| 18:30.38 | pimpwell | any asterisk engineers in NY? |
| 18:30.48 | newmedian | i.e. you might use ast_data if you were using stable, but once you go head you'd want Realtime and there's not real transition path? |
| 18:31.08 | Juggie | ast_data is close to using realtime static |
| 18:31.20 | Juggie | i'm not 100% |
| 18:31.52 | Juggie | i dont like the database design astdata uses, or realtime static for that matter |
| 18:32.41 | newmedian | Juggie: thanks. If you were using Realtime, you could theoretically share a MySQL database between multiple Asterisks performing redundancy? |
| 18:34.23 | ariel_ | newmedian, you can do that with out using a database |
| 18:35.54 | ariel_ | get two asterisk boxes set them up with CentOS 3.4 format them with QFS file system and then add hartbeat for the failover |
| 18:36.55 | newmedian | ariel: have you implemented that in the field? Aside from the config sharing issue, how does the redundant box take over the PRI(s) |
| 18:38.02 | ariel_ | newmedian, they don't it depends on how you set that part up. I have set this up for a telco in California. But they have an TNT that is brining in the PRI's into there system. |
| 18:38.52 | kapejod | newmedian: you might use one of these: http://www.sboost.org/~kapejod/ISDNguard.pdf ;-) |
| 18:40.52 | newmedian | kapejod: interesting |
| 18:40.53 | jontow | pimpwell; depends on what you call an engineer, and where you're talking about in NY.. its a big state.. more than most people understand. |
| 18:41.22 | newmedian | ariel: do you recall what TNT they are using? |
| 18:42.01 | ariel_ | newmedian, not off the top of my head. I can find out on monday. I have more work to do for them on monday. |
| 18:42.20 | newmedian | ariel: if you remember, that would be great, thanks. :) |
| 18:43.41 | newmedian | kapejod: (while this may be a moot question) how well does it work in practice? How to ensure that dead Asterisk isn't still sending heartbeat as the last remaining gasp task still alive. ;) |
| 18:45.12 | *** join/#asterisk sitelco (~sitelco@201.138.151.246) |
| 18:45.16 | ariel_ | newmedian, sounds like your wanting to put up your own voip service. |
| 18:45.18 | sitelco | hi all |
| 18:45.19 | kapejod | newmedian: if * still sends the hearbeat then userspace is still alive. |
| 18:45.31 | sitelco | i have a little question about digium cards |
| 18:45.36 | kapejod | newmedian: if userspace is still alive you can shutdown asterisk |
| 18:46.15 | sitelco | i want to buy a tdm400p but i don`t what kind of pci slot i need to this card |
| 18:46.16 | ariel_ | newmedian, if your system reports an failure you can have a script to shut it down. That way hartbeat works. |
| 18:46.18 | sitelco | ? |
| 18:46.36 | ariel_ | sitelco, 32bit normal pci 2.2 version |
| 18:46.51 | ariel_ | it works on 5 and 3 volt systems |
| 18:46.58 | sitelco | ah ok |
| 18:47.03 | sitelco | that is my doubt |
| 18:47.20 | sitelco | exist a recommend motherbord |
| 18:47.36 | Qwell | anything less then 10 years old should do |
| 18:48.12 | ariel_ | sitelco, I actually have been using Asus mb on clones and the rest I have been using Dell SC420 servers. Cheap and there working for me. |
| 18:48.16 | newmedian | ariel: quite possibly; Interested to see how VOIP regulation ugliness in Canada will work out. |
| 18:48.30 | *** join/#asterisk danalien (~danalien@danalien.user) |
| 18:49.00 | ariel_ | newmedian, it's ugly every where. |
| 18:49.07 | newmedian | I'm also in the R&D phase right now; Info on competitors offering PRIs is starting to filter in. Starting to work up the spreadsheets etc. |
| 18:49.40 | sitelco | the dell sc420 works? because in digium page say that system have some problems with asterisk |
| 18:50.26 | ariel_ | sitelco, I saw that. But I have about 4 customers with them on has 3 TDM boards one has 2 and the other just one all working just fine. But there with only one sata drive. |
| 18:50.52 | newmedian | Of course, if you/anyone has real-world info that they could pass along, e.g. for VOIP->PSTN and PSTN->VOIP scenarios, the PRI(s):Bandwidth. Yes, I know about generically figuring out which codec and then extrapolating bandwidth, but would be nice to hear some in-the-field info. |
| 18:50.55 | ariel_ | The problems I have seen with people is when they start adding raid. I also don't turn on hypertheradint. |
| 18:51.04 | *** join/#asterisk gdsm (~gdsm@mk-ns500-1.uk.tiscali.com) |
| 18:51.05 | ariel_ | hyper thearding... |
| 18:51.31 | sitelco | ariel in this case what kind of motherbord asus you recommed me to buy |
| 18:51.34 | sitelco | ? |
| 18:51.55 | ariel_ | Just a sec. |
| 18:51.57 | Zeeek | Thank You, NuFone! |
| 18:51.58 | sitelco | ok |
| 18:53.32 | Zeeek | my wife is travelling alone in the USA and thank to NuFone, she has an easy way to call me from anywhere at all: private homes, airports, bars... wait a minute! |
| 18:53.50 | Zeeek | what was that music I heard in the background? |
| 18:54.14 | ariel_ | sitelco, last one I used was VIA P4X533 |
| 18:54.30 | Qwell | Zeeek: Got busted for being in a bar? :p |
| 18:54.39 | Qwell | erm, she was in a bar? |
| 18:54.39 | Zeeek | I'm gonna kill her! |
| 18:54.42 | ariel_ | Zeeek, hehehe |
| 18:54.51 | Zeeek | NuFone will be a the trial |
| 18:55.01 | Zeeek | Crime de Passion (tm) |
| 18:55.06 | ariel_ | trust |
| 18:55.19 | Qwell | Zeeek: send JerJer a testimonial to put on the site. :P |
| 18:55.23 | Zeeek | no really, those toll-free DIDs rock! |
| 18:55.40 | Zeeek | How NuFone helped my divorce in twenty words or less |
| 18:55.56 | ariel_ | Zeeek, if they were able to get me one I would use them. But they have been out for weeks. |
| 18:55.57 | Zeeek | hey as long as she does call... |
| 18:56.16 | Zeeek | really? I did see they weren't giving them last week |
| 18:56.33 | Zeeek | there must be an exponential rise in demand for all providers |
| 18:56.44 | Zeeek | this shit is coming to the attention of the public now |
| 18:56.48 | ariel_ | I needed 2 for a customer I had to go with (argh livevoip) |
| 18:57.16 | Zeeek | I got one of those to test |
| 18:57.22 | Zeeek | at least I think I did |
| 18:57.36 | Zeeek | no wait, their site didn't work right and I didn't |
| 18:58.20 | ariel_ | livevoip and others have problems picking up there phones. Support from them at this time sucks. |
| 18:59.13 | sitelco | ariel waht is the brand of the motherboard do you use |
| 18:59.36 | Zeeek | ariel they are also having credit card gateway probs. |
| 19:00.15 | Zeeek | My order went through but the charge did not (an address difference on CC, I think) A lot of these sales web sites screw up the transaction |
| 19:00.18 | ariel_ | the one I posted was from asus |
| 19:00.44 | Zeeek | I'm running an old asus in production right now |
| 19:01.38 | Zeeek | CUSL2C |
| 19:01.47 | ariel_ | I wonder why digium is having so much trouble with dell systems. So far the ones I have are all up and running without any issue's. |
| 19:01.58 | ariel_ | Zeeek, I have my desktop with that mb.... |
| 19:02.00 | Zeeek | they must have some oddities |
| 19:02.08 | Juggie | i have two dell systems no problem |
| 19:02.25 | Zeeek | ariel_ I retired it from my own box when I could get asterisk running on the old PII |
| 19:02.30 | ariel_ | Juggie, I have about 8 of them out there and there all working jsut fine. |
| 19:03.17 | ariel_ | Zeeek, I have not used my desktop in over 6 months. I have been working from my laptop mainly now. |
| 19:03.46 | ariel_ | Just take it with me and off I go. When I come back home I sync it to my server and my data is updated. |
| 19:03.47 | dca_ | ariel_, tried teliax? they have support |
| 19:04.04 | ariel_ | I tried them could not get them on the phone last week either. |
| 19:04.25 | dca_ | seems like ever since broadvoice went down all the other providers have been getting slammed |
| 19:05.12 | *** join/#asterisk wildgoose (~edward@xunil.mail.wildgooses.com) |
| 19:05.14 | dca_ | anyone ever run asterisk on a Mac? |
| 19:05.25 | dca_ | like the G5 or X5 or whatever it is? |
| 19:05.44 | ariel_ | for my home I have been using race.com and I still have a did from voicepulse that has been working for over a year. |
| 19:06.15 | Zeeek | dca_ I've seen people in FWD forum that did |
| 19:06.58 | ariel_ | it's hit or miss these pass few weeks for most of the voip providers. I feel there is a major shake down coming soon. |
| 19:07.09 | Zeeek | the shaking began a while ago |
| 19:07.24 | Juggie | its much easier if you have your own pri's :) |
| 19:08.14 | ariel_ | Juggie, some of us can't afford pri's |
| 19:08.36 | Juggie | ariel_, i dont pay for them either :) |
| 19:11.33 | newmedian | The QuadBRI only works in Europe? http://www.junghanns.net/asterisk/page17.html oops kapejod has gone |
| 19:12.26 | *** join/#asterisk bofh42 (~bofh42@pD9EC0C8D.dip0.t-ipconnect.de) |
| 19:22.49 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
| 19:24.36 | *** join/#asterisk jeffik (~jeffik@CPE00080da2f8fc-CM0011aec8aa0a.cpe.net.cable.rogers.com) |
| 19:25.04 | *** join/#asterisk Deryl (pgpkeys@static-141-149-128-140.buff.east.verizon.net) |
| 19:27.24 | darwin35 | ManX you around |
| 19:29.49 | Zeeek | look in -stable |
| 19:30.49 | Blackvel | newmedian: its isdn |
| 19:30.59 | jeffik | hello all |
| 19:31.01 | Blackvel | newmedian: do you have isdn? |
| 19:31.57 | [Lamer] | Hi folks, how do I switch to use the Realtime Config? |
| 19:32.23 | jeffik | need macro to access voicemail from outside by dialing # |
| 19:32.26 | newmedian | Blackvel: I don't currently have any ISDN BRI, no, but I was looking at that and thinking of SOHOs that want digital but don't need the capacity for a PRI. In the PDF it says "The drivers can handle EuroISDN (ETS 300 102) signalling, support for National ISDN 1 (Q.931) is planned." |
| 19:33.37 | [Lamer] | Now I have asterisk-addons installed and res_mysql.conf,extconfig.conf setup |
| 19:33.44 | Blackvel | hm good question |
| 19:34.01 | Blackvel | never used national isdn nor do I know where the differences are |
| 19:34.22 | [Lamer] | anyone using Realtime with mysql? |
| 19:35.30 | Moc | forget about BRI in general, they will provide it, but very limited in features |
| 19:35.37 | Moc | I mean in Canada |
| 19:36.05 | newmedian | Moc: can you recommend a less expensive provider than Bell for PRIs where service != bad? |
| 19:36.07 | Moc | they dont allow to put more than 2 did on a BRI (1 per B-Channel) and it a static outgoing caller id |
| 19:36.18 | newmedian | Moc: useful information |
| 19:36.24 | tzanger | yeah NI2 here too |
| 19:36.32 | Moc | newmedian, allstream, gt, telus.. |
| 19:36.44 | Moc | bell is the most expensive of them all |
| 19:37.05 | tzanger | you can't get more than two DID on a MicroLink? |
| 19:37.19 | [Lamer] | How do I make a switch to use Realtime data in mysql? |
| 19:37.37 | Moc | on Bell yes you can get a few, but they are so expensive |
| 19:37.42 | shido6 | if I resold my PRI's and added a few blocks of hundred cheese burgers I would be cheaper than bell |
| 19:37.58 | Moc | tzanger, MicroLink = BRI, MegaLink = PRI ... I hate Bell for rebranding their stuff |
| 19:38.00 | gbdrbob | BRI can have as many did,msn as you want but many telcos won't provide it |
| 19:38.00 | file | mmm... cheese burgers |
| 19:38.12 | tzanger | yes I know microlink is BRI, that's what I asked... they don't allow more than 2 DID on a BRI? |
| 19:38.20 | [Lamer] | I set all up following http://www.voip-info.org/wiki-Asterisk+RealTime but it still reads what I put in sip.conf, extensions.conf,... |
| 19:38.40 | Moc | newmedian best solution is find a voip provider in your area |
| 19:39.14 | gbdrbob | I'd really like to use BT's home bri serivce here in the UK but they won't even provide caller id on the digital ports! |
| 19:39.20 | [Lamer] | Hi guys, anyone using Realtime config from database? |
| 19:39.21 | newmedian | Moc: or become a provider for SOHOs. hmm. |
| 19:39.42 | Moc | newmedian, I use unlimitel and Im happy... |
| 19:40.10 | [Lamer] | Hi folks, anyone using mysql to store config files? |
| 19:40.53 | Nugget | nobody new since you asked a minute ago, no. |
| 19:41.27 | pussfeller | are the table names hard coded into the mysql addin |
| 19:41.29 | [Lamer] | Nugget: thanks, at least you can hear me =) |
| 19:42.25 | [Lamer] | pussfeller: yes, and I added them to extconfig.conf |
| 19:42.31 | newmedian | Moc: I've been eyeing unlimitel. Which would be an alternative to becoming a provider for SOHOs, but if a SOHO was migrating from an existing POTS to get the advantage of digital, then aren't the per minute charges a conceptual/psychological barrier? Then they start thinking of every call being like long distance. |
| 19:42.58 | pussfeller | [Lamer], sorry, i was asking if in the addin I can set my own table names, or do I have to hack some c source |
| 19:43.01 | [Lamer] | pussfeller: like "extensions => mysql,asterisk" |
| 19:43.16 | Moc | well it all depend of the numbers of minutes your doing |
| 19:43.40 | Moc | if I do 10000 min/month, then unlimitel is less expensive than PRI |
| 19:43.43 | Moc | alot less.. |
| 19:44.00 | Moc | and it a question of ratio of usage of the time.. |
| 19:44.04 | Moc | time = line |
| 19:44.22 | [Lamer] | pussfeller: and in the extensions table I have "| 2 | local-extensions | 444 | 2 | SayNumber | 102 |" |
| 19:44.23 | newmedian | Moc: but if you're being the provider, then aren't you spreading the cost of the PRI against multiple SOHOs who only need x (say 1-4) lines? |
| 19:45.00 | *** join/#asterisk sitelco (~sitelco@201.138.151.246) |
| 19:45.01 | [Lamer] | pussfeller: that's fine :-) |
| 19:45.03 | *** join/#asterisk sandnigg0r (~niggerplz@66-55-197-254.gwi.net) |
| 19:45.23 | Moc | yes, that the idea, but it take a few T1 to get a good client/line ratio |
| 19:45.28 | Moc | depending of your usage.. |
| 19:45.43 | [Lamer] | pussfeller: I don't know if anyone will answer your question |
| 19:45.52 | sitelco | is posible asterisk have mfc/R2 E1 |
| 19:46.19 | [Lamer] | pussfeller: I've tried asking some questions for over 12 hrs now, hoping someone will answer |
| 19:46.36 | Moc | realtime aint very popular in general.. |
| 19:46.41 | [Lamer] | sitelco; yes, it is |
| 19:46.42 | Moc | only in 'big' implementation |
| 19:46.50 | newmedian | Moc: there's also the issue of lack of control. There seems to be some volatility in the providers in this space at the moment. If you're connecting SOHOs up to one of these providers and they go unreliable or bust, the portability of the number is an issue as well as the reliability of the service (and customer service). |
| 19:46.58 | [Lamer] | sitelco: use unicall library |
| 19:47.24 | Moc | newmedian, yep, Ive lost DID because of crappy provider... that resold Broadbox |
| 19:47.38 | [Lamer] | sitelco: E1 CAS with MFC/R2 |
| 19:47.52 | tzanger | ohhhh dear |
| 19:47.55 | tzanger | When is bedtime at Neverland Ranch? When the big hand touches the little hand. |
| 19:48.00 | Moc | I have a good relation with unlimitel, you talk directy to the guy who manage this so .. no middle man |
| 19:48.17 | Zeeek | tzanger eewwwwww |
| 19:48.21 | newmedian | Lamer/pussfeller: Sometimes it's a good day for questions on #asterisk, sometimes it's not a good day. It's not exactly a "paid support" channel. ;) |
| 19:48.52 | PatrickDK | this isn't a support channel |
| 19:48.58 | PatrickDK | it's just what we normally do |
| 19:48.59 | Zeeek | it isn't? Shit! |
| 19:49.15 | Zeeek | oh I thought you meant support hose |
| 19:49.37 | jeffik | anyone, access voice mail from outside |
| 19:49.49 | PatrickDK | jeffik, easy, press * |
| 19:49.49 | Zeeek | jeffik what d'ya need? |
| 19:49.59 | Zeeek | outside as in call in on a phone? |
| 19:50.18 | jeffik | when i press *i get 'welcome to the company directory' |
| 19:50.30 | Zeeek | you suing @home or something? |
| 19:50.34 | PatrickDK | jeff, guess you didn't set that up in the extentions file yet |
| 19:50.35 | jeffik | ys |
| 19:50.41 | Zeeek | oh. |
| 19:50.47 | jeffik | yes @home |
| 19:50.52 | Zeeek | @hole? |
| 19:51.06 | newmedian | jeffik, do you want just to access voicemail when you're remote, or do you want more functionality (like making calls through your * outbound on SIP/IAX2)? |
| 19:51.07 | jeffik | ok ok |
| 19:51.23 | Zeeek | see this is the problem. You need to know what extensions are |
| 19:51.41 | Zeeek | so @hole sets you up for a disappointment |
| 19:51.56 | Zeeek | it's great as long as you don't need to do anything else |
| 19:51.57 | jeffik | i want to dial my did and while the announcment is playing press * and have it say 'comedian mail, mail box, passowrd |
| 19:52.06 | Zeeek | read about the dialplan |
| 19:52.13 | newmedian | @home was useful to see a complete "working" setup and the resulting config files. Useful as a jump start to learn Asterisk, but not for Production use. |
| 19:52.18 | Zeeek | they say the wiki is good for this |
| 19:52.37 | Zeeek | newmedian yeah, exactly, then delete it all and begin again |
| 19:52.40 | shido6 | or hiring a consultant |
| 19:52.46 | jeffik | i looked there |
| 19:52.51 | Zeeek | and? |
| 19:53.01 | Zeeek | didn't find the five pages about dialplans? |
| 19:53.34 | jeffik | didn't find this specific i'll look in dial plan |
| 19:53.37 | newmedian | jeffik, did you ask in the Asterisk@Home forums? http://sourceforge.net/forum/forum.php?forum_id=420324 |
| 19:54.00 | jeffik | will go three next thanks |
| 19:55.38 | *** join/#asterisk anderiv (~anderiv@207-67-87-34.gen.twtelecom.net) |
| 20:01.22 | *** join/#asterisk |Vulture| (~V@199.227.253.212) |
| 20:01.34 | |Vulture| | anyone here use PRI --> FXS --> Fax Machine? |
| 20:03.30 | docelmo | I use FXO -> FSX -> Fax |
| 20:03.36 | Juggie | i dont, but it should work fine. |
| 20:03.51 | *** join/#asterisk easimon (~easimon@localhorst.kawo2.RWTH-Aachen.DE) |
| 20:03.55 | docelmo | But at my office we have a couple stations setup like that |
| 20:03.55 | *** join/#asterisk jdg (~jdg@CA03F957.adsl.mana.pf) |
| 20:04.38 | docelmo | VOIP -> Cisco -> CAS PRI -> NEC PBX -> FAX |
| 20:05.27 | |Vulture| | they are not sending.. didn't know if I need to set something special like echocancel etc. |
| 20:05.40 | |Vulture| | I turned off echo cancel on the PRI and the FXS |
| 20:06.01 | docelmo | are you doing PRI -> * -> FXS? |
| 20:06.11 | *** part/#asterisk jdg (~jdg@CA03F957.adsl.mana.pf) |
| 20:07.30 | |Vulture| | yes |
| 20:09.21 | *** join/#asterisk knigma-h (~mkn@shrewd.pub.knigma.org) |
| 20:10.21 | knigma-h | I'm trying to use meetme with just the ztdummy driver, and every time I join the conference I get "channel.c:1937 ast_request: No channel type registered for 'zap'" and "app_meetme.c:227 build_conf: Unable to open pseudo channel - trying device". Any ideas please? |
| 20:13.46 | *** join/#asterisk zeedo (~zeedo@www.reboot-robot.net) |
| 20:14.56 | *** join/#asterisk zeedo (~zeedo@www.reboot-robot.net) |
| 20:16.02 | *** part/#asterisk pimpwell (~pimpwell@ool-44c6ab45.dyn.optonline.net) |
| 20:17.27 | |Vulture| | anyone? |
| 20:18.19 | *** join/#asterisk DrJolo (~chatzilla@217.153.194.10) |
| 20:19.18 | tzafrir_laptop | knight_, is the module ztdummy loaded? |
| 20:19.35 | tzafrir_laptop | lsmod |grep ztdummy |
| 20:19.37 | *** join/#asterisk tld (~tld@42.80-203-178.nextgentel.com) |
| 20:20.27 | tzafrir_laptop | knigma-h, basically it seems like chan_zap is not loaded, right. |
| 20:20.42 | tzafrir_laptop | try: load chan_zap.so from the asterisk cli |
| 20:25.41 | knigma-h | right, I'll try that - thanks |
| 20:25.50 | knigma-h | yes, ztdummy is loaded |
| 20:27.23 | knigma-h | load chan_zap.so has silenced the warnings |
| 20:27.32 | knigma-h | cool - thanks. How do I make it load itself? |
| 20:28.19 | knigma-h | perhaps modules.conf shouldn't say "noload => chan_zap.so" ;) |
| 20:29.44 | knigma-h | maybe not: /usr/local/lib/asterisk/modules/chan_zap.so: Undefined symbol "ast_retrieve_call_to_death" |
| 20:30.28 | shido6 | err |
| 20:30.33 | shido6 | whats lsmod say |
| 20:30.37 | shido6 | whats ztcfg -vv say? |
| 20:31.02 | knigma-h | Notice: Configuration file is /usr/local/etc/zaptel.conf (I have not real hardware) |
| 20:31.12 | knigma-h | 10 2 0xc1972000 30000 zaptel.ko |
| 20:31.13 | JerJer | has someone updated zaptel lately? |
| 20:31.18 | knigma-h | 11 1 0xc19a2000 2000 ztdummy.ko |
| 20:32.07 | ariel_ | seems like the user list is down. I have not gotten any emails through for about an hour. |
| 20:32.34 | file | slow day |
| 20:32.54 | *** join/#asterisk edguy3 (~edguy@host-24-225-213-218.patmedia.net) |
| 20:33.32 | denon | JerJer: russell did on 5/11 |
| 20:33.42 | denon | bug #4051 |
| 20:33.42 | Zeeek | ²²²²ariel_ why not post a question about BV being down to check? |
| 20:33.46 | denon | that's the last thing I see |
| 20:34.17 | doolph | hello |
| 20:34.33 | doolph | anyone can help me, check http://pastebin.com/284314 pls |
| 20:34.36 | knigma-h | adding load => res_features.so seems to have sorted it - thanks for the pointers!! |
| 20:37.25 | doolph | argh |
| 20:38.02 | |Vulture| | ariel_: do you do PRI --> * --> fax? |
| 20:41.05 | Blackvel | ariel_: do you have BV problem? |
| 20:41.20 | |Vulture| | BV has been giving me nothing but crap the past week |
| 20:41.32 | ariel_ | |Vulture|, yes I have a few of them up |
| 20:41.40 | ariel_ | Blackvel, no I don't use BV |
| 20:41.46 | |Vulture| | ariel_: and it works fine? mine keeps giving errors |
| 20:41.53 | ariel_ | yes |
| 20:42.07 | |Vulture| | ariel_: do you have echo cancel turned off? |
| 20:42.23 | |Vulture| | or build your zconfig any special way? |
| 20:45.36 | doolph | ariel |
| 20:46.48 | *** join/#asterisk jets (~brian@xyharley.dsl.xmission.com) |
| 20:47.11 | jets | Do all of the B-channels restart when a PRI is idle every once in awihle |
| 20:47.20 | jets | B-channel 0/15 successfully restarted on span 3 |
| 20:47.25 | |Vulture| | jets: I get that too |
| 20:47.52 | |Vulture| | but doesn't cause a problem since it does it when idle |
| 20:48.03 | |Vulture| | I think its more of a ping for the D channel |
| 20:48.15 | jets | okay we had a ton of problems with dropped calls and before i og live again i'm making sure the pri's are happy and everything is normal |
| 20:48.20 | jets | :) |
| 20:48.30 | |Vulture| | jets: do a zttest |
| 20:48.43 | |Vulture| | make sure your lowest isn't below 99.85 |
| 20:49.31 | jets | i've been doing that and i made sure it has its own irq and /proc/interrupts has interrupts to the card |
| 20:50.15 | file | Oh Yeah! |
| 20:50.35 | jets | now i'll just fake calls between a bunch of pri's and go from there |
| 20:51.46 | |Vulture| | I am just having problems with faxes :( |
| 20:51.56 | |Vulture| | tyring to go back to STABLE and see if that fixes it |
| 20:53.13 | ariel_ | |Vulture|, what are you doing inbound via pri then to what device? |
| 20:53.48 | |Vulture| | Zap/26 |
| 20:53.59 | |Vulture| | FXS that connects to fax machine |
| 20:54.38 | ariel_ | ok on the fxs do you have echo turn off? |
| 20:55.03 | |Vulture| | no I didn't here is my conf |
| 20:55.26 | |Vulture| | http://pastebin.ca/11749 |
| 20:55.29 | ariel_ | I had to turn off all echo cancelation on my zap ports for fax to work correctly. |
| 20:55.45 | Zeeek | FXO too ? |
| 20:55.54 | *** join/#asterisk tekati (~captain@cpe-66-75-215-63.bak.res.rr.com) |
| 20:55.58 | |Vulture| | yea, the FXO is for the door thingie |
| 20:56.20 | Zeeek | no ariel_ I meant? |
| 20:56.20 | |Vulture| | you push a button, it calls the system, then you can hit * to open the door |
| 20:56.28 | |Vulture| | oh okay |
| 20:56.28 | *** join/#asterisk Sedorox (brandon@sedorox.staff.smartserv) |
| 20:56.33 | ariel_ | |Vulture|, what device are you using for the *? |
| 20:56.53 | |Vulture| | for the PRI or the FXS? |
| 20:57.05 | Zeeek | funny, I was in a Hilton hotel last week and their room phone sucked! Sounded worse than my cheapest IAX hardphone |
| 20:57.18 | ariel_ | your door phone. I am using viking door phones but I need to find something to open the srike plate |
| 20:57.27 | Zeeek | so bad people asked "you on that asterix thingie again?" |
| 20:57.28 | |Vulture| | lemme get the name 1s |
| 20:58.08 | |Vulture| | http://shop.store.yahoo.com/asihome/doorfonfroma.html |
| 20:58.43 | |Vulture| | works well |
| 20:58.49 | ariel_ | your settings for your zap port looks good like mine. |
| 20:59.17 | |Vulture| | just need the FXS set to echocancel=no right? |
| 20:59.43 | |Vulture| | I think it may be a head problem, it worked like 4 days ago when I was testing the PRI |
| 21:00.01 | |Vulture| | but then today when I installed it to the fax machine it keeps erroring |
| 21:00.01 | ariel_ | yes |
| 21:00.01 | PoWeRKiLL | anyone find a good driver to make tiff file under windows ? |
| 21:00.09 | ariel_ | hummm |
| 21:00.28 | PoWeRKiLL | ma amatsav ariel_ ? |
| 21:00.30 | ariel_ | I have my fax machine at my new office and it's not connected yet. So can't help you test it. |
| 21:01.38 | Zeeek | PoWeRKiLL are you in Paris? |
| 21:01.47 | |Vulture| | the heck... with 1.0.7 zaptel drivers my PRI card isn't coming online |
| 21:01.58 | |Vulture| | no lights, but ttool displays no alarms |
| 21:02.00 | |Vulture| | and sees the car |
| 21:02.01 | |Vulture| | d |
| 21:02.15 | PoWeRKiLL | Zeeek actuellement non |
| 21:02.25 | ManxPower | My SPA-2100 just does NOT want to talk G726 |
| 21:02.28 | Zeeek | you came over to our office, right? |
| 21:02.43 | Zeeek | I see your pseudo oce in a while but haven't caught you - how are things? |
| 21:02.47 | PoWeRKiLL | I will be in Paris on wed |
| 21:03.00 | *** join/#asterisk zeedo (~zeedo@www.reboot-robot.net) |
| 21:03.17 | PoWeRKiLL | Zeeek where are you from ? |
| 21:03.34 | Zeeek | minneapolis, MN originally - but didn't we meet ? |
| 21:04.19 | *** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net) |
| 21:06.16 | Sedorox | Can anyone give me feedback on Teliax? |
| 21:06.25 | |Vulture| | ooo with 1.0.7 looks like its sending.... |
| 21:06.26 | Sedorox | like.. does it go down often, how's service... |
| 21:06.54 | |Vulture| | looks like zaptel HEAD has a little bug for faxing |
| 21:07.49 | darwin35 | when using #include in a file is it #include " " or #include = ? |
| 21:08.01 | Sedorox | #include <iostream> |
| 21:08.22 | Sedorox | its either <> or "" |
| 21:08.23 | darwin35 | so you us <> |
| 21:08.27 | darwin35 | ok |
| 21:08.31 | Sedorox | I wanna say <> |
| 21:10.05 | ariel_ | "" |
| 21:10.21 | |Vulture| | ariel_: do you notice that faxing going through * is a lot slower? |
| 21:10.25 | ariel_ | ManxPower, does it do other codec's |
| 21:10.40 | ariel_ | |Vulture|, never paid attension to it. |
| 21:10.44 | |Vulture| | kk |
| 21:11.03 | ariel_ | I figured that once I had it working I did not care |
| 21:11.10 | |Vulture| | hahaha good idea |
| 21:11.26 | |Vulture| | if it goes fax --> PRI --> spandsp its fast as anything |
| 21:11.31 | file | You should all come to the PBX Developer's Conference, Cluecon, see http://www.cluecon.com/ for details! |
| 21:11.44 | ariel_ | in most of my setups SOHO there is always a fax machine. It's been working fine for my customers. |
| 21:12.02 | |Vulture| | ariel_: its working with 1.0.7 drivers now |
| 21:12.10 | |Vulture| | but I want my ChanSpy() :P |
| 21:12.14 | ariel_ | file, yes but some of us don't have the cash |
| 21:12.44 | file | find a way, or make a way! |
| 21:12.50 | file | to attend. |
| 21:13.05 | ariel_ | file, just trying to get more customers so I can. |
| 21:13.51 | ariel_ | This coming week I have another asterisk installation. |
| 21:14.11 | ariel_ | but it sure seems like I install them get them up and running and then they don't need me any more. |
| 21:14.22 | *** join/#asterisk _tekati_ (~captain@cpe-66-75-215-63.bak.res.rr.com) |
| 21:14.41 | ariel_ | it just works and other then maybe one or two changes a month there is no more calls to that customers. |
| 21:15.09 | file | :( |
| 21:15.17 | file | that's sad |
| 21:16.08 | ariel_ | These last 2 customers I also took over there Windows network so I am getting more work from them. But asterisk once it's setup correctly needs very little changes. |
| 21:16.11 | *** join/#asterisk citats (~james@duff.gnuinter.net) |
| 21:16.45 | file | so you're getting no recurring income? |
| 21:19.32 | ariel_ | some but not at level i need |
| 21:20.13 | *** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 21:22.08 | Chuji | ariel_ : You should try Asterisk hosting then |
| 21:22.08 | Chuji | ariel_ : Just resell a colo'd asterisk box |
| 21:22.15 | Chuji | ariel_ : Sell them dialtone and all |
| 21:22.36 | Chuji | It's not for all customers, but it sounds like you installation base is growing |
| 21:23.56 | |Vulture| | Chuji: so the customer just buys a T1 data? |
| 21:25.26 | *** join/#asterisk lattice (~lattice@S010600045ad57bb6.vc.shawcable.net) |
| 21:25.44 | Chuji | |Vulture| : Exactly, you resell them dialtone and internet over the same pipe |
| 21:26.18 | Chuji | Best suited for the small office, but it's a good solution |
| 21:26.40 | Blackvel | ok, i am sick of coding |
| 21:26.44 | ariel_ | actuall the install base is what is recommending the other offices so it's getting better. |
| 21:26.52 | Blackvel | time for a relaxing evening :) |
| 21:27.24 | Blackvel | was waiting for at least 1 hour for feedback from UK |
| 21:27.52 | Blackvel | now I don't want to wait any longer :) |
| 21:27.56 | Blackvel | I wish |
| 21:28.12 | Blackvel | at what time exactly? |
| 21:28.27 | ariel_ | Chuji, sounds like a good deal but most of the my customers don't want to depend on the internet for there service. Bellsouth is not that good down here. |
| 21:28.28 | Blackvel | takes me ages to get alive after shower and breakfast |
| 21:28.30 | FuriousGeorge | so where can i find an h323 gatekeeper? can i run my own with OSS (OpenH323?) |
| 21:29.00 | Chuji | ariel_ : No, you need to setup shop at a telco co/lo space |
| 21:29.16 | *** join/#asterisk rcam (~rcammobil@adsl-218-151-77.jax.bellsouth.net) |
| 21:29.19 | Chuji | ariel_ : You can get PRI(s) and internet, then resell it |
| 21:29.39 | Chuji | ariel_ : You would just provision a point 2 point from the colo to the customer |
| 21:29.42 | ariel_ | Chuji, yes I know that |
| 21:29.51 | Chuji | So they aren't dependent on the internet |
| 21:30.32 | ariel_ | Chuji, it's very expensive for t1 here and most of my customers are small and don't want to pay for that type of connection. |
| 21:30.39 | Blackvel | have a good day/evening/night :) |
| 21:30.47 | ariel_ | goodnight Blackvel |
| 21:31.01 | Chuji | Ahh, point to points are like 250 here |
| 21:31.46 | ariel_ | Here they start depending on distance 228.00 plus tax. And most of the ones I have looked at cost around 300 or more. |
| 21:32.33 | ariel_ | Chuji, I am about 15 to 20 miles outside the city limites of Miami so it's more then within the city. |
| 21:33.01 | Chuji | They vary around here too. If it's bell south all the way, they are high |
| 21:33.08 | Chuji | but if you can get XO, they are cheaper |
| 21:34.10 | ariel_ | xo depends on bs for there last mile. |
| 21:34.41 | ariel_ | But in any case I don't have the funds to start up my own pri's and a colo spot. |
| 21:35.23 | *** join/#asterisk kb1_kanobe (~krisbouti@h24-207-80-55.cst.dccnet.com) |
| 21:35.25 | *** join/#asterisk Trickyphillips (~Trickyphi@adsl-68-121-21-99.dsl.irvnca.pacbell.net) |
| 21:38.51 | *** join/#asterisk voiper (~none@pcp0010158348pcs.eatntn01.nj.comcast.net) |
| 21:39.19 | FuriousGeorge | lol, my box just said "Yuck! Error in buffer handling" |
| 21:40.12 | FuriousGeorge | how can the devel's expect me to take it serriously when my bix says yuck |
| 21:40.16 | doolph | http://pastebin.com/284314 |
| 21:40.18 | doolph | anyone? |
| 21:41.20 | *** join/#asterisk whachagonnado (~hallhouse@66.219.228.65) |
| 21:41.48 | FuriousGeorge | doolph: i dont know much about anything (im new), but i used to get segfaults when i was assuming gentoo made the linux26 modules for me, and it didnt (or didnt make thm right) did you make linux26? |
| 21:42.20 | FuriousGeorge | that is, if u are using a 2.6 kernel |
| 21:43.09 | FuriousGeorge | chuji, hate to evesdrop ;) but what is a co/lo? collaborative location? |
| 21:43.19 | file | colocation |
| 21:43.26 | FuriousGeorge | is it like... |
| 21:43.37 | file | it's a place where you can put machines/equipment |
| 21:43.47 | FuriousGeorge | you get a server with your telco (verizon in my case) and they allow you to resell bandwidth |
| 21:44.03 | doolph | FuriousGeorge nope, im on 2.4 |
| 21:44.13 | FuriousGeorge | sorry doolph, thats all i got :( |
| 21:44.19 | voiper | anyone uses oh323 here ? |
| 21:44.20 | file | doolph: have you tried starting asterisk in console and seeing where it barfs? ie: asterisk -vvvvc |
| 21:44.34 | file | FuriousGeorge: http://en.wikipedia.org/wiki/Colocation -> read that |
| 21:44.45 | whachagonnado | Hello, I am just starting to read the handbook and planning to install Asterisk some time in the next week or two. I was wondering, and I imagine this may be a can of worms, what distro to use. Which is most stable and or has the best/easiest package manager support. |
| 21:44.45 | whachagonnado | Does anyone have any input? |
| 21:44.47 | FuriousGeorge | voiper: no, but i can make it stop trying to register with gatekeeper ;) |
| 21:44.53 | doolph | file |
| 21:44.53 | doolph | [res_config_mysql.so] => (MySQL RealTime Configuration Driver) |
| 21:44.53 | doolph | == Parsing '/etc/asterisk/res_mysql.conf': Found |
| 21:44.53 | doolph | Ouch ... error while writing audio data: : Broken pipe |
| 21:44.53 | doolph | Segmentation fault |
| 21:44.53 | doolph | [root@www tmp]# |
| 21:44.57 | voiper | :) |
| 21:45.03 | file | doolph: if you paste something like that again I'll hurt you :) |
| 21:45.08 | doolph | sorry |
| 21:45.14 | doolph | mmm writing audio data |
| 21:45.17 | file | doolph: sounds like a bug in res_config_mysql, btw: it wasn't able to connect |
| 21:45.25 | file | so that might be why |
| 21:45.31 | doolph | what do you suggest |
| 21:45.39 | file | disable loading of it? |
| 21:45.41 | file | noload => res_config |
| 21:45.42 | doolph | how |
| 21:45.43 | file | gah |
| 21:45.45 | file | noload => res_config_mysql.so |
| 21:45.46 | file | in modules.conf |
| 21:45.52 | file | and I suggest reading documentation |
| 21:46.05 | shido6 | file |
| 21:46.07 | shido6 | pedo |
| 21:46.10 | doolph | umm my things maybe not work if i dissable that |
| 21:46.11 | file | it's shido! |
| 21:46.14 | Zeeek | that isn't the first time that suggestion was given today |
| 21:46.18 | file | doolph: Well then fix it so it connects! |
| 21:46.25 | file | doolph: cause right now it won't connect to your server or whatever |
| 21:46.58 | shido6 | should I scare ppl and put up ppls pics? |
| 21:47.07 | file | shido6: for what? |
| 21:47.46 | shido6 | boredom? |
| 21:47.51 | file | meh great idea |
| 21:48.55 | doolph | argh |
| 21:48.59 | file | yay headphones |
| 21:49.01 | doolph | everything is worst now |
| 21:49.27 | doolph | I cant believe it, I followed step by step without errors |
| 21:49.37 | |Vulture| | Oh I got the bestest pics evar! |
| 21:49.52 | file | lemme find a good pic of me |
| 21:49.57 | |Vulture| | http://www.janpro-fl.com/sex0r/IMG_0168.JPG |
| 21:49.59 | |Vulture| | there ya go |
| 21:50.31 | doolph | how gay |
| 21:50.33 | doolph | lol |
| 21:50.35 | |Vulture| | hahah |
| 21:50.47 | doolph | omg |
| 21:50.50 | |Vulture| | http://www.janpro-fl.com/sex0r/IMG_0167.JPG |
| 21:50.52 | |Vulture| | HHAHAHAA |
| 21:50.54 | doolph | AMP suck my balls |
| 21:51.01 | file | http://gallery.indigent-networks.com/AtriptoSanFrancisco/p3110144 |
| 21:51.03 | |Vulture| | the non-gay one |
| 21:51.04 | file | bkw on the left, me on the right |
| 21:51.20 | |Vulture| | Got VoIP? |
| 21:51.36 | file | voice |
| 21:52.16 | |Vulture| | zaptel head is fux0red |
| 21:52.24 | |Vulture| | for fax |
| 21:52.44 | file | as well everyone, I'll be speaking at Cluecon - so be sure to come! |
| 21:53.13 | doolph | about what |
| 21:53.32 | file | a showcase of voip things |
| 21:53.41 | file | Asterisk, SER, Cisco, Lucent, mmm |
| 21:53.52 | doolph | ah, i wish to go |
| 21:53.59 | file | probably ranting about SIP somewhat |
| 21:54.13 | doolph | there's something called SIP version 2? |
| 21:54.16 | file | yes |
| 21:54.25 | file | it replaced SIP version 1 ironically, a new RFC |
| 21:54.41 | FuriousGeorge | got voip, lol |
| 21:54.43 | doolph | what are asterisk using |
| 21:54.46 | file | 2 |
| 21:54.54 | file | FuriousGeorge: I wish it said Got VoIP |
| 21:54.59 | |Vulture| | who would have thought... SIP2 replaces SIP1? |
| 21:54.59 | file | it actually says "Got Voice?" |
| 21:55.14 | doolph | argh |
| 21:55.18 | FuriousGeorge | what does a voip moustache look like? |
| 21:55.22 | FuriousGeorge | it doesnt say that? |
| 21:55.26 | |Vulture| | * uses 1 |
| 21:55.28 | doolph | file you have your own manager eh |
| 21:55.29 | FuriousGeorge | oh |
| 21:55.30 | FuriousGeorge | i c |
| 21:55.58 | file | |Vulture|: eh? |
| 21:56.02 | ariel_ | asterisk uses 2 or so at least part of it. |
| 21:56.06 | file | yeah, 2 |
| 21:56.13 | file | everyone uses 2 now |
| 21:56.24 | doolph | 1 or 2 who care |
| 21:56.30 | file | quiet you |
| 21:56.31 | doolph | it's same for me lol |
| 21:56.48 | ariel_ | doolph, If your setting up amp you can't use res_ I told you to noload it some time ago. |
| 21:57.07 | doolph | ariel_ no i am getting more errors |
| 21:57.18 | file | I just checked, we are using 2 ;) |
| 21:57.27 | FuriousGeorge | file: having read that link you pasted re: co/lo, im just wondering: you can get a co/lo with your telco and resell their bandwidth? |
| 21:57.44 | ariel_ | doolph, I also recommend if your using amp to use stable it works and besides you don't use the other feature if your using amp. |
| 21:57.46 | file | FuriousGeorge: maybe |
| 21:58.02 | FuriousGeorge | gotcha |
| 21:58.24 | FuriousGeorge | that would be maybe depending on the telco, right |
| 21:58.28 | file | FuriousGeorge: yes |
| 21:58.39 | doolph | ariel_ I run asterisk@home in a box and it worked very cool, but I want to install it on a existing machine |
| 21:58.39 | FuriousGeorge | ok, i do gotcha |
| 21:59.26 | file | time for latest Andromeda |
| 21:59.32 | doolph | argh |
| 21:59.38 | doolph | i will reinstall this night again |
| 22:00.07 | doolph | will this error caused by no usb, zap hardware? |
| 22:00.09 | file | FuriousGeorge: you could also go to a regular colo place, make special arrangements to get a PRI from the telco dropped there, plus a regular internet connection, plus a P2P to customers |
| 22:00.15 | ariel_ | well bye folks I got to make dinner for my kid and my other girls. Then I have a party to go. My brother in-law is turning 50. |
| 22:00.24 | file | ariel_: yeekz old |
| 22:00.38 | ariel_ | old hummm he is only 3 years older then me.... |
| 22:00.43 | file | yes, you're old |
| 22:00.48 | doolph | file how do you add extensions? |
| 22:00.57 | file | doolph: what do you mean |
| 22:01.02 | file | to what? |
| 22:01.03 | doolph | with text editor or something like AMP |
| 22:01.14 | doolph | really? |
| 22:01.17 | file | yes... |
| 22:01.21 | FuriousGeorge | file: cool, thanks for the info, this chatroom opens new worlds to me. and i acheived my original goal of replacing the doorbell my landlord wouldnt fix with an asterisk install |
| 22:01.40 | FuriousGeorge | from there, the skies the limit, really |
| 22:01.51 | doolph | what if you need to add lot of extensions |
| 22:01.59 | whachagonnado | Okay, I'll ask a little more directly. Is Gentoo a good distro option for asterisk? |
| 22:02.05 | doolph | you need to edit example sip.conf and voicemail.conf, etc etc |
| 22:02.14 | file | doolph: custom system for configuration file generation |
| 22:02.28 | doolph | what you mean |
| 22:02.31 | file | plus I have a deployment system I wrote a few months ago for creating a web of asterisk servers, where it propogates to each |
| 22:02.34 | file | and that's all I'm saying. |
| 22:03.19 | FuriousGeorge | whachagonnado: a lot of people seem to use it. i suppose if you had a serious server doing real mission critical stuff, you may want to go with a red hat or something with that kind of support and whatnot |
| 22:04.14 | FuriousGeorge | i got a buddy who says i shouldnt use gentoo to have asterisk foreward calls over a zap channel from one specific line to another, the same thing a few dozen times a day |
| 22:04.40 | FuriousGeorge | then again, people and #gentoo will tell you gentoo makes a fine server |
| 22:04.46 | whachagonnado | how about a roll call? Whats everyone using as thier distros? |
| 22:05.01 | FuriousGeorge | BeOS |
| 22:05.05 | FuriousGeorge | j/k |
| 22:05.08 | whachagonnado | I really like gentoo personally |
| 22:05.25 | whachagonnado | But I do hear the support argument too. |
| 22:05.28 | file | whachagonnado: custom! |
| 22:05.32 | FuriousGeorge | mee too, thats why i tell my buddy ill take my chances. i never used anything else other than mandrake and i hated that |
| 22:05.43 | ManxPower | whachagonnado: You are going to have many people tell you yes and many people tell you no. For the most part there all linux distros are the same, except MAYBE you have to install your own kernel. |
| 22:05.57 | file | and if any of you come to Cluecon, you'll see what I use |
| 22:06.29 | whachagonnado | where is Cluecon? |
| 22:06.31 | doolph | ok ok |
| 22:06.39 | FuriousGeorge | where is this cluecon you all speak of |
| 22:06.53 | FuriousGeorge | vegas? |
| 22:06.56 | FuriousGeorge | baby? |
| 22:06.59 | Qwell | Chicago |
| 22:07.03 | ManxPower | FuriousGeorge: some city nobody has heard of. Chicago, I think. |
| 22:07.05 | doolph | and what's your name |
| 22:07.16 | doolph | Admission: $650.00 |
| 22:07.19 | ta[i]nted | i've heard of Chicago |
| 22:07.19 | doolph | great |
| 22:07.21 | FuriousGeorge | Chi-cah-go? |
| 22:07.25 | FuriousGeorge | never head of it |
| 22:07.31 | ta[i]nted | great band |
| 22:07.50 | FuriousGeorge | i still think about the karate kid |
| 22:07.54 | ta[i]nted | a little on the soft side |
| 22:07.58 | FuriousGeorge | ya |
| 22:08.07 | FuriousGeorge | now boston is just right |
| 22:08.10 | doolph | when is it? |
| 22:08.44 | file | August 3rd to 5th |
| 22:08.59 | file | and that $650 includes your hotel, plus lunches |
| 22:09.09 | doolph | ah it's far |
| 22:09.24 | doolph | and regist form is not working |
| 22:09.44 | file | works for me |
| 22:09.59 | Qwell | file: Did you have them make an ipv4 only hostname? :p |
| 22:10.05 | doolph | its blocking me then |
| 22:10.06 | file | Qwell: quiet you |
| 22:10.21 | file | now let me watch my Andromeda in peace and quiet |
| 22:19.33 | docelmo | I know Im a little late but I run CentOS 32/64bit cross the board |
| 22:19.50 | *** join/#asterisk zotz (~zotz@24.231.32.109) |
| 22:23.44 | PyroSteve | i just found an alternate provider to broadvoice |
| 22:23.56 | PyroSteve | someone local |
| 22:26.19 | jets | :) |
| 22:26.30 | jets | ne idaho *'rs |
| 22:41.02 | kb1_kanobe | how is the -head code feeling at the moment? No nasty 'PRI doesn't work' or anything right now? |
| 22:43.59 | MikeJ[Laptop] | Kram! |
| 22:45.47 | *** join/#asterisk tsp (~tyler@S01060080c825173c.vc.shawcable.net) |
| 22:46.08 | tsp | can somoene help me wtih alsa and asterisk? |
| 22:46.22 | tsp | I'm using alsa 1.0.8 and I hear a bunch of clicks during my calls. |
| 22:47.12 | FuriousGeorge | what softphone are you using? |
| 22:47.50 | tsp | FuriousGeorge: asterisk |
| 22:47.55 | tsp | asterisk is the only softphone I can get. |
| 22:48.00 | tsp | that's not gui based. |
| 22:48.20 | ManxPower | Ha! I figured it out! |
| 22:48.28 | *** join/#asterisk burrito7 (~Tommy@port-212-202-182-4.dynamic.qsc.de) |
| 22:48.37 | burrito7 | hi |
| 22:48.55 | ManxPower | The SIPura SPA-2100 uses RTP payload 2 for G726-32 (the old number), Asterisk uses payload 111 for G726-32 |
| 22:49.08 | gambolputty | there a command to pause without audio for waiting for dtmf? |
| 22:49.16 | gambolputty | for a selectable length of time? |
| 22:49.34 | FuriousGeorge | tsp: i dunno, if u had said x-lite, i would have told you to watch out for the codec. are you having bandwidth issues, perhaps |
| 22:49.39 | tsp | hmm |
| 22:49.55 | tsp | FuriousGeorge: it works just fine under windows. |
| 22:49.59 | tsp | FuriousGeorge: using sjphone. |
| 22:50.10 | tsp | FuriousGeorge: but with the alsa drivers it just pops and clicks. |
| 22:50.29 | tsp | I hear sound alright - but it just makes an annoying clicking. |
| 22:50.44 | FuriousGeorge | im not sure... do you ahve OSS compatibility to try that, jsut to see if it fixes it |
| 22:51.08 | *** join/#asterisk harryvv (~leonardo@S010600a0c93f6f7e.vs.shawcable.net) |
| 22:51.26 | tsp | hmm |
| 22:52.04 | kb1_kanobe | I've interfaced some Nortel PBX's to * using PRI/DMS100. They use enblock dialing, not overlap and, although I can set up the various dial patterns that trigger the digit spills there is still a hard-coded digit timeout in there. Thus, if someone pauses for four seconds while dialing the incomplete number spills and they end up dialing the rest of the digits in-band. Anyone ever built a 'capture missing digits' function into their dial |
| 22:52.24 | FuriousGeorge | if alsa is built into your kernel already, you should be able to make the stuff under Alsa--> OSS (Not the OSS thats by itself) and have it working |
| 22:53.20 | tzanger | kram: hey |
| 22:53.20 | tsp | I tried using chan_oss... not much difference - I still hear some distortion |
| 22:53.29 | tzanger | kram: I'm reading through expired norstar patents |
| 22:53.30 | tsp | the busy signal on fwd is a good example |
| 22:53.47 | tsp | thats probably a sound asterisk plays though |
| 22:53.55 | tzanger | did you know their conference engine only transmits the loudest 3 particpants of any conference? the rest are muted |
| 22:54.07 | tzanger | er two loudest |
| 22:54.37 | FuriousGeorge | tsp: and only when u use asterisk it pops |
| 22:55.43 | tsp | FuriousGeorge: yeah |
| 22:55.49 | tsp | only when I use asterisk |
| 22:56.29 | tzanger | kb1_kanobe: true, but the patents specifically are for 3+ (emphasis on the +) parties |
| 22:56.31 | FuriousGeorge | how much processor do u have? |
| 22:57.02 | FuriousGeorge | thats prob not it. im outta my league already, im afraid :( |
| 22:57.12 | tsp | 750 mhz |
| 22:57.26 | FuriousGeorge | kinda on the low side |
| 22:57.36 | tsp | yeah but I'm only using asterisk as a softphone |
| 22:57.39 | tsp | nothing else |
| 22:57.41 | FuriousGeorge | oh |
| 22:57.58 | tsp | all I need is a softphone thats console based and that works |
| 22:58.19 | FuriousGeorge | unfortunately, its kinda dead in here. usually its rampant with people trying to help you |
| 22:58.41 | FuriousGeorge | maybe asterisk is a bit too robust of a softphone for you to use |
| 22:59.54 | tsp | why is it too robust? |
| 23:00.52 | drumkilla | well, overkill is probably a better term |
| 23:01.22 | tsp | I just need working sound - the rest works fine |
| 23:02.26 | tsp | the phone sounds alright with oss but with some breakup |
| 23:02.43 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@bzq-218-62-72.cablep.bezeqint.net) |
| 23:04.14 | *** join/#asterisk Sedorox (brandon@sedorox.staff.smartserv) |
| 23:08.03 | tsp | can I send a sound file to a currently connection user? |
| 23:08.56 | ta[i]nted | yes |
| 23:09.06 | ta[i]nted | just call him/her up |
| 23:09.30 | ta[i]nted | but obviously they'll need a second line |
| 23:10.19 | tsp | oh |
| 23:10.27 | tsp | heh |
| 23:15.16 | *** join/#asterisk OzJames79 (~opera@CPE20320889-1842-1.gex.ncable.net.au) |
| 23:16.43 | *** join/#asterisk tld (~tld@80.203.70.227) |
| 23:18.37 | Zeeek | tsp transfer them to an extension that pays the file |
| 23:18.46 | Zeeek | 8 minutes later |
| 23:19.20 | Zeeek | or set the moh to that file, and put em on hold |
| 23:19.34 | Zeeek | what am I doing up at this hour? |
| 23:19.51 | tsp | oh |
| 23:19.52 | tsp | hmm |
| 23:20.03 | Zeeek | oh |
| 23:20.06 | Zeeek | ha |
| 23:20.12 | Zeeek | ooooh |
| 23:20.16 | Zeeek | the fireworks! |
| 23:30.03 | *** join/#asterisk RoyK (~roy@179.80-203-29.nextgentel.com) |
| 23:30.05 | OzJames79 | anyone here using InPhonex? |
| 23:30.38 | RoyK | anyone here experiencing heavy memory leaks? |
| 23:32.17 | Moc | Our toronto is very experience with memory leaks .. |
| 23:32.43 | Moc | we ordered so many Memory modules in the past few month, they are now expert ;) |
| 23:36.23 | Zeeek | I thought I had memory leaks, but in the end it was just anemia |
| 23:36.39 | file | my memory leaks so much I forgot it was leaking |
| 23:36.57 | Zeeek | it turned out the memory kept going down at 4AM |
| 23:37.07 | kb1_kanobe | royk: what is the environment and what are the symptoms? |
| 23:37.13 | Zeeek | that was exactly the time of a cronjob doing backups or something |
| 23:37.26 | Zeeek | just linux keeping the space for buffers |
| 23:37.45 | RoyK | kb1_kanobe: 1.0.7 sipfriends and agi |
| 23:38.11 | Zeeek | it's youir farfon calling home! |
| 23:38.16 | kb1_kanobe | I've no experience with that combination. Sorry. |
| 23:38.31 | Zeeek | wtf are you doing up at this hour RoyK? |
| 23:39.02 | RoyK | Zeeek: wtf are you? |
| 23:39.18 | Zeeek | I'm a bot of the real Zeeek who is sound asleep |
| 23:39.28 | RoyK | hehe |
| 23:39.53 | Zeeek | I was on CDT for 5 days now I can't sleep |
| 23:40.04 | Zeeek | wait there's porn on at this hour |
| 23:40.17 | Zeeek | Europe is so great for things that count |
| 23:40.54 | RoyK | Zeeek: any ideaa what sort of peple that can help me sort this out? |
| 23:41.05 | Zeeek | what the memory? |
| 23:41.39 | tzanger | hahahaha |
| 23:41.40 | tzanger | http://www.rentmydaughter.com/ |
| 23:41.51 | citats | RoyK: can you provide some details? |
| 23:41.53 | Zeeek | I found the film, gotta go |
| 23:42.05 | *** part/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 23:42.28 | RoyK | just heavy memoty |
| 23:42.36 | RoyK | memory leaks |
| 23:42.42 | RoyK | from 1.0.7 |
| 23:42.52 | citats | RoyK: in which modules? is it possible to reproduce? |
| 23:43.01 | RoyK | using sipfroemds amd agi |
| 23:43.21 | RoyK | always reprodcable |
| 23:43.36 | citats | can you recompile asterisk with astmm enabled? |
| 23:43.54 | RoyK | astmm? |
| 23:44.27 | citats | theres a line in the main Makefile |
| 23:44.31 | citats | make it look like this: |
| 23:44.32 | citats | MALLOC_DEBUG = -include $(PWD)/include/asterisk/astmm.h |
| 23:44.41 | citats | normally part is commented out |
| 23:44.46 | citats | then make clean and recompile and reinstall |
| 23:45.15 | RoyK | what is that_ |
| 23:45.26 | RoyK | ? |
| 23:45.43 | citats | a tool to help locate memory leaks |
