| 00:00.44 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 00:00.52 | *** join/#asterisk chaoscon (n=ph33r@smartserv/ceo/chaoscon) |
| 00:01.02 | *** join/#asterisk zeppelin_ (n=zeppelin@201.41.0.159) |
| 00:01.26 | RoyKa | ~nickometer L|NUX |
| 00:02.32 | Strom_C | ~nickometer Strom_C |
| 00:02.37 | Strom_C | woot |
| 00:03.03 | orlock | ~nickometer orlock |
| 00:03.06 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 00:03.51 | stronginteractio | can you set voicemail options per voicemail context? |
| 00:04.24 | stronginteractio | i know you can set global defaults, but what about context defaults? |
| 00:06.11 | De_Mon | stronginteractio I dont think so, but you can set mailbox specific settings |
| 00:08.17 | *** join/#asterisk Powerkill (n=PoWeRKiL@84.205.154.179) |
| 00:08.17 | Powerkill | hi |
| 00:08.26 | stronginteractio | ah that is not easy for me |
| 00:08.38 | stronginteractio | i am storing user mailbox information in a mysql database |
| 00:09.06 | *** join/#asterisk jeremy_g (n=j@83.233.40.109) |
| 00:09.10 | jeremy_g | hi |
| 00:09.13 | jeremy_g | i am fucked |
| 00:09.22 | jeremy_g | just did 150 sittups as a bet with a friend and won it |
| 00:09.28 | jeremy_g | and now my muscles are so frigging stiff |
| 00:09.38 | Strom_C | thank you for joining #offtopic |
| 00:09.44 | Strom_C | please continue to hold |
| 00:09.48 | Strom_C | someone will be with you soon |
| 00:09.54 | Strom_C | [new wave] |
| 00:11.24 | Strom_C | fifteen hundred pages of goodness |
| 00:12.22 | Powerkill | someone using vgsm card ? |
| 00:12.59 | *** join/#asterisk Defraz (n=t0tal@24-119-94-19.cpe.cableone.net) |
| 00:13.10 | *** join/#asterisk Gregabyte (i=greg@nat/digium/x-cd5b1d34223d7864) |
| 00:15.36 | *** join/#asterisk EyeCue (n=eyecue@unaffiliated/eyecue) |
| 00:16.14 | *** join/#asterisk neoalex (n=neoalex@user-0cdfjlp.cable.mindspring.com) |
| 00:16.30 | neoalex | hello |
| 00:16.47 | neoalex | can anyone help me find a simple tutorial on how to configure asterisk |
| 00:16.57 | Strom_C | ~docs |
| 00:16.58 | jbot | it has been said that docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 00:16.58 | Strom_C | ~book |
| 00:17.00 | jbot | well, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 00:17.14 | Strom_C | ~hafc |
| 00:17.15 | jbot | somebody said hafc was hire a freaking consultant. Look, if you're having difficulty understanding what you're doing and need a solution soon, you will be far better off hiring a competent consultant than continuing to pull your hair out. |
| 00:18.08 | *** part/#asterisk QbY (n=Kelvin@cm-64-221-172-88.dhcp.southerncoastalcable.net) |
| 00:18.22 | neoalex | yeah... well I'm too cheap to even get a landline... :)) |
| 00:18.45 | neoalex | thanks for your help Storm_C |
| 00:18.50 | *** join/#asterisk DasTech (n=DasTech@d47-69-194-47.col.wideopenwest.com) |
| 00:19.14 | *** join/#asterisk tengulre (n=tengulre@221.11.5.180) |
| 00:20.33 | *** join/#asterisk shuri (n=shuri@64.235.209.226) |
| 00:20.36 | *** join/#asterisk bintut (n=bintut@58.69.3.95) |
| 00:21.38 | bintut | anyone here running asterisk on debian etch? :) |
| 00:22.06 | shuri | bintut, yes |
| 00:22.32 | bintut | i just apt-get'ed asterisk here on this laptop.. i'm currently running debian etch.. but i'm directly connected thru pppoe |
| 00:23.28 | bintut | i just want to try this out.. how to make this work especially sip.. maybe for 4 to 5 hours or so because my public ip is not static though |
| 00:24.03 | bintut | any asterisk howto that is specific for debian available somewhere on the web? :) |
| 00:24.20 | riddlebox | bintut, you need to use dyndns.org then |
| 00:24.29 | intralanman | or noip |
| 00:24.58 | bintut | ok, i'll get a dyndns.org account.. i hope you guys can help me set this up.. :) |
| 00:25.05 | riddlebox | intralanman, do you know how to make a hot phone, where when you go offhook it dials one number automatically? |
| 00:25.53 | intralanman | i do not.... some ATA's could have a dialplan that would dial one number no matter what you dial though |
| 00:26.13 | intralanman | but if you find the answer you're really looking for, let me know too |
| 00:26.31 | intralanman | i've been wanting to do that for a while, but haven't found a way to do it without special hardware |
| 00:26.50 | riddlebox | there should be a way in programming to do it |
| 00:27.03 | riddlebox | a wildcard in the dial string or something |
| 00:28.26 | *** join/#asterisk diskace (i=diskace@dsl.speedline209.234.electronicbox.net) |
| 00:28.26 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 00:28.40 | diskace | hi guys |
| 00:28.52 | *** join/#asterisk chaoscon_ (n=ph33r@smartserv/ceo/chaoscon) |
| 00:29.36 | shuri | hi diskace |
| 00:29.44 | diskace | hey, shuri ! |
| 00:29.48 | diskace | ;) |
| 00:33.17 | DasTech | <==looking or a surrpoetjobis voip/asterisk/telecommuter |
| 00:33.19 | bintut | ok, i got the bintut.homelinux.org |
| 00:36.21 | bintut | what's a good gnome based softphone that supports sip? |
| 00:36.40 | intralanman | ekiga |
| 00:36.51 | intralanman | fka gnome meeting |
| 00:37.04 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 00:37.30 | bintut | you mean, i can use ekiga to connect to the asterisk that i'll setup on my laptop? :) |
| 00:37.43 | intralanman | yeah |
| 00:37.55 | bintut | ok.. i'll apt-get it.. :) |
| 00:38.23 | *** join/#asterisk JT (n=jon@unaffiliated/jt) |
| 00:38.46 | JT | damn this must be identified to join thing is really annoying |
| 00:40.18 | *** join/#asterisk freebsd_fan (n=ebola@hdkbib3.hdk.gu.se) |
| 00:40.40 | diskace | anyone had experience with tellabs echo canceller cards with digium cards or sangoma cards ? |
| 00:41.33 | diskace | look like being an alternative to octasic expensive EC module |
| 00:41.57 | *** join/#asterisk fall0ut (n=tim@c-69-180-250-113.hsd1.tn.comcast.net) |
| 00:43.24 | orlock | JT: went by st pauls cathedral on the way to work.. saw brockies hearse |
| 00:43.30 | orlock | complete with "BROCKY" plates |
| 00:43.31 | JT | hmm |
| 00:43.59 | orlock | city getting abit packed |
| 00:46.33 | bintut | intralanman: ekiga isn't in my repository.. i'm running debian etch here.. but gnomemeeting was installed.. can i use it instead? |
| 00:46.54 | intralanman | probably |
| 00:47.03 | bintut | ok, i'll try.. |
| 00:47.36 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 00:49.04 | *** join/#asterisk teknoprep (n=tekon@unaffiliated/teknoprep) |
| 00:49.12 | teknoprep | is there a way to monitor calls |
| 00:49.39 | diskace | hum, sad no reply to my last question :( |
| 00:49.48 | diskace | well, i got another one :) |
| 00:50.31 | diskace | it's quite difficult to answer but it's from your personnal experience |
| 00:50.57 | teknoprep | ? |
| 00:51.12 | diskace | i was wondering how many user/channel ratio would be safe on a residential voip environment ? |
| 00:51.26 | diskace | with only half-pri |
| 00:51.31 | diskace | and full pri |
| 00:51.42 | diskace | 11 / 23 channels respectively |
| 00:51.54 | diskace | what you guys think ? |
| 00:52.01 | diskace | 3 for 1 on half pri ? |
| 00:52.35 | diskace | i heard that 7 for 1 on a full pri is a good 'standard' for a full pri |
| 00:52.44 | intralanman | diskace: depends on the residents ;) |
| 00:52.50 | diskace | yeah, of course |
| 00:52.52 | diskace | :) |
| 00:53.02 | intralanman | 5 for 1 on half personally |
| 00:53.17 | *** join/#asterisk teknoprep (n=tekon@unaffiliated/teknoprep) |
| 00:53.25 | diskace | intralanman you had no problems |
| 00:53.27 | diskace | ? |
| 00:53.33 | intralanman | not so far |
| 00:53.38 | diskace | sweet |
| 00:53.55 | diskace | and thats for residential users ? |
| 00:54.14 | intralanman | yeah |
| 00:54.27 | *** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 00:54.45 | diskace | are you monitoring ? |
| 00:54.56 | intralanman | again, results may vary based on state, weight, haircolor, etc |
| 00:54.57 | intralanman | lol |
| 00:55.02 | diskace | haha |
| 00:55.08 | diskace | haircolor, you sure ? |
| 00:55.10 | intralanman | nah, not really "monitoring" |
| 00:55.17 | intralanman | yeah, blondes get more calls |
| 00:55.18 | intralanman | lmao |
| 00:55.22 | diskace | true ! |
| 00:55.23 | shuri | <PROTECTED> |
| 00:55.46 | diskace | i wonder if it is legal to ask when they signup |
| 00:55.49 | diskace | their hair color |
| 00:56.04 | intralanman | hmmmm |
| 00:56.11 | intralanman | that might depend on the state |
| 00:56.12 | diskace | please check this box if you are a sexy blonde |
| 00:56.34 | intralanman | "post pics for your directory listing" |
| 00:56.39 | diskace | lol |
| 00:57.43 | *** join/#asterisk teknoprep (n=tekon@unaffiliated/teknoprep) |
| 00:57.57 | diskace | intralanman, from your experience in the residential field |
| 00:58.10 | diskace | can you tell me when is your peak period in the day ? |
| 00:58.26 | intralanman | time of day, you mean? |
| 00:58.30 | diskace | yup |
| 00:58.41 | intralanman | 3-11 |
| 00:58.43 | intralanman | lol |
| 00:58.49 | intralanman | that's really not far off |
| 00:58.58 | intralanman | but that's EDT |
| 00:59.05 | *** join/#asterisk xAD (n=xAD@host144-199.pool8290.interbusiness.it) |
| 00:59.06 | diskace | 15h00 or 03:00 ? |
| 00:59.15 | intralanman | 1500 |
| 00:59.26 | intralanman | and really it's closer to 1700 |
| 00:59.35 | intralanman | just starts early on some days |
| 00:59.41 | diskace | when they get back home |
| 00:59.49 | intralanman | half days of school, fridays, etc |
| 00:59.51 | intralanman | yeah |
| 01:00.05 | diskace | you have emergency channels |
| 01:00.06 | diskace | ? |
| 01:00.17 | intralanman | yeah |
| 01:01.53 | intralanman | isn't that a legal "requirement" now? |
| 01:02.22 | diskace | hum |
| 01:02.29 | *** join/#asterisk Mportnoy (n=test@201.199.68.150) |
| 01:02.35 | diskace | well, i didn't mean e911 or basic 911 |
| 01:02.41 | diskace | just emergency channels |
| 01:02.41 | intralanman | oh |
| 01:02.52 | diskace | if your pri get full |
| 01:03.06 | diskace | you can route calls using another place |
| 01:03.16 | intralanman | well, i have a couple of PRIs just for emergency services |
| 01:03.22 | intralanman | which rarely ever get used |
| 01:03.29 | intralanman | it's actually pretty upsetting |
| 01:03.38 | diskace | i see |
| 01:03.55 | intralanman | but they're direct links to the emergency services |
| 01:04.10 | intralanman | so it's not like i can better utilize them in down times |
| 01:04.29 | diskace | you only have 1 half pri and couple pri's. .... |
| 01:04.33 | diskace | i don't understand |
| 01:04.58 | intralanman | different scenarios, different companies, etc |
| 01:05.05 | diskace | okk |
| 01:05.09 | diskace | nice |
| 01:05.26 | diskace | and from experience again |
| 01:05.44 | pyrom | What's the "preffered" Wakeup call script |
| 01:05.44 | intralanman | the company i "work" for has about 1000 customers (which isn't really that much) and 2 pri's for emergency services |
| 01:05.45 | diskace | what is the 'standard' ratio you use for a full pri |
| 01:06.02 | diskace | aside from blonde :) |
| 01:06.20 | intralanman | heheh, usually it's close to 3/1 or slightly less |
| 01:06.24 | intralanman | 2.7/1 |
| 01:06.31 | intralanman | +/- |
| 01:06.39 | intralanman | the 2.7 i just made up |
| 01:06.50 | intralanman | but usually about 3/1 |
| 01:07.34 | intralanman | pyrom: i prefer to write my own |
| 01:07.45 | intralanman | others have different preferences |
| 01:07.53 | pyrom | ok |
| 01:08.01 | diskace | 3:1 ? |
| 01:08.03 | pyrom | how do you issue the call? |
| 01:08.05 | pyrom | via cron? |
| 01:08.33 | diskace | 5:1 for a half pri and 3:1 for a full pri ? |
| 01:08.51 | intralanman | i run a cron job to pull info from a db and run the calls off of it |
| 01:08.57 | intralanman | diskace: you said full pri? |
| 01:08.59 | intralanman | my bad |
| 01:09.32 | diskace | well, yes i asked for a full pri and half (before) |
| 01:09.36 | diskace | you answered 5:1 for the half |
| 01:09.52 | *** join/#asterisk lero (n=rootz@201-1-24-71.dsl.telesp.net.br) |
| 01:09.55 | lero | hi |
| 01:09.58 | *** join/#asterisk type0 (i=type0@159-76-74-65.gci.net) |
| 01:09.59 | type0 | hey guys |
| 01:10.02 | diskace | and 3:1 for the full lol :) |
| 01:10.09 | intralanman | the 5:1 was kinda on the safe side..... i was thinking 3:1 is about average |
| 01:10.19 | intralanman | for a half |
| 01:10.25 | lero | i'm trying to compile zaptel and getting this: /bin/sh: /etc/udev/rules.d/zaptel.rules: Permission denied, but i'm doing this as root. |
| 01:10.39 | orlock | lero: its trying to execute it |
| 01:10.41 | diskace | ok |
| 01:10.41 | type0 | I have a client who wants to migrate to asterisk, but they currently have all rj11 handsets.. whats a good product to connect say.. 80 phones to an asterisk box? |
| 01:10.47 | diskace | and for a full ? |
| 01:10.57 | lero | orlock: yeah, but this file don't exist too. |
| 01:11.04 | orlock | ahh |
| 01:11.17 | lero | it do this first: build_tools/genudevrules > /etc/udev/rules.d/zaptel.rules |
| 01:11.35 | intralanman | diskace: it varies, slightly better than double usually |
| 01:11.49 | diskace | nice |
| 01:12.10 | diskace | just trying to have an idea |
| 01:12.26 | intralanman | diskace: just stay away from the blondes ;) |
| 01:12.38 | diskace | of course :P |
| 01:12.43 | diskace | ill have a special price for them |
| 01:12.43 | teknoprep | could someone help me with this... exten => *888,1,DBput(ivr/mode=bizhours) and exten => *999,1,DBput(ivr/mode=afterhours) |
| 01:12.43 | teknoprep | GotoIf($[${ivrmode} = "bizhours"]?daytime-ivr|s|1:nighttime-ivr|s|1) |
| 01:12.48 | teknoprep | still trying to understand it |
| 01:13.28 | teknoprep | i understand that you set a variable using *888 or *999 using DBput ... setting the ivr mode by using the *888 or *999 |
| 01:13.37 | teknoprep | the gotoif part is what i am trying to understand |
| 01:13.53 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 01:13.57 | teknoprep | gotoif ... is that gotoif statement complete and will it work with what i am attempting? |
| 01:14.00 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 01:16.03 | diskace | intralanman , thanks for sharing your experience |
| 01:16.06 | diskace | it is really appreciated |
| 01:16.10 | intralanman | sure, np |
| 01:16.23 | intralanman | teknoprep: is it not working? |
| 01:16.31 | intralanman | it looks ok at first glance |
| 01:16.36 | intralanman | second even for that matter |
| 01:16.51 | teknoprep | nono |
| 01:17.01 | teknoprep | i am just wanting to know how the GotoIf exactly works |
| 01:17.17 | teknoprep | someone helped me write that... and now i am just trying to understand it |
| 01:17.39 | intralanman | oh, ok |
| 01:17.43 | Qwell | teknoprep: if the variable ivrmode is "bizhours" then goto daytime-ivr|s|1, otherwise goto nighttime-ivr|s|1 |
| 01:17.50 | teknoprep | ahhh |
| 01:17.58 | teknoprep | now that makes sense |
| 01:18.00 | teknoprep | if else |
| 01:18.16 | teknoprep | that was my question.. how was it knowing that it was in afterhours mode |
| 01:18.19 | intralanman | exactly |
| 01:18.42 | intralanman | like "condition ? trueAction : falseAction" in C or PHP |
| 01:18.49 | teknoprep | yup |
| 01:19.00 | teknoprep | well i have really only ever programed in C# and VB |
| 01:19.01 | *** join/#asterisk aao_pwner (n=aao_pwne@c-24-21-91-140.hsd1.mn.comcast.net) |
| 01:19.11 | Qwell | C# has that too |
| 01:19.21 | intralanman | that's good to know |
| 01:19.34 | teknoprep | soooo |
| 01:19.35 | intralanman | that's one less thing i have to learn for that |
| 01:20.01 | teknoprep | i put the exten => in the extensions.conf |
| 01:20.20 | teknoprep | do i then make 2 ivr's ... one named bizhours and one named afterhours ? |
| 01:20.33 | intralanman | daytime-ivr |
| 01:20.40 | intralanman | and nighttime-ivr |
| 01:20.47 | teknoprep | oh yeah |
| 01:20.47 | teknoprep | lol |
| 01:20.58 | intralanman | but that's the idea, yeah |
| 01:21.14 | teknoprep | which also go in the extensions.conf |
| 01:21.24 | intralanman | usually |
| 01:21.32 | intralanman | s/usually/yes/ |
| 01:21.51 | intralanman | jbot is cool |
| 01:21.55 | intralanman | so helpful |
| 01:22.06 | intralanman | ~jbot |
| 01:22.08 | jbot | jbot is probably only marginally useful at best, He got a C- on his Turing Test, or a complete idiot, or a dolt |
| 01:23.05 | encode | haha |
| 01:24.12 | diskace | hey guys |
| 01:24.17 | diskace | i am leaving already |
| 01:24.29 | diskace | have to relax a bit |
| 01:24.43 | teknoprep | ahh nice |
| 01:24.54 | diskace | lol |
| 01:25.11 | teknoprep | to set this up in Freepbx i can record my IVR... set them up in Freepbx... then just copy and paste the settings from each IVR with a new name into the extensions.conf |
| 01:25.18 | teknoprep | this is easy |
| 01:25.21 | teknoprep | too easy |
| 01:25.38 | teknoprep | ty all for the help on that.. that was pretty much my only stumbling block |
| 01:28.14 | intralanman | hey teknoprep |
| 01:28.15 | *** part/#asterisk diskace (i=diskace@dsl.speedline209.234.electronicbox.net) |
| 01:28.20 | intralanman | i don't think that'll work out so well |
| 01:28.32 | intralanman | from what i remember, freepbx overwrites the config files |
| 01:28.44 | bintut | i have some questions regarding asterisk and sip.. i was given a username and password to connect to the pbx server of my friend located in another country. but he's not around yet. now, i want to try to connect to it. i already installed asterisk on this laptop and got a dyndns account already. what do i need to do in order to connect to the pbx (sip) server of my friend? |
| 01:28.50 | intralanman | you have to put that in extensions_custom.conf or something |
| 01:29.08 | teknoprep | yoyo |
| 01:32.09 | bintut | intralanman: i just apt-get'ed asterisk here.. what are the configurations i need to do in order to connect to the asterisk pbx/sip box of my friend? |
| 01:32.34 | intralanman | ummmm, look into the sip.conf, and then extensions.conf |
| 01:32.47 | intralanman | check voip-info.org for more info on those files |
| 01:32.56 | *** join/#asterisk zotz (n=zotz@24.244.163.225) |
| 01:33.08 | intralanman | also "The Book" (as it's been adequately named) |
| 01:33.13 | intralanman | ~book |
| 01:33.17 | jbot | well, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 01:33.17 | lero | what program i can use to test asterisk locally? just to call to it and see how it works basically? |
| 01:35.17 | *** join/#asterisk [Outcast] (n=outcast@222-154-72-242.jetstream.xtra.co.nz) |
| 01:36.57 | teknoprep | in my extensions.conf i have... exten => 14843351444,n,Goto(ivr-2,s,1) i would change that to exten => 14843351444,n,GotoIf($[${ivrmode} = "bizhours"]?daytime-ivr|s|1:nighttime-ivr|s|1) |
| 01:37.02 | teknoprep | is that correct ? |
| 01:37.38 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 01:39.16 | bintut | any gui tool to configure asterisk? |
| 01:39.52 | intralanman | bintut: maybe you should be in #freepbx |
| 01:39.57 | intralanman | or check out AMP |
| 01:39.59 | intralanman | ~amp |
| 01:40.13 | jbot | extra, extra, read all about it, amp is NOT supported here! People using it should join #freepbx (FreePBX is the new name of AMP) |
| 01:40.13 | bintut | i was given the username, password, realm/domain and pbx server only |
| 01:40.15 | intralanman | are you kidding? jbot doesn't know amp? |
| 01:40.17 | intralanman | oh |
| 01:40.18 | intralanman | lol |
| 01:41.23 | bintut | intralanman: amp and freepbx is not in my debian etch repository |
| 01:41.33 | bintut | hhmmm.. i think this is not easy to setup.. |
| 01:41.34 | bintut | :( |
| 01:42.00 | intralanman | i think you may be right |
| 01:42.04 | intralanman | :( |
| 01:42.12 | intralanman | got another machine you can just burn? |
| 01:42.21 | bintut | oh.. :( |
| 01:45.00 | bintut | intralanman: what are the configs i need to configure in order to connect to the other asterisk pbx server? |
| 01:45.16 | intralanman | sip.conf will get you "connected" |
| 01:45.40 | intralanman | in order to do anything useful, you'll need to look into extensions.conf also |
| 01:46.14 | bintut | i'm there already but it has so many variables.. |
| 01:46.44 | bintut | the info that was given to me are the following: username, password, realm/domain and pbx server only |
| 01:47.47 | intralanman | toward the bottom of the sip.conf sample file, there are sample peers that are close to what you need |
| 01:48.00 | intralanman | password is now known as secret |
| 01:48.21 | intralanman | and server is known as host |
| 01:48.37 | intralanman | keep that in mind and you'll do fine with the help of "The Book" |
| 01:48.50 | Qwell | I love how when I call Comcast, they can't hear me, but if I call my cell, it works just fine |
| 01:49.41 | intralanman | Qwell: Comcast uses samsungs or something ;) |
| 01:49.43 | intralanman | maybe avaya |
| 01:49.45 | intralanman | lol |
| 01:49.46 | *** part/#asterisk mtaht4 (n=m@207.47.5.58.static.nextweb.net) |
| 01:51.44 | teknoprep | with the exten => *888,1,DBput(ivr/mode=bizhours) |
| 01:51.44 | teknoprep | exten => *999,1,DBput(ivr/mode=afterhours) |
| 01:51.55 | teknoprep | i get a busy signal when i dial *888 or *999 |
| 01:52.07 | Qwell | why not just...use a GotoIfTime? |
| 01:52.11 | teknoprep | also it doesn't switch the ivr modes... it stays by default in the afterhours ivr |
| 01:52.18 | bintut | intralanman: ok.. |
| 01:52.20 | teknoprep | becuase the shop closes at random times |
| 01:52.37 | teknoprep | depends on when ppl leave.. and when clients are done for the day comming in |
| 01:52.53 | intralanman | teknoprep: you making the db connection ok? |
| 01:53.05 | Qwell | intralanman: astdb is files |
| 01:53.06 | teknoprep | db to mysql ? |
| 01:53.10 | Qwell | well, a file |
| 01:53.10 | teknoprep | oh |
| 01:53.33 | Qwell | teknoprep: Its busy because you're using freepbx |
| 01:53.47 | teknoprep | freepbx blocks it? |
| 01:53.54 | Qwell | No, but it makes everything stupid. |
| 01:56.52 | bintut | intralanman: in my /etc/asterisk/sip.conf, i got so many line on SECRETS, HOSTS, etc |
| 01:57.11 | *** join/#asterisk Terlouw (n=d3vious@proxy.amsterdam.intruder.nl) |
| 01:57.47 | Terlouw | people!, quick question... i need to register a premium rate number... any tips? |
| 01:57.55 | intralanman | bintut: did you read the docs i pointed you to? |
| 01:58.10 | *** join/#asterisk freebsd_fan (n=ebola@hdkbib3.hdk.gu.se) |
| 01:58.44 | bintut | intralanman: sorry, not yet.. i want to try to make this work first by editing the necessary configs in /etc/asterisk/sip.conf |
| 01:59.03 | intralanman | lol, those docs are to help you with the necessary configs |
| 02:01.01 | Terlouw | ,,,anyone? |
| 02:05.44 | *** join/#asterisk trelane (n=trelane@unaffiliated/trelane) |
| 02:06.51 | teknoprep | how do i check if dbput is writing the file when i dial an exten with dbput in it? |
| 02:07.29 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 02:11.30 | bintut | when trying to do the command of "/etc/init.d/asterisk start", i got this error: id: asterisk: No such user /etc/init.d/asterisk: line 49: [: =: unary operator expected Starting Asterisk PBX: start-stop-daemon: group `asterisk' not found |
| 02:12.02 | trelane | bintut, whose startup script are you using? |
| 02:12.15 | bintut | trelane: bundled with the debian etch |
| 02:13.39 | trelane | never used debian, sorry, sounds like you ought to be talking to the package maintainer |
| 02:13.50 | *** join/#asterisk lowlevel (n=Stuart@CPE0050ba71c82f-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 02:14.22 | bintut | oh my.. ;( |
| 02:14.38 | intralanman | does debian use /etc/init scripts? |
| 02:14.51 | intralanman | they're not in some rc directory or something? |
| 02:15.10 | encode | debian uses /etc/init.d scripts |
| 02:15.15 | *** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net) |
| 02:15.20 | teknoprep | exten => *888,1,DBput(ivr/mode=bizhours) and GotoIf($[${ivrmode} = "bizhours"]?daytime-ivr|s|1:nighttime-ivr|s|1) |
| 02:15.31 | encode | sounds like the script is attempting to start asterisk as teh user asterisk, which doesnt exist |
| 02:15.34 | teknoprep | why have ivr/mode in the first part.. and ivrmode in the second part with no / |
| 02:15.36 | teknoprep | ? |
| 02:17.22 | intralanman | teknoprep: where do you use dbget to set ivrmode? |
| 02:17.27 | intralanman | do you do that? |
| 02:17.44 | teknoprep | ? |
| 02:18.00 | intralanman | ${ivrmode} is a variable, right? |
| 02:18.07 | intralanman | where does it get set? |
| 02:18.20 | teknoprep | exten => *888,1,DBput(ivr/mode=bizhours) |
| 02:18.29 | teknoprep | wouldn't that set it? |
| 02:18.46 | intralanman | in the db, yes |
| 02:18.53 | intralanman | you need to set the var from the db |
| 02:18.58 | [TK]D-Fender | exten => *888,1,Set(DB(ivr/mode)=bizhours) |
| 02:19.39 | [TK]D-Fender | <PROTECTED> |
| 02:19.56 | [TK]D-Fender | teknoprep : those 2 lines have no apparent link to one another |
| 02:20.14 | teknoprep | hmmm |
| 02:20.24 | teknoprep | exten => *888,1,DBput(ivr/mode=bizhours) and exten => *999,1,DBput(ivr/mode=afterhours) |
| 02:20.25 | teknoprep | GotoIf($[${ivrmode} = "bizhours"]?daytime-ivr|s|1:nighttime-ivr|s|1) |
| 02:20.29 | teknoprep | is what i have |
| 02:20.37 | intralanman | [TK]D-Fender: thanks, i was having trouble getting that to come out the way it sounded in my head :-D |
| 02:22.14 | bintut | i already started my asterisk but sip is not running |
| 02:22.53 | bintut | afaik, sip's port is 5060 but my output of my "lsof -i | grep asterisk" are the following: |
| 02:22.58 | intralanman | bintut: "sip show settings" |
| 02:23.02 | [TK]D-Fender | GotoIf($["${DB(ivr/mode)}"="bizhours"]?daytime-ivr|s|1:nighttime-ivr|s|1) |
| 02:23.50 | [TK]D-Fender | ${ivrmode} has no meaning in those lines you showed us. Its a channel variable or constant that yuo do not imply was created anywhere |
| 02:23.52 | bintut | # sip show settings |
| 02:23.53 | bintut | -su: sip: command not found |
| 02:24.03 | intralanman | my bad |
| 02:24.05 | intralanman | in the cli |
| 02:24.07 | intralanman | asterisk -r |
| 02:24.12 | intralanman | then that other one |
| 02:24.31 | teknoprep | hmm this just isn't working for me at all |
| 02:24.37 | teknoprep | i got the busy signal to go away |
| 02:24.57 | teknoprep | but when i dial.. which i now have it set to *25 and *26 for testing... i get nothing.. which i guess is good.. |
| 02:25.07 | teknoprep | but when i dial the number the ivr's are not changed |
| 02:25.21 | teknoprep | exten => 14843351444,n,GotoIf($[${ivr/mode} = "bizhours"]?ivr-2|s|1:ivr-3|s|1) |
| 02:25.27 | Qwell | bah |
| 02:25.29 | teknoprep | exten => *25,1,Set(DB(ivr/mode)=bizhours) |
| 02:25.29 | teknoprep | exten => *26,1,Set(DB(ivr/mode)=afterhours) |
| 02:25.43 | Qwell | Goto(${DB(ivr/mode)}|s|1) |
| 02:27.12 | Qwell | ~docs |
| 02:27.14 | jbot | from memory, docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 02:27.14 | Qwell | ~wikis |
| 02:27.16 | jbot | it has been said that wikis is http://www.voip-info.org |
| 02:27.17 | Qwell | teknoprep: go read those |
| 02:27.21 | Qwell | and buy this |
| 02:27.22 | Qwell | ~book |
| 02:27.23 | jbot | [book] a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 02:27.27 | teknoprep | omfg |
| 02:27.29 | teknoprep | it works |
| 02:27.30 | teknoprep | w0ot |
| 02:27.51 | teknoprep | yay!!!!!!!! |
| 02:27.52 | teknoprep | lol |
| 02:28.05 | teknoprep | i would love to thank all of you for the tedious but much thanked help |
| 02:28.30 | teknoprep | exten => *25,1,Set(DB(ivr/mode)=ivr-2) |
| 02:28.31 | teknoprep | exten => *26,1,Set(DB(ivr/mode)=ivr-3) |
| 02:28.31 | teknoprep | <PROTECTED> |
| 02:28.43 | teknoprep | exten => 14843351444,n,Goto(${DB(ivr/mode)}|s|1) |
| 02:28.49 | teknoprep | is what i ended up with .. just to let you guys know |
| 02:28.50 | teknoprep | ty again |
| 02:29.16 | intralanman | ~pb |
| 02:29.19 | jbot | pb is, like, a place to paste your stuff without flooding the channel - try http://pastebin.com/ (BROKEN AND SUCKING NUTS), or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/, or http://paste.lisp.org/ for the lisp/scheme nuts, or http://paste-it.net |
| 02:29.20 | intralanman | just to let you know |
| 02:29.21 | intralanman | lol |
| 02:29.29 | teknoprep | the astdb file.. will be saved even after a reboot...? |
| 02:29.31 | Qwell | You should still buy the book |
| 02:29.47 | teknoprep | so that if i reboot i will still be at the last set setting with the *25 or *26 ? |
| 02:29.55 | Qwell | yes |
| 02:29.58 | teknoprep | cool |
| 02:30.34 | intralanman | Qwell: dbput and dbget are deprecated, no? |
| 02:30.54 | Qwell | intralanman: yes |
| 02:30.56 | *** join/#asterisk Guest^DJ (n=me@211.24.146.11) |
| 02:31.38 | Guest^DJ | hi guys, is there a way to spoof callerid on a PRI/E1 |
| 02:31.41 | bintut | intralanman: kindly check the following sites: |
| 02:31.49 | bintut | intralanman: http://paste.debian.net/13223 |
| 02:32.05 | bintut | intralanman: http://paste.debian.net/13222 |
| 02:32.20 | Lyfe | you can set the outgoing callerid, but that doesnt' mean that 1- your provider has to listen to you, nor 2- that anyone along the way from there needs to listen to it. |
| 02:32.40 | bintut | intralanman: http://paste.debian.net/13220 |
| 02:32.50 | bintut | intralanman: http://paste.debian.net/13221 |
| 02:33.03 | Guest^DJ | Lyfe: i did set SetCallerID but has no effect |
| 02:33.35 | Guest^DJ | it still shows up as the ISDN number, not as requested |
| 02:34.43 | Lyfe | Guest^DJ: Hmm. Well, i have a PRI (t1 though) and have been using SetCIDNum (since i'm being lazy and not yet playing with trying to 'fix' it) and that works. |
| 02:35.10 | intralanman | bintut: that's way too many pb's for me to look at now |
| 02:35.18 | intralanman | which ones are really important? |
| 02:35.20 | intralanman | lol |
| 02:35.28 | Qwell | pastebin those pastebin links |
| 02:35.29 | Lyfe | but, like I said, your telco doesn't actually have to listen to what you set, and neither does any other telco. If you can't get it, I'd ask your telco if you have the option to set your callerid information. |
| 02:35.44 | file | Qwell: Der Waffle Haus! |
| 02:35.52 | Qwell | file: omg, soon! |
| 02:36.00 | Qwell | I get to go whenever I want! |
| 02:36.05 | Qwell | Which will be like...probably never |
| 02:36.30 | Lyfe | Guest^DJ: It's very possible that they might not be listening to what callerid information you set, possibly because they don't trust you, but most likely because it's a feature they'd have to enable or whatnot, and haven't done it. |
| 02:36.31 | bintut | intralanman: the output of sip show settings are in http://paste.debian.net/13223 |
| 02:36.34 | intralanman | so who's going to astricon this year? |
| 02:36.43 | droops | im trying to go |
| 02:36.47 | intralanman | aside from the obvious |
| 02:36.48 | intralanman | lol |
| 02:36.49 | Qwell | intralanman: anybody whos anybody |
| 02:36.50 | Guest^DJ | Lyfe: i would try SetCIDNum and would serioulsy doubt telco would even know what i want |
| 02:36.53 | *** join/#asterisk profounded (n=pro@ool-44c4eae2.dyn.optonline.net) |
| 02:36.58 | file | except anybody |
| 02:37.02 | intralanman | i feel like i must be somebody then |
| 02:37.04 | Qwell | file: he's a nub though |
| 02:37.11 | Lyfe | Guest^DJ: sounds like you need a better telco. :) |
| 02:37.13 | Qwell | intralanman: nah, nobodys go too |
| 02:37.16 | bintut | intralanman: my /etc/asterisk/sip.conf is at http://paste.debian.net/13220 |
| 02:37.36 | Guest^DJ | Lyfe: where i live, only ONE |
| 02:37.40 | bintut | intralanman: my /etc/asterisk/extensions.conf is at http://paste.debian.net/13221 |
| 02:38.04 | intralanman | Guest^DJ: where do you live, that's painful |
| 02:38.05 | intralanman | lol |
| 02:38.20 | bintut | intralanman: i just want to make a call to my friend's local number who in the other country.. |
| 02:38.28 | Lyfe | sorry to hear that, Guest^DJ. Better luck in the future then. |
| 02:39.18 | Lyfe | it's pretty standard that telco's understand callerid though.. i wouldn't be suprised if they just think you're too stupid to do it, and are simply telling their system to set/override one then. |
| 02:41.39 | Guest^DJ | Lyfe: funny thing is every time i ask them, they just simply say NO, no such thing exist |
| 02:41.43 | *** join/#asterisk tengulre11 (n=tengulre@221.11.5.180) |
| 02:43.10 | Lyfe | it's certainly a possible feature. i can tell you that I can set the callerid of an outgoing call on my PRI to another number that doesn't map back to it. |
| 02:43.36 | JT | Guest^DJ: where do you live? |
| 02:45.07 | Guest^DJ | JT: so call freeworld Malaysia |
| 02:45.49 | JT | lol malaysia isn't very free |
| 02:46.02 | Guest^DJ | i know |
| 02:46.04 | Guest^DJ | hahah |
| 02:46.18 | Guest^DJ | ops, is the government listening ? |
| 02:47.14 | teknoprep | holy shit |
| 02:47.19 | teknoprep | i just used what you guys told me |
| 02:47.30 | teknoprep | and set it up using the custom app feature of freepbx |
| 02:47.31 | teknoprep | w0ot |
| 02:47.36 | teknoprep | feels good.. thanx again all |
| 02:51.33 | Guest^DJ | damn, SetCIDNum doesnt work |
| 02:55.25 | *** join/#asterisk profounded (n=pro@ool-44c4eae2.dyn.optonline.net) |
| 02:56.54 | Guest^DJ | Lyfe: i could use some voip provider in the US to pass callerid, any recommendation which one ? |
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| 03:00.55 | teknoprep | voicepulse |
| 03:01.04 | droops | junction networks lets you set it |
| 03:01.15 | intralanman | Guest^DJ: how much traffic are you planning on using? |
| 03:02.34 | Guest^DJ | depending on latency as provider is in the US, rough estimate 200k min |
| 03:10.27 | [Outcast] | anyone here from juction networks? |
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| 03:15.09 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 03:18.03 | *** join/#asterisk Bobcat991966 (n=chatzill@cpe-069-132-138-111.carolina.res.rr.com) |
| 03:19.56 | Bobcat991966 | Hello All, has anybody install Lumenvox speech rec on asterisk 1.2? |
| 03:20.47 | file | Bobcat991966: you should talk to support |
| 03:22.35 | Bobcat991966 | Ya, I have called them and asterisk support many times. I have been trying to find the lumenvox patch for asterisk. |
| 03:22.52 | file | there is a patch that adds speech capability to 1.2, and a binary module |
| 03:22.59 | Bobcat991966 | Yep |
| 03:23.06 | file | Lumenvox should give you a tarball of it |
| 03:23.25 | Bobcat991966 | It took me a week just to find that out and now that I know there is a patch I need to find and download it. |
| 03:23.42 | file | it's distributed with the Lumenvox module |
| 03:24.06 | Bobcat991966 | Lumenvox sent me an email today just before they closed telling me I needed to download it from their website but did not give me a link |
| 03:24.39 | Bobcat991966 | I was hoping somebody on the IRC might know the link and save me from having to wait until morning |
| 03:24.51 | file | nope |
| 03:25.01 | Bobcat991966 | Owell thanks file |
| 03:25.09 | file | there is no link that everyone can use, it's linked to your account I believe |
| 03:25.37 | Bobcat991966 | Hmmm its not on my account, just the licence manager and server |
| 03:25.58 | Bobcat991966 | and of course my licence file |
| 03:27.39 | Bobcat991966 | Is there a lumenvox irc? |
| 03:27.48 | file | haha... |
| 03:28.57 | Bobcat991966 | thats what I thought, I was under the assumtion that lumenvox was a relatively large company with 24 hours support....that what you get with you assume....you make and ASS out of U and ME. |
| 03:29.07 | file | on IRC? no... |
| 03:29.23 | file | I doubt anyone but the developers at Lumenvox know of IRC |
| 03:29.49 | Bobcat991966 | cool, I guess I will just have to wait till tomorrow to resolve this...thanks again file |
| 03:30.12 | file | I'm sitting back and watching sales/support duke it out over this distribution/licensing thing |
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| 03:38.52 | SomeJ | outcast : lookin to see why inbound dids are not working? |
| 03:41.54 | [Outcast] | SomeJ: yes |
| 03:44.46 | [Outcast] | SomeJ: are they having issues? |
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| 03:45.42 | predder | if I record a conversation, where does it get stored by default? |
| 03:47.19 | Strom_C | /var/spool/asterisk/monitor |
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| 03:59.19 | *** join/#asterisk profounded (n=pro@ool-44c4eae2.dyn.optonline.net) |
| 03:59.53 | predder | thankyou Strom_C. |
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| 04:11.49 | intralanman | anyone know of a way to do an ENUM lookup for all numbers in a domain? |
| 04:12.19 | intralanman | long story short, i lost my e164.org login info and can't remember what my block of numbers was |
| 04:14.03 | brookshire | Bobcat991966: did you get the link? |
| 04:14.04 | *** part/#asterisk Elysius (n=Elysius@pool-71-108-163-36.lsanca.dsl-w.verizon.net) |
| 04:14.51 | Bobcat991966 | Brookshire...nope |
| 04:15.23 | Bobcat991966 | by anychance do you know it? |
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| 05:07.09 | RestLessGemini | hello room, anyone with astribank experience? |
| 05:07.57 | CunningPike | RestLessGemini: tzafrir from Xorcom hangs out here.... I think he knows a little about them ;) |
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| 05:10.55 | profounded | anyone got any idea why my sound point 430 is hanging at processing configuartion |
| 05:11.03 | *** join/#asterisk RestLessGemini (n=rLg@202.61.49.248) |
| 05:22.32 | *** join/#asterisk lorinc (n=ang@caracas-2327.adsl.interware.hu) |
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| 05:23.22 | hmmhesays | what happened to lilo? |
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| 05:23.38 | RaYmAn-Bx | hmmhesays: he died. |
| 05:23.53 | hmmhesays | for real? |
| 05:23.55 | RaYmAn-Bx | yes |
| 05:24.07 | RaYmAn-Bx | check freenode.net under news |
| 05:24.33 | L|NUX | CunningPike : hey |
| 05:24.57 | CunningPike | L|NUX: Hi |
| 05:25.08 | L|NUX | how are you doing |
| 05:25.09 | *** join/#asterisk HaMYaI (i=BugKhaM@202.8.86.164) |
| 05:25.26 | CunningPike | L|NUX: Good thanks - wrestling with SugarCRM atm |
| 05:25.52 | sevard | sounds like a worthwhile fight. |
| 05:26.00 | hmmhesays | hmm i see |
| 05:26.04 | L|NUX | CunningPike : okay |
| 05:26.24 | sevard | what's up you greasy old bag of dicks |
| 05:26.35 | hmmhesays | nothing |
| 05:26.38 | hmmhesays | need money |
| 05:26.44 | sevard | sell your body |
| 05:27.17 | HaMYaI | anyone using FC5 with pci_hotplug? |
| 05:27.24 | L|NUX | sevard : i think you are in wrong channel ;) |
| 05:27.24 | L|NUX | E-1 channel 1 : DID no. : 82 31 500 2400 ~ 2699 [DNIS 50024 |
| 05:27.25 | L|NUX | ~ 50026] |
| 05:27.26 | L|NUX | E-1 channel 2 : DID no. : 82 31 500 2700 ~ 2999 [DNIS 50027 |
| 05:27.26 | L|NUX | ~ 50029] |
| 05:27.27 | L|NUX | shit |
| 05:27.28 | L|NUX | sorry |
| 05:27.40 | sevard | you best be sorry, i think you're in the wrong channel. |
| 05:28.14 | *** join/#asterisk mrbnet (n=sureal@CPE-24-163-167-237.mn.res.rr.com) |
| 05:28.38 | sevard | why can't fox play something better than back to back episodes of will and grace |
| 05:28.50 | sevard | why don't they play back to back episodes of family guy instead? |
| 05:28.51 | sevard | bastards. |
| 05:29.26 | RaYmAn-Bx | back to back of anything gets boring after about an hour anyways. |
| 05:29.56 | sevard | CunningPike: these astribanks are a lot cheaper than a tdm2400p, how do they sound? |
| 05:30.25 | sevard | RaYmAn-Bx: But I'd rather take an ice pick to my eyes then watch an episode of will and grace |
| 05:30.36 | CunningPike | sevard: No idea - never used one |
| 05:31.40 | Juggie | good daily show tonight |
| 05:31.44 | Juggie | w/ bill clinton |
| 05:32.10 | wasim | suddenly this cigar isn't all that appealing |
| 05:32.22 | JT | they use usb, of course they're cheap, the astribanks :) |
| 05:32.46 | sevard | wasim: what'chu smoking? |
| 05:33.22 | wasim | sevard: bolivar, royala habana |
| 05:33.22 | sevard | JT: sure, but if they have the same reliability as a 2400p then.. usb.. what's the difference. it's a cheaper interface. |
| 05:33.31 | sevard | i'm pretty curious about these little guys |
| 05:33.57 | wasim | whats the price on them? |
| 05:34.20 | JT | well that's highly unlikely, seeing one is pci and the other is usb |
| 05:35.07 | *** join/#asterisk Stephnie (i=Stephnie@u15157627.onlinehome-server.com) |
| 05:35.38 | sevard | I don't know. I have three camera boxen that are controlled by three types of cameras, one has a pci controller, one has a usb controller, and one is a server for entirelly network based cameras |
| 05:35.49 | sevard | and I see reliability across the board |
| 05:36.03 | Stephnie | if the call is connected with CANREINVITE=YES .... then should the call be disconnected if I stop Asterisk ??? |
| 05:36.44 | Juggie | can reinvite doesnt mean it will definitally reinvite |
| 05:36.57 | Juggie | but assume the re-invite happens, then yes. |
| 05:37.13 | Juggie | you will loose call control, but the call will stay up |
| 05:37.55 | Stephnie | but ... doesnt asterisk connects the caller to the server directly when canreinvite=yes? |
| 05:38.20 | Juggie | it should happen yes |
| 05:38.22 | Stephnie | but the calls get dropped when I stop asterisk... |
| 05:38.25 | *** join/#asterisk Apturaa (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 05:38.35 | Juggie | then it might not be happening |
| 05:38.38 | Juggie | check your sip debug |
| 05:38.41 | Juggie | and check your rtp debug |
| 05:39.06 | Stephnie | okey ... |
| 05:40.40 | Stephnie | reinvite is really happening ..b'coz I get the better call quality with re-invite ... |
| 05:41.07 | Apturaa | how so |
| 05:41.17 | Apturaa | how does it affect the call quality |
| 05:41.33 | X-Rob_ | it could affect _latency_ |
| 05:41.37 | X-Rob_ | but not quality |
| 05:41.56 | Apturaa | so you mean a few dropped packets.. |
| 05:42.01 | *** join/#asterisk mrbnet_ (n=sureal@cust-static-blk197-45.BHI.COM) |
| 05:42.16 | Stephnie | you want to check ? |
| 05:42.17 | X-Rob_ | no, because instead of the audio stream goign phone -> asterisk -> phone, it goes phone -> phone |
| 05:43.06 | Stephnie | my scenerio is |
| 05:43.22 | Apturaa | Stephnie http://www.voip-info.org/wiki-Asterisk+sip+canreinvite3 |
| 05:43.35 | Stephnie | Voxbone DID ==> Asterisk ==> SIP Carrier |
| 05:43.51 | Stephnie | I get good quality with canreinvite=yes |
| 05:44.25 | Stephnie | but I want to check that call shouldnt get dropped if I stop Asterisk ... |
| 05:45.26 | Apturaa | it will get dropped unless anyone else cares to say otherwise |
| 05:45.55 | Juggie | if you stop now |
| 05:45.57 | *** join/#asterisk jeffgus (n=jeffgus@38.119.60.2) |
| 05:45.59 | Juggie | er, 'stop now' |
| 05:46.06 | Juggie | as will close all the calls |
| 05:46.09 | Juggie | of course. |
| 05:46.23 | Juggie | but i believe if the process was to say, terminate, the calls would remain up. |
| 05:46.37 | Juggie | you could test by doing a kill -9 <asteriskpid> |
| 05:46.45 | Juggie | w/ canreinvite=yes |
| 05:46.56 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 05:47.29 | Stephnie | a kill -9 ???? thats a new thing for me ....can I get the more detail ? |
| 05:48.40 | Juggie | 'kill -9 ####' where #### = your asterisk pid |
| 05:48.41 | Stephnie | Juddie: it is a termination.. |
| 05:48.47 | Juggie | will just kill asterisk |
| 05:48.49 | Juggie | like it crashed |
| 05:49.14 | *** join/#asterisk pnlarsson (n=niklas@c83-248-0-248.bredband.comhem.se) |
| 05:49.35 | Stephnie | whats the difference in "stop now" and killing asterisk? :) |
| 05:50.01 | JT | stop now asks nicely |
| 05:50.05 | JT | and it shuts down calls |
| 05:50.22 | JT | -9 terminates the process with no warning |
| 05:50.38 | Juggie | stop now kills all your sip calls |
| 05:50.44 | Juggie | because it tells the peers to hangup |
| 05:50.52 | Juggie | kill -9 should give the end phones no warning |
| 05:50.56 | Juggie | and hence the call should stay up |
| 05:51.34 | Juggie | i've explained this 2-3 times |
| 05:51.42 | Juggie | if your not ready to accept what i've said and try it |
| 05:51.45 | Juggie | then i can do nothing more for you |
| 05:52.43 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it) |
| 05:53.46 | sevard | date |
| 05:53.49 | sevard | grr |
| 05:53.51 | sevard | tired |
| 05:54.13 | Apturaa | Interesting in the old days of Telephony CO techs used to listen in to two party conversations waiting for them to hang up so thay can switch the tip and ring wiring for maintence. |
| 05:54.25 | Stephnie | great! by the way .... if reinvite is happening then.... is a pentium 4 machine with 1 GB ram enough for 100 concurrent calls ? |
| 05:54.43 | Apturaa | less then that. |
| 05:54.49 | Apturaa | I think |
| 05:55.09 | *** join/#asterisk profounded (n=pro@ool-44c4eae2.dyn.optonline.net) |
| 05:55.18 | Apturaa | I was told my dual opteron can handle 84 calls at half a gig. The more cyles the cpu can push the more calls your server can handle. |
| 05:56.29 | Stephnie | I thought I can do load balancing with canreinvite=yes |
| 05:57.41 | CunningPike | Stephnie: You should possibly take a look at OpenSER if you want to go that route |
| 05:58.22 | Stephnie | SIP EXPRESS ROUTER? |
| 05:59.11 | Apturaa | SER is powerfull |
| 05:59.25 | CunningPike | Stephnie: YES |
| 05:59.26 | CunningPike | :) |
| 05:59.33 | *** join/#asterisk tengulre (n=tengulre@61.185.224.66) |
| 05:59.36 | Stephnie | I am gud in asterisk but havent worked in SER :( |
| 05:59.52 | Apturaa | gud? |
| 05:59.57 | Apturaa | :) |
| 06:00.03 | CunningPike | Stephnie: We will probably go to SER instead of another Asterisk server when we reach that point |
| 06:00.14 | Apturaa | CunningPike so when is the first AUG |
| 06:00.29 | CunningPike | Apturaa: ?? |
| 06:00.41 | Apturaa | Did you attned the last LUG at BCIT? |
| 06:01.10 | Stephnie | CunningPike: u mean load balancing ? |
| 06:01.16 | CunningPike | Apturaa: No - I tried to get a AUG up and running in Vancouver, but not a huge amount of interest |
| 06:01.25 | Apturaa | yea |
| 06:01.27 | Apturaa | :) |
| 06:01.29 | Apturaa | news on. |
| 06:01.31 | CunningPike | Stephnie: When our existing server reaches capacity |
| 06:01.39 | *** join/#asterisk tparcina (n=tparcina@lns01-0556.dsl.iskon.hr) |
| 06:01.47 | tparcina | hi channel! |
| 06:02.20 | tparcina | did you all sleap well? :)) |
| 06:03.18 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 06:05.54 | Stephnie | CunningPike: I would like to know more about your setup and SER's existance in a diagram :) |
| 06:06.09 | Stephnie | so that I should go for SER or not |
| 06:06.23 | CunningPike | Stephnie: You'll need to read up on it first |
| 06:06.47 | Stephnie | does it do load balancing inbetween 3 Asterisk servers? |
| 06:06.51 | *** join/#asterisk daysmen3 (n=primus@host86-139-118-16.range86-139.btcentralplus.com) |
| 06:19.46 | *** join/#asterisk profounded (n=pro@ool-44c4eae2.dyn.optonline.net) |
| 06:20.36 | *** join/#asterisk Master_PE (n=masterpe@sd511057f.adsl.wanadoo.nl) |
| 06:21.14 | L|NUX | can some help me with E1 |
| 06:21.25 | L|NUX | i have Sangoma Card which is installed and E1 is installed |
| 06:21.32 | L|NUX | but when it call on number nothing come on * |
| 06:25.02 | Stephnie | whats the official website of SER? |
| 06:25.09 | Strom_C | google.com |
| 06:25.42 | Stephnie | hehehe....official website of everything |
| 06:27.46 | sx-wks | Stephnie: except for .be news |
| 06:28.05 | CunningPike | Stephnie: Google for OpenSER |
| 06:28.13 | CunningPike | Or SIP Express Router |
| 06:28.16 | CunningPike | ~giyf |
| 06:28.21 | jbot | giyf is probably Google Is Your Friend, or see also: STFW |
| 06:28.38 | Stephnie | CunningPike: I need your suggestions first....before I go for SER |
| 06:28.53 | Stephnie | does it do load balancing between 3 Asterisk servers? |
| 06:28.54 | CunningPike | Stephnie: It's 2330 here - sorry |
| 06:29.02 | *** join/#asterisk lowlevel (n=Stuart@CPE0050ba71c82f-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 06:29.15 | CunningPike | Wish my eyes could close......... |
| 06:29.22 | Stephnie | cant u even say YES or NO ? |
| 06:29.23 | Stephnie | :p |
| 06:29.40 | Strom_C | cant you even do RESEARCH? |
| 06:30.08 | Strom_C | i mean, honestly, if you're going to be an irritating yo-yo about it... |
| 06:30.20 | CunningPike | Stephnie: Right from the goddam home page: "It is customizable, being able to feature as fast load balancer" |
| 06:30.36 | Stephnie | thanks CunningPike |
| 06:30.37 | CunningPike | Took me 10 seconds |
| 06:30.41 | CunningPike | Sheesh |
| 06:31.06 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 06:31.06 | Stephnie | Strom_C : experts are for suggestions....and I hope u r not one of them... |
| 06:31.12 | *** join/#asterisk Corydon76-home (i=peuce@pdpc/supporter/sustaining/Corydon76-home) |
| 06:31.12 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 06:31.20 | Stephnie | thanks Pike!!! |
| 06:31.32 | CunningPike | Stephnie: You're welcome-ish ;) |
| 06:31.40 | Strom_C | thank you come again |
| 06:31.48 | *** join/#asterisk jaike (n=jaike@203.115.188.120) |
| 06:32.54 | *** join/#asterisk MadRio (n=rio@80.255.176.161) |
| 06:33.08 | CunningPike | What's the relationship between SER and OpenSER, anyway |
| 06:34.10 | CunningPike | Ah - it's a fork - sort of....... |
| 06:34.15 | Strom_C | one has fangier zealots, I'd imagine |
| 06:34.16 | L|NUX | just SER created Relation ship :P |
| 06:34.17 | L|NUX | hehe |
| 06:34.19 | L|NUX | joking |
| 06:34.29 | L|NUX | CunningPike : and your are right its fork |
| 06:34.43 | CunningPike | L|NUX: Aye - just reading now..... |
| 06:35.52 | *** join/#asterisk Bobcat991966 (n=chatzill@cpe-069-132-138-111.carolina.res.rr.com) |
| 06:36.21 | CunningPike | OK - really going this time - later dudes |
| 06:36.45 | *** join/#asterisk Rahail (n=dd@ip66-104-51-162.z51-104-66.customer.algx.net) |
| 06:38.28 | *** join/#asterisk Blackjck (i=johnny@70-34-9-66.lmdaca.adelphia.net) |
| 06:39.32 | Blackjck | hey all, anyone know if its possible to use telrad digital handsets with asterisk? or suggestions where to look? I can't find any info. |
| 06:39.38 | *** join/#asterisk darkskiez (i=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 06:40.32 | JT | Blackjck: are they isdn phones? |
| 06:41.49 | Blackjck | JT: no.. 3015F for use with the telrad/conegy BX digital pbx |
| 06:41.51 | *** join/#asterisk adorah (n=admin@84.94.144.7.cable.012.net.il) |
| 06:42.35 | JT | if they're a proprietry digital handset, then they won't work directly with asterisk |
| 06:43.17 | Blackjck | ah.. is there a means of interfacing them w/ asterisk without the original PBX? |
| 06:44.03 | JT | maybe if you are an electrical engineer |
| 06:44.23 | JT | it's like anything, you don't run NEC handsets with a Nortel PBX |
| 06:44.27 | Blackjck | not that good of one... =P |
| 06:45.19 | Blackjck | yeah I was under the impression for some reason, maybe i misread an article, that since asterisk is software based there was a way of emulating the propreitary protocols to support, for example, nortel handsets |
| 06:45.56 | JT | there might be something that somewhat supports nortel handsets, not sure |
| 06:46.04 | JT | well you need the correct hardware interface too |
| 06:46.33 | Blackjck | yeah I figured that... can't find anything anywhere about telrad though, as it's not a very popular pbx it seems. |
| 06:47.36 | Strom_C | Blackjck: if everything could be done in software, then there would be no market whatsoever for T1 cards and analog telephone line interface cards |
| 06:48.42 | Blackjck | i didn't mean to imply that it was 100% software. obviously hardware that would be required, and likely even adapters to support different wiring for different handset manufactures |
| 06:48.54 | Blackjck | if such a thing existed. |
| 06:49.26 | Blackjck | i'm a n00b to asterisk, but I'm not an idiot! |
| 06:50.01 | Strom_C | there's some company whose name I forget that makes adapters for all sorts of proprietary digital telephone sets, but honestly, from experience, you will spend more money in labor trying to get your hackjob working than you will in parts just buying good-quality SIP telephones |
| 06:51.04 | *** join/#asterisk svenadh (n=sven@213.217.93.246) |
| 06:51.12 | Blackjck | ah... bummer. |
| 06:51.12 | *** join/#asterisk cjk (n=cjk@80.92.64.103) |
| 06:51.18 | cjk | hi |
| 06:51.24 | Strom_C | balls |
| 06:51.42 | Strom_C | Blackjck: plus the added difficulty of getting support down the line |
| 06:51.53 | Blackjck | you have any personal favs for business quality SIP phones w/ multi line conferencing and full duplex speaker? |
| 06:51.54 | cjk | i am using realtime but after changing the password of a "friend" i can no longer make calls... reloading iax helps... is there a better way? |
| 06:52.15 | Strom_C | Blackjck: my personal favorite is the cisco 7960 |
| 06:52.47 | Strom_C | some people like the polycoms, but I find that the polycoms do all sorts of DSP which supposedly makes the sound more "natural" but which sounds like ass through a cheese grater to me |
| 06:52.57 | Blackjck | hahahaha |
| 06:55.12 | Blackjck | have you tested the QoS control on the built in port on the 7960? |
| 06:55.24 | Strom_C | not personally, no |
| 06:55.31 | Blackjck | heard anything about it? |
| 06:55.58 | Strom_C | no - realistically, QoS isn't going to become an issue until you go out over your DSL connection |
| 06:56.19 | Strom_C | if you've got QoS problems on your LAN, then holy god does your network suck |
| 06:56.42 | Strom_C | and no amount of built-in anything is going to solve that problem |
| 06:56.43 | Blackjck | i'm thinking more in an office w/ 30 handsets... we work w/ digital media and guys routinely copy gb+ files over the lan |
| 06:56.47 | *** join/#asterisk HaMYaI (n=hamyai@ppp-58.8.9.216.revip2.asianet.co.th) |
| 06:57.03 | Blackjck | just wondering if their phone call will go to crap if they are in the middle of xfering a file and a call comes in |
| 06:57.09 | Strom_C | Blackjck: you've got two ethernet drops at each station, right? |
| 06:57.17 | Blackjck | nope. old building, just one. |
| 06:57.21 | *** join/#asterisk myshenka (n=spamyous@82.153.170.213) |
| 06:57.24 | Strom_C | *smacks forehead* |
| 06:57.30 | Blackjck | hence the interest in the QoS on the port =) |
| 06:57.40 | HaMYaI | Strom_C: are you using FC5? |
| 06:57.42 | Strom_C | who's the nub who decided "let's only pull one cat6 drop to each station!"? |
| 06:57.45 | Strom_C | HaMYaI: no |
| 06:58.04 | Blackjck | some guy that wired an old warehouse about 15 years ago. |
| 06:58.10 | Strom_C | 15 years? |
| 06:58.17 | Blackjck | that'd be my guess. |
| 06:58.17 | Strom_C | what the hell is your network running on? |
| 06:58.19 | Strom_C | cat4? |
| 06:58.21 | Strom_C | er |
| 06:58.23 | Strom_C | cat3? |
| 06:58.25 | Blackjck | no cat 5 was around |
| 06:59.08 | HaMYaI | Strom_C: I've got to restart my zaptel everytime I reboot my system to get the /proc/zaptel/1 for the right card |
| 06:59.12 | Blackjck | it was the equivalent of of what, 4Gbs FC now? |
| 06:59.32 | Strom_C | HaMYaI: what hardware do you have? |
| 06:59.34 | Blackjck | it's a "historic" building, so we're not supposed to run new conduit. |
| 06:59.45 | Strom_C | Blackjck: you dont have to run new conduit |
| 06:59.51 | Strom_C | just pull more cable |
| 07:00.13 | HaMYaI | Strom_C: TE110P and TDM400 on the P5LD2 SE mainboard |
| 07:00.24 | Blackjck | if it were that simple, don't you think we'd have done it already? =) |
| 07:00.53 | Strom_C | HaMYaI: so just reconfigure your modules so that the TE110P loads first |
| 07:00.53 | Strom_C | Blackjck: you're not using enough lube :) |
| 07:01.03 | Blackjck | ultimately anything is possible, it's weighing the cost/benefit. at the moment, if we could rely on the QoS in a handset switch, that'd be worth several thousand |
| 07:01.52 | Strom_C | HaMYaI: am I correct in assuming that it's the TDM400 that's currently autoloading first? |
| 07:01.55 | HaMYaI | Strom_C: that's what I did , in /etc/modprobe.d/zaptel, but it really depends on which card was detected first in the kernel |
| 07:02.31 | Strom_C | Blackjck: well, i guess it could theoretically work |
| 07:04.06 | HaMYaI | Strom_C: yeah, you're right but if TE110P is detected first in dmesg, it just doesn't have any impact |
| 07:04.14 | Blackjck | ill dig around for some reviews |
| 07:04.26 | Blackjck | what makes the 7960 your fav? |
| 07:04.39 | Strom_C | excellent sound quality and excellent build quality |
| 07:04.57 | Blackjck | and snom? |
| 07:05.04 | Strom_C | it's like there is literally /nothing/ separating you from the audio stream |
| 07:05.20 | HaMYaI | Strom_C: any other ways you can think of? |
| 07:05.20 | Strom_C | the snoms ive played with have lower quality switches in the keypad |
| 07:05.26 | Strom_C | plus the handsets are all weird shapes |
| 07:05.35 | Blackjck | the 7940 comparable, other than a few less buttons? |
| 07:05.44 | Strom_C | HaMYaI: no, there is no other way besides ensuring the modules load in a specific order |
| 07:05.50 | Strom_C | Blackjck: yes, besides the buttons they are the same |
| 07:07.14 | Blackjck | great. thx for your input! |
| 07:07.25 | Strom_C | welcome |
| 07:07.31 | Strom_C | and thats one thing i forgot to mention |
| 07:07.35 | HaMYaI | Strom_C: don't understand why kernel detects device in different manners |
| 07:07.55 | Strom_C | the handsets on the ciscos are the ONLY ones I've ever seen on voip phones that are actually designed with ergonomics in mind |
| 07:08.21 | Strom_C | you can cradle them between your shoulder and ear just like you could with the old Western Electric G-type handsets |
| 07:08.30 | Strom_C | HaMYaI: I am not a linux person (tm) |
| 07:08.41 | Makenshi | Strom, i can do that with our optipoint handsets |
| 07:08.52 | Strom_C | show me the optipoint handsets |
| 07:08.57 | Aurs | cradle them between your shoulder and ear != ergonimics :) |
| 07:09.10 | Strom_C | Aurs: no, it's not something that should be done often |
| 07:09.14 | *** join/#asterisk profounded (n=pro@ool-44c4eae2.dyn.optonline.net) |
| 07:09.25 | Strom_C | but if you need to do it momentarily while you rustle for papers, it's nice to have that option |
| 07:09.35 | Aurs | yes |
| 07:09.37 | Strom_C | versus the alternative where the handset slips off and falls all over the place |
| 07:09.55 | Makenshi | Strom, go look it up or something |
| 07:09.58 | tparcina | queues - does anybody use them? |
| 07:10.03 | Aurs | speakerphone is an opt |
| 07:10.09 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 07:10.22 | Aurs | tparcina: yes |
| 07:10.59 | Strom_C | Makenshi: I can't find a good picture of one that's /not/ on the phone |
| 07:11.00 | tparcina | Aurs: when * calls agent (because he has a call in queue for him), how can you change language for that call? |
| 07:11.26 | Makenshi | Strom, you shouldn't make such sweeping statements when you obviously have not seen all the handsets |
| 07:11.41 | tparcina | Strom_C: you are pretty good with asterisk, maybe you know this one |
| 07:11.57 | *** join/#asterisk tengulre (n=tengulre@222.90.66.156) |
| 07:12.01 | Strom_C | Makenshi: hence why i qualified with "that I've ever seen" |
| 07:12.07 | tparcina | Strom_C: when * calls agent (because he has a call in queue for him), how can you change language for that call? |
| 07:12.12 | *** join/#asterisk af_ (n=af@ip-202-199.sn2.eutelia.it) |
| 07:12.24 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:12.29 | Strom_C | so you should probably read more carefully before you start calling people out on statements they never made :) |
| 07:12.38 | Strom_C | tparcina: I have no idea |
| 07:12.48 | Aurs | tparcina: don't know |
| 07:12.50 | Strom_C | tparcina: I have only worked in an English-only environment |
| 07:13.23 | tparcina | to bad, i'm trying to solve that for a couple of days - still unsuscesfull :(( |
| 07:13.47 | *** join/#asterisk smash- (n=smash@c-24-22-112-218.hsd1.or.comcast.net) |
| 07:14.58 | Strom_C | anyway, I looked at all 22 pages of google image search results for "optipoint" and none of them show a handset to the point where I can tell what the dimensions are |
| 07:15.34 | tparcina | guys, maybe you have answer for this one. call is established, person is in AA menu. when he dials a number extension rings. Problem is that caller doesn't hear ringing.i have tried with "r" in dial command - didn't help. i have tried with ringing application, result was the same |
| 07:17.23 | Strom_C | how is the call coming in? |
| 07:18.25 | tparcina | Strom: from SIP |
| 07:19.05 | *** join/#asterisk firestrm (n=no@S01060013105c1c69.gv.shawcable.net) |
| 07:19.24 | Strom_C | what kind of endpoint? |
| 07:19.25 | tparcina | Strom: you think that has to do anything with tehnology (SIP, ZAP, IAX)? |
| 07:19.42 | Strom_C | what codec? |
| 07:19.48 | tparcina | endpoint also SIP |
| 07:20.16 | tparcina | codecs are ulaw - local network - doing testing, later will mouve to ZAP, PRI |
| 07:21.49 | Strom_C | odd |
| 07:21.56 | *** join/#asterisk Corydon76-home (i=six@pdpc/supporter/sustaining/Corydon76-home) |
| 07:21.56 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 07:21.58 | Strom_C | and the console shows "SIP/xxx is ringing"? |
| 07:22.00 | *** join/#asterisk dashu (n=root@p549C6AAC.dip.t-dialin.net) |
| 07:22.27 | dashu | hey can anyone help me with capi ? |
| 07:23.02 | tparcina | yes. and when i tried with aplication ringing (with and without app wait afther it) it didn't help also. |
| 07:23.30 | firestrm | can anyone help? i need some brainstorming, im all out ideas on a problem im having... Upgraded to the latest trixbox on a newer faster machine, udes EXACLY the same settings as i did on the old box, (which was working fine, at least on the IAX side of things) but now on the new machine, all my iax channels have severe static on them.. makes them pretty much unusable.. Ive carefully set all the gains on my fxs/fxo ports, no joy.. |
| 07:23.52 | hads | <PROTECTED> |
| 07:24.01 | Strom_C | firestrm: I'll tell you your first problem. You didn't read the topic. |
| 07:24.06 | Strom_C | tparcina: hmmmm |
| 07:24.16 | Strom_C | tparcina: what if you try a softphone |
| 07:24.52 | firestrm | hmm.. your right.. i was just kinda used to coming here and getting help.. sorry.. i'll move along.. |
| 07:25.12 | Strom_C | firestrm: what kind of hardware |
| 07:25.33 | tparcina | Strom_C: i have tried to make call from softphone (SJ phone) and hard phone (Cisco 7940) - it's the same |
| 07:26.02 | Strom_C | tparcina: what version of asterisk> |
| 07:26.16 | Strom_C | and what happens, just for giggles, if you try an iax softphone |
| 07:26.31 | firestrm | Strom_C, dell dimension 2400, with a tdm 400 1 fxo, 3fxs |
| 07:26.51 | Strom_C | firestrm: so you changed the server AND the hardware? |
| 07:26.51 | tparcina | Strom_C: Asterisk 1.2.5 |
| 07:26.54 | Strom_C | er |
| 07:27.01 | Strom_C | the server AND the software |
| 07:27.35 | tparcina | Strom_C: didn't tried IAX. i won't use IAX in production so didn't tried it in test version |
| 07:27.40 | firestrm | srry, trixbox 1.2.. checking software versions.. brb |
| 07:27.54 | Strom_C | well lets just test with iax to see what happens |
| 07:28.20 | tparcina | Strom_C: ok, thank you for trouble shooting |
| 07:28.43 | Strom_C | any time |
| 07:28.48 | Strom_C | this is an interesting problem to me :) |
| 07:30.10 | dashu | http://pastebin.ca/175515 can anyone help me with this ? :o |
| 07:31.10 | *** join/#asterisk Ahrimanes (n=michael@81.7.159.2) |
| 07:31.30 | tparcina | Strom_C: when you mentioned that diferent tehnology could act diferently i Have tested with call from ZAP. now, when call is comming from ZAP interface and when in AA menu he dials extension, ZAP caller hears ringing :)) and that is what I need because all cals in this menu will come from ZAP - I was just testing with SIP :)) |
| 07:31.52 | Strom_C | tparcina: yeah, weird. i'm not sure what the issue is. |
| 07:31.56 | tparcina | Strom_C: anyway, should this be reported as bug? |
| 07:32.06 | Strom_C | tparcina: you're running kind of an old version of asterisk |
| 07:32.15 | Strom_C | i'd try the newest one and see if that fixes it first |
| 07:32.19 | tparcina | Strom_C: yes, i know |
| 07:33.07 | tparcina | Strom_C: good point you got there. anyway, do you have a ling to page where they instruct when, why and how to report a bug? |
| 07:33.24 | Strom_C | i think it's at bugs.digium.com |
| 07:33.27 | Strom_C | though i could be wrong |
| 07:34.02 | *** join/#asterisk angryuser (n=magasin@i03v-213-44-169-43.d4.club-internet.fr) |
| 07:34.13 | angryuser | hi |
| 07:34.50 | angryuser | does anyone know irc channel dedicated to hylafax? |
| 07:37.45 | firestrm | Strom_C, asterisk 1.2.11 svn 40948, zaptel 1.2 |
| 07:38.13 | dashu | anyone knows what this line does ? exten => s,1,Dial,CAPI/@5409618:b |
| 07:38.23 | Strom_C | firestrm: yeah, but i asked you whether you replaced the server hardware too |
| 07:38.41 | Strom_C | dashu: there is obviously no one here right now who knows about or cares about CAPI |
| 07:39.05 | dashu | :o and why ? |
| 07:39.14 | dashu | i thought capi is the usual isdn thing |
| 07:39.32 | firestrm | Strom_C, yes hardware changed from a dell poweredge 4300 to a dell dimension 2400, 512m ram, 80 meg hdd. Turned off all unused perihials in bios, no usb, no sound, etc.. |
| 07:39.35 | dashu | whatelse can u use instead of capi ? |
| 07:39.49 | angryuser | vISDN |
| 07:40.08 | angryuser | only for Cologne chip's |
| 07:40.17 | angryuser | and euro lines |
| 07:40.17 | orlock | 80 meg? |
| 07:40.23 | *** part/#asterisk Blackjck (i=johnny@70-34-9-66.lmdaca.adelphia.net) |
| 07:40.41 | Strom_C | firestrm: pastebin the output of cat /proc/interrupts/ |
| 07:40.44 | Strom_C | er |
| 07:40.48 | Strom_C | /proc/interrupts |
| 07:40.53 | firestrm | roger.. brb |
| 07:42.31 | tparcina | does anybody know how does asterisk calculate estimiate time waiting? |
| 07:42.45 | tparcina | does anybody know how does asterisk calculate estimiate time waiting - in queue |
| 07:43.00 | Strom_C | tparcina: it takes an average of the last fifty dead hookers |
| 07:43.06 | firestrm | Strom_C, thanks.. i see the problem right here.. wctcm and eth 0 sharing.. now i just have to figure out how to change.. |
| 07:43.24 | tparcina | Strom_C: dead hookers? |
| 07:43.32 | Strom_C | tparcina: it's a joke |
| 07:43.50 | Strom_C | firestrm: please tell me you're not actually running a business on trixbox |
| 07:44.03 | tparcina | Strom_C: i thought that I don't understand something on english :)) |
| 07:44.34 | Strom_C | tparcina: "hooker" is slang for "prostitute" |
| 07:45.18 | firestrm | lol.. no... just my home box.. it mostly to keep the telemarketers away.. they hate ivr's.. and a 1-800 did ait a bad addition to any houshold too :) |
| 07:45.18 | tparcina | anyway, there should be some calculation. does anybody know how does it look like? |
| 07:46.00 | tparcina | Strom_C: thank you, i know what hooker means (i was in Baltimore for 3 and 1/2 months ;)) - but I just ges that it also means someting else, since you mentioned it :)) |
| 07:47.04 | firestrm | Strom_C, i think my little dell box would have a meltdown if i tried to run a business on it.. but it does impress the girls.. somthing different about a guy that has a toll free home number :) |
| 07:47.28 | Strom_C | firestrm: i had three toll-frees four years ago :) |
| 07:47.43 | Strom_C | and they rang my home phone, my mobile, and my fax, respectively |
| 07:47.44 | *** join/#asterisk S^P (n=ss@203.81.196.20) |
| 07:47.50 | Strom_C | actually, they still do :) |
| 07:48.44 | Rahail | Strom_C can you help me with this |
| 07:49.00 | Rahail | at this moment now one using my pbx however its saying 32 sip channel active |
| 07:49.11 | angryuser | is is possible to detect busy line in asterisk and automaticly use another one? |
| 07:49.14 | Rahail | 16 from my extension and 16 from my provider |
| 07:49.18 | Strom_C | pastebin the output of "show channels" |
| 07:49.21 | Strom_C | angryuser: yes |
| 07:50.07 | firestrm | Strom_C, great.. my bios has absolutly no irq/cardslot assignment ability.. this could get interesting.. |
| 07:51.13 | Strom_C | angryuser: read the following |
| 07:51.15 | Strom_C | ~docs |
| 07:51.17 | jbot | hmm... docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 07:51.19 | Strom_C | ~book |
| 07:51.21 | jbot | book is, like, a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 07:51.25 | angryuser | k |
| 07:52.18 | tparcina | ~book |
| 07:52.20 | jbot | book is, like, a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 07:52.26 | Rahail | Strom_C what is the pasbin site add |
| 07:52.33 | *** join/#asterisk vgster (n=vgster@170.252.64.1) |
| 07:52.51 | Rahail | ?pastbin |
| 07:52.56 | Rahail | ?? pasbin |
| 07:53.04 | Strom_C | it's spelled PASTEBIN |
| 07:53.06 | Strom_C | PASTE BIN |
| 07:53.09 | tparcina | Strom_C: who make this ~ thing work? (i'm not so familiar with irc) |
| 07:53.10 | Strom_C | can't you read? |
| 07:53.13 | Strom_C | ~pb |
| 07:53.15 | jbot | pb is, like, a place to paste your stuff without flooding the channel - try http://pastebin.com/ (BROKEN AND SUCKING NUTS), or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/, or http://paste.lisp.org/ for the lisp/scheme nuts, or http://paste-it.net |
| 07:53.34 | Rahail | ok |
| 07:53.36 | Rahail | here you go |
| 07:53.37 | Rahail | http://pastebin.ca/175927 |
| 07:54.03 | firestrm | Strom_C, do you know of any way to reassign an irq that a tdm400 is using.. I suspect im pooched, but you seem to be "strong in the ways of the force", any home or am i buying yet another peecee tomorrow? |
| 07:54.41 | Strom_C | firestrm: I don't really know. I'm a telephony person. I hate computers. |
| 07:55.05 | firestrm | s/home/hope.. its tough typing with big bandages on my hands (long story.. suffice it to say newborns and powertools dont mix) |
| 07:55.18 | x86 | Strom_C: lol |
| 07:55.59 | Strom_C | Rahail: just restart asterisk |
| 07:56.10 | Strom_C | sounds like maybe your phone and/or your provider is fucked |
| 07:56.28 | firestrm | Strom_C, Im with you on that.. except im a radio communications guy, computers hate me.. |
| 07:57.22 | S^P | Is there any setting of DTMFmode in ZAP channels? |
| 07:58.15 | Strom_C | S^P: no |
| 07:58.21 | S^P | after connecting to a zap channel, asterisk IVR donn't accept any key. |
| 07:58.29 | *** join/#asterisk tengulre11 (n=tengulre@221.11.5.180) |
| 07:58.41 | S^P | I press key but it will continue playing IVR menu. |
| 07:58.50 | Rahail | Strom_C i think its my phone becuase I have other extension |
| 07:59.02 | Rahail | its not doing it do you know what I should look for on my ATA |
| 07:59.33 | Strom_C | Rahail: I have no clue what you're asking me. |
| 07:59.45 | Rahail | http://pastebin.ca/175927 |
| 07:59.48 | Rahail | about this issue |
| 08:00.45 | Strom_C | Rahail: I know what the issue is |
| 08:00.57 | Strom_C | Rahail: but your last sentence just doesn't make any fucking sense |
| 08:01.23 | dashu | what is zaptel for ? |
| 08:01.28 | Rahail | ops my bad .... |
| 08:01.43 | Rahail | I am saying I have also other extension that we use make call ... |
| 08:02.08 | Rahail | so there not showing at sip channel so I am guessing something up with my ATA |
| 08:02.22 | Rahail | so what should I look on ATA to get this thing Resolv |
| 08:02.44 | Strom_C | Rahail: i still don't understand you. |
| 08:02.59 | Strom_C | I give up. |
| 08:03.02 | Rahail | :( |
| 08:03.06 | Rahail | let me try again... |
| 08:03.12 | dashu | what is zaptel for ? :p |
| 08:03.20 | Rahail | I think its something wrong with my ATA |
| 08:03.26 | Strom_C | dashu: GOOGLE |
| 08:03.32 | Rahail | so what option should i look for to stop that problem |
| 08:03.41 | Strom_C | Rahail: use a sledgehammer and then smash it to pieces |
| 08:03.58 | x86 | Strom_C: why do you even try? :) |
| 08:04.36 | Strom_C | x86: because I'm insane :) |
| 08:04.50 | x86 | seems to be highly likely ;) |
| 08:05.39 | x86 | either that or you have a wish to have a coronary explode heh |
| 08:05.50 | L|NUX | can some one help me with E1 + * |
| 08:06.07 | *** join/#asterisk Jedirl (n=asdf@213.162.200.226) |
| 08:06.09 | Jedirl | Hello |
| 08:06.22 | Jedirl | I'm getting: |
| 08:06.22 | Jedirl | Sep 19 09:58:44 ERROR[23408]: chan_zap.c:7017 mkintf: Channel 16 is reserved for D-channel. |
| 08:06.22 | Jedirl | Sep 19 09:58:44 ERROR[23408]: chan_zap.c:10311 setup_zap: Unable to register channel '193-207' |
| 08:06.50 | Strom_C | Jedirl: you screwed up in zapata.conf |
| 08:06.58 | Jedirl | I guess, but I don't know why |
| 08:07.09 | Jedirl | as my zaptel.conf channel ranges match my zapata.conf's |
| 08:07.10 | Strom_C | do you understand what d-channels are? |
| 08:07.23 | Jedirl | yup |
| 08:07.25 | Jedirl | signaling channels |
| 08:07.39 | Strom_C | and why you can't assign b-channel type definitions to them in zapata.conf? |
| 08:07.55 | Jedirl | But I haven't! |
| 08:07.56 | L|NUX | i am trying to configure sangoma dual e1/t1 card with e1 connected to the box |
| 08:08.06 | L|NUX | but when i call call not coming to asterisk |
| 08:08.13 | Strom_C | Jedirl: pastebin your zapata.conf |
| 08:08.14 | Jedirl | Strom_C: my zapata.conf matches my zaptel.conf |
| 08:08.20 | S^P | Strom_C: using mobile if i call my PSTN line, which is connected to asterisk, it play IVR but if i press any digit there is no activity and seems like asterisk is not reciving digits. |
| 08:08.21 | Strom_C | pastebin |
| 08:08.23 | Strom_C | your |
| 08:08.25 | Strom_C | zapata.conf |
| 08:08.31 | S^P | and this only happens with cell phones dillaing in. |
| 08:08.50 | tengulre11 | what 's the Originate mean in asterisk maanger interface? |
| 08:08.53 | Jedirl | http://pastebin.ca/175932 |
| 08:08.58 | S^P | if I use any hand set every thing works fine. |
| 08:09.07 | Strom_C | S^P: what happens if you call a regular phone from a cellphone and try and touchtone at yourself |
| 08:09.12 | L|NUX | Jedirl : can you help me |
| 08:09.14 | L|NUX | any one |
| 08:09.20 | Jedirl | tengulre11: the same as /var/spool/asterisk/outgoing/*.call :) |
| 08:09.29 | tzafrir | Jedirl, xpp/genzaptelconf could be hamdy to generate a reference working zaptel.conf and zapata.conf |
| 08:09.32 | Rahail | Strom_C any other suggestion |
| 08:09.39 | tzafrir | s/hamdy/handy/ |
| 08:09.53 | Jedirl | But my zapata.conf was working |
| 08:09.58 | Jedirl | until I added the 62 Zap group |
| 08:10.14 | Jedirl | and it matches exactly the span definition in zaptel.conf |
| 08:10.15 | Strom_C | why group 62? |
| 08:10.22 | Strom_C | why not something sane like group 5 |
| 08:10.34 | Strom_C | L|NUX: oh for fuck's sake |
| 08:10.39 | *** part/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 08:10.42 | Strom_C | L|NUX: is it a PRI? |
| 08:10.54 | tengulre11 | Jedirl: thank you very much!! |
| 08:11.12 | Jedirl | Strom_C: I have more than one PRI Zap groups |
| 08:11.16 | *** part/#asterisk svenadh (n=sven@213.217.93.246) |
| 08:11.44 | L|NUX | yeah |
| 08:11.45 | L|NUX | its PRI |
| 08:11.54 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 08:11.58 | *** join/#asterisk jpsaman (n=jpsaman@ip545278d7.speed.planet.nl) |
| 08:12.21 | Strom_C | L|NUX: so do your settings match the telco? can you make outbound calls? |
| 08:12.25 | Jedirl | this is very strange |
| 08:12.35 | tengulre11 | how to using manager interface to dial other SIP peer? like SIP2001->SIP2002. |
| 08:12.42 | L|NUX | no idea about that :( |
| 08:12.47 | L|NUX | its korean teleco |
| 08:13.28 | tengulre11 | is it redirect? |
| 08:13.34 | tengulre11 | is it redirect function? |
| 08:13.43 | tengulre11 | s/function/action |
| 08:13.54 | Strom_C | L|NUX: so call the telco and get the settings and then come back |
| 08:14.03 | L|NUX | tengulre11 : i think so |
| 08:14.06 | L|NUX | okies |
| 08:14.07 | L|NUX | sir |
| 08:14.42 | Strom_C | oh dont worry |
| 08:14.47 | tengulre11 | L|NUX: thanks |
| 08:14.50 | tengulre11 | I see. |
| 08:14.56 | Strom_C | come back in ten minutes and I'm sure I'll be explaining to someone what a telephone is |
| 08:15.11 | *** join/#asterisk UnderMine (n=paddy@host81-149-176-190.in-addr.btopenworld.com) |
| 08:15.17 | UnderMine | morning |
| 08:15.20 | L|NUX | tengulre11 : btw what should i ask with teleco |
| 08:15.30 | tengulre11 | :( |
| 08:15.48 | Strom_C | L|NUX: framing and line coding format |
| 08:15.56 | L|NUX | okies boss |
| 08:16.29 | x86 | in the US it's usually B8ZS / ESF |
| 08:16.34 | tengulre11 | any ideas? |
| 08:16.37 | x86 | sometimes AMI / D4 though |
| 08:16.53 | x86 | not sure about north korea heh |
| 08:17.04 | L|NUX | well its korean then i am sure it should be eruoisdn |
| 08:17.11 | Strom_C | NO NO NO |
| 08:17.14 | Strom_C | not signaling |
| 08:17.18 | x86 | they prolly use KIM / NUKE |
| 08:17.18 | Strom_C | FRAMING AND LINE CODING |
| 08:17.24 | Strom_C | I give up |
| 08:17.32 | Strom_C | dear everyone: |
| 08:17.43 | Strom_C | you must be THIS EXPERIENCED WITH TELEPHONY to use a PBX |
| 08:17.46 | Strom_C | [draws line] |
| 08:17.51 | x86 | hahahaha |
| 08:19.22 | Strom_C | ~book |
| 08:19.23 | jbot | book is, like, a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 08:19.23 | Strom_C | ~docs |
| 08:19.25 | jbot | docs is, like, Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 08:19.30 | Strom_C | GO READ PLZKTHX |
| 08:19.44 | L|NUX | ok boss |
| 08:20.13 | *** join/#asterisk HaMYaI (i=BugKhaM@202.8.86.164) |
| 08:20.56 | HaMYaI | hi, how do we add a user for oh323? |
| 08:21.32 | Strom_C | yes, that's exactly the button you press |
| 08:21.42 | x86 | heh |
| 08:21.55 | x86 | it worked for his purpose ;) |
| 08:21.58 | Strom_C | hahaha |
| 08:22.02 | *** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135) |
| 08:24.28 | *** join/#asterisk HaMYaI (i=BugKhaM@202.8.86.164) |
| 08:24.45 | x86 | he hit the wrong button... damn |
| 08:26.34 | Strom_C | heh wow |
| 08:27.59 | *** join/#asterisk darkskiez (n=mbryars@194.247.78.146) |
| 08:30.00 | x86 | if you were riding through the desert, why would you care what the hell the horse's name was? moreso, why would you even care to write a song about the lack of a name for the horse? |
| 08:30.47 | Strom_C | because you name your band after an entire pair of continents? |
| 08:31.16 | x86 | eh? |
| 08:31.36 | Strom_C | you're talking about "Horse with no name" by "America", right? |
| 08:31.37 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 08:31.43 | x86 | yeah |
| 08:31.52 | x86 | well, didnt know "america" was the artist |
| 08:32.02 | x86 | now i get your pun ;) |
| 08:32.10 | Strom_C | welcome to nineteen seventy-something-or-other |
| 08:32.13 | x86 | touche ;) |
| 08:32.48 | firestrm | Strom_C, fxo now on own interrupt.. problem remains.. DOOH!! |
| 08:33.02 | firestrm | er tdm-400 i mean.. |
| 08:33.11 | Strom_C | firestrm: how old is the card? |
| 08:33.34 | firestrm | about a year maybe 2.. |
| 08:33.51 | Strom_C | thats a 100% difference |
| 08:33.56 | Strom_C | is it a year, or is it two |
| 08:33.59 | firestrm | hmm... let me think.. |
| 08:34.34 | firestrm | probbly closer to 1.5.. purchased feb 2005 |
| 08:34.54 | Strom_C | ok, so you're out of luck on the free install support then |
| 08:35.03 | firestrm | oh ya.. |
| 08:35.07 | *** part/#asterisk jaike (n=jaike@203.115.188.120) |
| 08:35.19 | Strom_C | what about using the old box with the new software just for shits and giggles? |
| 08:35.45 | firestrm | but i really doubt its the card, calls though the fxo are clear as a bell.. even incoming did is clear.. just outgoing sucks.. |
| 08:36.26 | Strom_C | what are you calling out from? |
| 08:36.48 | firestrm | Strom_C, i guess i could.. i was hoping to isolate the problem before jumping off into another days adventure loading another box |
| 08:36.58 | Strom_C | what are you calling out from? |
| 08:37.43 | firestrm | calling out from my * box to a voip provider in edmonton link2. i think ther are called.. |
| 08:37.48 | Strom_C | NO NO NO |
| 08:37.56 | Strom_C | what are you using to PLACE THE CALL |
| 08:38.01 | Strom_C | a phone? a glue gun? |
| 08:38.10 | Strom_C | pure telepathy? |
| 08:38.50 | firestrm | ahh.. ok.. mind meld.. J/K. im using one of my fxs ports on the tdm card, with my phone attached |
| 08:39.26 | Strom_C | and calls through that sound just fine when calling out through the FXO port as well? |
| 08:40.13 | firestrm | Strom_C, perfect, i can even recieve faxes.. |
| 08:40.28 | Strom_C | so then it really ISN'T a TDM400 problem at all |
| 08:40.39 | Strom_C | and you've just sent me on a wild goose chase |
| 08:41.42 | firestrm | no i dont think so.., ummm.. i started out calling it an iax problem, you correctly pointed me to interrupts, which were incorrect, but didnt cure the problem.. so now we are back at iax problem.. sorry.. |
| 08:42.19 | Strom_C | maybe i'm just tired |
| 08:42.27 | sxpert-work | morning |
| 08:42.41 | firestrm | no problem, im in the same boat.. ive benn going at this for 16h now.. |
| 08:42.56 | Strom_C | ah |
| 08:43.08 | Strom_C | well i was totally thrown by the fact that you mentioned fucking with fxo/fxs gain |
| 08:43.21 | Strom_C | so my mind threw out "iax problem" |
| 08:43.22 | tparcina | i'm testing something with asterisk tapi. he mentions "manager window" - what is manager window? |
| 08:43.45 | Strom_C | this is why one should never add unnecessary detail to troubleshooting questions |
| 08:43.45 | Strom_C | firestrm: blame your iax provider |
| 08:44.01 | firestrm | lol... i was just venting about the process that took most of the day.. setting gains properly is a PITA |
| 08:44.36 | firestrm | Strom_C, im leaning that way.. its just annoying that it came up just as i upgraded.. |
| 08:44.57 | Strom_C | firestrm: gain is easy |
| 08:45.08 | Strom_C | firestrm: call the local milliwatt test and set the gain based on that |
| 08:45.13 | Strom_C | takes all of two minutes |
| 08:45.22 | firestrm | but im also beginning to think its a wierd planetary alignment thingy where the provider pootched just as i was upgrading.. |
| 08:45.56 | firestrm | Strom_C, ya.. right.. it took me 4 hours to social engineer the number from telus.. |
| 08:46.25 | Strom_C | well that's your own damned fault for living in territory where the telco doesn't publish these things ;) |
| 08:46.27 | firestrm | but now that i have it, i wont take me as long next time.. |
| 08:46.34 | firestrm | lol.. |
| 08:48.23 | firestrm | not only doesnt publish, mis identifies, i Called a friend who works for telus, and they had it under 2100 hz test tone number.. it didnt make any sense.. but when i called it, it want no 2100 hz. sometimes telus couldnt find their a$$ with both hands.. |
| 08:48.44 | Strom_C | are you in alberta or are you in british columbia? |
| 08:48.47 | firestrm | bc |
| 08:48.54 | Strom_C | thats what I thought |
| 08:48.58 | firestrm | lol |
| 08:49.03 | Strom_C | BC Tel used to be partially owned by GTE |
| 08:49.07 | Strom_C | and you know what that means |
| 08:49.14 | Strom_C | "Get Telephone Elsewhere" |
| 08:49.26 | firestrm | rotfl.. |
| 08:49.42 | firestrm | rotflmao.. |
| 08:49.51 | firestrm | that is SOOOOO true |
| 08:50.33 | firestrm | and dont get me started about shaw.. jamming voip.. yeesh.. |
| 08:51.09 | *** join/#asterisk stargazer_gr (n=stargr@adsl98-12dynamic.salonica.acn.gr) |
| 08:51.47 | Strom_C | O canada, O canada....um...O canada, O canada! |
| 08:52.13 | firestrm | rodeo song or beer song.. ohhh. o canada.. our home's on native land.... |
| 08:52.19 | x86 | there's something about maple leafs and black trenchcoats in there now... |
| 08:52.51 | Strom_C | vancouver is a nice enough place, but man, that city contains like three Starbucks for every man, woman, and child in the entire province. |
| 08:53.39 | firestrm | im in victoria, so here starbucks is the underdog.. much to my suprise when i moved here its a serious coffee on every block |
| 08:54.38 | Strom_C | I live in a nice neighborhood of Los Angeles where there's this really awesome coffeeshop a block away |
| 08:54.53 | firestrm | well, its off to bed for 1h, then up for my daughter's 3am feeding.. |
| 08:54.54 | Strom_C | the only problem is that it tends to be filled with too many screenwriters. |
| 08:54.58 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 08:56.12 | firestrm | Strom_C, try a pepperbomb.. somebody has been launching them on city buses here.. clear em out in seconds.. |
| 08:56.16 | x86 | i live in the middle of a cornfield in illinois where you have to drive a good 20 minutes to get to a coffeeshop |
| 08:56.21 | x86 | 30 if you want wifi ;) |
| 08:56.35 | L|NUX | Strom_C : hey now my pri is connected |
| 08:56.50 | firestrm | lol.. thats what i do for a living.. wifi on every block.. |
| 08:56.53 | Strom_C | ah ok...what did you do to it |
| 08:57.21 | L|NUX | Strom_C : well i used signalling = pri_cpe |
| 08:57.26 | L|NUX | and its works |
| 08:57.40 | Strom_C | aigh |
| 08:57.43 | Strom_C | like i said |
| 08:57.48 | firestrm | g-night Strom_C, thanks a bunch for the help : |
| 08:57.53 | Strom_C | you must be THIS EXPERIENCED with telephones to touch the PBX |
| 08:57.57 | Strom_C | [redraws line] |
| 08:58.16 | *** part/#asterisk firestrm (n=no@S01060013105c1c69.gv.shawcable.net) |
| 08:58.41 | L|NUX | Strom_C : okay another question when i try to call on did |
| 08:58.58 | L|NUX | it will ring bell but nothing come up on * cli |
| 09:00.20 | Aurs | L|NUX: what level of verbosity do you have? |
| 09:00.29 | L|NUX | 200 |
| 09:00.30 | Strom_C | negative seventy-nine |
| 09:00.48 | L|NUX | i did set verbose to 200 |
| 09:00.50 | Aurs | hehe |
| 09:00.57 | L|NUX | :-> |
| 09:02.13 | L|NUX | any idea |
| 09:02.34 | Aurs | set verbose 200 in cli? |
| 09:02.39 | L|NUX | yeah |
| 09:02.53 | Aurs | (don't think there's any point in using 200, but I guess it doesn't hurt) |
| 09:03.02 | L|NUX | yupz |
| 09:03.33 | Aurs | and when you call through that asterisk, nothing comes up in cli? |
| 09:03.57 | L|NUX | well nah |
| 09:04.05 | L|NUX | when i call on any DID nothing comes up |
| 09:04.09 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 09:04.30 | *** join/#asterisk lionelp (n=lionel@ip-128.net-82-216-65.rev.numericable.fr) |
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| 09:13.39 | lionelp | Hi JiBees :) |
| 09:13.48 | JiBees | hey lionel ;) |
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| 09:21.00 | mitcheloc | hmm...i just set the clock on my desk |
| 09:21.10 | mitcheloc | it'd been blinking for about 10 months now |
| 09:22.20 | *** join/#asterisk Stephnie (i=Stephnie@u15157627.onlinehome-server.com) |
| 09:22.23 | Stephnie | hi |
| 09:22.28 | Stephnie | how to kill asterisk ? |
| 09:22.43 | angryuser | is it possible to dial when ${foo}=dial(zap/1/number) and s,1,${foo}? |
| 09:22.52 | mitcheloc | killall asterisk |
| 09:24.58 | angryuser | or should i aks in other way, how execute the chain saved in variable? |
| 09:25.16 | *** join/#asterisk RoyKa (n=roy@ti211210a080-4037.bb.online.no) |
| 09:26.01 | angryuser | any ideas? |
| 09:26.26 | Stephnie | mitcheloc: let me check plz |
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| 09:40.25 | dashu | ;ntmode=yes ;if isdn card operates in nt mode, set this to yes | How do i know if my isdn cards operates in nt mode ? |
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| 09:46.49 | puzzled | morning |
| 09:49.35 | jeremy_g | question: if i set the dtmf method in sip phone different from the one i set in the extension, what happens? e.g. if i set exten=123,1,SIPDtmfMode(info) then what dtmf restrictions apply on caller, are the required to support sip info also? what if they dont |
| 09:49.49 | *** join/#asterisk angryuser (n=aster@i03v-213-44-169-43.d4.club-internet.fr) |
| 09:57.08 | angryuser | can someone help me? i have ${foo} = Dial(Zap/1/number) i need execute what is inside chain, s,1,${foo} does not work |
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| 10:00.42 | stoffell | dashu: do you *want* it to operate in nt mode? |
| 10:02.00 | angryuser | maybe my q is too stupid?:) |
| 10:02.28 | stoffell | angryuser: paste your config on pastebin ? |
| 10:02.35 | dashu | i dont even know what nt mode is >< |
| 10:02.50 | stoffell | dashu: okay, read on voip-info everything on BRI / ISDN |
| 10:02.53 | stoffell | ~docs |
| 10:03.01 | jbot | methinks docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 10:03.13 | angryuser | sek |
| 10:05.38 | angryuser | [office4] |
| 10:05.38 | angryuser | exten => 201,1,Dial(Sip/201) |
| 10:05.38 | angryuser | exten => 200,1,Dial(Sip/200) |
| 10:05.38 | angryuser | exten => 202,1,Dial(Sip/202) |
| 10:05.38 | angryuser | exten => 203,1,Dial(Sip/203) |
| 10:05.39 | angryuser | exten => 208,1,Dial(IAX2/208) |
| 10:05.41 | angryuser | exten => _006XXXXXXXX,1,Set(FOO=${IF($[ ${PORT} = 1]?Dial(Zap/4/0ww${EXTEN}):Macro(portables3,${EXTEN:2}))}) |
| 10:05.44 | angryuser | exten => _006XXXXXXXX,2,${PORT}) |
| 10:05.46 | angryuser | ;exten => _006XXXXXXXX,1,Macro(portables3,${EXTEN:2}) |
| 10:06.17 | angryuser | i just need execute what is inside $FOO |
| 10:06.30 | angryuser | sorry last line is bad |
| 10:06.37 | RoyKa | ~pb |
| 10:06.40 | jbot | [pb] a place to paste your stuff without flooding the channel - try http://pastebin.com/ (BROKEN AND SUCKING NUTS), or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/, or http://paste.lisp.org/ for the lisp/scheme nuts, or http://paste-it.net |
| 10:07.11 | angryuser | ok sorry neveh heard of pasebin;) |
| 10:07.17 | angryuser | *pastebin |
| 10:08.42 | angryuser | http://pastebin.com/789818 |
| 10:08.48 | dashu | aaah :o asterisk doesnt want to start anymore |
| 10:09.14 | dashu | loading chan_capi.so failed hmmm |
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| 10:20.12 | HaMYaI | anyone using oh323 at all? |
| 10:20.22 | *** join/#asterisk Op3r (n=Op3r@61.28.166.162) |
| 10:20.47 | Op3r | whats the best voip termination to uk? |
| 10:21.09 | HaMYaI | I can't get the client to register, it always goes to the default context |
| 10:21.54 | HaMYaI | type=user and gatekeeper=DISABLE |
| 10:21.56 | HaMYaI | any idea? |
| 10:24.05 | jeremy_g | angryuser: hey, what does exten:2 refer to in Macro(somemacro,${EXTEN:2}, what are u doing to this variable? |
| 10:24.54 | angryuser | exten:2 will cut you variable |
| 10:25.17 | angryuser | like 0032365 exten:2 = 32365 |
| 10:30.25 | jeremy_g | kool! |
| 10:33.25 | jeremy_g | angryuser:whats your problem man |
| 10:33.41 | RoyK | jeremy_g: http://www.voip-info.org/wiki/view/Asterisk+Expressions |
| 10:34.37 | jeremy_g | thanks RoyK |
| 10:35.59 | stoffell | MrChimpy: yes, use groups.. |
| 10:36.06 | angryuser | i got $FOO = dial(zap/1number) i need to execute it |
| 10:36.23 | stoffell | angryuser: did you already paste it on pastebin.ca ? |
| 10:36.37 | stoffell | angryuser: oh, i see it, .. |
| 10:36.38 | angryuser | yes http://pastebin.com/789820 |
| 10:36.49 | stoffell | ok, lookin |
| 10:37.37 | stoffell | angryuser: u might want to try http://pastebin.ca , the pastebin.com site is slooow.. |
| 10:37.58 | jeremy_g | user calls, * picks up and asks user to enter the destination number and press # when done, now how to make * read the digits entered? |
| 10:38.21 | jeremy_g | i know how to do with agi, but i dont want. -performance considerations |
| 10:38.37 | jeremy_g | i need extension constrcuts for this |
| 10:38.47 | angryuser | http://pastebin.ca/176035 |
| 10:38.51 | jeremy_g | read digit entered :P sth like |
| 10:39.13 | stoffell | angryuser: ok, got it (much faster), what line is it .. ? |
| 10:39.47 | angryuser | [office4] line 83 |
| 10:40.18 | angryuser | just need to execute data in var ${foo} |
| 10:40.38 | stoffell | angryuser: hm, and what does the CLI "do" if you execute it? |
| 10:42.01 | jeremy_g | please pay some heed to my query too folks :-> |
| 10:42.02 | angryuser | http://pastebin.ca/176040 |
| 10:42.39 | stoffell | jeremy_g: there's a function for that I think, on voip-info |
| 10:43.32 | angryuser | oh sorry forget to make reload on the last one this one is good http://pastebin.ca/176041 |
| 10:43.35 | stephane_ | reboot-time |
| 10:43.36 | stoffell | angryuser: it seems it doesn't execute the 2nd line.. |
| 10:44.32 | stoffell | angryuser: it now tries to execute.. hm, can't you use ..,2,Dial($FOO) ? i think that should work better |
| 10:45.19 | angryuser | it wont work because $FOO =Macro or Dial allready |
| 10:45.27 | stoffell | angryuser: or, between ,1, and ,2, do a NoOp(${FOO}) |
| 10:45.37 | angryuser | ok |
| 10:45.45 | stoffell | to see what $FOO looks like.. |
| 10:46.00 | stoffell | yes, but you could rewrite it to make $FOO not contain a dial ? |
| 10:47.06 | angryuser | http://pastebin.ca/176046 noop added |
| 10:48.05 | angryuser | i cant rewrite it becaus $FOO can contain Macro or can contain Dial |
| 10:48.56 | stoffell | angryuser: hm, but it does it, and then tries to spawn ...,3, .. ? |
| 10:48.56 | *** join/#asterisk MacoStefX (n=stephane@nostromo.cabale.net) |
| 10:51.08 | angryuser | yes it tries, withot result;) |
| 10:51.50 | angryuser | well the main idea is that i dont know ho to execute the chain inside $foo |
| 10:51.54 | stoffell | angryuser: it gives an error "No application '${FOO}.." |
| 10:52.03 | angryuser | exactly |
| 10:52.16 | stoffell | angryuser: i guess you can't execute variables.. it might 'need' an application.. |
| 10:52.41 | stoffell | angryuser: you could rewrite it to do a Goto(${FOO}) ? |
| 10:52.59 | jeremy_g | whats the difference b/w digit time out and response time out? |
| 10:53.00 | angryuser | il try |
| 10:55.18 | angryuser | not working |
| 10:55.39 | angryuser | anyway stoo i got some ideas ho to resolve it, thank you four your help |
| 10:56.13 | angryuser | il write you back the solution |
| 10:56.23 | stoffell | okay, g'luck |
| 10:56.32 | angryuser | brb in 1.5hrs |
| 10:58.28 | dashu | http://pastebin.ca/176049 it just doesnt want to work can anyone tell me why ? |
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| 11:09.40 | LiFeX | hiya |
| 11:09.45 | *** join/#asterisk klem_ (n=klem@klem.estpak.ee) |
| 11:09.53 | LiFeX | I just have a short question regarding digium hw |
| 11:10.05 | LiFeX | We tried to install 2 TDM400P cards in one Asterisk server, and the driver doesn't load for the second one.. any ideas ? |
| 11:10.47 | klem_ | hi .. hope some developer can help me - channels/chan_modem_i4l.c line 434 |
| 11:11.20 | klem_ | I cannot understand where that "f" got his value |
| 11:11.41 | klem_ | i sometimes got "dropped frame" errors and sometimes not |
| 11:11.44 | LiFeX | I can see the 2 cards in /proc/pci file, but when I check dmesg, nothing happens to the second card.. all the modules on the first card are recognized.. |
| 11:11.54 | *** join/#asterisk franck (n=franck@tikiwiki/franck) |
| 11:12.11 | klem_ | possible buffer overflow? |
| 11:12.15 | klem_ | or? |
| 11:15.22 | LiFeX | hi klem_ .. I guess dropped frame occurs mostly because of buffer problems |
| 11:15.45 | LiFeX | I got lot of them when stress testing a Soekris PC a few months ago |
| 11:16.32 | klem_ | well I thought that I can at least read C but that pice of code is something I do not understand |
| 11:16.44 | klem_ | if u have some time, please look at it |
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| 11:31.34 | jeremy_g | shud i hate voip-wiki |
| 11:31.44 | jeremy_g | it says asterisk 1.2 has a Read() app |
| 11:32.05 | jeremy_g | for * to read digits entered by the caller |
| 11:32.31 | jeremy_g | but asterisk-CLI>show applications doesnt show Read() anywhere |
| 11:33.05 | RoyKa | show application read? |
| 11:34.06 | jeremy_g | show applications |
| 11:34.24 | jeremy_g | doesnt list an app as Read() while http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Read |
| 11:34.35 | jeremy_g | shows it exist for 1.2 |
| 11:35.07 | jeremy_g | sorry ignore by shit talk |
| 11:35.42 | jeremy_g | i was whistling all this out of my ass ; didnt load the app_module for it :-> |
| 11:38.06 | *** join/#asterisk eject_ck (n=eject@rubin-gw.neocm.com) |
| 11:38.09 | eject_ck | I found that InternetCalls support SIP http://www.internetcalls.com/en/sip.html and now I want use it with my Asterisk. Can anybody point me & |
| 11:38.21 | *** join/#asterisk tparcina (n=tparcina@lns01-0556.dsl.iskon.hr) |
| 11:38.49 | tparcina | asttapi - does anybody use it? |
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| 12:04.19 | tparcina | why it's so quite? |
| 12:05.07 | klem_ | probably we are in wrong timezone? |
| 12:05.33 | LiFeX | yupp.. .quite possibler |
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| 12:10.24 | stoffell | guess you guys are in europe? :p |
| 12:10.32 | tparcina | I planned to use asttapi, but is't so complicated (incoming calls still don't work for me). so i have decaided to try siptapi. does any body use siptapi? |
| 12:10.44 | DarKnesS_WolF | i hate a fast question " for analog phone lines " using a TDM400 FXS/FXO cards the only codec that works is sliner ? |
| 12:11.28 | tparcina | DarKnesS_WolF: what do you mean by - onlay codec that works? |
| 12:12.22 | DarKnesS_WolF | tparcina: what i want to know i'm using ULAW from the SIP phones but i notice that astersik doing translation to slin for the analog phone lines .. so i wanted to know what codec the analog phone line using and if i can change it |
| 12:15.18 | tparcina | DarKnesS_WolF: i get it, but i don't know the answer |
| 12:15.48 | DarKnesS_WolF | tparcina: thx for help :-) i'm trying voip-info also |
| 12:17.02 | tparcina | asttapi, siptapi, anytapi :)) does anybody use anything ot this? |
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| 12:23.49 | Dr-Linux | anybody knows about Gammu sms system? |
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| 12:26.52 | LiFeX | anybody using 2 digium cards in one asterisk server ?? |
| 12:27.16 | MrChimpy | i was, yes. |
| 12:27.17 | LiFeX | eventually 2 TDMP400P |
| 12:27.33 | MrChimpy | oh, I was using TE411 |
| 12:27.39 | LiFeX | we have problems with the 2 cards |
| 12:27.50 | LiFeX | we see both in /proc/pci |
| 12:27.59 | jontow | just make sure they don't share irqs with anything, ever. |
| 12:28.02 | LiFeX | but the driver doesn't load for the second |
| 12:28.02 | jontow | not even eachother. |
| 12:28.12 | MrChimpy | alternatively get sangoma |
| 12:28.23 | jontow | the driver doesn't have to load in most cases.. you just ztconfig it |
| 12:28.26 | LiFeX | the IRQ affects also the driver loading? .. |
| 12:28.48 | jontow | if you're using 4*FXO on both cards, load the driver and setup zaptel and zapata.conf correctly, then ztcfg |
| 12:28.51 | Dr-Linux | LiFeX: yes i'm using |
| 12:28.57 | *** join/#asterisk kannan (n=kannan2@59.144.22.222) |
| 12:29.04 | tzafrir | ztcfg . ztcfg just configures the already loaded zaptel spans/channels |
| 12:29.23 | jontow | exactly -- if the kernel module is loaded, why try to load it a second time? |
| 12:29.28 | kannan | hello all |
| 12:29.49 | LiFeX | I see these for all the modules on the first card : Module 1: Installed -- AUTO FXO (FCC mode) |
| 12:29.57 | LiFeX | but nothing for the second |
| 12:30.07 | LiFeX | and no green LEDs on the 2. |
| 12:30.24 | jontow | :( |
| 12:30.32 | eject_ck | Hi all. I have account on SIP server (sip.internetcalls.com) - and want make call from my SIP network (with Asterisk installed) throught this account. How can I make it? |
| 12:30.52 | kannan | I am using astgulcient suite with asterisk, can I ask in this room? |
| 12:31.21 | kannan | astguiclient |
| 12:31.53 | LiFeX | so, any other ideas |
| 12:31.59 | LiFeX | ? |
| 12:32.38 | LiFeX | if you can't see any LEDs on the card, what does that mean? |
| 12:32.52 | LiFeX | not even red, or yellow |
| 12:33.20 | LiFeX | I guess it wasn't able to load the driver module |
| 12:33.32 | LiFeX | ..or? |
| 12:33.57 | *** join/#asterisk Ox0F0-0FF (n=pierre@200.216.238.226) |
| 12:34.19 | mut | ARRRRR ye salty dogs! remeber when in the board meetings today its talk like a pirate day! |
| 12:35.10 | eject_ck | can anybody answer for my question |
| 12:36.31 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.116) |
| 12:37.42 | *** join/#asterisk h3x0r4t0r (n=hex@ip68-224-236-92.lv.lv.cox.net) |
| 12:37.55 | *** join/#asterisk RoyK (n=roy@ti211210a080-0359.bb.online.no) |
| 12:40.55 | *** join/#asterisk trelane` (n=trelane@unaffiliated/trelane) |
| 12:41.31 | *** join/#asterisk af_ (n=af@ip-202-199.sn2.eutelia.it) |
| 12:44.06 | Makenshi | i don't suppose anyone has worked with a cisco unified mcs appliance? (eg 7815/7825) |
| 12:49.23 | *** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk) |
| 12:50.09 | Dr-Linux | what is good SMS program for asterisk? |
| 12:50.23 | *** join/#asterisk h3x0r (n=hex@ip68-224-236-92.lv.lv.cox.net) |
| 12:51.07 | *** join/#asterisk row (n=row@plusnet.infidels.br0ke.me.uk) |
| 12:51.51 | JiBees | SipSak no ? |
| 12:52.40 | *** join/#asterisk Druken (n=jdumais@CPE000854de4ec0-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 12:55.53 | DarKnesS_WolF | hm UnderMine what module ur using ? FXS or FXO ? |
| 12:56.19 | Dr-Linux | what is good SMS program for asterisk? anybody knows? |
| 12:57.08 | DarKnesS_WolF | Dr-Linux: i love gsmsend " libgsm " and also sms-server i don't know exactly the name it's on freshmeat search SMS server |
| 12:57.54 | UnderMine | DarKnesS_WolF: fxs - verbosity 10 shows me enough now to see why think a typo somewhere so channel not defined |
| 12:59.23 | *** join/#asterisk supjigatr (n=syslod@152.53.16.10) |
| 13:01.40 | supjigatr | If I have a mp3 for MOH that sounds very slow and metallic how do I fix it? |
| 13:01.45 | Dr-Linux | DarKnesS_WolF: is it AstSMS? |
| 13:01.50 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 13:01.50 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:02.05 | DarKnesS_WolF | Dr-Linux: oh no.. it's a sperated application i use it with GSM modem |
| 13:02.07 | Dr-Linux | DarKnesS_WolF: can you please help me to find the link for it? |
| 13:02.20 | Dr-Linux | oo |
| 13:02.32 | Dr-Linux | DarKnesS_WolF: can't i use any opensource SMS? |
| 13:02.47 | DarKnesS_WolF | Dr-Linux: freshmeat and search on SMS |
| 13:02.55 | DarKnesS_WolF | it's GPL and opensource program |
| 13:03.25 | Dr-Linux | DarKnesS_WolF: there are many, but not sure which one should i use |
| 13:03.37 | Dr-Linux | i looking the name: Gammu SMS |
| 13:04.25 | *** join/#asterisk remiss (i=bofh@225.84-48-68.nextgentel.com) |
| 13:04.52 | UnderMine | s/talk/take/ |
| 13:06.06 | Dr-Linux | DarKnesS_WolF: check this >> http://www.gammu.org/wiki/index.php?title=Gammu:Main_Page |
| 13:07.32 | Druken | Dr-Linux: you should give me chanserv access on dal |
| 13:08.22 | Dr-Linux | Druken: sure what you need? |
| 13:08.49 | *** join/#asterisk robl^ (n=robl@dsl093-025-218.hou1.dsl.speakeasy.net) |
| 13:12.52 | *** join/#asterisk GTX (n=charlie@pdpc/supporter/monthlybronze/GTX) |
| 13:13.11 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:15.23 | GTX | What is termcap? |
| 13:15.35 | GTX | Keep getting an error up when make'ing Asterisk |
| 13:16.01 | GTX | Nevermind, found problem :) |
| 13:16.29 | GTX | maybe not, same problem |
| 13:16.52 | sxpert-work | GTX: what are you trying to run on? |
| 13:16.57 | GTX | Debian 3.1 |
| 13:18.13 | sxpert-work | shouldn't have any problems. |
| 13:18.48 | GTX | configure: error: termcap support not found |
| 13:18.48 | GTX | make: *** [editline/libedit.a] Error 1 |
| 13:21.50 | GTX | Can Asterisk be used to login to an exsisting SIP account and play back things etc like a virtual switchboard? |
| 13:22.57 | DarKnesS_WolF | Dr-Linux: nop gammu is for personal phones |
| 13:23.04 | DarKnesS_WolF | Dr-Linux: let me get u the name of the SMS server i'm usng |
| 13:23.36 | Dr-Linux | DarKnesS_WolF: okey thanks |
| 13:23.51 | stoffell | hm.. is there a way to (centrally from a server) get the network statistics (jitter) from a polycom phone? (so I can script it into an .rrd file) |
| 13:24.41 | DarKnesS_WolF | Dr-Linux: http://freshmeat.net/projects/smstools/ |
| 13:24.53 | Dr-Linux | DarKnesS_WolFThanks |
| 13:25.03 | DarKnesS_WolF | ur wlecome |
| 13:25.42 | Druken | man.... i r tired |
| 13:25.59 | JT | arr |
| 13:27.22 | *** join/#asterisk junix|work (n=andrew@64.221.73.154.ptr.us.xo.net) |
| 13:27.51 | junix|work | I am looking for a tutorial on setting up asterisk on Debian Sarge, I don't want to compile.... |
| 13:27.57 | GTX | Could someone answer my question? |
| 13:28.02 | Druken | man, i was watching tv last night, some show called wife swap... and oh my god... this one chick was soo anal rententive, i wanted to smack the tv |
| 13:28.33 | sxpert-work | Druken: how so ? |
| 13:28.41 | robl^ | Druken: just give her a laxative. works better ;-) |
| 13:29.21 | Druken | she actually lived by "people judge a book by it's cover".... |
| 13:29.33 | *** part/#asterisk GingerRoger (n=GingerDo@oak.palepurple.co.uk) |
| 13:29.49 | Druken | and yeah, everything had a place, and if it wasen't absolutly perfect, it had to be redone |
| 13:29.59 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 13:31.09 | Druken | was sick.. hehe |
| 13:34.43 | *** join/#asterisk `Sauron (i=sauron@h-69-3-12-50.hstqtx02.covad.net) |
| 13:40.12 | angryuser | can someone tell me why my globalvar PORT is not changed back to 0 at the end of the macro here? http://pastebin.ca/176169 |
| 13:44.59 | angryuser | it is really a tiny macro;) |
| 13:45.46 | fourcheeze | angryuser: you need to tell Dial to continue after hanging up |
| 13:46.40 | S^P | angryuser: exten => s,3,Dial(Zap/3/0033${ARG1},,g) << try this |
| 13:49.27 | angryuser | ,,g not worked, my var $PORT is still changed one time to 1 and it stays like that |
| 13:49.56 | Qwell | because you're EXPLICITLY hanging up the call... |
| 13:50.21 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 13:51.05 | angryuser | its strange, but if i dont add this line to the end of macro, my line dont detect a hang up |
| 13:51.08 | [TK]D-Fender | :O |
| 13:51.13 | Qwell | [TK]D-Fender: http://pastebin.ca/176169 :) |
| 13:51.20 | angryuser | and it stays open for a moment;) |
| 13:51.30 | Qwell | angryuser: sure doesn't look like the end to me |
| 13:51.52 | [TK]D-Fender | Qwell: DUMB.... 10 shades of it! |
| 13:52.22 | [TK]D-Fender | Qwell: Bludgeon that user with.... THE BOOK |
| 13:53.14 | Qwell | angryuser: so, yeah.. move the Hangup down a line |
| 13:53.16 | eject_ck | how must I configure asterisk to allow calling to another sip server with additional number |
| 13:53.46 | eject_ck | see softphone -> my asterisk server -> other SIP server |
| 13:53.56 | [TK]D-Fender | angryuser: Was that pastebin Qwell just pasted yours? |
| 13:54.01 | angryuser | it was my first thought, i did moved hungup , not helped |
| 13:54.22 | angryuser | yes Fender |
| 13:58.19 | stoffell | GTX: libedit problem fixed? |
| 13:58.26 | GTX | stoffell, yeah mate :) |
| 13:58.45 | stoffell | good :) www.debian.org/distrib/packages is your friend :) |
| 13:58.53 | GTX | indeed |
| 14:00.03 | [TK]D-Fender | angryuser: What are your trying to accomplish with that macro? |
| 14:00.13 | [TK]D-Fender | angryuser: And please show me how you're calling it. |
| 14:00.24 | angryuser | http://pastebin.ca/176185 ext.config |
| 14:01.01 | stoffell | [TK]D-Fender: oh, quick one.. you know if it's possible to fetch some stats from the polycom's,like jitter,etc.. ? (from a central server) |
| 14:01.05 | angryuser | well it is simple i need that at the start ${PORT} = 1 and at the end of the macro ${PORT}=0 |
| 14:01.13 | [TK]D-Fender | angryuser: What is that point of that variable? |
| 14:02.11 | angryuser | i got a simple IF test to detect is this channels is being used |
| 14:02.15 | *** join/#asterisk hank (n=hank@netwichtig.de) |
| 14:03.02 | [TK]D-Fender | angryuser: You should use "ChanIsAvail" for that. |
| 14:03.46 | dashu | exten => 5409618/01759140088,3,Hangup is 5409618 the number of my asterisk server and 01759140088 the number of the caller ? |
| 14:04.31 | *** join/#asterisk I-MOD (i=opticron@c-71-207-209-230.hsd1.al.comcast.net) |
| 14:04.48 | eject_ck | how realize redirecting call to another SIP server (when user inster addional number before number 9 for example) ? |
| 14:06.15 | angryuser | thx fender its working now |
| 14:08.19 | [TK]D-Fender | angryuser: Words cannot express how BAD this line is... exten => _X.,1,Macro(intern2,${EXTEN}) |
| 14:10.06 | GerbilWrk | has anyone experienced problems with asterisk routing calls with wild cards on the 4th digit in 10 digit format? |
| 14:10.39 | angryuser | fender, explain yourself |
| 14:15.39 | [TK]D-Fender | angryuser: You should never use a global pattern match like that especially with other pattern matches in the same context. |
| 14:15.55 | [TK]D-Fender | angryuser: 1 of those 2 possibilities will lock you out of the other |
| 14:18.29 | angryuser | it has been working good for 3 weeks now |
| 14:19.51 | *** join/#asterisk f0urtyfive (i=f0urtyfi@c-67-165-5-232.hsd1.ct.comcast.net) |
| 14:19.59 | [TK]D-Fender | angryuser: You've gotten lucky. _X. means every number. you are mixing that with other matches in the same context. _X. might have taken over all calls. |
| 14:21.48 | [TK]D-Fender | angryuser: And your Set's do nothing in there. |
| 14:23.44 | angryuser | which one?(line) |
| 14:24.02 | [TK]D-Fender | angryuser: ALL OF THEM. |
| 14:24.22 | [TK]D-Fender | angryuser: exten => s,2,Set(${ARG1}=${EXTEN}) |
| 14:24.39 | [TK]D-Fender | angryuser: Thats not how you call set, and you are LUCKY it failed otherwise nothing work ourk. |
| 14:24.52 | [TK]D-Fender | work* |
| 14:26.12 | Qwell | s/rk ou/uld wo/ :p |
| 14:26.30 | angryuser | thank you for help of course but, fender you starting to piss me off a bit;) me ext,conf is working for me, if i encounter any problems il try to solve them, dont bother |
| 14:27.49 | *** join/#asterisk Corydon76-home (i=gold@pdpc/supporter/sustaining/Corydon76-home) |
| 14:27.49 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 14:28.33 | angryuser | i mean i have learned aster 3 days a go, i cant write a professional ext.con from the firs time;) |
| 14:28.44 | [TK]D-Fender | Qwell: i/d/o/n/t/r/e/a/d/r/e/g/e/x/o/k/p/l/z/t/h/x/b/i/b/i |
| 14:29.31 | nettie | hey guys, anyone know why a sip phone host could show as UNSPECIFIED after a couple of minutes the phone registered? |
| 14:29.52 | nettie | the good thing is I can make calls |
| 14:29.58 | nettie | the bad is I cant receive them |
| 14:30.16 | nettie | it's like asterisk forgets about it |
| 14:30.19 | *** join/#asterisk rajiv|work (n=rajiv@gentoo/developer/rajiv) |
| 14:30.24 | nettie | and the phone doesnt re-register |
| 14:30.32 | nettie | any idea please? |
| 14:30.38 | Juggie | set nat=yes qualify=yes for that peer. |
| 14:30.42 | Qwell | rajiv|work: hey |
| 14:32.22 | nettie | Juggie it's already like that |
| 14:32.42 | nettie | all my peers are nat=yes and qualified |
| 14:32.54 | nettie | that's the only one which has problems |
| 14:33.12 | *** join/#asterisk kannan (n=kanna@61.246.91.36) |
| 14:33.26 | Juggie | nettie, then it could be a bad nat device |
| 14:33.31 | Juggie | or misbehaving |
| 14:33.35 | nettie | uhmm |
| 14:33.40 | Juggie | try setting your registration time really low |
| 14:33.44 | Juggie | i set mine to 60seconds |
| 14:33.44 | nettie | do you think configure stun on it could help? |
| 14:33.50 | Juggie | i dont know. |
| 14:33.51 | nettie | I did that as well |
| 14:33.56 | nettie | I configured it 60 |
| 14:33.58 | nettie | and also 30 |
| 14:34.00 | nettie | didnt help |
| 14:34.03 | Juggie | you have other phones behind nat that are fine |
| 14:34.07 | nettie | sure |
| 14:34.08 | Juggie | its just this one thats problematic? |
| 14:34.16 | nettie | I have couple of polys which are fine |
| 14:34.29 | Juggie | but they are behind nat correct? |
| 14:34.29 | nettie | this is an utstartcom f1000g vowifi phone |
| 14:34.33 | nettie | right |
| 14:34.38 | *** join/#asterisk nvzn (n=nvzn@dormir.dreaming.org) |
| 14:34.42 | Juggie | different nat routers or all the same |
| 14:35.15 | nettie | same router |
| 14:35.18 | nettie | same place |
| 14:35.24 | Juggie | then it might be the phone |
| 14:35.39 | Juggie | does it show qualify when it initially registers? |
| 14:36.02 | nettie | sure |
| 14:36.08 | nettie | it shoes the ip and all the info |
| 14:36.18 | Juggie | with a ms delay? |
| 14:36.20 | nettie | when I do another sip show peers |
| 14:36.24 | nvzn | im using an iax softphone to connect to my asterisk box, and also to the fwdOUT network directly. I call myself on my cellphone, and in both cases I cannot get proper audible sound in the cell phone. Seems distorted and there is a kind of digital musical sound in the background |
| 14:36.28 | nettie | it's unspecified |
| 14:36.35 | nettie | of course after a couple of mins |
| 14:38.23 | nvzn | when i say digital musical sound i mean its not uncomfortable, not actual music. more like garbled bits and bytes if i ever heard them |
| 14:39.51 | nettie | Juggie sure |
| 14:39.59 | nettie | it shows me the qualify delay |
| 14:40.52 | *** join/#asterisk QbY (n=Kelvin@cm-64-221-172-125.dhcp.southerncoastalcable.net) |
| 14:41.21 | QbY | I'm having a problem getting transfers to work with some Polycom phones. The Transfer key, nor the # key work. The transfer key puts me on hold, and the # key, does nothing.. Suggestions? |
| 14:41.27 | QbY | IP501s |
| 14:44.41 | QbY | # now works. |
| 14:45.10 | trelane_ | transfer works fine on my snoms |
| 14:45.37 | *** join/#asterisk creativx (n=creadure@196.82-134-19.bkkb.no) |
| 14:46.34 | *** join/#asterisk marv[work] (n=timr@64.89.118.139) |
| 14:46.54 | *** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim) |
| 14:49.22 | *** join/#asterisk unixgeek (n=unixgeek@216-220-234-197.exploremaine.com) |
| 14:54.47 | [TK]D-Fender | QbY: you press [transfer] on the 501 and YOU get put on hold and the screen doesn't react? |
| 14:56.08 | QbY | [TK]D-Fender.. Exactly. |
| 14:56.48 | [TK]D-Fender | QbY: Thats whacked... |
| 14:56.55 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 14:57.22 | QbY | in the CLI, it actually shows the other phone ringing.. but i never come off of hold.. |
| 15:00.08 | mut | http://tinyurl.com/zgm2t |
| 15:00.11 | mut | go willie |
| 15:01.27 | *** join/#asterisk Nebukadneza (n=daddel9@i3ED6F4D3.versanet.de) |
| 15:03.12 | *** join/#asterisk Brixius (n=Brixius@PDN-VBA.OnvoyInc.fw.onvoy.net) |
| 15:04.09 | mut | god i hate VAD |
| 15:05.07 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 15:05.08 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:06.56 | *** part/#asterisk Gregabyte (i=greg@nat/digium/x-cd5b1d34223d7864) |
| 15:07.26 | *** join/#asterisk Tr4d3r (n=Tr4der@200.71.51.147) |
| 15:08.07 | *** join/#asterisk pdt (n=ptinsley@209.12.249.243) |
| 15:08.17 | Brixius | HEllo |
| 15:08.49 | pdt | anybody got suggestions on a good centrally manageable phone other than the polycom's? Some people just seem to hate them... |
| 15:09.04 | Tr4d3r | Hello, i'm having some pauses playing the greeting ... the processor is 97 %idle ... how can i find what is the problem? |
| 15:09.06 | Mportnoy | QBY: I have polycoms 301 and 501 and we I hit transfer, the transfer never makes it |
| 15:09.48 | pdt | Mportnoy: to parking lot or other phones? |
| 15:11.48 | Tr4d3r | Hello again :) , i'm having some pauses playing the greeting ... the processor is 97 %idle ... how can i find what is the problem? |
| 15:11.53 | *** join/#asterisk nsgn (n=nsgn@cpe-66-69-197-173.austin.res.rr.com) |
| 15:11.57 | nsgn | mornin' all |
| 15:12.34 | *** join/#asterisk eKo1 (n=eKo1@190.4.7.90) |
| 15:12.55 | QbY | hrmm.. Tr4d3r.. Do You get the 501 Internal Server Error too? |
| 15:13.09 | *** join/#asterisk eBody (n=ehernand@207.71.51.162) |
| 15:13.29 | Tr4d3r | Nop |
| 15:13.35 | Tr4d3r | all is working fine |
| 15:13.46 | Tr4d3r | but the sound some times pause |
| 15:14.23 | *** join/#asterisk mercestes (n=merceste@216.54.143.242) |
| 15:14.49 | *** join/#asterisk s0lid (n=jlq@61.28.130.145) |
| 15:14.50 | mercestes | Anyone remember the syntax to reprogram the "Do Not Disturb" key on a Polycom 301? |
| 15:14.55 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 15:15.36 | nvzn | anyone knows why sound is so garbled coming out of my iax softphone? |
| 15:16.39 | *** part/#asterisk eject_ck (n=eject@rubin-gw.neocm.com) |
| 15:16.39 | nsgn | does anybody know if the WIP330 is the only wi-fi voip phone with a browser? |
| 15:16.43 | nsgn | i cant seem to find any others |
| 15:18.07 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
| 15:19.12 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
| 15:19.37 | jeremy_g | nvzn:take the softphone out of your __, voice would be clear then :p |
| 15:20.37 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
| 15:21.18 | *** join/#asterisk Sakimustafa (n=yusuf@202.133.14.226) |
| 15:21.22 | dashu | do you know a good site where i can find what stuff i can put behind exten => and what it does |
| 15:22.09 | sivana | do PRIs support ss7? |
| 15:22.15 | rajiv|work | Qwell: pong |
| 15:22.52 | QbY | dashu.. http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf |
| 15:23.12 | dashu | thaanks |
| 15:28.00 | *** join/#asterisk sb_mx (n=sb_mx@dsl-200-67-149-101.prod-empresarial.com.mx) |
| 15:28.23 | *** join/#asterisk intralanman (n=lanman@pool-70-104-160-230.norf.east.verizon.net) |
| 15:29.40 | Mportnoy | QBY : I have the Sip 500 error |
| 15:29.45 | dashu | not finding what i wanted to know there T_T |
| 15:30.23 | *** join/#asterisk Nitesh (n=Nitesh@65.48.63.178) |
| 15:30.29 | Mportnoy | QBY : digium does not know how to fix it, I call them and we spend 4 hours trying everything I jump from 1.2.5 to ver 1.2.10 and since that upgrade transfer got #$#$# |
| 15:31.24 | mog | sivana, ss7 is like pri they are both isdn protocols, so no pri cant support ss7 |
| 15:31.36 | Nitesh | Hello everyone... Can anyone help me with DTMF problems... |
| 15:32.08 | *** join/#asterisk Alystair (n=bob@CPE0009a3009d22-CM00407b85fb2b.cpe.net.cable.rogers.com) |
| 15:32.17 | sivana | mog: so if I wanted an ss7 link to my upstream provider, that would be a separate physical link? |
| 15:32.22 | dashu | can anyone tell me what these 3 lines do ? exten => 5409618/01759140088,1,Wait,1 |
| 15:32.22 | dashu | exten => 5409618/01759140088,2,AGI,callback.agi |
| 15:32.22 | dashu | exten => 5409618/01759140088,3,Hangup |
| 15:32.40 | Cresl1n | ss7!!!! |
| 15:32.50 | mog | correct |
| 15:32.50 | mog | or it would be your only link |
| 15:32.55 | mog | Cresl1n, is the expert here though |
| 15:32.59 | sivana | is ss7 superior to pri? |
| 15:33.03 | Cresl1n | yes |
| 15:33.05 | Juggie | dashu, someone calls 5409618 and if their callerid=01759140088 it runs that script. |
| 15:33.09 | Cresl1n | very much so |
| 15:33.17 | mog | depends on what your doing though |
| 15:33.22 | mog | you might not need the extra info etc |
| 15:33.22 | mog | it provides |
| 15:33.34 | dashu | aaah ok ^^ |
| 15:33.37 | dashu | thank you |
| 15:33.37 | Juggie | it provides a shit ton |
| 15:33.47 | Juggie | we get ss7 over ip (kind of) |
| 15:33.55 | sivana | I see... just trying to learn what exactly it is and how I can use it |
| 15:33.55 | Juggie | and get real time CDR/CPM's |
| 15:35.20 | FlatFoot | anyone got the _ALERT_INFO working on the Snom 190 ???? |
| 15:35.34 | Juggie | isnt _ALERT_INFO for cisco? |
| 15:36.07 | dashu | anyone here knows some php ? .o |
| 15:36.12 | Juggie | yes |
| 15:36.30 | *** join/#asterisk smackus (n=ckwall@63.149.122.93) |
| 15:36.33 | dashu | can u look at a script of mine and tell me what it does ? :ppp its really small |
| 15:36.39 | FlatFoot | Juggie: it is listed on the wiki as the snom |
| 15:36.41 | Juggie | pastebin.ca it. |
| 15:36.44 | dashu | ok |
| 15:36.49 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.72.40) |
| 15:37.03 | Nitesh | Can anyone help me with DTMF problems... I have my asterisk connected to Nextone and for some odd reasons I can not pass the DTMF tones... the remote IVR does not understand the tones when pressed.... |
| 15:38.19 | Nitesh | When I call any 1800 number.. i get the IVR prompts.. but when asked to press 1 for sales , press 2 for tech.... the IVR does not undertand the Tones.... |
| 15:38.25 | RoyK | anyone that knows what "Please can you activate the Asterisks equilivant = Facility-Information-Element-Component. " is supposed to mean? |
| 15:38.40 | Nitesh | I have PolyCom and Snom phones.. both phones are doing the same... |
| 15:38.56 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
| 15:39.39 | Nitesh | Is it Asterisk problem.. or Codec Problem...? I am using G711u |
| 15:40.22 | Nitesh | Please help |
| 15:40.22 | smackus | i have a queue http://pastebin.ca/176324 and i have added 3 members dynamically to it using add queue member Local/3585@CMG to test (for example). When I dial the extension for the test queue, it presents to the first logged in extension and rings it for the 15 seconds. After that I expect for it to ring to the next phone logged into the queue, but it doesnt. It rings the same phone again. Once I answer that call, then it does the same thing at the next p |
| 15:40.34 | *** join/#asterisk Qb3rt (n=jhgjkgui@kyle.colba.net) |
| 15:41.28 | dashu | juggie: http://pastebin.ca/176049 from line 45 |
| 15:41.30 | Qb3rt | in asterisk flash operator panel is there a manner to show if someone is on DND?? |
| 15:41.41 | *** join/#asterisk bmg505 (n=leon@c1-73-6.rndf.isadsl.co.za) |
| 15:41.55 | Cresl1n | RoyK: what are you asking that in reference to? |
| 15:43.06 | dashu | brb |
| 15:45.20 | RoyK | Cresl1n: trying to make call forward number presentation work through the telco's switch |
| 15:46.17 | [TK]D-Fender | smackus: It keeps trying that same agent because they are still the "least recent" and * has no way to know if they are on a call so it keeps trying over and over. |
| 15:46.30 | [TK]D-Fender | smackus: I suggest only RRMEMORY strategy |
| 15:47.27 | smackus | hmmm.. I will try that. |
| 15:47.28 | smackus | thanks |
| 15:47.42 | Katty | mew. |
| 15:48.15 | dashu | re |
| 15:48.45 | dashu | Juggie: seen my pastebin ? |
| 15:49.30 | Juggie | it just writes a call to the spool folder |
| 15:50.26 | dashu | :o oh |
| 15:50.33 | Juggie | it just generates a call |
| 15:50.37 | *** join/#asterisk supjigatr (n=syslod@152.53.16.10) |
| 15:50.38 | Juggie | considering it called callback |
| 15:50.46 | dashu | ah k nice :p |
| 15:50.50 | Juggie | it would appear someone wants to abuse the free incomming calls of their cellphone |
| 15:50.57 | Juggie | so they call, hangup and * calls them back |
| 15:51.00 | Juggie | and then they place a call :) |
| 15:51.04 | dashu | hehe |
| 15:51.23 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 15:51.33 | dashu | k :p good |
| 15:53.04 | *** part/#asterisk pdt (n=ptinsley@209.12.249.243) |
| 15:53.28 | smackus | [TK]D-Fender: so with rrmemory, does it take into account the agent penalty? or is that going to be ignored? in my head it cant work. just wondering if you know |
| 15:53.49 | Nitesh | Hey Dashu... you got any idea with DTMF problems with Asterisks..? |
| 15:54.05 | dashu | XD ah sorry im an asterisk newb |
| 15:54.28 | supjigatr | Anyone know of a cable modem with built in sip that can be purchased and configured? |
| 15:54.32 | dashu | :o im working with a week with it but still got no idea what im really doing lol |
| 15:54.47 | [TK]D-Fender | smackus: I think penalty applies over every strategy except RINGALL. |
| 15:54.47 | dashu | since not with |
| 15:55.01 | Qb3rt | in asterisk flash operator panel is there a manner to show if someone is on DND?? |
| 15:55.13 | [TK]D-Fender | smackus: But that those of equal priority are the ones selected from using the strtaegy |
| 15:55.29 | smackus | that makes sense. |
| 15:55.36 | Nitesh | HELP WITH DTMF PROBLEM.... PLEASE HELP |
| 15:56.03 | Nitesh | ANY SUGGESTIONS... |
| 15:56.08 | Qwell | LOSE THE CAPS |
| 15:56.13 | Nitesh | Sorry |
| 15:56.16 | file | Nitesh: ...please don't do that, that lowers the amount of people who want to help you |
| 15:56.25 | Nitesh | sorry |
| 15:57.37 | *** join/#asterisk chaoscon (n=ph33r@smartserv/ceo/chaoscon) |
| 15:58.11 | mog | to 0 |
| 15:58.21 | Qwell | mog: from 0 |
| 15:58.29 | Qwell | to...a number...less than 0 |
| 15:58.56 | mog | exactly now people want to harm you |
| 16:00.05 | *** join/#asterisk pabluss (n=aquicamb@200.75.1.29) |
| 16:01.04 | Nitesh | To all, |
| 16:01.04 | Nitesh | Do anyone have DTMF problem with Asterisk. I have polycom and snom phone register to Asterisk version 1.2.9. Both phones can not send DTMF tones to remote IVR prompts. I changed the dtmfmode to auto, rfc2833, inband but did not help... I am using G711u to terminate the calls... Can anyone help me with this problem... Thanks. |
| 16:01.37 | pabluss | Hi i have a doubt about my Arescom netdsl 1000 and asterisk, by default my ADSL modem has all the ports closed, I want to know which ports i need to open for asterisk... thanks in advance for the comments |
| 16:02.03 | *** join/#asterisk bintut (n=bintut@58.69.229.44) |
| 16:02.03 | Nitesh | My Asterisk is connected via Nextone SoftSwitch. |
| 16:02.44 | fourcheeze | Nitesh: * and nextone should both be set to use rfc2833 |
| 16:03.00 | jeremy_g | can * detect dtmf mode of the incoming call |
| 16:03.08 | Nitesh | Thanks fourcheeze.... any other suggestion |
| 16:03.17 | [TK]D-Fender | anyone able to point me in the direction of a simple CLI tool to do a DIFF against 2 files? |
| 16:03.18 | fourcheeze | clients also should use rfc2833 unless they don't have it |
| 16:03.23 | [TK]D-Fender | (linix CLI that is) |
| 16:03.23 | jeremy_g | i am reminded of sth SIPDtmfmode=auto but docs dont support it |
| 16:03.27 | Nitesh | yes... * detects incoming dtmf tones |
| 16:03.42 | jeremy_g | Nitesh:how? |
| 16:03.56 | jeremy_g | Nitesh:how can i know the incoming dtmf type |
| 16:04.11 | *** join/#asterisk s0lid (n=jlq@61.28.130.145) |
| 16:04.16 | Nitesh | I have IVR setup on ma * box... and it does detect... |
| 16:04.32 | jeremy_g | i basically want to read digits from incoming calls with all types of dtmfs |
| 16:04.48 | jeremy_g | Nitesh:when it detects, does it tell u also |
| 16:05.17 | Nitesh | yes... I did a test to play back the dialed digits and it plays back |
| 16:06.31 | Nitesh | my problem is while calling out... for example calling 1800flowers.... when asked to press 1 to place an order... the remote IVR does not understand the tone... |
| 16:06.46 | *** join/#asterisk hmmhesays (n=hmmhesay@24-117-135-28.cpe.cableone.net) |
| 16:07.38 | jeremy_g | Nitesh:did u test from two different sip clients with different dtmf modes |
| 16:09.23 | jeremy_g | what is talk off "when asterisk incorrectly detects dtmf when it should not" |
| 16:09.25 | Nitesh | My incoming is coming via Sansay SSW... I havent tried any other SIP clients |
| 16:09.32 | jeremy_g | what happens if talkoff happens |
| 16:12.37 | jeremy_g | please answer my question |
| 16:12.39 | junix|work | how do i log into the SugarCRM using trixbox? |
| 16:13.55 | *** join/#asterisk klem__ (n=klem@80-235-112-227-dsl.trt.estpak.ee) |
| 16:14.12 | Qwell | junix|work: see channel topic |
| 16:14.33 | *** join/#asterisk saftsack (n=saftsack@p54A7C5C2.dip.t-dialin.net) |
| 16:14.38 | Nitesh | i dont have talkoff setup anywhere |
| 16:15.21 | klem__ | hi..is there any asterisk guru also familiar with asterisk source code? |
| 16:15.36 | jeremy_g | klem_:#asterisk-dev |
| 16:15.42 | klem__ | thnx |
| 16:15.55 | mog | contribute to the matt heather laptop fund and im sure you can find some ^_^ |
| 16:16.02 | jeremy_g | :D |
| 16:18.00 | jeremy_g | can anyone plz answer, i want to read dtmf digits from a softphone irrespective of whether it is using rfc2833,inband or info. i wanna read its keys. how do i support them? is there a SIPDtmfmode(auto) |
| 16:18.00 | *** join/#asterisk De_Mon (n=de_mon@fl-69-69-144-77.dyn.embarqhsd.net) |
| 16:18.30 | jeremy_g | [TK]D-Fender:say something |
| 16:18.35 | jeremy_g | :) |
| 16:19.12 | jeremy_g | shud i set relaxdtmf=yes |
| 16:19.19 | jeremy_g | thats the only thing i can think of now |
| 16:21.27 | *** join/#asterisk af_ (n=af@ip-202-199.sn2.eutelia.it) |
| 16:21.29 | Alystair | http://www.newegg.com/Product/Product.asp?Item=N82E16833122077R <-- is this a good switch for SIP phone aggrigation? |
| 16:21.32 | Alystair | (PoE, etc) |
| 16:21.58 | *** join/#asterisk af_ (n=af@ip-202-199.sn2.eutelia.it) |
| 16:24.21 | [TK]D-Fender | Alystair: I wouldn't.. PoE is only functional on 12 of those ports |
| 16:25.23 | mut | has anyone here done any redundancy or load balance of multiple asterisk servers? |
| 16:25.28 | CunningPike | http://www.sineapps.com/news.php?rssid=1502 - nothing on asterisk.org........ |
| 16:25.31 | mut | wondering what method they used |
| 16:25.51 | Qwell | CunningPike: unveiled != released |
| 16:26.04 | Alystair | only 12? |
| 16:26.11 | *** join/#asterisk docelmo (n=vircuser@pool-68-163-49-48.phil.east.verizon.net) |
| 16:26.14 | CunningPike | Qwell: unveiled => feature list? :) |
| 16:26.23 | Qwell | CunningPike: something like that |
| 16:26.34 | CunningPike | Qwell: So, where is it? :D |
| 16:26.35 | Qwell | it even specifically says "Asterisk 1.4 will be available for download on Digium's website (www.digium.com) in October." |
| 16:26.43 | |Vulture| | CunningPike: it says October... |
| 16:26.44 | Alystair | where do you see only 12 |
| 16:27.34 | CunningPike | |Vulture|: I meant the feature list, not the software |
| 16:27.37 | Alystair | ah their site says so |
| 16:29.07 | jeremy_g | mut:yeah |
| 16:29.08 | *** join/#asterisk geoffl (n=geoff@gjctech.plus.com) |
| 16:29.22 | [TK]D-Fender | Alystair: http://www.compuplus.com/i-Netgear-FS726TPNA-ProSafe-24-Port-10100-Smart-Switch-with-2-Gigabit-Ports-and-PoE-1005304~mfr-Netgear-category-NETWORKING.html?sid=wqi0i4dj031t958 |
| 16:29.27 | mut | jeremy_g? |
| 16:29.45 | [TK]D-Fender | Alystair: Be extremely careful. Almost all Netgear routers w/ PoE only support it on HALF the ports |
| 16:30.10 | jeremy_g | mut:its trivial, on top of openmosix, lvs,... <other ways based on redundancy - network, process> |
| 16:30.17 | jeremy_g | mut: what hit are you getting |
| 16:30.21 | *** join/#asterisk justinu|laptop (n=Justin@12.44.122.130) |
| 16:30.39 | jeremy_g | mut: our problem was irq latencies so had to buy better hardware :( |
| 16:30.50 | mut | we're not running into issues right now |
| 16:31.02 | mut | i'm just working on plans to make stuff balanced/redundant |
| 16:31.14 | mut | web/mail/sip |
| 16:31.38 | *** join/#asterisk madero (n=rbc@213.155.203.44) |
| 16:32.37 | madero | hello! we need to put a client on-line to allow people to listen an-audio stream. any suggestion? thx! |
| 16:33.25 | |Vulture| | [TK]D-Fender: thats a great switch for the price 12 PoE for sub $300 |
| 16:33.27 | jeremy_g | jabber |
| 16:34.25 | [TK]D-Fender | |Vulture|: And for < $400 you could get all 24 w/ PoE with greater load capacity and better managed. |
| 16:35.02 | |Vulture| | from where? |
| 16:35.19 | jeremy_g | Is it possible to configure channel SIP to be able to handle more than one |
| 16:35.19 | jeremy_g | DTMF signalling. I am basically using * as a SIP voicemail server however I |
| 16:35.19 | jeremy_g | have a mutually exculise scenario with phones that use one or the othe DTMF |
| 16:35.19 | jeremy_g | signalling. |
| 16:35.33 | madero | anything flash or java able to provide a sip/whatever protocol compatible with asterisk from a web browser?? |
| 16:35.58 | |Vulture| | madero: a flash/java softphone? |
| 16:36.34 | madero | |Vulture|: wow .. thanks :)! have you tried any of them? any good suggestion? |
| 16:36.51 | |Vulture| | madero: no I was just saying.. is that what your looking for? |
| 16:36.59 | madero | |Vulture|: yes :)) |
| 16:37.00 | |Vulture| | I havent seen any |
| 16:37.03 | eKo1 | I've seen activex softphones. |
| 16:37.21 | madero | yes .. I've seen those ... but what about linux/mac users? |
| 16:37.25 | [TK]D-Fender | http://www.tigerdirect.com/applications/SearchTools/item-details.asp?EdpNo=1815261&CatId=596 |
| 16:37.27 | madero | firefox? |
| 16:37.29 | eKo1 | It pretty much sucks as. |
| 16:37.32 | [TK]D-Fender | http://www.tigerdirect.com/applications/SearchTools/item-details.asp?EdpNo=1205369&CatId=868 |
| 16:37.45 | [TK]D-Fender | There, 2 for < $400 |
| 16:37.50 | sxpert-work | anyone tested the sync cable for the TE407P/TE412P ? |
| 16:37.55 | |Vulture| | Linksys I would try |
| 16:38.05 | madero | |Vulture|: linksys? |
| 16:38.05 | jeremy_g | why i am being ignored :( |
| 16:38.09 | Alystair | awesome |
| 16:38.10 | [TK]D-Fender | |Vulture|: I own 2 of the the D-Link and the work great. |
| 16:38.13 | |Vulture| | ooo and QoS |
| 16:38.20 | Alystair | all I need to find is a PoE enabled electric shaver and I'm in business |
| 16:38.31 | file | jeremy_g: because this is not a guaranteed method of getting an answer, if someone wants to help you they will - otherwise they won't |
| 16:39.10 | |Vulture| | [TK]D-Fender: which do you like better the linksys or the dlink? |
| 16:39.48 | [TK]D-Fender | |Vulture|: Haven't tried the LInksys, just noting its there. I jsut bought a 48 Port Linksys switch. only have GBIC uplonk, no copper :( |
| 16:40.19 | JonR800 | jeremy_g: or people might not know, but the answer is yes. read up on sip.conf. you can specify signalling for each device. |
| 16:40.24 | *** join/#asterisk sb_mx (n=sb_mx@dsl-200-67-149-101.prod-empresarial.com.mx) |
| 16:40.32 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 16:41.22 | geoffl | Hi. I'm having difficulty registering a SIP service. Everything I've tried results in authentication failure except user:secret@remote_host, which results in an error, "WARNING[3145]: chan_sip.c:9768 handle_response_register: Got 200 OK on REGISTER that isn't a register" ... Help! |
| 16:42.08 | madero | so ... nobody here has ever tryed a java/flash softphone???? |
| 16:42.53 | *** join/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 16:45.04 | jeremy_g | JonR800:I know how to specify signalling for each device and thats what i cant do. i want to support all three dtmf modes for any incoming call |
| 16:46.07 | jeremy_g | file: :( yeah it seems that way. why does my mind say dtmfmode=auto is possible but there is no documentation to support it |
| 16:46.11 | jeremy_g | tell me what to read |
| 16:49.23 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 16:51.08 | *** join/#asterisk zeppelin_ (n=zeppelin@201.41.0.159) |
| 16:52.32 | ManxPower | you normally want rfc2833 for dtmfmode |
| 16:53.01 | ManxPower | for example of a call has inband dtmf, then dtmf won't work for ANY codec EXCEPT ulaw and alaw. |
| 16:54.55 | intralanman | and not reliably then |
| 16:55.17 | intralanman | maybe that's too harsh |
| 16:55.25 | intralanman | i just hate inband |
| 16:55.34 | GTX | Do you need libgsm installed to playback sound files? |
| 16:56.27 | *** join/#asterisk clyrrad (n=ddd@CPE001195f553c7-CM0011aea484a4.cpe.net.cable.rogers.com) |
| 16:59.15 | ManxPower | GTX, not for asterisk |
| 16:59.33 | jeremy_g | ManxPower:I know all this my friend. I know all this. Tell me what i dont know. Awww!! |
| 16:59.34 | ManxPower | not reliable == not work |
| 17:00.16 | jeremy_g | ManxPower:inband cant work on compressed codecs. that is obvious. |
| 17:00.28 | jeremy_g | s/codecs/audio streams |
| 17:01.14 | Alystair | [TK]D-Fender: what phones do you use in your setup? |
| 17:01.35 | [TK]D-Fender | Alystair: Pretty much exclusively Polycom. |
| 17:01.38 | jeremy_g | [TK]D-Fender:i was really expecting an answer from you. |
| 17:02.01 | [TK]D-Fender | jeremy_g: At least you have identified the source of your disappointment.... |
| 17:02.27 | Alystair | ok I'm on the right track the n |
| 17:03.03 | Alystair | How do you config polycom's anyway? |
| 17:03.17 | jeremy_g | [TK]D-Fender:no, i was clear in my question from the beginning |
| 17:03.21 | jeremy_g | :( |
| 17:03.55 | bintut | anyone here can help me test my setup if this is already running ok or not? please give me a call at: sip://bintut@bintut.homelinux.org:5060 |
| 17:04.02 | [TK]D-Fender | Alystair: The sme way anyone sane would : provisioning server. I prefer and recomment FTP |
| 17:04.40 | *** join/#asterisk jhildebrand (n=hildjj@corp-fw-main.jabber.com) |
| 17:04.49 | bintut | my configs located at http://paste.debian.net/13292 kindly check it if on the right track already.. |
| 17:05.02 | bintut | i'm not sure if this is ok already |
| 17:05.26 | jeremy_g | bye bye folks |
| 17:05.29 | jeremy_g | its just not my day today |
| 17:05.53 | jhildebrand | can someone give me a pointer to docs on the extension.conf "hint" argument? |
| 17:05.57 | [TK]D-Fender | bintut: You are several miles out on that one... |
| 17:06.24 | Alystair | [TK]D-Fender, where would I find documentation about the whole thing? |
| 17:06.26 | bintut | [TK]D-Fender: what do you mean? |
| 17:06.28 | ManxPower | jhildebrand, extensions.conf.sample was not helpful? |
| 17:06.33 | [TK]D-Fender | jhildebrand: http://www.voip-info.org/wiki/view/Asterisk+presence |
| 17:07.05 | jhildebrand | ManxPower: no... i'm a newb to extensions.conf, so it was too much to process at once. |
| 17:07.11 | jhildebrand | TK: thx. that should get me started. |
| 17:07.13 | [TK]D-Fender | Alystair: www.polycom.com Go download the admin guide and you'll need to download the SIP & BootROM images you intend to run on them |
| 17:07.13 | bintut | where can i find the entire list of parameters for sip.conf and extensions.conf files? |
| 17:07.47 | [TK]D-Fender | bintut: check on the WIKI |
| 17:08.11 | bintut | [TK]D-Fender: where is it located? |
| 17:08.21 | [TK]D-Fender | bintut: Look at my link above |
| 17:08.27 | bintut | ok |
| 17:09.46 | jhildebrand | hm. so if I have a hint priority, if the phone has registered, hint more-or-less equals 1, and if not the rule is skipped? |
| 17:10.08 | CunningPike | Has anyone used Druid? http://www.voiceroute.net/site/index.php?p=druid |
| 17:11.02 | [TK]D-Fender | jhildebrand: * will track the refferd-to device and report status back to a phone that is told to monitor the exten the hint is on. |
| 17:11.22 | *** part/#asterisk geoffl (n=geoff@gjctech.plus.com) |
| 17:14.15 | Alystair | [TK]D-Fender: are you using IP501's? |
| 17:14.30 | [TK]D-Fender | Alystair: I own every desktop model they make. |
| 17:15.02 | Alystair | Know anyone in Canada who can make a big bundle? |
| 17:15.07 | [TK]D-Fender | Alystair: If you are planning on PoE, IP 430's are a more economical choice. I will say that I would suggest the 501 over it though for raw usability. |
| 17:15.56 | CunningPike | Alystair: We plumped for the 501 |
| 17:16.27 | jhildebrand | so, how would I parse the meaning of this, from the sample extensions.conf: |
| 17:16.29 | jhildebrand | exten => 6245,n(dial),Dial(${HINT},20,rtT) |
| 17:17.17 | *** join/#asterisk ToTo (n=ToTo@host149-109-dynamic.58-82-r.retail.telecomitalia.it) |
| 17:19.01 | *** join/#asterisk MRH2 (n=Mr_happy@host-83-146-30-242.bulldogdsl.com) |
| 17:19.16 | *** join/#asterisk chaoscon_ (n=ph33r@smartserv/ceo/chaoscon) |
| 17:19.27 | hmmhesays | bah, faxing is such a b1atch |
| 17:19.38 | MRH2 | hey anyone using a poly ip4000 with firmware 2.0.1? |
| 17:20.06 | MRH2 | and reinvites on |
| 17:21.22 | [TK]D-Fender | CunningPike: IP 501 screen is worth paying for over the 430 if cost effectiveness isn't in question. |
| 17:21.28 | hmmhesays | asterisk support t.38 passthru yet? |
| 17:21.32 | [TK]D-Fender | CunningPike: Much nicer to work with. |
| 17:22.26 | CunningPike | [TK]D-Fender: We had decided on the 501 before the 430 came out, but I understand it's the same screen as the 301, and that didn't exactly blow our skirt up |
| 17:22.58 | CunningPike | hmmhesays: In 1.4, apparently |
| 17:23.10 | CunningPike | No Druid users? |
| 17:24.27 | [TK]D-Fender | CunningPike: Well its not as "nice", but it does have lighted key indicators, a pixel screen over the IP 301, integrated PoE, etc. Technically it still as good a general choice as any phone as compared to the IP 501 |
| 17:24.41 | [TK]D-Fender | CunningPike: Just that the 501's larger screen feels nicer |
| 17:24.53 | CunningPike | [TK]D-Fender: And it fits our logo :) |
| 17:25.44 | [TK]D-Fender | CunningPike: That too ::) I just did my first Idle image & auto-answer setup this past weekend on mine |
| 17:25.48 | *** join/#asterisk Sakimustafa (n=yusuf@202.133.14.226) |
| 17:26.06 | MRH2 | but now poly has thrown IP650 in the mix (with G722) and chrome bits |
| 17:26.21 | CunningPike | MRH2: And an extra $100 a set |
| 17:27.02 | MRH2 | yeah for $100 i want a colour screen and giga ethernet |
| 17:27.24 | [TK]D-Fender | Yeah, the IP 650 is too pricy for me to suggest. |
| 17:27.27 | hmmhesays | and I want a dead hooker in my trunk |
| 17:27.32 | hmmhesays | .... ooops ... |
| 17:27.58 | CunningPike | You silly, twisted boy :P |
| 17:28.05 | MRH2 | maybe if you order in quantity... |
| 17:28.18 | supjigatr | Anyone know of a cable modem with built in sip that can be purchased and configured? |
| 17:28.34 | MRH2 | draytek does one |
| 17:28.36 | [TK]D-Fender | supjigatr: EW! |
| 17:28.48 | [TK]D-Fender | supjigatr: Why would you want to do that? Leave them seperate |
| 17:28.57 | supjigatr | TK: Yea it suck but nessacary. |
| 17:29.08 | supjigatr | Rural cable operator requires it. |
| 17:29.55 | MRH2 | actually supjigator maybe the draytek is just adsl |
| 17:30.45 | *** join/#asterisk bintut (n=bintut@124.106.140.153) |
| 17:30.53 | sevard | hmmhesays: i got a couple you can have, but you'll have to off them for me |
| 17:30.54 | Dr-Linux | anybdoy ever use Gammu program? |
| 17:31.35 | supjigatr | Yea it seems to be a hard item to find. |
| 17:31.35 | bintut | [TK]D-Fender: my laptop shuts off.. i didn't noticed my battery life.. :( |
| 17:31.45 | supjigatr | Motoroa makes it but won't sell them. |
| 17:31.51 | bintut | [TK]D-Fender: what's was the link again to the wiki? |
| 17:31.53 | supjigatr | SA makes it but won't sell it. |
| 17:35.04 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 17:35.09 | *** join/#asterisk ambriento (n=melcon@200.192.160.100) |
| 17:35.43 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 17:35.44 | [TK]D-Fender | ~docs |
| 17:35.46 | jbot | extra, extra, read all about it, docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 17:36.07 | supjigatr | ~cablemodme |
| 17:36.10 | supjigatr | ~cablemodem |
| 17:39.45 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 17:40.09 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 17:40.39 | *** join/#asterisk DasTech (n=DasTech@d47-69-194-47.col.wideopenwest.com) |
| 17:41.00 | DasTech | anyone here got a vox mo=dulein dev for asterisk ? |
| 17:41.11 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 17:41.16 | *** join/#asterisk C6Vette (n=info@72-166-37-114.dia.static.qwest.net) |
| 17:41.57 | *** join/#asterisk SomethingISODD (n=dan@h109.42.63.69.cable.ottr.cablerocket.net) |
| 17:42.07 | *** join/#asterisk Arno[Slack] (i=100@master.infinityperl.org) |
| 17:42.27 | SomethingISODD | Hello all anyone know of any did providers that has parry sound ontario dids? |
| 17:44.53 | DasTech | anyone here get sphinx to to work with asterisk ? |
| 17:48.01 | *** join/#asterisk rollergrrl (n=0x3e44d@71-213-6-123.slkc.qwest.net) |
| 17:48.43 | *** join/#asterisk Seb7 (n=sebast@host217-34-0-169.in-addr.btopenworld.com) |
| 17:50.57 | *** join/#asterisk iCEBrkr (i=icebrkr@69.9.167.70) |
| 17:54.44 | *** join/#asterisk deeLer (n=trillian@d54C38572.access.telenet.be) |
| 17:54.51 | deeLer | hello |
| 17:55.26 | deeLer | any Belgian users here ? i'd like to know how to make my callerID work with asterisk .... |
| 17:55.32 | stoffell | hi deeLer :) |
| 17:55.35 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 17:55.41 | deeLer | hi stoffell |
| 17:55.42 | deeLer | :) |
| 17:55.47 | stoffell | u could be my neighbour :p |
| 17:56.07 | deeLer | kben van roeselare, en jij? |
| 17:56.20 | charles___ | I see darkness |
| 17:56.26 | stoffell | let's talk english here, i'll talk flemish in private ;) |
| 17:56.33 | deeLer | sure :) |
| 17:56.34 | charles___ | Hey guys, what kind of preemtiveness do you recommend for Asterisk |
| 17:56.45 | charles___ | 100hz regular server ? |
| 17:56.52 | deeLer | Hz ? |
| 17:57.04 | stoffell | charles___: in normal circumstances, i'd do that, yes |
| 17:57.16 | sevard | I thought sendmail was highly vunerable to exploits. What does #asterisk suggest for sending voicemail by email? |
| 17:57.33 | stoffell | sevard: use whatever you prefer, i prefer postfix.. |
| 17:57.55 | sevard | i'm thinking at least exploit free :| |
| 17:58.15 | hmmhesays | how many special people change |
| 17:58.22 | hmmhesays | how many lives are living strange |
| 17:58.23 | *** join/#asterisk gmfm (i=gmfm@rtr.enterprisemtg.net) |
| 17:58.29 | hmmhesays | where were you while we were getting high |
| 17:58.53 | stoffell | sevard: use the one your most familiar with.. |
| 17:58.58 | stoffell | sevard: i'd prefer postfix or qmail |
| 17:59.06 | *** join/#asterisk Magicianx (n=magician@116-22.dr.cgocable.ca) |
| 17:59.11 | *** join/#asterisk chaoscon (n=ph33r@smartserv/ceo/chaoscon) |
| 17:59.14 | sevard | stoffell: I'm familiar with sendEmail.pl :) |
| 17:59.41 | stoffell | sevard: well, ... that's a great one too.. |
| 17:59.53 | sevard | I don't think it supports attachments though |
| 17:59.56 | hmmhesays | you can use whatever the hell you want |
| 17:59.57 | sevard | it's pretty slick though. |
| 18:00.00 | hmmhesays | you can use phpmailer if you want |
| 18:00.03 | stoffell | sevard: uhm, i think it does though... |
| 18:00.24 | sevard | hmmhesays: I just don't think anyone ought to be using sendmail these days. |
| 18:00.26 | DasTech | is there a page or url where people post thier agi's they write or post addons ? |
| 18:00.58 | stoffell | DasTech: voip-info is the best resource.. |
| 18:01.10 | hmmhesays | I have some stuff on www.thelostpacket.org |
| 18:04.21 | stoffell | hmmhesays: nice |
| 18:05.25 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 18:06.16 | charles___ | going to leave at 250hz |
| 18:07.00 | *** join/#asterisk adorah (n=admin@84.94.144.7.cable.012.net.il) |
| 18:07.14 | *** join/#asterisk jeffgus (n=jeffgus@38.119.60.2) |
| 18:08.08 | *** join/#asterisk loxza (n=gonzalo@VA1-1F-u-0063.mc.onolab.com) |
| 18:08.11 | *** join/#asterisk nortex (n=breeves@snapper.titanspecialties.com) |
| 18:08.44 | CunningPike | Today is International 'Talk Like A Pirate Day' - http://www.talklikeapirate.com/piratehome.html |
| 18:08.47 | loxza | hi, anyone could tell me if it is possible to use g.729 library from intel in a soekris/wrap board? |
| 18:15.41 | *** join/#asterisk chaoscon_ (n=ph33r@smartserv/ceo/chaoscon) |
| 18:16.12 | mog | loxza, yes |
| 18:16.14 | tamp4x | is it possible to bind to multiple sip ports? |
| 18:16.32 | Druken | CunningPike: i didn't realize that was the season opener... |
| 18:19.22 | loxza | mog, and what would be the right flavour? pentium? |
| 18:20.25 | *** join/#asterisk chaoscon (n=ph33r@smartserv/ceo/chaoscon) |
| 18:20.25 | mog | i think there is an unoptimized build |
| 18:20.35 | mog | i believe the soekris is only a 586 based system |
| 18:21.04 | *** join/#asterisk myshenka (n=spamyous@82.153.170.213) |
| 18:22.31 | *** join/#asterisk Ebola (n=Ebola@81-86-155-65.dsl.pipex.com) |
| 18:23.05 | Sakimustafa | What is the billing software for asterisk? |
| 18:23.30 | DasTech | a2billing is one |
| 18:23.44 | DasTech | google |
| 18:23.51 | DasTech | ts your friend |
| 18:24.27 | Sakimustafa | How to configure it give me some link pls |
| 18:25.17 | carrar | Home made SQL querys |
| 18:25.36 | DasTech | goggle it |
| 18:25.43 | carrar | http://www.postgresql.org/ |
| 18:25.53 | DasTech | or get trixbox |
| 18:27.51 | hmmhesays | there is all kinds out there |
| 18:27.53 | hmmhesays | a2bill |
| 18:27.56 | hmmhesays | is pretty full featured |
| 18:28.11 | hmmhesays | otherwise you can generate your own bills with asterisk stat |
| 18:28.14 | SomethingISODD | can anyone recommend any good wifi phone that will connect to hotspots? |
| 18:28.53 | [TK]D-Fender | SomethingISODD: Get one running full WinCE. You'll need a browse for some spots and its going to be PRICEY |
| 18:29.38 | hmmhesays | linksys wip-300 works 'ok' and can do site surveys otherwise the zyxel 2000w is 'ok' |
| 18:30.18 | hmmhesays | but I agree something running wince would be much better |
| 18:30.34 | SomethingISODD | [TK]D-Fender basically i need something that will work with my wireless router, and different ones here in town, that i can get access on |
| 18:31.18 | [TK]D-Fender | SomethingISODD: Depends on the hotspot. Some require HTTP auth, etc. WIP-300 is a decent looking basic one, but I haven't tried it yet. |
| 18:31.19 | hmmhesays | wip-300 |
| 18:31.29 | [TK]D-Fender | hmmhesays: You used one? |
| 18:31.30 | hmmhesays | i have one sitting next to me, it is .. "ok" |
| 18:32.01 | SomethingISODD | wip-300 can you tell me where to get info on that phone? |
| 18:32.07 | hmmhesays | linksys website |
| 18:32.11 | hmmhesays | voipsupply website |
| 18:32.12 | SomethingISODD | oh ok |
| 18:32.45 | SomethingISODD | thanks guys. one more question do you guys know of any did provider that has northen ontairo |
| 18:32.57 | hmmhesays | vitelity might |
| 18:33.09 | SomethingISODD | who? |
| 18:33.39 | hmmhesays | http://www.vitelity.net/?p=network |
| 18:33.56 | SomethingISODD | ok thanks |
| 18:34.22 | loxza | mog, thanks, I will give it a try |
| 18:41.24 | *** join/#asterisk jrprado (i=jrprado@201.47.13.55.adsl.gvt.net.br) |
| 18:47.34 | *** join/#asterisk chadkouse (n=chadkous@165.236.120.14) |
| 18:47.50 | chadkouse | what is the best way to monitor or even barge in on a call taking place through asterisk ? |
| 18:48.24 | mog | chan_spy and whisper paging |
| 18:48.50 | Alystair | baseball bat |
| 18:48.55 | chadkouse | thanks |
| 18:49.05 | *** join/#asterisk De_Mon (n=de_mon@fl-69-69-138-55.dyn.embarqhsd.net) |
| 18:49.09 | mog | hmm baseball bat didnt think of that |
| 18:50.20 | trelane_ | *WHAM* I'll take that call! |
| 18:52.13 | jrprado | Hi, I need help for accont create with IP authenticate not user and pass.. ?? |
| 18:52.49 | mog | allow guest calls from ip |
| 18:52.50 | mog | done |
| 18:54.14 | jrprado | how ? |
| 18:55.20 | *** join/#asterisk `Sauron (i=sauron@h-69-3-12-50.hstqtx02.covad.net) |
| 18:58.17 | *** join/#asterisk JohnJacob (n=dhorner@pool-71-127-86-243.aubnin.fios.verizon.net) |
| 19:00.09 | hmmhesays | host=1.2.3.4 in sip or iax.conf |
| 19:00.23 | *** join/#asterisk dandan (i=dandan@pacanka.com) |
| 19:00.28 | dandan | hello all :) |
| 19:00.36 | dandan | anyone has 1.1.11.zip for GXP-2000? |
| 19:00.46 | dandan | it has been pulled from BET website... |
| 19:00.51 | dandan | *BETA |
| 19:03.06 | SomethingISODD | hey any one here running fedora core 5 and live in brantford or woodstock or norwich and would dub me a cd |
| 19:06.49 | *** join/#asterisk chaoscon (n=ph33r@smartserv/ceo/chaoscon) |
| 19:06.53 | dandan | anyone has 1.1.11.zip for GXP-2000? |
| 19:07.43 | Trazz | Whats the mode stable * in a box now? |
| 19:07.56 | *** join/#asterisk Gunnar (n=gunnar@101.Red-80-24-171.staticIP.rima-tde.net) |
| 19:08.00 | Trazz | Like prebuilt image with web gui, etc.. |
| 19:09.42 | *** join/#asterisk Jam0r (i=Jamie@88-110-231-32.dynamic.dsl.as9105.com) |
| 19:11.25 | Jam0r | Has anyone got any idea why I'm getting "chan_sip.c:6846 handle_response: Forbidden - wrong password on authentication for INVITE" when the user/pass is deffinately right |
| 19:12.51 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-166-30.red.bezeqint.net) |
| 19:13.02 | *** join/#asterisk ^Sandro^ (n=lkjsdf@internal--router.canaca.com) |
| 19:13.05 | ^Sandro^ | hello |
| 19:13.43 | ^Sandro^ | anyone here ? I have a questions. I installed asterisk ( latest version ) and everythins is running ok exept the music on hold. I installed the mpg123 and its going but when i call queue it plays the sound all distorted |
| 19:13.46 | ^Sandro^ | any ideas ? |
| 19:13.55 | ^Sandro^ | i have looked at every file and nothing |
| 19:15.55 | *** join/#asterisk rowter (n=Silver@201.135.9.97) |
| 19:15.57 | trelane_ | install asterisk-addons |
| 19:16.03 | trelane_ | it's got an mp3 player built in |
| 19:16.14 | ^Sandro^ | ok cool how do i call it |
| 19:16.14 | ^Sandro^ | ? |
| 19:16.18 | ^Sandro^ | i got that installed |
| 19:16.24 | trelane_ | it just runs if you're set up correctly |
| 19:16.28 | trelane_ | hang on |
| 19:16.32 | trelane_ | let me get you my musiconhold.conf |
| 19:16.36 | ^Sandro^ | cool ty |
| 19:16.46 | trelane_ | [default] |
| 19:16.46 | trelane_ | mode=files |
| 19:16.47 | trelane_ | directory=/var/lib/asterisk/mohmp3 |
| 19:16.47 | trelane_ | random=yes |
| 19:17.53 | ^Sandro^ | so thats it.. that is all i put in there |
| 19:17.54 | ^Sandro^ | ? |
| 19:18.35 | ^Sandro^ | bingo |
| 19:18.40 | ^Sandro^ | dahm that mpg123 |
| 19:18.41 | ^Sandro^ | :) |
| 19:18.46 | ^Sandro^ | its working ok now |
| 19:18.48 | ^Sandro^ | ty very very much |
| 19:19.26 | GTX | Has anyone got any idea why I'm getting "chan_sip.c:6846 handle_response: Forbidden - wrong password on authentication for INVITE" when the user/pass is deffinately righ |
| 19:19.52 | trelane_ | GTX, sip debug and paste it? |
| 19:19.55 | trelane_ | err pastebin it |
| 19:19.59 | trelane_ | ~pb |
| 19:20.00 | jbot | hmm... pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/ (BROKEN AND SUCKING NUTS), or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/, or http://paste.lisp.org/ for the lisp/scheme nuts, or http://paste-it.net |
| 19:22.36 | GTX | trelane, http://pastebin.ca/176550 |
| 19:22.36 | [TK]D-Fender | GTX: Its not lying. you only THINK you did it right. |
| 19:23.28 | GTX | It is lying, [TK]D-Fender |
| 19:23.38 | *** join/#asterisk chaoscon_ (n=ph33r@smartserv/ceo/chaoscon) |
| 19:23.53 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 19:23.59 | [TK]D-Fender | GTX: Where's the rejection message in there? |
| 19:24.07 | Qwell | GTX: in sip.conf, set stoplyingtomeplease=yes in [general] |
| 19:24.33 | GTX | It's not sec, Ill do another paste |
| 19:24.52 | Alystair | yay for $186 phone bills \o/ |
| 19:26.10 | GTX | [TK]D-Fender - trelane : http://pastebin.ca/176555 |
| 19:27.37 | *** join/#asterisk dasenjo_ (n=dasenjo@208.195.215.116) |
| 19:27.57 | *** join/#asterisk dahunter3 (n=dahunter@pool-71-110-86-116.lsanca.dsl-w.verizon.net) |
| 19:29.20 | *** join/#asterisk anthonyl (n=anthony@c-67-167-211-3.hsd1.il.comcast.net) |
| 19:30.44 | trelane_ | Qwell, bastard |
| 19:31.45 | *** join/#asterisk f0urtyfive (i=f0urtyfi@c-67-165-5-232.hsd1.ct.comcast.net) |
| 19:33.43 | trelane_ | GTX, would you paste the connection block from sip.conf please to pb? |
| 19:34.06 | GTX | Jam0r is the tech he'll paste it |
| 19:35.16 | *** join/#asterisk p1p (i=tjcomp91@mail.comp911.com) |
| 19:36.24 | *** join/#asterisk Ox0F0-0FF (n=pierre@200.216.238.226) |
| 19:38.52 | Jam0r | trelane_ see pm |
| 19:39.34 | *** join/#asterisk l-fy (n=l-fy@yate/developer/l-fy) |
| 19:40.33 | *** part/#asterisk l-fy (n=l-fy@yate/developer/l-fy) |
| 19:45.45 | rowter | whats a good way to copy audio files to another server, via sshfs, rsync? |
| 19:45.54 | Qwell | scp? |
| 19:46.23 | rowter | Qwell, compacting and then scp and then delete? |
| 19:46.44 | Qwell | probably not gonna get a whole hell of a lot of compression on audio files |
| 19:48.30 | *** join/#asterisk _alex_mx_ (n=alex@200.78.229.18) |
| 19:50.30 | rowter | Qwell, true |
| 19:53.45 | *** join/#asterisk trevarthan (n=trevarth@c-71-226-190-251.hsd1.ga.comcast.net) |
| 19:54.47 | trevarthan | hello. I'm using trixbox and the queue login extension is 1700*. However, I'm using a Linksys spa942 phone and it seems to dislike the * character. Basically, it won't let me dial the extension. Does anyone know how to fix that? |
| 19:57.54 | *** join/#asterisk zotz (n=zotz@24.244.163.225) |
| 19:57.58 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
| 19:58.14 | Assid | heya |
| 19:58.37 | Assid | err.. is there a way to make the mic of the polycom a bit less sensitive to background noise |
| 19:58.56 | Assid | but not work like walkie talkie |
| 19:59.16 | trevarthan | The asterisk extension config line looks like this: |
| 19:59.16 | trevarthan | exten => 1700*,1,Macro(agent-add,1700,) |
| 19:59.48 | trevarthan | unfortunately, it's trixbox/freepbx, so I can't really change the * character without causing trouble. |
| 20:00.23 | x86 | err |
| 20:00.28 | x86 | you want #freepbx |
| 20:00.42 | x86 | see the topic |
| 20:01.48 | *** join/#asterisk darkskiez (i=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 20:02.50 | *** join/#asterisk xAD (n=xAD@host144-199.pool8290.interbusiness.it) |
| 20:03.32 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 20:03.44 | robl^ | trevarthan: freepbx uses asterisk at the core, but is a separate product. you are better off asking those who know the specifics of freepbx |
| 20:09.04 | [TK]D-Fender | Assid: Adjust the mic gain in your provisioning files. |
| 20:09.09 | *** part/#asterisk _alex_mx_ (n=alex@200.78.229.18) |
| 20:09.31 | [TK]D-Fender | trevarthan: Your problem is with your phone. You need to go into its config and change it's dialplan. |
| 20:13.02 | *** join/#asterisk adorah (n=admin@84.94.144.7.cable.012.net.il) |
| 20:14.49 | Assid | [TK]D-Fender: voice.gain? |
| 20:15.28 | Assid | tx right ? |
| 20:16.51 | [TK]D-Fender | Assid: think so... something like that. |
| 20:16.53 | *** part/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 20:17.54 | Assid | sholdnt it be voice.ns ? |
| 20:18.09 | Assid | for the background noise supression? |
| 20:18.45 | Assid | okay o keep getting confused . -1 is 1 point lower than FULL right ? |
| 20:18.49 | [TK]D-Fender | Assid: I never did this personally. I'd read the admin guide if I were you and experiment a bit |
| 20:19.32 | [hC] | any of you guys using handset lifters with polycom phones? |
| 20:19.45 | [TK]D-Fender | [hC]: I have. DON'T |
| 20:19.45 | [hC] | Im using a plantronics hl10 but it definitely is not fully compatible |
| 20:19.52 | [hC] | Yeah, this is just not working out. |
| 20:20.00 | [TK]D-Fender | [hC]: Did try to warn you.... |
| 20:20.04 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 20:20.06 | *** mode/#asterisk [+o mog] by ChanServ |
| 20:20.13 | [hC] | Since there is no mouthpiece to catch the handset, it slides down and you cant hang up properly |
| 20:20.58 | [TK]D-Fender | [hC]: Our solution was to puta screw into the lifter top-bar that would engage the hook-switch and then leaev the handset off entirely. |
| 20:21.09 | [TK]D-Fender | [hC]: Ugly & easy hack. |
| 20:21.20 | [TK]D-Fender | [hC]: And a sad thing to do to such a nice phone. |
| 20:21.30 | [hC] | oh i see what you mean |
| 20:21.49 | trevarthan | [TK]D-Fender: yeah, gay regex dialplan support. Got it working. Thanks! |
| 20:21.53 | *** part/#asterisk trevarthan (n=trevarth@c-71-226-190-251.hsd1.ga.comcast.net) |
| 20:21.57 | [hC] | yeah that doesnt work exactly for this application... if i stick something underneath the bottom of the handset, so when it lifts it doesnt fall, that works |
| 20:22.03 | [hC] | but its hard to keep it aligned in a day to day setting |
| 20:22.58 | [TK]D-Fender | [hC]: My way you leave the handset COMPLETELY off. |
| 20:23.49 | [hC] | [TK]D-Fender: yeah.. i understand.. my clients use the handset occasionally |
| 20:24.22 | Assid | [TK]D-Fender: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg137845.html <-- check that |
| 20:24.24 | [TK]D-Fender | [hC]: You can keep it plugged, just leave it flat on the table. |
| 20:24.36 | [hC] | I think i also fried my 7970 today by spilling coffee on it :( |
| 20:24.39 | [TK]D-Fender | [hC]: Next time.... just get them a cordless phon on an ata! |
| 20:24.39 | [hC] | that was a bit of a pissoff |
| 20:25.22 | [hC] | Im going to look into using the headset jack and see what polycom offers for leaving the headset light on and dealing with calls that way |
| 20:25.23 | [hC] | i dunno |
| 20:26.36 | [TK]D-Fender | [hC]: Seriously. Stop screwing around and get them proper phones... |
| 20:26.57 | [hC] | the only thing i can do for these guys is cisco 7940s |
| 20:27.00 | [hC] | to be on par |
| 20:28.05 | *** join/#asterisk hypa7ia (i=hypatia@judecca.aculei.net) |
| 20:28.28 | *** join/#asterisk aydiosmio (i=aydiosmi@judecca.aculei.net) |
| 20:28.36 | aydiosmio | I'm back, biatches! |
| 20:28.41 | [TK]D-Fender | [hC]: Screw Par... they are LOSING functionality in attempting to become "wireless" It is getting fuglier and LESS functional by the minute... |
| 20:28.54 | luke-jr_work | Does #include sort when given a wildcard? |
| 20:28.56 | [hC] | I agree. |
| 20:30.30 | luke-jr_work | and does #include work in AEL2? |
| 20:30.35 | [TK]D-Fender | [hC]: If you want them to feel more exectutive get them *shudder* a WifI SIP phone. |
| 20:31.12 | [hC] | heh |
| 20:31.18 | murf | luke-jr_work: AEL2 supports #include "filename" |
| 20:31.21 | [hC] | [TK]D-Fender: i just rolled out a bunch of linksys wip300's for a client |
| 20:31.24 | aydiosmio | IP phone with cordless handset ftw. |
| 20:31.25 | [hC] | [TK]D-Fender: that was....... fun. |
| 20:31.26 | [hC] | :| |
| 20:31.42 | [TK]D-Fender | [hC]: Less than enjoyable? What kind of issues? |
| 20:31.53 | trelane_ | [hC], there was a really really nifty phone from RCA/Thompson, no idea if it ever made it out of R&D |
| 20:31.54 | luke-jr_work | murf: not #include "directory/*.conf"? |
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| 20:32.04 | aydiosmio | haha |
| 20:32.13 | aydiosmio | include with wildcard. that's a little creepy |
| 20:32.16 | murf | luke-jr_work: sorry, nope. |
| 20:32.23 | trelane_ | aydiosmio, noone cares |
| 20:32.27 | [TK]D-Fender | luke-jr_ : I'd guess that like any sane language you should NOT be allowed to load by wildcard... the order could matter. |
| 20:32.39 | trelane_ | aydiosmio, err disregard and apologies I was scrolled to "I'm back biatches" |
| 20:32.46 | aydiosmio | ah |
| 20:32.47 | aydiosmio | haha |
| 20:32.50 | trelane_ | yeah oops :) |
| 20:32.51 | trelane_ | sorry :) |
| 20:33.06 | luke-jr_work | [TK]D-Fender: well, other * confs allow wildcards... =p |
| 20:33.10 | trelane_ | if you wish feel free to insert a "noone cares" at that point (if it makes you feel better) |
| 20:33.25 | aydiosmio | duely noted |
| 20:33.34 | luke-jr_work | [TK]D-Fender: I'm trying to split switch cases into their own files... |
| 20:33.58 | murf | luke-jr_work: AEL2 is "special". sorry! |
| 20:34.08 | luke-jr_work | :( |
| 20:35.31 | murf | luke-jr_work: What's your situation? If I were to add this feature, how much use would it be? |
| 20:36.28 | luke-jr_work | murf: I want to switch() on the extension, and have non-default cases split out into their own files |
| 20:36.54 | *** join/#asterisk chaoscon (n=ph33r@smartserv/ceo/chaoscon) |
| 20:36.54 | luke-jr_work | so ext 2301 would be in its own conf if non-default |
| 20:37.19 | *** join/#asterisk southtel (n=slester@76.17.115.183) |
| 20:37.34 | [TK]D-Fender | luke-jr_ : how big is your setup? |
| 20:37.45 | murf | luke-jr_work: The problem is this: the includes are evaluated at compile-time, not run time... |
| 20:38.12 | luke-jr_work | murf: that's fine, I was expecting to do an asterisk -x 'ael reload' |
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| 20:40.04 | murf | luke-jr_work: OK, just so you know. I've got a wiki page (voip-info) for AEL2, where I'm handling enhancement requests. Feel free to add this there. If you don't I'll do it for you when I get around to it (after 1.4 is forked). |
| 20:41.23 | luke-jr_work | murf: but wiki'd requests get put in 1.4? :) |
| 20:42.36 | murf | luke-jr_work: They might have gotten in 1.4, if they got in the list early enough, and interested me enough to get them done in time. But right now, the freeze is on. |
| 20:42.41 | *** join/#asterisk pjz (n=pj@zachs.place.org) |
| 20:42.43 | southtel | Is there a way to "loopback" a call? |
| 20:43.16 | pjz | what's the best bootrom to run on polycom 501 phones? |
| 20:43.30 | southtel | I'm trying to have a call go from one device to another, but have a separate cdr entry for the inbound and outbound portions. |
| 20:43.43 | luke-jr_work | murf: wouldn't this be a bug fix since every other * config takes wildcard includes? ;p |
| 20:43.50 | [TK]D-Fender | pjz: 3.2.2. |
| 20:44.09 | pjz | [TK]D-Fender: from spip_ssip_botrom_3_2_2.zip ? |
| 20:44.40 | murf | luke-jr_work: I tried that approach with some other stuff I did. I can guarantee the answer is no. |
| 20:44.50 | [TK]D-Fender | pjz: Looks about right. |
| 20:45.38 | [TK]D-Fender | pjz: I have found that SIP 2.0.1. and BootROM 3.2.2. is very stable and increases audio quality on the IP 430 a lot as well. Its also "zippier" than previous releases |
| 20:45.55 | *** join/#asterisk chaoscon_ (n=ph33r@smartserv/ceo/chaoscon) |
| 20:46.18 | pjz | [TK]D-Fender: ah, cool. thanks for the info |
| 20:48.17 | luke-jr_work | murf: other stuff was inconsistent with the rest of asterisk? |
| 20:48.34 | murf | luke-jr_work: mainly because the original AEL (in 1.2) didn't have #include, and the AEL2 version doesn't have wildcards. This would definitely rate as an enhancement. But, hey, I'll supply a patch for stuff that didn't make it into 1.4. Life isn't over yet! |
| 20:49.43 | pjz | [TK]D-Fender: are the .cfg files compatible? |
| 20:49.58 | pjz | [TK]D-Fender: with 1.6.2 |
| 20:50.36 | luke-jr_work | murf: then I need to compile it >.> |
| 20:50.47 | [TK]D-Fender | pjz:No, this is a major release. I'd suggest a complete rebuild. |
| 20:51.08 | pjz | [TK]D-Fender: eep |
| 20:51.29 | murf | luke-jr_work: ha, compiling is the ****EASY**** part! |
| 20:51.35 | pjz | [TK]D-Fender: if I upgrade the bootrom do I have to upgrade the sip app? |
| 20:51.36 | luke-jr_work | murf: not without a compiler |
| 20:52.04 | murf | luke-jr_work: well, you've got a month or two to find and install one! |
| 20:52.17 | luke-jr_work | murf: I don't install compilers on production systems ;) |
| 20:53.16 | murf | luke-jr_work: Well, that's an excellent policy. You'll have to play with this on a dev system or somenthing! |
| 20:53.16 | [TK]D-Fender | pjz: Why are you looking to change the bootrom? |
| 20:53.34 | luke-jr_work | murf: ;p |
| 20:53.42 | *** join/#asterisk jaramani (n=inarticu@199.60.97.1) |
| 20:53.46 | pjz | [TK]D-Fender: I started off with a bunch of polycom 500s but now I have soem 501s |
| 20:53.51 | Alystair | I'm still confused how you would do rollovers using DIDs |
| 20:54.11 | murf | luke-jr_work: cheer up! Things can **always** get worse! |
| 20:54.16 | pjz | [TK]D-Fender: and the 501s won't boot with the old bootrom on the server |
| 20:54.46 | luke-jr_work | murf: yeah, like AEL support in 1.2 |
| 20:55.03 | pjz | [TK]D-Fender: because they come with 3.1 or something installed |
| 20:55.40 | murf | luke-jr_work: I've already published a patch for 1.2 for AEL2... is that better or worse? |
| 20:55.45 | luke-jr_work | I thought I heard ETA 1 week on a 1.4 beta a few weeks ago... o.O |
| 20:55.54 | luke-jr_work | murf: that requires compiling to use =p |
| 20:56.18 | luke-jr_work | and seeing that I've already worked around the bugs in my AEL files, it's not worth the trouble =p |
| 20:56.37 | murf | luke-jr_work: and I thought I had it bad. Just remember everyone, to allow plus/minus 6 months to any release date! |
| 20:57.00 | murf | luke-jr_work: <just joking, btw!> |
| 20:57.24 | [TK]D-Fender | pjz: I rant old releases on 501's just fine |
| 20:57.34 | pjz | [TK]D-Fender: old releases of what? |
| 20:57.40 | murf | luke-jr_work: Wait a minute. Take a good look at AEL2. I think you'll like it! |
| 20:58.03 | pjz | [TK]D-Fender: ah, looks like if I just take the bootrom off the server they run the old app that's on the server |
| 20:58.15 | luke-jr_work | murf: yes, I can't wait until 1.4 beta 1 =p |
| 20:58.31 | luke-jr_work | beta isn't supposed to be stable, why put it off? =p |
| 20:58.34 | [TK]D-Fender | pjz: 30X, 430, 50X, 6XX all support the latest SIP & BR. I'd suggest upgrading them all |
| 20:58.54 | murf | luke-jr_work: (and everybody else) hopefully you won't have to wait very long...! |
| 20:59.06 | pjz | [TK]D-Fender: yeah, I need to do that.. but I need to test them first to make sure my config-builder works with tiem |
| 20:59.34 | [TK]D-Fender | pjz: What builder do you use? How many phones do you manage? |
| 20:59.52 | *** join/#asterisk bigjb (n=nnnnbigj@195.60.10.114) |
| 21:00.00 | murf | luke-jr_work: There are some major features that would go missing if we released now. It'll be worth the wait. |
| 21:00.04 | pjz | [TK]D-Fender: I manage around 35 phones and I use a custom builder that I wrote in python |
| 21:00.08 | jaramani | serious * noob here<---- I want to integrate * with Nortel KSU to extend locals to remote softphones <challenge anyone?> |
| 21:00.21 | luke-jr_work | murf: all I'm waiting for is XMPP/Jingle and AEL2 =p |
| 21:00.40 | [TK]D-Fender | pjz: I use the magic of "search & replace" and am rebuilding my 30 over here (did my 3 at home already) |
| 21:01.14 | murf | luke-jr_work: saw some jingling going on... hang on a little longer! |
| 21:01.24 | luke-jr_work | hmm |
| 21:01.31 | Trazz | Whats the most stable * in a box now with the web gui, etc. now ? I have been using freepbx but have to reboot it often |
| 21:01.42 | luke-jr_work | http://www.voip-info.org/wiki/view/AEL+Example+Snippets suggests I really should post my extensions.ael somewhere |
| 21:02.39 | pjz | [TK]D-Fender: http://www.place.org/~pj/fullstarconf/fullstarconf.py.bowlderized |
| 21:02.49 | murf | luke-jr_work: Go for it. Add some good examples to the pile! |
| 21:02.59 | *** join/#asterisk Defraz (n=t0tal@fw.centrisys.com) |
| 21:03.07 | pjz | [TK]D-Fender: ICYC |
| 21:03.44 | murf | luke-jr_work: oops, gtg. kid stranded in town. C U later...! |
| 21:03.55 | *** join/#asterisk hmmhesays (n=hmmhesay@24-117-135-28.cpe.cableone.net) |
| 21:04.12 | luke-jr_work | heh |
| 21:04.23 | hmmhesays | yup exactly |
| 21:05.19 | jaramani | serious * noob here<---- I want to integrate * with Nortel KSU to extend locals to remote softphones <challenge anyone?> |
| 21:05.37 | Trazz | Whats the most stable * in a box now with the web gui, etc. now ? I have been using freepbx but have to reboot it often |
| 21:05.44 | Trazz | Tk ? |
| 21:05.55 | hmmhesays | why do you have to reboot it? |
| 21:06.09 | Trazz | it stops answering or allowing calls out |
| 21:06.14 | hmmhesays | jaramani: what does the nortel consist of? |
| 21:06.22 | Trazz | a reboot fixes it |
| 21:06.29 | Trazz | or a restart of * |
| 21:06.41 | Trazz | i have a cron to restart * daily |
| 21:06.44 | Trazz | i want to stop that |
| 21:07.00 | jaramani | hmmhesays: Nortel Meridian ICS, T1 in... lots of station modules and 8 ATA's |
| 21:07.11 | hmmhesays | Trazz: turn debug on |
| 21:07.19 | hmmhesays | jaramani |
| 21:07.20 | *** join/#asterisk Gunnar (n=gunnar@101.Red-80-24-171.staticIP.rima-tde.net) |
| 21:07.51 | jaramani | hmmhesays: ? |
| 21:08.04 | hmmhesays | i was going to say something but I lost my thought |
| 21:08.13 | hmmhesays | jaramani: how do you want to integrate it? |
| 21:08.15 | jaramani | lol |
| 21:09.05 | jaramani | hmmhesays: not sure, I could attach to the ATA's and allow forwarding from the local sets to the * implementation? |
| 21:10.20 | hmmhesays | what kind of ata's? |
| 21:11.54 | jaramani | hmmhesays: Nortel ATA devices -> convert KSU lines to standard analog to allow modems/fax/machines/non-nortel handsets |
| 21:11.58 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 21:12.31 | hmmhesays | jaramani, gotcha |
| 21:12.41 | hmmhesays | so you'd need to hook up some fxo boxes |
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| 21:14.20 | *** join/#asterisk aptura (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 21:14.26 | hmmhesays | jarami, pretty simple affair |
| 21:14.36 | hmmhesays | you could put a gateway on the trunk side of your pbx too |
| 21:14.46 | hmmhesays | I have done that a few times |
| 21:15.46 | jaramani | hmmhesays: that's what my rapid bit of research says (FXO). Is it possible to go right to the lines (no ATA) with FXS? |
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| 21:16.02 | hmmhesays | "right to the lines" ? |
| 21:16.40 | jaramani | hmmhesay: the ports that the nortel phones are connected to |
| 21:19.16 | hmmhesays | highly unlikely you can plug an ata right into the ports, usually pbx's run a propietary type of signaling |
| 21:20.32 | jaramani | as you say though, simplest would be to use a bunch of FXO's to the ATA lines... give them extensions and allow users to call forward to them |
| 21:21.43 | hmmhesays | yep |
| 21:22.19 | [TK]D-Fender | jaramani: What kind & how many lines do you take into your KSU right now? How many remote extensions are you looking to add? |
| 21:22.59 | [TK]D-Fender | jaramani: And how many existing Norstar stations do you have? |
| 21:23.19 | hmmhesays | and [TK]D-Fender come in to answer the questions i'm to lazy to ask |
| 21:23.21 | hmmhesays | ;) |
| 21:23.32 | jaramani | T1 in (20 lines?) -> looking to extend 2 to 4 out with *.... got about 40 stations in operation |
| 21:23.35 | hmmhesays | * s/answer/ask |
| 21:24.00 | hmmhesays | jarmani: you looking to extend a couple did's out to ip extensions? |
| 21:24.09 | [TK]D-Fender | jaramani: Do you have an extra T1 port on it? |
| 21:24.34 | *** join/#asterisk japerry (n=falc0n@216.231.51.209) |
| 21:24.38 | jaramani | extending DID's out would be good!... |
| 21:24.43 | hmmhesays | [TK]D-Fender: remember that discussion we had about quintum awhile back... this is the perfect scenario |
| 21:24.44 | jaramani | no... no extra T1 port |
| 21:24.59 | *** join/#asterisk bigjb (n=bigjb@195.60.10.114) |
| 21:25.09 | hmmhesays | jarmani: you can put a quintum gateway inline with the t1 and pick off did's you want to send out IP |
| 21:25.18 | [TK]D-Fender | jaramani: Have you considered ditching your Norstar setup as a whole for *? |
| 21:25.35 | *** join/#asterisk profounded (n=pro@ool-44c4eae2.dyn.optonline.net) |
| 21:25.40 | jaramani | quintum gateway?... I'll check that out... |
| 21:25.59 | robl^ | you can get a norstar digital set to SIP gateway.. works with asterisk.. |
| 21:26.00 | hmmhesays | you can put one inline between the telco and the trunk side of the pbx and pick off numbers based on dnis |
| 21:26.09 | jaramani | Yes, we'll go complete VOIP in the next 18 months... got lots of money in the analog still.... :( |
| 21:26.21 | [TK]D-Fender | And the Norstar gateway can be had REAL cheap nowadays/.... |
| 21:26.48 | robl^ | real cheap? last I saw was like $1200 for 24 port |
| 21:27.13 | hmmhesays | that's not bad |
| 21:27.39 | aydiosmio | Quintum? yipes. |
| 21:27.47 | [TK]D-Fender | robl^: $1000 now.... |
| 21:28.04 | *** join/#asterisk Winkie (n=urmom@cpc2-stre3-0-0-cust344.bagu.cable.ntl.com) |
| 21:29.04 | robl^ | and then use Aastra phones.. they are very Nortel like... so they won't feel foreign to the users |
| 21:30.13 | DasTech | polycoms rock and easy o setup anduse |
| 21:30.37 | hmmhesays | spa-942's are pretty good to, i'm working on auto config for mass provisioning of those right now |
| 21:30.41 | robl^ | polycoms are good.. aastra is good. snom is good.. |
| 21:31.12 | hmmhesays | CunningPike: fun, I was down there for training a few months back |
| 21:31.21 | hmmhesays | i much prefer mediatrix 1124's then |
| 21:31.33 | robl^ | Aastra though, are very similar in to Nortel phones.. Aastra actually manufactures phones for Nortel |
| 21:31.52 | CunningPike | hmmhesays: It was a tossup between Mediatrix and AudioCodes |
| 21:31.53 | *** join/#asterisk Winkie (n=urmom@cpc2-stre3-0-0-cust344.bagu.cable.ntl.com) |
| 21:32.15 | *** join/#asterisk lorinc (n=ang@caracas-2592.adsl.interware.hu) |
| 21:32.28 | hmmhesays | CunningPike: oh man I love the 1124's with all the features, mass provisioning capabilities |
| 21:32.48 | *** join/#asterisk bigjb (n=nbigjb@195.60.10.114) |
| 21:32.52 | CunningPike | hmmhesays: WTF were you when I was asking which one to buy?? :P |
| 21:32.56 | hmmhesays | audiocodes has a prettier gui |
| 21:33.05 | CunningPike | Pretty is goo |
| 21:33.11 | CunningPike | s/goo/good/ |
| 21:33.28 | hmmhesays | CunningPike: i'm where were you all of the times i've been spouting on about the 1124's? |
| 21:33.45 | hmmhesays | wow |
| 21:33.51 | [TK]D-Fender | robl^: What I was referring to is the Norstar gateway so they could use their existing Digitral phones |
| 21:33.54 | *** part/#asterisk DasTech (n=DasTech@d47-69-194-47.col.wideopenwest.com) |
| 21:33.59 | hmmhesays | extract something that meaningful out of that sentence |
| 21:34.00 | hmmhesays | lol |
| 21:34.08 | CunningPike | hmmhesays: Asleep at the wheel |
| 21:34.22 | CunningPike | hmmhesays: lol - I didn't even notice until you pointed it out |
| 21:34.23 | hmmhesays | I need a new guitar |
| 21:34.34 | CunningPike | hmmhesays: Because that will help you type better? |
| 21:34.37 | hmmhesays | CunningPike: classic case of fingers in front of hands |
| 21:34.49 | CunningPike | hmmhesays: Ah |
| 21:34.49 | hmmhesays | CunningPike: cause the ladies love guitarists |
| 21:34.58 | CunningPike | Cresl1n: Ewww |
| 21:35.10 | hmmhesays | my les paul is too fat for my fingers, and i'm indecisive |
| 21:35.18 | Cresl1n | I used to play guitar |
| 21:35.18 | aydiosmio | chuck it! |
| 21:35.26 | Cresl1n | then it got ganked :-) |
| 21:35.29 | *** join/#asterisk adorah (n=admin@84.94.144.7.cable.012.net.il) |
| 21:35.31 | Cresl1n | so no more guitar for a while |
| 21:36.00 | hmmhesays | I've got my eyes on a hamer double cut away, but then I talked to the guitarist from avenged sevenfold... and was thinking of going with a schecter |
| 21:36.30 | CunningPike | hmmhesays: Telecaster |
| 21:36.41 | [TK]D-Fender | ewww |
| 21:36.59 | aydiosmio | I often take instrument advice from mediocre muscians as well:0 |
| 21:37.01 | CunningPike | [TK]D-Fender: Made by........ Fender!! :D |
| 21:37.33 | hmmhesays | Tele's can be ok if you put some fat strings on them |
| 21:38.05 | X-Rob_ | Mmmm, fat strings *droool* |
| 21:38.16 | [TK]D-Fender | CunningPike: Fender doesn't do what I need. 24 fret double-locaking w/ floating trem. :) I play a Dean currently, and previously Ibanez |
| 21:39.03 | CunningPike | It's the fins........ |
| 21:39.29 | hmmhesays | [TK]D-Fender: that's right you're an 80's nut |
| 21:39.40 | hmmhesays | did you ever watch that video i posted the other day? |
| 21:39.46 | aptura | Its noteable that my step dad had a play in the 60s music erra as a group manager,big band booking agent and formed groups. |
| 21:39.50 | aydiosmio | all hail satriani |
| 21:39.53 | hmmhesays | synyster gates makes that schecter sing |
| 21:39.57 | [TK]D-Fender | hmmhesays: Yeah I time where music & songwriting still mattered :) |
| 21:40.10 | hmmhesays | [TK]D-Fender: it is getting back that weay |
| 21:40.10 | aptura | CunningPike did not know you liked music |
| 21:40.11 | hmmhesays | *way |
| 21:40.18 | aydiosmio | hey, not fair to generalize. |
| 21:40.24 | CunningPike | [TK]D-Fender: They still matter - it's just that no-one does it any more |
| 21:40.38 | [TK]D-Fender | aydiosmio: Satch is cool.... Via is abrasive, Malmsteen is a self-plagiarizing egotist |
| 21:40.43 | CunningPike | aptura: Purely as a consumer - I have no ability whatsoever |
| 21:40.43 | *** join/#asterisk Jeekay (n=jeekay@yellow.rgh.org) |
| 21:40.52 | hmmhesays | i disagree: atreyu, avenged sevenfold |
| 21:41.04 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-48-32.cybersurf.com) |
| 21:41.07 | aptura | CunningPike okay :) my wife in heavy into music as a cunsumer. |
| 21:41.12 | aydiosmio | yes, I'm sure, it's okay. |
| 21:41.27 | Jeekay | Whenever I try to call my grandstream phone from my Windows softphone (3CX), Asterisk returns a 404 Not Found. I can call the softphone from the grandstream just fine. What might be wrong / what can I do to get more debug information? |
| 21:41.32 | aptura | Went to school with the perlband drummer who was eventually kicked out of the group ;) |
| 21:41.34 | aptura | BRB |
| 21:41.42 | *** join/#asterisk asternick (i=asdas@222.126.38.74) |
| 21:41.46 | hmmhesays | that's why it is so fun going to see good bar bands, they play the best music from lots of era's |
| 21:41.56 | *** join/#asterisk kannan (i=1001@61.8.147.164) |
| 21:41.59 | asternick | hello can anyone help me |
| 21:42.21 | hmmhesays | asternick: the gerbil is dead dude, let it go |
| 21:42.27 | aydiosmio | I'd really rather listen to a jukebox than listen to someone butcher "Little Wing" |
| 21:42.31 | [TK]D-Fender | Jeekay: 404 means your dialplan doesn't have a match for the number dialed |
| 21:42.38 | CunningPike | Try pandora.com |
| 21:42.42 | hmmhesays | aydiosmio: i'd rather see someone good play it live |
| 21:42.46 | *** join/#asterisk ambriento (n=melcon@200.192.160.100) |
| 21:42.53 | CunningPike | asternick: Ask |
| 21:42.56 | CunningPike | ~justask |
| 21:43.00 | Jeekay | It does though, that's the strange thing. The extension I'm trying is 8192, and I have exten => 8192,blah entries in the appropriate contexts |
| 21:43.03 | hmmhesays | ~hmmhesays |
| 21:43.05 | jbot | you are, like, not really here... |
| 21:43.29 | [TK]D-Fender | Jeekay: pastebin your sip.conf (masking passwords), and your extensions.conf |
| 21:43.35 | aydiosmio | does anyone know anything about computers? |
| 21:43.36 | CunningPike | Jeekay: You're going to have pastebin your...... what he said |
| 21:43.43 | Jeekay | Will do - thanks :) one sec |
| 21:44.23 | CunningPike | asternick: Ask your question in the channel, so all may benefit from my wisdom |
| 21:44.31 | Jeekay | just as a quick sidequestion - is there a way to see all extensions that have currently registered? |
| 21:44.39 | hmmhesays | and my shenanigans |
| 21:44.41 | CunningPike | Jeekay: 'sip show peers' |
| 21:44.44 | asternick | can anyone help me because i have a PBX server running on a localnet and my clients are outside another local net |
| 21:45.00 | asternick | any idea on inux congifugration on that thing |
| 21:45.09 | Jeekay | ah |
| 21:45.15 | asternick | any idea on linux congifugration on that thing |
| 21:45.21 | Jeekay | Sorry for burning your time... I had the softphone in the wrong context - many thanks for your help! |
| 21:45.34 | hmmhesays | set your externip's or use openvpn asternick |
| 21:45.45 | *** join/#asterisk saftsack (n=oliver@p54A7C5C2.dip.t-dialin.net) |
| 21:45.56 | asternick | ah gonna try that |
| 21:46.13 | *** join/#asterisk phatmonkey (i=nobody@81.2.121.150) |
| 21:46.33 | [TK]D-Fender | NEXT!!!!! |
| 21:47.12 | asternick | well thanks hmmhesays |
| 21:47.42 | hmmhesays | openvpn just rocks |
| 21:47.47 | hmmhesays | that way they are on the same network |
| 21:47.48 | *** join/#asterisk c4t3l (n=c4t3l@69.15.174.114) |
| 21:48.39 | aydiosmio | software vpn is tacky |
| 21:48.40 | phatmonkey | how can i have a list of prefixes in the dialplan (_500. etc), with a "catch all" at the bottom for any that didn't match? using _X. would just match everything regardless of whether a prefix matched it. is there a way to bail out of the current context that i can add to the prefix extensions? |
| 21:49.44 | hmmhesays | phatmonkey: use an include |
| 21:50.32 | phatmonkey | hmmhesays, how exactly? |
| 21:51.15 | Druken | mut: you around? |
| 21:51.53 | hmmhesays | phatmonkey [mycontext] include => mycontext2; exten => _500.,1,NoOp(500 extensions); [mycontext2] exten => _X.,1,NoOp(Catch the rest) |
| 21:51.59 | hmmhesays | that should do'er |
| 21:52.08 | asternick | hmmhesays i've tried downloading the iso image of trixbox and write it on a cd but it doesn't work so i just use a VMware player right now |
| 21:52.13 | asternick | would that be just ok? |
| 21:52.16 | Jeekay | is there a list of all the 'operations' you can use in extensions (NoOp, Dial, Hangup, etc) anywhere? |
| 21:52.34 | hmmhesays | feel free to donate me a 10'spot for that phatmonkey |
| 21:52.40 | phatmonkey | hmmhesays, ah i see, of course |
| 21:53.51 | *** join/#asterisk Brijn (n=bas@204.244.176.116.net-conex.com) |
| 21:53.53 | hmmhesays | bah, I can't find the tab for "unbelievable" anyway |
| 21:53.54 | Brijn | Hello all |
| 21:54.32 | hmmhesays | s/anyway/anywhere |
| 21:54.33 | *** join/#asterisk groogs (n=greg@d38-54-164.commercial1.cgocable.net) |
| 21:55.44 | Brijn | I have a question that probably comes up a lot.. But google & voip-info.org didn't help :( I have an * box in India, and I want to play MoH to get a feel for audio quality. I set up an IAX trunk and can phone MoH without problems. But as soon as it anwers it says "Start playing" and then "Stopped playing" |
| 21:55.56 | Brijn | The format_mp3 is there, and ztdummy is loaded |
| 21:57.03 | hmmhesays | is there mp3's in your moh directory? |
| 21:58.00 | Brijn | yes |
| 21:58.09 | Brijn | default install with sounds |
| 21:59.18 | *** join/#asterisk AlexDaSilva (n=asterisk@static-71-245-156-4.alb.east.verizon.net) |
| 21:59.56 | *** join/#asterisk zeppelin_ (n=zeppelin@201.41.0.159) |
| 22:00.16 | stubert | Brijn: how are you calling moh in your dialplan? |
| 22:00.17 | Jeekay | Hm.. if I have a dialplan that looks like exten => 100,1,Dial(SIP/100,30,t), how can I play different things depending on whether the person didnt answer or the extension is invalid? Both cases seem to just go onto the next priority. |
| 22:00.26 | X-Rob_ | Brijn, check /var/log/asterisk/full |
| 22:00.40 | X-Rob_ | also do a 'moh files show' |
| 22:00.49 | X-Rob_ | make sure that asterisk can actually _See_ the files |
| 22:00.56 | CunningPike | Jeekay: Old way: Have both an n+1 and an n+101 priority |
| 22:01.06 | CunningPike | Jeekay: New way, check ${DIALSTATUS} |
| 22:01.22 | justinu|laptop | except when app_dial forgets to set DIALSTATUS |
| 22:01.34 | Jeekay | I can only make it go to 102 if the user actively BUSYs the call |
| 22:01.35 | Brijn | X-Rob_: Hmmm, doesn't show any files.. Maybe permission issue |
| 22:01.39 | sx-wks | justinu|laptop: then that's a bug |
| 22:01.41 | Jeekay | if its just not answered it goes onto prio 2 |
| 22:01.56 | Jeekay | of course the other question is.. how do I use DIALSTATUS? :) |
| 22:02.02 | CunningPike | Jeekay: Precisely - isn't that what you wanted? |
| 22:02.09 | sx-wks | Jeekay: use GotoIf |
| 22:02.21 | Brijn | X-Rob_: Everybody can read, so that should not be the problem |
| 22:02.27 | Jeekay | I need to be able to distinguish between an unanswered call (phone is there, phone rang, noone answered) versus a call to a non-existant extension |
| 22:02.55 | Jeekay | oh wait |
| 22:03.07 | Jeekay | I've just been dumb again havn't I.. A non-existant extension won't be configured at all. |
| 22:03.15 | CunningPike | Jeekay: :D |
| 22:03.34 | Jeekay | Right I see why... when I dial non-existant extensions, it rings busy |
| 22:03.43 | Jeekay | I need that do to the 'the extension you have dialed is invalid' thing |
| 22:04.37 | AlexDaSilva | Hello everyone could I ask a question ? DTMF detection from some callers but not all incoming zap channel |
| 22:04.49 | CunningPike | Jeekay: Then you need an 'i' extension in the current context and make it PlayBack() the appropriate sound file |
| 22:04.59 | CunningPike | AlexDaSilva: Inband? |
| 22:05.00 | Jeekay | you're a star |
| 22:05.03 | Jeekay | thankyou |
| 22:05.09 | AlexDaSilva | not sure |
| 22:05.16 | CunningPike | AlexDaSilva: Find out |
| 22:05.18 | CunningPike | :D |
| 22:05.33 | AlexDaSilva | how i thought inband would only apply to a SIP not ZAP channel |
| 22:05.41 | X-Rob_ | Brijn, have you restarted asterisk? (MOH changes only get noticed on a restart) |
| 22:06.53 | CunningPike | Brijn: reload res_musiconhold.so should be enough in most cases |
| 22:07.34 | Jeekay | hmm.. when I dial an invalid extension, asterisk keeps sending a 404 instead of following my dialplan.. |
| 22:08.07 | pigpen | anyone know of any iax providers with a single channel LD monthly rate? |
| 22:08.58 | Jeekay | oh. the asterisk info pages suggests that i doesn't infact work? |
| 22:10.33 | Brijn | X-Rob_: Ahh, i just reload.. Let me restart' |
| 22:11.33 | AlexDaSilva | CunningPike, wouldn't inband apply only to a SIP channel ? The problem I'm having is on a ZAP channel, I know I said it before but you didn't respond, really need the help |
| 22:11.37 | [hC] | hmm. is there a way to have a queue call ring for a certain amount of time if all members are ringing, but ring for longer if all the members were busy? |
| 22:12.15 | CunningPike | AlexDaSilva: Hmm - are you receiving the calls with SIP phones? |
| 22:12.30 | pigpen | wow...that many ....hehe...longshot. |
| 22:12.38 | AlexDaSilva | callers can't go passed the autoaattendant |
| 22:13.15 | CunningPike | AlexDaSilva: Ah, I see - try playing with your gains |
| 22:13.27 | CunningPike | AlexDaSilva: In zapata.conf |
| 22:13.43 | AlexDaSilva | uhu i do have rxgain to 25 cause they complained about volume |
| 22:13.53 | Brijn | X-Rob_: How do I get the full log? I only have messages in var/log/asterisk |
| 22:14.05 | CunningPike | AlexDaSilva: That sounds really high - is that rxgain? |
| 22:14.09 | AlexDaSilva | yes |
| 22:14.25 | CunningPike | AlexDaSilva: Back it off a ways and see if that cures the DTMF |
| 22:15.40 | [hC] | what method do you guys use to set rx/txgain? and do you usually end up having to do it per line (on, say, a 5-6 line system) or do you just set it globally? |
| 22:15.56 | AlexDaSilva | CunningPile I'll try that then thank you. It's only a problem from one person with a Cingular Cell phone and a couple of other ramdom callers. I can never recreate the issue when I call |
| 22:18.14 | Jeekay | http://pastebin.com/790250 is my extensions file.. any ideas why the [invalid] context at the bottom isn't working? (my phones are in the [intern] context defined at the top |
| 22:18.24 | Brijn | X-Rob_ / CunningPike : I evenr ebooted the server. Still no files listed in "moh files show", the directory listed in musiconhold.conf is there, and it has the three default files... They are readable by owner,group,other |
| 22:19.12 | CunningPike | AlexDaSilva: Cell phones are notoriously problematic with DTMF |
| 22:20.04 | CunningPike | Jeekay: Try pastebin.ca - pastebin.com is fscked up |
| 22:20.12 | jaramani | got sucked into the "work" blackhole for a moment... circled the event horizon at light speed... |
| 22:20.35 | Jeekay | http://pastebin.ca/176753 |
| 22:20.47 | CunningPike | Brijn: Pastebin the output from 'moh files show' |
| 22:21.25 | Brijn | No need to post it, it's empty ;-) |
| 22:21.53 | jaramani | I'll investigate > T1<->Nortel KSU<->ATA<->*FXO<->Remote Softphone some more |
| 22:22.00 | AlexDaSilva | CunningPike, unfortunately I can't tell that to my client, specially when he has clients calling in and complaining. Thanks for your help lowering the rxgain has actually made it worse but i'll keep playing with these settings |
| 22:22.03 | jaramani | thanks for the input |
| 22:23.04 | X-Rob_ | Brijn, check to make sure the directory has the correct permissions. Try to CD into it as the asterisk user |
| 22:23.16 | Brijn | CunningPike: http://pastebin.ca/176755 has the musiconhold.conf and dirlist |
| 22:23.21 | Brijn | X-Rob_: let me check |
| 22:23.25 | X-Rob_ | Brijn, apart from that, check to make sure you're loading format_mp3 before res_musiconhold in modules.conf |
| 22:23.48 | CunningPike | Jeekay: I misled you - 'i' is for auto-attendants. You need a [default] context with an _X. extension that plays the invalid sound |
| 22:24.35 | Jeekay | Where is a good reference for all the various commands (like GotoIf) the variables (DIALSTATUS?) and these patterns (_X.) ? |
| 22:24.49 | Brijn | X-Rob_: * is running as root |
| 22:24.51 | Jeekay | the documentation linked off the asterisk website as Digiums Documentation doesn't seem terribly helpful |
| 22:25.27 | CunningPike | ~thewiki |
| 22:25.32 | jbot | methinks thewiki is at http://www.voip-info.org/wiki-Asterisk |
| 22:25.32 | Brijn | X-Rob_: It's a recent install, modules.conf has autoload and specific load of res_musiconhold. Added format_mp3.so and stop/started *, no change |
| 22:25.33 | CunningPike | ~thebook |
| 22:25.35 | jbot | thebook is, like, a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 22:26.08 | X-Rob_ | Brijn, format_mp3 _above_ res_musiconhold? |
| 22:26.27 | Brijn | yes |
| 22:26.40 | X-Rob_ | don't know then 8) |
| 22:26.52 | CunningPike | Brijn: Change your mode to 'files' if you want to use format_mp3.so |
| 22:27.01 | Brijn | X-Rob_: I enabled full logging, not much goes into that file :( |
| 22:27.05 | X-Rob_ | [default] |
| 22:27.05 | X-Rob_ | mode=files |
| 22:27.05 | X-Rob_ | directory=/var/lib/asterisk/mohmp3/ |
| 22:27.05 | X-Rob_ | random=yes |
| 22:27.06 | fall0ut | anybody used the 7941/7961 phones yet? |
| 22:27.13 | X-Rob_ | That's what your moh context should look like |
| 22:27.19 | X-Rob_ | (with or without random=yes) |
| 22:27.54 | Jeekay | I take it the _X. has to be the last extension defined to avoid causing trouble? |
| 22:28.58 | Brijn | X-Rob_: Bingo!!! |
| 22:29.53 | CunningPike | Jeekay: No, it does has to be in an otherwise empty default context - you never normally use the default context for anything |
| 22:30.11 | CunningPike | Jeekay: At least, that's what we've done |
| 22:30.27 | Brijn | Hmm, i would probably be a nice idea to ship * with a working config ;-)) This was a brand new install |
| 22:30.38 | Brijn | s/i/it/ |
| 22:31.25 | Jeekay | If it's in a context that nothing uses, how does it ever get called? |
| 22:31.33 | CunningPike | Brijn: Pretty hard to do, given the variety of environments out there. I do, however, agree that the default install should use native MOH (format_mp3.so) |
| 22:31.49 | CunningPike | Jeekay: Your calls will go there if there is no match anywhere else |
| 22:32.25 | Brijn | And I read the comments: I read the "file" description as: Use this if you have all the MoH files in ulaw/alaw/gsm format so that no conversion is needed |
| 22:32.41 | Jeekay | is [default] 'special' in that way? |
| 22:32.51 | Brijn | CunningPike: Should I open a "bug" in mantis? Would that help |
| 22:32.56 | CunningPike | Jeekay: Yes |
| 22:33.22 | Jeekay | hm. if I move it out of my used context, it just rings busy again |
| 22:33.22 | *** join/#asterisk myshenka (n=spamyous@82.153.170.213) |
| 22:33.39 | CunningPike | Brijn: Yes - we use ulaw files to match our call codec |
| 22:34.04 | CunningPike | Brijn: It's not a bug - asterisk doesn't work out of the box - you have to configure it |
| 22:35.03 | *** part/#asterisk myshenka (n=spamyous@82.153.170.213) |
| 22:35.18 | Brijn | CunningPike: True, but by changing musiconhold.conf to this setup it would work in 100% of the installs I think? Now it will not work in any setup after a fresh install? |
| 22:35.52 | *** join/#asterisk bigjb (n=bigjb@195.60.10.114) |
| 22:36.27 | CunningPike | Brijn: This setup requires something out of asterisk-addons - not sure if that could be incorporated into the main product or not |
| 22:36.52 | teleweb | hi |
| 22:37.06 | CunningPike | Jeekay: Pastebin your CLI output and your current extensions.conf |
| 22:37.19 | teleweb | what's the easiest way to check whether your rtp streams runs through asterisk or directly from phone to phone? |
| 22:37.23 | Brijn | CunningPike: OK, tx! I'm now listening to an MP3 from India to Vancouver ;-) For a 32kbit uplink via shitty internet, the quality is surprisingly good |
| 22:38.19 | Brijn | Is iax2 show netstats functional? |
| 22:38.59 | teleweb | I'm talking about sip |
| 22:40.26 | CunningPike | teleweb: wireshark |
| 22:41.43 | Brijn | Can I force MoH to be cached in memory.. If I browse quickly in a textfile on the PBX I hear it in the MoH playback :( or will the Linux disk cache do that automayicly after the first round |
| 22:42.10 | CunningPike | Brijn: I think it is - I think it only gives stats with an active call |
| 22:42.42 | CunningPike | Brijn: That shouldn't be happening with local moh files..... |
| 22:43.04 | teleweb | www.wireshark.org does not work :( |
| 22:43.06 | Brijn | CunningPike: I thought I had enabled jitterbuffer, but I probably overwrote that file at some point.. Resetting it now |
| 22:43.20 | teleweb | is there no easier way, without installing extra software? |
| 22:43.46 | CunningPike | teleweb: There's no way that I know of to be certain, afaik |
| 22:45.00 | *** join/#asterisk Ox0F0-0FF (n=pierre@200.216.238.226) |
| 22:46.32 | CunningPike | teleweb: But I am far from an expert.... |
| 22:46.38 | *** join/#asterisk twisted[work] (n=twisted@pdpc/supporter/active/twisted) |
| 22:46.38 | *** mode/#asterisk [+o twisted[work]] by ChanServ |
| 22:46.49 | CunningPike | teleweb: wireshark is useful anyway :) |
| 22:46.54 | *** join/#asterisk SwK[Work] (n=SwK@64.89.118.139) |
| 22:46.58 | teleweb | thanks :) |
| 22:47.05 | teleweb | but can you access the site? :p |
| 22:47.25 | teleweb | ah!!! it works now |
| 22:47.33 | teleweb | was down a minute ago |
| 22:48.04 | Brijn | teleweb: The first DNS lookup probably failed.. You sometimes see that, if the DNS needs more then 2 seconds to answer |
| 22:48.18 | Brijn | Next request will work fine, since your local DNS then has the asnwer |
| 22:48.51 | teleweb | that could be the reason yes |
| 22:49.32 | hmmhesays | ah ah ah ah staying alive? |
| 22:49.55 | teleweb | related question: is it possible for 2 sip phones at the same location to use a hosted asterisk on the internet, but send their rtp media path directly to one another |
| 22:50.04 | CunningPike | "If I can't have you, I don't want nobody, baby..." |
| 22:50.16 | teleweb | and still keep the transfer functionality of * ? |
| 22:50.46 | teleweb | (i can use sip info for dtmf if that helps) |
| 22:51.16 | CunningPike | teleweb: It might work if you use INFO or RFC2388 for DTMF and there's no NAT to bugger up the reinvite |
| 22:51.29 | CunningPike | teleweb: Try it! :) |
| 22:51.47 | teleweb | well, there is nat between the phones and *, but not between the phones themselves :) |
| 22:52.55 | *** join/#asterisk teknoprep (n=tekon@unaffiliated/teknoprep) |
| 22:53.19 | teleweb | where should i install that shark anyway? :) can I just put it on a pc in the home lan? (i don't feel like letting it swim in the hosting pbx) |
| 22:54.58 | Brijn | Not a 100% asterisk question. But I try to compile speex since that seems to be the lowest BW (free) codec. It fails with configure: error: C++ preprocessor "/lib/cpp" fails sanity check... Anyone an idea what that might be |
| 22:56.09 | CunningPike | teleweb: wireshark can go anywhere where it can see the packets |
| 22:56.42 | CunningPike | teleweb: The NAT between asterisk and your phones _might_ wreck the reinvite - you'll have to try it |
| 22:57.42 | *** join/#asterisk RoyK (n=roy@ti211210a080-0359.bb.online.no) |
| 22:57.53 | *** join/#asterisk w12zard (n=w12zard@206.186.72.162) |
| 22:58.31 | w12zard | Howdy |
| 22:58.51 | *** join/#asterisk anthonyl (n=anthony@c-67-167-211-3.hsd1.il.comcast.net) |
| 22:59.23 | *** join/#asterisk asternick (i=asdas@222.126.38.74) |
| 22:59.40 | Jeekay | if, from my SIP phone registered to my local asterisk, I dial a numnber that is handed off to another SIP provider |
| 22:59.50 | Jeekay | is the RTP stream between my phone and the provider, or my phone to asterisk to the provider? |
| 22:59.51 | asternick | can anyone help me download that openVPN |
| 22:59.57 | asternick | please |
| 23:00.39 | CunningPike | Jeekay: Most likely the latter |
| 23:00.41 | eKo1 | Help you download? |
| 23:00.50 | CunningPike | Everyone pull!!!!! |
| 23:00.53 | asternick | yeah |
| 23:01.01 | asternick | how to download it |
| 23:01.08 | w12zard | Is there a way to execute an AGI script after sending a busy signal? |
| 23:01.22 | w12zard | can I log congestion to a database? |
| 23:01.31 | *** join/#asterisk ToTo (n=ToTo@host149-109-dynamic.58-82-r.retail.telecomitalia.it) |
| 23:01.31 | anthonyl | openvpn.org .. |
| 23:03.24 | Brijn | Haha, iax2 show netstats after I started a download on a 32kbit line: |
| 23:03.24 | Brijn | Jit Del Lost % Drop OOO Kpkts |
| 23:03.24 | Brijn | 4961 4270 16777201 22 53 933 18 |
| 23:03.42 | w12zard | I'm trying to make a cheap callback service for my cellphone |
| 23:03.54 | *** join/#asterisk justinu|laptop (n=Justin@12.44.122.130) |
| 23:03.56 | w12zard | where I get my PBX to call me back and then I can dial out on my home line |
| 23:03.59 | w12zard | Is that possible? |
| 23:05.05 | eKo1 | w12zard: yes |
| 23:05.32 | eKo1 | You can have your PBX call two parties and unite them so-to-speak. |
| 23:05.39 | eKo1 | your * PBX that is. |
| 23:05.53 | justinu|laptop | re-unite pangea! |
| 23:06.12 | w12zard | it would have to call me first and give me access to DISA or something...right? |
| 23:07.21 | eKo1 | No, it would call your cellphone and call the number you want to call from your cellphone. |
| 23:08.20 | eKo1 | How are you going to do that...well, I'll leave that to you to figure out. |
| 23:09.05 | w12zard | I'm just wondering if I can initiate the callback from my phone in the first place |
| 23:09.20 | *** join/#asterisk RoyK (n=roy@ti211210a080-0359.bb.online.no) |
| 23:09.45 | *** join/#asterisk QbY (n=Kelvin@cm-64-221-172-88.dhcp.southerncoastalcable.net) |
| 23:09.51 | eKo1 | Yes I said. |
| 23:09.56 | hmmhesays | of course you can |
| 23:09.57 | w12zard | oops sorry |
| 23:09.59 | QbY | server crashed, and was restarted--now i'm getting echo*CLI> iax2 debug |
| 23:09.59 | QbY | No such command 'iax2' (type 'help' for help) |
| 23:09.59 | QbY | echo*CLI> sip debug |
| 23:09.59 | QbY | No such command 'sip' (type 'help' for help) |
| 23:12.31 | hmmhesays | so type help and see what it tells you in its seemingly broken state |
| 23:13.08 | QbY | i've already rebooted the box |
| 23:13.20 | QbY | is anyone else having sponatenous crashes with 1.2.12.1? |
| 23:13.30 | QbY | no warnings int he log or nothing--just poof.. dead. |
| 23:14.06 | hmmhesays | turn debug on to check the last thing that happens before it crashed |
| 23:14.38 | QbY | debug the last event in the log? |
| 23:15.43 | w12zard | ok thanks |
| 23:16.01 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 23:19.23 | Nugget | http://www.joelonsoftware.com/items/2006/09/19b.html <-- a good example of why I hate (most) mobile phones. |
| 23:20.39 | Mportnoy | Anyone knows how to fix error SIP 500 INTERNAL "SERVER ERROR" |
| 23:20.47 | *** join/#asterisk robin_sz (n=robin@adsl.redpoint.org.uk) |
| 23:21.17 | robin_sz | people, what is the general opinion of the aastra phones, 9113i for example, good, bad or ugly? |
| 23:21.37 | robin_sz | sorry, 9133i |
| 23:22.08 | robin_sz | looks nice in the phot and is cheap ... but there again the GXP2000 looked nice ... |
| 23:23.13 | *** join/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
| 23:23.23 | redder86 | hrmm... it appears to be habitual now for an Asterisk release to be followed the next week by a .1 oops-fix. How very... telling |
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| 23:26.51 | *** join/#asterisk dasenjo (n=dasenjo@63.245.86.95) |
| 23:27.02 | dasenjo | Hi everybody! |
| 23:28.52 | sting3r | can someone tell me why 1.1 kills my conferences? |
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| 23:29.04 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
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| 23:32.04 | dasenjo | I need a FreeBSD user that can help me test destar, a Management Interface for asterisk. |
| 23:32.16 | *** join/#asterisk RoyK (n=roy@ti211210a080-0359.bb.online.no) |
| 23:32.23 | dasenjo | volunteers? |
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| 23:40.22 | CunningPike | Nugget: I love Joel - rofl |
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