| 00:03.00 | *** join/#asterisk jksM (jks@193.189.93.254) |
| 00:03.07 | *** join/#asterisk Jhirley (~Jhirley@adsl-3-193-57.mia.bellsouth.net) |
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| 00:14.51 | Jhirley | what would cause an IP 601 to stay in DND mode ? Besides the button on the phone ? |
| 00:24.52 | *** join/#asterisk riddlebox (~riddlebox@75-132-225-75.dhcp.stls.mo.charter.com) |
| 00:31.53 | *** join/#asterisk matt_d (~matt@70.134.98.183) |
| 00:35.14 | Kobaz | Jhirley: the phones xml config |
| 00:35.21 | Kobaz | you might have divert configured |
| 00:36.16 | Jhirley | ty, let me look |
| 00:38.59 | *** part/#asterisk Anomizer (~Anomizer@adsl-225-4-178.mia.bellsouth.net) |
| 00:55.05 | antiwire | Where would I go about enabling *86 to allow any user to connect to voicemail and be prompted to enter their mbox number and password? |
| 01:01.31 | iamthelostboy | im running asterisk 1.6.2.0, call-limit is definitely working, setting it to 1 means only 1 call is possible, though setting busylevel=1 in sip.conf alters the information when i do a sip show peer XXX, the phone will still recieve calls |
| 01:02.56 | TJNII | antiwire: exten => *86,1,VoiceMailMain() |
| 01:04.32 | antiwire | thanks TJNII |
| 01:04.46 | antiwire | For some reason, it keeps telling me that my login is incorrect |
| 01:05.10 | antiwire | I'm entering my MB #, hitting pound, entering my password which is just 1234 right now and hitting poind |
| 01:05.22 | TJNII | Does it not seem to react, and then say that as if it is timing out? |
| 01:05.48 | antiwire | Nope, I can see my digits in asterisk CLI but it says login incorrect |
| 01:05.59 | antiwire | It seems to know that I am entering digits and the pound |
| 01:06.24 | antiwire | my client is ekiga |
| 01:08.44 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 01:13.23 | *** join/#asterisk Jhirley (~Jhirley@adsl-145-0-148.mia.bellsouth.net) |
| 01:21.29 | antiwire | I'm set to RFC2833 and I can navigate my mail box if I set a specific dial plan code to just send me to mail box but if I use exten => *86,1,VoiceMailMain() and enter my mb # and password as defined in voicemail.conf it says login incorrect still |
| 01:24.22 | antiwire | for example, using this exten => 999,1,VoiceMailMain(102@mbox) accepts my password |
| 01:24.23 | *** join/#asterisk riddlebox (~riddlebox@75-132-225-75.dhcp.stls.mo.charter.com) |
| 01:24.42 | TJNII | Check the default voicemal context. |
| 01:25.43 | antiwire | Can I specify the VM context inside of VoiceMailMain() ? |
| 01:26.08 | antiwire | in my voicemail.conf, I have only [mbox] and two mailbox entries under that |
| 01:26.35 | TJNII | Not sure, never tried. Check the docs. |
| 01:28.19 | *** join/#asterisk neurosys (~neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
| 01:28.30 | antiwire | TJNII: Thanks! fixed it |
| 01:28.35 | antiwire | It was the context problem |
| 01:29.18 | antiwire | I've been working with Shoretel and Zultys systems and I decided I wanted to see how it's done using asterisk. This is awesome stuff |
| 01:42.28 | *** join/#asterisk Kumbang (~kumbang@rusnas.paume.itb.ac.id) |
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| 02:37.57 | p3nguin | kobaz: http://www.wsu.edu/~brians/errors/its.html |
| 02:41.54 | ChannelZ | You seriously need a girlfriend |
| 02:43.36 | jaytee | definitely |
| 02:43.49 | TJNII | ChannelZ++ |
| 02:43.54 | jaytee | grammar nazis have way too much free time on their hands |
| 02:44.23 | ChannelZ | Oh, his hands are busy |
| 02:44.46 | TJNII | ZING! |
| 02:52.25 | *** join/#asterisk random_mike (~MrT@deathstar.corp.adam.com.au) |
| 02:53.17 | random_mike | Hey all, can anyone advise if it is possible to execute a command line entry, to return the number of active sip channels on an asterisk server? |
| 02:53.44 | random_mike | I thought perhaps "asterisk -rx sip show channels" would work, but I get an error :( |
| 02:54.41 | ChannelZ | do asterisk -rx "sip show channels" |
| 02:55.10 | random_mike | oh bash requires quotes |
| 02:55.12 | random_mike | ofcourse |
| 02:55.16 | random_mike | thanks :) |
| 02:55.23 | ChannelZ | well it's many things that are really one argument |
| 02:56.08 | random_mike | works awesomely :) |
| 02:56.09 | random_mike | thanks |
| 02:56.30 | random_mike | asterisk -rx "sip show channels" | grep "active SIP channels" | awk {' print $1 '} |
| 02:56.43 | random_mike | returns how many calls in process on my asterisk server using SIP :) |
| 02:57.21 | ChannelZ | hurray! |
| 02:58.17 | *** join/#asterisk rgsteele (~rgsteele@c-71-230-37-172.hsd1.pa.comcast.net) |
| 03:03.39 | *** join/#asterisk nbash (~NickBenne@ip68-229-161-58.om.om.cox.net) |
| 03:10.08 | TJNII | "There is no such thing as a PNP FET." Why are ignorant people so vocal? |
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| 03:25.25 | Jhirley | guys, I have a Polycom 601 that is set in DND Mode no matter what I try ? I have blown away the phone , downloaded firmware 3.1.6 from polycom but still no luck ?Anyone have any ideas or input ? |
| 03:29.29 | *** join/#asterisk riddlebox (~riddlebox@75-132-225-75.dhcp.stls.mo.charter.com) |
| 03:32.19 | nbash | I'm seeing astrisk eat 100% cpu...anyone have any tips? or suggestions |
| 03:39.04 | dlynes | nbash, what're you running it on? |
| 03:41.08 | nbash | debian |
| 03:41.23 | nbash | looking at a scrpt to fix the issue atm |
| 03:41.33 | nbash | guess its something to do with running -c |
| 03:41.50 | dlynes | nbash, i meant what processor are you running it on? |
| 03:41.51 | doneir | i doubt it, i'm running asterisk -cvvvvgd and it's fine (debian) |
| 03:42.11 | doneir | this is with a digium card though |
| 03:42.12 | nbash | reading here http://svnview.digium.com/svn/asterisk/branches/1.4/contrib/init.d/rc.debian.asterisk?view=markup&pathrev=251309 |
| 03:42.38 | nbash | I'm on a vps |
| 03:44.20 | dlynes | nbash, which version of asterisk? |
| 03:44.37 | nbash | https://issues.asterisk.org/view.php?id=16784 |
| 03:45.14 | nbash | 1.6.2.6 |
| 03:45.28 | nbash | just did a build on a fresh system with it |
| 03:46.09 | *** join/#asterisk Jhirley_ (~Jhirley@adsl-161-2-232.mia.bellsouth.net) |
| 03:46.42 | dlynes | 1.6.2.6? The latest is only 1.6.2.5 |
| 03:47.33 | dlynes | nvm....somebody hasn't updated the topic yet |
| 03:48.16 | nbash | yeah I just DL and built it....just telling ya what the sorce said |
| 03:48.39 | nbash | I removed the -c from 1 line and doing a reboot to test |
| 03:49.20 | dlynes | nbash, hrm...weird...my startup script doesn't have a '-c' switch |
| 03:49.50 | dlynes | nbash, I'm running asterisk 1.6.1.8 |
| 03:50.03 | dlynes | on Debian Lenny, if it makes a difference |
| 03:50.51 | nbash | fixed it...my "asterisk" in /etc/init.d had this line "start-stop-daemon --start --oknodo --background --exec $DAEMON -- $ASTARGS -c" I removed the -c and rebooted |
| 03:51.35 | nbash | dont know if its a new bug but from that link above it looks to have been around awhile |
| 03:51.47 | dlynes | nbash, weird....I've just got this: 71 start-stop-daemon --start --oknodo --exec $DAEMON -- $ASTARGS |
| 03:51.51 | nbash | just did a "top" command and its empty |
| 03:52.01 | nbash | meaning its idle |
| 03:52.04 | dlynes | see how I don't have the --background, either? |
| 03:52.19 | nbash | did you do apt-get to install? |
| 03:52.23 | dlynes | hell, no |
| 03:52.24 | nbash | or build yourself |
| 03:52.28 | nbash | hmm |
| 03:52.29 | dlynes | built it myself |
| 03:52.38 | nbash | dont know then |
| 03:52.39 | dlynes | I'd never install asterisk from binary |
| 03:52.57 | p3nguin | Someone had to compile it. |
| 03:54.43 | dlynes | what's the 'c' wart mean beside a package name in aptitude again? |
| 03:57.26 | nbash | on Lenny myself |
| 03:57.40 | nbash | opps |
| 03:59.40 | ManxPower-work | you want to do a "make install" to install the asterisk boot scripts |
| 04:00.27 | p3nguin | I thought it was make config. |
| 04:01.52 | nbash | it is make config |
| 04:02.20 | *** join/#asterisk OrNix (~ornix@l151-249-47.static.cn.ru) |
| 04:07.18 | ManxPower-work | Sorry, correct. "make config" |
| 04:32.01 | ChannelZ | and then do "make mybed" |
| 04:32.16 | TJNII | make asandwich |
| 04:36.15 | hhkahya | how can we making to work a2billing:P anyone now the a2billing working mechanism ? |
| 04:39.07 | *** join/#asterisk jhirley (~jhirley@adsl-3-192-201.mia.bellsouth.net) |
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| 04:39.51 | jmcdowell | Hello all |
| 04:40.48 | jmcdowell | has anyone ever experienced the following message.. "Invalid SIP message - rejected , no callid" |
| 04:41.11 | jmcdowell | I think it's related to the polycom side of things. |
| 04:42.00 | antiwire | what happens if you force a CID in the dial plan? |
| 04:42.24 | jmcdowell | I didn't know you could.. |
| 04:42.33 | jmcdowell | You mean from the phone it's self? Or within asterisk ? |
| 04:43.06 | jmcdowell | It should be noted, those messages appear even when the phone system is 100% idle. |
| 04:44.24 | p3nguin | "itself" |
| 04:45.03 | jmcdowell | I will look at it.. |
| 04:45.12 | jmcdowell | I will have to rtfm on what you stated. |
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| 05:02.15 | jmcdowell | I should only see 1 asterisk instance running.. |
| 05:02.19 | jmcdowell | Right ? |
| 05:02.59 | *** join/#asterisk voipmonk (~shido6@dsl-67-204-1-83.acanac.net) |
| 05:04.30 | jmcdowell | hmmmm |
| 05:05.17 | jmcdowell | I have 27 "asterisk -f -vvvg -c" processes listed.. |
| 05:05.29 | jmcdowell | Those of course appear to be part of a tree, but still. |
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| 05:17.43 | sawgood | Are the "Asterisk add ons" part of the AsteriskNOW 1.5 ISO, and/or are they added if you update after installing AsteriskNOW? |
| 05:25.59 | *** part/#asterisk sawgood (~sawgood@173-13-158-25-sfba.hfc.comcastbusiness.net) |
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| 06:45.15 | vandebo | feel free to point me to a more appropriate channel - Any recommendations for a per minute outbound provider (usa)? I was using voicepulse, but they've instituted a monthly minimum. |
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| 08:41.36 | contrabanda | hi, i need help with G729 codec. Where can i get good one? |
| 08:42.18 | kaldemar | contrabanda: from digium |
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| 09:12.01 | ik_5 | hello |
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| 09:39.19 | Akiraaa | What would/do you use to give users the ability to send/receive SMS with a VoIP network? The setup is small: 4 mobile phone lines via GSM-IP gateways. |
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| 10:12.03 | J4zen | Hi guys, i've been looking all over for a proper VoIP (asterisk compatible) DECT phone. I've tried and deployed a lot of SNOM M3's and have tried some M9's (unfortunatly they have faulty firmware and have been recalled). Anyone have good expierences with another DECT phone for desk/office usage? |
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| 10:15.41 | *** join/#asterisk OlafsenM (~omorika@193.198.31.85) |
| 10:26.12 | *** join/#asterisk pisg (~pisg@intrush.pl) |
| 10:27.18 | pisg | hi, i have problem when i`m using X-lite i can call, when i`m using SIP phones and set login and pass X-lite i can call and have : -- Executing [06681406XX@outline:1] Dial("SIP/konradnowak-00000000", "DAHDI/g0/06681406XX") in new stack == Everyone is busy/congested at this time (1:0/0/1) |
| 10:27.38 | pisg | what i must change in SIP phone if i can calling |
| 10:29.53 | kaldemar | pisg: DAHDI/g0/06681406XX is most likely your problem |
| 10:30.36 | kaldemar | pastebin your extension |
| 10:33.01 | pisg | kaldemar: http://pastebin.com/0SLmJiaV |
| 10:33.56 | kaldemar | if those are real you're dialing 06681406XX with your phone. dial a number. |
| 10:34.19 | OlafsenM | hi |
| 10:34.21 | OlafsenM | im originating calls through AMI, but i dont get any CDR records |
| 10:34.26 | OlafsenM | <PROTECTED> |
| 10:34.32 | OlafsenM | <PROTECTED> |
| 10:34.33 | OlafsenM | ? |
| 10:34.43 | tuxx- | Is there a way i can let a phone reload its sip subscriptions? I have an aastra 57i with extensionpad atm. And somehow if someone logs out of their phone (got hotdesking) it loses its sip subscription for that BLF led. |
| 10:34.46 | pisg | when i`m use X-lite i can dialing, when i`m use user/pass in SIP phone i see this |
| 10:34.49 | OlafsenM | in fact, i do have records in cdr-csv |
| 10:34.59 | OlafsenM | but i dont get any CDR manager events |
| 10:35.06 | pisg | kaldemar: Everyone is busy/congested at this time (1:0/0/1) |
| 10:36.05 | kaldemar | pisg: show the whole cli output for a call |
| 10:37.18 | pisg | kaldemar: http://pastebin.com/LbggFNzw only console i can see this |
| 10:37.37 | pisg | in asterisk -rvvv |
| 10:38.36 | pisg | but when i`m use X-lite i can calling |
| 10:38.41 | pisg | in SIP phone not ;/ |
| 10:38.43 | kaldemar | you should see more. "core set verbose 10" in cli and try again. |
| 10:40.56 | pisg | kaldemar: hmm CLi see me this same |
| 10:41.33 | kaldemar | there's no way that's ALL the output for a call. |
| 10:44.51 | pisg | i change core set verbose to 10 and see this same not more options |
| 10:45.47 | pisg | soory more i can see this : > Saved useragent "SIP201 (lp201sip.101b)" for peer konradnowak |
| 10:47.53 | tuxx- | Is there a way i can let a phone reload its sip subscriptions? I have an aastra 57i with extensionpad atm. And somehow if someone logs out of their phone (got hotdesking) it loses its sip subscription for that BLF led. \ |
| 10:48.31 | kaldemar | pisg: well, don't dial literal X's with your phone. |
| 10:54.58 | pisg | kaldemar: so i must change this in extensions.conf ? |
| 10:55.29 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
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| 10:59.32 | kaldemar | pisg: no, unless your configs are fake. |
| 11:00.12 | pisg | config in hardware SIP phone ? |
| 11:00.22 | pisg | or config asterisk |
| 11:00.33 | Icheb | Would anyone here perhaps know when a new version of the 1.6.2 (or 1.6.3) branch is planned as bugfix release? - I'm desperately searching for a new version with issue 16729 fixed ;) |
| 11:01.16 | *** join/#asterisk pif (~ldm@zenon.apartia.fr) |
| 11:06.03 | kaldemar | pisg: you're not showing me what happens when you make a call so i don't know. |
| 11:08.16 | tzafrir_laptop | Iamnacho, there won't be a branch 1.6.3 . Next release beanch will be 1.8 |
| 11:08.30 | pisg | kaldemar: http://img683.imageshack.us/img683/4805/sipso.jpg this is my SIP phone setting and when i call i console i see : http://pastebin.com/LbggFNzw |
| 11:08.32 | tzafrir_laptop | In branch 1.6.2, 1.6.2.6 was recently released |
| 11:09.08 | tzafrir_laptop | Icheb, why not grab latest release and apply the patch yourself? |
| 11:10.13 | Chainsaw | 1.6.2.6 isn't in the topic though. |
| 11:10.23 | Chainsaw | Is that a security release or a proper release? |
| 11:10.56 | J4zen | pisg: enable sip debugging, set verbosity to 20, place a call and pb all output |
| 11:13.17 | Icheb | tzafrir_laptop, I don't like running non official versions on production systems |
| 11:14.06 | tzafrir_laptop | Chainsaw, somebody forgot to update the topic, then |
| 11:15.02 | Icheb | And due to the fact no svn rev is mentioned, nor any files are mention, it would probably mean having to run the trunk version, don't know if that's stable enough |
| 11:16.25 | Icheb | okay, that's a diff for the other one, might try that on my dev environment, but still, I believe this to be quite important for a lot of people, so that's why I'm curious for a roadmap to this being included |
| 11:19.43 | pisg | J4zen: http://pastebin.com/1y8X4E1M |
| 11:23.15 | *** join/#asterisk waa (~waa@balrog.credipar.com.br) |
| 11:28.05 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 11:31.17 | *** join/#asterisk Quasar-1922 (~quasar@mail.kdj.nl) |
| 11:33.25 | J4zen | pisg: It states your problem right in the debug: Unable to create channel of type 'DAHDI' (cause 58 - Bearer capability not available) |
| 11:33.31 | J4zen | probably due to : No translator path exists for channel type DAHDI (native 0x4c) to 0x100 |
| 11:34.15 | J4zen | sounds like a codec issue of some sorts, im not sure. |
| 11:34.40 | pisg | J4zen: i have this problem when unisg Pentragram lp-201 Sip phone, but when use X-lite all ist fine |
| 11:34.49 | kaldemar | the phone is trying to use G.729 codec, but asterisk doesn't have one. |
| 11:34.56 | kaldemar | pisg: ^^ |
| 11:35.01 | pisg | :O |
| 11:35.12 | kaldemar | change the codec in the phone configuration. |
| 11:35.25 | J4zen | there you go :) |
| 11:39.53 | *** join/#asterisk af_ (~getsmart@88-149-240-251.dynamic.ngi.it) |
| 11:47.21 | pisg | kaldemar J4zen, now working, thx, big VODKA when you come to poland |
| 11:53.35 | J4zen | pisg: your welcome |
| 11:53.38 | J4zen | you're* |
| 11:53.44 | atis_work | pisg: vodka?? no zubrowka? |
| 11:55.25 | *** join/#asterisk creativx (~creadurex@197.82-134-19.bkkb.no) |
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| 12:02.12 | pisg | atis_work: i dont like zubrowka |
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| 12:59.44 | Quasar-1922 | Hey all... Does anyone here have experience with the redfone fonebridge2 boxes? |
| 13:02.30 | *** join/#asterisk coppice (~chatzilla@59.192.17.210.dyn.pacific.net.hk) |
| 13:09.16 | *** join/#asterisk asteriskATmarmuD (~mundt@193.158.65.23) |
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| 13:12.51 | Katty | morning |
| 13:13.57 | rttrey | Good morning. |
| 13:14.59 | rttrey | Have a quick question. The daylight savings time change didnt carry over to my phones. The time on the server is correct using the 'date' command. Any ideas real quick? |
| 13:15.31 | Katty | polycoms don't use your server |
| 13:15.41 | Katty | they use whatever snmp ip address you feed them |
| 13:16.15 | Katty | they also have settings on when to make the switch over. |
| 13:16.27 | [TK]D-Fender | rttrey: Devices need to know what their time zone is. |
| 13:16.30 | Katty | you can read all about it with the guide that comes with them |
| 13:16.44 | [TK]D-Fender | rttrey: and North America's rules changed 2 years ago |
| 13:17.24 | rttrey | Yes they are polycoms... thank you |
| 13:18.41 | *** join/#asterisk Faithful (~Faithful@202.6.145.116) |
| 13:19.06 | *** join/#asterisk TheDavidFactor (~chatzilla@nc-71-0-16-133.dhcp.embarqhsd.net) |
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| 13:24.44 | *** part/#asterisk muiro (~muiro@unaffiliated/muiro) |
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| 13:29.09 | *** join/#asterisk andres833 (~andres833@190.144.75.22) |
| 13:32.38 | asteriskATmarmuD | using vicidial i get the following message when an angent logs into the GUI: "no channel type registered for 'Zap'" |
| 13:34.06 | asteriskATmarmuD | I want to use DAHDI, vicidial people told me DAHDI and Zap are used synonymously in the backend |
| 13:34.38 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 13:39.10 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 13:40.17 | [TK]D-Fender | asteriskATmarmuD: There is a flag that lets you use the ZAP channel name type with DAHDI, but you have to configure it to do so. Go read the samples to find the precise name for it |
| 13:42.43 | *** join/#asterisk csiadmin (~csiadmin@81.144.152.52) |
| 13:47.30 | *** join/#asterisk moy (~chatzilla@74.12.134.118) |
| 13:49.17 | *** join/#asterisk Pitel (~pitel@ip-89-103-6-29.karneval.cz) |
| 13:53.10 | *** join/#asterisk miloux (~KVIrc@milu.rit.se) |
| 13:54.05 | Katty | wooooooo what a morning |
| 13:54.13 | asteriskATmarmuD | [TK]D-Fender: thx a lot... any hint on where to find that flag? I've been searching the internet for days now... |
| 13:56.19 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:56.19 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:58.00 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 14:00.54 | [TK]D-Fender | asteriskATmarmuD: In the SAMPLE CONFIGS like I told you |
| 14:01.18 | Quasar-1922 | Hey I've setup a IAX trunk between two boxes and if I call from one to another I get: |
| 14:01.18 | Quasar-1922 | [14:27:32] WARNING[5084]: channel.c:700 ast_best_codec: Don't know any of 0xe000 formats |
| 14:01.18 | Quasar-1922 | [14:27:32] ERROR[5084]: chan_iax2.c:7678 socket_process: No best format in 0xe000??? |
| 14:01.18 | Quasar-1922 | [14:27:32] NOTICE[5084]: chan_iax2.c:7685 socket_process: Rejected connect attempt from 192.168.78.3, requested/capability |
| 14:01.36 | *** join/#asterisk shader (~user@janustw.tavve.com) |
| 14:01.48 | Katty | >.< |
| 14:01.51 | Katty | i keep beeping |
| 14:02.47 | kaldemar | Quasar-1922: you need to allow some codecs. |
| 14:03.06 | Quasar-1922 | I did... |
| 14:03.11 | Quasar-1922 | I allowed ulaw |
| 14:03.14 | Quasar-1922 | sorry |
| 14:03.15 | shader | is this an acceptable channel to ask questions regarding hardware accessories to an asterisk based phone system? or is there some other channel I should visit? |
| 14:03.21 | Quasar-1922 | alaw,ulaw,g729 in that order |
| 14:03.47 | kaldemar | Quasar-1922: show it |
| 14:03.57 | Quasar-1922 | and then it uses g729 but i find the quality not good enough so if I leave out g729 to use Alaw i get an error. the one i pasted above |
| 14:05.12 | csiadmin | hi everyone, I'm using monitor to record incoming calls but they're being saved as in/out files. I've sox installed, set with MONITOR_EXEC, monitor flagged with 'm' to mix - any other suggestions for what else I could check? |
| 14:05.12 | *** join/#asterisk rgsteele (~rgsteele@207.106.239.81) |
| 14:05.15 | Quasar-1922 | iax.conf: allow=alaw,ulaw,g729 |
| 14:05.16 | Quasar-1922 | sip.conf: allow=alaw,ulaw,g729 |
| 14:05.40 | kaldemar | Quasar-1922: show all the configuration for the peers |
| 14:05.44 | kaldemar | ~pb |
| 14:05.45 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 14:06.05 | *** join/#asterisk ManxPower-work (~manxpower@234.sub-75-254-56.myvzw.com) |
| 14:06.32 | leifmadsen | Katty: beep beep |
| 14:06.48 | Katty | >.< |
| 14:06.52 | Katty | WHY I OUTTA JUST |
| 14:06.54 | Katty | HUG YOU! |
| 14:06.57 | Katty | hugs leifmadsen |
| 14:07.29 | Katty | frowns |
| 14:07.33 | *** join/#asterisk Dibri (~gavit@pop1.isgroup.sr) |
| 14:07.34 | Katty | squirrely on my new feeder. |
| 14:07.48 | *** join/#asterisk dmz (~dmz@64.203.207.101.dyn-cm-pool-54.hargray.net) |
| 14:07.56 | Katty | amazing how they can climb a sheer 5ft pole |
| 14:08.23 | Kobaz | it's not exactly sheer though |
| 14:08.36 | Katty | erhmm. |
| 14:08.42 | Katty | fine. vertical. |
| 14:08.43 | Kobaz | there's fine indentations and ridges in the wood that they can grab |
| 14:08.44 | Quasar-1922 | http://pastebin.com/HAZhDC36 |
| 14:08.44 | Kobaz | heh |
| 14:08.48 | ManxPower-work | Don't you hate it when you basically (nicely) tell someone they are an idiot, and they are so clueless they actually thank you? |
| 14:08.54 | Katty | Kobaz: the pole isn't wood. |
| 14:08.59 | Katty | Kobaz: but it could be textured. |
| 14:09.06 | Kobaz | mm |
| 14:09.10 | ManxPower-work | Pole Dancing? |
| 14:09.16 | Katty | climbing. |
| 14:09.35 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 14:09.53 | Katty | infobot: crittercam |
| 14:09.54 | infobot | crittercam is probably Katty's live broadcast of The Nut House @ http://ustre.am/8H5d and The Fuzzy Ferret Flat @ http://ustre.am/bEBU |
| 14:09.57 | *** join/#asterisk creativx (~creadurex@197.82-134-19.bkkb.no) |
| 14:10.05 | kaldemar | Quasar-1922: you have two boxes, so there are two relevant configurations |
| 14:10.27 | Katty | now he's eating the woodpecker suet :< |
| 14:10.27 | shader | can anyone recommend a usb handset for a linux softphone? |
| 14:10.38 | Katty | shader: handset? |
| 14:10.53 | shader | desk phone |
| 14:10.59 | Quasar-1922 | i know but both are the same |
| 14:11.00 | Kobaz | yeah it's a phone without the phone part, and it's usb |
| 14:11.14 | Quasar-1922 | just the other way round ;-) |
| 14:11.15 | Katty | i didn't know they made one |
| 14:11.20 | Kobaz | shader: just get a polycom |
| 14:11.25 | Katty | which would be very interesting |
| 14:11.31 | Katty | because then i could use network vpn stuffs |
| 14:11.40 | Katty | theoretically |
| 14:11.45 | shader | Kobaz: any particular model? |
| 14:11.53 | Katty | polycoms aren't usb. |
| 14:11.56 | Katty | so it's not what he wants. |
| 14:12.18 | Kobaz | shader: 331's are nice... they have two ethernet ports |
| 14:12.44 | kaldemar | Quasar-1922: what codec are you using on the phone? |
| 14:12.54 | Kobaz | shader: good luck getting any usb phone working in linux |
| 14:13.11 | Quasar-1922 | uhm, alaw since that has preference... it's a polycom 430 |
| 14:13.17 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 14:13.40 | Katty | shader: http://www.telephonydepot.com/Catalog/Polycom-Phones/Polycom-Soundpoint-IP-331 |
| 14:13.42 | Quasar-1922 | it's both asterisk 1.4.21.2 btw |
| 14:14.39 | ManxPower-work | the problem with the 3xx Polycoms is that the screen is so small (2-lines) that all the REALLY cool stuff (microbrowser, etc) just won't work well. |
| 14:14.43 | asteriskATmarmuD | [TK]D-Fender: ok, thx. found something useful in asterisk.conf "dahdichanname=no" ... now my analog phones are useless again... wish me luck... bye |
| 14:14.59 | *** join/#asterisk Squeeb (~Debian-ex@host81-149-117-179.in-addr.btopenworld.com) |
| 14:15.01 | Katty | ManxPower-work: it's good for a basic phone without the frilly stuffs. |
| 14:15.12 | Katty | ManxPower-work: but i do agree it's difficult to have multiple calls waiting for you. |
| 14:15.20 | ManxPower-work | Katty, so are many of the Linksys phones |
| 14:15.27 | Katty | ManxPower-work: so a terrible receiptionist phone, or for anyone who deals with more than call at a time |
| 14:15.40 | Squeeb | Hi, I'm looking for some decent documentation about queues.conf, detailed information regarding the options you can use in both the global section and per-queue sections |
| 14:16.03 | kaldemar | Quasar-1922: use disallow=all before the allow lines and don't allow codecs you don't have (=g729). if this doesn't help, enable iax debug and pastebin the cli output for a failed call. |
| 14:16.15 | ManxPower-work | Squeeb, the info in UPGRADE*.txt and the stuff on doc/ (i'm sure there's a queues document in there) |
| 14:16.20 | Squeeb | hmm |
| 14:16.21 | Squeeb | ok |
| 14:16.42 | Squeeb | Why isn't this stuff on the documentation / support sections of the digium / asterisk website though? |
| 14:16.46 | Squeeb | :/ |
| 14:16.55 | ManxPower-work | UPGRADE*.txt for you would mostly so you know what older Asterisk commands map to which newer commands for when you are reading out of date docs. |
| 14:17.13 | Quasar-1922 | Kaldemar.. ok.. I do have g729 .. This works.. Just when I leave out g729 it goes wrong.. I'll fetch the iax debug one moment |
| 14:17.18 | ManxPower-work | Squeeb, um, because the official place for official asterisk docs is the source directory? |
| 14:17.19 | *** join/#asterisk FirstSgt (~cheney@host2.complimentsinternational.com) |
| 14:17.32 | Katty | Squeeb: why don't you call them and bitch about it. |
| 14:17.34 | ManxPower-work | Quasar-1922, how many G729 licenses do you have? |
| 14:17.35 | Squeeb | Lol |
| 14:17.38 | Katty | Squeeb: i'm sure they would LOVE hearing from you. |
| 14:17.39 | FirstSgt | Anyone kknow of a good linux softphone (SIP Phone)? |
| 14:17.46 | Squeeb | Katty: everybody loves hearing from me |
| 14:17.47 | ManxPower-work | All softphones suck |
| 14:18.17 | FirstSgt | ManxPower-work: ok, then lemme rephraze |
| 14:18.26 | Katty | FirstSgt: i like zoiper. |
| 14:18.36 | FirstSgt | cool... |
| 14:18.38 | FirstSgt | i'll look |
| 14:18.43 | FirstSgt | twinkle wouldn't compile for me |
| 14:18.49 | ManxPower-work | No matter how you phrase it, they still suck. |
| 14:19.01 | Katty | gives ManxPower-work coffee |
| 14:19.23 | Katty | no more talking until you've finished 1 cup of coffee. |
| 14:19.30 | FirstSgt | ManxPower-work: I wasn't asking about their level of suckage, so I believe you missunderstood my question. which why I was rephrasing it. |
| 14:19.48 | [TK]D-Fender | FirstSgt: Ekiga |
| 14:19.57 | FirstSgt | ManxPower-work: Does anyone know the top 3 linux softphones? |
| 14:19.58 | [TK]D-Fender | ~ekiga |
| 14:19.58 | infobot | [~ekiga] Ekiga is an OSS SIP soft-phone for Windows, MacOSX, and Linux (Requires GTK+ which is included with the binary releases) that can be found at http://www.ekiga.org |
| 14:20.02 | FirstSgt | [TK]D-Fender: thanks |
| 14:20.11 | FirstSgt | Katty: Thank you too... I will try both |
| 14:20.35 | FirstSgt | I think I have GKT+ libs |
| 14:21.57 | *** join/#asterisk ccesario (~ccesario@189-19-6-236.dsl.telesp.net.br) |
| 14:24.27 | Quasar-1922 | hey kaldemar... i've tried but the weird thing is.. from one box to another works fine all the time but the other way round works in maybe 10% of all cases |
| 14:24.34 | Quasar-1922 | i''ve pasted the iax2 debug |
| 14:24.34 | Quasar-1922 | http://pastebin.com/gH9tE86w |
| 14:24.41 | *** join/#asterisk Akiraaa (~Akiraaaa@79.112.26.154) |
| 14:26.34 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 14:28.07 | *** join/#asterisk benngard (~benngard@90-230-92-67-no148.tbcn.telia.com) |
| 14:29.41 | kaldemar | Quasar-1922: you're getting cause 17, which should mean busy. show also the verbose cli output with the iax debug. |
| 14:30.13 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 14:31.19 | Quasar-1922 | kaldemar.. that's a little hard since we have 200 phones connected to it making calls right now ;-) |
| 14:31.31 | Quasar-1922 | unless there's a way to show only my sip extension? |
| 14:32.09 | kaldemar | no. that's a parsing job then. |
| 14:35.02 | *** join/#asterisk Ad-Hoc (~nimbus@62.169.216.185) |
| 14:36.00 | [TK]D-Fender | Quasar-1922: This is an IAX2 call. What does it have to do with SIP? <---- |
| 14:40.00 | *** join/#asterisk undertuga (~undertuga@213-205-80-134.net.novis.pt) |
| 14:40.06 | undertuga | Hi there! |
| 14:40.32 | Katty | ohai |
| 14:40.36 | Katty | did you bring me breakfast |
| 14:41.26 | andrebarbosa | Hi |
| 14:41.37 | andrebarbosa | anyway knows what bug this note is talking about: https://support.counterpath.com/default.asp?W324 |
| 14:41.42 | andrebarbosa | ? |
| 14:41.48 | andrebarbosa | anyone* |
| 14:42.29 | undertuga | Just started messing with asterisk (using asteriskNow distro), and i'm having some trouble while trying to connect some SIP clients to it! I had already created some extensions, but the client gives me 408 timeout while trying to connect. I heard about setting up bindport to 5060, since asterisk is not bindng itself to it. Can someone help me out on here do i change that property? Thanks in advance! |
| 14:42.54 | *** join/#asterisk joesuffceren (~chatzilla@ip68-104-167-226.ph.ph.cox.net) |
| 14:43.03 | Katty | infobot: asterisknow |
| 14:43.04 | infobot | asterisknow is probably based on Asterisk, but is difficult to support in #asterisk for a number of reasons. Please seek support in #asterisknow instead. |
| 14:43.21 | undertuga | roger that! |
| 14:43.24 | undertuga | thanks! |
| 14:44.54 | KavanS | to switch from zaptel to dahdi I need to replace "Zap" with what in the config file? |
| 14:45.15 | Katty | erm |
| 14:45.16 | Katty | dahdi |
| 14:45.32 | Katty | checks KavanS's caffeine levels |
| 14:46.06 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
| 14:46.17 | [TK]D-Fender | KavanS: Any play that would use it |
| 14:46.24 | Katty | http://i.imgur.com/Tnwxk.jpg |
| 14:46.29 | [TK]D-Fender | KavanS: And my guess would be... DAHDI <- |
| 14:46.34 | KavanS | ok |
| 14:46.41 | Katty | ^- me, but not me. |
| 14:46.57 | joesuffceren | having some trouble with a te122. I'm running asterisk 1.4.29.1, dahdi 2.2.1, libpri 1.4.10.2. I'm getting random reboots and hangs, and lots of errors in my server's event log related to the te122 and the pci-e slot it's in. I do notice that it seems to be sharing an interrupt with the USB controller which I know is a bad thing. Running dahdi_test for 5 minutes, though, produced no results... |
| 14:46.59 | joesuffceren | ...worse than 99.993%, which I think indicates that's not a problem. Any other thoughts? |
| 14:48.48 | Katty | what is a te122 |
| 14:49.02 | Naikrovek | te121 +1 |
| 14:49.13 | Katty | and what's that |
| 14:49.31 | Quasar-1922 | <[TK]D-Fender> -> since my phones use sip.... |
| 14:50.03 | joesuffceren | sorry. typo. it's a te121 |
| 14:50.05 | [TK]D-Fender | Quasar-1922: You are showing us an IAX2 error. that has precisely nothing to do with SIP |
| 14:50.13 | *** join/#asterisk garymc (~chatzilla@host81-148-109-86.in-addr.btopenworld.com) |
| 14:50.24 | Katty | [TK]D-Fender: oh |
| 14:50.30 | Katty | [TK]D-Fender: i must show you my latest photo |
| 14:50.32 | [TK]D-Fender | Quasar-1922: Next you'll associate the cupholder in your car to your transmission problems... |
| 14:50.32 | spenguin[work] | pokes Katty --E |
| 14:50.38 | *** join/#asterisk bsaxon (~bsaxon@12.68.234.174) |
| 14:50.44 | Katty | squeaks |
| 14:50.52 | Quasar-1922 | it does if codecs for iax2 and sip are different which causes asterisk to transcode the conversation |
| 14:50.59 | Katty | spenguin[work]: at first, i thought that was a flag. |
| 14:51.06 | spenguin[work] | heh |
| 14:51.11 | Katty | spenguin[work]: and thought i needed to consult my readme file for additional flag information |
| 14:51.27 | spenguin[work] | haha |
| 14:52.51 | spenguin[work] | trident, damn |
| 14:52.52 | [TK]D-Fender | Quasar-1922: We'd see that in the SIP debug.... |
| 14:52.56 | spenguin[work] | Ive been thinking :p |
| 14:53.03 | spenguin[work] | I knew the hindi word |
| 14:53.51 | Katty | [TK]D-Fender: http://farm1.static.flickr.com/67/203584249_a9fb9cbb87_b.jpg |
| 14:54.16 | *** join/#asterisk jpeeler (~jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:54.25 | Quasar-1922 | <[TK]D-Fender> hmm. yeah.. but you're right, it doen't seem to be sip related i guess since alaw works fine on both boxes.. |
| 14:54.31 | Katty | [TK]D-Fender: and |
| 14:54.52 | [TK]D-Fender | Quasar-1922: .... You have not shown a reasonably complete sample to date. |
| 14:55.05 | Katty | [TK]D-Fender: http://farm3.static.flickr.com/2647/3757499110_9456d2bea1_o.jpg |
| 14:55.14 | [TK]D-Fender | Quasar-1922: And introduce unvalidate elements as you go |
| 14:55.51 | [TK]D-Fender | Katty: nice... |
| 14:56.16 | joesuffceren | katty: te121 is a digium T1/PRI card |
| 14:56.22 | Katty | joesuffceren: ah. k |
| 14:56.31 | Katty | joesuffceren: i mostly work with sangoma stuff (= |
| 14:56.41 | *** join/#asterisk leo66 (~chatzilla@187-032-044-130.static.ctbctelecom.com.br) |
| 14:56.52 | Quasar-1922 | does anyone here have experience with redfone fonebridge2 boxes by the way? |
| 14:57.45 | Quasar-1922 | i was looking for an alternative for our sangoma's here since we have 2 pbx-es with a104d's and we need backup boxes for both.. get's a little too expensive and complicated i guess putting in two backup sangoma's for just backup box.. |
| 14:58.59 | [TK]D-Fender | Quasar-1922: Avoid. Hear little but trouble about them in here |
| 14:59.49 | *** join/#asterisk AlHafoudh (~alhafoudh@77.93.192.244) |
| 14:59.57 | AlHafoudh | hi all |
| 15:00.17 | Katty | anyone good with landscaping |
| 15:00.36 | Kobaz | scapes some land |
| 15:00.42 | coppice | yeah. Capability Brown |
| 15:00.58 | Katty | i think landscaping was a poor choice of words. |
| 15:01.08 | Katty | is anyone familiar with planting blueberry bushes |
| 15:01.08 | Quasar-1922 | <[TK]D-Fender> hmm, ok thanks. do you have another good suggestion for external hardware able to do failover? |
| 15:01.14 | Naikrovek | Katty: pubic topiary? |
| 15:01.17 | *** join/#asterisk asteriskmonkey (~philip@69.77.169.14) |
| 15:01.23 | Katty | Naikrovek: that statement does not parse. |
| 15:01.26 | Katty | Naikrovek: please try again. |
| 15:01.41 | Katty | i will draw a pretty picture. |
| 15:01.43 | Kobaz | syntax error near 'public' |
| 15:01.45 | Katty | maybe that will help. |
| 15:01.58 | [TK]D-Fender | Quasar-1922: AudioCodes Mediant series |
| 15:01.59 | Kobaz | topiary - Of or pertaining to ornamental gardening; produced by cutting, trimming, etc.; topiarian. |
| 15:02.37 | Quasar-1922 | Fender, I'll have a look, thanks. |
| 15:03.01 | Kobaz | Katty: i think you kind of dig a hole and push the bush in... make sure it gets plenty of water... and replace the topsoil from the hole with good stuff from a nursury |
| 15:03.53 | spenguin[work] | has a planted fish tank |
| 15:04.02 | Kobaz | Katty: you'll need to soak it thoroughly after planting, and keep it well watered for the first week or two |
| 15:04.02 | spenguin[work] | Im still learning scaping it |
| 15:04.22 | Katty | Kobaz: how about sun requirements? |
| 15:04.32 | Kobaz | Katty: they don't like shade |
| 15:05.01 | Katty | Kobaz: and how well do you think it would do in a huge planter, rather than in the ground |
| 15:05.02 | coppice | Katty: if you are planting you need plenty of manure. the marketing dept should be able to provide you with some |
| 15:05.14 | Kobaz | Katty: unless you love watering... not very well |
| 15:05.19 | spenguin[work] | not plenty |
| 15:05.27 | spenguin[work] | you could burn the plant out and kill it |
| 15:05.46 | AlHafoudh | by default, SIP sends password in clear text or is it secure? |
| 15:06.08 | Kobaz | AlHafoudh: generally it's an md5 challenge based auth |
| 15:06.21 | Katty | Kobaz: wouldn't it retain more water by being in a planter? |
| 15:06.25 | Kobaz | Katty: no |
| 15:06.35 | Katty | Kobaz: the planter doesn't have holes. |
| 15:06.35 | Kobaz | Katty: it will dry out much faster |
| 15:06.39 | Katty | Kobaz: i'm not following your logic |
| 15:06.40 | Kobaz | Katty: that's even worse |
| 15:06.50 | Kobaz | then the plant will drown and die |
| 15:06.54 | asteriskmonkey | AlHafoudh: clear |
| 15:07.10 | Katty | hmmmm |
| 15:07.16 | Katty | k |
| 15:07.21 | Katty | uploads photo |
| 15:07.22 | *** join/#asterisk moos3 (~rgenthner@216.52.121.66) |
| 15:07.29 | leo66 | Hello all... I'm trying to use chan_mobile to call from asterisk to outside phones, but i cant get audio working. I can dial from my extension to outside using bluetooth but i dont hear any sound when the call is answered.. can someone help me? |
| 15:07.54 | Kobaz | it's going to get more airflow in a planter since it's higher off the ground. and now it no longer has access to the moisture from the ground, so it's now solely dependend on consistant rain and/or watering |
| 15:08.35 | Kobaz | Katty: and if you have no holes in the bottom, and you dump a gallon of water into it... then the opposite will happen... it's going to take forever to dry out, the roots will rot, and it'll croak |
| 15:09.12 | [TK]D-Fender | LeDoes it work both ways jsut within *? |
| 15:09.14 | Katty | Kobaz: http://imagebin.org/88941 |
| 15:09.14 | Kobaz | Katty: all non-marsh plants need a cycle of wet/dry for proper balance |
| 15:09.29 | Katty | Kobaz: black is the house, grey the decks, and the brown is the mulch area |
| 15:09.43 | Katty | Kobaz: blue is obviously the proposed area for the blueberry bush, to accomodate for proper sunlight. |
| 15:09.53 | Kobaz | looks good |
| 15:09.57 | Kobaz | how much sun does that spot yet? |
| 15:09.58 | Kobaz | get |
| 15:10.05 | Quasar-1922 | here's some more info: http://pastebin.com/Lb0RwuNt |
| 15:10.06 | Katty | Kobaz: i'm also guessing that putting a blueberry bush in the ground, with potting soil, and covering it mulch is probablyh a bad idea |
| 15:10.06 | Kobaz | and what direction is the sun coming from |
| 15:10.15 | Katty | Kobaz: sun comes from 'behind' the house. |
| 15:10.24 | [TK]D-Fender | leo66: Does it work both ways jsut within *? |
| 15:10.27 | Kobaz | as in the top? |
| 15:10.38 | Katty | Kobaz: the bush's view of the sun will be obstructed till probably 9 or 10 in the morning, but full sun after that |
| 15:10.43 | Katty | Kobaz: yes the 'top' of the photo |
| 15:10.55 | Kobaz | what sort of siding do you have? is it reflective/white ? |
| 15:11.02 | Katty | it's white. |
| 15:11.04 | Katty | gets a photo |
| 15:11.15 | spenguin[work] | Katty: you just bought the plant? |
| 15:11.29 | spenguin[work] | Id normally keep it in the pot, untill it gets a lil bigger |
| 15:11.42 | Kobaz | that's going to bounce extra heat/light against the bush... if there is high sun intensity, it will make it higher... and you may get leaf burn |
| 15:11.54 | [TK]D-Fender | Quasar-1922: allow=alaw,ulaw <-- split this |
| 15:11.58 | Kobaz | depends how much sun you get |
| 15:12.07 | *** join/#asterisk Corydon76-lap (~Corydon76@nat/digium/x-buvlvswpkrnpeoeu) |
| 15:12.07 | *** mode/#asterisk [+o Corydon76-lap] by ChanServ |
| 15:12.09 | Katty | Kobaz: http://farm4.static.flickr.com/3188/3021635539_237bf0f45f_o.jpg |
| 15:12.28 | Katty | Kobaz: that photo was taken late in the evening |
| 15:12.41 | Kobaz | oh okay, partial shade |
| 15:12.43 | Kobaz | you should be fine |
| 15:12.45 | Katty | Kobaz: the corner of the house you're looking at is the corner i'd plant the bush on |
| 15:13.03 | Kobaz | so the back of the house is south-facing |
| 15:13.04 | Katty | spenguin[work]: i haven't bought the bush yet, but it's not in a planter...it's wrapped in plastic |
| 15:13.13 | Katty | Kobaz: not quite south |
| 15:13.19 | Katty | Kobaz: more like south east |
| 15:13.31 | Kobaz | yeah that's fine |
| 15:13.34 | Katty | Kobaz: close enough |
| 15:13.43 | Katty | Kobaz: where would you suggest i put the strawberry planter? |
| 15:13.58 | Kobaz | at my parents house there's a sun room in the back, the house is completely south facing... and we tried tomato plants behind the house, they got cooked |
| 15:14.08 | Kobaz | the sunroom reflects so much heat it's insane |
| 15:14.18 | spenguin[work] | too much sun is bad for these plants |
| 15:14.23 | *** join/#asterisk babbio (~matteo@host-78-13-24-238.cust-adsl.tiscali.it) |
| 15:14.24 | [TK]D-Fender | Quasar-1922: I also don't see matching peer names oin there which make it look un-authed which leaves you with [general] for your matching |
| 15:14.25 | *** join/#asterisk sourcode (~code@ppp-115-87-200-93.revip4.asianet.co.th) |
| 15:14.30 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:14.38 | Kobaz | Katty: stawberry is a vine... where would you like a vine? |
| 15:14.56 | *** join/#asterisk sourcode (~code@ppp-115-87-200-93.revip4.asianet.co.th) |
| 15:15.11 | *** join/#asterisk hluesea (~hulusikah@88.247.127.66) |
| 15:15.30 | babbio | hi guys i'm new on asterisk....i'm trying to use trixbox but after installing....when i try to make a call i have an error "trixbox kernel: FXO PCI Master Abort" what should i do? |
| 15:15.33 | Kobaz | most people don't have ample space to properly grow strawberries |
| 15:15.34 | babbio | thanks |
| 15:15.37 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 15:15.43 | ManxPower-work | ~trixbox |
| 15:15.44 | infobot | trixbox is probably SH1TB0X. Basically a CRAPPY, closed source distro. STAY AWAY! |
| 15:15.57 | ManxPower-work | ~freebpx |
| 15:17.00 | babbio | please i would like to learn using asterisk....i tried trixbox for the beginning because i read it is more easy but i'm having this error...please help me |
| 15:17.19 | Qwell | babbio: We can't help you with that here. |
| 15:17.23 | Qwell | Go ask them |
| 15:17.27 | [TK]D-Fender | babbio: You have a PCI issue with their distro. Go take it up ther. THis isn't an ASTERISK problem |
| 15:17.30 | Kobaz | babbio: it's going to be very difficult to help you, since trixbox puts 39284792387423 lines of code on top of asterisk |
| 15:17.41 | Qwell | Kobaz: actually they don't |
| 15:17.50 | Kobaz | they have their gooey |
| 15:18.08 | Kobaz | it's probably not that many lines, but it's not vanilla |
| 15:18.11 | leo66 | [TK]D-Fender yes |
| 15:18.36 | [TK]D-Fender | leo66: then your problem is the other leg of the call to * and has nothing to do with the first |
| 15:19.01 | Kobaz | Katty: bonsai is much easier... i like my cacti |
| 15:19.25 | babbio | you think i could use asterisknow or is it too hard for the beginning? |
| 15:20.09 | Kobaz | Katty: lemme see i have a picture of the bonsai |
| 15:20.30 | *** join/#asterisk ccesario (~ccesario@189-19-6-236.dsl.telesp.net.br) |
| 15:20.44 | bmoraca_work | bonsai kitty? |
| 15:21.28 | Kobaz | no like bonsai bonsai |
| 15:21.32 | Kobaz | little trees |
| 15:21.39 | leo66 | [TK]D-Fender voip, sip, pstn calls are ok. Do you think the problem is bluetooth configuration? |
| 15:21.51 | bmoraca_work | i know, i was making a joke: http://m.blog.hu/zi/zizyandfanny/image/bonsai_kitten.png |
| 15:22.15 | *** join/#asterisk Netgeeks (~chris@173-11-68-157-SFBA.hfc.comcastbusiness.net) |
| 15:22.16 | bmoraca_work | this is a better picture: http://www.hizone.info/data/2003/07/09/images/cat_in_a_bottle.jpg |
| 15:22.32 | Kobaz | hah |
| 15:23.47 | petern_ | that's not a bottle |
| 15:23.58 | seanbright | details details |
| 15:24.03 | bmoraca_work | i know, but i didn't name the picture...so...what do you want? |
| 15:24.35 | spenguin[work] | Kobaz: bonsai takes aaaages |
| 15:24.42 | Kobaz | it does |
| 15:24.46 | spenguin[work] | Id want a bonsai mango plant |
| 15:25.06 | spenguin[work] | can it be tried on quicker growing plants like say the papaya? |
| 15:25.27 | Kobaz | bonsai works the best with trees/shrubs that have small leaves |
| 15:25.46 | spenguin[work] | hrm |
| 15:25.47 | Kobaz | ficus works really well, and you can get them to do some cool stuff, like air roots |
| 15:26.09 | spenguin[work] | a banyan bonsai would be cool too |
| 15:26.39 | moos3 | can anyone tell me why my context listen4extn only gets one digit ? http://pastie.org/870416 |
| 15:26.41 | leo66 | [TK]D-Fender: everything is well configurated. I have bluetooth working on * 1.4, but i cant get audio in/out on my new * 1.6 install. |
| 15:27.04 | Kobaz | i would get my bonsai trees from my parents house if i had a good spot inside to grow them |
| 15:27.17 | Kobaz | i only have like one window that gets direct sun for like 2 hours out of the day |
| 15:28.43 | *** join/#asterisk CoderForLife (~Miranda@cpe-174-101-155-51.cinci.res.rr.com) |
| 15:29.04 | [TK]D-Fender | leo66: You just confirmed that BT to/from * is bidirectional. Therefor the problem is the other leg of the call |
| 15:29.08 | Corydon76-lap | like totally? |
| 15:29.18 | Kobaz | hah yeah |
| 15:29.20 | Kobaz | sorry :P |
| 15:29.31 | Kobaz | i have a ficus kinda like this one http://web.mawebcenters.com/hollowcreekbonsai/images/9F49.jpg |
| 15:30.16 | spenguin[work] | Kobaz: http://bonsai-plants.net/banyan-bonsai.php |
| 15:30.25 | spenguin[work] | man they look mad |
| 15:30.56 | spenguin[work] | http://www.fukubonsai.com/images/2a4.jpg |
| 15:31.36 | Kobaz | not a very good picture.. but |
| 15:31.37 | Kobaz | http://mvbonsai.com/galeries/2008%20MVBC%20Show/DSCF0002.JPG |
| 15:31.58 | Kobaz | the bushy one in the orange pot third from the left |
| 15:32.10 | Katty | returns |
| 15:32.20 | spenguin[work] | all those are yours Kobaz ? |
| 15:32.34 | Kobaz | is one of my dads plectranthus |
| 15:32.36 | bmoraca_work | a banyan bonsai would be way cool |
| 15:32.42 | Katty | Kobaz: that looks kinda like a ... |
| 15:32.46 | coppice | vertically challenged trees |
| 15:32.48 | spenguin[work] | hrm I think I want to do either a orange bonsai |
| 15:32.48 | Katty | Kobaz: well, not what i expected |
| 15:32.52 | Kobaz | hehe |
| 15:32.54 | spenguin[work] | or banyan bonsai |
| 15:32.57 | Katty | Kobaz: i always think of miniture cedar trees. |
| 15:33.09 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.157) |
| 15:33.16 | Katty | Kobaz: i'd rather have flowering plants, or plants that produce fruit tho |
| 15:33.18 | spenguin[work] | Kobaz: how old is the oldest tree? |
| 15:33.29 | Kobaz | spenguin[work]: 25 years is the oldest one we have |
| 15:33.42 | spenguin[work] | man |
| 15:33.42 | Katty | Kobaz: i also have a 'pocket' planter for the strawberry plants |
| 15:34.15 | spenguin[work] | older than me :s |
| 15:34.25 | Kobaz | the plectranthus in the orange pot is about 20ish years old i think |
| 15:34.27 | Katty | Kobaz: http://i.ehow.com/images/a04/r2/uq/make-strawberry-planter-800X800.jpg <- looks kinda like that |
| 15:34.40 | Katty | Kobaz: there are several tiers to it, with openings for the plants to grow through |
| 15:34.58 | Katty | Kobaz: i'm just not sure where to place the planter |
| 15:35.08 | Katty | Kobaz: they require lots of sun, right? |
| 15:35.32 | Kobaz | aughh |
| 15:35.36 | Kobaz | don't use one of those |
| 15:35.50 | Kobaz | unless you get a really big one |
| 15:36.02 | Katty | why? |
| 15:36.24 | Kobaz | they don't hold water very well, and there's really not much space for the fruits to poke out |
| 15:36.27 | spenguin[work] | Kobaz: I could grow a bonsai outdoors? |
| 15:36.32 | Kobaz | you'll get maybe like 10 strawberries from one of those |
| 15:36.45 | Kobaz | spenguin[work]: most people do actually... that's the 'real' way to do bonsai |
| 15:36.51 | spenguin[work] | ok |
| 15:37.03 | Kobaz | but it's much easier and comfortable to do inside |
| 15:37.07 | spenguin[work] | ill sneak it in on occasions |
| 15:37.22 | Kobaz | it's bad to change it's environment frequently |
| 15:37.38 | Quasar-1922 | <[TK]D-Fender> i've split the allow strings but still same problem.. also what do you mean by mathcing peer names? i have this in my dialplan for calling the other box: exten => _4XXX,1,Dial(IAX2/nl-ale-pbx01@nl-ams-pbx01/${EXTEN}@office) |
| 15:37.42 | Kobaz | the people who do outdoor bonsai, leave them outside all season, and then bring them in for the late fall through spring |
| 15:37.54 | Katty | Kobaz: so i'd do better to build a raised bed for them to grow in? |
| 15:38.00 | Kobaz | Katty: sure |
| 15:38.01 | spenguin[work] | hrm, here I just have - monsoons - summer - winter |
| 15:38.07 | spenguin[work] | s/I/we |
| 15:38.25 | Kobaz | so you get the wet season, and then monsoon season? |
| 15:38.42 | Katty | Kobaz: i'm starting to think that mulching over the whole area is a bad idea |
| 15:38.45 | *** join/#asterisk Skeeter- (skeeter@c216.218.2-65.clta.globetrotter.net) |
| 15:38.53 | joesuffceren | spenguin[work]: you in phoenix? |
| 15:39.13 | joesuffceren | weather sounds similar at least |
| 15:39.57 | Katty | Kobaz: let's say i wanted to divide that brown area up into segments |
| 15:40.19 | moos3 | can anyone help me with my ivr issue |
| 15:40.24 | Katty | Kobaz: it already has a border around in...how would you divide it up into segments without making it look odd? |
| 15:40.36 | Kobaz | Katty: i dunno, i'm not much of a designer |
| 15:40.43 | Kobaz | i just know how to not kill the plants |
| 15:40.45 | spenguin[work] | joesuffceren: nah far off - India |
| 15:41.07 | ManxPower-work | Not mulch of a designer? 8-) |
| 15:41.13 | babbio | I've installed asterisknow, i have already the freepbx installed but when i select the "administrator page" on the freepbx page it ask me for a username and password....what should i insert? |
| 15:41.13 | joesuffceren | spenguin[work]: haha. not even close. |
| 15:41.13 | Katty | Kobaz: putting in flowers, plants, and bushes in there, and putting mulch in there be bad for the plants? |
| 15:41.16 | Kobaz | Katty: you can use the plastic outdoor edge molding |
| 15:41.22 | spenguin[work] | :> |
| 15:41.27 | Katty | Kobaz: i have edge molding around the entire area now |
| 15:41.32 | *** join/#asterisk shiley (~chatzilla@122.165.61.71) |
| 15:41.42 | Kobaz | Katty: mulch helps keep weeds out, they help keep moisture so you don't have to water as often |
| 15:41.43 | Katty | Kobaz: i just think that putting landscaping rocks or what not inside the molded flower bed would just look stupid |
| 15:41.49 | spenguin[work] | joesuffceren: summers are pretty damn hot - we are heading for one atm :S |
| 15:41.53 | Kobaz | which may or may not be good for what your planting |
| 15:41.59 | Katty | Kobaz: so mulching on top of a freshly planted blueberry bush would be okay? |
| 15:42.08 | *** join/#asterisk asteriskATmarmuD (~mundt@193.158.65.23) |
| 15:42.26 | Kobaz | should be fine... if the area gets good sun and airflow, the mulch wont stay soggy from watering |
| 15:42.38 | shader | do any of you use a sip trunk service that you're happy with? |
| 15:42.50 | Katty | Kobaz: what would you recommend planting in that shady area in front of the house? |
| 15:43.00 | Katty | Kobaz: i would prefer plants that come back each year |
| 15:43.01 | Kobaz | sip trunk? is that like a tree branch that's laying in the lake? |
| 15:43.34 | Kobaz | there's all kinds of stuff that does well in the shade |
| 15:43.45 | *** join/#asterisk Deeewayne (~dwayne@75.76.254.162) |
| 15:43.46 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:43.52 | Kobaz | flowering? non-flowering? |
| 15:43.58 | Katty | flowering |
| 15:44.02 | Kobaz | mums are really easy |
| 15:44.07 | joesuffceren | shader: a lot of it depends on scale and requirements. I've been pretty happy with Teliax for small scale deployments |
| 15:44.11 | Quasar-1922 | [TK]D-Fender-> do the audiocodes mediant units have hardware echo cancel? |
| 15:44.16 | Katty | i have some mums |
| 15:44.20 | [TK]D-Fender | Quasar-1922: yes |
| 15:44.20 | Katty | they were very easy |
| 15:44.23 | Kobaz | and they come in like 23947982374 different varities |
| 15:44.34 | Katty | nods |
| 15:44.38 | Katty | what about non flowering? |
| 15:44.42 | Katty | something about 12 inches high |
| 15:45.09 | Kobaz | you could do something evergreen |
| 15:45.16 | Kobaz | and trim it back every year |
| 15:45.29 | Kobaz | any non-flower is going to need trimming to keep it small |
| 15:45.38 | spenguin[work] | try mango Katty |
| 15:45.39 | Kobaz | we have some juniper in the front |
| 15:46.11 | Kobaz | http://navigator.gardenpilot.com/AnnualsFullShade.html |
| 15:46.11 | spenguin[work] | mango or lemons |
| 15:46.13 | Kobaz | there's a nice list |
| 15:46.14 | shader | joesuffceren: have you tried faxing with Teliax? |
| 15:46.35 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 15:46.51 | joesuffceren | shader: DIE FAXING DIE... no, I haven't, sorry |
| 15:46.54 | Kobaz | rhododendrons are cool |
| 15:47.04 | Kobaz | we have one on the other side of the front |
| 15:47.18 | Kobaz | it's kinda grown like a bonsai |
| 15:47.23 | Kobaz | rather than a bushy lump |
| 15:47.47 | joesuffceren | shader: I do know that they support it to some level and they also have a fax>email service |
| 15:47.58 | shader | joesuffceren: interesting |
| 15:48.29 | moos3 | [TK]D-Fender: can you help me figure out my menu issue? |
| 15:48.39 | shader | has anyone else had some success with faxing over sip trunks? |
| 15:48.40 | *** join/#asterisk ralonso (~ralonso@140.Red-88-2-26.staticIP.rima-tde.net) |
| 15:48.50 | Kobaz | Euonymus is a pretty cool little shrub |
| 15:48.53 | Quasar-1922 | <[TK]D-Fender> ok, sounds good.. how do you connect them to asterisk, using SIP?? |
| 15:49.20 | Kobaz | i like plants with variegated leaves |
| 15:49.35 | [TK]D-Fender | moos3: ... youdon't even HAVE an enten in there to match! |
| 15:49.40 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 15:49.51 | [TK]D-Fender | moos3: Evereything = invalid. SO the second you hit something it = "i" |
| 15:50.03 | ManxPower-work | The n00bs, oh gawd the n00bs! |
| 15:50.08 | *** part/#asterisk ManxPower-work (~manxpower@234.sub-75-254-56.myvzw.com) |
| 15:50.16 | Kobaz | Euonymus japonicus has variegated leaves |
| 15:50.26 | moos3 | so i'll missing is the _XXXX,s part? |
| 15:50.47 | [TK]D-Fender | moos3: or "whatever". You have nothing to dial in there period |
| 15:51.36 | Katty | omnomnomnomnomnombreakfast |
| 15:51.41 | Katty | http://www.ustream.tv/channel-popup/squirrel-critter-cam |
| 15:51.43 | Katty | omnomnomnom |
| 15:51.50 | Qwell | squirrel cam isn't breakfast! |
| 15:51.56 | Kobaz | Katty: go to a nursery and ask what grows well in your area... and just window shop all the stuff that looks cool, and get something |
| 15:52.16 | Katty | Kobaz: that requires people skills :< |
| 15:52.27 | Katty | Kobaz: i got the animal skills, but not the people skills |
| 15:52.31 | Kobaz | heh |
| 15:52.44 | p3nguin | lol |
| 15:52.55 | moos3 | [TK]D-Fender, have the following on the out side of the menu, exten => _XXXX,1,Macro(stdexten,${EXTEN},SIP/${EXTEN},70) ;Catch all extensions not defined above do i need to move that into the menu context? |
| 15:53.26 | p3nguin | ~book |
| 15:53.27 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 15:53.29 | Kobaz | moos3: it might be helpful to paste the *rest* of your dialplan... random snippits dont help much |
| 15:53.47 | [TK]D-Fender | moos3: I think you need to understand that your context has no extens and that you have to put some in there or INCLUDE other contexts that already contain things you want them to be able to dila |
| 15:53.50 | [TK]D-Fender | dial* |
| 15:54.28 | Kobaz | moos3: and yes... please read the book, it'll answer many many questions |
| 15:54.45 | Kobaz | goes back to writing unit tests |
| 15:55.10 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-172.dslextreme.com) |
| 15:55.40 | ralonso | anyone know a "web" sip softphone? |
| 15:55.53 | p3nguin | zoiper has one. |
| 15:56.01 | moos3 | Kobaz: thanks |
| 15:56.37 | p3nguin | *sigh* I need to take a gateway offline, but the phones depend on it... and I don't want to wait until night to do the work. Bother. |
| 15:56.45 | *** join/#asterisk ParanoyaM (~kvirc@93.183.233.119) |
| 15:56.48 | *** join/#asterisk fifer (~fifer@67.208.108.228) |
| 15:57.16 | ParanoyaM | Hi, can anybody help me with call limits? i tried to limit a concurent calls and it is not working for me |
| 15:57.20 | moos3 | heres another question, our current dailplan says to enter you extension at the begining, but its only getting part of the extension ideas? |
| 15:57.55 | ParanoyaM | sip show peer mypear gives me : Call limit: 10 |
| 15:58.08 | p3nguin | paranoyam: In sip.conf, in a peer definition, call-limit=2 will limit the device to two calls. |
| 15:58.43 | ParanoyaM | p3nguin: i have 3 options: |
| 15:58.45 | ParanoyaM | call-limit=10 |
| 15:58.45 | ParanoyaM | Asterisk sip call-limit=10 |
| 15:58.45 | ParanoyaM | incominglimit=10 |
| 15:58.49 | ParanoyaM | not one works for me |
| 15:59.06 | p3nguin | It seems to be working like you have configured it to work. |
| 15:59.15 | ParanoyaM | in general in sip.conf i tried to enable limitonpeers |
| 15:59.17 | p3nguin | You configured it for 10, you said it was limited to 10. |
| 15:59.28 | ParanoyaM | yes, but i see 13 calls in up state |
| 15:59.32 | Katty | what do i want for lunch |
| 15:59.50 | spenguin[work] | tandoori |
| 15:59.58 | ParanoyaM | p3nguin: here it is: |
| 16:00.00 | ParanoyaM | 26 active channels |
| 16:00.00 | ParanoyaM | 13 active calls |
| 16:00.04 | Katty | what do i want for lunch, that i can make here, with minimal effort |
| 16:00.14 | Katty | 30 minutes, max |
| 16:00.18 | spenguin[work] | omlette |
| 16:00.43 | ParanoyaM | p3nguin: i have only one peer. |
| 16:00.50 | *** join/#asterisk jelly-bean (~jelly-bea@75-148-103-190-Utah.hfc.comcastbusiness.net) |
| 16:01.11 | jelly-bean | does anyone have the command to convert the asterisk ulaw .wav files into a .mp3 file |
| 16:01.23 | *** join/#asterisk Poincare (~jefffnode@213.219.184.23) |
| 16:01.29 | ParanoyaM | p3nguin: any idea? |
| 16:01.37 | p3nguin | jelly-bean: see "file convert" on your Asterisk CLI. |
| 16:01.51 | Kobaz | file convert does mp3s now? |
| 16:01.56 | p3nguin | I don't know. |
| 16:02.00 | Kobaz | i don't think so |
| 16:02.06 | p3nguin | Well then that's too bad. |
| 16:02.27 | [TK]D-Fender | It doesn't |
| 16:02.42 | [TK]D-Fender | jelly-bean: LAME <- |
| 16:03.13 | Kobaz | jelly-bean: first you need to convert from ulaw to pcm signed: sox -v 3 src.wav -e signed-integer dest.wav |
| 16:03.31 | Qwell | Why would you want to use mp3? |
| 16:03.35 | ralonso | anyone know another sip softphone web like zoiper? |
| 16:03.40 | Kobaz | jelly-bean: and then you make your mp3: lame -b <bitrate> -m m dst.wav dst.mp3 |
| 16:03.42 | Qwell | That's an unnecessary conversion.. |
| 16:03.44 | p3nguin | ralonso: One isn't enough? |
| 16:04.03 | Kobaz | Qwell: lame won't encode properly if you don't pre-convert |
| 16:04.20 | Qwell | Kobaz: you're missing the point |
| 16:04.22 | atis_work | /opt/sox/bin/sox -M -t raw -r 8000 -s -w /mnt/gluster/voip/monitor/2010/03/08/call-R1-1268063614.2-1-in.sln -t raw -r 8000 -s -w /mnt/gluster/voip/monitor/2010/03/08/call-R1-1268063614.2-1-out.sln -t mp3 /mnt/gluster/voip/monitor/2010/03/08/call-R1-1268063614.2-1.mp3s8 |
| 16:04.28 | atis_work | working on it :) |
| 16:04.34 | ralonso | is to compare |
| 16:04.35 | Kobaz | well yeah, mp3 is silly, since it tends to be bigger than the original wav |
| 16:04.53 | atis_work | Kobaz: it depends on sample rate |
| 16:05.05 | Kobaz | qwell: but... if you say, want to stream call recordings over the web with a flash player... well then you need mp3 |
| 16:05.12 | ParanoyaM | p3nguin: here is sip show peer http://pastebin.ru/311086 and sip.conf : http://pastebin.ru/311087 |
| 16:05.25 | ParanoyaM | so any ideas why call limitation is not working? |
| 16:05.29 | atis_work | Kobaz: and you need MP3 11KHz |
| 16:05.36 | atis_work | not 8 as from asterisk |
| 16:05.51 | atis_work | anyone wants to try streaming ogg directly? |
| 16:06.11 | atis_work | Flash has known compatibility problems with 8khz mp3 |
| 16:06.12 | *** join/#asterisk fofware (~chatzilla@190.229.137.89) |
| 16:06.16 | p3nguin | paranoyam: line 018 is invalid. |
| 16:06.26 | ParanoyaM | p3nguin: which file? |
| 16:06.27 | p3nguin | paranoyam: in sip.conf, that is. |
| 16:06.35 | *** join/#asterisk mayfield (~mayfield@76-250-152-224.lightspeed.snantx.sbcglobal.net) |
| 16:07.22 | ParanoyaM | p3nguin: it is not an issue i can remove it but still it is not working, i will try now |
| 16:07.45 | [TK]D-Fender | ParanoyaM: limitonpeers=yes ......... type=friend .................ISN'T A FRIGGEN PEER |
| 16:07.51 | [TK]D-Fender | :D |
| 16:08.09 | [TK]D-Fender | ParanoyaM: type=peer ,_ |
| 16:08.13 | [TK]D-Fender | ^ |
| 16:08.28 | ParanoyaM | [TK]D-Fender: can you explain as for user pleas |
| 16:08.48 | ParanoyaM | [TK]D-Fender: i am not expert in asterisk |
| 16:09.26 | babbio | guys....i have a question...i have installed asterisknow but i think it comes with centos server because i have no giu login.....now my problem is that i would like to use visual dialplan so i need the Xserver...how could i do? |
| 16:09.27 | ParanoyaM | [TK]D-Fender: you mean this type=friend |
| 16:09.47 | ParanoyaM | [TK]D-Fender: i tried both peer and friend, doesn't work for me |
| 16:10.11 | ParanoyaM | [TK]D-Fender: but changed back to peer |
| 16:10.18 | joesuffceren | babbio: don't run X on your pbx. it's running a webserver which will let you access the Freepbx GUI. Access the IP address of your PBX from another computer's browser |
| 16:10.34 | *** join/#asterisk Anomizer (~Anomizer@mx.onboard.com) |
| 16:11.00 | babbio | yes i already done it |
| 16:11.16 | babbio | but i need to execute visual dialplan on the pbx server |
| 16:11.25 | Anomizer | Hello, I was wondering if anyone had a recommendation as to the best Asterisk book (Regardless of Price) |
| 16:11.28 | babbio | so i need X on the pbx server machine |
| 16:11.34 | Qwell | ~buybook |
| 16:11.34 | infobot | [~buybook] You can buy "Asterisk the Future of Telephony" at http://www.oreilly.com/catalog/9780596510480/ so go buy it SERIOUSLY |
| 16:12.04 | jelly-bean | Kobaz: i had been using sox and lame before with diff settings. just tried yours. that works too. except in both cases, LAME is adding blips and squeaks to the audio. u can still make out what is said but it has added all this random noise |
| 16:12.14 | Anomizer | Nice, thanks guys!!! |
| 16:13.13 | leifmadsen | w00t :) |
| 16:13.31 | leifmadsen | places another nickle in his jar |
| 16:13.42 | jelly-bean | atis_work: sox FAIL formats: no handler for given file type `mp3' |
| 16:13.51 | Kobaz | jelly-bean: in my commandline there are some gain settings to bump up the volume ... take out the -v3 |
| 16:13.52 | ParanoyaM | here is new sip show peer: http://pastebin.ru/311088 and sip.conf: http://pastebin.ru/311089 |
| 16:14.10 | ParanoyaM | right now |
| 16:14.12 | ParanoyaM | 26 active channels |
| 16:14.12 | ParanoyaM | 13 active calls |
| 16:14.14 | jelly-bean | Kobaz: did so. i usually also see errosr while playing like: mpg123: Can't rewind stream by 5 bits!6% |
| 16:14.15 | jelly-bean | <PROTECTED> |
| 16:14.25 | Kobaz | hmm |
| 16:14.29 | ParanoyaM | p3nguin: |
| 16:14.32 | ParanoyaM | [TK]D-Fender: |
| 16:14.39 | ParanoyaM | maybe you know where my mistake? |
| 16:14.42 | Kobaz | jelly-bean: a different mp3 player? |
| 16:15.02 | jelly-bean | also while encoding with sox i get: sox wav: Length in output .wav header will be wrong since can't seek to fix it |
| 16:15.14 | Kobaz | jelly-bean: oh that sounds bad |
| 16:15.23 | Kobaz | what sox version? |
| 16:16.15 | fifer | I have a * 1.4.29 system and I'm setting up a new Aastra phone. I have a 480i that is working fine wiht my * setup but the 6731i I'm setting up is not getting dtmf to * during the call. |
| 16:16.40 | fifer | They are setup almost identically with their dial plans and dtmf settings the same. |
| 16:17.23 | jelly-bean | Kobaz: sox: SoX v14.0.0 |
| 16:17.23 | fifer | When calling VM the * does not get anything from the 6721i when I try to enter a pw nor when entering an extensions on a foreign phone system. |
| 16:17.29 | *** join/#asterisk pabelanger (~pabelange@yoda.kanatek.com) |
| 16:17.33 | fifer | Works fine with the 480i |
| 16:17.47 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 16:18.00 | pabelanger | yo: Anybody using 'Skype for Asterisk' with 1.6.2? |
| 16:18.07 | jelly-bean | Kobaz: what player do u use with the mp3? |
| 16:18.11 | ChannelZ | I'm running it on 1.6.1 |
| 16:18.29 | Kobaz | jelly-bean: mplayer |
| 16:18.33 | [TK]D-Fender | ParanoyaM: I'm not seeing updated configs, nor sample failed calls, channel dumps, etc |
| 16:18.38 | jelly-bean | Kobaz: yep thats what im using |
| 16:18.54 | Kobaz | i thought you said mpg123 |
| 16:19.00 | ParanoyaM | [TK]D-Fender: here is sip show inuse and core show channels : http://pastebin.ru/311090 |
| 16:19.01 | joesuffceren | pabelanger: not supported on 1.6.2: http://www.digium.com/en/docs/SFA/sfa_faq.php |
| 16:19.09 | jelly-bean | Kobaz: that error comes from mplayer tho |
| 16:19.13 | Kobaz | k |
| 16:19.26 | ParanoyaM | [TK]D-Fender: and here is new sip show peer: http://pastebin.ru/311088 and sip.conf: http://pastebin.ru/311089 |
| 16:19.36 | Kobaz | umm.. hmm... you're outputting ulaw... you know, i think i'm converting from gsm |
| 16:19.51 | Kobaz | i do Monitor(foo.WAV) |
| 16:19.52 | Qwell | joesuffceren, pabelanger: That is out of date. It does indeed work with 1.6.2 |
| 16:19.57 | Kobaz | well MixMonitor rather |
| 16:20.03 | jelly-bean | my original file is: ulaw.wav: RIFF (little-endian) data, WAVE audio, ITU G.711 mu-law, mono 8000 Hz |
| 16:20.19 | jelly-bean | i got my file from Oreka btw. |
| 16:20.20 | joesuffceren | Qwell: thanks. sorry, pabelanger |
| 16:20.32 | jelly-bean | its supposed to be encoded with ulaw |
| 16:20.43 | Kobaz | /var/spool/asterisk/monitor/1268120828.397058.WAV: RIFF (little-endian) data, WAVE audio, GSM 6.10, mono 8000 Hz |
| 16:20.54 | Kobaz | that's my source recordings |
| 16:20.55 | *** join/#asterisk Slugs_ (Slugs_@c-76-97-205-31.hsd1.ga.comcast.net) |
| 16:21.11 | Kobaz | jelly-bean: did you try without the sox conversion? and convert from wav to mp3? |
| 16:21.18 | jelly-bean | Kobaz: do you run your asterisk recordings on the same box as the rest of the pbx? or separate? |
| 16:21.37 | pabelanger | Qwell: Thanks for the confirmation. Will have to wait for digium support then to help get it up and running. |
| 16:21.46 | jelly-bean | Kobaz: yes Oreka has option to save in pcm instead of ulaw so i tried that and lame will convert it but its still got the same squeeks after. so its definitely during the lame conversion |
| 16:21.50 | Slugs_ | What ports could sip listen on, that 'probably' would not be blocked by the firewall if 5060 is being blocked? |
| 16:21.55 | jelly-bean | Kobaz: the original wav sounds great |
| 16:21.56 | Kobaz | jelly-bean: everything used to be all done on one box... now we have an A/B setup for failover, with a db server that rsyncs the recordings and cdr's and encodes them |
| 16:22.25 | Kobaz | jelly-bean: but the biggest issue was using Monitor() rather than MixMonitor() |
| 16:22.26 | [TK]D-Fender | ParanoyaM: http://www.voip-info.org/wiki/view/Asterisk+sip+incominglimit |
| 16:22.45 | leifmadsen | pabelanger: ya, I'm using it on 1.6.2 latest branch right now |
| 16:22.47 | Kobaz | Monitor() has a serious design flaw that affects audio bridging when you have high disk io |
| 16:22.50 | jelly-bean | Kobaz: we saw the load of the box drop dramatically once we moved recording, mixing, compressing, and archiving processes to their own box |
| 16:23.11 | Kobaz | once we switched to MixMonitor, all load issues dissapeared |
| 16:23.11 | [TK]D-Fender | ParanoyaM: appears to be incoming anyway, not outgoung. Use GROUP() |
| 16:23.23 | jelly-bean | interesting |
| 16:23.40 | ParanoyaM | [TK]D-Fender: i can't use group. it will mix up all my routing |
| 16:23.49 | Kobaz | jelly-bean: Monitor() does audio recording and audio bridging in the same thread... so if you have high disk io, you will drop audio in the call itself |
| 16:23.52 | [TK]D-Fender | ParanoyaM: how so? |
| 16:24.20 | ParanoyaM | [TK]D-Fender: i am using macro that distribute calls between gateways, and operates with hangupcases |
| 16:24.27 | Kobaz | MixMonitor has a seperate thread that saves the audio, and mixes it at the same time |
| 16:24.35 | Kobaz | *much* less post processing, no post-mixing needed |
| 16:24.53 | atis_work | i prefer post-processing into stereo :) |
| 16:25.10 | atis_work | at night time when there are no calls |
| 16:25.11 | [TK]D-Fender | ParanoyaM: So? Everywhere you'd dial out that peer, just use a group count check |
| 16:25.19 | leifmadsen | pabelanger: huh, actually, just updating to the LATEST 1.6.2 branch (since a few days ago) seems to have broken it :) |
| 16:25.47 | jelly-bean | Kobaz: how does your rsync run? do you have it in a bash script or are there rsync options to watch continuously for new files, copy them over, and delete from source? |
| 16:25.55 | ParanoyaM | [TK]D-Fender: honestly say i am unable to make this, i was helped to write the macro |
| 16:25.57 | Kobaz | jelly-bean: it runs every 5 minutes |
| 16:26.23 | Kobaz | jelly-bean: it would be cool it rsync had inotify support to do hotcopy from the source upon file update/create |
| 16:27.01 | leifmadsen | Kobaz: does unison help with that? |
| 16:27.17 | Kobaz | unison? |
| 16:27.19 | leifmadsen | oh probably not -- I think I understand what you just said now :) |
| 16:27.25 | Kobaz | danielson? |
| 16:27.28 | leifmadsen | :) |
| 16:27.37 | Slugs_ | What ports could sip listen on, that 'probably' would not be blocked by the firewall if 5060 is being blocked? |
| 16:27.43 | leifmadsen | ~unison |
| 16:27.44 | infobot | it has been said that unison is a nice tool to syncronise files between two systems, it uses the rsync transferprotocol and can be used over ssh or over socket, apt-get install unison, or at http://www.cis.upenn.edu/~bcpierce/unison/ |
| 16:27.49 | leifmadsen | nice :) |
| 16:27.58 | Kobaz | leifmadsen: rewriting Monitor() has been on my todo list though |
| 16:28.08 | leifmadsen | Kobaz: w00t! :) |
| 16:28.13 | leifmadsen | that code is pretty old I suspect |
| 16:28.17 | Kobaz | probably |
| 16:28.21 | Kobaz | and a lot of people still use Monitor |
| 16:28.25 | Kobaz | and it's written crappily |
| 16:28.38 | leifmadsen | agreed on the 2nd point :) |
| 16:28.40 | Kobaz | leifmadsen: it took me 6 months to find the problem |
| 16:28.45 | leifmadsen | ouch |
| 16:28.46 | [TK]D-Fender | ParanoyaM: Guess you'd better learn the dialplan. It is 95% of * you know... |
| 16:28.56 | leifmadsen | Kobaz: then I suspect many people would be happy with any fixes :) |
| 16:29.14 | Kobaz | leifmadsen: random audio was droping from calls... and it's like wtf... no t1 slips, no bit errors no nothing,... asterisk was just losing random audio frames |
| 16:30.16 | leifmadsen | Kobaz: o.O ouch |
| 16:30.42 | Kobaz | so... if you do not explicily need individual leg recording... MixMonitor is highly recomended |
| 16:30.50 | Squeeb | Hmmm, I'm trying to get announce-frequency to work, however I can't understand why it only seems to work when the timeout has expired in queues.conf |
| 16:31.02 | Kobaz | Squeeb: because of the way dialplan works |
| 16:31.05 | jelly-bean | Unison-hackers inotify: http://lists.seas.upenn.edu/pipermail/unison-hackers/2006-October/000521.html |
| 16:31.09 | Squeeb | ah |
| 16:31.16 | Kobaz | Squeeb: you can only do one thing at a time... you can ring a phone, or you can play a track... not both |
| 16:31.22 | *** join/#asterisk timeshell (~timeshell@gw.lusi.on.ca) |
| 16:31.22 | Squeeb | I see. |
| 16:31.39 | Squeeb | So what's the announce-frequency for if it can't announce during playback of musiconhold ? |
| 16:31.50 | Kobaz | Squeeb: because of that... the only way you'll play a track/announcement, is if you're not ringing a phone... and the only way to not ring a phone, is to hit the timeout |
| 16:31.51 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 16:32.00 | Squeeb | I see.. |
| 16:32.06 | Kobaz | Squeeb: it's pretty useless for me... i had to write my own call queue module from scratch |
| 16:32.13 | Squeeb | :/ |
| 16:32.40 | Kobaz | ~2000 lines of perl... it works great |
| 16:32.43 | Squeeb | Doesn't sound fun at all |
| 16:32.54 | Kobaz | although i'm still writing tests for it |
| 16:32.57 | Kobaz | three weeks later |
| 16:32.58 | Quasar-1922 | <[TK]D-Fender> those audiocodes mediants are quite expensive ;-) |
| 16:32.58 | Kobaz | heh |
| 16:33.09 | Squeeb | i see |
| 16:33.21 | Squeeb | if I didn't have music on hold? |
| 16:33.23 | Squeeb | and had ringing instead |
| 16:33.26 | Kobaz | same problem |
| 16:33.27 | Squeeb | would that help? |
| 16:33.28 | Squeeb | arse |
| 16:33.38 | jelly-bean | http://code.google.com/p/lsyncd/ |
| 16:33.42 | Kobaz | the best you can do... is open up an audio editor... load up some music... and paste in your announcements |
| 16:33.59 | Kobaz | every 30 seconds, paste in your 'thank you for calling' |
| 16:34.08 | Squeeb | yea, we're trying to announce queue position |
| 16:34.13 | Squeeb | so erm .. can't really do that :P |
| 16:34.16 | Kobaz | heh |
| 16:34.30 | Kobaz | okay so... you'll have to write a seperate script/daemon that uses chanspy whisper |
| 16:34.42 | Squeeb | I see |
| 16:34.47 | ParanoyaM | [TK]D-Fender: thank you |
| 16:34.52 | Kobaz | you need third party call control, is what you need |
| 16:34.58 | Kobaz | and the call queue module does not give you that |
| 16:35.04 | Squeeb | sod it, I'll just wang the timeout to 5 :P |
| 16:35.33 | Kobaz | it was just the most annoying thing in the world that phones would stop ringing when you play tracks |
| 16:35.40 | Squeeb | 99yea |
| 16:35.42 | Kobaz | but there's nothing you can do about that without a huge rewrite effort |
| 16:35.44 | Squeeb | it is a bit of fail |
| 16:36.30 | spenguin[work] | hey what are these called exactly - 'COMPLETEAGENT', 'COMPLETECALLER','TRANSFER' , 'EXITWITHTIMEOUT' |
| 16:36.43 | spenguin[work] | they are in the queue_logs |
| 16:36.46 | spenguin[work] | table |
| 16:36.46 | Kobaz | spenguin[work]: return codes, or reason codes |
| 16:36.50 | spenguin[work] | ok |
| 16:36.53 | *** join/#asterisk Warp4 (~robert.wo@cpe-69-204-103-227.buffalo.res.rr.com) |
| 16:40.20 | *** part/#asterisk babbio (~matteo@host-78-13-24-238.cust-adsl.tiscali.it) |
| 16:42.59 | fifer | I'm having issues with dtmf comming from one of my sip phones. I have a 480i that is working fine with my * setup but the 6731i I'm setting up is not getting dtmf to * during the call. |
| 16:43.27 | fifer | They are configured almost identically with dtmf and dial plan the same. |
| 16:44.29 | jelly-bean | is there a way to convert this wav files to smaller wavs? i mean when i did wav to mp3 conversion on 8,000+ files it went from 9gb to 1gb. but then i got squeaks added to the mp3 by lame. is there a way to compress a wav, so i can avoid lame? |
| 16:44.58 | fifer | They are both set at RTP for DTMF method and Force RFC2833 Out-of-Band |
| 16:45.28 | fifer | This is * 1.4.29 |
| 16:46.56 | fifer | I saw something about a "dtmf debug" cli command but it is either depricated or in a newer version of * as I do not have it. |
| 16:49.04 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 16:49.55 | *** join/#asterisk mnick86 (~mnick86@whhem00002.cip.uni-regensburg.de) |
| 16:50.57 | thehar | Has anyone ever interopped with Verlocity Networks? |
| 16:51.11 | *** join/#asterisk Brack10 (~tbrackett@66.35.2.50) |
| 16:52.58 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 16:54.39 | *** join/#asterisk stix_ (~stix@x1-6-00-1e-2a-87-8d-aa.k1.webspeed.dk) |
| 16:57.25 | leifmadsen | fifer: that command doesnt' exist |
| 16:57.37 | leifmadsen | fifer: never has -- enable DTMF debugging via logger.conf |
| 16:58.11 | leifmadsen | lunch! |
| 17:02.41 | *** join/#asterisk timeshell (~timeshell@206.248.136.108) |
| 17:03.01 | [TK]D-Fender | jelly-bean: audacity is scriptable. |
| 17:07.55 | *** join/#asterisk hfb (~hfb@pool-98-112-219-90.lsanca.dsl-w.verizon.net) |
| 17:09.10 | fifer | @leifmadsen: Thanks! |
| 17:09.56 | *** join/#asterisk xLP (~test@mail-out.lpcorp.fr) |
| 17:10.32 | *** part/#asterisk csiadmin (~csiadmin@81.144.152.52) |
| 17:12.40 | Kobaz | russellb: okay i got something for you |
| 17:12.42 | Kobaz | er |
| 17:13.30 | *** part/#asterisk rttrey (~trey@209.208.18.121) |
| 17:13.45 | *** join/#asterisk rttrey (~trey@209.208.18.121) |
| 17:14.20 | *** join/#asterisk DennisG (DennisG@2002:541e:88d0:0:4d26:dc05:7f65:70fb) |
| 17:26.20 | fifer | I have confirmed with dtmf debugging in the log enabled that my new Aastra 6731i's are not getting dtmf to * |
| 17:26.53 | fifer | I now have 3 6731i's setup so I'm fairly sure it is not a hardware issue wiht a particular phone. they all have the latest firmware |
| 17:29.46 | DennisG | do you use the rfc2833 dtmf mode ? |
| 17:35.45 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 17:35.50 | *** part/#asterisk pabelanger (~pabelange@yoda.kanatek.com) |
| 17:36.28 | *** join/#asterisk mrtelnet (~mr.telnet@intouchpharma.com) |
| 17:36.29 | fifer | DennisG: Yes, on both Aastra models |
| 17:36.41 | fifer | Working on the 480i not on the 6731i |
| 17:37.41 | mrtelnet | I have an issue where on an attended transfer of caller a from b to c, asterisk does not appear to be forwarding rtp from c to a. Canreinvite is no and ther is no natting or firewalls enabled. I have a pcap if anyone can look. |
| 17:39.25 | Brack10 | Is there a list of phones tested with asterisk? |
| 17:39.34 | *** join/#asterisk Bokhuval (~user@cpe-70-112-22-94.austin.res.rr.com) |
| 17:39.43 | Qwell | Brack10: if it's SIP, it'll probably work |
| 17:41.32 | *** part/#asterisk lftsy (~lftsy@leonhart.leurent.eu) |
| 17:41.34 | Bokhuval | I have some questions about system requirements, but first wanted to ask would a small office (1 incoming number, about 10 extensions) need standalone/physical hardware or could asterisk be satisfied running on a VM? |
| 17:41.35 | fifer | Also just about any analog phone will work with an ATA box |
| 17:42.12 | fifer | Bokhuval: due to the nature of how Asterisk runs, it is best to always use dedicated hardware |
| 17:42.25 | [TK]D-Fender | mrtelnet: If canreinvite=no ... RTP isn't SUPPOSED to go between A * C |
| 17:42.25 | Nugget | telnet is eeeeeeevil! |
| 17:42.29 | fifer | That said, for your situation, you might have some OLD hardware that woudl work just fine |
| 17:42.37 | *** join/#asterisk DagMoller (~aguirre@unaffiliated/dagmoller) |
| 17:42.42 | DagMoller | tzafrir_laptop, hi |
| 17:42.51 | fifer | A P-4 with 1GB ram and a 40GB HD would do nicely |
| 17:42.54 | tzafrir_laptop | DagMoller, hi |
| 17:43.04 | mrtelnet | [TK]D-Fender: no, it's not, but it should be relayed by asterisk to the original caller. That does not apppear to be happening. |
| 17:43.14 | DagMoller | tzafrir_laptop, good news: monast now can handle multiple servers status |
| 17:43.23 | Bokhuval | fifer: I think I can scrounge that kindof hardware up. |
| 17:43.49 | [TK]D-Fender | mrtelnet: pastebin your configs and the failed call |
| 17:43.50 | [TK]D-Fender | ~pb |
| 17:43.51 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 17:43.53 | [TK]D-Fender | ^^^^^^ |
| 17:44.13 | fifer | Bokhuval: The only issue with HD size is the ammount of VM space though 40GB is still probably plenty |
| 17:45.03 | Katty | mmm |
| 17:45.11 | Katty | cheese stuffed digorno pizza |
| 17:45.26 | Bokhuval | fifer: it's a small private practice office and voicemail gets checked/cleared every day. What about bandwidth? They have a 10mb/1mb cable connection. I would guess it to be plenty but how many concurrent calls do you think that could handle? |
| 17:45.29 | p3nguin | bokhuval: I can't imagine you would have only 10 extensions... I have over 200 extensions, and I only have less than a dozen phones. |
| 17:45.35 | fifer | Katty: Try the new Flatbread versions!! |
| 17:45.41 | Katty | i don't care for them. |
| 17:45.47 | Katty | i like the cheese stuffed. |
| 17:45.53 | Bokhuval | p3nguin: it's a doctor's office. 1 doctor, a couple staff. |
| 17:46.02 | p3nguin | No IVRs? |
| 17:46.08 | p3nguin | No Voicemails? |
| 17:46.11 | [TK]D-Fender | Bokhuval: You could fit a few simultaneous calls through that |
| 17:46.57 | Bokhuval | p3nguin: there will be voicemail, yes. IVR I don't know - I'm just getting started on this whole project :] |
| 17:46.59 | fifer | Bokhuval: You only need to worry about the bandwidth if you are using SIP trunk(s) or connecting 1 or more phones from outside the office, if you are just using an existing ptns line and all the phones are in the office, it is not an issue |
| 17:47.08 | p3nguin | If you use g.729 and an IAX2 trunk, you might get more than just a few calls. |
| 17:47.42 | fifer | Even if you are doing a bit of both, you are still probably fine, but having some ability to do QoS will make a diference |
| 17:47.53 | Brack10 | Qwell: do all the features work on Cisco 7000 series phones with a SIP image? |
| 17:48.40 | Bokhuval | I will look into tweaking the iptables set on the firewall machine, but yes it's definitely doable. |
| 17:48.59 | Katty | that's what she said |
| 17:49.01 | Bokhuval | as for connecting to the outside world, was looking at broadvoice |
| 17:49.41 | Brack10 | Bokhuval: I had them at home. Terrible terrible terrible |
| 17:49.45 | Brack10 | Horrible |
| 17:49.58 | Bokhuval | any recommendations instead then? |
| 17:50.13 | Brack10 | I had vonage a while back, they're a lot better |
| 17:50.30 | p3nguin | I'm satisfied with VoIP.ms. |
| 17:50.32 | Bokhuval | Will they accept connections from a personally maintained BPX setup? |
| 17:50.36 | Bokhuval | pbx, rather |
| 17:50.44 | Brack10 | Last time I checked, yeah |
| 17:50.45 | Katty | infobot: itsp-us |
| 17:51.52 | Katty | infobot: itsp |
| 17:51.53 | infobot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 17:51.53 | Katty | frowns |
| 17:51.53 | Katty | infobot: are you broken |
| 17:51.55 | Katty | okay who broke infobot |
| 17:51.55 | Katty | blames Qwell |
| 17:51.55 | fifer | Someone fed him cheese? |
| 17:51.55 | dddh | DennisG: ~itsp |
| 17:51.56 | fifer | Or stuffed crust pizza?? ;-) |
| 17:51.56 | Bokhuval | I've used vonage for years for my home phone and had good experiences. will have to check them out for this too. |
| 17:52.01 | *** join/#asterisk theHub (~theHub@69.177.93.21) |
| 17:52.07 | Katty | infobot: itsplist0-us |
| 17:52.09 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 17:52.11 | Katty | infobot: itsplist-us |
| 17:52.12 | infobot | from memory, itsplist-us is Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net, or http://flowroute.com , http://voip.ms |
| 17:52.15 | Katty | there. |
| 17:52.20 | Katty | apparentl i just fail today. |
| 17:52.25 | Katty | frowns |
| 17:52.28 | Katty | +y... |
| 17:53.05 | Katty | Kobaz: they have blueberry bushes at the nursery |
| 17:53.10 | Katty | Kobaz: several varieties |
| 17:53.21 | fifer | this is weird, I can't find any hint what might be wrong with my Aastra 6731i setup. DTMF is NOT getting to * during call |
| 17:53.44 | Katty | test with a sip phone |
| 17:53.52 | Katty | and tell us if that works |
| 17:53.56 | fifer | I can't find anything of consiquence that is diference between the their sip config and my 480i, which works fine |
| 17:54.02 | [TK]D-Fender | fifer: When would it get to * OUTSIDE of a call? |
| 17:54.14 | fifer | OK, got me there ;-) |
| 17:55.35 | xLP | Anyone saw "Nobody picked up in 2000 ms" before? How can I tweak that? I find several things about it on Google, but nothing useful. |
| 17:55.46 | Katty | change the timeout |
| 17:56.28 | xLP | in Dial() ? |
| 17:56.34 | Katty | frowns. |
| 17:57.17 | Katty | xLP: take the asterisk book and put it on your head. |
| 17:57.25 | Katty | xLP: sit like that for 4 hours...maybe something will seep in |
| 17:58.08 | [TK]D-Fender | ~OSMOSIS |
| 17:58.09 | infobot | [~osmosis] Osmosis is the act of beating yourself on the head repeatedly with THE BOOK, until some measure of absorption has occured ... or at least until your unconsciousness restores peace to the channel ... |
| 17:58.11 | [TK]D-Fender | ^^^^^^^^^^^ |
| 17:58.15 | xLP | lol |
| 17:58.55 | [TK]D-Fender | xLP: Would also help if you showed us WHERE this was happening.. |
| 17:59.03 | [TK]D-Fender | xLP: PASTEBIN a sample call that does this |
| 17:59.05 | [TK]D-Fender | ~pb |
| 17:59.06 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 17:59.08 | [TK]D-Fender | ^^^ |
| 17:59.16 | Katty | hmm. cake. hmmmmmm. |
| 17:59.27 | *** join/#asterisk nickaugust (~anonymous@rrcs-71-42-53-182.se.biz.rr.com) |
| 17:59.29 | Katty | think i'm too full for cake :< |
| 17:59.35 | Katty | which is a shame |
| 17:59.50 | xLP | sorry for thinking my problems are so common |
| 18:00.03 | xLP | or obvious... however, if Katty meant the timeout in Dial(), she was right |
| 18:00.12 | xLP | thx Katty |
| 18:00.37 | Katty | i meant timeout. |
| 18:00.57 | Katty | but whatever. |
| 18:01.08 | Katty | it's not worth frowning over again. |
| 18:01.23 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 18:03.01 | Bokhuval | Is there a difference between functionality of asterisk (downloading/compiling/installing) and AsteriskNOW iso kickstart? |
| 18:03.43 | Naikrovek | yes |
| 18:04.03 | Katty | no idea, never used asterisknow |
| 18:04.20 | Naikrovek | asterisknow is a whole linux distro, with a lot of things tacked on to asterisk. asterisk is just asterisk |
| 18:04.34 | Katty | sounds... |
| 18:04.35 | Katty | closed. |
| 18:05.15 | Naikrovek | asterisknow is a digium product and it's fully open meaning you can fiddle with any part of it (linux, freepbx, asterisk, fop, etc) but tweaking any breaks the others |
| 18:05.33 | Naikrovek | but this is not the channel for asterisknow |
| 18:05.40 | Katty | eww. |
| 18:05.42 | Naikrovek | you will want #asterisknow if you need help with it |
| 18:05.42 | Bokhuval | Yes, the iso build is based on CentOS. Let me rephrase the question - will I miss out on any features of getting/building it myself as a standard package? |
| 18:05.46 | Katty | i don't want that crap. |
| 18:06.01 | Katty | i'd guess stability for one |
| 18:06.19 | Naikrovek | Bokhuval: asterisknow and its kin all shield you from the dialplan. freepbx is powerful, but not as powerful as having your own access to the dialplan |
| 18:06.31 | p3nguin | If you use FreePBX at all, you lose out on all the fine-grained tuning ability you get without it. |
| 18:06.34 | Naikrovek | Bokhuval: if you want to install/maintain an asterisk install of any size, you will not want asterisknow |
| 18:07.08 | p3nguin | ~asterisknow |
| 18:07.09 | infobot | [asterisknow] based on Asterisk, but is difficult to support in #asterisk for a number of reasons. Please seek support in #asterisknow instead. |
| 18:07.10 | Naikrovek | by "of any size" i mean "of any size larger than a few endpoints" |
| 18:07.14 | Bokhuval | *nod* |
| 18:07.53 | Naikrovek | if you wind up liking asterisk, which i'm sure you will, you'll want to do some more complicated things with it, and freepbx (asterisknow) will limit you in a lot of ways |
| 18:08.01 | Bokhuval | I don't mind a longer build if it means better functionality and control :] there wasn't much info about it, just 'download and start!' so figured I'd ask. |
| 18:08.02 | p3nguin | It might be okay for your small doctor's office, though. |
| 18:08.08 | Naikrovek | yes |
| 18:08.15 | p3nguin | It really depends on what you need and what you might need later. |
| 18:08.23 | Katty | i would not want any unstable platform in a doctor's office. |
| 18:08.29 | Katty | that'd make me wibble |
| 18:08.30 | thehar | what other SBCs besides ACME would anyone recommend? |
| 18:08.39 | Bokhuval | I'd be willing to bet we'll need it. 2 more doctors coming on this year and all the extra overhead. |
| 18:08.39 | Katty | what if they have an emergency and have to call the hospital? |
| 18:08.42 | p3nguin | That's why AsteriskNOW would be good for an office... it's not unstable at all. |
| 18:08.44 | seanbright | unstable? |
| 18:08.49 | Naikrovek | Katty: it's stable, reliably stable, as long as you don't fiddle with it |
| 18:08.54 | seanbright | Katty: have you ever used asterisknow? |
| 18:09.01 | Katty | seanbright: i've used freepbx. |
| 18:09.08 | Katty | seanbright: and from what Naikrovek says that's on asterisknow |
| 18:09.08 | seanbright | how about asterisknow? |
| 18:09.11 | Naikrovek | start fiddlin' and you're in unsupported-land right away |
| 18:09.24 | Katty | my experience with freepbx is awful, at best. |
| 18:09.34 | Naikrovek | it is certainly good at what it does |
| 18:09.34 | Katty | anything with the world freepbx now turns on RED WARNING LIGHTS |
| 18:09.50 | Katty | if ya'll think it's stable then that's fine |
| 18:09.52 | Naikrovek | it just doesn't do everything that the more discerning phone system admin would require |
| 18:09.55 | Katty | but i wouldn't touch it with a 10ft pole |
| 18:09.56 | seanbright | well anecdotal stories aside |
| 18:10.20 | p3nguin | If y'all think it's unstable, you'd be mistaken. |
| 18:10.20 | Katty | anecdotal stories aside, i have not used asterisknow |
| 18:10.26 | seanbright | wonderful |
| 18:10.27 | seanbright | thanks |
| 18:10.33 | Katty | seanbright wins arguement. |
| 18:10.41 | Katty | Katty 0 ~ sean 1 |
| 18:10.41 | Bokhuval | is gonna go rummage around and do some builds, thanks for the info everyone! |
| 18:11.12 | Katty | i hope he doesn't use freepbx at a doctor's office :< |
| 18:11.20 | Katty | what if it breaks :< |
| 18:11.24 | *** join/#asterisk stix_ (~stix@212.99.255.54) |
| 18:11.24 | Naikrovek | well |
| 18:11.27 | Katty | he won't know how to fix it :<<< |
| 18:11.33 | Katty | meanwhile someone's appendix is rupturing!!! |
| 18:11.33 | Naikrovek | no doctor's office is going to have its own IT guy |
| 18:11.41 | *** part/#asterisk asteriskATmarmuD (~mundt@193.158.65.23) |
| 18:11.42 | seanbright | whenever i see "never used it" and "unstable" in the same breath, it turns on RED FUD LIGHTS |
| 18:12.15 | Kobaz | haha |
| 18:12.23 | p3nguin | If he chooses to not use FreePBX in a doctor's office and the system breaks, he's still in the exact same position. |
| 18:12.26 | Katty | if i were setting up a doctor's office i think i'd use a completely hosted solution |
| 18:12.36 | Naikrovek | Katty: my point exactly |
| 18:12.47 | Katty | no |
| 18:12.47 | Naikrovek | Katty: you don't see IT closets in doctor offices |
| 18:12.50 | Katty | i don't think you get it |
| 18:12.57 | mrtelnet | [TK]D-Fender: http://pastebin.com/693xGiGk (dropping audio after transfer) |
| 18:13.04 | Kobaz | Naikrovek: we set up it closets for the doctors we've set up |
| 18:13.06 | Katty | i mean like put the phone in their office and have it point to some server farm in chicago |
| 18:13.10 | p3nguin | I've installed many a server for doctors' offices. |
| 18:13.12 | Naikrovek | Kobaz: really... |
| 18:13.16 | Kobaz | Katty: and when the internet goes out? |
| 18:13.18 | Katty | their server goes down, atlanta takes over |
| 18:13.22 | seanbright | because the network connection never goes down |
| 18:13.27 | Katty | cellphones |
| 18:13.28 | seanbright | this is a doctor's office after all |
| 18:13.30 | seanbright | they are high tech |
| 18:13.32 | Kobaz | Katty: heh |
| 18:13.36 | Katty | I AM SKEERED |
| 18:13.44 | Naikrovek | Kobaz: i guess the doctors here are more associated with hospitals here, all the IT rooms here are in the hospitals, and they have dedicated IT staff for those giant places |
| 18:13.48 | Kobaz | Katty: what about incoming calls? |
| 18:13.51 | seanbright | my doctor's office still has the carbon paper credit card processing system |
| 18:13.57 | Katty | errr cellphones |
| 18:14.06 | Katty | yeah well credit card's a bit different |
| 18:14.14 | Katty | it's not quite as bad as rupturing appendix |
| 18:14.17 | seanbright | but behind the scenes, they must have redundant high speed internet connectivity |
| 18:14.20 | [TK]D-Fender | mrtelnet: canreinvite=no <- should be under [general] as should "nat=yes" |
| 18:14.21 | Kobaz | Katty: so they'll have to get online to set up call forwareding from their now not-working internet connection? |
| 18:14.29 | seanbright | i just know it |
| 18:14.34 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 18:14.38 | Katty | Kobaz: erm surely the providor can forward their calls to another number |
| 18:14.42 | seanbright | i haven't seen their internet connections... but i would guess they are unstable |
| 18:14.44 | Katty | Kobaz: if this is a hosted solution |
| 18:14.51 | Kobaz | Katty: or they could have a local phone system, and not have to worry about that stuff at all |
| 18:15.04 | [TK]D-Fender | mrtelnet: Also, what do you have forwarded to your *? |
| 18:15.08 | Kobaz | unless the power went out, which your kinda screwed anyway |
| 18:15.20 | Katty | we just need telepathy |
| 18:15.25 | seanbright | Kobaz: all doctor's offices have redundant power supplies and backup generators |
| 18:15.32 | Katty | send brain waves towards the hospital when the appendix starts to rupture |
| 18:15.33 | seanbright | everyone knows that |
| 18:15.35 | Kobaz | seanbright: not the ones around here |
| 18:15.36 | Kobaz | heh |
| 18:15.49 | p3nguin | What does all that have to do with Asterisk, anyway? |
| 18:15.54 | Kobaz | the only customer we have that has a backup generator is an answering service |
| 18:16.02 | Katty | since when do we talk about asterisk in here |
| 18:16.10 | Katty | don't be redonkulus |
| 18:16.23 | Katty | Kobaz: that's most unfortunate |
| 18:16.33 | Katty | Kobaz: medical offices should be required to have backup generators |
| 18:16.39 | seanbright | wow |
| 18:16.43 | p3nguin | Here, nursing homes and clinics have generators, but Dr. Joe's office doesn't. |
| 18:16.47 | seanbright | this conversation has jumped the shark |
| 18:16.47 | Kobaz | actually we just did set up a small call center in a hospital... they have generators actually |
| 18:16.54 | Katty | that's good :> |
| 18:16.59 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 18:17.03 | Naikrovek | hospitals should |
| 18:17.06 | Katty | yes |
| 18:17.11 | Katty | cause if you have someone on life support |
| 18:17.17 | Katty | and the hospital has no generator |
| 18:17.20 | Katty | :<<<<<<< |
| 18:17.20 | Naikrovek | heh |
| 18:17.28 | p3nguin | _________________________________________ |
| 18:17.33 | Katty | ^- exactly |
| 18:17.58 | Katty | ^- lawsuit |
| 18:18.31 | Naikrovek | howcome UNPLUGGING someone is playing god? isn't plugging them in playing God? |
| 18:18.43 | p3nguin | yep |
| 18:18.48 | Katty | let's not drag god into it |
| 18:18.53 | seanbright | how about that asterisk open source telephony platform? |
| 18:19.03 | seanbright | let me ask a question... asterisk: |
| 18:19.04 | Naikrovek | seanbright: god made asterisk |
| 18:19.04 | Katty | it's you know....okay i guess |
| 18:19.06 | Naikrovek | duhhhhhh |
| 18:19.07 | seanbright | have you used it? |
| 18:19.11 | Katty | nope |
| 18:19.13 | Naikrovek | seanbright: yes |
| 18:19.13 | seanbright | and if not, is it unstable? |
| 18:19.16 | Katty | i'm just here for the stimulating conversation |
| 18:19.24 | Katty | with seanbright |
| 18:19.40 | Katty | it is the hilight of my afternoon |
| 18:20.45 | p3nguin | If St. Louis is the gateway to the west, what about the people that are already in the west? Wouldn't it be the gateway to the east for those people? |
| 18:21.11 | *** part/#asterisk DagMoller (~aguirre@unaffiliated/dagmoller) |
| 18:21.40 | Naikrovek | p3nguin: if that were the only way from west to east |
| 18:21.50 | Naikrovek | st. louis is a little nutty with their gateway thing |
| 18:21.59 | p3nguin | But it's not really the only way from the east to the west. |
| 18:22.03 | Naikrovek | you can't even drive under the arch |
| 18:22.13 | p3nguin | You can if you have a Hoverround. |
| 18:22.14 | Naikrovek | was there a few days ago |
| 18:22.27 | Katty | the arch is overrated |
| 18:22.31 | Naikrovek | yes |
| 18:22.34 | Katty | much better things to do in stl |
| 18:22.50 | p3nguin | Like have bread and/or spaghetti? |
| 18:22.54 | Katty | no |
| 18:22.58 | Naikrovek | the sexateria for one |
| 18:23.03 | Katty | like feeding the fish at the botanical gardens |
| 18:23.10 | Katty | lost a ring in there once :< |
| 18:23.13 | Katty | fell right off my hand. |
| 18:23.15 | seanbright | or going to starbucks |
| 18:23.23 | seanbright | just... awe inspiring |
| 18:23.34 | Katty | japanese festival at botanical gardens is lots of fun. |
| 18:23.42 | Katty | complete with sumo wrestlers n everything |
| 18:23.49 | seanbright | japanese guy at starbucks is wonderful as well |
| 18:23.58 | seanbright | go now, thank me later. |
| 18:24.09 | *** join/#asterisk Tim_Toady (~moi@77.49.45.81.dsl.dyn.forthnet.gr) |
| 18:24.28 | Katty | i also liked feeding the pgymy goats at grants' farm, but i think they're closed now |
| 18:24.39 | seanbright | they've move them all over to the starbucks |
| 18:24.43 | seanbright | crazy i know |
| 18:24.47 | Katty | :< |
| 18:24.53 | Katty | so THAT"S what happened. |
| 18:25.14 | Naikrovek | use them for milk |
| 18:25.43 | *** join/#asterisk Z_God (~julius@wlan225137.mobiel.utwente.nl) |
| 18:25.52 | Naikrovek | i heard them and smelled maneur. then i noticed goat poop. get it? coffee = maneur |
| 18:26.05 | Naikrovek | ... |
| 18:26.11 | Naikrovek | well i thought it was funny |
| 18:26.23 | Naikrovek | because coffee = nasty? ah nevermind |
| 18:26.48 | seanbright | coffee isn't nasty |
| 18:26.54 | Naikrovek | i disagree |
| 18:27.08 | seanbright | well you're entitled to your opinion |
| 18:27.13 | seanbright | as wrong as it is |
| 18:27.14 | hardwire | ~broadvoice |
| 18:27.15 | infobot | extra, extra, read all about it, broadvoice is Follow the config instructions at http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup but also beware: http://www.broadvoice.com/president_msg.html |
| 18:27.16 | seanbright | :P |
| 18:27.28 | Naikrovek | hehe |
| 18:27.28 | Katty | they see seanbright trollin |
| 18:27.29 | Katty | they hatin |
| 18:27.39 | Katty | does a lil dance |
| 18:27.40 | Naikrovek | i never understood why people dump scalding hot liquid down their mouths |
| 18:27.44 | mrtelnet | [TK]D-Fender: Thanks for the reccomendation, it appears that if any member puts the call on hold and takes it off, the issue goes away? Does this mean anything to you? |
| 18:27.45 | hardwire | anybody have a cached copy of the links above? |
| 18:27.47 | Naikrovek | down their throats, rather |
| 18:27.53 | Katty | Naikrovek: idk |
| 18:28.02 | seanbright | Naikrovek: you can let it cool off first |
| 18:28.03 | Katty | Naikrovek: sometimes i do that with hot cocoa |
| 18:28.05 | hardwire | archive.org ftw |
| 18:28.22 | Naikrovek | seanbright: it still stinks and tastes like poo when cold |
| 18:28.28 | seanbright | fair enough |
| 18:28.37 | seanbright | i personally have never tasted poo |
| 18:28.41 | seanbright | but i will take your word for it |
| 18:28.49 | Naikrovek | i've tasted it involuntarily |
| 18:28.52 | Katty | i'm gonna start callin you trollface |
| 18:28.55 | Naikrovek | not pleasant |
| 18:29.10 | [TK]D-Fender | mrtelnet: First I'd undo your templating attempts and split this normally. NExt I'd want a sample callw ith SIP debug and the cleaned configs |
| 18:29.11 | seanbright | good times. |
| 18:30.28 | hardwire | broadvoice is saying I'm registering too early to their 60 second expirey |
| 18:30.29 | hardwire | sigh |
| 18:30.39 | hardwire | they didn't define their boundary |
| 18:30.55 | hardwire | just that they don't have 20 million users calling them so it's my problem |
| 18:30.59 | Katty | would a hug make it better? |
| 18:31.29 | Naikrovek | temporarily |
| 18:32.21 | hardwire | make that 30 seconds |
| 18:32.52 | *** join/#asterisk mpe (~mpe@0x4dd624b2.adsl.cybercity.dk) |
| 18:34.45 | p3nguin | Isn't it fairly simple to NOT register so often? |
| 18:35.03 | Katty | that doesn't mean you can't complain about it |
| 18:35.27 | hardwire | p3nguin: the force the expiration |
| 18:35.27 | p3nguin | I would fix it, but I would continue complaining only if it was a huge bother. |
| 18:35.48 | hardwire | apparently the result that I get from them.. even though it's a 200 OK.. isn't enough to stop asterisk from reregistering several seconds later |
| 18:36.20 | p3nguin | Don't they have other people using Asterisk successfully? |
| 18:36.32 | hardwire | they claim to |
| 18:36.46 | hardwire | I'm sure they do. We switched from asterisk 1.2 to 1.6 and crap hit the fan |
| 18:36.54 | Katty | :< |
| 18:36.57 | hardwire | but even beofre that we had similar issues |
| 18:37.00 | Naikrovek | quite a jump |
| 18:37.27 | hardwire | Naikrovek: I just wish they had appropriate documentation for Asterisk users. |
| 18:37.44 | Katty | bummer. |
| 18:37.46 | hardwire | if they are aware of fixups to help interoperate with their platform.. We'd like to know |
| 18:38.12 | p3nguin | Lots of providers offer some asterisk configurations, but I haven't seen a single one that is 100% accurate. |
| 18:38.19 | hardwire | yar |
| 18:38.26 | Katty | at least it's somethin (= |
| 18:39.01 | Naikrovek | hardwire: well, once you get it working well you can give them the documentation and they can publish it to help others |
| 18:39.16 | p3nguin | I bet they won't, though. |
| 18:39.22 | Naikrovek | yeah |
| 18:39.26 | Naikrovek | i bet they won't too |
| 18:39.30 | *** join/#asterisk DennisG (~DennisG@84.30.136.208) |
| 18:39.36 | Katty | well he can blog it, so it's searchable from google |
| 18:39.38 | p3nguin | Because most people seem to not give a shit about service. |
| 18:39.43 | Naikrovek | Katty: even better |
| 18:39.54 | *** join/#asterisk stix (~stix@80.72.152.153) |
| 18:40.43 | hardwire | I should try registering from openser to see if I really can isolate it to asterisk |
| 18:41.00 | p3nguin | or use a soft phone. |
| 18:41.09 | hardwire | I'd rather not since those are all business calls |
| 18:41.12 | hardwire | very busy lines |
| 18:41.20 | hardwire | I can assuredly route it using a proxy |
| 18:43.19 | hardwire | https://issues.asterisk.org/view.php?id=15052 |
| 18:43.23 | hardwire | that's what I may be up against |
| 18:43.28 | hardwire | lol @ the 3rd comment |
| 18:46.26 | Naikrovek | 666@evil |
| 18:51.49 | *** join/#asterisk voipmonk (~shido6@dsl-67-204-1-83.acanac.net) |
| 18:52.10 | Brack10 | Hey is it possible to trunk video calls over an IAX2 trunk? |
| 18:52.36 | DennisG | Brack10, not that i know |
| 18:52.56 | DennisG | so far as i know, video support is a SIP feature. but don't blame me if i'm wrong |
| 18:53.03 | Brack10 | K |
| 18:53.26 | Brack10 | so with that said, best way to integrate dialplan between multiple * boxes is with dundi then? |
| 18:53.29 | *** join/#asterisk pabelanger (~pabelange@yoda.kanatek.com) |
| 18:53.39 | Brack10 | with sip trunks |
| 18:53.39 | DennisG | ooh i'm wrong haha |
| 18:53.40 | DennisG | http://www.voip-info.org/wiki/view/Asterisk+video |
| 18:55.14 | DennisG | but i have only tried it with SIP |
| 18:55.46 | DennisG | and that works great but you get a lot more traffic over your (fiber)lines |
| 18:56.08 | Brack10 | Yeah. I'm planning on a pretty limited video deployment |
| 18:56.16 | DennisG | nice :) |
| 18:56.17 | *** join/#asterisk benngard (~benngard@90-230-92-67-no148.tbcn.telia.com) |
| 18:56.29 | Brack10 | just a few important peeps and some conference rooms |
| 18:56.32 | [TK]D-Fender | Brack10: Passing a call is passing a call. No need for DUNDi |
| 18:56.33 | DennisG | i have only used it in a MAN environment |
| 18:57.07 | DennisG | D-Fender, he want to use multiple boxes ;) |
| 18:57.48 | [TK]D-Fender | DennisG: So? |
| 18:58.17 | hardwire | ahha |
| 18:58.18 | Brack10 | [TK]D-Fender: I thought dundi is for sharing dialplans between servers |
| 18:58.20 | hardwire | I found a/the problem |
| 18:58.26 | hardwire | broadvoice isn't asking me to authenticate at all |
| 18:58.43 | hardwire | if I've registered in the past it only wants a ping from me.. so I never get back "unauthorized - please do something about that" |
| 18:58.49 | [TK]D-Fender | Brack10: For the bits that he probably needs there is no need for dundi. |
| 18:59.10 | hardwire | so asterisk counteracts with "cmoooooooooon let me send you my fancy auth bits." |
| 18:59.13 | hardwire | a lot |
| 18:59.20 | [TK]D-Fender | DennisG: make a peer, pass a call. uses dialplan. big deal. When do you not know the pattern you want to reach teh otehr side as? |
| 19:00.06 | Katty | teh otehr |
| 19:00.07 | Brack10 | [TK]D-Fender: Oh I get it. I think I was misreading its purpose |
| 19:00.32 | Brack10 | [TK]D-Fender: As long as I know the dialplan on the other side it's not necessary? |
| 19:00.33 | [TK]D-Fender | DennisG: stop using the channel-notice msg. Just say it in channel |
| 19:00.52 | [TK]D-Fender | DennisG: DUNDi is NOT "load-balancing" |
| 19:01.00 | [TK]D-Fender | DennisG: It is a search order, nothing more. |
| 19:01.20 | [TK]D-Fender | DennisG: And you STILL need to AUTH calls you pass via it |
| 19:01.28 | DennisG | Yeah oke but you can use/misuse that feature :) |
| 19:01.39 | DennisG | yes that true d-fender :) |
| 19:01.51 | DennisG | and better know without the notice? :P |
| 19:02.03 | shader | anyone use skype-for-asterisk? |
| 19:02.25 | DennisG | no sorry shader, i only use hardphones and x-lite as softphone |
| 19:02.53 | shader | well, I was thinking of the servie provision side, but thanks |
| 19:02.55 | *** part/#asterisk jelly-bean (~jelly-bea@75-148-103-190-Utah.hfc.comcastbusiness.net) |
| 19:02.58 | shader | *service |
| 19:03.07 | shader | DennisG: what hardphones do you use? |
| 19:06.33 | [TK]D-Fender | shader: There is a cost/channel, and you can't use unlimite accounts with it |
| 19:06.49 | *** join/#asterisk Bokhuval (~user@cpe-70-112-22-94.austin.res.rr.com) |
| 19:09.49 | mrtelnet | Can anyone reccomend paid support? |
| 19:10.20 | mrtelnet | *Can anyone reccomend (a good company that offers) paid support? |
| 19:10.20 | hardwire | I recommend paying somebody.. sure. |
| 19:10.21 | *** join/#asterisk bn-7bc (bjarne@mac.lan.noare-1.holmedal.net) |
| 19:10.27 | Bokhuval | I've installed asterisk debian packages, connected via asterisk -r, and 'http show status' says it should be there. but can't connect to 8080 and netstat on the system shows that nothing is listening there. ideas? |
| 19:10.27 | Qwell | mrtelnet: http://store.digium.com/products.php?category_id=93 |
| 19:10.28 | hardwire | digium offers paid support. |
| 19:11.16 | shader | [TK]D-Fender: is it a monthly fee per channel? I can't find pricing info online |
| 19:11.22 | hardwire | should registration attemps always be followed up with a 401 and then a 200? |
| 19:11.35 | Qwell | shader: SFA is a one-time license, per channel |
| 19:12.18 | *** join/#asterisk benngard (~benngard@90-230-92-67-no148.tbcn.telia.com) |
| 19:12.58 | joesuffceren | shader: in case you don't know, SFA is per minute use only. You can't use any of the "plans |
| 19:13.07 | joesuffceren | that Skype offers |
| 19:13.59 | seanbright | buys a L4 3 year support contract |
| 19:14.06 | seanbright | Qwell: should i just make the check out to you? |
| 19:14.08 | bmoraca_work | interesting...adtran gives an automatic 10% hit to the "quality rating" of a phone call simply by using g729 |
| 19:14.16 | Qwell | seanbright: C.A.S.H., please |
| 19:14.33 | shader | joesuffceren: I'm aware |
| 19:14.33 | seanbright | k |
| 19:14.37 | Qwell | that stands for "per-Case Asterisk Support Hotline" |
| 19:14.43 | seanbright | heh |
| 19:14.53 | shader | I was hoping to use it for international calls, and use an unlimited sip trunk for domestic |
| 19:15.08 | shader | just looking at options really |
| 19:16.39 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 19:17.32 | *** join/#asterisk magronez (~eusei@unaffiliated/magrao/x-2903) |
| 19:18.28 | shader | can anyone recommend a sip trunk provider with support for T.38, international calls, and DIDs? |
| 19:19.20 | DennisG | what is your SIP traffic per month ? |
| 19:19.35 | DennisG | like 10 calls per month or more like 10K calls per month ? |
| 19:19.37 | *** join/#asterisk authorized (~asdfg@206.173.193.56.ptr.us.xo.net) |
| 19:19.47 | shader | low |
| 19:19.58 | Bokhuval | I haven't been able to find any useful messages (errors, 'file/directory does not exist', anything) in any logfiles either. Wondering how asterisk can think it's got an open http port when it obviously doesn't. |
| 19:20.09 | [TK]D-Fender | [15:19]<shader>low <- is that metric, or imperial? |
| 19:20.14 | shader | lol |
| 19:20.17 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 19:20.36 | DennisG | haha :P |
| 19:21.11 | DennisG | try voipbuster? or try a lot of different voip providers with a postpaid account of 5 dollars and see which is the best? |
| 19:21.19 | *** join/#asterisk githogori (~githogori@SJC-Office-DHCP-135.mail-abuse.org) |
| 19:21.25 | shader | it's probably 10-15 calls a day |
| 19:21.37 | shader | maybe more |
| 19:21.37 | p3nguin | bokhuval: Why are you even bothering with the asterisk gui? |
| 19:21.37 | DennisG | ooh oke |
| 19:21.57 | shader | 11 people, but low traffic |
| 19:22.06 | shader | so probably 2-3 trunks |
| 19:22.16 | shader | plus 2 fax machines |
| 19:22.54 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 19:23.10 | Bokhuval | p3nguin: because it's there and that's what the setup instructions said to do? :P |
| 19:23.19 | p3nguin | I can believe that. |
| 19:23.40 | p3nguin | My advice to you is to forget about it for now, then go back and shut it down later. |
| 19:24.11 | *** join/#asterisk X-TaZ (~X-TaZ@78.242.36.126) |
| 19:26.12 | Bokhuval | I can't shutdown what isn't actually open, but whatever. I've got asterisk installed because I would like to learn a bit about it before actually getting a paid trunk service. Apparently getting into the GUI to glance around at things is out, so are there any suggestions on what I should look at/poke through next? |
| 19:26.27 | Qwell | ~book |
| 19:26.28 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 19:26.33 | [TK]D-Fender | Bokhuval: What GUI? Installed how? |
| 19:27.09 | *** join/#asterisk niekvlessert (~niek@5ED25657.cable.ziggo.nl) |
| 19:27.15 | *** part/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 19:27.19 | Bokhuval | [TK]D-Fender: 'apt-get install asterisk' /etc/asterisk/http.conf - enabled=yes port=8080, asterisk -r 'http show status' says it's running. |
| 19:27.30 | Bokhuval | but it's not :P |
| 19:28.02 | [TK]D-Fender | Bokhuval: Well so far... you don't HAVE a GUI. |
| 19:28.08 | p3nguin | asterisk-gui |
| 19:28.39 | [TK]D-Fender | Bokhuval: The GUI that Digium made using that framework isn't part of that package you installed |
| 19:28.55 | [TK]D-Fender | Bokhuval: It is completely separate add-on, and isn't even being maintained |
| 19:29.23 | [TK]D-Fender | Bokhuval: If you are looking to evaluate * for your use... this is not the means by which you should be doing so |
| 19:29.41 | Bokhuval | oy. *shakes head* not even going to ask why. So, repeat of last question - sans-gui, what should I look through next? |
| 19:29.51 | Qwell | ~book |
| 19:29.52 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 19:29.52 | p3nguin | THE BOOK |
| 19:29.54 | [TK]D-Fender | ^^^ |
| 19:30.00 | [TK]D-Fender | Bokhuval: It |
| 19:30.06 | [TK]D-Fender | Bokhuval: Its be said coutnless time |
| 19:30.12 | [TK]D-Fender | Bokhuval: Its be said countless tiems |
| 19:30.14 | [TK]D-Fender | gah |
| 19:30.27 | p3nguin | I'll fix it: It's been said countless times. |
| 19:30.41 | Bokhuval | chuckles and passes a cookie "It's been a long day and it's barely past lunch." |
| 19:30.56 | Bokhuval | Looks like I'll be heading down to B&N then.. |
| 19:31.07 | LemensTS | . |
| 19:31.16 | p3nguin | Is something wrong with your browser? |
| 19:31.25 | Qwell | many people prefer actual books.. |
| 19:31.37 | p3nguin | Suit yourself. |
| 19:31.54 | Bokhuval | No, but I love this marvelous piece of technology called a book - it's delightfully intuitive :] |
| 19:32.14 | Qwell | p3nguin: besides, leifmadsen gets to put nickels in his jar every time |
| 19:32.14 | p3nguin | I read the book online so I didn't have to go to B&N. |
| 19:32.46 | p3nguin | I've read it in PDF and HTML. |
| 19:32.52 | Bokhuval | I telecommute, getting to leave the house during the day for some fresh air would be nice. |
| 19:32.59 | p3nguin | :) |
| 19:33.24 | leifmadsen | Bokhuval: thanks for sponsoring my nickle jar. Perhaps some day I'll save up enough nickles to update the book and release it for free again. |
| 19:33.33 | Bokhuval | lol |
| 19:33.41 | Naikrovek | i want an e-reader for the asterisk book but i don't know of any current ereaders that support pdf well enough to display it properly |
| 19:34.18 | Bokhuval | I'll likely get a digital copy at some point too, but for now I like having something in my hands. |
| 19:34.34 | Qwell | leifmadsen: sign my pdf! |
| 19:34.45 | [TK]D-Fender | revokes Qwell's DRM |
| 19:35.54 | leifmadsen | Qwell: gpg --armor --output tfotv2.pdf.sig --detact-sig tfotv2.pdf |
| 19:36.23 | Qwell | sells it on eBay |
| 19:38.08 | hardwire | hmm.. 6 minutes of registration attempts always resolving to a 200 |
| 19:38.11 | hardwire | no auth |
| 19:38.15 | hardwire | no 401 |
| 19:38.17 | hardwire | this is annoying |
| 19:41.37 | hardwire | interesting. |
| 19:41.55 | hardwire | anybody experienced a problem where the VoIP registrar returns an answer too quickly? |
| 19:42.12 | hardwire | I'm seeing a 200 back nearly immediately.. but asterisk is logging a timeout and a retry attempt |
| 19:42.17 | hardwire | the tags line up |
| 19:42.37 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.227.117.dsl.dyn.forthnet.gr) |
| 19:44.21 | hardwire | interesting.. the call sequence number keeps getting reused. |
| 19:44.27 | hardwire | arghs a little |
| 19:49.26 | bmoraca_work | you know, customers who are morons and can't seem to do simple things are bad enough...but when it's a colleague who is also a tech, it just pisses me off like crazy! |
| 19:49.28 | *** join/#asterisk benngard (~benngard@90-230-92-67-no148.tbcn.telia.com) |
| 19:49.41 | niekvlessert | hello! Can I do a call pickup from a call in a queue with no agents? |
| 19:50.29 | seanbright | niekvlessert: #asterisk-dev |
| 19:50.32 | seanbright | (just kidding) |
| 19:50.34 | niekvlessert | lol |
| 19:50.40 | benngard | ofc i do it everey day |
| 19:50.58 | bmoraca_work | niekvlessert: probably. my experience with call pickup is that it wants to pick the call up based on the original extension and context, not necissarily where it's currently ringing |
| 19:53.36 | niekvlessert | benngard & bmoraca_work: thanks |
| 19:53.49 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-147.cablep.bezeqint.net) |
| 19:54.17 | niekvlessert | another one: can I do barge in with Asterisk? Like the real barge in, so the secretary can talk to the boss and that the client isn't hearing it |
| 19:54.35 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 19:55.06 | [TK]D-Fender | niekvlessert: ""sore show application chanspy |
| 19:55.13 | [TK]D-Fender | niekvlessert: core show application chanspy |
| 19:55.38 | bmoraca_work | yes, however it will likely take some outside influence (AGI or AMI) to do well, and you're not going to find a canned solution...depending, of course, on the exact details you want |
| 19:56.04 | seanbright | chanspy with whisper mode |
| 19:56.19 | leifmadsen | aye |
| 19:56.23 | niekvlessert | ah! very useful |
| 19:56.38 | niekvlessert | was looking into meetme and stuff |
| 19:56.52 | leifmadsen | reading through the application documentation is VERY useful |
| 19:57.00 | leifmadsen | you can find all sorts of cool features |
| 19:57.46 | niekvlessert | i know i know, but somehow google didn't show me this one |
| 19:58.07 | niekvlessert | and i seem to remember it wasn't possible, it's in the dcap exam, but I think that is with meetme |
| 19:59.19 | niekvlessert | i did read the 1.8 coming up stuff though :) at least a quick read |
| 19:59.23 | benngard | and reading the source code could be vere usefull to ;) |
| 20:00.05 | *** join/#asterisk sleazye (~ernesto@64-71-24-110.static.wiline.com) |
| 20:00.27 | benngard | is not a programmer but have learnt alot from it |
| 20:00.31 | *** part/#asterisk sleazye (~ernesto@64-71-24-110.static.wiline.com) |
| 20:00.47 | niekvlessert | benngard: I've been trying to adapt patches, then you learn, true |
| 20:01.46 | benngard | niekvlessert: guess so, i am a better tester then programmer |
| 20:02.26 | niekvlessert | another quick question which I couldn't find the answer for; is it true that using AMI to originate calls CDR information is missing? |
| 20:02.28 | niekvlessert | in 1.4 |
| 20:02.52 | niekvlessert | benngard: me neither, I'm more like a software engineer without the patience for coding ;) |
| 20:04.54 | benngard | niekvlessert: about cdr info from originate a call from ami, think it is even in trunk, let me check, sec |
| 20:05.12 | benngard | it is even not in |
| 20:05.16 | benngard | in* |
| 20:05.24 | benngard | spells like shit |
| 20:06.33 | benngard | niekvlessert no cdr when u originate a call from AMI, at least not in trunk |
| 20:06.54 | niekvlessert | benngard: thanks! we have a switchboard that does that, so that's bad |
| 20:07.02 | Katty | dialplan reload includes extensions.conf, right? |
| 20:07.15 | Katty | it would seem silly not to |
| 20:07.27 | niekvlessert | could I simulate a way in the dialplan maybe... |
| 20:07.40 | bmoraca_work | dialplan reload IS extensions.conf |
| 20:08.27 | bmoraca_work | hey Katty, do you like shrimp scampi? i found a really good recipe for it yesterday. interested? |
| 20:08.38 | Katty | are you kiddin me |
| 20:08.41 | Katty | course i'm interested |
| 20:08.44 | Katty | send it to me -> |
| 20:08.58 | hardwire | sweet.. apparently I'm blacklisted now from broadvoice.. |
| 20:09.05 | hardwire | at least one particular registration server |
| 20:09.07 | Katty | :< |
| 20:09.18 | bmoraca_work | Katty: http://www.foodnetwork.com/recipes/food-network-kitchens/shrimp-scampi-recipe/index.html ....wuper simple and easy. |
| 20:09.41 | *** join/#asterisk xpot-mobile (~xpot@66.60.101.91) |
| 20:09.41 | bmoraca_work | Katty: i dropped the shrimp to 1lb and mixed it with some cooked linguini...it was great |
| 20:09.52 | niekvlessert | benngard: what about transfer and things like that, will it update stuff? |
| 20:10.17 | Katty | reads |
| 20:10.32 | Katty | i'm guessing vermouth is a kind of wine? |
| 20:10.43 | bmoraca_work | next time i make it, though, i'm going to double or triple the butter, vermouth, garlic, and lemon juice so that i have more of the pan sauce...mixed with 1/2 lb of linguini, it ended up being a bit dry |
| 20:10.54 | bmoraca_work | yep, it's fortified, spiced white wine |
| 20:11.01 | benngard | niekvlessert: u mean if u transfer the call with AMI? |
| 20:11.02 | bmoraca_work | spiced/herbed |
| 20:11.04 | Katty | is it in the wine aisle? |
| 20:11.10 | Katty | or the vinegar aisle |
| 20:11.10 | niekvlessert | benngard: ye |
| 20:11.15 | bmoraca_work | usually in the booze aisle |
| 20:11.18 | Katty | k |
| 20:11.19 | benngard | let me check |
| 20:11.23 | bmoraca_work | you'll find it by the gin, i think |
| 20:11.23 | Katty | takes notes |
| 20:11.30 | benngard | just need asnother phone |
| 20:11.37 | Katty | did you use unsalted butter? |
| 20:11.44 | bmoraca_work | i used salted butter |
| 20:11.47 | Katty | k |
| 20:11.53 | Katty | did you take the tails off? |
| 20:12.00 | bmoraca_work | just adjust how much salt you put on the shrimp if you don't have unsalted handy |
| 20:12.03 | bmoraca_work | yes, tails came off |
| 20:12.07 | Micc | I set notifyringing=no in sip.conf in general and in individual sip accounts, but it still changes the state to ringing. |
| 20:12.09 | X-TaZ | Hi. I'm having trouble with trunking 2 asterisk servers with IAX2. |
| 20:12.15 | X-TaZ | I got this error : [Mar 15 21:11:32] WARNING[3307]: chan_iax2.c:7820 socket_process: Call rejected by 192.168.1.128: No authority found |
| 20:12.17 | Micc | Is notifyringing broken? |
| 20:12.19 | Katty | bmoraca_work: i have a habit of using already cooked shrimp |
| 20:12.30 | X-TaZ | My iax's files are at http://pastebin.com/gL5a88BB |
| 20:12.30 | bmoraca_work | Katty: don't in this dish...it won't taste right |
| 20:12.46 | X-TaZ | And my extensionsconf aare at http://pastebin.com/GgY05vZ8 |
| 20:12.52 | Katty | bmoraca_work: i'm not sure how to cook shrimp is why |
| 20:12.53 | bmoraca_work | even if you have to buy frozen, get raw shrimp. cooking in the butter and garlic is what gives the dish its flavor |
| 20:13.05 | Katty | bmoraca_work: my parents never liked shrimp... so i never really learned that one |
| 20:13.05 | niekvlessert | benngard: https://issues.asterisk.org/view.php?id=12007 |
| 20:13.06 | *** join/#asterisk korihor (~korihor@201.210.226.98) |
| 20:13.30 | bmoraca_work | Katty: just follow the instructions in the recipe. it's way easy. once they're orange, they're done. they go quick, too, like 4 minutes total |
| 20:13.33 | X-TaZ | i spent hours searching for it, now i ask there |
| 20:13.52 | bmoraca_work | Katty: if you use pre-cooked, they'll get OVERcooked and will be mushy. use raw and they'll be nice and crisp |
| 20:14.42 | bmoraca_work | i also used dried parsley cause i didn't have any fresh |
| 20:14.55 | niekvlessert | X-Taz, I see something |
| 20:15.01 | X-TaZ | Really ? |
| 20:15.17 | Micc | how can I get blf lights not to flash for ringing phones? I think people only want to know if someone is on the phone, not to see all the phones in the office ringing at once. |
| 20:15.20 | niekvlessert | when you first do exten => _XXX,1, this is the first action |
| 20:15.30 | hardwire | hrm.. sipsak can register and forward right? |
| 20:15.43 | niekvlessert | but only exten => _XXX,2, is the second action concerning this |
| 20:15.58 | niekvlessert | exten => _2XX has to start with a ,1 all over |
| 20:16.12 | X-TaZ | ok |
| 20:16.26 | Katty | bmoraca_work: i'm having a hard time following this wording. pat shrimp dry, place in pan cook until not foamy? |
| 20:16.32 | Katty | bmoraca_work: then flip and cook other side for 1 minute? |
| 20:16.36 | niekvlessert | right asterisk guys??? I don't do a lot of dialplan hacking :) |
| 20:17.04 | benngard | niekvlessert seems not to be working either |
| 20:17.37 | bmoraca_work | Katty: you know how when you melt butter in a pan it gets a bit of a white foam? once it's up to temperature, the butter will be mostly clear. that's when you add your shrimp. as far as patting it...they just want to make sure you're not adding excess water to your hot, melted butter (causes splatter) |
| 20:17.42 | benngard | but as i said I am running trunk (with posgres) so i could be wrong |
| 20:17.47 | niekvlessert | benngard: lol, it's 9 PM overhere, just enjoying my evening with a little hobby-ing :) |
| 20:17.48 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 20:18.01 | Katty | bmoraca_work: yeah i know about butter doing that...it's call clarifying... |
| 20:18.05 | benngard | same here |
| 20:18.07 | X-TaZ | I'm still having [Mar 15 21:17:34] WARNING[3297]: chan_iax2.c:7820 socket_process: Call rejected by 192.168.1.129: No authority found |
| 20:18.07 | X-TaZ | <PROTECTED> |
| 20:18.07 | X-TaZ | <PROTECTED> |
| 20:18.25 | [TK]D-Fender | Micc: mod the source. |
| 20:18.30 | Katty | bmoraca_work: i thought maybe the shrimp was supposed to do something |
| 20:18.32 | *** join/#asterisk ChrisWi (~admin@mx2.wwserver.net) |
| 20:18.53 | niekvlessert | X-TaZ: iax2 show peers? |
| 20:18.59 | bmoraca_work | Katty: nope...just sits there in butter and garlic until it's orange (flipped halfway through to get both sides). |
| 20:19.06 | Katty | k |
| 20:19.08 | niekvlessert | benngard: where u from? |
| 20:19.20 | benngard | sweden, gothenburg |
| 20:19.22 | X-TaZ | 200/Maurice 192.168.1.128 (D) 255.255.255.255 4570 Unmonitored |
| 20:19.22 | X-TaZ | server1 192.168.1.129 (D) 255.255.255.255 4569 OK (6 ms) |
| 20:19.23 | Micc | TKD-Fender, I thought thats what notifyringing was for. Can I just mod the source for notifyringing or is that for something else? |
| 20:19.46 | bmoraca_work | Katty: like I said, though...if you want to mix it with pasta, I'd double or triple the butter, garlic, vermouth, and lemon juice. if you want to eat the shrimp plain, then the amounts are fine |
| 20:20.25 | Katty | nods |
| 20:20.31 | bmoraca_work | i used 31ct shrimp...it would be probably better with 25ct or even less. i wouldn't use jumbo prawns though. |
| 20:20.35 | Katty | did you put any vegetables in there? |
| 20:20.35 | *** join/#asterisk sulex (~sulex@host-78-14-170-90.cust-adsl.tiscali.it) |
| 20:20.58 | niekvlessert | X-TaZ: where's your outgoing stuff? |
| 20:20.59 | bmoraca_work | Katty: nope. i had salad on the side. scampi shouldn't have veggies mixed in, though it does go well with steamed veggies |
| 20:21.08 | niekvlessert | try dialing voicemailmain at the other server first |
| 20:21.29 | niekvlessert | benngard: nice, NL, Utrecht overhere |
| 20:21.38 | X-TaZ | niekvlessert ? i just want my iax2 ring and can call I dont need voicemails |
| 20:21.51 | niekvlessert | benngard: finally, 6 degrees celcius |
| 20:21.55 | X-TaZ | my outgoing stuff ? in extensionsconf ? |
| 20:22.00 | niekvlessert | X-TaZ: yes |
| 20:22.32 | X-TaZ | [fromiax] is this thing, no ? |
| 20:22.36 | Katty | bmoraca_work: hrmmmmk |
| 20:22.45 | Katty | bmoraca_work: salad doesn't go over well here. |
| 20:22.50 | Katty | bmoraca_work: it's a chore to get him to eat his veggies |
| 20:24.01 | bmoraca_work | understandable. steamed broccoli would go well, i think...with baby carrots, i think. but you can be creative and try different things. |
| 20:24.02 | niekvlessert | http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/I_sect14_tt670.htm |
| 20:24.23 | niekvlessert | we should start a channel asterisk_food........ |
| 20:24.33 | niekvlessert | will be a big succes |
| 20:24.38 | bmoraca_work | lol |
| 20:24.46 | X-TaZ | This url is not found :x |
| 20:24.58 | bmoraca_work | anyway, i've got to go fix a colleague's screwup...be back later. |
| 20:25.01 | niekvlessert | sry htmL it is :) |
| 20:25.13 | X-TaZ | Thanks :p |
| 20:25.51 | Katty | niekvlessert: mmmyeah. you haven't been around long have you |
| 20:26.13 | [TK]D-Fender | checkout time, later all |
| 20:26.31 | niekvlessert | Katty: well, I'm hanging around in places where I need stuff for, asterisk-dev, xen, netfilter, stuff like that |
| 20:27.06 | joesuffceren | My TE121 issue was a bad piece of hardware. The hardware rev. I had was recalled, but my supplier didn't follow the recall. :-/ |
| 20:27.37 | Katty | they never do |
| 20:28.25 | Katty | you think toyota's gonna call every person in with the posibility of a sticking gas pedal? |
| 20:29.24 | *** join/#asterisk nightrid3r (kvirc@41.214.205.87) |
| 20:31.01 | Katty | ohai |
| 20:33.59 | Corydon76-lap | Katty: now it looks like Toyota has a fault in their little black boxes, too |
| 20:34.25 | *** join/#asterisk darkdrgn2k (~darkdrgn2@CPE000c419e662f-CM0011aea0fa16.cpe.net.cable.rogers.com) |
| 20:34.51 | Corydon76-lap | since they clearly did not record all of the events leading up to an unplanned acceleration event |
| 20:34.53 | darkdrgn2k | ok im tryiing to use ASterisk Call Manager to do a "click to call" |
| 20:34.57 | darkdrgn2k | i managed to get to this poin |
| 20:34.58 | darkdrgn2k | http://pastebin.ca/1841598 |
| 20:35.09 | darkdrgn2k | right now it alles the extension FIRST then the client |
| 20:35.11 | Katty | Corydon76-lap: also very unfortunate |
| 20:35.13 | darkdrgn2k | is there any way we can do it backwards |
| 20:35.19 | darkdrgn2k | (ie call the client FIRST then the extension) |
| 20:37.49 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 20:38.11 | *** join/#asterisk timeshell (~timeshell@gw.lusi.on.ca) |
| 20:38.27 | *** join/#asterisk stix (~stix@80.72.152.153) |
| 20:38.40 | *** join/#asterisk DennisG (~DennisG@84.30.136.208) |
| 20:39.06 | niekvlessert | Is toyata running windows? |
| 20:39.20 | darkdrgn2k | niekvlessert: windows me i beleave |
| 20:39.38 | Kobaz | i thought it was a combination of ce, me, and nt |
| 20:39.49 | Kobaz | it was like code named windows CeMeNt |
| 20:40.03 | darkdrgn2k | Kobaz: nice |
| 20:40.35 | Kobaz | perfect for all those odd do it yourself jobs |
| 20:40.39 | niekvlessert | clever, to combine the best windows editions |
| 20:41.02 | Kobaz | yeah i always thought windows cement was a great idea |
| 20:41.08 | darkdrgn2k | Kobaz: i hear they are gonna cut a hole in the bottom of the drivers side.. for flinstone like breaking |
| 20:41.11 | Kobaz | everything just sticks together so nicely |
| 20:41.32 | niekvlessert | it's like windows foundation, but a little different |
| 20:50.41 | *** join/#asterisk a1fa (~a1fa@unaffiliated/a1fa) |
| 20:50.43 | a1fa | hi |
| 20:52.05 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 20:54.44 | *** join/#asterisk Takapa (vegard@svanberg.no) |
| 20:56.00 | darkdrgn2k | ok figured that out |
| 20:56.16 | darkdrgn2k | how can i identify the status of the call that is made from the asterisk manager? |
| 20:56.44 | Kobaz | strange |
| 20:56.58 | Kobaz | one of my t1 links just died |
| 20:57.41 | Kobaz | oh, there's it's back |
| 20:59.41 | Chainsaw | Kobaz: Some miles away, a telco engineer tugged on the wrong plug. |
| 20:59.42 | a1fa | has anyone had their sip uplink account breached? |
| 21:00.31 | Kobaz | heh |
| 21:00.49 | Kobaz | Chainsaw: actually this is a t1 run that's between a pbx and a sangoma card, and it's 20 feet from me in the lab |
| 21:01.29 | *** join/#asterisk DJF5 (~email@84-105-183-83.cable.quicknet.nl) |
| 21:01.34 | freezey | any good doc to get dahdi to work with t1 digium carD? |
| 21:02.02 | a1fa | anyone had a recent breach on their uplink sip? |
| 21:02.14 | a1fa | i am trying to see if this is a isolated case |
| 21:02.28 | [TK]D-Fender | darkdrgn2k: Its a channel like any other |
| 21:02.44 | *** join/#asterisk andres833 (~andres833@190.144.75.22) |
| 21:07.32 | Tim_Toady | i want a fax setup that will allow the faxes to be received on the fax machine but at the same time keep a copy (tiff) for every arriving fax on the asterisk server the fax machine is connected to, any hints? |
| 21:08.50 | [TK]D-Fender | Tim_Toady: receive in *, then call your own fax machine and send to it |
| 21:09.33 | Tim_Toady | that was my idea too, but i thought it was a bit lame :D |
| 21:10.26 | Kobaz | so i added my dahdi checks for not indicating when already answered |
| 21:10.27 | Kobaz | er |
| 21:13.09 | *** join/#asterisk stix_ (~stix@80.72.152.153) |
| 21:13.23 | a1fa | lol |
| 21:13.25 | a1fa | i figured it out |
| 21:13.32 | a1fa | one of my extensions did not have a password |
| 21:14.21 | a1fa | lol |
| 21:15.17 | leifmadsen | burn |
| 21:15.31 | a1fa | :P |
| 21:15.33 | a1fa | not really |
| 21:15.57 | a1fa | i wonder why did they mass dial |
| 21:16.08 | a1fa | 1-509 area code |
| 21:20.23 | a1fa | crap |
| 21:20.29 | a1fa | they called people asking for credit card numbers |
| 21:20.55 | Naikrovek | yep |
| 21:20.58 | Naikrovek | that is how they roll |
| 21:21.25 | Naikrovek | gotta strengthen those passwords |
| 21:21.27 | *** join/#asterisk AlHafoudh (~AlHafoudh@icm7-orange.orange.sk) |
| 21:21.36 | Naikrovek | mine are all 16 chars of gibberish |
| 21:21.43 | Naikrovek | randomly generated |
| 21:21.55 | Naikrovek | no two extensions ahve the same passwd, etc. |
| 21:22.07 | *** join/#asterisk riddlebox (~riddlebox@75-132-225-75.dhcp.stls.mo.charter.com) |
| 21:23.09 | a1fa | yeah |
| 21:23.14 | a1fa | mine are the same way |
| 21:23.20 | a1fa | it was a debug account i used |
| 21:23.26 | Naikrovek | d'oh |
| 21:23.35 | a1fa | and forgot to disable that one day |
| 21:23.44 | a1fa | the scammers came from Korea |
| 21:23.48 | Naikrovek | i think there's a way to tell asterisk to lock a sip account out if there are multiple badd password attempts on it |
| 21:24.00 | a1fa | anyway to tell asterisk to reply "no peer found" |
| 21:24.03 | Naikrovek | but that wont' work if you don't have a password, those are the first attempts i think |
| 21:24.05 | a1fa | on any |
| 21:24.18 | Naikrovek | you can tell asterisk that that extension can only connect from a particular IP i think |
| 21:24.31 | Naikrovek | 'host=dynamic' is the way around it |
| 21:24.35 | Naikrovek | and is the default i believe |
| 21:24.52 | Naikrovek | 'host=123.123.123.123' would be more ideal if the ip is indeed static |
| 21:25.19 | Naikrovek | or you can toss a firewall on the machine and only allow connections from wherever is authorized if you know the netblocks |
| 21:26.09 | Naikrovek | i do that also heh |
| 21:26.30 | a1fa | asterisk -r |
| 21:26.32 | a1fa | sip show peers |
| 21:26.34 | a1fa | lol |
| 21:29.25 | *** join/#asterisk aandrade (~aandrade@187.58.239.123) |
| 21:29.36 | *** join/#asterisk mpe (~mpe@0x4dd624b2.adsl.cybercity.dk) |
| 21:32.19 | bmoraca_work | does anyone know how to check the time on a Cisco 7940 from the telnet interface? |
| 21:33.45 | *** join/#asterisk kotp (~vgoff@96.2.187.66) |
| 21:38.09 | mnick86 | hey, is there a way if a SIP client registers to asterisk, that a extension is beeing executed in the dialplan ? |
| 21:39.36 | Kobaz | mnick86: not that i know of... in order to run dialplan code you need an audio path... and registration is a control message, not a call |
| 21:39.40 | Kobaz | mnick86: what are you trying to do? |
| 21:41.30 | mnick86 | I am trying to check if a SIP client is registered or not |
| 21:42.42 | [TK]D-Fender | mnick86: You just asked 2 completely different things |
| 21:42.58 | Katty | i have a hypothetical question. |
| 21:43.15 | mnick86 | when a client registers to asterisk and a extension gets executed I could set a variable |
| 21:43.28 | [TK]D-Fender | mnick86: 1) Execute an extension if a SIP device registers. and 2) Just check if a device already has registered |
| 21:43.35 | [TK]D-Fender | mnick86: Which is it? |
| 21:44.09 | Katty | you call an IVR. intermittently on this IVR when you put in any 4 digit extension, it occasionally goes dead silent. Upon enabling said debug you find that the dtmf doesn't really show up right. After finding this information out you dial the IVR again and put in 4 digit extensions until it goofs up...you sit at the dead silence, then put in the last two digits of the 4 digit extension and BAM it dials. hypothetically, is that a dtmf problem or more |
| 21:44.33 | Katty | s/said/sip |
| 21:44.57 | *** join/#asterisk gardo (~gardo@125.212.88.237) |
| 21:46.01 | mnick86 | <[TK]D-Fender>: either one would work for me |
| 21:46.57 | Katty | http://pastebin.com/kJPbX5SC <- some debug log. |
| 21:47.36 | Katty | after line 10 silence. |
| 21:47.47 | Katty | after putting in 44 again at lines 11-14, RINGGGGGGGGG |
| 21:48.22 | freezey | any reason asterisk will start using safe_asterisk but not using the init.d script to start asterisk? cant find anywhere it is choking on either |
| 21:48.25 | *** join/#asterisk cosmicwombat (~cosmicwom@69.7.44.68) |
| 21:48.44 | Katty | freezey: can you manually kick off asterisk with the script |
| 21:49.13 | freezey | yeah i just run safe_asterisk |
| 21:49.14 | freezey | and it starts up |
| 21:50.07 | doneir | check /var/log/messages or the asterisk /var/log/asterisk/messages file for info on startup, perhaps running 'asterisk -cvvvvgd' may also help |
| 21:50.55 | Katty | freezey: i'm not sure if that answers the question that i asked. |
| 21:51.54 | Katty | freezey: when you run the script manually, what happens |
| 21:51.54 | freezey | if i run safe_asterisk it starts up no problem... but if i run /etc/init.d/asterisk start it fails on exit code 1 |
| 21:53.19 | *** join/#asterisk lanning (~lanning@208.87.235.224) |
| 21:53.22 | asteriskmonkey | freezy: type asterisk -vvvvvvvvvvvvvvvvvvc |
| 21:53.23 | [TK]D-Fender | mnick86: "sip show peer [theperr]" |
| 21:53.27 | asteriskmonkey | see what it kicks you out on |
| 21:53.34 | X-TaZ | niekvlessert : after tried to adapt the confs, i finally copied the whole configs on http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/I_sect14_tt670.html |
| 21:53.47 | niekvlessert | any luck then? |
| 21:53.49 | asteriskmonkey | freezey: might also be a scrupts/user permssion |
| 21:53.56 | freezey | asteriskmonkey: i am root |
| 21:54.00 | X-TaZ | Now i get a [Mar 15 22:52:16] NOTICE[3299]: chan_iax2.c:7642 socket_process: Rejected connect attempt from 192.168.1.128, request '2001@phones' does not exist |
| 21:54.06 | X-TaZ | :s |
| 21:54.06 | freezey | asteriskmonkey: yeah that starts it fine |
| 21:54.35 | asteriskmonkey | freezy : go to the /usr/src directory and go into your asterisk src folder and type make config to rebuilt the init file |
| 21:54.41 | Katty | freezey: and what does the log show after the exit? |
| 21:55.18 | Katty | freezey: it would be hilarious if it exited because asterisk was already running |
| 21:55.22 | X-TaZ | but i still have OK(6ms ) in the iax2 show peers |
| 21:55.44 | niekvlessert | can you give me ssh access? now i want to know... we can use screen |
| 21:55.58 | niekvlessert | so you can see what i'm doing |
| 21:56.21 | asteriskmonkey | freezey : try asterisk -r if you think its already running |
| 21:56.29 | freezey | i can execute the script directory from src and it works |
| 21:56.32 | freezey | just the init.d script doesnt |
| 21:56.35 | freezey | which isnt that big of a deal to me |
| 21:56.36 | *** part/#asterisk a1fa (~a1fa@unaffiliated/a1fa) |
| 21:56.39 | Katty | freezey: k, then get rid of it |
| 21:56.40 | *** join/#asterisk a1fa (~a1fa@unaffiliated/a1fa) |
| 21:56.44 | freezey | yeah thast the plan |
| 21:56.45 | freezey | screw it |
| 21:56.50 | Katty | freezey: and symlink the working one into the appropriate rc.level |
| 21:56.54 | a1fa | anyway to tell asterisk to reply to all failed attempts as peer not found |
| 21:56.55 | Katty | freezey: or whatever |
| 21:56.59 | niekvlessert | btw, try 1001 on the phone you tried 2001 on |
| 21:57.15 | Katty | freezey: you can also 'make config' during the asterisk compile |
| 21:57.42 | Katty | freezey: well it's the last thing you do generally on a compile, so you can probably stop asterisk make config, and then reboot |
| 21:58.28 | freezey | k thanks |
| 21:58.30 | freezey | fair enough |
| 21:58.31 | X-TaZ | niekvlessert i'd like, but my vmware networking mode cant support it :x |
| 21:58.47 | X-TaZ | i'm using a special compat mode to be able to use wireshark |
| 21:58.53 | niekvlessert | ssh -R ? |
| 21:59.06 | *** part/#asterisk asteriskmonkey (~philip@69.77.169.14) |
| 21:59.10 | *** join/#asterisk xpot-mobile (~xpot@66.60.101.91) |
| 21:59.15 | X-TaZ | i mean, i dont have any web access from my 2 vm's |
| 21:59.43 | Kobaz | heh |
| 21:59.55 | Katty | wonders what web access is for |
| 21:59.56 | X-TaZ | i can post again my config files if it can be usefull |
| 22:00.14 | X-TaZ | internet connectivity if you prefer |
| 22:01.20 | Katty | also, anyone ever had this thing where one ear starts to ring pretty loud and everything /else/ in that ear fades out in the background and gets quieter? ringing lasts for just a few seconds then goes away... |
| 22:02.01 | Naikrovek | Katty: yes |
| 22:02.04 | Naikrovek | it's called tinnitis |
| 22:02.12 | Katty | no, tinnitus is a constant ringing of the ears. |
| 22:02.14 | Katty | which i also have |
| 22:02.20 | Katty | thanks to 3 days of prozac *sigh* |
| 22:02.49 | Katty | Naikrovek: what i'm talking about lasts only a few brief seconds, and dims the audio around it |
| 22:03.00 | Naikrovek | well the intermittent variety is tinnitis, too. so says my doctor |
| 22:03.11 | Naikrovek | but he's not always right, i'm learning |
| 22:03.17 | Katty | how often do you get it? |
| 22:03.19 | Naikrovek | can't trust anyone to have proper information anymore |
| 22:03.27 | Naikrovek | Katty: couple times a week, couple minutes at a time |
| 22:03.35 | Katty | hmm. |
| 22:03.52 | Katty | i'm getting it at least once a day, but only 3 or 4 seconds tops |
| 22:04.18 | Naikrovek | it is like there's a mixer in my head. slowly ups the audio on the "eeeeeeeeeee" noise and everything else fades |
| 22:04.29 | Naikrovek | then anywhere from 30s to 3m later it reverses |
| 22:05.00 | Katty | yeah but you don't get it multiple times every single day |
| 22:05.09 | Naikrovek | now that i think about it, i used to |
| 22:05.13 | Naikrovek | but not anymore |
| 22:05.20 | Katty | i used to get it like...maybe once a month /maybe/ |
| 22:05.22 | Naikrovek | relatively rare for me these days |
| 22:05.47 | Naikrovek | all i can say is that it's normal for me |
| 22:05.50 | Naikrovek | so maybe normal for you? |
| 22:07.48 | Katty | dunno |
| 22:07.56 | Katty | i will call an audiologist and ask |
| 22:08.07 | Katty | tho they will probably want mme to come in and have a visit |
| 22:08.58 | *** part/#asterisk bsaxon (~bsaxon@12.68.234.174) |
| 22:10.30 | *** join/#asterisk Shazaum (~Shazaum@189.73.100.18) |
| 22:10.54 | ChannelZ | Katty: I have that but not often.. like once or twice a year if I had to guess |
| 22:11.27 | mnick86 | Someone tries to call SIP/test ... how can I check if SIP/test is registered or not ? |
| 22:11.35 | ChannelZ | sip show peers |
| 22:11.43 | mnick86 | from the dialplan |
| 22:11.49 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 22:12.51 | ChannelZ | try the function SIPPEER |
| 22:13.13 | *** join/#asterisk NightMonkey (~NightMonk@pdpc/supporter/professional/nightmonkey) |
| 22:13.21 | ChannelZ | ${SIPPEER(SIP/text,status)} perhaps |
| 22:13.37 | ChannelZ | err SIP/test even |
| 22:14.55 | *** part/#asterisk atis_work (~atis_work@193.238.212.171) |
| 22:15.44 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 22:16.22 | *** join/#asterisk Whtsup (~sssi@203.81.226.170) |
| 22:16.25 | Whtsup | hello |
| 22:16.37 | Shazaum | hi |
| 22:17.04 | Whtsup | Got SIP response 400 "Bad Request" back from 192.168.1.110 |
| 22:17.13 | Whtsup | i m getting this error |
| 22:17.30 | Kobaz | ~details |
| 22:17.31 | infobot | If you want help on a topic, you HAVE to say more than "it doesn't work, help!" or else you'll get no help whatsoever. Give as many details as you can or else no one can give any suggestions. |
| 22:17.42 | Whtsup | when i call to my gsm gateway |
| 22:17.50 | Kobaz | more details |
| 22:17.55 | Shazaum | morr |
| 22:18.01 | Kobaz | console log, sip config |
| 22:18.01 | Whtsup | ok |
| 22:18.21 | Kobaz | moarrrr |
| 22:18.32 | Shazaum | hehehe |
| 22:18.40 | lanning | om nom nom nom |
| 22:18.44 | Whtsup | [102] |
| 22:18.44 | Whtsup | username=102 |
| 22:18.45 | Whtsup | mysecret=102 |
| 22:18.45 | Whtsup | host=dynamic |
| 22:18.45 | Whtsup | type=friend |
| 22:18.45 | Whtsup | qualify=yes |
| 22:18.45 | Whtsup | context=from-testing |
| 22:18.50 | Whtsup | my sip confif |
| 22:18.52 | Whtsup | config |
| 22:19.06 | lanning | learn pastebin... |
| 22:19.17 | Kobaz | ~pb |
| 22:19.18 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 22:19.20 | Whtsup | ok |
| 22:19.53 | Shazaum | Kobaz: good bot |
| 22:20.12 | Kobaz | whose a bot |
| 22:20.47 | ChannelZ | mnick86: Actually I lied, it's just ${SIPPEER(test,status)} (no SIP/ on the front, just the peer name) |
| 22:24.05 | Whtsup | http://pastebin.com/iFsZsGPg |
| 22:24.10 | Whtsup | here is my pastebin |
| 22:26.19 | *** join/#asterisk hluesea (~hulusikah@88.247.127.66) |
| 22:26.30 | freezey | tried to make it easier for people to manage the asterisk system by putting trixbox gui in front but its just horrible thing is useless |
| 22:29.02 | ChannelZ | mmmm gooey |
| 22:29.20 | freezey | just wanted to make it easy for people lol but its just not going to happen these gui's are just crap |
| 22:29.32 | *** join/#asterisk ruben23 (~ITadmin@122.55.48.243) |
| 22:30.27 | ChannelZ | yes... yes they are. |
| 22:30.59 | freezey | my patience went to 0 |
| 22:31.16 | freezey | now i am just going to have to reinstall system and teach people howto add conf rooms and extensions |
| 22:32.04 | Brack10 | Is it possible to transfer a call that's on a trunk to an extension on another Asterisk server via IAX2 trunk? |
| 22:32.25 | Brack10 | like an incoming call from a PRI |
| 22:33.46 | X-TaZ | No i'm absolutely sure : http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/I_sect14_tt670.html dont works on asterisk 1.4 |
| 22:34.21 | *** part/#asterisk shader (~user@janustw.tavve.com) |
| 22:35.16 | *** join/#asterisk seba (~seba@p57BDFCDC.dip.t-dialin.net) |
| 22:35.27 | seba | hi |
| 22:36.20 | X-TaZ | now i'm getting [Mar 15 23:35:37] WARNING[5767]: chan_iax2.c:3038 create_addr: No such host: 2001 |
| 22:36.20 | X-TaZ | [Mar 15 23:35:37] WARNING[5767]: app_dial.c:1202 dial_exec_full: Unable to create channel of type 'IAX2' (cause 3 - No route to destination) |
| 22:36.34 | *** join/#asterisk bn-7bc (bjarne@mac.lan.noare-1.holmedal.net) |
| 22:36.46 | Brack10 | X-TaZ: was that directed at me? |
| 22:38.48 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 22:39.07 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 22:41.42 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 22:42.04 | seba | i want to connect with multiple softphones from my computers to my asterisk per sip. do i have to set up an extra sip account for every phone or can i use the same for all? i think with many accounts my dialplan could get realy complicated |
| 22:42.25 | X-TaZ | Brack10 no |
| 22:42.26 | X-TaZ | :p |
| 22:42.43 | X-TaZ | i'm working a a IAX2 trunk too |
| 22:43.32 | Brack10 | X-TaZ: thanks for the sweet document. Any idea of that's possible? |
| 22:45.47 | ChannelZ | Brack10: Perhaps I'm misunderstanding the question, but of course |
| 22:48.01 | Brack10 | ChannelZ: like someone calls in on the PRI, an attendant picks up the call, and they transfer it to an extension on another asterisk install via an iax2 trunk |
| 22:48.15 | ChannelZ | seba: only one device can be registered at a time |
| 22:48.27 | ChannelZ | Brack10: yeah |
| 22:49.05 | Brack10 | K |
| 22:49.24 | seba | hm.. okay, extra accounts it is.. thx |
| 22:49.35 | ChannelZ | Brack10: extension 555 does a Dial(IAX2/somedude) - you transfer a call to 555... |
| 22:51.55 | Brack10 | thanks |
| 22:52.24 | *** join/#asterisk xpot-mobile (~xpot@66.60.101.91) |
| 22:53.42 | *** join/#asterisk Dibri (~gavit@pop1.isgroup.sr) |
| 22:57.09 | *** join/#asterisk Jhirley (~Jhirley@adsl-145-35-115.mia.bellsouth.net) |
| 22:57.43 | Kobaz | pasting pasting |
| 22:59.10 | lmsteffan | <PROTECTED> |
| 23:00.10 | Jhirley | whats that website where you can paste errors ? |
| 23:04.51 | ChannelZ | pastebin.com |
| 23:04.56 | ChannelZ | or pastebin.ca |
| 23:05.13 | Jhirley | thank you |
| 23:17.00 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 23:19.07 | *** join/#asterisk miknix (~miknix@gentoo/developer/miknix) |
| 23:19.27 | miknix | hello all |
| 23:21.55 | miknix | I'm thinking as asterisk as a potential solution to interface some SIP clients to a analog phone at my parents house. I'm really new to this matter and concepts (specially about modems and phone lines). I would appreciate some directions about this matter |
| 23:23.25 | miknix | I have a computer to put asterisk on. it has internet connection and a 56k modem |
| 23:25.56 | miknix | my doubt is if 1) I can use asterisk for this setup 2) if I connect the computer modem to the home's landline, how would I call the other phones connected to the same line? (it has only a phone number) |
| 23:26.01 | NightMonkey | Howdy. Is there a command to syntax check asterisk config files without starting a new instance? |
| 23:26.09 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 23:26.49 | NightMonkey | I want to validate a config before restarting the daemon. |
| 23:27.26 | ChannelZ | hmm not that I know of |
| 23:27.42 | file | validate in what way? |
| 23:28.18 | NightMonkey | file: Sanity check. I'm upgrading from asterisk 1.2 (yeah, I know) to 1.6, and want to make sure I haven't snuffed up anything. |
| 23:28.33 | file | nope, no way really |
| 23:29.05 | NightMonkey | Sweet. Well, thanks for validating the lack of validation. ;) |
| 23:29.31 | NightMonkey | Apache has it, so I thought * might have it by now. |
| 23:29.57 | file | validating it is rather difficult |
| 23:30.32 | file | you can certainly check the basic syntax, but otherwise you'd have to know what applications are present and their arguments plus limits on arguments |
| 23:30.35 | Qwell | Apache configs are all startup-time. Asterisk doesn't have that luxury. |
| 23:30.50 | file | it's just fun |
| 23:30.51 | NightMonkey | file: Well, perhaps validation is too specific a term. A syntax check would rock. Not a check that what you're configuring makes any sense in the real world. |
| 23:31.21 | Qwell | NightMonkey: well, basic syntax hasn't changed any. a 1.2 syntax will work in 1.6 |
| 23:31.36 | Qwell | again - basic syntax only. it's up to apps to say if the actual args are correct |
| 23:32.11 | NightMonkey | Qwell: Thanks, yeah, I'm trying to be too lazy. :) |
| 23:33.00 | NightMonkey | holds nose before diving in |
| 23:36.13 | Brack10 | When I dial an incorrect extension on my softphone, is that Asterisk telling me the call cannot be completed as dialed or my softphone? |
| 23:36.28 | Brack10 | the speaker indicator moves up and down as if it was actually receiving audio |
| 23:36.47 | NightMonkey | Wow, it Just Started(tm). Amazing. |
| 23:36.51 | Qwell | Did you setup Asterisk to play that message? |
| 23:37.08 | Brack10 | I installed asterisknow |
| 23:37.34 | Qwell | freepbx is doing that |
| 23:37.35 | Qwell | ~freepbx |
| 23:37.36 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 23:37.54 | Brack10 | K |
| 23:38.12 | *** join/#asterisk lmsteffan (~laurent@reef.ac-noumea.nc) |
| 23:38.17 | Brack10 | I'm just testing asterisk with this for now |
| 23:38.20 | NightMonkey | I guess the "Comedian Mail" VoiceModel just upped her speaking speed a bit and raised her pitch a half octave? Funny. |
| 23:38.34 | NightMonkey | knows this is very old news |
| 23:43.17 | fifer | anyone have any experience with out-of-band dtmf issues on * with 6731i's |
| 23:43.25 | Qwell | NightMonkey: You would probably be the first to notice that |
| 23:43.33 | Qwell | or, rather, the first to say something about it, anyways |
| 23:45.02 | Qwell | NightMonkey: what happened, was all the prompts were re-recorded in higher quality. I never imagined the prompts would be any different, though with that specific prompt, I can see it |
| 23:46.08 | NightMonkey | Qwell: Well, I'm not complaining. They do sound nicer, for sure. |
| 23:46.22 | Qwell | wasn't taking it as a complaint :p |
| 23:46.33 | NightMonkey | I can't believe that the upgrade was soo easy. Amazing. |
| 23:46.42 | Qwell | that is indeed curious though |
| 23:46.49 | NightMonkey | waits for the other shoe to drop. |
| 23:47.06 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 23:48.21 | NightMonkey | Hrm. Seem to have lost some mailbox custom configs, like recorded prompts. |
| 23:48.38 | ChannelZ | sound files go in a different place now |
| 23:48.44 | ChannelZ | using the language structure |
| 23:49.01 | ChannelZ | make sure anything you recorded is in the right place |
| 23:49.27 | ChannelZ | /var/lib/asterisk/sounds/en/..... more than likely |
| 23:49.37 | NightMonkey | ChannelZ: Ah, thanks, I'll check. |
| 23:50.22 | ChannelZ | if it's people's mailbox greetings and things, then thats something different.. |
| 23:50.28 | *** join/#asterisk Tech_Travis (~Administr@cpe-76-168-191-127.socal.res.rr.com) |
| 23:50.41 | NightMonkey | <PROTECTED> |
| 23:51.14 | Shazaum | Tech_Travis ti |
| 23:52.03 | Tech_Travis | Shazaum: ti? |
| 23:52.12 | Shazaum | nth |
| 23:52.25 | Shazaum | this full win |
| 23:52.26 | Shazaum | :P |
| 23:53.31 | NightMonkey | Hrm. greet files are still in /var/spool/asterisk/voicemail/... |
| 23:54.05 | NightMonkey | Oh, wait, there's a "greet" subdirectory for each extension... perhaps that moved? |
| 23:54.18 | mrtelnet | I have one way audio after a attended transfer only after upgrading from 1.6.0.1 to 1.6.2.5, any ideas? |
| 23:54.59 | ChannelZ | NightMonkey: Hmm no I still have 'greet.wav' and such in the root directory, not inside the 'greet' directory |
| 23:55.29 | ChannelZ | oh but you said you were coming from 1.2 - not sure how it was setup. |
| 23:55.47 | ChannelZ | I upgraded from 1.4 to 1.6 but didn't have to move any of my voicemail junk |
| 23:56.04 | NightMonkey | ChannelZ: Hrm. Let me check the loaded modules... |
| 23:57.04 | ChannelZ | NightMonkey: http://pastebin.com/E54P3kGX there's a listing of one of my mailboxes if it helps figure it out |
| 23:59.37 | NightMonkey | ChannelZ: Thanks! |
