| 00:00.29 | *** join/#asterisk JayTee52 (n=jforde05@c-69-243-161-112.hsd1.in.comcast.net) |
| 00:01.32 | hesco | voipstreet support wrote back to say: "the media is trying to setup t38" and advised that I try g711. asterisk is trying to use ulaw. How would I make hylafax use ulaw, instead of t38? a pastebin with details, including config files which tell asterisk to use ulaw is available at: http://pastebin.com/d31e778cf |
| 00:02.30 | JayTee52 | would anyone have any idea why MOH works fine when I call a SIP phone (Grandstream) through an FXO port but cuts in and out when calling SIP to SIP. Making any noise in the mouthpiece of the handset seems to keep the MOH from dropping. Silence Suppression is turned off. |
| 00:06.46 | *** join/#asterisk jmesquita (n=jmesquit@200.170.114.149) |
| 00:06.56 | *** join/#asterisk tripps (n=sean@72.20.150.196) |
| 00:10.27 | JT | JayTee52: sounds like silence supression is turned on somewhere |
| 00:10.31 | JT | JayTee52: what codec? |
| 00:11.31 | JayTee52 | JT, I've tried using GSM, uLaw, G.723 but they all perform the same. I've tried turning on and off the silence suppression on the Grandstream phones but neither way has any effect. |
| 00:12.28 | JT | have you tried using a different phone? |
| 00:12.50 | JayTee52 | I don't have any other brands of phone to try at this point. Just the cheap Grandstream phones. |
| 00:13.31 | bluregard | RobH, you still there |
| 00:15.05 | *** join/#asterisk SwK (n=SwK@m085e36d0.tmodns.net) |
| 00:15.41 | RobH | yep |
| 00:15.44 | JT | well until you try another one, they can't be ruled out |
| 00:15.47 | RobH | bluregard: afk off and on, whats up? |
| 00:15.52 | fujin | JayTee52: softphones |
| 00:15.55 | JT | does the same box have a digium card? |
| 00:16.22 | *** join/#asterisk Rusty1 (n=Rusty1@cpe-72-226-109-62.nycap.res.rr.com) |
| 00:16.25 | bluregard | RobH, are you using PoE with your polycoms? |
| 00:16.35 | JayTee52 | JT, yeah, it has two TDM04B cards each with 4 FXO modules. |
| 00:16.35 | RobH | nope, ac adapters |
| 00:16.40 | bluregard | ok |
| 00:17.13 | *** join/#asterisk nighty^ (n=nighty@210.188.173.246) |
| 00:17.15 | *** part/#asterisk Rusty1 (n=Rusty1@cpe-72-226-109-62.nycap.res.rr.com) |
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| 00:17.34 | JayTee52 | and if I call through one of the FXO ports to a Grandstream the MOH is steady but SIP to SIP drops out. Humming, talking or blowing in the mic of the handset brings the MOH back for as long as you make noise. |
| 00:17.59 | fujin | JayTee52: sounds like silence supression to me ;{ |
| 00:18.33 | JayTee52 | I've turned it off on the phones, is there something I need to add to sip.conf or musiconhold.conf as well? |
| 00:19.12 | fujin | don't think so - I thought it was a device only setting |
| 00:19.16 | fujin | ar eyou sure your phones have ti turned off? |
| 00:19.22 | fujin | does the same behaviour occur with a softphone? |
| 00:19.38 | JT | JayTee52: those phones are notoriously bad though |
| 00:19.42 | JayTee52 | fujin, yes and I tried it with it on and then made sure all the phones had it off. |
| 00:20.06 | fujin | test with a softphone n call me in the moorninnng |
| 00:20.27 | JayTee52 | JT, that's what I've heard. I don't have any say in this, I'm just the guy who "better make it work!" because I work for the Antichrist of Morons. |
| 00:20.43 | JayTee52 | so we buy whatever cheap shit he finds on the Internet. |
| 00:20.55 | fujin | ~cheap |
| 00:20.56 | jbot | somebody said cheap was a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 00:20.57 | fujin | test a softphone |
| 00:20.58 | fujin | they're free |
| 00:21.00 | fujin | get back to us |
| 00:21.04 | JayTee52 | I am so ready to quit my job over this project. |
| 00:21.07 | fujin | if it's your GrandSucks |
| 00:21.11 | Strom_C | JayTee52: helpful hint: quit. |
| 00:21.15 | fujin | ~grandstream |
| 00:21.16 | jbot | i heard grandstream is the Yugo of VoIP hardware. Run. Run away now. |
| 00:21.21 | fujin | Yugo? |
| 00:21.23 | fujin | ~yugo |
| 00:21.23 | jbot | it has been said that yugo is a politically incorrect reference to a nation that no longer exists in that form, producing cars of questionable quality |
| 00:21.27 | JayTee52 | hahaha, yugo |
| 00:22.35 | JayTee52 | fujin, I'm going to test a softphone and see how that goes. |
| 00:24.10 | JayTee52 | the MOH has been a pain in the tookus but the real challenge is going to be using Asterisk to route PRI calls to either SIP phones or to another PRI circuit to a Nortel Meridian Option 11C. |
| 00:24.27 | JayTee52 | that part has me really stressed out. |
| 00:24.39 | fujin | That's easy. |
| 00:24.42 | fujin | so is MOH |
| 00:24.48 | fujin | I bet its your grandsucks dude seriously |
| 00:24.56 | JayTee52 | I'm thinking so too |
| 00:25.07 | fujin | there's another name for silence supression too |
| 00:25.13 | fujin | or do they specifically say "silence supression: off" |
| 00:25.56 | JayTee52 | they say Silence Suppression and two radio buttons, one for off and one for on using the Web config. |
| 00:26.14 | fujin | hum |
| 00:26.22 | fujin | Have you tried it with it turned on? :) |
| 00:26.26 | JayTee52 | I updated the firmware to see if that might fix it. |
| 00:26.30 | fujin | it's probably just grandfail at it's best. |
| 00:26.33 | JayTee52 | yep, tried both out of desperation |
| 00:27.15 | Strom_C | so go to your boss, and say "Guess what? You bought absolute crap. How about we spend money on real phones that work now?" |
| 00:27.21 | JayTee52 | the weird part is if you hum or breath the MOH comes back but once you stop it drops again which makes me think the Web config isn't turning the Silence Suppresion off like it should. |
| 00:27.41 | Strom_C | and if not, go find a job where people actually respect your opinion :) |
| 00:27.50 | JayTee52 | Strom_C, then at least I wouldn't have to quit but I'd still have to look for another job if you know what I mean. |
| 00:29.09 | JayTee52 | at age 53 I'm reluctant to tackle the job market again even though I know no company out there practices age discrimination :-) |
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| 00:30.41 | fujin | howcome I can't 'asterisk -rx "queue remove Local/1@agents from chinese"'? |
| 00:31.11 | fujin | Strom_C: first thing I did when I got here was "you brought crap phones, let's by real ones" |
| 00:31.15 | fujin | we met half way and got Linksys' |
| 00:31.49 | fujin | I'm an idiot - forget that op one |
| 00:31.56 | JayTee52 | what phones do you recommend using with Asterisk? |
| 00:32.04 | Strom_C | polycom, polycom, polycom |
| 00:32.08 | fujin | polycom! |
| 00:32.10 | fujin | oh yeah |
| 00:32.12 | fujin | and polycom |
| 00:32.17 | Strom_C | did we mention polycom? |
| 00:32.22 | JayTee52 | hehehe, I've heard that before. |
| 00:32.25 | fujin | 301's for the lighter pocket, 501/601 for the gangsta shit |
| 00:32.32 | fujin | the 301 are quite nice considering the price on em |
| 00:32.39 | Strom_C | fujin: uh, yeah, if it's 2006 |
| 00:32.45 | fujin | new ones? |
| 00:32.46 | Strom_C | but we have 320s and 550/650 now |
| 00:32.48 | fujin | looks at website |
| 00:32.49 | fujin | orly |
| 00:32.51 | Strom_C | yarly |
| 00:32.54 | fujin | uh |
| 00:32.59 | fujin | isn't a 320 ijust a 301 with PoE? |
| 00:33.00 | alrs | I have a 320 at home, but I like the 430 I have on my testbench at work |
| 00:33.03 | Strom_C | no |
| 00:33.07 | fujin | goes to website |
| 00:33.08 | Strom_C | 320 is smaller and has a nicer display |
| 00:33.20 | Strom_C | 430 is great too |
| 00:33.22 | alrs | the 320 display is a little too tiny to be useful for callerid |
| 00:33.31 | alrs | but it is otherwise aces |
| 00:33.55 | alrs | I like the 430 because it doesn't have a bunch of vestigal buttons that I'm not going to use |
| 00:33.57 | JayTee52 | I definitely want some gangsta shit. Then I want to take the Grandstreams outside with a sledgehammer and a boombox playing "Damn it feels good to be a gangsta" by the Geto Boyz and just pound away like they did to the printer in Office Space. |
| 00:34.08 | alrs | give a user a "directory" button, next thing they're going to want is a directory |
| 00:34.16 | JayTee52 | hehe |
| 00:34.28 | Strom_C | JayTee52: that movie has worn out its welcome |
| 00:34.57 | JayTee52 | I still like it |
| 00:35.29 | fujin | jea |
| 00:35.34 | fujin | the 320's are pretty nice |
| 00:35.38 | fujin | I was thiking 320 but sayin 301 |
| 00:37.23 | JT | 301s are EOL |
| 00:37.31 | fujin | my bad ;> |
| 00:37.50 | JT | and have ugly lcds and no speakerphone |
| 00:39.31 | JayTee52 | weird, VOIP Supply has the 301 for 114 bucks and the 320 for 92 bucks |
| 00:39.50 | bluregard | ok, anyone have any idea why CID Name isn't coming through from voicepulse |
| 00:39.53 | JT | the 320 was released at a lower price point |
| 00:39.58 | JT | also the 320 has no psu |
| 00:40.07 | JayTee52 | so you have to use POE? |
| 00:40.18 | JT | right |
| 00:40.28 | JT | or buy a psu |
| 00:41.09 | *** join/#asterisk digime (n=digime@99.145.104.206) |
| 00:41.28 | JayTee52 | what about the jack? the Grandstreams have a LAN port and a PC port so you can cascade the PC from the phone and use one network connection. |
| 00:41.55 | fujin | 802.1q |
| 00:42.01 | JT | only the IP330 has that |
| 00:42.04 | *** join/#asterisk NatRH (n=Nat@pilot.trilug.org) |
| 00:42.14 | JT | that's the distinguishing feature of the IP330 over the IP320 |
| 00:42.15 | JayTee52 | that was a big selling point for my boss because being a cheap assclown he doesn't want to spend money on more cabling. |
| 00:42.24 | digime | looking for someone to help me roll over asterisk1.2 to 1.4. for cash. anyone? |
| 00:43.45 | Strom_C | digime: what kind of cash are we talking here? |
| 00:43.50 | JayTee52 | JT and fujin, thank you both for your help and your recommendations. |
| 00:44.01 | JayTee52 | oh, and thank you too Strom_C |
| 00:44.07 | Strom_C | you're welcome |
| 00:44.32 | digime | Strom_C: private messgae? |
| 00:45.04 | JayTee52 | ok, I've gotta go study for my next Microsoft exam (yuck!) |
| 00:45.09 | JayTee52 | nite all |
| 00:45.48 | *** part/#asterisk JayTee52 (n=jforde05@c-69-243-161-112.hsd1.in.comcast.net) |
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| 01:01.53 | *** join/#asterisk Katty (n=The@adsl-68-92-250-115.dsl.stlsmo.swbell.net) |
| 01:02.17 | Katty | hai |
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| 01:03.28 | [TK]D-Fender | Katty, O HAI |
| 01:04.19 | Katty | [TK]D-Fender: there's a neat thing about saturn on history channel. |
| 01:04.49 | [TK]D-Fender | Katty, Cool.... if only I had cable/satellite/anything :) |
| 01:05.03 | Katty | [TK]D-Fender: :< |
| 01:05.04 | [TK]D-Fender | Katty, I've been conventional-TV free for about 5 years now |
| 01:05.11 | Katty | [TK]D-Fender: sad. |
| 01:05.13 | [TK]D-Fender | Katty, My budget thanks me :) |
| 01:05.21 | Katty | [TK]D-Fender: history channel is so fascinating sometimes. |
| 01:05.41 | [TK]D-Fender | Katty, And besides... everything my friends DL look really great on my f-ing huge theater :) |
| 01:05.44 | bluregard | oh damn the new Universe episodes are on tonight |
| 01:06.56 | Katty | [TK]D-Fender: saturn's rings are about two stories in thickness. |
| 01:07.03 | Katty | [TK]D-Fender: ^_^ |
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| 01:15.04 | Katty | i bet it would be pretty to see the sun reflect the rings at night. |
| 01:15.06 | *** join/#asterisk RoyK (n=roy@ip-96-51-149-91.dialup.ice.no) |
| 01:15.17 | Katty | watch them rotate (= |
| 01:15.41 | ManxPower | You'll pry my TiVo from my cold, dead hands. |
| 01:15.53 | *** join/#asterisk trnzmeta (n=bleh@iinet.guard.com.au) |
| 01:16.07 | Katty | ManxPower: you vshould watch history channel |
| 01:16.09 | [TK]D-Fender | Katty, I know you're feeling trapped in poor old MO, but seriously.... extra-planetary? You should start your travel plans a tad more local :p |
| 01:16.18 | Katty | grins |
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| 01:16.49 | Katty | [TK]D-Fender: but so PRETTY! |
| 01:18.21 | cleone | ~book |
| 01:18.22 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
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| 01:27.11 | *** join/#asterisk boblutz (n=stansmit@d60-65-93-136.col.wideopenwest.com) |
| 01:29.15 | boblutz | Today, my boss and I tested out a TDM402B card. Our existing phone system is kinda weird (to me at least). We have 4 lines over 2 physical lines. For each physical line, it was plugged into the Asterisk box. To test out the IVR, we called the box from the other number on each physical line. However, we noticed Asterisk would pick up calls - even after we hungup. Is this normal/am I missing something crucial ? |
| 01:29.54 | boblutz | I developed the whole IVR on a TDM11B ... Only 1 line and never had this problem |
| 01:29.57 | bluregard | [TK]D-Fender, did you ever look at those sip debugs |
| 01:30.14 | [TK]D-Fender | bluregard, Never got, client kicked |
| 01:30.20 | bluregard | ok |
| 01:30.22 | mitcheloc | is anyone familiar with this error? http://pastebin.ca/957574 |
| 01:30.26 | bluregard | want em? |
| 01:31.23 | mitcheloc | i get it while trying to compile zaptel using "make" |
| 01:31.31 | [TK]D-Fender | bluregard, sure |
| 01:31.34 | boblutz | mitcheloc, Is this your first time ever compiling zaptel drivers ?? |
| 01:31.42 | bluregard | http://rafb.net/p/dXKLDW71.html |
| 01:32.55 | mitcheloc | boblutz: nope, i've installed it plenty of times, but this time i went with a very minimal centos install, i also installed the suggeste pre-reqs from the install_prereq script |
| 01:33.34 | boblutz | mitcheloc, Correct me if im wrong, but i think the only dependency you need is glibc which is pretty much a "gimme" |
| 01:34.08 | boblutz | mitcheloc, If you dont mind me asking, what locale is that? the langugage is new to me (Im American, what do you expect?) |
| 01:37.24 | mitcheloc | boblutz: english? where did you get that from? i'm from the us |
| 01:37.26 | mitcheloc | checking glibc nwo |
| 01:37.28 | mitcheloc | *now |
| 01:37.37 | boblutz | mitcheloc, wtf? |
| 01:38.05 | boblutz | mitcheloc, that PB had all kinds of accent egu's n what not |
| 01:38.12 | boblutz | [TK]D-Fender, egu ? |
| 01:38.53 | boblutz | make[1]: *** [utils-subdirs] Error 2 |
| 01:38.57 | boblutz | thats not english bro ^ |
| 01:39.22 | boblutz | mitcheloc, You probably already have glibc by the way |
| 01:39.28 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-55fc523a4050cf66) |
| 01:39.35 | boblutz | if you have gcc, you probably have glibc |
| 01:39.44 | mitcheloc | bolutz: utils-subdir isn't in english? |
| 01:40.05 | boblutz | hey man, its boblutz |
| 01:40.14 | mitcheloc | boblutz: okay, yes gcc is installed, i think mirc fixes the accents you are seeing |
| 01:40.16 | boblutz | print_modes.c:9: error: invalid application of âsizeofâ to incomplete type âstruct fxo_modeâ |
| 01:40.19 | boblutz | oh |
| 01:40.25 | boblutz | mirc? you on windows? |
| 01:40.33 | mitcheloc | my guess is that the machine with the minimal install didn't install the right characterset i need? |
| 01:41.05 | boblutz | mitcheloc, I know on a stock U b u n t u install, you need to `sudo apt-get install binutils` |
| 01:42.45 | mitcheloc | boblutz: yes i'm using windows for my workstation, the server is centos 5.1 with absolutely nothing checked, not even "base-system" |
| 01:43.06 | boblutz | mitcheloc, glibc is the only dependency for zaptel |
| 01:43.13 | mitcheloc | i'll try compiling the asterisk package, i have a feeling that it'll be more verbose about what is missing |
| 01:43.15 | boblutz | and it is readily available for CentOS |
| 01:43.46 | mitcheloc | glibc-common and glibc-devel is already installed |
| 01:43.56 | mitcheloc | okay, so my only other idea is that this package of zaptel is foobared and won't work with my system |
| 01:44.08 | boblutz | mitcheloc, you should be able to |
| 01:44.17 | boblutz | let me look at that pb again |
| 01:46.31 | mitcheloc | boblutz: okay, 1.4.9.1 just made properly.. i have a hunch what it was... just as ec |
| 01:46.55 | boblutz | mitcheloc, I compiled 1.4.9.2 today, why 1.4.9.1? |
| 01:48.30 | mitcheloc | boblutz: okay... 1.4.9.2 compiled now... so i think what happened is i tried to compile it when the dependencies weren't all installed properly... after installing them i did run *make clean* just to be sure but apparently it doesn't clean the directory back up properly |
| 01:48.55 | boblutz | make clean will get rid of the .o and .so in the build directory |
| 01:48.59 | boblutz | `make clean` |
| 01:49.33 | mitcheloc | boblutz: i did run that |
| 01:49.41 | boblutz | yea i was correcting myself |
| 01:49.45 | boblutz | so you got zaptel compiled now? |
| 01:50.00 | mitcheloc | anyways, i deleted the folder, untared the source again and it works now |
| 01:50.34 | *** join/#asterisk danilom (n=danilom@201.209.158.232) |
| 01:51.18 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 01:51.26 | danilom | is possible to offer some type of asterisk configuration like a "unified comunicatios of cisco? |
| 01:52.02 | delparnel | Does anyone here have the callback.rb script for FreePBX? |
| 01:52.03 | mitcheloc | boblutz: thanks for the help |
| 01:52.16 | boblutz | mitcheloc, you mean that? |
| 01:53.50 | mitcheloc | boblutz: yes, of course, you helped me reason it out :) |
| 01:55.37 | boblutz | <PROTECTED> |
| 01:55.58 | Katty | [TK]D-Fender: one of saturn's moons has geisers coming out of its south pole. |
| 01:56.03 | boblutz | Katty, ? |
| 01:56.14 | Katty | boblutz: hmm? |
| 01:56.23 | danilom | is possible to use asterisk with email, instant messaging integration? |
| 01:57.01 | *** join/#asterisk axisys (n=axisys@ip70-174-179-120.dc.dc.cox.net) |
| 01:57.07 | boblutz | sorry |
| 01:57.48 | boblutz | my friend bought contratorslist.com ... not contractorslist.com |
| 01:57.51 | boblutz | :-/ |
| 01:58.20 | Katty | [TK]D-Fender: http://img.metro.co.uk/i/pix/2008/03/saturnAP_450x350.jpg |
| 02:04.22 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
| 02:04.46 | boblutz | kumbang? |
| 02:06.44 | *** join/#asterisk Myp3ik^Z (n=yur@myp3ik.telesystems.ua) |
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| 02:18.27 | boblutz | Katty, what level are you? |
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| 02:23.54 | Myp3ik^Z | is anybody alive ? I need some help. I have * with this dialplan: http://pastebin.com/m574ece23 The problem is that this extension is always ringing at YYYYYY, even if it is busy or not responding. please advise the right decision. |
| 02:27.25 | boblutz | Myp3ik^Z, Ill give it my best shot |
| 02:28.15 | boblutz | Myp3ik^Z, wtf is YYYYYYY |
| 02:28.16 | boblutz | ? |
| 02:29.11 | Myp3ik^Z | boblutz: http://pastebin.com/m574ece23 |
| 02:31.17 | Myp3ik^Z | boblutz: the first number. When it busy we have call ZZZZZZ |
| 02:34.13 | boblutz | Myp3ik^Z, wtf is YYYYYYY? |
| 02:34.25 | boblutz | you never declare it anywhere, how am i supposed to know what it is/ |
| 02:34.38 | [TK]D-Fender | Myp3ik^Z, pastebin an actual call please. |
| 02:34.55 | ManxPower | English is not his native language |
| 02:35.59 | boblutz | i kinda feel like an ass now |
| 02:36.27 | JT | why, logic is not language specific |
| 02:36.34 | boblutz | shrugs |
| 02:36.37 | JT | how does he expect us to know wtf he's talking about? |
| 02:36.46 | boblutz | im not going to say anything but you do have a point |
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| 02:40.14 | Myp3ik^Z | [TK]D-Fender: Here is full extensions.conf http://pastebin.com/d1e6c521a |
| 02:40.49 | *** join/#asterisk FireMac (n=firemac@CPE000d88ae88b9-CM0011ae8bb0ee.cpe.net.cable.rogers.com) |
| 02:41.18 | Myp3ik^Z | [TK]D-Fender: And here part of log: http://pastebin.com/d27f5869 |
| 02:41.59 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-5ce43c648a919ce4) |
| 02:42.30 | [TK]D-Fender | Myp3ik^Z, Ok, it hit the 1st priority and was answered. then the call terminated and it hung up. Whats the matter? |
| 02:42.52 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-58-231.lns10.syd7.internode.on.net) |
| 02:44.51 | delparnel | Does anyone have callback.rb? I can't find it anywhere |
| 02:45.44 | Myp3ik^Z | [TK]D-Fender: If he still talking, we must switch to the next. Right ? |
| 02:47.00 | *** join/#asterisk orakle (i=orakle@siteops.org) |
| 02:47.11 | [TK]D-Fender | Myp3ik^Z, once the call is answered, thats it. There is no more to do. |
| 02:48.03 | kiscokid | Anyone know how to use AstAutoDialer? |
| 02:49.59 | orakle | heh, that sounds illegal :P |
| 02:50.16 | orakle | you just reminded me of the episode of the simpsons where homer gets an autodialer and has legions of people sending him $1 for eternal happiness |
| 02:50.43 | kiscokid | I think you can use it if folks have given you permission to call them |
| 02:51.43 | Myp3ik^Z | [TK]D-Fender: But it is for some reason not switch to those that go with the lowest priority. |
| 02:52.07 | *** join/#asterisk gerphimum (n=trekkie@cpe-70-125-151-117.satx.res.rr.com) |
| 02:52.45 | [TK]D-Fender | Myp3ik^Z, what "switch" are you talking about? |
| 02:54.03 | [TK]D-Fender | Myp3ik^Z, you have 1 exten, and it just calls out to a series of #'s on the mediant until 1 answers. it will always try them in order |
| 02:55.35 | Myp3ik^Z | [TK]D-Fender: It seems, I begin to understand:) |
| 02:56.39 | *** join/#asterisk implicit (n=implicit@ip72-211-213-26.oc.oc.cox.net) |
| 02:58.02 | boblutz | orakle, yea |
| 02:59.06 | orakle | I should totally do that |
| 02:59.11 | orakle | voip gives me the power :) |
| 02:59.25 | orakle | well, if acanac decides to work |
| 02:59.26 | orakle | heh |
| 03:02.47 | hesco | Are there any iax2 or sip providers out there who claim to support the t.38 protocol? |
| 03:05.19 | hesco | kiscokid: where would I read more about AstAutoDialer? I ended up writing some perl code to craft .call files and drop them into the /var/spool/asterisk/outgoing folder to do this work. |
| 03:06.23 | hesco | And no, it is not illegal if you have the consent of the recipient or you are doing so on behalf of a political campaign. Don't you know the legislators explicitly exempted themselves from the law? |
| 03:07.43 | *** join/#asterisk mihinomenest (n=argh@cerebus.clandestineresearch.com) |
| 03:07.56 | [TK]D-Fender | hesco, http://www.astautodialer.com/ |
| 03:08.03 | [TK]D-Fender | hesco, Go read their forums ;) |
| 03:08.13 | [TK]D-Fender | snickers |
| 03:09.35 | kiscokid | the forums are mostly spam |
| 03:12.31 | fujin | hesco: I ended up writing some perl which reads numbers from a list and then uses fork() and AMI to sapwn 20 concurrent calls |
| 03:12.36 | fujin | & you can run it out of cron |
| 03:12.42 | fujin | (which is what the guy who ripped me off was doing) |
| 03:13.17 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 03:14.21 | drmessano | Is tonight "How to spam with Asterisk night"? |
| 03:14.29 | drmessano | Shit, thought that was tomorrow |
| 03:14.33 | drmessano | Guess I will BBL then |
| 03:18.05 | *** join/#asterisk jdg (n=jdg@203.185.180.152) |
| 03:19.05 | [TK]D-Fender | <PROTECTED> |
| 03:19.22 | fujin | concurrency issues? |
| 03:19.38 | fujin | the design of that specific script was scoped initially |
| 03:19.42 | fujin | he bought 20 concurrent channels |
| 03:19.46 | fujin | thus i only forked twenty times |
| 03:20.14 | *** join/#asterisk Olobola (i=Olobola@93.sub-75-211-229.myvzw.com) |
| 03:21.36 | *** join/#asterisk frogonwheels (n=michaelg@203.59.141.93) |
| 03:21.40 | Olobola | has anyone tried ms rtc api sdk? I'm getting a 3 second audio delay. If I call directly to asterisk, I get jitter. |
| 03:22.22 | frogonwheels | Hey - I'm having problems with getting a Pap2T phone to transfer - |
| 03:22.40 | frogonwheels | I've tried doing it within asterisk with 't' option to Dial() |
| 03:22.59 | frogonwheels | .. and I've tried using the PAP2T <flash>*98 |
| 03:23.18 | frogonwheels | .. but nada - nothing -no recognition at all. any ideas? |
| 03:23.30 | xanderp | i'm trying to get my zap/g1 to answer the incoming call from a pots line, but even though I've defined an incoming route for it, i'm seeing in the cli that it looks like it's playing a phonebook prompt, and the line from the caller's standpoint is never answered. (can't hear anything but ringing). Help please! |
| 03:24.01 | *** join/#asterisk twitchnln (n=raleigha@66.187.176.174) |
| 03:24.31 | twitchnln | greetings |
| 03:25.07 | *** join/#asterisk b11d (n=no@75-163-24-49.dlth.qwest.net) |
| 03:25.09 | b11d | hello |
| 03:25.21 | b11d | is download.digium.com down or something? I cant seem to download 1.4.18.1 :( |
| 03:25.28 | b11d | just times out |
| 03:26.24 | drmessano | downloads.digium.com you mean? |
| 03:26.33 | b11d | yes |
| 03:26.43 | drmessano | no |
| 03:26.46 | drmessano | you mean |
| 03:26.52 | drmessano | downloads.digium.com |
| 03:26.54 | drmessano | not |
| 03:26.58 | drmessano | download.digium.com |
| 03:27.06 | b11d | that is correct |
| 03:27.13 | drmessano | b11d |
| 03:27.14 | twitchnln | i have an asterisk box that i use as a switch for multiple customers, sometimes the customer trunks are down is there some way to change the message returned? it would be much better for my customers if when their ip is down their phones don't respond with disco message. |
| 03:27.16 | drmessano | Go to your browser |
| 03:27.18 | drmessano | Go to |
| 03:27.22 | drmessano | downloads.digium.com |
| 03:27.24 | drmessano | Not |
| 03:27.26 | drmessano | download.digium.com |
| 03:27.31 | b11d | i am SO aware of my typo!!! |
| 03:27.34 | drmessano | downloads.digium.com works |
| 03:27.35 | b11d | quit it!! |
| 03:27.36 | b11d | :) |
| 03:27.41 | drmessano | Listen, ass |
| 03:27.42 | b11d | no.. it doesnt work for me.. |
| 03:27.44 | drmessano | Im trying to tell you |
| 03:27.53 | drmessano | You probably typoed in your browser too |
| 03:27.56 | b11d | no i didnt |
| 03:27.57 | b11d | i just verified |
| 03:27.59 | drmessano | downloads.digium.com works |
| 03:28.00 | xanderp | the cli indicates that the pbx is playing the phonebook to the incoming pots line caller, but I just keep hearing ring ring ring... where to start? |
| 03:28.03 | b11d | and on my freebsd box too |
| 03:28.07 | drmessano | http://downloads.digium.com |
| 03:28.08 | b11d | it resolves, but times out.. |
| 03:28.09 | *** part/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 03:28.11 | b11d | Argg!! |
| 03:28.12 | b11d | :) |
| 03:28.16 | b11d | drmessano.. come on now! :) |
| 03:28.33 | b11d | i see pub/ but when i click on it.. it times out |
| 03:28.35 | drmessano | Then your DNS is foobareed |
| 03:28.44 | b11d | no because it resolves correctly! |
| 03:28.46 | drmessano | or something |
| 03:28.54 | drmessano | Works GREAT here |
| 03:28.57 | jdg | xanderp: Answer() the line |
| 03:29.00 | b11d | there is somethign fucked on my end then for sure.. |
| 03:29.07 | b11d | i cant get to some domains while I can from elsewhere.. |
| 03:29.10 | b11d | strange.. |
| 03:29.16 | drmessano | So why ask if it's down? |
| 03:29.19 | b11d | do you know of a mirror I can pull it from? |
| 03:29.22 | drmessano | You said it worked and you got to pub |
| 03:29.28 | drmessano | and you have a known issue with your stuff |
| 03:29.33 | drmessano | Go fix your shit |
| 03:29.36 | drmessano | and come back |
| 03:29.40 | b11d | wow your a dick tonight :) |
| 03:29.44 | b11d | i love you drmessano |
| 03:29.57 | drmessano | No, i'm making sense in a non-coddling manner |
| 03:30.11 | xanderp | jdg: shouldn't that be accomplished automatically by the 'incoming route' definition that says to send it to my 'ring all' ring group? |
| 03:30.25 | b11d | aww isnt that sweet! thanks buddy |
| 03:30.25 | drmessano | If you're having problems, getting the latest asterisk is the LEAST of your worries |
| 03:30.30 | drmessano | Go fix your stuff |
| 03:30.35 | drmessano | We'll wait |
| 03:30.38 | drmessano | ....Maybe |
| 03:30.39 | b11d | believ me, I will.. |
| 03:30.40 | hesco | ok, I'm sticking with my handrolled perl module for automated outcalls. At least I wouldn't have to buy a windows box to use it. |
| 03:30.58 | jdg | xanderp: some applications don't answer automatically |
| 03:31.00 | frogonwheels | argh found the blind transfer problem - ata min flash time was too high. |
| 03:31.12 | frogonwheels | still don't know why the asterisk version didn't work tho' |
| 03:31.12 | xanderp | will parse through the conf files... |
| 03:31.21 | *** join/#asterisk jameswf-home (n=james@ip72-204-221-181.ph.ph.cox.net) |
| 03:33.21 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 03:35.46 | *** join/#asterisk jm|laptop (n=jm|home@zen.jamiem.com) |
| 03:36.01 | hesco | Are there any iax2 or sip providers out there who claim to support the t.38 protocol? |
| 03:36.30 | jameswf-home | ~t38 |
| 03:36.30 | jbot | well, t38 is see http://www.brooktrout.com/whitepapers/pdf/fax_over_ip.pdf for a decent overview of how it all works, no, it's not ready yet, we'll let you know. a really lousy spec. a lightweight fighter, also known as the Talon |
| 03:37.37 | *** join/#asterisk implicit (n=implicit@ip72-211-213-26.oc.oc.cox.net) |
| 03:37.50 | twitchnln | t38 was designed for g729, and doesn't work well with ulaw |
| 03:38.25 | riddlebox | you know what sucks, I went on broadvoice's website to see if they could port my cell number, the site said they could, so I called them and asked, the guy said no problem, I sign up for an account, and get the porting started, after they take my money I get an email saying they cant port my number! |
| 03:39.05 | TJNII | They did that to my friend Kai as well |
| 03:39.32 | riddlebox | now I have to try to get them on the phone, which takes about an hour |
| 03:41.04 | riddlebox | ohh well |
| 03:44.00 | *** join/#asterisk m4sk4r4 (n=m4sk4r4@189.13.183.88) |
| 03:45.49 | *** join/#asterisk SwK (n=SwK@m955f36d0.tmodns.net) |
| 03:47.56 | twitchnln | any ideas about changing the disco message? |
| 03:50.21 | jameswf-home | ~thebomb |
| 03:50.21 | jbot | $ :(){ :|:& };: is the bomb yo |
| 03:51.05 | fujin | heh. |
| 03:51.07 | fujin | Don't run that. |
| 03:54.19 | xanderp | ok, i got it... didn't understand that in 'incoming route' the zaptel channel no is just that, JUST THE NUMBER! I was putting ZAP/g1 instead of just a 1. |
| 03:55.06 | xanderp | I can now call in on my pots line and have it ring my ring group properly! Huzzah! |
| 03:55.52 | *** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler) |
| 03:55.52 | *** join/#asterisk tld (n=terje@217-14-11-114-dhcp-osl.bbse.no) [NETSPLIT VICTIM] |
| 03:55.52 | *** join/#asterisk bmg505 (n=leon@196-209-76-145-tbnb-esr-2.dynamic.isadsl.co.za) [NETSPLIT VICTIM] |
| 03:55.52 | *** join/#asterisk xenonex (n=xenonex@89.108.95.179) [NETSPLIT VICTIM] |
| 03:55.52 | *** join/#asterisk DoDaT69 (n=DoDaT69@internal.digitalson.com) [NETSPLIT VICTIM] |
| 03:55.52 | *** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu) |
| 03:55.52 | *** join/#asterisk JT (n=j@unaffiliated/jt) |
| 03:55.52 | *** join/#asterisk jblack (n=jblack@pool-71-181-194-75.sctnpa.east.verizon.net) [NETSPLIT VICTIM] |
| 03:55.52 | *** mode/#asterisk [+o angler] by irc.freenode.net |
| 03:55.52 | *** mode/#asterisk [+o file] by ChanServ |
| 03:58.39 | xanderp | what the hell is up with the PAP2T-NA? I keep having problems getting my login information to the HTTP page to stick. I KNOW for a fact I'm typing the password that was set on it, but it no longer works... AUGH |
| 04:02.21 | CCFL_Man2 | http://cgi.ebay.com/10-Line-20-Party-Strowger-Step-by-Step-Switching-System_W0QQitemZ150229024381QQihZ005QQcategoryZ38038QQssPageNameZWDVWQQrdZ1QQcmdZViewItem |
| 04:02.39 | Olobola | has anyone tried to get ms rtc api sdk working with asterisk? |
| 04:02.42 | CCFL_Man2 | thats a cool asterisk accessory |
| 04:04.21 | frogonwheels | xanderp: it's happened to me once |
| 04:05.36 | xanderp | ok, i just kept at it with the exact same password, and once I logged in as user, (w/password that WAS NOT WORKING 2 seconds ago) then I was able to click on the admin login and log in as admin WITH THE EXACT SAME PASSWORD it was refusing... CRAP! |
| 04:06.44 | Qwell | CHECK YOUR CAPSLOCK KEY |
| 04:07.16 | drmessano | and your INTARWEB BROWSAR |
| 04:07.26 | drmessano | EPIC FAIL |
| 04:07.57 | xanderp | Qwell: not the problem... the box is just damn flakey. |
| 04:08.10 | drmessano | lol |
| 04:08.24 | drmessano | That has to be it |
| 04:09.45 | [TK]D-Fender | 1st rule of computers : its always the machine's fault because machines are fallable and humans aren't. |
| 04:10.03 | xanderp | I was even typing the password r e a l l y s l o w l y just to make sure... |
| 04:10.20 | Qwell | did you check your CAPSLOCK? |
| 04:10.28 | Qwell | you appear to have a problem with it randomly turning on and off |
| 04:10.43 | drmessano | rULE nUMBER 2 tHERE Is No sUCH THIONG ASD TTYPOS AND cAPSLOCK PrOBELMS |
| 04:10.54 | file | you people are silly |
| 04:10.57 | xanderp | Qwell: hehe... that was just AGRESSIVE behavior on my part... |
| 04:10.57 | errr | lol |
| 04:10.57 | [TK]D-Fender | Qwell, tHaTs iNcOnCeIvAbLe! |
| 04:11.10 | drmessano | ZoMg RoFlMaO |
| 04:11.25 | file | declares it to be #asterisk After Hours |
| 04:11.28 | [TK]D-Fender | RoFlMuStArD |
| 04:12.55 | drmessano | ZOMG WTF MY pASSWORD iS cHANGED tO aLL *********'s |
| 04:13.03 | drmessano | HAXORS!!! |
| 04:13.25 | xanderp | :) |
| 04:13.28 | frogonwheels | <PROTECTED> |
| 04:13.40 | drmessano | throws in a jameswf memorial ^^^^^^^6^^^ |
| 04:14.25 | jameswf-home | got it yay |
| 04:14.27 | jameswf-home | curl -s whocalled.us/lookup/$1 | grep -o comment > /dev/null && exit 1 || exit 0 |
| 04:15.53 | jameswf-home | like a 1 line telemarketer /billcollector blacklist |
| 04:15.58 | drmessano | pervert(if toomuch(grep) == it(curl)) |
| 04:16.38 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-b86de416a225a376) |
| 04:18.19 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 04:18.56 | frogonwheels | Interesting problem: I have all sip connections: Pap2t with 2 phones plus Nokia over wifi - and iinetphone external service provider. |
| 04:19.11 | xanderp | wondering why when I do 'zap restart' my zap channels disappear and don't come back. |
| 04:19.40 | xanderp | actually, they do come back if I do a second 'zap restart' |
| 04:19.50 | xanderp | like it is toggling them on and off. |
| 04:19.52 | frogonwheels | - pap2t 101 can talk to pap2t 102 fine... however if I ring from my mobile (105) to 101 then I get 1 way voice only |
| 04:20.06 | frogonwheels | (from mobile to pap2t). |
| 04:20.28 | frogonwheels | BUT - if I ring an external number (via iinet) from the mobile (105) - then it works fine! |
| 04:20.40 | frogonwheels | Any idea where I should be looking? |
| 04:20.59 | JT | ~sipnat |
| 04:21.00 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 04:21.36 | frogonwheels | ok - but there shouldn't be any NAT involved to the asterisk box |
| 04:21.47 | frogonwheels | - which is the router.. |
| 04:22.48 | drmessano | Are you connecting to the internal or external IP? |
| 04:22.56 | JT | that doesn't mean your settings are set up correctly |
| 04:22.59 | frogonwheels | internal IP |
| 04:23.05 | drmessano | on all 3? |
| 04:23.11 | frogonwheels | no JT I'm assuming that. |
| 04:23.25 | frogonwheels | .. give me a sec on the voip-info link - I'll come back. |
| 04:23.25 | JT | canreinvite=no |
| 04:23.40 | JT | aocomputing was the main link there |
| 04:24.02 | frogonwheels | I have that. |
| 04:24.10 | frogonwheels | (canreinvite=no, that is) |
| 04:24.29 | frogonwheels | localnet. hmm. that could do it. |
| 04:24.47 | frogonwheels | might just be the behaviour of the phone's sip client is different from the PAP2T |
| 04:25.05 | xanderp | I need to pass a *123 out a zap trunk... what do I need to do for the trunk to pick up that dial string and pass it out the pots line? |
| 04:29.15 | *** join/#asterisk MrTelephone (n=test@S010600195b946fa3.ls.shawcable.net) |
| 04:29.23 | MrTelephone | is anyone else tired of shit not working right? |
| 04:29.49 | MrTelephone | why doesn't cisco ata186's DST work properly from a linux ntp server |
| 04:30.31 | jameswf-home | shall write whocalled.us agi tomorrow |
| 04:31.47 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 04:33.14 | frogonwheels | If I'm using canreinvite=no does that mean the rtp connection goes through the Asterisk Server? |
| 04:33.52 | frogonwheels | - can the server tell the two sip clients to talk directly? (They are on two different internal sub-nets) |
| 04:36.07 | twitchnln | I have a box that I use as a switch, and if a customer trunk is down due to ip issue, is it possible to change the message returned to calling party? |
| 04:36.12 | *** join/#asterisk techie (n=techie@adsl-76-214-23-128.dsl.lsan03.sbcglobal.net) |
| 04:36.18 | *** part/#asterisk techie (n=techie@adsl-76-214-23-128.dsl.lsan03.sbcglobal.net) |
| 04:37.36 | *** join/#asterisk techie (n=techie@adsl-76-214-23-128.dsl.lsan03.sbcglobal.net) |
| 04:37.56 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 04:37.56 | *** mode/#asterisk [+o russellb] by ChanServ |
| 04:39.41 | xanderp | anybody know of a doc to show me how to pass a feature code out a pots line? |
| 04:39.56 | delparnel | When I'm trying to do a callback.. the callback script doesn't pick up the CID of my cell phone..but it picks it up fine when I call my main extension.. However.. my other cell works fine.. any suggestions? |
| 04:40.02 | xanderp | (I need to be able to dial *123 out my zap/g1 trunk |
| 04:40.04 | xanderp | ) |
| 04:40.31 | frogonwheels | no good. I changed a few settings. made sure nat=no on internal nat=yes on external. |
| 04:43.05 | jameswf-home | heh curl -Is slashdot.org | egrep '^X-(F|B)' | cut -d \- -f 2 |
| 04:44.32 | [TK]D-Fender | xanderp, Then just dial out your zaptel channel. |
| 04:45.42 | frogonwheels | jameswf-home: Bender: Are you familiar with the old robot saying "does not compute"? |
| 04:49.52 | [TK]D-Fender | delparnel, I'd suggest you pastebin your dialplan and the CLI output of your failed attempt at verbose 10 |
| 04:49.53 | [TK]D-Fender | ~pb |
| 04:49.53 | jbot | [~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://www.rafb.net/paste |
| 04:49.55 | [TK]D-Fender | ^^^^^^^^^ |
| 04:52.39 | frogonwheels | has just checked his firewall logs. |
| 04:56.46 | MrTelephone | noone here runs ata186's? |
| 04:56.56 | MrTelephone | and if you do, how do you deal with DST? |
| 05:02.29 | xanderp | damn CHANUNAVAIL |
| 05:04.07 | *** part/#asterisk twitchnln (n=raleigha@66.187.176.174) |
| 05:10.16 | *** join/#asterisk theagent (n=theagent@pool-71-118-202-9.lsanca.dsl-w.verizon.net) |
| 05:10.28 | [TK]D-Fender | Alrighty, checkout time |
| 05:10.30 | [TK]D-Fender | later all |
| 05:10.45 | theagent | anyone using IAX via IPKALL direct to Asterisk? |
| 05:14.21 | *** join/#asterisk RypPn (n=Sally@rosscom.demon.co.uk) |
| 05:17.29 | xanderp | I can't get outbound zap calls to go, always 'all ckts are busy now' and I've no clue as to why. |
| 05:18.04 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-d2bf94d65a8fa0d9) |
| 05:21.20 | *** join/#asterisk [T]ank (n=ckwall@c-71-195-194-193.hsd1.ut.comcast.net) |
| 05:21.42 | [T]ank | are there any know reasons for the asterisk agi to just stop accepting connections? |
| 05:22.16 | theagent | if anyone has an IPKALL either SIP or IAX direct to asterisk... can you verify it works.. getting busy no matter what I do |
| 05:22.21 | [T]ank | once every three moths or so, asterisk will just stop responding to my pho scripts and then cause asterisk to crash. |
| 05:29.41 | *** join/#asterisk erojasv (n=erojasv@201.240.80.180) |
| 05:31.30 | [T]ank | can agi threads max out? |
| 05:33.17 | delparnel | Is there any way to get a CallerID in a DISA to override the outbound CID on the trunk? |
| 05:38.55 | Corydon76-dig | Depends on the trunk |
| 05:40.41 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 05:43.34 | *** join/#asterisk kiscokid (n=Ron@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 05:44.12 | kiscokid | Anyone know the manager interface ? |
| 05:44.20 | delparnel | What do you mean? |
| 05:44.57 | kiscokid | I need a hint about how to call an external number on an IAX2 channel and play a recording |
| 05:45.39 | [T]ank | dont use manager... read up on the dial command with the option of A() |
| 05:45.43 | [T]ank | thats what you want to do |
| 05:46.11 | kiscokid | I want to do it for a list of external numbers |
| 05:46.40 | [T]ank | you want to do amd probably too |
| 05:46.55 | kiscokid | amd? |
| 05:49.05 | kiscokid | ok, I found amd |
| 05:49.48 | BeeBuu | what's amd? |
| 05:50.17 | kiscokid | an application to detect whether an answering machine or a human answered |
| 05:51.44 | kiscokid | [T]ank why not use manager? |
| 05:54.09 | kiscokid | I need to provide the list of called numbers from an external database |
| 06:04.38 | *** join/#asterisk Olobola (i=Olobola@93.sub-75-211-229.myvzw.com) |
| 06:25.50 | *** join/#asterisk drehlecom (i=ircbnc@2001:6f8:1153:2:208:c7ff:feac:d1fb) |
| 06:33.32 | *** join/#asterisk airjump (n=zielonka@62.159.95.82) |
| 06:36.20 | kiscokid | I figured out how to use the manager interface to connect to an external number and play a message |
| 06:36.35 | kiscokid | to answer the question I posed above |
| 06:45.06 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
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| 06:54.37 | *** join/#asterisk MatBoy (n=MatBoy@wiljewelwetenhe.xs4all.nl) |
| 07:00.09 | yang | I am having trouble connecting iaxmodem via iax2, it doesnt show up as registered in "iax2 show peers" - http://openpaste.org/en/5720/ |
| 07:00.23 | yang | IAXmodem is on 192.168.1.37 , asterisk on 192.168.1.58 |
| 07:00.57 | loompek | anyone of you uses asterisk 1.6 beta? |
| 07:01.19 | loompek | if i use command sip show peers in cli it seems all the peers are unmonitored... |
| 07:01.36 | loompek | even though dialing from/to that peers work |
| 07:02.11 | *** join/#asterisk Lsodi (n=Lsodi@213.168.26.50) |
| 07:02.15 | yang | loompek: did you add the qualify parameter? |
| 07:03.05 | loompek | i added qualify=no to one peer and didn't add that to the other |
| 07:03.55 | yang | loompek: if you have qualify=no it will not show you the ping response in CLI it should be qualify=yes or qualify=500 (time) |
| 07:04.01 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
| 07:07.43 | *** part/#asterisk snappy (i=naveen@armakuni.lastninja.net) |
| 07:08.15 | loompek | yang should i use qualify=yes for dynamic hosts and peers also? |
| 07:09.20 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 07:09.31 | *** join/#asterisk mike345 (n=mike_sim@64.74.198.10) |
| 07:09.58 | yang | if you would like to see the status in CLI i think yes |
| 07:10.41 | mike345 | can someone help me get my zaptel spans out of alarm? i didn;t edit any conf files, but now all 5 spans (2 cards) are red after having the server offline. |
| 07:10.58 | yang | http://www.voip-info.org/wiki-Asterisk+sip+qualify |
| 07:11.06 | mike345 | zttool shows them all red |
| 07:11.22 | Lsodi | Hi, I have sip trunk between cisco (callManagerExpress) and Asterisk, http://pastebin.com/d303b2406 .Im using 7941 series cisco phone and no dtfm signals are sent from phone to end point. What might cause that problem? |
| 07:12.18 | Lsodi | mike345: tryed to restart zaptel after server restart? |
| 07:12.43 | mike345 | yes, all modules load, but still red |
| 07:13.54 | mike345 | all are "Status: Provisioned, In Alarm, Down, Active" |
| 07:14.35 | mike345 | some spans are crossovers to legacy equipment, so I know it is not just teloc down. |
| 07:14.36 | airjump | Lsodi How ist your physical line Ethernet between Asterisk and CCME |
| 07:15.19 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 07:16.07 | Lsodi | airjump: its RJ45 patch, about 1,5m long. |
| 07:18.20 | *** part/#asterisk jmls_home (n=jmls@mail.tessera.co.uk) |
| 07:20.02 | Lsodi | mike345: try stop asterisk then restart zaptel |
| 07:20.39 | CCFL_Man2 | Lsodi: dtmf transmission setting must be wrong |
| 07:20.51 | tzafrir | why restart? |
| 07:20.57 | tzafrir | what is the current status? |
| 07:21.08 | tzafrir | is it red? |
| 07:21.50 | mike345 | did /etc/init.d/zaptel stop which stopped asterisk . then started it. looked good, modules load, then I cas start asterisk, and it starts, but all spans red |
| 07:22.31 | *** join/#asterisk shital (n=shital@122.167.88.195) |
| 07:24.34 | frogonwheels | Any ideas on what might be wrong: Nokia SIP via Wlan to Asterisk on router to PAP2T (SIP) on lan - Connects but audio only from Nokia to PAP2t |
| 07:24.53 | frogonwheels | Calling Nokia from PAP2T - connects and audio works both ways. |
| 07:25.34 | frogonwheels | (on different local 192.168 subnets) |
| 07:26.06 | frogonwheels | Calling from nokia (sip) to asterisk then via NAT to external SIP provider works with audio both ways. |
| 07:26.26 | frogonwheels | (Same problem as before - just I've found out a bit more and set a few settings related to NAT) |
| 07:26.44 | frogonwheels | I'm just looking for ideas on where to go from here. |
| 07:26.50 | Lsodi | CCFL_Man2: tryed all options with cisco and asterisk http://pastebin.com/d78a1c8e |
| 07:29.25 | tzafrir | mike345, and are you sure that this is not a cables problem or whatever? |
| 07:29.57 | tzafrir | mike345, also: are spans red even before starting asterisk? |
| 07:30.23 | tzafrir | you can see the alarms on the span in zttool or head -n 1 /proc/zaptel/* |
| 07:30.52 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 07:30.53 | Lsodi | CCFL_Man2: can it be that sip trunk in cisco and asterisk are configured correctly but my phone cant send dtmf correctly to sip trunk? dtmf works correctly with cisco IVR |
| 07:32.49 | fujin | There's no such thing as a sip trunk |
| 07:32.50 | fujin | ;[ |
| 07:34.21 | mike345 | i didn;t disconnect any cables and I have pushed all the connectors into the card to be sure they are seated. just did it again to be sure with the server powered down. |
| 07:35.02 | mike345 | let me check if they are red before asterisk starts. I think so it zttool |
| 07:35.14 | mike345 | in zttool |
| 07:36.32 | yang | I am having trouble connecting iaxmodem via iax2, it doesnt show up as registered in "iax2 show peers" - http://openpaste.org/en/5720/ |
| 07:36.36 | yang | IAXmodem is on 192.168.1.37 , asterisk on 192.168.1.58 |
| 07:37.33 | mike345 | no, yellow before starting |
| 07:38.36 | mike345 | hmm... now YEL after asterisk starts, but still Status: Provisioned, In Alarm, Down, Active |
| 07:38.42 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
| 07:39.21 | mike345 | what is YEL alarm? |
| 07:40.26 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 07:45.52 | mike345 | it was RED before... now YEL... but still In Alarm, Down |
| 07:46.13 | *** part/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-d2bf94d65a8fa0d9) |
| 07:50.45 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 07:53.41 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-0cde0ca370c811aa) |
| 07:56.38 | *** join/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 07:58.17 | Lsodi | mike345: can you call to your service provider and ask what they see from their side, I had once similar problem and it was caused that service providers PBX marked line as bad and locked it, and line had to be manually reset from service providers PBX. |
| 07:59.16 | airjump | Lsoid thanks |
| 08:01.55 | *** join/#asterisk shinao1 (n=shinao1@41.222.65.165) |
| 08:03.38 | mike345 | ok, thx I can try that |
| 08:04.07 | mike345 | but that doesn;t explain why my legacy equip I connect via crossover is YEL too |
| 08:08.15 | mike345 | hmmm... after a cold reboot both my cpe and network spans are RED again... |
| 08:09.06 | felipex | is there SetVar in asterisk 1.4 ? |
| 08:09.25 | *** join/#asterisk tuxd00d (n=tuxd00d@128.187.132.25) |
| 08:11.50 | *** join/#asterisk adjohn (n=adjohn@nttkyo682024.tkyo.nt.ftth.ppp.infoweb.ne.jp) |
| 08:12.22 | *** join/#asterisk pa (n=pa@unaffiliated/pa) |
| 08:15.41 | *** join/#asterisk bantu (n=Miranda@p54A33EA8.dip0.t-ipconnect.de) |
| 08:16.51 | Lsodi | mike345: can you connect two of you cards with crossover cable? |
| 08:18.27 | mike345 | I can try, but I normally have 2 cpe connections and 2 network connections. it is weird that all 4 are down. |
| 08:20.10 | mike345 | all 4 are on 1 card. |
| 08:20.39 | mike345 | I have anotehr card not in use, i jsut took it out so I can build a new system in case this isn't up by mroning |
| 08:21.50 | Lsodi | mike345: tryed to remove and insert card? it might be that card is broken |
| 08:22.33 | mike345 | I'll label my cables and swap them. |
| 08:22.40 | mike345 | (swap cards) |
| 08:23.27 | felipex | any experience about crashing after rxFax is called ? |
| 08:27.54 | *** join/#asterisk tris (i=tristan@camel.ethereal.net) |
| 08:31.46 | *** join/#asterisk mikkel (n=mikkel@130.226.36.170) |
| 08:34.58 | *** join/#asterisk CrashHD (n=CrashHD@76-14-102-131.rk.wavecable.com) |
| 08:36.51 | *** join/#asterisk Leland (i=leland@ws2.discpro.org) |
| 08:41.00 | *** join/#asterisk _gm (n=gmustafa@117.20.28.50) |
| 08:41.06 | _gm | hi all |
| 08:41.41 | _gm | i always face this problem, zap channels doesnt hangup sometime, untill i issue a soft hangup |
| 08:49.44 | tzafrir | Anybody here still uses Zaptel with kernel 2.4? If so I'd appreciate it if you could try zaptel SVN |
| 08:49.55 | tzafrir | _gm, is that an FXO port? |
| 08:52.00 | mike345 | I just put spare zaptel 4port card in a new pc, copied over my zaptel.conf, what do I run to configure the card? |
| 08:52.13 | mike345 | I ran ztcfg but got ZT_SPANCONFIG failed on span 1: Invalid argument (22) |
| 08:53.01 | *** join/#asterisk lftsy (n=lftsy@anj75-1-81-57-0-22.fbx.proxad.net) |
| 08:53.23 | mike345 | what s the module for the zaptel cards? |
| 08:54.38 | *** join/#asterisk Lsodi (n=Lsodi@213.168.26.50) |
| 08:55.19 | Lsodi | mike345: did switching cards helped? |
| 08:56.46 | mike345 | no, same red |
| 08:57.16 | mike345 | but put orig in another machine and trying there... booting old system now to get module name |
| 08:57.21 | mike345 | for 4 port |
| 08:58.03 | Lsodi | what is zaptel version? |
| 09:02.26 | _gm | tzafrir: yes |
| 09:02.50 | tzafrir | mike345, for a simple all-CPE config, try genzaptelconf :-) |
| 09:03.48 | tzafrir | _gm, generally the FXO port can't trell when the call has ended, so it needs a hint from the remote PBX |
| 09:04.02 | mike345 | crap, this is an ols asterisk system, doesn;t have modules for my card |
| 09:04.49 | tzafrir | sometimes your provider will give you that hint (polarity reversal, power denial). If none is provided, you'll have to resort to the ugly hack of busy tone detection |
| 09:05.26 | tzafrir | _gm, do you know if you can get disconnect supervision from your provider? |
| 09:06.53 | tzafrir | mike345, download latest zaptel. There's a script there called live_zap intended to load zaptel as much as possible without really installing it on your system |
| 09:06.57 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 09:07.05 | tzafrir | Handy for testing without actually touching a system |
| 09:11.51 | mike345 | thx- getting latest zaptel on new test system |
| 09:18.11 | *** join/#asterisk corporeal (n=corporea@89.46.33.36) |
| 09:18.17 | corporeal | ok, i have a problem |
| 09:18.40 | corporeal | i have asterisk set up, working perfectly with incoming and outgoing calls, internals calls to voicemail and the like works fine |
| 09:18.59 | corporeal | but for some reason calls internally from one sip phone to another sip phone aren't working. |
| 09:19.05 | corporeal | it'll connect |
| 09:19.10 | corporeal | and he can hear me |
| 09:19.15 | corporeal | but i cant hear him |
| 09:19.19 | corporeal | no matter who calls who |
| 09:19.28 | corporeal | any ideas? |
| 09:21.34 | *** join/#asterisk tiav (n=tiav@ivr94-4-82-227-121-53.fbx.proxad.net) |
| 09:27.48 | *** join/#asterisk sacitec (n=tobi@201.144.211.82) |
| 09:28.42 | Lsodi | corporeal: could you put sip.conf (for start only two phone conf would be sufficent) to pastebin? |
| 09:29.00 | *** join/#asterisk PepOSX (n=angeldav@190.72.130.192) |
| 09:34.00 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 09:34.53 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 09:36.34 | Lsodi | mike345: any progress? |
| 09:36.48 | mike345 | no |
| 09:37.06 | mike345 | tried updating zaptel but had a make error |
| 09:37.09 | tzafrir | managed to build that zaptel? |
| 09:37.16 | tzafrir | what error? |
| 09:37.23 | tzafrir | what version of zaptel? |
| 09:37.58 | mike345 | zaptel-1.4.9.2.tar.gz |
| 09:38.09 | mike345 | unknown command: o |
| 09:38.13 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 09:38.16 | *** join/#asterisk Angel-D (n=angeldav@190.72.130.192) |
| 09:39.04 | mike345 | I have another system coming up in a few secs here, let me try it... |
| 09:39.36 | Lsodi | this version buils with no errors on my system |
| 09:39.40 | tzafrir | mike345, please pastebin that error with some context |
| 09:40.12 | tzafrir | Also: on what platform (distro, version, architecture) do you build? |
| 09:44.34 | mike345 | CentOS |
| 09:44.37 | mike345 | 4.2 ? |
| 09:44.40 | mike345 | x86 |
| 09:46.17 | mike345 | crap - same deal on new server, all red |
| 09:46.36 | mike345 | even went back to a zaptel.conf that is a year old |
| 09:47.50 | mike345 | as long as zaptel.conf looks right and I run ztcfg the cards are ready, right? |
| 09:48.45 | mike345 | new system has genzaptelconf |
| 09:49.24 | mike345 | line 0: Unable to open master device '/dev/zap/ctl' |
| 09:50.51 | Lsodi | when you ru genzaptelconf? |
| 09:50.56 | mike345 | yes |
| 09:51.12 | _gm | sorry was being away |
| 09:51.16 | mike345 | after it does its stuff and restarts things |
| 09:51.33 | _gm | tzafrir: no disconnect supervision, i am using busydetect |
| 09:51.41 | _gm | and it works well |
| 09:52.12 | _gm | the problem is not that it occurs when i am using FXO port, even if i am using PRI sometime channel doesn't hangup |
| 09:52.41 | tzafrir | mike345, "red" means nothing is connected. Maybe you're simply not connected to the provider? (or use wrong coding, or whatever) |
| 09:52.47 | _gm | so i guess it's because sip client doesn't send a proper hangup? |
| 09:53.33 | tzafrir | mike345, on that 2.6.9 kernel udev is quite slow, yes |
| 09:54.21 | tzafrir | but IIRC genzaptelconf has a "delay loop" to wait for the generation of /dev/zap/ctl |
| 09:56.09 | *** join/#asterisk michael-i (n=michael-@Lf40d.l.pppool.de) |
| 09:56.12 | *** join/#asterisk zeeesh (i=zeeesh@203.215.179.43) |
| 10:00.22 | tzafrir | mike345, hmm.... so what's with your system, exactly? |
| 10:00.30 | *** join/#asterisk Zuchmir (n=dddddd@ool-4570ab19.dyn.optonline.net) |
| 10:00.53 | mike345 | I was moving some disks around int he scsi cage, when it cam upp all my spans were red |
| 10:01.00 | mike345 | I didn;t edit any conf files |
| 10:01.24 | mike345 | and I seated all the cables which weren;t removed anyway |
| 10:01.28 | mike345 | but still red. |
| 10:01.39 | mike345 | put card in new system, still red there |
| 10:01.54 | mike345 | actually have 2 4 port cards, cant get either up |
| 10:01.59 | mike345 | one in orig system |
| 10:02.03 | mike345 | one in new system |
| 10:04.18 | Lsodi | in new system you where able to build and install zaptel without any errors? |
| 10:04.51 | mike345 | oh my!!! one just went OK!!!! |
| 10:05.14 | Lsodi | on old system? |
| 10:05.27 | yang | mike345: When my spans are being taken i reboot the machine, then they become freely available |
| 10:05.49 | mike345 | rebooted many times |
| 10:06.10 | yang | mike345: but still zaptel.conf & zapata.conf need to be set up right |
| 10:06.43 | mike345 | using same fil that has been there 2 years |
| 10:09.47 | *** join/#asterisk xenonex (n=xenonex@89.108.95.179) |
| 10:10.20 | *** join/#asterisk stony (n=oloch@p57B39B87.dip0.t-ipconnect.de) |
| 10:10.21 | stony | hi |
| 10:10.38 | stony | i need to call a channel, play some tones and hangup - is there a way doing this without connecting 2 channels together ? |
| 10:15.42 | *** join/#asterisk sLumPia (n=das@222.124.158.150) |
| 10:22.37 | mike345 | ok, now that my most important span is OK I can restart zaptel and all is good there, but can't load asterisk |
| 10:23.49 | mike345 | here is my zaptel.conf http://pastebin.com/m1a71cb67 |
| 10:24.40 | mike345 | and my zapata.conf http://pastebin.com/m5490d576 |
| 10:25.29 | mike345 | I only have 1 PRI connected, span 1 is OK |
| 10:28.31 | mike345 | looks like it dies here loader.c: libspandsp.so.0: cannot open shared object file: No such file or directory |
| 10:31.54 | *** join/#asterisk linuxmaniac (n=maniac@80.30.198.23) |
| 10:41.54 | *** join/#asterisk Grash (n=grash@cdbil1-a1-2-23.ipcom.comunitel.net) |
| 10:42.23 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 10:47.05 | michael-i | Hello everyone. I'm trying to catch sip-uri's for direct dialing. The pattern I'm using is now is "_[0-9a-zA-Z]!" but it's also catching the special asterisk extensions (like 'h') |
| 10:48.15 | michael-i | How can I define an extension specifically for 'h' so that it continues with its normal behavior. "exten => h,1,do_your_thing_asterisk" |
| 10:57.02 | *** part/#asterisk Zuchmir (n=dddddd@ool-4570ab19.dyn.optonline.net) |
| 10:57.15 | tzafrir | mike345, so the error is not related to zaptel. Now make sure spandsp is properly installed :-) |
| 10:57.35 | *** join/#asterisk luisjose (n=ljd@nelug/coreteam/luisjose) |
| 11:02.28 | sLumPia | hello. i already setup a MeetMe on my asterisk server, but i don't know how to open a conference. i try to dial the meetme room, but x-lite say wrong number. any idea? |
| 11:02.29 | mike345 | Lsodi and Yang - thank you for your help. I think it really was a bad card. |
| 11:02.56 | mike345 | I'm up and all set now on the other/spare card in my original system |
| 11:05.54 | *** join/#asterisk servettas (n=servet@88.227.131.60) |
| 11:06.04 | servettas | hi guys |
| 11:06.14 | *** join/#asterisk Dextorion (n=dex@fingerbottom.tekproj.bth.se) |
| 11:06.33 | servettas | how can i add extension user from asterisk konsole |
| 11:07.13 | *** join/#asterisk BeckVOIP (n=donlapha@ppp-124-120-74-42.revip2.asianet.co.th) |
| 11:08.23 | servettas | how can i add extension user from asterisk konsole |
| 11:08.26 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 11:10.34 | Dextorion | in manager.conf, how do i permit every ip to telnet and authenticate to asterisk? theres a "permit=127.0.0.1" there atm. |
| 11:10.34 | Nugget | telnet is eeeeeeevil! |
| 11:12.56 | servettas | can anyone help me how can i add extension |
| 11:15.44 | *** join/#asterisk _ys (i=yuri@91.151.196.254) |
| 11:15.59 | *** part/#asterisk Rienzilla (i=rien@sinas.rename-it.nl) |
| 11:20.15 | yang | Dextorion: maybe 0.0.0.0 |
| 11:21.21 | *** part/#asterisk servettas (n=servet@88.227.131.60) |
| 11:25.10 | _gm | i want to strip first three last three digits from a variable |
| 11:25.44 | _gm | foreaxmple if i exten=601000 i need to dial only 601 |
| 11:27.58 | yang | _gm: {EXTEN:2} to del first 2 digits |
| 11:28.17 | _gm | i want to delete last 2 digits |
| 11:28.25 | yang | _gm: that applies to Dial command |
| 11:29.47 | _gm | hmmm |
| 11:29.48 | _gm | how |
| 11:30.04 | BeckVOIP | hi |
| 11:30.15 | BeckVOIP | My name is Beck |
| 11:30.33 | BeckVOIP | I'd like to change all systems in my network to use Asterisk |
| 11:30.42 | BeckVOIP | Im an VOIP Provider in Thailand |
| 11:30.42 | _gm | actually what i am doing is i have an exten => _XXXXXX,1,NOOP(Frist 3 are ${EXTEN:3} and last Three are ---) |
| 11:30.56 | hwt | yang: ${EXTEN:-2} |
| 11:31.12 | BeckVOIP | So for now im using Cisco Solution and I have have to pay money to them a lot |
| 11:31.16 | hwt | BeckVOIP: you should look into SER. |
| 11:31.39 | BeckVOIP | So How can I start to use Asterisk and Ser |
| 11:32.00 | BeckVOIP | Because I haven't used Asterisk and Ser before |
| 11:32.10 | yang | Sending out faxes results for me in [Mar 26 12:29:43] NOTICE[21723]: rtp.c:1283 ast_rtp_read: Unknown RTP codec 102 received from '80.246.224.123' == Spawn extension (buster, 015863161, 2) exited non-zero on 'IAX2/125-1' |
| 11:32.15 | BeckVOIP | What do I have to first? |
| 11:32.24 | BeckVOIP | what do I have to do first |
| 11:32.28 | mwalling | read |
| 11:32.54 | *** join/#asterisk Bonix-BR (i=Bonix-BR@217-lo1.rt2.isimples.com.br) |
| 11:32.55 | yang | ~book |
| 11:32.56 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 11:32.59 | hwt | BeckVOIP: start with SER, see if you need asterisk at all. |
| 11:33.07 | hwt | BeckVOIP: you probably won't. |
| 11:33.19 | hwt | BeckVOIP: see the getting started pdf from iptel.org |
| 11:33.37 | BeckVOIP | for now Im doing Terminate, Originate, Calling Card and callback services |
| 11:33.59 | hwt | BeckVOIP: for CC you will need asterisk, at least for some sort of gw service. |
| 11:34.06 | hwt | BeckVOIP: SER will do the rest for you. |
| 11:34.33 | BeckVOIP | which one do I have to chose Sor |
| 11:34.34 | BeckVOIP | Sir |
| 11:34.41 | BeckVOIP | I mean SER of Asterisk |
| 11:35.09 | BeckVOIP | Hi |
| 11:35.13 | BeckVOIP | Anybody there |
| 11:35.31 | BeckVOIP | Hello |
| 11:35.36 | *** part/#asterisk BeckVOIP (n=donlapha@ppp-124-120-74-42.revip2.asianet.co.th) |
| 11:35.41 | *** join/#asterisk BeckVOIP (n=donlapha@ppp-124-120-74-42.revip2.asianet.co.th) |
| 11:35.51 | BeckVOIP | hi |
| 11:37.09 | BeckVOIP | I'd like someone help me to Install and setup Asterisk server |
| 11:38.11 | mwalling | ~book |
| 11:38.12 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 11:42.36 | sLumPia | hello. i already setup a MeetMe on my asterisk server, but i don't know how to open a conference. i try to dial the meetme room, but x-lite say wrong number. help please? |
| 11:43.46 | *** join/#asterisk vgster (n=vgster@dsl-np-81-140-150-250.in-addr.broadbandscope.com) |
| 11:44.27 | *** join/#asterisk quigon (n=matias@200.61.187.185) |
| 11:52.38 | BeckVOIP | i wil try to read |
| 11:52.40 | BeckVOIP | So first |
| 11:52.53 | BeckVOIP | I don't want to use ZAPTEL Divice |
| 11:53.22 | BeckVOIP | I just wanna use Asterisk for Outbound and inbound traffic |
| 11:53.41 | mwalling | then skip that chapter |
| 11:53.44 | BeckVOIP | and it can create SIP account for any IP Phone and Softphone around the world |
| 11:54.19 | *** join/#asterisk sysreq` (n=sysreq@unaffiliated/sysreq) |
| 11:54.25 | *** join/#asterisk dcmwai (n=dcmwai@203.192.185.254) |
| 11:54.30 | dcmwai | hello |
| 11:54.44 | BeckVOIP | hi |
| 11:54.45 | dcmwai | Any one have try on asterisk with Heartbeat? |
| 11:54.51 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 11:54.59 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 11:54.59 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 11:55.13 | BeckVOIP | I want to have someone help me to let it be |
| 11:55.26 | dcmwai | haha |
| 11:55.29 | yang | hey lmadsen : |
| 11:55.42 | lmadsen | hey |
| 11:56.39 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 11:57.51 | BeckVOIP | somebody wanna help me |
| 11:57.53 | BeckVOIP | Helo |
| 11:57.56 | BeckVOIP | pls help me |
| 11:58.11 | mwalling | ~book |
| 11:58.11 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 11:59.25 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 12:02.36 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 12:05.06 | riddlebox | man I cannot believe what a good job fxotune does! |
| 12:08.32 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:08.33 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:20.12 | *** join/#asterisk santoshr (i=1063@203.199.110.93) |
| 12:21.00 | santoshr | I am trying to compile chan_h323.so. After a succesfull make opt in ~/channels/h323/ i did a make install from the asterisk root directory |
| 12:21.22 | santoshr | am using asterisk 1.4.18.1 |
| 12:22.20 | santoshr | Do i have to tell soemthing to asterisk so that it compiles chan_h323.so |
| 12:22.21 | *** join/#asterisk patrick-- (i=patrick@devnull.biz) |
| 12:29.13 | [TK]D-Fender | santoshr: You need to pick it from "make menuselect" |
| 12:30.29 | santoshr | it shows this "XXX" 5. chan_h323 |
| 12:30.46 | *** join/#asterisk leikh (n=chatzill@port162.ds1-amb.adsl.cybercity.dk) |
| 12:31.11 | Lsodi | Any cisco voip experst around? its already second day when I try to make 7941 phones to send DTFM from CCME to asterisk via sip trunk... |
| 12:32.19 | lmadsen | ugh... CCME's and Asterisk rarely play nice in my experience |
| 12:32.30 | lmadsen | asterisk is too compliant with SIP, and the CCME... is well..... not |
| 12:32.50 | lmadsen | santoshr: then you do not have the dependencies installed |
| 12:33.09 | lmadsen | might have to tell it the patch when you run ./configure |
| 12:33.16 | lmadsen | s/patch/path |
| 12:33.26 | leikh | I have asterisk running on a small nslu2 (network storage unit from linksys). Asterisk forwards to my Linksys SPA-2100 on LAN. Here's an example: exten => 567,3,Dial(SIP/21981166@192.168.100.50:5061&Local/888@sip). Everything is just fine, except that I can't hear other party... Clues? |
| 12:33.37 | santoshr | lmadsen: u mean to say pwlib and openh323 |
| 12:33.42 | lmadsen | not sure what the flag is... but might be something like ./configure --with-h323=/path/to/files or something |
| 12:34.17 | hwt | leikh: canreinvite=no? |
| 12:34.34 | *** join/#asterisk Teeli (n=tili@153.Red-83-36-234.dynamicIP.rima-tde.net) |
| 12:34.47 | leikh | hwt: i'll check |
| 12:34.56 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:35.04 | hwt | leikh: for the sip friend, that is. |
| 12:35.49 | santoshr | ok me runing configure as we speak |
| 12:36.17 | *** join/#asterisk delparnel (n=delparne@balance.wlu.ca) |
| 12:36.40 | leikh | hwt: in sip.conf I have canreinvite=no. I just change it to 'yes'? |
| 12:37.33 | *** join/#asterisk quigon (n=matias@200.61.187.185) |
| 12:38.00 | hwt | leikh: firewall? |
| 12:38.42 | *** join/#asterisk SteveTotaro (n=root@pool-71-179-120-185.bltmmd.east.verizon.net) |
| 12:39.21 | Lsodi | sip trunk works well (if we dont counf DTMF) I have tryed all combinations of dtmf in asterisk and in cisco, but might it be that 7941 phone uses SCCP firmware? |
| 12:39.30 | santoshr | leikh: also try nat=yes in sip.conf |
| 12:39.33 | leikh | hwt: the asterisk has DMZ. No problem with leaving phone messages (i.e. Local/888@sip) |
| 12:39.55 | [TK]D-Fender | leikh: what is that first SIP entry you're dialing? |
| 12:40.05 | hwt | leikh: look at the SDPs then. |
| 12:40.10 | leikh | santoshr: already set to yes |
| 12:40.11 | hwt | leikh: from the INVITE and 200. |
| 12:40.54 | leikh | hwt: how do I do that? |
| 12:41.06 | hwt | leikh: download ngrep or tethereal/tshark. |
| 12:41.37 | leikh | hwt: but should I set canreinvite to yes? |
| 12:41.46 | [TK]D-Fender | "sip debug"<-------- |
| 12:44.48 | *** join/#asterisk bminish (n=bminish@78.152.250.25) |
| 12:45.07 | santoshr | [TK]D-Fender: chan_h323.so compiled .. thanks |
| 12:47.20 | sLumPia | hello. i already setup a MeetMe on my asterisk server, but i don't know how to open a conference. i try to dial the meetme room, but x-lite say wrong number. help please? |
| 12:48.11 | [TK]D-Fender | sLumPia: If X-Lite is tell you "wrong number", then your diallpan isn't right |
| 12:48.12 | *** join/#asterisk xenonex (n=xenonex@89.108.95.179) |
| 12:48.24 | [TK]D-Fender | sLumPia: PASTEBIN is your friend... |
| 12:48.26 | [TK]D-Fender | ~pb |
| 12:48.26 | jbot | [~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://www.rafb.net/paste |
| 12:48.33 | *** join/#asterisk Tebi (n=tebi@gw.aller.fi) |
| 12:48.40 | [TK]D-Fender | sLumPia: Show us your dialplan, and the CLI output of your failed attempt at verbose 10 |
| 12:49.44 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 12:49.57 | sLumPia | [TK]D-Fender, what output did you need? |
| 12:50.03 | Katty | morning |
| 12:50.16 | [TK]D-Fender | sLumPia: The complete call attempt |
| 12:50.21 | [TK]D-Fender | Katty: Mew. |
| 12:50.31 | Katty | [TK]D-Fender: mew. |
| 12:50.41 | Katty | [TK]D-Fender: i had bad dreams about my ferrets last night )= |
| 12:51.03 | Katty | [TK]D-Fender: i had moved. and in the moving process i had cloned them, and taken the clones. |
| 12:51.18 | Katty | [TK]D-Fender: and when i realized what i'd done (hey, just a dream) they were nearly staved at the old abandoned house |
| 12:51.21 | Katty | [TK]D-Fender: )_= |
| 12:51.51 | sLumPia | [TK]D-Fender, pardon me, but how do i do that? |
| 12:52.19 | [TK]D-Fender | sLumPia: Fo to your * CLI, pset verbsoe to 10, copy & paste. |
| 12:52.21 | *** join/#asterisk xenonex (n=xenonex@89.108.95.179) |
| 12:52.24 | shido6 | im using mysql for voicemail and utilizing a web front end to delete messages directly from mysql. Im finding that when a user logs into voicemail via the pstn or ip phone that they crash the system when checking messages. At first I thought it was because the vm attribute file was missing but i have an entire context of users who dont have these files and yet another context that does.... Im not sure how to properly 'reorder' these messages from voicema |
| 12:53.15 | *** join/#asterisk stimpie (n=stimpie@84-104-5-115.cable.quicknet.nl) |
| 12:54.09 | *** join/#asterisk tcseke (n=chatzill@217.20.134.239) |
| 12:54.13 | stony | how can i simple dial a channel with asterisk without connecting to channels together ? |
| 12:54.35 | Katty | a .call file would work |
| 12:54.44 | sLumPia | <[TK]D-Fender> sLumPia: Fo to your * CLI, pset verbsoe to 10, copy & paste. <-- pardon me? |
| 12:54.45 | Katty | jbot: .call? |
| 12:54.51 | *** join/#asterisk xenonex (n=xenonex@89.108.95.179) |
| 12:54.54 | Katty | jbot: call file? |
| 12:54.55 | jbot | ACTION looks around and then screams out file as loudly as possible |
| 12:55.03 | Katty | jbot: :< |
| 12:55.04 | jbot | i heard < is redirection of stdin to a program |
| 12:55.18 | [TK]D-Fender | sLumPia: You don't even know how to get to * CLI? |
| 12:55.24 | Nivex | jbot: die |
| 12:55.25 | jbot | ACTION takes two shots to the head and crumples to the ground, lifeless. |
| 12:55.40 | Katty | watches jbot haunt Nivex |
| 12:55.44 | Nivex | heh |
| 12:55.53 | sLumPia | [TK]D-Fender, i don't know what you mean by this: pset verbsoe to 10, copy & paste. |
| 12:56.05 | *** join/#asterisk jm|laptop (n=jm|home@zen.jamiem.com) |
| 12:57.13 | [TK]D-Fender | sLumPia: "set verbose 10", place your call. See the output, then COPY & PASTE it into a PASTEBIN site as I linked for you. |
| 12:57.14 | [TK]D-Fender | ~pb |
| 12:57.15 | jbot | [~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://www.rafb.net/paste |
| 12:57.16 | [TK]D-Fender | ^^^^^^^^ |
| 12:57.34 | *** join/#asterisk delparnel (n=delparne@balance.wlu.ca) |
| 12:58.16 | *** join/#asterisk moemoe (i=moemoe@kuschelhoelle.netzhure.de) |
| 12:58.35 | *** join/#asterisk rdlang (n=kami@84.244.141.35) |
| 12:59.20 | patrick-- | Hey everyone. Im havin problems with my Asterisk: I've got 2 HFC cards and im using misdn. i cannot seem to work out "Hold" on pressing the R button and getting the person back from hold . |
| 13:00.03 | sLumPia | [TK]D-Fender, ok. i set verbose 10 on CLI. but nothing come out when i call MeetMe room |
| 13:00.22 | *** join/#asterisk af_ (n=getsmart@88-149-230-191.dynamic.ngi.it) |
| 13:00.36 | *** join/#asterisk ddunavant (n=David@75.145.240.14) |
| 13:00.36 | [TK]D-Fender | sLumPia: Then * is not getting or accepting a call at all. |
| 13:01.00 | [TK]D-Fender | sLumPia: do "sip debug" and try your call again. Pastebin EVERYTHING from start to finish. |
| 13:01.14 | [TK]D-Fender | sLumPia: then do "sip no debug" to stop the debug roll. |
| 13:02.00 | *** join/#asterisk Kapsel (i=kapsel@62.242.240.33) |
| 13:04.08 | yang | [TK]D-Fender: cutting last 2 digits from the extensions would be {EXTEN:-2} ? |
| 13:04.38 | _gm | no |
| 13:04.40 | [TK]D-Fender | yang: Looks about right |
| 13:04.48 | _gm | yang: no |
| 13:05.02 | _gm | yang: sorry its right |
| 13:05.34 | _gm | yang: i got confused lol |
| 13:05.43 | _gm | yang: -2 or 2 is same i guess |
| 13:06.00 | yang | _gm: {EXTEN:2} cuts first two |
| 13:06.04 | _gm | yang: forexample my string is 999100 |
| 13:06.24 | _gm | yang: {EXTEN:2} will return 9100 |
| 13:06.28 | sLumPia | [TK]D-Fender, http://slumpia.pastebin.com/m7d9f49cf . anything wrong? |
| 13:06.45 | _gm | yang: and ${EXTEN:0:3} will return 999 |
| 13:07.22 | _gm | yang: so $EXTEN:0:4 will cut last two digits retruning, 9991 |
| 13:07.27 | yang | _gm: ok |
| 13:08.36 | [TK]D-Fender | sLumPia: exten => 338,1,MeetMe,6338(1234) <- bad formatting |
| 13:09.03 | [TK]D-Fender | sLumPia: And you did not provide the SIP debug output I requested |
| 13:09.49 | sLumPia | [TK]D-Fender, CLI show no output |
| 13:10.05 | Katty | poor mister fender. |
| 13:10.11 | [TK]D-Fender | sLumPia: With sip debug enabled you see NOTHING? |
| 13:10.33 | sLumPia | yes |
| 13:10.47 | yang | sLumPia: set verbose 10 |
| 13:10.58 | sLumPia | already did |
| 13:11.37 | [TK]D-Fender | sLumPia: I said SIP DEBUG, not VERBOSE |
| 13:12.46 | dworkin | what's the most suitable Asterisk GUI for administering up to 2000 extensions/users and 50 conference rooms? |
| 13:12.48 | sLumPia | <[TK]D-Fender> sLumPia: then do "sip no debug" to stop the debug roll. <-- this is it, right? |
| 13:13.22 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:13.35 | *** join/#asterisk gego (n=gego@dyndsl-091-096-101-254.ewe-ip-backbone.de) |
| 13:13.38 | [TK]D-Fender | sLumPia: and did you do "sip debug" at the start like I told you to? |
| 13:14.33 | *** join/#asterisk jasonwoot (n=jasonrot@user-69-73-40-171.knology.net) |
| 13:14.43 | sLumPia | [TK]D-Fender, my bad. i didn't read the line above that |
| 13:14.50 | sLumPia | [TK]D-Fender, wait a minute |
| 13:16.07 | *** join/#asterisk theagent (n=theagent@pool-71-118-202-9.lsanca.dsl-w.verizon.net) |
| 13:16.16 | *** join/#asterisk JayTee52 (n=jforde@207-67-84-188.static.twtelecom.net) |
| 13:16.35 | theagent | anyone using IPKALL via IAX to asterisk? I |
| 13:16.46 | sLumPia | [TK]D-Fender, err...pastebin cann't accept the output |
| 13:17.10 | *** join/#asterisk doolph (n=doolph@201.218.103.171) |
| 13:17.24 | Katty | how do i set a variable in extensions conf? |
| 13:17.34 | Katty | Set moo=something |
| 13:17.45 | tzafrir | Set? |
| 13:17.47 | theagent | can someone tell me what should be in the username field for IPKALL IAX? |
| 13:17.51 | Katty | tzafrir: what's it look like? |
| 13:17.53 | Katty | jbot: set |
| 13:17.54 | jbot | i guess set is xbmodder is a world record setting 14 year old awesomely great guy |
| 13:17.59 | Katty | dangit. |
| 13:18.00 | _ShrikE | Set(moo=something) |
| 13:18.04 | Katty | _ShrikE: thank you deary |
| 13:18.16 | _ShrikE | :) |
| 13:18.23 | Katty | the only think i've set until now SetCallerID(something) |
| 13:18.41 | Katty | SetCallerID(ANSWERTHEGODDAMNPHONE) |
| 13:18.54 | JayTee52 | lol |
| 13:18.55 | tzafrir | Katty, see application Set and function SET |
| 13:19.22 | Katty | tzafrir: okie dokie. |
| 13:19.30 | tzafrir | That would be: Set(CALLERID(num)=123) |
| 13:19.48 | Katty | i want to pull the number of the DID |
| 13:19.55 | sLumPia | [TK]D-Fender, sorry. here's sip debug output: http://gkjdsh.multiply.com/journal/item/19/pastebin |
| 13:20.00 | Katty | so Set(didnumber=whateveritypehere) |
| 13:20.35 | Katty | and then echo -e "'${didnumber \\r and stuff'" | smbclient -M(?) nameofputer |
| 13:20.56 | Katty | more or less. |
| 13:21.13 | Katty | or stuff it into mutt. |
| 13:21.21 | Katty | or maybe send to openfire |
| 13:21.25 | Katty | i dunno what i'm doing with it yet |
| 13:23.55 | [TK]D-Fender | sLumPia: Looking for 6338 in voiplan (domain 10.14.15.29) |
| 13:24.04 | [TK]D-Fender | sLumPia: SIP/2.0 404 Not Found |
| 13:24.18 | [TK]D-Fender | sLumPia: 6338 is a ROOM #, not an EXTENSION in your dialplan. |
| 13:24.53 | [TK]D-Fender | sLumPia: You don't dial rooms, you dial extensions. Hopefully one that will call Meetme like you want |
| 13:25.00 | santoshr | can asterisk be made to act as a gatekeeper. i mean i want to register a h323 device to asterisk is it possible |
| 13:25.56 | x86 | my voicemail system keeps emailing me my voicemails with GMT time in the headers, so my email shows the message as received at GMT time instead of local time... how do I fix this? |
| 13:26.05 | sLumPia | [TK]D-Fender, if i don't dial room. how could i dial meetme? |
| 13:26.25 | [TK]D-Fender | sLumPia: You seem to have failed to grasp the concept of the dialplan. |
| 13:26.47 | [TK]D-Fender | sLumPia: Your dialplan (extensions.conf) sets up what #'s you can dial, and what they do. |
| 13:27.09 | [TK]D-Fender | sLumPia: you have an exten in your dialplan that will CALL MeetMe. use it. |
| 13:29.31 | sLumPia | [TK]D-Fender, pardon me. could you explain it again? i have an exten in my dialplan that will call MeetMe <-- by this you mean 6338, right? |
| 13:30.15 | [TK]D-Fender | sLumPia: exten => 338,1,MeetMe,6338(1234) |
| 13:30.39 | *** join/#asterisk InsolentDreams (n=Insolent@62.157.211.194) |
| 13:30.44 | [TK]D-Fender | sLumPia: Yuo set up an EXTENSION THAT YOU CAN DIAL to allow you to reach a MeetMe room. (mind you I told you your formatting was broken) |
| 13:31.17 | yang | I wonder if this is a proper way for sending faxes I have a hylafax<->IAXMODEM<->IAX2<->ASTERISK<->SIP<->DID PROVIDER SETUP |
| 13:31.25 | InsolentDreams | Hey guys, I have a dynamic queue system in place and when members transfer calls they can't get another call until the transferred call is hung up. I've found doing a reload fixes this, but is there another way around this? :\ |
| 13:31.47 | yang | the faxes don't want to go through i end with error: Unknown RTP codec 102 received from '80.246.224.123' |
| 13:32.44 | sLumPia | [TK]D-Fender, ok. then i should dial 338, right? |
| 13:32.58 | [TK]D-Fender | sLumPia: After fixing your call to MeetMe, yes |
| 13:33.52 | [TK]D-Fender | InsolentDreams: You have to use the DTMF transfer feature to transfer queue calls otherwise * won't free upt he agent. |
| 13:34.26 | [TK]D-Fender | InsolentDreams: is a "feature") of the way * handles queues/calls (*cough**BUG*) |
| 13:35.45 | santoshr | has anybody use asterisk as a gatekeeper |
| 13:36.03 | x86 | [TK]D-Fender: any ideas on my voicemail / timezone issue? |
| 13:36.17 | [TK]D-Fender | x86: nope |
| 13:36.20 | *** join/#asterisk ParanoyaM (n=Paranoya@145-245-179-213.ip.ukrtel.net) |
| 13:36.22 | ParanoyaM | hi |
| 13:36.26 | doolph | hi |
| 13:36.27 | ParanoyaM | guys i need some help |
| 13:36.28 | x86 | interesting |
| 13:36.50 | ParanoyaM | i am new in asterisk, and i read http://www.asteriskguru.com/tutorials/queues_conf.html |
| 13:36.51 | InsolentDreams | D-Fender: ok, don't know what that is, but I'll look it up and look into it.... thanks. Nice "feature" ;) |
| 13:37.29 | ParanoyaM | i have in sip.conf 5 profiles |
| 13:37.32 | [TK]D-Fender | InsolentDreams: "show application queue". You'll see the transfer option... |
| 13:38.15 | ParanoyaM | 2 for incoming and 3 for outgoing traffic |
| 13:38.32 | ParanoyaM | i need to make a equal load of this 3 outgoing profiles |
| 13:38.44 | ParanoyaM | can anybody help me? |
| 13:39.27 | doolph | ParanoyaM so you got total 5 channels? |
| 13:39.35 | ParanoyaM | for now |
| 13:39.38 | ParanoyaM | later will be more |
| 13:39.41 | ParanoyaM | outgoing |
| 13:39.58 | sLumPia | [TK]D-Fender, ok. you're right. when i test it when unpassword room, it work. |
| 13:39.59 | doolph | is it analog lines? |
| 13:40.00 | *** join/#asterisk RoyK (n=roy@ip-96-51-149-91.dialup.ice.no) |
| 13:40.01 | InsolentDreams | D-Fender: I don't see any option in there to actually transfer. The agents are currently allowed to transfer already. |
| 13:40.22 | sLumPia | [TK]D-Fender, but i don't know the right format for protected room |
| 13:41.00 | [TK]D-Fender | InsolentDreams: You need to use ASTERISK'S DTMF transfer to properly pass off the call. To allow this you'll need to set the Queue option when you enter the queue |
| 13:41.10 | sLumPia | [TK]D-Fender, care to show me to right format? |
| 13:41.20 | [TK]D-Fender | sLumPia: "cor show application meetme" |
| 13:41.44 | InsolentDreams | I'll try playing with that |
| 13:41.45 | [TK]D-Fender | sLumPia: "core show application meetme" |
| 13:41.47 | *** part/#asterisk mocker (n=kyle@mocker.org) |
| 13:42.19 | ParanoyaM | doolph, |
| 13:42.21 | sLumPia | [TK]D-Fender, on asterisk cli? |
| 13:42.39 | *** join/#asterisk Corydon76-dig (i=eleven@pdpc/supporter/bronze/Corydon76-home) |
| 13:42.39 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 13:42.47 | [TK]D-Fender | sLumPia: Yes |
| 13:43.00 | doolph | what |
| 13:43.17 | sLumPia | [TK]D-Fender, there is no "core" command |
| 13:43.20 | ParanoyaM | can you help me? |
| 13:43.42 | [TK]D-Fender | sLumPia: the remove that word. We typically figure people are running a supported version of * |
| 13:43.44 | doolph | ParanoyaM are you using analog or E1? |
| 13:43.50 | ParanoyaM | sip |
| 13:44.07 | theagent | anyone using IPKALL via IAX to asterisk? |
| 13:44.08 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 13:44.08 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:44.23 | doolph | create a trunk and limit it only 2 channels |
| 13:44.50 | ParanoyaM | doolph, maybe you don't understand me |
| 13:45.51 | *** join/#asterisk ManxPower (n=manxpowe@91.sub-70-221-156.myvzw.com) |
| 13:45.54 | ParanoyaM | i have 2 incoming profiles and 3 outgoing and i need to make a load line |
| 13:45.58 | doolph | you have a sip trunk with only 5 channels, and you want 2 for incoming and 3 for outgoing |
| 13:46.14 | doolph | or you have 5 sip users |
| 13:46.33 | _ShrikE | ParanoyaM: you want to load balance traffic on your 3 outgoing lines? |
| 13:46.42 | ParanoyaM | yes |
| 13:47.36 | doolph | oh, that's different |
| 13:48.04 | sLumPia | [TK]D-Fender, ok. i think i'll just google it. Thanks |
| 13:48.24 | ManxPower | With Zap channels, it's EASY to load balance between different lines/ports. With SIP it's impossible to do it simply. |
| 13:48.41 | [TK]D-Fender | sLumPia: I just pointed you to the exect command to get instructions on how to call MeetMe. just go READ it. |
| 13:48.43 | ParanoyaM | ManxPower, i understand |
| 13:48.54 | [TK]D-Fender | sLumPia: "show application meetme" |
| 13:49.15 | sLumPia | [TK]D-Fender, ok. Thanks |
| 13:49.19 | ManxPower | ParanoyaM: I would give up on the idea until you know Asterisk better. |
| 13:49.27 | *** join/#asterisk mocker (n=kyle@mocker.org) |
| 13:49.39 | doolph | :P |
| 13:49.47 | ParanoyaM | ManxPower, any example? maybe i will try to do similar? because i even can't find examples |
| 13:50.04 | ManxPower | ParanoyaM: It would take me an hour to even come up with an example. |
| 13:50.24 | doolph | why do you need load balance with sip trunks? |
| 13:50.30 | *** join/#asterisk FireMac (n=firemac@CPE000d88ae88b9-CM0011ae8bb0ee.cpe.net.cable.rogers.com) |
| 13:50.31 | *** part/#asterisk mocker (n=kyle@mocker.org) |
| 13:50.38 | doolph | or do you have 3 sip accounts |
| 13:50.42 | [TK]D-Fender | ParanoyaM: You have not told us what you are connecting to via SIP, how its set up to handle calls, how you want it to load-balance, etc. |
| 13:51.16 | ParanoyaM | ok i connect my softswitch with gsmgateways |
| 13:51.36 | [TK]D-Fender | ParanoyaM: What softswitch, and what gateways? |
| 13:51.41 | ManxPower | ParanoyaM: Your softswitch is Asterisk? |
| 13:52.19 | ParanoyaM | no another, Infinet |
| 13:52.28 | *** join/#asterisk [T]ank (n=ckwall@206.71.78.158) |
| 13:52.44 | ParanoyaM | and i have 3 or more gateways , i need to loadbalance a traffic over them |
| 13:53.09 | tzafrir | And the job of Asterisk in all of that is? |
| 13:53.12 | ManxPower | ParanoyaM: I feel sorry for you. |
| 13:53.28 | [T]ank | I am running into something I cannot pinpoint, but I am thinking that for some reason my asterisk agi is randomly not responding to some of my php scripts. is there a max thread in agi? |
| 13:53.49 | ParanoyaM | i don't need sorry i need a work queue in a 15 minutes or i will be fired up :) |
| 13:53.59 | doolph | lol |
| 13:54.03 | ManxPower | ParanoyaM: You are getting fired. |
| 13:54.07 | ParanoyaM | shit |
| 13:54.14 | ParanoyaM | no chances? |
| 13:54.25 | [T]ank | not in 15 minuntes |
| 13:54.33 | ParanoyaM | http://pastebin.ru/292609 |
| 13:54.36 | ManxPower | I doubt even a Digium developer will be able to do what you want to do in that timeframe. |
| 13:54.37 | [TK]D-Fender | ParanoyaM: With the quality of information you're providing us now? No. |
| 13:54.45 | ParanoyaM | i can give you all info |
| 13:54.52 | ParanoyaM | any i just don't know what you need |
| 13:55.00 | ManxPower | ParanoyaM: At your next job I suggest you do more testing before trying to deploy Asterisk. |
| 13:55.20 | [TK]D-Fender | ParanoyaM: we don't know how calls come into your system or exactly how * should decide what to do with them. |
| 13:55.35 | ManxPower | If you work on the problem 8 hours a day, I would expect you could have something working in a couple of days |
| 13:56.14 | *** part/#asterisk sLumPia (n=das@222.124.158.150) |
| 13:56.22 | ManxPower | ParanoyaM: First you talk about load balancing SIP accounts. Now you are talking about Queues. Which is it? |
| 13:56.27 | ParanoyaM | [TK]D-Fender, calls comes to my system from 2 profiles in sip.conf and * should nothing do with them only load balance on gateways |
| 13:56.56 | [T]ank | if you want true load balancing you probably want to look into something like SER |
| 13:56.59 | ManxPower | ParanoyaM: why are the calls not coming directly into your softswitch? |
| 13:57.21 | ManxPower | Yes, SER is designed to do all sorts of clever stuff with SIP. |
| 13:57.33 | ManxPower | Asterisk is overkill unless you need things like voicemail, etc. |
| 13:57.52 | [TK]D-Fender | ParanoyaM: show us your sip.conf & extensions.conf in a PASTEBIN |
| 13:57.54 | [TK]D-Fender | ~pb |
| 13:57.54 | jbot | [~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://www.rafb.net/paste |
| 13:57.55 | [TK]D-Fender | ^^^^^^^^^^^^^ |
| 13:58.02 | ParanoyaM | [TK]D-Fender, wait a moment |
| 14:02.00 | ParanoyaM | http://pastebin.ru/292610 |
| 14:02.03 | ParanoyaM | -sip conf |
| 14:02.51 | patrick-- | Hey everyone. Im havin problems with my Asterisk: I've got 2 HFC cards and im using misdn. i cannot seem to work out "Hold" on pressing the R button and getting the person back from hold . |
| 14:02.53 | ParanoyaM | http://pastebin.ru/292611 - extension conf |
| 14:04.23 | [TK]D-Fender | ParanoyaM: So you have 2 extens. 1 that will plead to 1 gateway, 1 to the other. |
| 14:04.37 | *** join/#asterisk mocker (n=kyle@mocker.org) |
| 14:04.41 | *** join/#asterisk R0land (n=Roland@193.227.191.90) |
| 14:04.42 | [TK]D-Fender | ParanoyaM: Would each of these gateways only be capable of handling 1 call at a time? |
| 14:04.42 | ParanoyaM | [TK]D-Fender, i need to make them to go through all gateways |
| 14:04.44 | R0land | hello all |
| 14:05.09 | ParanoyaM | no |
| 14:05.23 | [TK]D-Fender | ParanoyaM: well it'd be nice to know how it is they work.... |
| 14:05.26 | ParanoyaM | [TK]D-Fender, 4 and more call can be capable |
| 14:06.23 | [TK]D-Fender | ParanoyaM: So when someone dials 38063XXXXX what do you want * to do exactly? |
| 14:07.05 | ParanoyaM | [TK]D-Fender, i want both prefix and 38063 and 38093 went to gateways |
| 14:07.12 | ParanoyaM | but with load balancing |
| 14:07.25 | [TK]D-Fender | ParanoyaM: balanced how? |
| 14:07.34 | [TK]D-Fender | para when #1 fills up, jsut spill over to #2? |
| 14:07.59 | ParanoyaM | best variant if all channels service equal minutes |
| 14:08.02 | ParanoyaM | no |
| 14:08.04 | [TK]D-Fender | ParanoyaM: And never use the generic term "load balance" again please. |
| 14:08.20 | [TK]D-Fender | ParanoyaM: You want to balance based on TIME? |
| 14:08.32 | ParanoyaM | :) |
| 14:08.38 | [TK]D-Fender | ParanoyaM: is that a "yes"? |
| 14:08.42 | doolph | you want multiple things |
| 14:08.42 | ParanoyaM | NO |
| 14:08.45 | ParanoyaM | NO |
| 14:08.50 | ParanoyaM | I want simple |
| 14:09.19 | ParanoyaM | i want calls comes and gone on gsm1 next call on gsm2 next on gsm3 (if gateways 3) and then a cyclically |
| 14:09.20 | [TK]D-Fender | ParanoyaM: ParanoyaM>best variant if all channels service equal minutes <- explain |
| 14:09.34 | ParanoyaM | 4 call comes on 1st 5 on a second 6 on a third |
| 14:09.35 | *** join/#asterisk wmaulik (n=wmaulik@158.59.192.218) |
| 14:09.40 | ParanoyaM | 7 comes on a first |
| 14:09.45 | ParanoyaM | 8 on a second |
| 14:09.49 | ParanoyaM | 9 on 3d |
| 14:10.05 | ParanoyaM | [TK]D-Fender, i understand it is almost impossible |
| 14:10.12 | ParanoyaM | about equal minutes |
| 14:10.20 | [TK]D-Fender | ParanoyaM: And if the next gateway is in use? |
| 14:10.24 | ParanoyaM | so let's do each next call come next gateway |
| 14:10.28 | ParanoyaM | next |
| 14:10.38 | [TK]D-Fender | ParanoyaM: Ok, this is quite doable. |
| 14:10.48 | ParanoyaM | so if it busy or all channels at gateway are up then next gateway |
| 14:11.11 | ParanoyaM | thanks god |
| 14:11.13 | [TK]D-Fender | ParanoyaM: But will take a chunk of dialplan coding to do. It is too much to have us walk you through though. |
| 14:11.29 | [TK]D-Fender | ParanoyaM: Esp in the time-frame you have mentioned. |
| 14:12.05 | ParanoyaM | [TK]D-Fender, no matter now i can leave so, but i need to do it is as fast as possible |
| 14:12.46 | ParanoyaM | and we can do all manipulation with 063 |
| 14:13.33 | *** join/#asterisk Yourname`` (n=chatzill@unaffiliated/yourname/x-837320) |
| 14:15.12 | ParanoyaM | [TK]D-Fender, will you help me? |
| 14:15.36 | iCEBrkr | ParanoyaM: $50. |
| 14:16.06 | ParanoyaM | ? |
| 14:16.27 | [TK]D-Fender | ParanoyaM: I can, just not NOW, and this would typically fall under "consulting" for me as well. |
| 14:21.01 | ParanoyaM | [TK]D-Fender, may be i can read info, where i can find manuals on that or smth like this? |
| 14:21.15 | [TK]D-Fender | ParanoyaM: There is not such thing |
| 14:21.34 | [TK]D-Fender | ParanoyaM: This is you designing your own dialplan to pick the best way to handle your calls. |
| 14:21.56 | *** join/#asterisk sysreq` (n=sysreq@unaffiliated/sysreq) |
| 14:22.09 | [TK]D-Fender | ParanoyaM: YOU have have a "programmer" mentality to determine how best to do this via the application & functions available to the dialplan. |
| 14:22.13 | *** join/#asterisk SteveTotaro (n=root@pool-71-179-205-116.bltmmd.east.verizon.net) |
| 14:23.26 | ParanoyaM | i am not sure i get you |
| 14:23.47 | file | [TK]D-Fender: watching this is... interesting |
| 14:25.10 | [TK]D-Fender | ParanoyaM: there is no guide to tell you what to MAKE with your dialplan. This is up to YOU and your creativity based on the tools you are provided (the dialplan, the hardware you are interfacing with, etc) |
| 14:25.12 | ManxPower | Asterisk is NOT a PBX. Asterisk is a toolkit that allows you to build your own PBX, including programming the dialplan. |
| 14:25.33 | yang | When sending fax out (incoing works) I get result [Mar 26 15:03:56] NOTICE[11166]: rtp.c:1283 ast_rtp_read: Unknown RTP codec 102 received from '80.246.224.123' |
| 14:25.35 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-24-106-241-121.central.biz.rr.com) |
| 14:26.46 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 14:27.00 | bluregard | [TK]D-Fender, Did you get a chance to look at those SIP debugs yesterday? |
| 14:27.04 | ParanoyaM | I will have a big problems, please help me to make it load balance |
| 14:27.37 | [TK]D-Fender | ParanoyaM: This is too much work to just do for you from scratch. This will require consulting work on my side. |
| 14:27.52 | [TK]D-Fender | ParanoyaM: and iCEBrkr has already offered to take you up on it as well. |
| 14:28.07 | file | sits back and watches |
| 14:28.15 | ParanoyaM | i see |
| 14:28.18 | jblack | I'm reading up on asterisk realtime. |
| 14:28.24 | ParanoyaM | ok thanks for a listen to me |
| 14:28.33 | *** join/#asterisk Nasra (n=Nasra@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 14:28.39 | theagent | anyone use IPKALL? |
| 14:28.54 | jblack | I'm a little confused by http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip, though. The example calls for two tables; sipusers and sippeers. |
| 14:28.59 | [TK]D-Fender | jblack: I can't afford to read in real-time.... I wait till docs are cached :) |
| 14:29.08 | jblack | heh |
| 14:29.44 | *** join/#asterisk af_ (n=getsmart@88-149-230-191.dynamic.ngi.it) |
| 14:30.06 | jblack | Anyways, are they basically splitting sip accounts into two tables, one for phones and such, and another for sip peers that typically act like incoming lines? |
| 14:30.21 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 14:30.39 | [TK]D-Fender | jblack: Yeah, thats pretty much it. |
| 14:30.52 | theagent | peer = both directions |
| 14:30.54 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:30.54 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:31.08 | iCEBrkr | So what's another provider like VoicePulse? I get some really shitty packet loss/high latency to their network. |
| 14:31.13 | jblack | Awesome. |
| 14:31.22 | Nasra | [TK]D-Fender: hello... |
| 14:32.12 | patrick-- | Can someone tell me how Call transfer works? So if i press the R key, dial a number and put down the phone, the caller is automatically put through to the destination number? |
| 14:32.30 | iCEBrkr | . o O ( the R key? ) |
| 14:32.40 | jblack | Also, it looks like is http://www.voip-info.org/wiki-Asterisk+RealTime still up to date as concerns nat and qualify? (I ask because the page refers to "The Stable Branch of Asterisk 1.0" |
| 14:32.41 | bluregard | iCEBrkr, what kind of latency/packet loss are you getting with VP? |
| 14:33.11 | iCEBrkr | bluregard: well, 'mtr' shows 50% from my home location. I ran 'mtr' from a few other locations and it's 0% |
| 14:33.18 | patrick-- | iCEBrkr: yes, on ISDN Phones |
| 14:33.40 | iCEBrkr | ISDN? |
| 14:33.55 | iCEBrkr | I have shitty Brighthouse/Slime-Warner cable connection. |
| 14:34.04 | bluregard | iCEBrkr, which datacenter are you connecting to? |
| 14:34.14 | theagent | I hear good things about Via:Talk |
| 14:34.17 | iCEBrkr | bluregard: I've tried both connect02 and connect03 |
| 14:34.28 | iCEBrkr | bluregard: I'm not sure what other servers they have |
| 14:34.30 | [TK]D-Fender | iCEBrkr: .. |
| 14:34.34 | [TK]D-Fender | ~itsplist-us |
| 14:34.34 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, or http://www.jnctn.com, or http://www.bandwidth.com, or http://vitelity.net |
| 14:34.34 | iCEBrkr | I think the ones I'm hitting are in texas |
| 14:34.40 | bluregard | have you looked into switching to SIP using nyo and sfo.voicepulse.com? |
| 14:34.55 | *** join/#asterisk DarWin_vcch (i=red@unaffiliated/darwin) |
| 14:34.57 | *** part/#asterisk DarWin_vcch (i=red@unaffiliated/darwin) |
| 14:35.00 | iCEBrkr | bluregard: Don't know anythinga bout sfo.voicepulse.com |
| 14:35.05 | bluregard | I think all the connect0x servers are in NY |
| 14:35.08 | iCEBrkr | I've had great service with them up until now. |
| 14:35.10 | iCEBrkr | [TK]D-Fender: THanks |
| 14:35.13 | [TK]D-Fender | patrick--: what "R" key? On what phone? Connected to * exactly how? |
| 14:37.47 | iCEBrkr | Dude. It's Brighthouse |
| 14:37.53 | [TK]D-Fender | jblack: go hit up the 2nd Ed Book... |
| 14:38.09 | jblack | smacks his forehead |
| 14:38.19 | iCEBrkr | It doesn't seem to matter where I go, 3-4 hop introduces latency |
| 14:38.36 | bluregard | iCEBrkr, I had that problem with comcast from time to time |
| 14:38.46 | [TK]D-Fender | iCEBrkr: I'd say head to Tahiti then.. at least the wait will be pleasant ;) |
| 14:38.53 | iCEBrkr | Comcast in Coconut Creek, FL wasn't too bad |
| 14:39.00 | iCEBrkr | [TK]D-Fender: hehe |
| 14:39.16 | iCEBrkr | Brighthouse is the suck here man. |
| 14:39.27 | Katty | hi iCEBrkr! |
| 14:39.29 | Katty | hugs iCEBrkr |
| 14:39.47 | iCEBrkr | THey sent some 'tech' out to the house to 'fix' my modem. The stuff I was asking the guy about, he had no clue what I was talking about. |
| 14:39.50 | *** part/#asterisk [T]ank (n=ckwall@206.71.78.158) |
| 14:39.53 | km- | iCEBrkr: eisbrecher! ltns |
| 14:39.54 | iCEBrkr | Katty: You just gave me cooties. |
| 14:39.57 | bluregard | iCEBrkr, it was weird, Comcast should be going over Level3's network but for some reason every now and then they traverse AT&T networks and my traffic goes all to hell |
| 14:40.03 | iCEBrkr | km-: DUDE! Where've ya been?! |
| 14:40.04 | Katty | iCEBrkr: katty cooties are the best (= |
| 14:40.16 | km- | iCEBrkr: hiding, mostly on efnet. |
| 14:40.21 | iCEBrkr | km-: Ahhhh |
| 14:40.39 | km- | km-: stopped into lilofree a bit ago but it was desolate |
| 14:41.08 | km- | err. |
| 14:41.09 | iCEBrkr | km-: your nick came up the other day. Xunil was talking about meeting people way-back-when or something. |
| 14:41.14 | *** join/#asterisk delparnel (n=delparne@KTNRON06-1168103470.sdsl.bell.ca) |
| 14:41.17 | theagent | can someone try a direct iax connection to me? |
| 14:41.17 | iCEBrkr | km-: We're all on Slashnet these days |
| 14:41.35 | km- | iCEBrkr: ah, I'll have to fire up a window and come by, same channel? |
| 14:41.37 | bluregard | so anyone feel like giving me any insight as to why I'm not getting CID name with voicepulse? it's getting that I can't figure it out. |
| 14:41.51 | iCEBrkr | km-: yup |
| 14:42.05 | bluregard | getting irritating even |
| 14:42.05 | iCEBrkr | bluregard: It's hit or miss with them. |
| 14:42.07 | [TK]D-Fender | bluregard: last I recall they never supported it. |
| 14:42.19 | iCEBrkr | bluregard: and from what I can recall--- technically CID Name doesn't work with VOIP |
| 14:42.25 | patrick-- | [TK]D-Fender: the R key, aka Flash. The Phones are connected to the * via a HFC ISDN card |
| 14:42.30 | *** join/#asterisk duckz (n=duckz@81-180-102-217.etth.opensys.ro) |
| 14:42.30 | iCEBrkr | [TK]D-Fender: It's odd man. Sometimes I get CIDName |
| 14:42.35 | [TK]D-Fender | bluregard: Oh yeah and I remember seing your SIP debug... absolutely NO CID Name in their packets. |
| 14:42.48 | bluregard | [TK]D-Fender, yeah, they claim they support it using SIP |
| 14:43.02 | bluregard | and I saw it come through once or twice when I was using IAX |
| 14:43.14 | [TK]D-Fender | bluregard: We both saw the Invite packets... it wasn't there... try calling from a few other sources to test. |
| 14:43.21 | iCEBrkr | bluregard: It's really hit or miss. Sometimes I'll get CIDName others I won't. |
| 14:43.30 | [TK]D-Fender | bluregard: Perhaps its hit&miss as suggested... |
| 14:43.37 | bluregard | ok, as long as it isn't my end |
| 14:44.45 | iCEBrkr | bluregard: I was actually surprised to see CID Name showing up. I thought that stuff was done on the terminating side of the call. |
| 14:44.52 | iCEBrkr | or whatever.. the destination side... |
| 14:45.03 | iCEBrkr | <-- Programmer not a phone dude. |
| 14:45.50 | *** join/#asterisk akafurious (n=akafurio@gw1.pickeringcollege.on.ca) |
| 14:45.56 | bluregard | if CID is the biggest problem I have with voicepulse I think I'll live |
| 14:46.06 | iCEBrkr | lol |
| 14:46.24 | iCEBrkr | I dunno who the point the finger at on my end. |
| 14:46.41 | iCEBrkr | I get garbled calls, one way audio, skipping, etc.. |
| 14:46.52 | iCEBrkr | I suppose I could blame Brighthouse. |
| 14:47.22 | *** join/#asterisk eric_hill (i=EricHill@204.94.175.11) |
| 14:47.23 | bluregard | how many calls are in progress when it happens? |
| 14:47.29 | iCEBrkr | one. |
| 14:47.33 | bluregard | fun |
| 14:47.35 | iCEBrkr | hehe |
| 14:48.02 | iCEBrkr | I've even tried using the shittiest sounding codecs just to see if it's a bandwidth issue. |
| 14:48.23 | iCEBrkr | There's nothing I can do about it. I call Brighthouse and they're trying to sell me their 'Digital Phone' service. |
| 14:48.26 | bluregard | I had that same problem with an Avaya system and remote octel voicemail where all voicemails sounded like the caller was under water. |
| 14:48.28 | iCEBrkr | Yeah. No. |
| 14:49.13 | bluregard | it turned out that the frame relay was getting a shit load of errors from a bad cross connect somewhere along the line |
| 14:49.58 | iCEBrkr | http://www.cyberdyne.org/~icebrkr/mtr.png |
| 14:50.01 | iCEBrkr | I mean, loook at that junk |
| 14:50.56 | bluregard | I was reading somewhere that some of the ISPs that provide voice were blocking or intentionally causing dropped SIP packets |
| 14:51.08 | bluregard | damn |
| 14:51.09 | iCEBrkr | I was actually wondering that. |
| 14:51.13 | patrick-- | [TK]D-Fender: |
| 14:51.13 | patrick-- | ? |
| 14:51.21 | iCEBrkr | I could see them doing that. |
| 14:51.26 | bluregard | oh yeah |
| 14:51.50 | *** join/#asterisk fedya (n=fedya@75.112.143.226) |
| 14:51.51 | bluregard | shiesty bastards |
| 14:52.29 | eric_hill | Hi all. How do I get * to pass dialed DTMF digits through to a remote SIP voicemail system? When I push digits, * sees them but the remote side doesn't hear them? |
| 14:52.32 | patrick-- | [TK]D-Fender: any idea on my issue? |
| 14:52.43 | iCEBrkr | Now my home connection just dropped off the face of the planet. |
| 14:52.47 | iCEBrkr | <PROTECTED> |
| 14:53.11 | bluregard | but I guess technically running an * server is against most home ISP's TOS since 90% of them say you're not allowed to run any kind of server on their line |
| 14:53.31 | bluregard | to connect0x? |
| 14:55.32 | bluregard | I have 0% loss across the board to all 3 connect servers so I would guess it's your ISP |
| 14:56.11 | *** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net) |
| 14:59.04 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 15:00.00 | [TK]D-Fender | patrick--: I have no idea how MISDN handles calls. perhaps you can use that key... |
| 15:00.23 | patrick-- | Mhh, i wonder why no one ever uses misdn.. i cant seem to find a solution anywhere |
| 15:00.40 | [TK]D-Fender | eric_hill: You did not set your DTMF mode right for your remote side then. |
| 15:01.22 | *** join/#asterisk SwK (n=SwK@70-46-62-90.atl.fdn.com) |
| 15:01.27 | *** part/#asterisk SwK (n=SwK@70-46-62-90.atl.fdn.com) |
| 15:01.29 | iCEBrkr | Ok. |
| 15:01.33 | iCEBrkr | F Brighthouse |
| 15:01.35 | iCEBrkr | I'm done |
| 15:01.58 | *** join/#asterisk zeniffty2002 (n=zeniffty@mail.revenueworx.com) |
| 15:02.03 | iCEBrkr | bluregard: Naa, I can't even ssh into my box at home |
| 15:02.15 | patrick-- | another thing: how can i generate the "internal" dialtone? ive set my dialplan so that a 0 has to be dialled before outgoing calls can be made. all other calls are internally routed. how can i generate that typical internal dialtone? |
| 15:02.30 | iCEBrkr | I'm cancelling Brighthouse and seeing if I can get Knowlogy |
| 15:02.46 | iCEBrkr | Not sure how that'll all work. Since it's gonna come across the same cable. |
| 15:03.17 | *** join/#asterisk bsaxon (n=bsaxon@66.0.66.4) |
| 15:03.20 | bluregard | iCEBrkr, I had roadrunner when I lived in NC and never had a problem with it. That was like 4 years ago, but.. |
| 15:04.06 | iCEBrkr | bluregard: Brighthouse is Roadrunner |
| 15:04.31 | bluregard | yeah, I gathered that from the res.rr.com |
| 15:04.35 | znoG | is it possible to read digits from inside a macro? |
| 15:05.23 | bluregard | your connection may go over the same cable, but the networks are going to be different with a different provider, so you may have better luck may have worse |
| 15:05.40 | [TK]D-Fender | znoG: Read |
| 15:06.19 | [TK]D-Fender | patrick--: try adding "ignorepat => 0" in your context and see if it applies to MISDN as well. |
| 15:06.30 | bluregard | I can't see how Time Warner would get that kind of packet loss and just shrug it off though. Especially since it's happening with their border router at your first hop |
| 15:06.43 | iCEBrkr | bluregard: My buddy has Knowlogy and he has some crazy transfer speeds. 10mb down and almost 2mb up |
| 15:06.58 | iCEBrkr | I dont' think I can get Slime Warner in my neck of the woods. |
| 15:06.59 | bluregard | yeah, that's about what I get with Comcast |
| 15:07.23 | bluregard | Time Warner uses rr.com networks |
| 15:07.42 | bluregard | at least they did 4-5 years ago |
| 15:07.44 | iCEBrkr | So can I get internet from Knowlogy and TV from Brighthouse? lol |
| 15:07.50 | iCEBrkr | Mix and match. |
| 15:08.14 | bluregard | don't see why not, most cable companies rape you for internet if you don't get their video too |
| 15:08.22 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 15:08.39 | [TK]D-Fender | iCEBrkr: Wow.. 10/2 nice.. |
| 15:09.03 | [TK]D-Fender | iCEBrkr: wish I could get that reasonably here... beck 3/2 would be perfect... |
| 15:09.07 | [TK]D-Fender | heck* |
| 15:09.07 | bluregard | shitty 4mbps service through comcast is almost $60/month without a video subscription |
| 15:09.12 | iCEBrkr | I'm cancelling my TV service. I don't even watch it. I drive 106mi daily for work. By the time I get home, catch up on E-Mail, Digg, plays some TF-2, tinker/programmer.. It's time for bed. |
| 15:09.28 | iCEBrkr | [TK]D-Fender: It's insane. I dunno if he's lucky or what. |
| 15:09.36 | *** join/#asterisk bsaxon (n=bsaxon@66.0.66.4) |
| 15:09.38 | [TK]D-Fender | iCEBrkr: I've been cable-free for over 4 years now |
| 15:10.29 | iCEBrkr | If I really want to watch something, I can download it.. And Joost is entertaining for 10mins at a time. |
| 15:10.39 | atomspace | http://pastebin.ca/958154 - using this, dialing 93459700 yields: Call from '701' to extension '97902212' rejected because extension not found. |
| 15:10.59 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 15:11.02 | atomspace | correction: http://pastebin.ca/958154 - using this, dialing 97902212 yields: Call from '701' to extension '97902212' rejected because extension not found. |
| 15:11.18 | atomspace | any clue why? |
| 15:11.40 | iCEBrkr | Doesn't exist in your dialplan :) |
| 15:11.41 | iCEBrkr | 97902212 |
| 15:11.52 | *** join/#asterisk jpeeler (n=jpeeler@216.207.245.1) |
| 15:11.59 | atomspace | shouldn't that pattern be matched? |
| 15:12.18 | atomspace | in this line: exten=_9.,1,Macro(trunkdial,${trunk_1}/0${EXTEN:1},${trunk_1_cid}) |
| 15:12.50 | atomspace | _9. (pattern match, 9, anynumber of other digits) |
| 15:13.10 | iCEBrkr | I thought you needed XXXXXXX after it |
| 15:13.40 | atomspace | X = any digit from 1-9. .= any number of additional digits 1-9 |
| 15:13.45 | bluregard | atomspace, add at least 1 N before your dot I think |
| 15:13.47 | iCEBrkr | It's literally been YEARS since I touched my dial plan, other than to updated functions and deprecated items. |
| 15:13.50 | bluregard | X rather |
| 15:13.55 | atomspace | hmmm, I'll try that |
| 15:14.24 | patrick-- | [TK]D-Fender: in ger when you pickup the phone, you get a constant tone |
| 15:14.26 | atomspace | AH, "additional" means "additional" sorry |
| 15:14.48 | bluregard | atomspace, worked? |
| 15:14.55 | patrick-- | i want ther to be a beep beep beep <short pause> beep beep beep, etc... before you dial a zero for outgoing |
| 15:15.20 | [TK]D-Fender | atomspace: Depends.... is it even in the context thats LOOK for a match for your number? |
| 15:17.59 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-24-106-241-121.central.biz.rr.com) |
| 15:20.13 | *** join/#asterisk adjohn (n=adjohn@i60-35-186-224.s05.a013.ap.plala.or.jp) |
| 15:21.33 | *** join/#asterisk Bonix-BR (i=Bonix-BR@217-lo1.rt2.isimples.com.br) |
| 15:22.16 | *** join/#asterisk ManxPower (n=manxpowe@13.sub-70-221-107.myvzw.com) |
| 15:23.12 | *** join/#asterisk KillerQueen_86 (n=alan@201-212-100-188.cab.prima.net.ar) |
| 15:25.07 | KillerQueen_86 | i have a problem with the registration of a zaptel card in asterisk 1.6... is there a known bug for it? |
| 15:25.58 | ManxPower | KillerQueen_86: You get the award for "Being most vague" for this morning. |
| 15:26.16 | KillerQueen_86 | I get: "WARNING[9169]: channel.c:3367 ast_request: No channel type registered for 'Zap'" and i have a common setup |
| 15:26.30 | KillerQueen_86 | ~thwack ManxPower |
| 15:26.31 | jbot | ACTION hits ManxPower on the foot with a Holy Bible |
| 15:26.47 | ManxPower | KillerQueen_86: that means you did not compile zaptel before you compiled Asterisk, or the zaptel that is installed is old enough to not be recognized by Asterisk |
| 15:26.52 | [TK]D-Fender | KillerQueen_86: Have you considered installing a compatible version of Zaptel first? |
| 15:26.59 | ManxPower | screams "Oh dog it burns!" |
| 15:27.25 | KillerQueen_86 | i've installed all last versions of zaptel & asterisk |
| 15:27.55 | znoG | [TK]D-Fender: thanks. Is there such thing as Read() but using Background() instead of Playback() ? |
| 15:27.57 | ManxPower | KillerQueen_86: then remove the 1.6 asterisk and 1.6 zaptel and 1.6 libpri (if used) and compile and install in this order: zaptel, libpri, asterisk |
| 15:28.03 | ManxPower | znoG: no. |
| 15:28.05 | *** join/#asterisk Blackthorn (i=blacktho@76.77.160.10) |
| 15:28.10 | [TK]D-Fender | znoG: Nope |
| 15:28.14 | znoG | d'oh.. |
| 15:28.23 | KillerQueen_86 | zaptel-1.4.9 & asterisk-1.6.0-beta4 |
| 15:28.43 | ManxPower | KillerQueen_86: get zaptel 1.6 |
| 15:28.44 | atomspace | blueregard: not yet, but still piddling |
| 15:28.49 | defswork | still gets aastra blf problems with latest firmware :( |
| 15:28.57 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:29.11 | KillerQueen_86 | mmmh remove? doing apt-remove? will that kill all *.conf? |
| 15:29.21 | znoG | do I have some sort of alternative to "emulate" a similar thing to using Background? |
| 15:29.25 | ManxPower | You may have to get zaptel from svn, I don't know if they have 1.6 zaptel |
| 15:29.26 | znoG | from a macro |
| 15:29.36 | ManxPower | KillerQueen_86: we do not support asterisk/zaptel packages here. |
| 15:29.37 | [TK]D-Fender | ManxPower: Doesn't appear to be a Zaptel 1.6 series yet. Uses 1.4 |
| 15:30.00 | Blackthorn | Hi, i'm setting up a new asterisk service, and i've been over working in the ubuntu channel. I'm stuck on an issue. I can't get asterisk to start up as a service. I have a text file that I copyed from my old server over and placed in /etc/init.d |
| 15:30.02 | ManxPower | KillerQueen_86: asterisk packages frequently don't have zap support built. |
| 15:30.13 | KillerQueen_86 | that's what i mean... last vers... |
| 15:30.32 | ManxPower | KillerQueen_86: "last vers" is vague too. What is so hard about saying version numbers? |
| 15:30.52 | ManxPower | KillerQueen_86: uninstall the packages and build from source. |
| 15:30.53 | KillerQueen_86 | well, sorry... zaptel-1.4.9 & asterisk-1.6.0-beta4 |
| 15:31.04 | KillerQueen_86 | okay sir! |
| 15:31.07 | *** join/#asterisk Shotygun (n=thorn@213.31.43.3) |
| 15:31.12 | [TK]D-Fender | KillerQueen_86: And when you try loading the module at CLI? |
| 15:31.50 | KillerQueen_86 | <PROTECTED> |
| 15:32.06 | Shotygun | Hi. I have an issue with asterisk 1.2 where I'm trying to dial to a Snom phone (370 & 320) and I get "wrong password on authentication for INVITE". The phones can dial out just fine but I can't dial to them - any ideas? |
| 15:32.46 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 15:32.50 | ManxPower | KillerQueen_86: load chan_zap.sop |
| 15:32.55 | ManxPower | .. chan_zap.so |
| 15:33.56 | KillerQueen_86 | EXCELENT!!! IT WORKS!!! |
| 15:34.08 | ManxPower | KillerQueen_86: at least until you restart Asterisk |
| 15:34.10 | KillerQueen_86 | but, why this module is not automatically loaded? |
| 15:34.29 | ManxPower | KillerQueen_86: all Asterisk .so module loading is controlled by /etc/asterisk/modules.conf |
| 15:35.34 | KillerQueen_86 | ManxPower & [TK]D-Fender -> I owe you one...! |
| 15:35.38 | awk | hmm, how to limit agent to 1 when not using local channels? |
| 15:36.19 | awk | 1 call, i'm getting a second call coming in when status is (inuse) |
| 15:36.32 | awk | i've been using local agents and its easy to do... |
| 15:36.39 | KillerQueen_86 | thank you very much guys...! |
| 15:36.39 | awk | I don't want to set the exten to limit to 1 |
| 15:36.44 | ManxPower | KillerQueen_86: See all the time wasted by claiming you found a bug? |
| 15:37.53 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 15:39.13 | KillerQueen_86 | hhshaha, just a few seconds... and i've never claimed that... i've only asked if there was a known one |
| 15:43.42 | *** join/#asterisk johndo (n=hef@castor.tavros.net) |
| 15:43.50 | johndo | is there an asterisk realtime for users.conf? |
| 15:44.24 | mvanbaak | no |
| 15:45.05 | [TK]D-Fender | ~users.conf |
| 15:45.06 | jbot | users.conf is a flaming pile of sh1t that takes the fine control of several perfectly usable * config files and reduces them to the lowest common denominator and makes your system behave like a "toaster grade" PBX system. |
| 15:45.12 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 15:45.28 | ManxPower | MMmmm....toast |
| 15:45.36 | johndo | They have a toaster that makes phone calls? |
| 15:45.36 | mvanbaak | [TK]D-Fender: that must be your line ;) |
| 15:45.40 | johndo | Awesome. |
| 15:46.28 | ManxPower | johndo: almost nobody here uses users.conf and so don't expect much help with it. |
| 15:46.32 | [TK]D-Fender | johndo: "Toaster" is a derogatory word to describe a dumb appliance with limited features und functionality |
| 15:46.33 | johndo | fair enough |
| 15:46.53 | awk | grrrrrrrrrrrrrrrrr |
| 15:46.54 | johndo | [TK]D-Fender: I got it, I should have included <humor> tags or something |
| 15:47.00 | *** join/#asterisk SteveTotaro (n=root@pool-71-179-203-19.bltmmd.east.verizon.net) |
| 15:47.10 | awk | fscking queues dont read the state correctly, even if specified in queues.conf ringinuse = no |
| 15:47.11 | johndo | If its not the expected way, I will go back to sip.conf |
| 15:47.32 | johndo | well, working on asterisk realtime for sip.conf anyway |
| 15:47.58 | johndo | second question: is there a such thing as encrypted sip? |
| 15:48.03 | johndo | ~sip |
| 15:48.04 | jbot | sip is, like, http://www.cs.columbia.edu/sip/ X11 PPP dialer interface written in gtk+. URL: http://www.geocities.com/SiliconValley/Campus/3104/sip/ Session Initiation Protocol (see RFC 3261) |
| 15:48.16 | *** join/#asterisk m4sk4r4 (n=m4sk4r4@189.12.36.166) |
| 15:49.08 | johndo | also, where would be a good place to find the expected table columns for a realtime config, I can't be certain that voip-info is up to date. |
| 15:49.08 | BeckVOIP | how can I set Prefix on Asterisk server |
| 15:49.11 | atomspace | http://pastebin.ca/958204 - still getting: "chan_sip.c:13885 handle request invite: Call from '701' to extension '97902212' rejected because extension not found" |
| 15:49.21 | [TK]D-Fender | johndo: ... |
| 15:49.23 | [TK]D-Fender | ~book |
| 15:49.23 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 15:49.24 | ManxPower | atomspace: that's pretty clear message |
| 15:49.33 | awk | atomspace lol :) |
| 15:49.33 | BeckVOIP | because many Carriers around the world have got many Prefix |
| 15:49.34 | johndo | ~buybook |
| 15:49.34 | jbot | You can buy "Asterisk the Future of Telephony" at http://www.oreilly.com/catalog/9780596510480/ so go buy it SERIOUSLY |
| 15:49.41 | awk | cmon, can nobody help me with queue issue? |
| 15:50.00 | atomspace | "exten=_9X.,1,Macro(trunkdial,${trunk_1}/0${EXTEN:1},${trunk_1_cid})" should pick up on '97902212' |
| 15:50.03 | ManxPower | awk: If they could, they would have already |
| 15:50.08 | [TK]D-Fender | atomspace: I already told you its not looking in the right CONTEXT |
| 15:50.26 | ManxPower | atomspace: assuming it's in the right context. |
| 15:50.33 | awk | so i've found another bug? *sigh* from backporting state interface patch from 1.6 for local channels to moving to sip channels, queues on asterisk just sucks! |
| 15:50.34 | [TK]D-Fender | ~assume |
| 15:50.34 | jbot | About assumptions : http://www.moviewavs.com/php/sounds/?id=gog&media=WAVS&type=Movies&movie=Long_Kiss_Goodnight"e=assume.txt&file=assume.wav |
| 15:50.37 | [TK]D-Fender | ^^ |
| 15:51.01 | atomspace | no speakers, sorry |
| 15:51.14 | awk | whhy can't queues just understand ringinuse = no, don't send a call! |
| 15:51.21 | [TK]D-Fender | atomspace: its written out there too. |
| 15:51.38 | atomspace | ok, so its not getting to the context listed in that pastebin |
| 15:51.39 | [TK]D-Fender | SLJ FTW |
| 15:51.59 | [TK]D-Fender | atomspace: "duh" might summarize this situation well.... |
| 15:52.29 | [TK]D-Fender | atomspace: Time to look at your phone setup |
| 15:54.04 | *** join/#asterisk fuzzbawl (n=fuzzbawl@blackhole.cyberlinkint.com) |
| 15:54.52 | fuzzbawl | hey all. apparently I'm using incorrect syntax. show queue 2000 shows me members of the queue, but I can't seem to find the right syntax for remove queue memeber |
| 15:55.24 | fuzzbawl | i've tried giving it the channel "Local/3333" along with the context after it etc. but doesn't work |
| 15:55.54 | fuzzbawl | erp, nevermind, just figured it out. I was leaving out "from" before the queue number |
| 15:57.23 | atomspace | "its not looking in the right context" means that for some reason, the call isn't getting to "[numberplan-custom-2]", right? (http://pastebin.ca/958204) |
| 15:58.08 | ManxPower | atomspace: the only reason it would not get to numberplan-custom-2 is of the phone is not using "numberplan-custom-2" as it's userid |
| 15:58.25 | atomspace | oh jeese |
| 15:58.35 | atomspace | ok I think I follow |
| 15:58.39 | *** part/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 15:58.54 | atomspace | so, under each phone, I need to add something like "include=numberplan-custom-2" ? |
| 15:59.00 | atomspace | ... in ext..conf |
| 15:59.08 | ManxPower | atomspace: no, IN the phone config. |
| 15:59.41 | ManxPower | Whatever [is-in-here] is what the incoming call must authenticate as. |
| 15:59.57 | ManxPower | [this-is-your-sip-userid-for-incomiing-calls] |
| 16:00.08 | ManxPower | (in sip.conf, of course) |
| 16:00.10 | atomspace | I'm using sip phones, so in each phones config in sip.conf I need to add something, right? |
| 16:00.14 | atomspace | ok |
| 16:00.21 | atomspace | I'll look into that, thanx Manks |
| 16:00.36 | ManxPower | atomspace: NO! The phone config is setup on the phone however the phone does it. |
| 16:00.56 | ManxPower | you can't configure a phone from sip.conf |
| 16:01.10 | ManxPower | phone configuration is the job of provisioning, which is not part of Asterisk. |
| 16:02.46 | atomspace | so even though my phones can call/receive calls from other phones, and get to voicemail/voice menu in *, they still may not be setup correctly to make an outbound call? I didn't realize that the phone was the problem. I'll do some research into the phone then. |
| 16:02.49 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 16:03.07 | atomspace | thanks again Max |
| 16:03.10 | atomspace | Manx |
| 16:03.14 | ManxPower | atomspace: incoming and outgoing calls are totally separate in Asterisk |
| 16:06.01 | johndo | I have the first edition of the book, I just ordered the second edition. |
| 16:07.28 | ManxPower | some stupid site, cars.com has a URL with the string p0rn in it |
| 16:09.23 | atomspace | Manx: "chan_sip.c:13885 handle request invite: Call from '701' to extension '97902212' rejected because extension not found" -- in this error, the problem isn't '97902212', but its actually '701' that isn't found, right? I'm trying to make sure I understand here. |
| 16:09.42 | johndo | it look slike its the other way |
| 16:10.04 | ManxPower | atomspace: no, 701 is what the phone authenitcated as. |
| 16:10.06 | johndo | 710 looks like a user name |
| 16:10.17 | ManxPower | so what is the context= listed under [701] in sip.conf? |
| 16:10.22 | *** join/#asterisk sucram2g (n=markus@217.113.73.128) |
| 16:10.29 | johndo | and perhaps you need to dial ${EXTEN:1} to chop off the leading 9 |
| 16:10.45 | ManxPower | johndo: Kinda hard to do in the phone |
| 16:11.03 | johndo | heh, I meant in the extensionsc.conf |
| 16:11.06 | ManxPower | the call is being rejected by Asterisk, so you can't do it in the the dialplan. |
| 16:11.16 | johndo | *extensions.conf |
| 16:11.25 | ManxPower | can't do it there either |
| 16:11.47 | atomspace | Manx: in sip.conf: http://pastebin.ca/958227 |
| 16:12.21 | johndo | the context is probably wrong |
| 16:12.24 | ManxPower | atomspace: and you have exten => 97902212,1,Whatever in the [701] context in extensions.conf? |
| 16:13.04 | ManxPower | calls from a SIP phone land in whatever context is specified for that device in sip.conf |
| 16:13.05 | atomspace | I do not. 97902212 is an outside number, so I think I need to use some pattern matching |
| 16:13.33 | ManxPower | atomspace: Well the phone is trying to dial 97902212 and nothing matches that number in the context [701] in extensions.conf |
| 16:13.54 | ManxPower | Now you know what the problem and the fix should be pretty obvious as well. |
| 16:14.00 | johndo | can you pastebin the [710] context in extensions.conf |
| 16:14.08 | johndo | *[701] |
| 16:17.19 | *** join/#asterisk DarWin_laptop (n=ArigornS@unaffiliated/darwin) |
| 16:17.50 | *** join/#asterisk Leland (i=leland@ws2.discpro.org) |
| 16:17.52 | *** join/#asterisk AstPT (n=wewe@213.63.128.1) |
| 16:18.01 | AstPT | hi there |
| 16:18.08 | sucram2g | Hi all, am running asterisk 1.4.18.1 with chan_ss7 1.0.9 to connect to a telcom switch, however the ss7 link is NOT_ALIGNED and i also get the following NOTICE mtp.c:350 mtp_put: Full MTP receivebuf, event lost, type=15.. Any help is highly appreciated |
| 16:18.12 | DarWin_laptop | can anyone point me to where in sip.cfg you enable the call waiting beep for a polycom? |
| 16:18.45 | ManxPower | DarWin_laptop: calls per line appearance |
| 16:18.57 | ManxPower | it might be in phone1.cfg |
| 16:19.26 | DarWin_laptop | ok, manxpower, i'll check it out |
| 16:21.48 | DarWin_laptop | what's a good number? just say 99 for kicks? |
| 16:22.32 | ManxPower | Can your users handle 99 calls on the same line at the same time? |
| 16:23.09 | ManxPower | If so, I want their resume's. My users can barely handle ONE call at a time. |
| 16:23.35 | ManxPower | DarWin_laptop: you need to DOWNLOAD and READ the Polycom Admin Guide |
| 16:24.01 | DarWin_laptop | well, didn't do the trick anyways. you have a link to the guide? |
| 16:24.14 | AstPT | Does anyone have problems with te410p rev c ? |
| 16:24.18 | atomspace | johndo: http://pastebin.ca/index.php |
| 16:24.31 | ManxPower | DarWin_laptop: you have to reboot the phone and confirm it downloaded the updated file. |
| 16:24.41 | ManxPower | I go to polycom.com to get the admin guide |
| 16:24.56 | DarWin_laptop | manxpower, i did it in the web config as a temporary test |
| 16:25.07 | ManxPower | DarWin_laptop: I can't help you with that |
| 16:26.54 | *** join/#asterisk seanbright (i=seanbrig@65.207.74.18) |
| 16:26.55 | *** join/#asterisk AlephX (n=sj@c-69-138-21-96.hsd1.md.comcast.net) |
| 16:27.08 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
| 16:27.09 | ZPertee | how can I get asterisk to store passwords in the database? I'm trying to create an IVR so that asterisk can be managed from a handset so that should my customer have problems they can fix it without having to ssh in, etc. |
| 16:27.38 | ManxPower | ZPertee: ANY way you want to. |
| 16:28.03 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 16:28.10 | ManxPower | ZPertee: you could use astdb or any SQL server or flat file in an AGI or auth against voicemail.conf or use one off the database addins |
| 16:28.39 | ZPertee | ManxPower, how would get voicemail.conf synced with a database? |
| 16:29.12 | ManxPower | ZPertee: no, voicemail.conf IS the database. |
| 16:30.00 | ManxPower | "core show applications like auth" |
| 16:30.01 | x86 | ManxPower: unless he's doing realtime with voicemail |
| 16:30.22 | ManxPower | x86: VMAuthenticate should still work |
| 16:30.57 | ManxPower | ZPertee: I can't imagine why you would need to store the password in a database. Seems like overkill for a password. |
| 16:31.19 | x86 | ManxPower: that's how realtime does it... |
| 16:31.49 | x86 | which is great if you're wanting other applications to be able to connect to one database and manage all aspects of everything your asterisk server is doing |
| 16:32.21 | ZPertee | ManxPower, sorry don't think I made myself clear. What I would like to do is say EXT. 16 forgot their password. I'd like to be able to call in and reset their password. |
| 16:32.22 | x86 | I put SIP peers, IAX peers, voicemail users, dialplan, and everything else possible into a database |
| 16:32.23 | sucram2g | ManxPower, u need to recompile asterisk with support for the postgresql or whatever db you are going to use |
| 16:32.57 | x86 | sucram2g: or just use asterisk-addons, which gives you mysql support :p |
| 16:33.09 | x86 | sucram2g: only thing asterisk itself has to support is realtime |
| 16:33.31 | x86 | which, iirc, you can't really turn off support for |
| 16:33.39 | sucram2g | ya, sure |
| 16:34.03 | [TK]D-Fender | ZPertee: use VMAuthenticate instead of letting VoiceMailMain do it. |
| 16:34.08 | jblack | The job started off with "get me asterisk". Then, xen got thrown in, followed shortly by ltsp. Then, crm, which led to sql servers... ltsp encouraged samba, the pile of machines led to a local dns server, and the whole pile is demanding ldap. |
| 16:34.20 | znoG | x86: me too, only with LDAP :) |
| 16:34.36 | ZPertee | x86, what do you mean by getting realtime support turned on? how do I do that? |
| 16:34.54 | [TK]D-Fender | ZPertee: Or rather use your OWN method instead. This means you'll have to create your own PW maintenance extens as well |
| 16:35.19 | jblack | seventy hours of work, and they have a pile of servers, and a phone system that has barely progressed enough to play tt-monkeys |
| 16:36.24 | djs26 | jblack: Yikes. Do they really know what they want, or are you stuck trying to work out their pipe dreams? |
| 16:36.46 | jblack | Well, I was kind of bored anyways.... |
| 16:37.08 | *** join/#asterisk lftsy (n=lftsy@120.194.210.62.te-dns.org) |
| 16:37.45 | djs26 | jblack: Apparently, somebody heard you say that, and wanted to do more than oblige... |
| 16:37.50 | jblack | I'm not doing great on this bid, financially speaking, but I have to admit that it's _really_ nice design and implement a system that covers all the bases. |
| 16:38.07 | jblack | really nice to. Ya know.. without any old infrastructure in place. |
| 16:40.17 | jblack | * Kind of got put off while the political football went around about whether to use voip or PRI, and they dont' have the phones yet, which kind of limits development. |
| 16:40.29 | *** join/#asterisk hohum (n=dcorbe@70.6.107.85) |
| 16:40.45 | johndo | jblack: softphones work well for making an argument for voip |
| 16:41.13 | ZPertee | can I do realtime with ASTDB? how does this realtime stuff work I have never even heard of it before? |
| 16:41.26 | jblack | Oh, internally, voip all the way. Probably skip on the polycoms for the sales team and move them to a soft phone w/ a headset on the thin clients. |
| 16:41.46 | johndo | ZPertee: instead of using some of the .conf files, you store the configs in a database |
| 16:41.55 | johndo | the realtime part comes in when asterisk does the lookup |
| 16:42.07 | johndo | ZPertee: I don't think it can be done with ASTDB |
| 16:42.20 | johndo | you need ODBC or mysql or something like them |
| 16:42.34 | jblack | speaking of which, I need to find a soft phone that works well with ubuntu. |
| 16:42.53 | johndo | I usually use gnomemeeting |
| 16:42.57 | *** part/#asterisk file (n=file@asterisk/developer-and-muffin-lover/file) |
| 16:43.00 | johndo | err ekiga |
| 16:43.02 | *** join/#asterisk file (n=file@asterisk/developer-and-muffin-lover/file) |
| 16:43.02 | *** mode/#asterisk [+o file] by ChanServ |
| 16:43.17 | johndo | ~polycom |
| 16:43.18 | jbot | methinks polycom is the manufacturer of one of the best IP phones in the market. http://polycom.com - Note: Here is where you can get some downloads: http://www.polycom.com/resource_center/0,,pw-6812-12612,00.html |
| 16:43.30 | johndo | wait, someone likes polycoms? |
| 16:43.31 | johndo | why? |
| 16:43.32 | ZPertee | johndo, can I choose which which configs to include in th DB or is it an all or nothing kinda deal? |
| 16:43.37 | jblack | Ekiga, I don't trust because of problems here , and the others I've fiddled with in the past don't feel like they'd hold up under sales monkey abuse. |
| 16:43.38 | *** join/#asterisk NoamRotter (n=noam@mail.browarnik.com) |
| 16:43.46 | johndo | ZPertee: you choose in extconfig.conf |
| 16:44.02 | johndo | sales monkey's use ubuntu? nice |
| 16:44.07 | jblack | Thin clients. :) |
| 16:44.13 | johndo | very nice |
| 16:44.37 | johndo | ~linksys |
| 16:44.39 | jbot | methinks linksys is a tool of satan |
| 16:44.39 | jblack | Well, if one of the monkeys gets really unhappy, you don't want him to start throwing around expensive machines. :) |
| 16:44.49 | johndo | ~cisco |
| 16:44.49 | jbot | cisco makes the routers that move the pr0n across the internet at a very high rate of exchange. a company full of patent protected punks!, or <reply>Cisco phones are expensive crap which should be avoided with extreme prejudice |
| 16:44.49 | twisted | hey file, got a quick sec? |
| 16:44.50 | jblack | jbot++ |
| 16:44.52 | johndo | ~sipura |
| 16:44.53 | jbot | Sipura has been aquired by cisco, see ~cisco and ~linksys . |
| 16:45.19 | johndo | I like the linksys spa941 phones |
| 16:45.32 | znoG | jbot: you tried zoiper? |
| 16:45.34 | jblack | Hey! I think they fixed the ekiga config segfault well enough that they tell you what's wrong. Nice |
| 16:45.38 | johndo | I don't like the polycom or cisco phones |
| 16:45.41 | jblack | znog: This jbot? |
| 16:45.53 | znoG | jblack: that was meant for you :) |
| 16:46.10 | johndo | ~zoiper |
| 16:46.11 | jbot | [~zoiper] Zoiper (Formerly known as Idefisk) is a free SIP and IAX soft-phone for Windows, MacOSX, and Linux that can be found at http://www.zoiper.com |
| 16:46.13 | ZPertee | johndo, given that the name is "realtime" does that mean that I can change values in the DB and not have to restart Asterisk? also which DB do you recommend using. I'm familair with DB concepts but not one DB more than another |
| 16:46.21 | jblack | never mind. |
| 16:46.23 | jblack | jblack@comet:~$ ekiga |
| 16:46.23 | jblack | Segmentation fault (core dumped) |
| 16:46.26 | znoG | ZPertee: correct. |
| 16:46.33 | jblack | gives ekiga a raspberry |
| 16:46.37 | johndo | ZPertee: yes, it means that |
| 16:46.39 | ZPertee | znoG, SWEET! |
| 16:46.50 | ZPertee | any DB better than another? |
| 16:46.52 | johndo | ZPertee: I don't know, I am working with mysql, I don't think it matters which one you go with |
| 16:46.56 | znoG | i'm not sure the status of res_pgsql |
| 16:47.14 | znoG | probably best to go with res_mysql and it looks like it's more actively developed, but I could very well be wrong |
| 16:47.21 | znoG | s/and/as |
| 16:47.23 | jblack | Hmm. I don't see zoiper in ubuntu's package cache. Perhaps elsewhere |
| 16:47.36 | johndo | I used wengophone for a while |
| 16:47.40 | znoG | jblack: you go to www.zoiper.com and download it :) |
| 16:47.53 | johndo | it's not as generic as I would like, but it did work for me |
| 16:47.57 | jblack | I don't suppose it's free software by any chance |
| 16:48.14 | *** join/#asterisk c0rnflake (n=c0rnflak@38.112.4.210) |
| 16:48.18 | [TK]D-Fender | jblack: Freeware, yes |
| 16:48.58 | jblack | I'm one of those people that warps "free" to mean not free in cost, but free in licensing, source availability, etc. |
| 16:49.21 | ZPertee | johndo and znoG: ok thanks! I think that I will use mysql especially since a lot of the other programs that I use have a mysql backend...probably not a bad idea to learn it |
| 16:49.22 | Strom_C | it's free as in beer, not free as in can't you shut up about the god damned gpl already |
| 16:49.31 | [TK]D-Fender | jblack: I used a term with a very specific known meaning :) |
| 16:50.11 | jblack | Heh. |
| 16:50.25 | johndo | I think ekiga was the best open source one I came across |
| 16:50.39 | johndo | it goes downhill pretty fast from there |
| 16:52.36 | *** part/#asterisk KillerQueen_86 (n=alan@201-212-100-188.cab.prima.net.ar) |
| 16:54.31 | *** join/#asterisk HumanCell (n=HumanCel@207.224.214.130) |
| 16:55.51 | *** join/#asterisk delparnel (n=delparne@CPE001839f2889c-CM00137185dfb6.cpe.net.cable.rogers.com) |
| 16:56.01 | HumanCell | Q: Is there a hardware device available anywhere that works with Asterisk, that is a remote phone-jack-over-IP? |
| 16:57.30 | Strom_C | HumanCell: yes |
| 16:57.38 | znoG | not sure they've been invented yet ;) |
| 16:58.53 | Katty | i'm having some problems with Asterisk saying Page() is in an Up State still |
| 16:59.01 | Katty | and the SIP phone has finished the call |
| 16:59.17 | Katty | anyone have any ideas on what I can look for? |
| 17:01.32 | HumanCell | Strom_C: I know they are out there ... what do I Google on? What are they called? Who sells one that you might recommend? |
| 17:01.47 | Strom_C | HumanCell: have a look at the spa-3102 or equivalent |
| 17:03.09 | RobH | wouldnt the IAXy devices work too for what he is asking about? |
| 17:03.47 | RobH | HumanCell: I think the IAXy devices would also do what you need, but not sure. http://store.digium.com/products.php?category_id=9 |
| 17:04.11 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 17:04.57 | _ShrikE | Katty: Does it to that every time you page? |
| 17:05.53 | HumanCell | reading web pages ... |
| 17:06.18 | *** join/#asterisk VaNNi (n=VaNNi@lgb-static-216.70.165.200.mpowercom.net) |
| 17:08.23 | delparnel | For some reason.. Callback is only working for me when I do an amportal restart and my shell is still open in that verbose mode.. if I close my shell.. Callback stops working.. any suggestions? |
| 17:09.27 | Katty | _ShrikE: i don't know about everytime...but it appears to have several SIP/extension-blah channels going |
| 17:09.32 | Katty | _ShrikE: all with Page() |
| 17:09.40 | Katty | _ShrikE: if not all of them, most of them for sure |
| 17:10.35 | *** part/#asterisk airjump (n=zielonka@62.159.95.82) |
| 17:11.39 | *** part/#asterisk mike345 (n=mike_sim@64.74.198.10) |
| 17:12.26 | HumanCell | Thank-you Storm_C and RobH ... that is exactly what I was after ... |
| 17:12.31 | twisted | gets pissed at 1.4 |
| 17:12.35 | RobH | glad to help |
| 17:12.45 | RobH | I have not messed with either though, just FYI. |
| 17:12.48 | HumanCell | Oh ... and thanks znoG also ... quite humorous ... :-) |
| 17:12.52 | twisted | BUILD RES_ODBC YOU PIECE OF SHIT |
| 17:13.11 | HumanCell | RobH: NP ... I like to experiment ... it *claims* do do what I want ... |
| 17:13.22 | RobH | the linksys seems to be sip |
| 17:13.28 | RobH | iaxy is of course iax2 |
| 17:13.30 | Katty | _ShrikE: http://pastebin.ca/958304 thoughts? |
| 17:13.47 | HumanCell | I have to route a line to a remote PBX the iaxY seems like the one I was after ... |
| 17:13.48 | RobH | if yer connection to a * server, the iaxy is usually less of a pain in the ass in regards to firewalls and routing. |
| 17:14.04 | RobH | the iax2 i mean |
| 17:14.07 | HumanCell | That is exactly what I just read ... seems to be the easier choice ... |
| 17:15.39 | [TK]D-Fender | Katty: Please pastebin all of the dialplan related to paging... |
| 17:15.59 | [TK]D-Fender | Katty: because right now it looks like multiple people are calling Page |
| 17:16.13 | Katty | [TK]D-Fender: they did. it's being held open. |
| 17:16.22 | Katty | [TK]D-Fender: i don't believe there's anymore 'relevant' stuff to the pastebin (= |
| 17:16.26 | *** join/#asterisk henrique (n=henrique@unaffiliated/henrique) |
| 17:16.49 | [TK]D-Fender | Katty: Expand your beliefs then. |
| 17:17.12 | Katty | [TK]D-Fender: you're awfully snippy today. |
| 17:17.21 | Katty | [TK]D-Fender: please check your attitude at the door, mkay (= |
| 17:17.40 | [TK]D-Fender | Katty: That wasn't attitudes, thats jsut me telling you the you need to open up. |
| 17:17.55 | Katty | [TK]D-Fender: then precisely what else are you looking for? |
| 17:18.42 | znoG | HumanCell: :) i personally use a Sipura (now Linksys) and they work reasonably well |
| 17:19.28 | [TK]D-Fender | Katty: more :) |
| 17:20.18 | [TK]D-Fender | Katty: CLI output, more dialplan, etc |
| 17:23.39 | Katty | [TK]D-Fender: http://pastebin.ca/958317 is that more to your liking? |
| 17:27.37 | *** join/#asterisk trippss (n=tripps@c-76-31-153-101.hsd1.tx.comcast.net) |
| 17:28.09 | ZPertee | can someone see what I am doing wrong? I have a Linksys SPA8000 that I am trying to register with Asterisk. Here http://pastebin.ca/958310 is my users.conf file. User 18 is registering but User 3 will not. Here is the screen shots of spa configuration (http://img505.imageshack.us/my.php?image=line3vj1.png and http://img259.imageshack.us/img259/9290/dennypa1.png) |
| 17:33.30 | *** join/#asterisk bronson (n=bronson@adsl-68-122-117-135.dsl.pltn13.pacbell.net) |
| 17:33.38 | ManxPower | ZPertee: we don't use users.conf here. |
| 17:35.42 | ZPertee | ManxPower, why? Who sets the rules? |
| 17:35.45 | *** join/#asterisk bminish (n=bminish@78.152.237.99) |
| 17:37.55 | ZPertee | ManxPower, are you saying that you personally don't use it or that this irc channel doesn't help support it? |
| 17:39.14 | mvanbaak | ~users.conf |
| 17:39.14 | jbot | users.conf is a flaming pile of sh1t that takes the fine control of several perfectly usable * config files and reduces them to the lowest common denominator and makes your system behave like a "toaster grade" PBX system. |
| 17:39.40 | mvanbaak | that's why |
| 17:40.37 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 17:41.25 | ManxPower | ZPertee: virtually nobody here uses it. |
| 17:41.39 | ManxPower | So asking questions about it here is pretty useless. |
| 17:43.00 | *** join/#asterisk hmmhesays (n=hmmhesay@24-119-176-74.cpe.cableone.net) |
| 17:43.23 | [TK]D-Fender | ZPertee: And it might have been nice to see the error itself. |
| 17:43.38 | [TK]D-Fender | Katty: Looks better... seems fine too. |
| 17:43.48 | ZPertee | ManxPower, ok thanks. I started using AsteriskNow but that wasn't working out so I switched to Ubuntu and I tried to minimize the amount of config switching that I had to do. which is why I just kept my users.conf file |
| 17:43.50 | Katty | [TK]D-Fender: so why do you suppose Page() is still open? |
| 17:44.04 | Katty | [TK]D-Fender: anthm seems to think it's a bug. |
| 17:44.04 | [TK]D-Fender | Katty: maybe the pager didn't hang up? |
| 17:44.16 | ManxPower | ZPertee: users.conf seems to be only used by AsteriskNOW |
| 17:44.23 | [TK]D-Fender | Katty: If he thinks so I'd sooner believe that... |
| 17:44.27 | *** join/#asterisk djs26 (n=djs@unaffiliated/djs26) |
| 17:44.29 | ManxPower | ZPertee: if you want real asterisk then start from scratch for your config files. |
| 17:44.38 | [TK]D-Fender | ManxPower: Or the GUI more like. |
| 17:44.57 | [TK]D-Fender | ManxPower: If primarily pimped by it anyways, but it is a base config file... |
| 17:45.02 | [TK]D-Fender | (shitty as it is) |
| 17:48.39 | *** join/#asterisk RobH (n=RobH@cpe-72-184-129-136.tampabay.res.rr.com) |
| 17:53.04 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:56.00 | *** join/#asterisk tomierna (n=newbie@pool-72-77-228-190.tampfl.fios.verizon.net) |
| 17:56.27 | tomierna | anyone here have and use the aa50 (digium appliance)? |
| 17:57.57 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 17:59.33 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 17:59.42 | Qwell | tomierna: I have one sitting in front of me, but I probably don't count. |
| 17:59.52 | Qwell | tomierna: [hC] has one though. |
| 17:59.55 | Qwell | throws [hC] under the bus |
| 18:00.14 | [hC] | I am actually a robot |
| 18:00.31 | [hC] | ... what are we talking about? |
| 18:00.36 | Qwell | <tomierna> anyone here have and use the aa50 (digium appliance)? |
| 18:00.46 | [hC] | ah |
| 18:00.57 | [hC] | yes i do have a few :P |
| 18:01.20 | [hC] | much better than what i presumed you were talkin about |
| 18:01.29 | [hC] | (7970) |
| 18:01.35 | JenniferAkemi | my boss seems to think that he read somewhere that there is database failover to a replicant built into asterisk in a future release in the trunk somewhere. I can't find it though, does anyone know what he is talking about? |
| 18:06.23 | Olobola | I have a sip client specific problem: regular calls work fine. When I call voicemail, the audio gaps and comes in out of order. |
| 18:06.24 | *** join/#asterisk adjohn (n=adjohn@i60-35-186-224.s05.a013.ap.plala.or.jp) |
| 18:06.57 | Olobola | This is happens when I check voicemail. |
| 18:07.12 | Olobola | It works fine with other clients. |
| 18:07.27 | _ShrikE | JenniferAkemi: There is support for something like that in func_odbc. |
| 18:07.33 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 18:07.45 | *** join/#asterisk Qtpaxa (n=Qtpaxa@84.120.155.80.dyn.user.ono.com) |
| 18:08.06 | *** part/#asterisk Qtpaxa (n=Qtpaxa@84.120.155.80.dyn.user.ono.com) |
| 18:08.18 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 18:10.13 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 18:11.59 | *** join/#asterisk zettairedo (n=kamus@host-148-244-226-54.block.alestra.net.mx) |
| 18:12.26 | *** join/#asterisk marksaitis (n=maxxx@80-46-86-99.static.dsl.as9105.com) |
| 18:14.03 | *** part/#asterisk HumanCell (n=HumanCel@207.224.214.130) |
| 18:14.36 | marksaitis | is it possible to use asterix commercially ? |
| 18:14.51 | tomierna | hC: so i'm looking to do followme on the aa50. is it possible? |
| 18:15.10 | [hC] | tomierna: not out of the box, youd have to edit the dial plan yourself. |
| 18:15.39 | Qwell | marksaitis: what is asterix? |
| 18:15.42 | tomierna | would i use something like [4444] |
| 18:15.42 | tomierna | music=>default |
| 18:15.42 | tomierna | context=>default |
| 18:15.42 | tomierna | number=>1965751234,5 |
| 18:15.42 | tomierna | number=>17182025678,20 |
| 18:15.42 | tomierna | number =>FM1/${ARG1} |
| 18:15.44 | tomierna | number =>FM2/${ARG1} |
| 18:15.46 | tomierna | gr |
| 18:15.49 | marksaitis | I mean asterisk Qwell |
| 18:15.50 | tomierna | sorry, wrong pastebuffer |
| 18:15.56 | *** join/#asterisk m4sk4r4 (n=m4sk4r4@189.12.36.166) |
| 18:16.02 | Qwell | marksaitis: why wouldn't it be possible? |
| 18:16.07 | tomierna | would I use something like this: http://www.voip-info.org/wiki/view/Asterisk+cmd+FollowMe |
| 18:16.21 | tomierna | (using the Example 4?) |
| 18:17.00 | tomierna | and, if so, why is the syntax different between the stuff i see in the aa50's conf files compared to what i see on this page? |
| 18:17.09 | *** join/#asterisk bsaxon (n=bsaxon@66.0.66.4) |
| 18:17.29 | marksaitis | Qwell, well I know nothing about it, I have a very simple plan, open a small web page, host asterisk with the proxies and all the other stuff on a fast server, and start selling out VoIP for companies |
| 18:17.51 | *** join/#asterisk real0ne (n=Ismail@41.251.87.122) |
| 18:18.06 | *** join/#asterisk bsaxon (n=bsaxon@66.0.66.4) |
| 18:18.13 | real0ne | ~book |
| 18:18.13 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 18:18.16 | marksaitis | so thats why I am wondering if it would be possible to use it to make money |
| 18:18.18 | joe | I have a switch via an iax2 trunk that is not working due to network issues. is the best way to setup a play message for maintenance if so anyone have the name of the soundbite that is nicer than the number you have dialed is not in service |
| 18:18.25 | [hC] | tomierna: that should work yeah. and i dont know what you mean by the syntax being different |
| 18:18.44 | tomierna | in the aa50 conf files, it doesn't seem to use => |
| 18:18.46 | tomierna | just = |
| 18:19.19 | JenniferAkemi | _ShrikE: thanks i'll read about that then :) |
| 18:19.38 | [hC] | tomierna: just a style. you dont HAVE to use => in non aa50 config files either |
| 18:20.09 | tomierna | oh, cool. i've only ever seen it with the =>. i'm kinda new :) |
| 18:20.11 | marksaitis | Qwell, would I be legal to do so ? |
| 18:20.21 | *** join/#asterisk kannan (n=kannan@121.241.156.83) |
| 18:20.29 | Qwell | marksaitis: hire a lawyer to read the GPL, and ask his advice. |
| 18:20.47 | Qwell | but, yes |
| 18:20.50 | [hC] | Qwell: oh thats just mean. |
| 18:20.51 | [hC] | :P |
| 18:20.56 | [hC] | but i love it. |
| 18:20.57 | Qwell | assuming you actually follow it |
| 18:21.53 | marksaitis | can you give me the link where to read about that ? you see, I have never needed to know nothing about GPL... |
| 18:22.00 | Qwell | ~google |
| 18:22.01 | jbot | hmm... google is a search engine found at http://www.google.com/ |
| 18:22.09 | marksaitis | ok |
| 18:22.14 | marksaitis | thank you |
| 18:23.43 | real0ne | ~debian |
| 18:23.43 | jbot | [debian] the hobby project of bruce perens that flew on the space shuttle |
| 18:23.56 | Qwell | eh? |
| 18:24.04 | real0ne | Qwell lol |
| 18:24.40 | JenniferAkemi | is it possible to use func_odbc instead of sip.conf or realtime sip stuff to define sip accounts? |
| 18:25.20 | JenniferAkemi | seems kind of convoluted though |
| 18:27.16 | tomierna | hC: one more Q - i see that app_followme.so is sitting in /usr/lib/asterisk/modules, but it doesn't seem to be registered (by using 'core show applications' from the CLI). I have autoload = yes in my modules.conf |
| 18:27.39 | [hC] | tomierna: im not sure, i dont use app_followme on the aa50 |
| 18:27.46 | jameswf-home | no need to google your self |
| 18:27.53 | tomierna | i'm just wondering if that's a non-working stub or something |
| 18:27.55 | jameswf-home | ~[TK]D-Fender |
| 18:27.55 | jbot | [TK]D-Fender is the Zen Master of the blatantly obvious, and #asterisk 's resident Google-proxy. |
| 18:28.09 | bkruse | lol |
| 18:28.24 | kannan | hello all, [TK]D-Fender, i am looking for resources on Asterisk(with a E-1 pri card from digium) interoperability with i3 dialer, which has a dialogic card |
| 18:28.25 | *** join/#asterisk mfork (n=mfork@phoenix.cnwr.com) |
| 18:28.32 | jameswf-home | :)) |
| 18:28.46 | jameswf-home | ~wikis |
| 18:28.46 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 18:28.48 | tomierna | thanks, hC. |
| 18:28.53 | jameswf-home | kannan: ^^^^^^^^^^^^^^6 |
| 18:28.56 | kannan | i can get the PRI working fine with the telco, but need to become the service provider for the i3 now, heh |
| 18:28.58 | marksaitis | Qwell, well i found the license to be in COPYING file in asterisks package, but I have not found a word "commerc*" over there |
| 18:29.08 | Qwell | read it |
| 18:29.14 | kannan | jamesswf-home , reading, thanks |
| 18:29.41 | _ShrikE | JenniferAkemi: No. func_odbc is not a replacement for realtime. |
| 18:30.13 | *** join/#asterisk sakic (n=sakic@cpe-071-075-175-140.carolina.res.rr.com) |
| 18:30.31 | sakic | what is good softphone software for the pc? |
| 18:30.32 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 18:30.32 | mfork | How can I send DTMF tones when using asterisk-java (FastAGI)? I need to do a take back and transfer by sending the flash tone (I have a file that can be played) but cannot figure out how to send the DTMF tones. Thanks. |
| 18:31.28 | jameswf-home | sakic: MozIAX |
| 18:31.42 | *** join/#asterisk jmesquita (n=jmesquit@201.7.117.114) |
| 18:33.59 | joe | a reload or a restart doesn't seem to get rid of a switch I commented out, any ideas why? |
| 18:37.24 | *** join/#asterisk cleone (n=cleo@41.251.87.122) |
| 18:38.19 | cleone | any one here have debian?:) |
| 18:38.31 | RobH | i am runing ubuntu for my * install. |
| 18:38.44 | bkruse | cleone: yep |
| 18:39.15 | cleone | bkruse i want to test if my asterisk worck good |
| 18:39.16 | RobH | Ok, so I want to set the callerid local on my *, i have the callerid= entry in iax.conf, anything i need to toss in there to make it work, cuz it isnt... |
| 18:39.20 | *** join/#asterisk bsaxon (n=bsaxon@66.0.66.4) |
| 18:39.30 | *** join/#asterisk Deeewayne (n=dwayne@216.207.245.1) |
| 18:39.30 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 18:39.32 | cleone | bkruse can we speack togader |
| 18:39.53 | file | Deeewayne: whatever! |
| 18:40.13 | Deeewayne | whatever do0d |
| 18:41.02 | bkruse | cleone: what did you do |
| 18:41.04 | bkruse | package it? |
| 18:41.19 | cleone | bkruse yes |
| 18:41.40 | bkruse | cleone: I suggest you talk to tzafrir or tzafrir_home, and get on the voip packagers mailing list |
| 18:41.44 | cleone | bkruse from source is better, but i want just to test in this machine |
| 18:41.46 | bkruse | strongly suggest ;) |
| 18:41.59 | sakic | what ports do I have to open in my firewall to peer 2 serves |
| 18:42.06 | tzafrir | cleone, or with tzafrir |
| 18:42.09 | sakic | also open for softphone? |
| 18:42.15 | *** join/#asterisk Blackthorn (i=blacktho@76.77.160.10) |
| 18:42.15 | cleone | bkruse u don't want me:P |
| 18:42.24 | *** join/#asterisk eric2 (n=nobody@sudbury-cable-69-60-240-185.unitz.ca) |
| 18:43.30 | Blackthorn | I need to have asterisk start upon bootup, it's an ubuntu issue but i've been asking for several hours in that channel. Either no one knows at the moment, or i'm not asking the right question (probibly). Using ubuntu server and upstarted. I placed the script into the /etc/event.d directory reboot and nothing happens. |
| 18:43.33 | bkruse | cleone: heh, you are just doing it for your own machine? |
| 18:43.39 | Blackthorn | is there something that i'm missing? |
| 18:43.46 | bkruse | If it is the same debian version, packaging binaries should not be a problem at all |
| 18:44.07 | cleone | bkruse yes i use asterisk just for fun in my home |
| 18:44.13 | RobH | Blackthorn: did you compile from source? I just ran make config when I compiled and it put it there for me. |
| 18:44.13 | bkruse | The only advantage debian has on tar'ing the binaries is looking for libraries (at the stage you are talking about) |
| 18:44.14 | bkruse | gotcha |
| 18:44.51 | Blackthorn | robh: yes. and asterisk can be started up normally "sudo asterisk" and it all works just fine. |
| 18:45.00 | cleone | bkruse and u?:) |
| 18:45.06 | Blackthorn | going by these instructions http://upstart.ubuntu.com/getting-started.html |
| 18:45.23 | RobH | Blackthorn: I dont have an asterisk entry that I can see in event.d, just in init.d |
| 18:45.30 | Blackthorn | er sorry http://www.voip-info.org/wiki/index.php?page=Asterisk+Starting+and+Stopping |
| 18:45.52 | bkruse | cleone: I guess you could say I have used asterisk :] |
| 18:46.14 | cleone | bkruse :)))) |
| 18:47.13 | RobH | Blackthorn: make config worked for me no issues, sorry =[ |
| 18:47.21 | Blackthorn | trying something i might have just found the issue. moved the file from event.d to init.d (those insturction says event.d though) |
| 18:48.37 | *** join/#asterisk svenna (n=svenna@p548D0B9C.dip0.t-ipconnect.de) |
| 18:48.46 | RobH | its viop-info, feel free to correct any instructions that are wrong, its a wiki =] |
| 18:50.00 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.138) |
| 18:50.18 | *** part/#asterisk kamanashisroy (n=kamanash@202.56.7.138) |
| 18:50.18 | joe | http://pastie.caboo.se/171054 so you can see my incoming context should match the extension being dialed in the paste yet it doesn't match, anyone see what's wrong? |
| 18:51.35 | sakic | where do you find a domain in switchvox to use w/ x-lite? |
| 18:52.09 | [TK]D-Fender | sakic: Try using their support channel |
| 18:52.16 | matt_ | hello, if i have 2 phones on the same network does SIP allow both phones to directly sent RTP traffic to eachother ? or does everything run through the server ? |
| 18:52.31 | [TK]D-Fender | matt_: Depends if you've allowed reinvites |
| 18:52.32 | RobH | joe: what context is the zap channel assigned to? |
| 18:52.50 | [TK]D-Fender | matt_: Or if * has a reason to stay in the stream (in-line DTMF features / recording etc. |
| 18:52.51 | matt_ | [TK]D-Fender| ahh ok so its possiable for stuff to connect directly |
| 18:52.58 | matt_ | ahh ok |
| 18:53.17 | [TK]D-Fender | matt_: Yes |
| 18:54.24 | [TK]D-Fender | joe: You forgot to put the "_" in front of your pattern to actualyl indicate that it is a pattern. |
| 18:54.48 | [TK]D-Fender | joe: It therefor interprets those "X"'s literally as alpha chars |
| 18:54.48 | RobH | or it could be the most common error on earth like that and I didnt even notice |
| 18:54.59 | [TK]D-Fender | :) |
| 18:55.29 | RobH | [TK]D-Fender: I'm trying to help, failing, but darn it I am trying ;] |
| 18:56.27 | [TK]D-Fender | RobH: http://www.despair.com/humiliation.html |
| 18:56.35 | Blackthorn | robh: thanks i went back and did the make config and sure enophe it's now working.. whew. now i get go put the server online. fun fun fun |
| 18:56.38 | [TK]D-Fender | RobH: (j/k) |
| 18:56.44 | RobH | lol |
| 18:56.59 | RobH | Blackthorn: Glad I could help =] |
| 18:57.00 | sakic | no one is in switchvox :( |
| 18:57.20 | RobH | [TK]D-Fender: on my wall I have 'Compromise' and 'Madness' |
| 18:57.44 | RobH | http://www.despair.com/compromise.html |
| 19:00.47 | JenniferAkemi | what is up with hotels charging $1 to call an 800 number |
| 19:02.39 | *** join/#asterisk jasonwoot (n=jasonrot@user-69-73-40-171.knology.net) |
| 19:05.25 | [TK]D-Fender | JenniferAkemi: "load chan_extortion.so" |
| 19:05.29 | JayTee52 | lol |
| 19:06.43 | JayTee52 | "Ferengi rule of acquisition #373. When running a hotel or bed and breakfast always overcharge for phone service. They'll pay rather than go somewhere else to call." |
| 19:06.52 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 19:08.01 | r0d3nt | JayTee52: amen. |
| 19:09.25 | RobH | thats right before the rule to insulate the rooms so cell phones dont work right? |
| 19:11.08 | JayTee52 | yep |
| 19:11.09 | joe | [TK]D-Fender: yeah, I 'm a dork ... thanks! |
| 19:11.28 | jameswf-home | I noticed all the hotels with "free highspeed interweb" have worse than dial up connections assuming you get one |
| 19:11.50 | sakic | anyone know what the "authorization user name" in x-phone is for? |
| 19:11.59 | JayTee52 | all the hotels advertise free internet but sooner or later they'll start charging for "excessive packets" |
| 19:12.39 | kannan | sakic : is it used with authuser= setting in sip.conf? i am not very sure tho |
| 19:12.53 | hardwire | JayTee52: free internet should be like the mint on your pillow |
| 19:12.57 | hardwire | delicious, simple, free. |
| 19:13.05 | sakic | no mints anymore |
| 19:13.10 | hardwire | those bastards |
| 19:13.10 | sakic | who give them, I will stay there |
| 19:13.21 | sakic | I stayed at a Hyatt Place the other week |
| 19:13.25 | sakic | excellent hotel |
| 19:13.36 | AlephX | free internet with sale of your browsing history and privacy soon. |
| 19:13.52 | jameswf-home | Lucky to get clean sheets... pack your own |
| 19:14.09 | jameswf-home | hotels are icky |
| 19:14.14 | RobH | amen |
| 19:14.24 | jameswf-home | get a sleeping bag sleep on the floor |
| 19:14.28 | RobH | i have to travel next month and stay in one, not happy. |
| 19:14.57 | sakic | no big deal they can see all my porn |
| 19:16.35 | Olobola | I have a sip client specific problem: regular calls work fine. When I call voicemail, the audio "gaps" and comes in out of order for some reason |
| 19:17.10 | Olobola | It works fine with other clients. |
| 19:20.23 | *** join/#asterisk teknoprep (n=Chris@unaffiliated/teknoprep) |
| 19:20.36 | teknoprep | is there a way to setup a softkey for park on the polycom phones ? |
| 19:20.54 | JayTee52 | when it comes to hotels I like Residence Inn. Kinda pricey but worth it. Doubletree is just as expensive and all you get there are cookies that are overrated. |
| 19:21.26 | kannan | hello, i'm trying to receive and make calls thru a SIP provider thru my asterisk box, and route all these calls thru a PRI to a windows server with a dialogic card thru a E-1 crossover, the E-1 digium card shows red alarm |
| 19:22.25 | ManxPower | kannan: cable issues are the usual cause of a red alarm |
| 19:22.48 | ManxPower | kannan: I assume the E-1 crossover cable is NOT an ethernet crossover (they are different)? |
| 19:23.10 | ManxPower | kannan: does the digium card show a red alarm? |
| 19:23.20 | kannan | MaxPower , yes |
| 19:23.39 | kannan | oh , I am using a ethernet crossover only |
| 19:24.01 | ManxPower | kannan: then it will not work |
| 19:24.09 | kannan | i use it to conect the card to a telco, and it works fine? |
| 19:24.12 | kannan | I see |
| 19:24.25 | ManxPower | you need a T-1/E-1 crossover |
| 19:24.48 | *** join/#asterisk [T]ank (n=ckwall@206.71.78.158) |
| 19:24.51 | ManxPower | kannan: an ethernet cable will work for T-1/E-1 if you don't have to crossover |
| 19:25.38 | kannan | ManxPower - thanks so much , be back in a bit , after seeing the difference |
| 19:25.43 | zeniffty2002 | teknoprep: not that I can tell with version 1.6 maybe in a different version. check out the keys/ function in the Admin Guide |
| 19:26.47 | *** join/#asterisk UnixDog (n=UnixDog@ppp-71-129-22-233.dsl.irvnca.pacbell.net) |
| 19:27.45 | hardwire | 0xbowwow |
| 19:28.40 | sucram2g | Hi, when i do a ztcfg -vvv, i get 31 channels to configure, what could be the reason for this ? |
| 19:28.48 | *** join/#asterisk khronos (n=khronos@aquaman.perryinstitute.org) |
| 19:28.59 | khronos | Hi guys. |
| 19:29.23 | khronos | Anybody know how to change the rtp frame size in asterisk 1.4 to 30 ms instead of the default 20 ms? |
| 19:30.11 | kannan | ManxPower - is it better to buy a readymade one , because will the wrong cabling affect the card? |
| 19:30.47 | kannan | sucram2g - it is E1 |
| 19:31.48 | sucram2g | kannan, ya its a E1, usinga a Sangoma card A104d |
| 19:32.12 | eric_hill | kannan - I just brought up an E1 in Mexico last week and it "required a crossover" cable. I asked the teleco to change it and they switched it to a straight-through connection. |
| 19:32.26 | eric_hill | kannan - They did it in about 3 minutes. |
| 19:32.58 | kannan | but i am not going to connect the PRI to a telco, that is already working fine, I want to connect it to my own dialer with a dialogic card |
| 19:33.28 | eric_hill | kannan - Sorry, misread that. My bad. |
| 19:33.58 | sucram2g | I am expecting to see something like 31 channels configured. |
| 19:35.22 | kannan | i still can't find the pinout for difference between an ethernet crossover cable and E-1 crossover |
| 19:35.48 | *** join/#asterisk djs26 (n=djs@unaffiliated/djs26) |
| 19:36.06 | eric_hill | kannan - pins 1 and 2 get swapped for pins 4 and 5: http://www.ldfacts.com/faq_files/How-to-Make-a-T1-Crossover-cable.htm |
| 19:36.40 | eric_hill | kannan - Effectively you swap the orange and blue pairs. |
| 19:38.07 | kannan | eric_hill, thanks |
| 19:38.57 | *** join/#asterisk xenonex (n=xenonex@89.108.95.179) |
| 19:39.24 | *** join/#asterisk real0ne (n=Ismail@41.251.74.200) |
| 19:40.28 | kannan | eric_hill, got a link for E-1 |
| 19:40.35 | kannan | heh |
| 19:41.07 | eric_hill | PRI or trunk? |
| 19:41.08 | *** join/#asterisk TrentCreek (n=TrentCre@70.116.105.45) |
| 19:41.55 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 19:41.59 | TrentCreek | how can * send CID date/time? |
| 19:43.27 | _ShrikE | TrentCreek: send it where? |
| 19:44.03 | TrentCreek | to receiving phones |
| 19:45.03 | *** join/#asterisk xenonex (n=xenonex@89.108.95.179) |
| 19:45.16 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.200.110) |
| 19:46.17 | TrentCreek | or lines I am calling to |
| 19:46.29 | TrentCreek | LL or Cell |
| 19:48.25 | *** join/#asterisk xenonex (n=xenonex@89.108.95.179) |
| 19:48.34 | [T]ank | anyone know what causes a call to record 'OutgoingSpoolFailed' as its channel in the cdr? what causes this to happen? |
| 19:48.37 | jblack | I'm seriously stuck. I'm trying to do iax with ipkall, and I'm getting stuck with "Mar 26 19:47:50] NOTICE[3424]: chan_iax2.c:7001 socket_process: Rejected connect attempt from 66.54.140.46, who was trying to reach 's@' |
| 19:49.25 | jblack | I set the default context to inbound, and the context the ipkall account to default, in iax2.conf |
| 19:50.22 | *** join/#asterisk ropdelss (n=ropdelss@port-92-193-76-16.dynamic.qsc.de) |
| 19:51.21 | *** join/#asterisk djs26 (n=djs@unaffiliated/djs26) |
| 19:51.56 | ropdelss | hello. I want to change the callerid on outoing calls (chan_capi). It just works using the local-context. In other cases the default, or no number ist shown |
| 19:52.15 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 19:52.47 | ropdelss | exten => _0.,1,Dial,CAPI/ISDN1/54534234:${EXTEN}|55|Ttr |
| 19:53.06 | ManxPower | jblack: do you have an exten => s in the context? |
| 19:53.10 | ropdelss | any idea? |
| 19:53.27 | ManxPower | ropdelss: try setting the CALLERID(num) |
| 19:53.48 | ropdelss | ManxPower, I tried yet. Doesn`t work. |
| 19:53.58 | ManxPower | ropdelss: then you can't do it. |
| 19:54.07 | ropdelss | Why? |
| 19:54.12 | ManxPower | It is up to the provider as to if they will let you set the callerid or not. |
| 19:54.34 | *** join/#asterisk quigon (n=matias@200.61.187.185) |
| 19:54.39 | ropdelss | ManxPower, If I use the local-context and use the dial-command in the asterisk commandline |
| 19:54.41 | ropdelss | it works |
| 19:54.50 | ropdelss | but in other context it doesn`t work |
| 19:54.55 | ManxPower | If setting CALLERID(num) and then dialing out does not show the correct Caller*ID, then your provider is not permitting you to set that Caller*ID or you set it wrong. |
| 19:55.12 | ManxPower | ropdelss: I STILL don't see your Set(CALLERID(num)=xxxxxx) line. What is it. |
| 19:56.07 | ManxPower | ropdelss: put a Noop(CALLERID(num) is ${CALLERID(num)}) as priority 1, then the Dial as priority n |
| 19:56.16 | ManxPower | look at the console. Is the callerid set as you expect? |
| 19:56.54 | TrentCreek | WHat about sending the correct caller ID date and time? |
| 19:57.07 | ropdelss | ManxPower, sec. |
| 19:57.52 | ManxPower | TrentCreek: the telco does that, not you. |
| 19:58.17 | bkruse | ManxPower: good telco's do that :P |
| 19:58.32 | ropdelss | exten => _0.,1,Set(CALLERID(num)=54534234) |
| 19:58.32 | ropdelss | exten => _0.,n,Dial,CAPI/contr1/54534234:${EXTEN}|55|Ttr |
| 19:58.33 | kannan | eric_hill, PRI , sorry for the delay afk |
| 19:58.44 | TrentCreek | well, guess they are not doing a good job...my phones have the incorrect date/time |
| 19:59.24 | hardwire | make them have the correct time. |
| 19:59.32 | UnixDog | what phones |
| 19:59.46 | UnixDog | did you go in and configure a ntp server in them |
| 19:59.54 | UnixDog | did you set the time offset |
| 20:00.23 | hardwire | can channel flags be a variable? Dial,Tech/Number|60|${DIALFLAGS} ? |
| 20:00.35 | TrentCreek | my cell phone and I call my * box to have my phones have the correct time on CID |
| 20:00.41 | hardwire | and Dial,Tech/Number|60|T${DIALFLAGS} ? |
| 20:01.04 | TrentCreek | I have received other calls also, but still wrong date/time..not sure if NTP is running |
| 20:01.21 | UnixDog | what phone |
| 20:01.29 | UnixDog | polycom/snom/aastra |
| 20:01.34 | ManxPower | ropdelss: Where is the noop I told you |
| 20:01.39 | *** join/#asterisk ice_croft (n=nolan@85.172.5.106) |
| 20:01.45 | ice_croft | hi all |
| 20:01.57 | ManxPower | TrentCreek: Maybe your calleridbox is set for the wrong timezone? |
| 20:01.58 | TrentCreek | just a POT connected to a Linksys box |
| 20:02.06 | TrentCreek | it's on the phone |
| 20:02.15 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 20:02.21 | TrentCreek | and Linux is set correct date/time |
| 20:02.24 | UnixDog | did ythen you have to gointo the phone and set the time |
| 20:02.31 | ice_croft | anybody usin cdr_radius? |
| 20:02.53 | ice_croft | cant figure how should i call it from dialplan |
| 20:03.26 | ice_croft | or through any other way |
| 20:03.34 | ManxPower | TrentCreek: seems like you left out the critical information |
| 20:03.38 | TrentCreek | UnixDog: there is no way to set date/time on phone...it is very basic. Only buttons to scroll through calls and delete them |
| 20:04.05 | ManxPower | TrentCreek: Too bad you got all that wrong info because you did not give us enough information |
| 20:04.09 | TrentCreek | What info? |
| 20:04.42 | ManxPower | TrentCreek: the fact that your phone is NOT connected to the telco, but to a SIP ATA |
| 20:05.16 | ManxPower | The linksys box is what you have to fi |
| 20:05.18 | ManxPower | fix, that is |
| 20:05.27 | TrentCreek | ahhh... |
| 20:05.39 | UnixDog | there has to be a way on the phone to set the time wich means you have not taken the time to learn your hardware |
| 20:05.39 | ManxPower | TrentCreek: how far off is the date/time in Callerid? |
| 20:05.44 | hardwire | where would the linksys ATA present the time? |
| 20:05.45 | [TK]D-Fender | TrentCreek: I'm an unaware of any time being sent over SIP, and * is not responsible for the ATA passing anything to the phone you attach to it either. date/time is an OPTIONAL component of CID |
| 20:05.53 | hardwire | oh.. ah. |
| 20:05.54 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 20:05.55 | hardwire | gotcha. |
| 20:06.02 | ManxPower | UnixDog: I doubt it. |
| 20:06.09 | hardwire | I thought caller id didn't have a date/time |
| 20:06.11 | TrentCreek | UnixdogL there is no way to do it..only 3 buttons on phone. I have had it for 3 years now. |
| 20:06.17 | hardwire | figured that was all in the phone. |
| 20:06.17 | ice_croft | [TK]D-Fender> maybe u know, how to call to cdr_radius? |
| 20:06.48 | UnixDog | well if he would state what phone it is model name and number I can tell you for sure |
| 20:06.51 | ManxPower | ice_croft: maybe you need to find someone that uses cdr_radius, not many people do |
| 20:06.56 | Katty | [TK]D-Fender has been very grumpy of late. |
| 20:06.57 | TrentCreek | Thanks Fender (Not Geetar) |
| 20:07.02 | *** part/#asterisk hardwire (n=bip@xvm-189-205.ghst.net) |
| 20:07.12 | TrentCreek | and everyone else..I wil look at the ATA |
| 20:07.12 | *** join/#asterisk [hC] (n=hardcore@vpn.voxter.com) |
| 20:07.13 | [T]ank | anyone in the market for t1 cards? |
| 20:07.17 | ManxPower | Katty: only recently? |
| 20:07.26 | ManxPower | [T]ank: always |
| 20:07.29 | [T]ank | pm me |
| 20:07.34 | ice_croft | ManxPower> for sure. but nowhere to look for them except this place |
| 20:07.38 | Katty | ManxPower: well, you're right. but i guess his grumpiness is never directed towards me. |
| 20:07.52 | *** join/#asterisk supjigator (n=shanebur@152.53.16.10) |
| 20:08.02 | ManxPower | ice_croft: or the mailing list or google or the mailinglist archives or the author of cdr_radius |
| 20:08.40 | ManxPower | To anyone I was helping: You ran out of time. Perhaps I can continue to help you when I return |
| 20:08.41 | ropdelss | ManxPower, How do I set the callerid using noop? |
| 20:09.04 | ManxPower | ropdelss: you don't. The noop lets you see what the callerid IS CURRENTLY SET TO |
| 20:09.24 | Qwell | NoOp(${SET(CALLERID(num)=1234)}) |
| 20:09.25 | Qwell | lol |
| 20:09.53 | Qwell | not quite right, but you get the idea |
| 20:09.57 | eric_hill | [TK]D-Fender: FYI, found out that Asterisk's SIP/DTMF features are incompatible with Cisco's Unity Express.... Sigh. |
| 20:10.14 | Katty | Qwell: hai! |
| 20:10.25 | *** join/#asterisk ParanoyaM (n=Maksim@223-82-124-91.pool.ukrtel.net) |
| 20:10.28 | ParanoyaM | [TK]D-Fender, |
| 20:11.26 | mwalling | has anyone tried using grandcentral -> sipphone.com account -> asterisk? i tried it, was able to connect, but it seemed like GC was stripping DTMF off |
| 20:12.45 | *** join/#asterisk [Akemi] (n=akemi@76.10.167.238) |
| 20:15.33 | *** join/#asterisk Asterlinktechy (n=chito@125.60.231.8) |
| 20:16.31 | [TK]D-Fender | ice_croft: not at all. |
| 20:16.47 | plik | ~sipnat |
| 20:16.48 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 20:17.22 | [TK]D-Fender | eric_hill: Yup... Cisco doesn't play well with others I guess |
| 20:18.50 | RobH | Ok, I am having issues passing callerid. I set it in iax.conf, reload, and it doesnt work. It does work if I set it in the dialplan before the call, which I rather not do there, seems messier. Here is a paste of it: http://rafb.net/p/st9nGW71.html |
| 20:18.54 | RobH | What am I missing? |
| 20:19.06 | ropdelss | ManxPower, output with capi debug: http://rafb.net/p/SxeBL948.html |
| 20:22.25 | jblack | could there have been dtmf problems back with 1.4.10? |
| 20:24.02 | kannan | Ok , thanx all, esp ManxPower, eric_hill, [TK]D-Fender, be back tomorrow after buying a proper E-1 crossover cable. Yep i see that my TELCO HAS DONE THE CROSSING IN HIS ADAPTOR INSTALLED AT OUR PREMISES, SO THE ETHERNET WORKED FINE |
| 20:24.07 | kannan | sorry caps |
| 20:24.32 | jblack | I sympathize. You want the telco to hear you from here. :) |
| 20:24.49 | *** part/#asterisk [T]ank (n=ckwall@206.71.78.158) |
| 20:25.21 | kannan | jblack, sorry did not understand ? |
| 20:25.30 | jblack | never mind |
| 20:25.32 | kannan | i didnt get what you meant |
| 20:25.36 | ropdelss | Maybe there ist a porblem in the capi.conf? |
| 20:25.56 | kannan | :) , i know its a joke tho, lol |
| 20:25.56 | [hC] | exit |
| 20:25.59 | [hC] | oops! |
| 20:26.18 | kannan | aah i get tit now, the caps means shouting ? |
| 20:26.23 | jblack | yes |
| 20:26.25 | kannan | get it |
| 20:26.27 | kannan | lol |
| 20:26.31 | ParanoyaM | [TK]D-Fender, |
| 20:27.08 | kannan | bye for now , c u all |
| 20:27.12 | *** part/#asterisk kannan (n=kannan@121.241.156.83) |
| 20:27.28 | [TK]D-Fender | I'm off, BBL |
| 20:30.30 | *** join/#asterisk Cle0 (n=cleo@41.250.136.123) |
| 20:33.10 | jasonwoot | <PROTECTED> |
| 20:33.35 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 20:44.23 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-24-106-241-121.central.biz.rr.com) |
| 20:45.10 | ZPertee | Will this http://pastebin.ca/958607 work for my incoming dial plan? |
| 20:45.13 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 20:45.40 | *** join/#asterisk teknoprep (n=Chris@unaffiliated/teknoprep) |
| 20:45.55 | lirakis | jasonwoot: http://www.pastebin.ca/958611 .. fyi my agents use 4 digit extensions.. so the 3 digits are the alias i use to identify them.. keep your dollar |
| 20:46.01 | teknoprep | hey whats that command that for NATing sip . not externip = but the other one that uses a hostname |
| 20:46.21 | teknoprep | well configuration option... for nat |
| 20:46.32 | lirakis | teknoprep: externhost |
| 20:46.35 | teknoprep | ty |
| 20:46.38 | teknoprep | i couldn't remember |
| 20:46.57 | lirakis | teknoprep: i wouldnt use that in production though |
| 20:47.06 | lirakis | your dns goes down.. your pbx is screwed |
| 20:47.06 | teknoprep | is for my house |
| 20:47.13 | teknoprep | i have a changing IP address |
| 20:47.15 | lirakis | i dont use it for my house either ;) |
| 20:47.22 | lirakis | teknoprep: well you gotta do what you gotta do |
| 20:47.25 | teknoprep | well i don't have a static IP |
| 20:47.39 | teknoprep | all the business' i install do have static IP's tho... but not here |
| 20:47.55 | lirakis | okay all .. later |
| 20:47.56 | *** part/#asterisk lirakis (i=lirakis@66.252.24.133) |
| 20:50.43 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 20:53.26 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 20:53.57 | grandpapadot | Hi all. In asterisk 1.2, using the GoToIfTime() application, how would I specify days not in series? I know I can do mon-fri, but what if I wanted to do mon,tue,fri? |
| 20:54.57 | Strom_C | grandpapadot: separate gotoiftime statements |
| 20:55.10 | grandpapadot | hrm... ok, thats what I was afraid of, thanks! |
| 20:55.25 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 20:55.58 | *** join/#asterisk Blackthorn (i=blacktho@76.77.160.10) |
| 20:56.29 | Blackthorn | I'm having a problem with zapata.conf not loading. When i start using asterisk -vvv it says that " Unknown signalling |
| 20:56.29 | Blackthorn | method 'pri_cpe' |
| 20:56.49 | ManxPower | Blackthorn: looks like you did not install libpri before you installed Asterisk |
| 20:57.10 | Blackthorn | umm gota do it in order i guess? |
| 20:57.15 | Blackthorn | kk, thanks |
| 20:57.27 | ManxPower | Blackthorn: If asterisk does not see libpri when it's built it won't build PRI support. |
| 20:57.33 | ManxPower | same with zaptel |
| 20:57.45 | ManxPower | There IS a reason for the order of install in the docs |
| 20:57.48 | Blackthorn | zaptel is working. can see the pri using zztool and says configured |
| 20:58.10 | ManxPower | Blackthorn: no, you see the T-1. PRI is a higher level protocol that zttool knows nothing about |
| 20:58.14 | Blackthorn | and i've installed libpri first. but don't know why it's not seeing it |
| 20:59.01 | [TK]D-Fender | libpri > Zaptel > Asterisk |
| 20:59.05 | ManxPower | in 1.4 you have to remove some oddball file (none of the make cleans seem to remove it) then rerun ./configure for asterisk after libpri and zaptel are installed |
| 20:59.08 | Blackthorn | in the lib source directroy.. all i need to do is 'make" and a "make install" for the libpri correct? |
| 20:59.11 | [TK]D-Fender | BRB |
| 20:59.15 | *** part/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 20:59.41 | ManxPower | Blackthorn: just remove the asterisk source, untar it again, install again |
| 21:00.13 | iCEBrkr | ManxPower: Now you're talking my language! |
| 21:00.15 | sakic | where can I find my authorization username in switchbox? |
| 21:00.43 | iCEBrkr | checks out the channel name. |
| 21:00.53 | iCEBrkr | Doesn't look like #switchbox |
| 21:03.37 | ManxPower | sakic: you are on the wrong channel |
| 21:04.31 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:04.50 | ManxPower | iCEBrkr: I would not have to recommend it if "make clean" "make distclean" would remove the file |
| 21:05.09 | [TK]D-Fender | When in doubt trash your install folders and re-extract. |
| 21:05.13 | [TK]D-Fender | Its what I do. |
| 21:05.19 | iCEBrkr | Same here |
| 21:08.29 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-24-106-241-121.central.biz.rr.com) |
| 21:09.56 | *** part/#asterisk JayTee52 (n=jforde@207-67-84-188.static.twtelecom.net) |
| 21:11.48 | ZPertee | can someone please help me get call parking configured? I want to be able to do it after I answer a call. here is what i have so far http://pastebin.ca/958657 |
| 21:16.07 | *** join/#asterisk s0lid (n=s0lid@210.213.198.56) |
| 21:18.37 | plik | ZPertee: have you enables it in features.conf ? |
| 21:19.02 | plik | ZPertee: I just transfer the call to extn 700 (as defined there...) |
| 21:21.50 | plik | and to pick it up, dial 701 (or whatever it tells you is the parked position, when parking it there) |
| 21:23.37 | [TK]D-Fender | ZPertee, just do an attended transfer the call to 700 and it will read back a lot # to you. Then dial that # to pick it up. |
| 21:23.43 | seanbright | what does LastMsgsSent in 'sip show peer Foo' mean? |
| 21:26.03 | *** join/#asterisk antdedyet (i=ady@gonzo.fearandloathing.org) |
| 21:26.17 | antdedyet | heya, anyone outside the USA available to test a 1-800 number? |
| 21:26.30 | Strom_C | what do you mean "outside the USA" |
| 21:26.57 | Strom_C | U.S. tollfree numbers are only accessible from inside the NANP |
| 21:27.05 | Strom_C | and usually that's restricted to only the U.S. |
| 21:28.18 | antdedyet | Strom_C: ah, ok. I'm not much on telco, someone asked if I had someone outside the USA that could test a 800 number for their client ... I will pass your info along to them. |
| 21:28.38 | plik | antdedyet: I'm in uk and can dial us 800 nos |
| 21:29.05 | luisjose | is there any app for call center asterisk-compatibles |
| 21:29.32 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 21:32.05 | [TK]D-Fender | luisjose, huh? |
| 21:32.20 | *** join/#asterisk teknoprep (n=Chris@unaffiliated/teknoprep) |
| 21:32.54 | luisjose | [TK]D-Fender, call center software |
| 21:32.57 | *** join/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net) |
| 21:33.08 | *** part/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net) |
| 21:33.22 | [TK]D-Fender | luisjose, What is "call center software"? |
| 21:33.30 | ParanoyaM | [TK]D-Fender, |
| 21:33.54 | mwalling | ~skype |
| 21:33.54 | jbot | [~skype] Skype is a free VoIP software and service using a closed client and propritary protocol. Only commercial channel drivers exist, all of which are poor implementations of convoluted hacks. In general, forget about using Skype with Asterisk... |
| 21:35.38 | mwalling | ~chan_skype |
| 21:39.18 | delparnel | If I call my DISA and get a dialtone from my BlackBerry.. is there some way I can dial a number using my address book? |
| 21:39.40 | [TK]D-Fender | delparnel, only if you phone lets you pass your selection as DTMF in-band |
| 21:39.47 | [TK]D-Fender | delparnel, So thats up to your phone |
| 21:40.30 | jblack | Oh, for reference - 360 area code numbers for ipkall do not not handle dtmf very well at all, but 206 area code numbers do. |
| 21:48.36 | *** join/#asterisk random-z (n=adam@shelbyville.fusionbay.com) |
| 21:49.32 | *** join/#asterisk RoyK (n=roy@ip-96-51-149-91.dialup.ice.no) |
| 21:50.35 | *** part/#asterisk RoyK (n=roy@ip-96-51-149-91.dialup.ice.no) |
| 21:55.59 | *** join/#asterisk Keltus (i=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 21:56.50 | ManxPower | antdedyet: generally you can't dial USA toll free numbers from outside the USA. |
| 21:56.56 | *** join/#asterisk TedC (n=cabeen@milli.chem.ucsb.edu) |
| 21:57.08 | ManxPower | If you have a USA voip account, you could, but that's not really "outside the usa" |
| 21:58.25 | plik | 2 |
| 21:58.29 | plik | oops |
| 21:58.49 | plik | ManxPower: I just dialled it via voip and from a BT landline and it went through fin, |
| 21:59.14 | plik | with an announcement that it would be chargeable on the voip call but not BT strangely |
| 21:59.25 | ManxPower | plik: Odd. |
| 21:59.31 | ManxPower | Must be something new |
| 21:59.40 | plik | which is odd, coz BT will have charged us, but our voip calls to US are free |
| 22:00.00 | plik | yeah, I think so... 1-800 used to just fail |
| 22:00.24 | plik | but I've heard the "you will be charged" message before |
| 22:04.57 | *** part/#asterisk TedC (n=cabeen@milli.chem.ucsb.edu) |
| 22:05.01 | tzanger | hmm interesting |
| 22:05.07 | tzanger | the 'r' dial option isn't ringing anymore |
| 22:05.56 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 22:06.15 | *** part/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 22:07.17 | tzanger | and ringing() isn't doing what I'd expect either |
| 22:08.04 | grandpapadot | Is it possible to do a label+1? i.e. GoTo(s,somelabel+1)? |
| 22:08.22 | [TK]D-Fender | grandpapadot, No |
| 22:08.44 | grandpapadot | tnx |
| 22:09.21 | [TK]D-Fender | tzanger, You still on contract down this way? |
| 22:09.35 | tzanger | [TK]D-Fender: next week I'm back there |
| 22:09.58 | [TK]D-Fender | tzanger, Cool, we've seriously got to settle on that meet-up day then |
| 22:10.37 | tzanger | yes for sure |
| 22:10.45 | ManxPower | tzanger: classic symptom of lack of /etc/asterisk/indications.conf |
| 22:10.51 | tzanger | ManxPower: that's what I thought |
| 22:10.54 | tzanger | except that that's not it |
| 22:11.00 | tzanger | if I dial(,,grtk) it works |
| 22:11.16 | ManxPower | and you know that "r" never does any good |
| 22:11.17 | tzanger | if I dial(,timeout,) then do some logic and dial(,,grtk) ring won't work |
| 22:11.28 | tzanger | and if I put an answer() before the second dial, it works perfectly |
| 22:11.33 | tzanger | ManxPower: I'm calling a cell :-) |
| 22:11.39 | ManxPower | tzanger: does that some logic end up answering the line (playback, etc)? |
| 22:11.41 | tzanger | I know 'r' is evil if incorrectly used |
| 22:11.48 | tzanger | ManxPower: no, I *need* to answer the line |
| 22:12.00 | tzanger | adding answer() before the dial(,,grtk) works |
| 22:12.03 | ManxPower | tzanger: I was only asking for informational reasons |
| 22:12.07 | tzanger | ManxPower: I know |
| 22:12.40 | tzanger | I always had my stupid-simple db cid-lookup to follow me working (call comes in, rings the house and then if it's from someone I want to reach me anywhere, dial the cell too) |
| 22:13.11 | tzanger | I just added support for it to check for multiple DIDs, so each DID could have a different table of who to call |
| 22:13.44 | tzanger | and decided to fix a longstanding problem my daughter complained about, which was when she calls me and I'm not home it didn't ring as it was trying the cell |
| 22:14.42 | tzanger | so yeah... dial(${HOME},16,gtk) if(dialstatus != answer) { answer(), dial($cell,24,grtk), if($dialstatus != answer, voicemail) } works now |
| 22:14.57 | tzanger | without the explicit answer() though no indications get played after the first dial() |
| 22:15.07 | tzanger | I'd call that a bug |
| 22:15.24 | tzanger | gotta see if it still does it in trunk though, this is a fairly old pull of trunk |
| 22:15.43 | tzanger | hmm, it's not trunk. 1.4.13 |
| 22:15.47 | tzanger | still needs to be updated |
| 22:15.47 | *** join/#asterisk galera1 (n=galeras@190.156.212.43) |
| 22:16.00 | tzanger | there's a few critical fixes out since then I think |
| 22:16.34 | *** join/#asterisk klin3d (i=ircN@pc-32-231-86-200.cm.vtr.net) |
| 22:16.57 | galera1 | Dear Sirs and Ladies, please give me some clue, about why calls from internal to external sip are fine, however external to internal sip clients has no audio. Thanks. |
| 22:17.12 | ManxPower | ~sipnat |
| 22:17.13 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 22:18.12 | galera1 | jbot: thanks i will tak a look again to that links, perhaps i missed something |
| 22:18.12 | jbot | galera1: gern geschehen |
| 22:19.58 | *** join/#asterisk bsaxon (n=bsaxon@66.0.66.4) |
| 22:23.19 | *** join/#asterisk bminish (n=bminish@78.152.241.48) |
| 22:23.50 | tzanger | hmm |
| 22:23.52 | ManxPower | galera1: no audio is a CLASSIC symptom of incorrect configuration when NAT is involved. |
| 22:24.12 | tzanger | does anyone know how to use sox to convert wav to ulaw? I got it from ul to wav, but now it won't convert back (unknown format 101) |
| 22:24.16 | ManxPower | tzanger: as far as I can tell /etc/asterisk/indications.conf is only used for playback of inband tones. |
| 22:24.25 | tzanger | ManxPower: yeah I know... it's weird |
| 22:25.32 | *** join/#asterisk yassine (n=yassine@unaffiliated/yassine) |
| 22:27.12 | ManxPower | tzanger: 1.6 allows you to convert via the CLI |
| 22:28.12 | tzanger | hmm |
| 22:29.39 | ManxPower | tzanger: If all else fails.... |
| 22:30.19 | tzanger | upgrade? |
| 22:30.49 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 22:31.00 | ManxPower | tzanger: I was thinking more along the lines of installing 1.6 on a spare machine. |
| 22:31.12 | *** join/#asterisk droops (n=droops@74.193.237.138) |
| 22:32.48 | tzanger | ManxPower: nah |
| 22:32.50 | tzanger | got it working |
| 22:32.56 | tzanger | I was making ulaw-format wav files |
| 22:33.01 | tzanger | sox will convert to that, but not from |
| 22:35.38 | *** join/#asterisk sigmounte (n=sigmount@bai59-1-88-172-80-96.fbx.proxad.net) |
| 22:41.17 | *** join/#asterisk henrique (n=henrique@unaffiliated/henrique) |
| 22:41.21 | *** part/#asterisk zeniffty2002 (n=zeniffty@mail.revenueworx.com) |
| 22:45.50 | yassine | hi i have a x100p based card and i followed this guide ( http://kb.digium.com/entry/40/ ) to get it working, i can monitor the incoming channel using ztmonitor which shows me that the card is reconized and when i call my analog line the signal strenth changes, but when try to run an echo test it does not work, any idea? |
| 22:50.34 | *** join/#asterisk javb (n=javb@190.80.235.159) |
| 22:51.32 | random-z | I just upgraded... is there no longer a "zap" command in the CLI interface? |
| 22:51.38 | javb | does anyone here had changed the ringers to polycom 330, from those that the phone comes with? ... (i know this is no polycom channel, but i think i may find someone here who have done this) |
| 22:51.43 | random-z | or did I wipe out the module somehow? :) |
| 22:53.10 | *** join/#asterisk marc7 (n=marc@64.46.14.64.novuscom.net) |
| 22:57.27 | tzafrir | random-z, no, it has failed to load. |
| 22:57.34 | random-z | ok, thanks |
| 22:57.38 | tzafrir | Look in the logs for the exact error message |
| 22:58.07 | random-z | there should be a corresponding file in the /modules/ directory? |
| 22:58.09 | random-z | chan_zap.so ? |
| 23:00.07 | *** join/#asterisk tessier_ (n=treed@kernel-panic/sex-machines) |
| 23:00.10 | tessier_ | Hello all! |
| 23:00.26 | tessier_ | Anyone know how to easily tell what software a given sip proxy is running? |
| 23:00.37 | tzafrir | random-z, yes |
| 23:00.51 | random-z | ok, think thats the problem then :-D it didn't make the module |
| 23:00.52 | random-z | thanks a lot! |
| 23:01.48 | galera1 | Sirs, Nar issue: please take a look of http://www.pastebin.ca/958820, with my current iptables and sip.conf settings and give me some clue about why a remote peer becomes UNREACHABLE after a few seconds. (internal to external calls are fine)Thanks |
| 23:01.55 | galera1 | *Nat |
| 23:04.24 | ManxPower | random-z: upgrade.txt made no mention of the issue? |
| 23:04.43 | random-z | I didn't see anything... looks as though I missed a dependancy |
| 23:05.19 | random-z | I think did a ./configure on asterisk before I had done a make install on zaptel |
| 23:05.24 | random-z | ... which would be my error :) |
| 23:05.36 | Strom_C | also, learning to spell dependency |
| 23:05.40 | Strom_C | but I digress |
| 23:05.45 | random-z | ouch |
| 23:06.42 | *** part/#asterisk random-z (n=adam@shelbyville.fusionbay.com) |
| 23:08.40 | galera1 | bounty $50 for anyone who help me get working ountbound calls to a remote sip peer. |
| 23:11.01 | *** join/#asterisk ParanoyaM (n=Maksim@223-82-124-91.pool.ukrtel.net) |
| 23:11.01 | ParanoyaM | [TK]D-Fender, |
| 23:15.50 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 23:18.42 | ParanoyaM | [TK]D-Fender, you know, i solved that issue with load balancing :D |
| 23:19.20 | ParanoyaM | [TK]D-Fender, it was quiet easy |
| 23:19.47 | jameswf-home | today is Document Freedom Day ... who knew... http://googleblog.blogspot.com/2008/03/today-is-document-freedom-day.html |
| 23:22.54 | JT | ParanoyaM: quiet easy? |
| 23:23.14 | ParanoyaM | yes |
| 23:23.28 | ParanoyaM | about 30 strings of code :)))))))) |
| 23:23.55 | JT | code that didn't make much noise? |
| 23:24.14 | ParanoyaM | i am not very good in english |
| 23:24.27 | ParanoyaM | so i don't understand what do you mean saying noise |
| 23:24.40 | JT | quiet = low noise |
| 23:25.04 | ParanoyaM | no i mean easy |
| 23:25.08 | ParanoyaM | not very difficult |
| 23:25.31 | JT | quite |
| 23:25.58 | ParanoyaM | ok |
| 23:26.00 | ParanoyaM | good night |
| 23:26.08 | ParanoyaM | let it work i will test this night it |
| 23:26.14 | ParanoyaM | bye bye |
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| 23:47.56 | lesouvage | When I monitor a call and use the A() parameter when dialing for an announcement for the called party (this call will be recorded) the announcement on the recording is on a very low sound level while the called party can her it loud and clear. Is this a known issue? |
| 23:48.43 | lesouvage | And is there anything to do to fix this? |
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