| 00:03.19 | *** join/#asterisk jks (~jks@193.189.93.254) |
| 00:05.47 | *** join/#asterisk tris (tristan@camel.ethereal.net) |
| 00:06.21 | *** join/#asterisk SaiSoma (~SaiSoma@adsl-074-167-136-030.sip.mob.bellsouth.net) |
| 00:11.09 | *** join/#asterisk fnordus (~dnall@70.70.0.215) |
| 00:15.21 | *** join/#asterisk [netman] (~netman@31.Red-83-45-0.dynamicIP.rima-tde.net) |
| 00:18.15 | *** join/#asterisk moy (~moy@69.33.25.200) |
| 00:21.45 | Kobaz | <PROTECTED> |
| 00:21.57 | Kobaz | how do i get channels to exit with a zero status |
| 00:22.04 | Kobaz | Hangup(0) doesn't do it |
| 00:22.55 | nny | is moh reload depricated in 1.6.2? |
| 00:23.01 | Kobaz | the arg to hangup is cause code... which i assume comes from channel.c... so Hangup(5) would be normal clearing... but that still doesn't do it |
| 00:24.31 | nny | nm |
| 00:24.35 | nny | permission ftl |
| 00:26.36 | nny | ugh |
| 00:27.26 | nny | so I have the moh folder with group RWX and asterisk as the group, also asterisk is running *as* asterisk.asterisk. I do a moh reload and it doesn't pick up any changes to the folder. all files within have group R on them, what am I missing? |
| 00:28.44 | nny | however a restart of asterisk picks them up |
| 00:30.16 | nny | reloading the module doesn't work either |
| 00:32.00 | nny | nm lol |
| 00:32.01 | nny | https://bugs.digium.com/view.php?id=16744 |
| 00:32.04 | nny | hooray! |
| 00:44.39 | nny | btw if i ever type hooray, I DEMAND it be read in the voice of Zoidberg |
| 00:44.53 | nny | *m* |
| 00:54.34 | *** part/#asterisk nny (~scott@64.203.239.83) |
| 01:00.20 | Katty | hi |
| 01:09.37 | Katty | [TK]D-Fender: http://farm4.static.flickr.com/3107/3125404776_f1c79e313f_o.jpg |
| 01:13.56 | Katty | [TK]D-Fender: i shall name the photo Scnozzle |
| 01:14.03 | Katty | [TK]D-Fender: err Schnozzle |
| 01:18.36 | *** join/#asterisk ex-parrot (~ex-parrot@2401:f000:3:0:21a:4dff:fe0d:7e59) |
| 01:18.46 | ex-parrot | hi all, I'm having trouble getting chan_zap to build |
| 01:19.01 | ex-parrot | it's showing with XXX through the selection checkbox in make menuselect |
| 01:19.12 | ex-parrot | despite (to the best of my knowledge) me having all the pre-reqs selected |
| 01:20.16 | Katty | what did you become after a parrot? |
| 01:20.43 | *** join/#asterisk dansays (~dansays@CPE-24-209-154-36.wi.res.rr.com) |
| 01:21.11 | ex-parrot | Katty: an ex-parrot |
| 01:21.36 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 01:21.42 | Katty | are you still a parrot then? |
| 01:22.07 | ex-parrot | depends how you look at it.... |
| 01:23.20 | Katty | elaborate |
| 01:24.29 | ex-parrot | Katty: http://en.wikipedia.org/wiki/Dead_Parrot |
| 01:24.38 | Katty | that requires too much effort. |
| 01:25.35 | ex-parrot | I'm astounded, to be honest, that you're not familiar with the Monty Python Dead Parrot sketch. |
| 01:25.55 | spenguin[work] | hey Katty |
| 01:26.48 | Katty | smoooooooooooooooooooooth operator |
| 01:27.01 | spenguin[work] | :> |
| 01:27.06 | spenguin[work] | hows the critters |
| 01:27.22 | Katty | good, good. you see the schnozzle photo? |
| 01:27.29 | spenguin[work] | nope |
| 01:27.56 | Katty | spenguin[work]: http://farm4.static.flickr.com/3107/3125404776_f1c79e313f_o.jpg |
| 01:28.15 | spenguin[work] | lol cute |
| 01:28.24 | spenguin[work] | his nose isnt wet |
| 01:28.39 | Katty | not for the photo |
| 01:28.42 | Katty | that'd just look gross |
| 01:28.59 | *** join/#asterisk dansays (~dansays@CPE-24-209-154-36.wi.res.rr.com) |
| 01:29.01 | spenguin[work] | hehe |
| 01:29.18 | spenguin[work] | you wipe his nose before the photo? |
| 01:29.23 | spenguin[work] | :\ |
| 01:29.38 | Katty | yes |
| 01:29.46 | Katty | and the camera lense a few times |
| 01:29.49 | Katty | do to kisses |
| 01:29.51 | Katty | er due |
| 01:29.51 | spenguin[work] | haha |
| 01:30.06 | spenguin[work] | I find it cute |
| 01:31.09 | *** join/#asterisk ruben23 (~AGENT@122.55.48.243) |
| 01:32.08 | Katty | the google superbowl ad was cute |
| 01:32.17 | Deeewayne | Katty, g'night! |
| 01:32.24 | Katty | Deeewayne: herroes. |
| 01:32.41 | Deeewayne | mmm..... heroes .... |
| 01:32.48 | ex-parrot | does anyone know what the check leading to "The Zaptel installation on this system appears to be broken." in configure is? I'm having trouble understanding the configure script in enough detail to see what might be causing the check to fail |
| 01:33.49 | ex-parrot | aha, it's just checking if the .h file exists at all, that's simple enough |
| 01:34.28 | spenguin[work] | whats an ex-parrot |
| 01:34.33 | ruben23 | hi im try to dial a number 8663002323 and im getting---> http://pastebin.com/m10ab3b1e anyone have idea. |
| 01:35.17 | ex-parrot | spenguin[work]: c.f Monty Python's Dead Parrot sketch |
| 01:35.54 | spenguin[work] | ruben23: what do you get with 'zap show status' |
| 01:37.33 | *** join/#asterisk tris (tristan@camel.ethereal.net) |
| 01:39.10 | ruben23 | spenguin[work]:im doing sip |
| 01:41.06 | spenguin[work] | hrm |
| 01:41.17 | *** join/#asterisk Badrobot- (~badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 01:41.19 | spenguin[work] | anyone here use a linksys SPA 3102? |
| 01:41.40 | Katty | i don't |
| 01:42.01 | spenguin[work] | k, now the rest 200 |
| 01:42.11 | Katty | :P |
| 01:42.29 | spenguin[work] | hrm .. I was wondering if it was switched off would the telephone line still work? |
| 01:43.15 | hluesea | i am using pap2t. SPA 3102 moslty like the pap2t |
| 01:43.37 | hluesea | nope line is cut down automatically |
| 01:43.40 | carrar | Didn't LinkSys go out of biz? |
| 01:43.52 | spenguin[work] | ok, since the phone line would be wired through the SPA 3102 |
| 01:44.06 | spenguin[work] | carrar: they were bought by cisco |
| 01:44.10 | carrar | I know |
| 01:44.50 | Katty | the taco bell commercial was terrible |
| 01:44.58 | Katty | they hsould have advertised healthy options |
| 01:45.16 | carrar | I think they were appealing to a different demographic |
| 01:45.31 | carrar | the none healthy one |
| 01:45.43 | hluesea | spenguin : i rememberred now, sorry, phone line is still up when the switched off |
| 01:45.55 | Katty | the audi commercial went off the deep end |
| 01:46.05 | spenguin[work] | hluesea: ah k thanks! |
| 01:46.19 | Katty | but i do like the screaming chicken denny commercials. |
| 01:46.24 | *** part/#asterisk ex-parrot (~ex-parrot@2401:f000:3:0:21a:4dff:fe0d:7e59) |
| 01:46.45 | Katty | it's just funny |
| 01:47.04 | spenguin[work] | hasnt watched TV in a while : |
| 01:47.14 | Katty | spenguin[work]: hulu.com/adzone |
| 01:47.19 | spenguin[work] | oh |
| 01:47.25 | Katty | oooh honda had a squirrel add :> |
| 01:47.28 | Katty | go honda! |
| 01:49.20 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 01:49.21 | spenguin[work] | godaddy has odd adds |
| 01:49.33 | Katty | yeah |
| 01:49.37 | Katty | you gonna watch them all? |
| 01:50.19 | spenguin[work] | lol no |
| 01:50.50 | spenguin[work] | but thx for the link |
| 01:50.52 | spenguin[work] | :> |
| 01:51.07 | ruben23 | spenguin[work]:..? |
| 01:51.27 | spenguin[work] | ruben23: Im not sure about the error there |
| 01:51.51 | Katty | k |
| 01:51.55 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 01:51.56 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 01:52.02 | Katty | mister madsen |
| 01:52.10 | ruben23 | spenguin[work]: what i imake a separte extensions for it to do outbound dial |
| 01:52.23 | leifmadsen | Katty: !!! |
| 01:53.22 | *** join/#asterisk pawz (~pawz@ppp118-208-77-174.lns20.bne4.internode.on.net) |
| 01:54.11 | spenguin[work] | ruben23: I dont understand |
| 01:54.30 | *** join/#asterisk titter (~titter@c-76-101-240-142.hsd1.fl.comcast.net) |
| 01:54.54 | ruben23 | spenguin[work]: i think asterisk si blocking the number not to be dialed. |
| 01:54.55 | *** join/#asterisk gnufan (~hardev@76.91.81.190) |
| 01:55.05 | spenguin[work] | http://lists.digium.com/pipermail/asterisk-users/2005-June/105358.html |
| 01:55.10 | spenguin[work] | if it helps |
| 01:55.11 | *** join/#asterisk chendy (~chatzilla@58.250.11.38) |
| 01:56.01 | gnufan | i'm looking for Predictive Dialer app (open source ofcourse..) to integrate w/ asterisk. Any recommendation? |
| 01:56.03 | *** join/#asterisk rizwank (~rizwank@76.89.131.47) |
| 01:56.16 | spenguin[work] | http://www.voipuser.org/forum_topic_15861.html |
| 01:56.22 | leifmadsen | gnufan: likely not available for free if that's what you're asking -- you'll have to build your own, or find a commercial one |
| 01:57.08 | gnufan | darn it! |
| 01:57.17 | rizwank | I've been having trouble getting a GotoIf to work correctly --- exten => 1,n,GotoIf($[${failedlookup} = 1],FailedLookup) $failedlookup is = 1 ,but it doesn't go, returns a parse error. Later in the script, there is a exten=>1,n(FailedLookup),blah blah entry. What's wrong with the goto? |
| 01:57.22 | Katty | hugs on leifmadsen |
| 01:58.01 | leifmadsen | ;) |
| 01:58.11 | Katty | what? i'm not worth a hug? |
| 01:58.15 | Katty | )=< |
| 01:58.21 | leifmadsen | rizwank: wrong syntax |
| 01:58.28 | leifmadsen | rizwank: you have a comma where you should have a ? |
| 01:58.41 | rizwank | oh really? |
| 01:58.44 | ruben23 | <PROTECTED> |
| 01:58.44 | leifmadsen | yes really |
| 01:58.48 | ruben23 | work great |
| 01:58.50 | leifmadsen | I'm not even lying! |
| 01:58.53 | rizwank | is it in ternary format? a ? b : c |
| 01:58.54 | ruben23 | and powerful enough |
| 01:58.55 | rizwank | or just a? b |
| 01:58.55 | ManxPower-work | rizwank: Also what if ${failedlookup} is empty |
| 01:59.11 | rizwank | its a noop |
| 01:59.13 | ManxPower-work | rizwank: the specific format depends on your version of Asterisk. What does "core show application gotoif" show? |
| 01:59.15 | leifmadsen | rizwank: look at the formatting for GotoIf() with 'core show application GotoIf' |
| 01:59.22 | gnufan | let me google that |
| 01:59.42 | ruben23 | gnufan:if you need help |
| 01:59.47 | ruben23 | im happy to assist |
| 01:59.53 | rizwank | good to know |
| 01:59.54 | rizwank | thanks |
| 01:59.55 | leifmadsen | GotoIf($[value1 = value2]?true:false) |
| 02:00.13 | Katty | hmm. coca cola's new slogan is Open Happiness. |
| 02:00.15 | Katty | :/ |
| 02:00.31 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 02:00.36 | spenguin[work] | are they giving anything free? |
| 02:00.53 | Katty | you don't get it do you |
| 02:01.00 | spenguin[work] | nope |
| 02:01.03 | Katty | the effect of caffeine and high fructose corn syrup on the body |
| 02:01.11 | spenguin[work] | :\ |
| 02:01.25 | spenguin[work] | I hate the caffeine |
| 02:01.30 | leifmadsen | Katty: that's why I don't drink pop anymore -- soooo bad for you. There is a new study out linking it to pancreatic cancer too |
| 02:01.38 | Katty | their slogan should have been Open Diabetes |
| 02:01.44 | Katty | or Open Morbid Obesity |
| 02:01.50 | Katty | Open High Blood Pressure |
| 02:01.53 | Katty | etc |
| 02:02.19 | leifmadsen | Cigarettes: Cancer Sticks |
| 02:02.23 | leifmadsen | Pop: Cancer juice! |
| 02:03.09 | Katty | but no, let's market it as Open Happiness |
| 02:03.13 | Katty | scowls |
| 02:03.45 | Raden | lol |
| 02:03.48 | Katty | all these poor kids... |
| 02:03.48 | Raden | hi Katty |
| 02:03.53 | Katty | hi Raden (= |
| 02:03.55 | Katty | hugs Raden |
| 02:04.00 | Raden | hugs Katty |
| 02:04.11 | Raden | bmoraca_work, you around ? |
| 02:05.47 | gnufan | Here is good list of Predictive Dialers: http://www.voip-info.org/wiki/view/Predictive+dialer |
| 02:06.05 | *** part/#asterisk gnufan (~hardev@76.91.81.190) |
| 02:06.21 | Raden | Katty, you know much about Asterisk Realtime database ? |
| 02:09.40 | Katty | can't say i do )= |
| 02:10.27 | Raden | darn |
| 02:10.49 | Katty | but if you need a good shrimp recipe, i'm your girl! |
| 02:11.56 | Katty | hmm. riddick is shredding a toy. |
| 02:12.02 | Katty | stuffing is everywhere |
| 02:13.25 | Raden | LMAO |
| 02:13.48 | Raden | id never be lucky enough for you to be my girl Katty :P |
| 02:14.11 | Katty | psh |
| 02:14.32 | Raden | what (= |
| 02:15.08 | Raden | I dont get it i know all these women like sooo many but there all so boring and unintelligent :( |
| 02:15.49 | Raden | maybe i expect to much, I dunno |
| 02:15.56 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 02:16.00 | Raden | so whats Katty upto tonight ? |
| 02:16.02 | Katty | most people are very intersting |
| 02:16.12 | Raden | some.. |
| 02:16.12 | Katty | you just have get to know them (= |
| 02:17.00 | Raden | you talk to the average woman about computer anything or physics or science anything and yeah watch them smile and nod :( |
| 02:18.04 | Raden | people all have there interesting qualities,,, |
| 02:19.29 | Raden | maybe im the odd one :( |
| 02:19.46 | Raden | gets sickening being alone and a single parent all the time |
| 02:22.11 | seanbright | welcome to #asterisk after dark |
| 02:22.42 | seanbright | puts on some jazz and turns down the lights |
| 02:23.05 | seanbright | heh |
| 02:27.03 | *** join/#asterisk Katty (~asteriska@mail.copi-rite.com) |
| 02:27.28 | *** join/#asterisk Moc (~mochouina@pbx.moctel.com) |
| 02:27.52 | *** part/#asterisk Moc (~mochouina@pbx.moctel.com) |
| 02:28.37 | Katty | hrmm, disconnecty |
| 02:30.12 | Raden | wb, katty |
| 02:30.16 | Raden | brb 30 |
| 02:33.30 | random_mike | can anyone advise a definitive list of failure codes generated by asterisk, as seen through the AMI? |
| 02:34.20 | random_mike | by failure codes, I mean the return code on a call status |
| 02:34.28 | random_mike | ie not answered, busy, etc, etc |
| 02:37.38 | Katty | i have reached the end of superbowl commercials :< |
| 02:38.24 | Katty | how sad. |
| 02:42.18 | random_mike | 0, 1, 4, 8 etc |
| 02:42.20 | random_mike | anyone? |
| 02:42.30 | Katty | 16, 32, 64 |
| 02:42.44 | random_mike | cool what do they represent Katty ? |
| 02:42.46 | seanbright | random_mike: what version of asterisk? |
| 02:42.49 | Katty | your mom. |
| 02:43.08 | random_mike | seanbright, 1.4.28 (i believe) |
| 02:43.38 | random_mike | Katty, im Australian so it'd be my mum, non the less. Go back to watching SuperBowl commercials. You were more helpful then! |
| 02:43.50 | Katty | simmer down fancy pants. |
| 02:43.58 | Katty | you might get your panties in a bunch. |
| 02:44.01 | *** join/#asterisk Daejeo (~889f1078@gateway/web/freenode/x-jckztqcceygwokge) |
| 02:44.04 | *** join/#asterisk dansays (~dansays@CPE-24-209-154-36.wi.res.rr.com) |
| 02:44.24 | *** join/#asterisk shmaltz (~chatzilla@mail.dmaven.com) |
| 02:44.35 | Daejeo | hi |
| 02:44.40 | Katty | ohai |
| 02:44.46 | random_mike | hi2u2 |
| 02:44.56 | Daejeo | hey katty. how are you? |
| 02:45.04 | Daejeo | i do remember you |
| 02:45.06 | Katty | hmm. that's a good question |
| 02:45.10 | Katty | sits and thinks |
| 02:45.17 | Katty | my feet are cold. |
| 02:45.19 | Katty | head hurts just a bit |
| 02:45.21 | Raden | mine 2 |
| 02:45.23 | Katty | but otherwise, just peachy |
| 02:45.32 | *** join/#asterisk albertoandrade (~albertoan@189.34.121.136) |
| 02:45.32 | Katty | how're you Daejeo |
| 02:45.38 | random_mike | Katty appears to be in one of 'those' moods Daejeo ! |
| 02:45.51 | Katty | moods? |
| 02:45.53 | Katty | are you kiddin me |
| 02:45.55 | Katty | i'm always like this |
| 02:46.03 | random_mike | oh have mercy! |
| 02:46.06 | Daejeo | how about massage? |
| 02:46.11 | Raden | wow |
| 02:46.12 | Daejeo | you may feel warm |
| 02:46.23 | Katty | i'm good. |
| 02:46.25 | Daejeo | phone massage? |
| 02:46.30 | Katty | massage in a bottle |
| 02:46.34 | random_mike | this is #asterisk right? D: |
| 02:46.38 | Katty | yep. |
| 02:46.43 | random_mike | ok so guru |
| 02:46.46 | Katty | or at least it has been for the last four years |
| 02:46.54 | random_mike | guru's gather round and solve my riddle! |
| 02:47.09 | Katty | most of them are asleep |
| 02:47.17 | Katty | but keep trying, someone might wake up. |
| 02:47.49 | random_mike | yea there's usually a few lurkers who are around at this time (13:17 ACDT where I am) who assist |
| 02:48.13 | Katty | speaking of lurkers |
| 02:48.15 | Katty | dave is missing, again |
| 02:48.17 | Katty | also |
| 02:48.21 | Katty | has anyone seen jblack? |
| 02:48.26 | Katty | jblack: pinggggg |
| 02:48.31 | random_mike | never before have I seen jblack |
| 02:49.14 | Katty | jblack: well i hope you say hi soon :< i've not spoken with you in awhile--wondering how you are |
| 02:49.15 | Daejeo | blank things |
| 02:49.28 | Katty | goes to check on laundry |
| 02:49.34 | random_mike | https://wiki.xivo.fr/index.php/AMI_Commands#ExtensionState |
| 02:49.40 | random_mike | this is what I was after |
| 02:49.41 | Daejeo | what r u washing? |
| 02:49.47 | Daejeo | under? |
| 02:49.49 | Daejeo | p? |
| 02:49.54 | random_mike | I gather there are no other ExtensionStates ? |
| 02:51.01 | Daejeo | d |
| 02:51.23 | *** join/#asterisk jpvoip (~jpvoip@76.73.16.26) |
| 02:51.42 | jpvoip | hello |
| 02:51.51 | random_mike | olleh |
| 02:52.45 | random_mike | -1 = Extension not found, 0 = Idle, 1 = In Use, 2 = Busy, 4 = Unavailable, 8 = Ringing, 16 = On Hold |
| 02:52.46 | jpvoip | installed asterisk 1.6 using apt-get... now im trying to CDR be loaded on a mysql DB.... installed asterisk-addons but im getting some errors on make and make install... and . *cdr*mysql*.so dont appears |
| 02:52.51 | Daejeo | kemi_modnar |
| 02:53.04 | Katty | Daejeo: clothing. |
| 02:53.19 | Daejeo | kemi_modnar r u there? |
| 02:53.25 | random_mike | sorry to hear it jpvoip im using Centos (no mysql back end) and all seems peachy |
| 02:53.50 | Daejeo | keim_modnar r u there? |
| 02:54.04 | random_mike | Daejeo, it would appear not |
| 02:54.11 | random_mike | however I am here, if it pleases you |
| 02:54.28 | Daejeo | smart guy |
| 02:54.30 | Daejeo | u got me |
| 02:54.55 | random_mike | :D |
| 02:58.29 | *** join/#asterisk xpot (~xpot@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 03:02.07 | jpvoip | damn mysql >( |
| 03:04.30 | *** join/#asterisk Xetrov` (~xetrov@unaffiliated/xetrov/x-827361) |
| 03:07.07 | Katty | returns |
| 03:07.18 | jaytee | just like Batman |
| 03:07.26 | Raden | teleports in |
| 03:07.38 | Katty | hehe |
| 03:07.40 | Katty | hugs jaytee |
| 03:07.49 | jaytee | spinhugs Katty |
| 03:07.53 | jaytee | wheeeeee! |
| 03:08.04 | Raden | thinks kattie holds world record for hugging |
| 03:08.17 | Katty | hugs brighten everyone's day |
| 03:08.21 | Katty | embrace the idea. |
| 03:08.30 | jaytee | thinks raden is bat shit crazy for believing he can teleport |
| 03:08.39 | Raden | lol |
| 03:08.41 | Katty | maybe he's a slider. |
| 03:08.46 | Katty | maybe, he's the professor |
| 03:08.52 | Raden | ;D |
| 03:09.03 | Katty | or maybe he's Qball. |
| 03:09.11 | jaytee | one of my favorite bumper stickers: Heisenberg slept here.....or somewhere nearby. |
| 03:09.19 | Katty | lol |
| 03:09.32 | *** join/#asterisk Scorpio2007 (~Scorpio20@jose-tc.ctc.biz) |
| 03:09.34 | Katty | too cute |
| 03:13.00 | Raden | Katty, what do you do for work ? you seem so knowledgeable |
| 03:13.48 | Katty | i work for a smaller company, of about 30 people. |
| 03:14.12 | Katty | i provide service and support for all the IT/computer/networking stuffs and the asterisk/samsung/toshiba phone systems |
| 03:14.30 | jpvoip | ok |
| 03:14.56 | jpvoip | mysql installed, modules loading ok.. cdr isnt registering... how can i test if everything between asterisk and mysql are ok? |
| 03:15.11 | Katty | my company sells everything office related from shredders, to mice, to huge MFPs, and video surv stuff...furniture...laptops...etc |
| 03:15.40 | Katty | i'm samsung and toshiba certified, and i have a few mcps in 2003 server |
| 03:16.05 | Katty | it's not a fancy job tho |
| 03:16.24 | Raden | interesting |
| 03:16.55 | Raden | how do you know how Heisenberg is ? and why you seem to know so much about physics ????? |
| 03:17.35 | Katty | i know who heisenberg is because i love quantum theory |
| 03:17.58 | Katty | maybe that has something to do with the physics question too |
| 03:18.28 | Katty | i don't know a lot, really. i just like watching youtube videos that seem interesting and don't involve a guy hopping around going uhh uhhh yeahhh uhh |
| 03:18.32 | jpvoip | cdr_addon_mysql.c: Unable to query table description!! Logging disabled. |
| 03:22.18 | *** join/#asterisk riddlebox (~james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 03:22.25 | Katty | :>>>> |
| 03:22.27 | Katty | hugs riddlebox |
| 03:22.53 | Katty | jaytee: on a dark desert highway |
| 03:23.04 | Katty | jaytee: warm smell of colitas |
| 03:23.04 | jaytee | Raden, don't underestimate Katty, there's a way more there than meets the eye and it runs deep |
| 03:23.17 | jaytee | rising up through the air |
| 03:23.28 | Katty | i heard the mission bell |
| 03:23.36 | riddlebox | hey whats up |
| 03:23.43 | Raden | lol :) |
| 03:23.52 | jaytee | And I was saying to myself, this could be Heaven and this could be Hell |
| 03:23.54 | Katty | riddlebox: snow |
| 03:23.57 | Katty | riddlebox: which is falling down |
| 03:24.00 | jpvoip | cdr_addon_mysql.c: Unable to query table description!! Logging disabled. <- any idea? |
| 03:24.05 | riddlebox | how much you getting? we dont have to much yet |
| 03:24.13 | Katty | jpvoip: i'm gonna say....that you're having an issue with a query |
| 03:24.24 | Katty | jpvoip: and that if maybe you enabled logging, you could see more about what's going on |
| 03:24.37 | ChannelZ | jpvoip: bad privileges for the mysql user? |
| 03:24.40 | Katty | riddlebox: that is a good question. let me consult the national weather service |
| 03:24.58 | riddlebox | :) |
| 03:25.17 | Katty | 2-4 inches possible |
| 03:25.40 | Katty | not enough to stay home tomorrow :< |
| 03:27.03 | Raden | 6-14 here |
| 03:27.36 | Katty | great googa mooga |
| 03:27.46 | ChannelZ | googly-moogly |
| 03:27.57 | Katty | no that's from the snickers commerical |
| 03:28.02 | Katty | googa mooga is from the evolution movie |
| 03:28.10 | Katty | which if you've not seen it, watch it EL PRONTO |
| 03:28.22 | ChannelZ | I prefer peanuts and caramel |
| 03:28.34 | Katty | Diabetus Bar |
| 03:28.46 | ChannelZ | delish |
| 03:29.23 | jpvoip | ChannelZ: yes! tks!!!! |
| 03:29.29 | *** join/#asterisk shidan (~63e31495@gateway/web/freenode/x-nessggswtmzpnoul) |
| 03:30.00 | ChannelZ | sure |
| 03:37.04 | Katty | >.< |
| 03:37.16 | Katty | ryan just told me elvis died in 77 because there Can Be Only One |
| 03:37.37 | ChannelZ | Who was the other one? |
| 03:38.08 | Katty | elvis, the King |
| 03:39.06 | Katty | jaytee: ryan's song is You light up my life, by debbie boone |
| 03:39.30 | jaytee | Well why do we dream in metaphors? |
| 03:39.30 | jaytee | Try to hold on to something we couldn't understand. |
| 03:39.30 | jaytee | Couldn't understand. |
| 03:39.30 | jaytee | Tell me why must we argue loudly? When peace is our one, |
| 03:39.30 | jaytee | Salvation I couldn't understand |
| 03:39.31 | jaytee | Couldn't understand |
| 03:39.55 | ChannelZ | There once was a man from Nantucket |
| 03:40.16 | jaytee | the island where The Change started |
| 03:40.25 | Katty | ^- babylon 5 |
| 03:40.28 | Katty | ^- delene |
| 03:40.40 | jaytee | Dies The Fire |
| 03:40.46 | Katty | ^- pilot episode |
| 03:40.47 | jaytee | S.M. Stirling |
| 03:40.57 | ChannelZ | I never got into that show |
| 03:41.00 | Katty | babylon 5 <3 |
| 03:41.00 | jaytee | I have all 5 seasons of Babylon 5 |
| 03:41.09 | Katty | if you see something |
| 03:41.13 | Katty | WITH EIGHT LEGS |
| 03:41.15 | Katty | comin this way |
| 03:41.22 | Katty | let me know--i have to kill it before it develops language skills |
| 03:41.30 | *** join/#asterisk jo8330 (~4c0a99b0@gateway/web/freenode/x-quhlrowxjhlwwhye) |
| 03:41.40 | Katty | (see something this big) |
| 03:41.46 | jaytee | "No one, ever listens to poor Zathras no, he's quite mad they say. It is good that Zathras does not mind, has even grown to like it." |
| 03:41.57 | Katty | zathras talks to bugs |
| 03:42.02 | Katty | zathras sometimes eat bugs |
| 03:42.05 | Katty | very good for protein |
| 03:42.22 | jblack | katty: pooonnngg |
| 03:42.25 | Katty | jaytee: beep beep :> |
| 03:42.42 | jblack | Yeah, I've been kinda quiet, I guess. |
| 03:43.18 | Katty | hugs jblack |
| 03:43.28 | Katty | jblack: it is good to see you :> how've you been? |
| 03:44.40 | Katty | jaytee: oh not that my feelings would have any meaning for you |
| 03:44.44 | jblack | I guess I've been great. |
| 03:44.53 | Katty | jaytee: but why are you having these people POKE ME all over the place |
| 03:44.56 | jblack | Truth is, I don't really know? Kinda up, kinda down |
| 03:45.01 | Katty | jaytee: i'm a perfectly reasonable artifical personality |
| 03:45.17 | Katty | jaytee: no one's evercomplained before...but i guess some people just aren't happy unless they screwin it up for everyone else! |
| 03:45.21 | Katty | jaytee: you know you really should stand up straight |
| 03:45.35 | Katty | jaytee: your mother and i have been worried about it for years... we've been trying to talk to you about it |
| 03:45.40 | jaytee | Katty, what are you going on about? |
| 03:45.44 | Katty | jaytee: pzzzzzPING |
| 03:45.47 | Katty | what?! |
| 03:45.51 | jblack | How about you? |
| 03:46.45 | Katty | jaytee: season 3, episode 11, ceremonies of light and dark |
| 03:46.55 | jo8330 | hi all. does anyone know what codec is applied for PRI? |
| 03:47.02 | jaytee | ok, didn't remember |
| 03:47.09 | Katty | jaytee: sparky the computer |
| 03:47.20 | Katty | jaytee: garibaldi pulls out his ppg and shoots the speaker int he turbolift |
| 03:47.20 | jaytee | talking to Garibaldi? |
| 03:47.23 | Katty | mhmm |
| 03:47.27 | Katty | jblack: i am goodly :> |
| 03:47.35 | jaytee | yeah, now it's coming back to me |
| 03:47.58 | Katty | luck? i don't believe in luck....... |
| 03:48.32 | *** join/#asterisk rvhi (~chatzilla@udp219966uds.hawaiiantel.net) |
| 03:48.56 | jblack | How are the critters doing? |
| 03:49.15 | jo8330 | i'm guessing G.711 |
| 03:49.26 | Katty | jblack: oh i suppose okay. |
| 03:49.40 | Katty | jblack: http://farm4.static.flickr.com/3107/3125404776_f1c79e313f_o.jpg (= |
| 03:49.58 | jo8330 | nice doggy. |
| 03:50.59 | rvhi | after asterisk running, i do a 'lsmod | grep dahdi_dummy', it shows 'dahdi_dummy 5604 0', it is normal? Isn't the used by field supposed to be greater than 0? |
| 03:51.44 | jblack | That's a fantastic pic of Riddick. |
| 03:51.52 | Katty | thanks :> |
| 03:51.57 | *** part/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 03:52.20 | Katty | he has a very cute schnozzle |
| 03:52.21 | [TK]D-Fender | Katty: I concur. Great quality and creative shot |
| 03:52.48 | jblack | fb is a wreck tonight. |
| 03:53.08 | Katty | oh? |
| 03:53.16 | Katty | what's going on on the big fb |
| 03:53.31 | jaytee | they changed the whole format |
| 03:53.37 | jblack | They even fucked up the css |
| 03:53.43 | jaytee | and I quit playin Mafia Wars |
| 03:53.47 | Katty | looks |
| 03:53.56 | ChannelZ | works fine here |
| 03:54.14 | Katty | oh...it does look different. hmmmmmm |
| 03:54.17 | jblack | ChannelZ: Fucked up as now forcing a fixed width. |
| 03:55.08 | ChannelZ | too bad they've never managed to implement a sensible layout scheme in html yet |
| 03:55.30 | Katty | that would load too quickly |
| 03:55.34 | Katty | can't have that |
| 03:56.18 | ChannelZ | Anyway guess I'm lucky, the width is pretty much right for my browser window |
| 03:56.50 | Katty | mine too |
| 03:57.08 | Katty | i wonder how it looks on the irock. |
| 03:57.10 | Katty | i mean ipad. |
| 03:57.30 | ChannelZ | Gorgeous, beautiful, magical |
| 03:57.57 | jaytee | heard Apple's got another, larger tablet model coming out soon, the iMaxipad |
| 03:58.07 | Katty | hehehhe |
| 03:58.09 | Katty | heheheh |
| 03:58.11 | Katty | oh hhh that's good |
| 03:58.11 | ChannelZ | Light, Medium and Heavy iPads |
| 03:58.35 | Katty | do they come with refreshing wipes too? |
| 03:58.38 | Katty | >;P |
| 03:59.00 | jblack | Facebook is like that crazy girlfriend you dated for 2 months that was apt to show up with half of her hair dyed green and the other half shaved off. |
| 03:59.11 | Katty | ^- my sister |
| 03:59.13 | jaytee | post that!!!!! |
| 03:59.22 | jblack | ok |
| 03:59.26 | jaytee | hehe |
| 03:59.37 | ChannelZ | http://www.youtube.com/watch?v=1ZS8HqOGTbA |
| 04:01.55 | Katty | niceee |
| 04:02.00 | Katty | youtube won't buffer |
| 04:02.01 | Katty | >.< |
| 04:02.05 | Katty | WHO STOLE MY BUFFERS |
| 04:02.11 | ChannelZ | Al Gore |
| 04:02.14 | Xetrov` | runs |
| 04:02.46 | jblack | I hope you guys mark line on that one.. 'specially you, jaytee. |
| 04:02.58 | jaytee | mark line? |
| 04:03.48 | jblack | Where'd the post go? |
| 04:04.06 | jblack | There it is, on most recent. |
| 04:04.22 | jblack | We can like our own posts? |
| 04:05.08 | jaytee | you can, but it's kind of tacky |
| 04:05.26 | jblack | Yeah. |
| 04:05.57 | jaytee | but hey, we live in an age of shameless self-promotion so it'll fly |
| 04:06.45 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 04:07.06 | Katty | http://www.youtube.com/watch?v=DgLvT3yOF70 |
| 04:08.01 | jaytee | jblack, I reposted it |
| 04:08.22 | jaytee | but I said it was reposted from a friend, didn't try to plaigarize |
| 04:09.31 | *** join/#asterisk smooth_penguin (~smoove@59.95.15.213) |
| 04:10.04 | jpvoip | man, i luv * |
| 04:10.13 | Katty | smooooooooooooooooth operator |
| 04:10.20 | jaytee | Sade |
| 04:10.24 | smooth_penguin | heya Katty :> |
| 04:10.39 | smooth_penguin | is just back from the night shift |
| 04:10.44 | smooth_penguin | must get in bed quick |
| 04:10.51 | smooth_penguin | :s |
| 04:12.00 | Katty | buhbye |
| 04:12.18 | ChannelZ | time for Hoarders |
| 04:13.35 | *** join/#asterisk sjobeck (~sjobeck@74.85.224.166) |
| 04:13.44 | [TK]D-Fender | jaytee: Copying from 1 person = plagiarism. Copying from 100 people = research. |
| 04:13.52 | *** part/#asterisk sjobeck (~sjobeck@74.85.224.166) |
| 04:13.53 | jaytee | lol |
| 04:14.26 | jaytee | plus I spelled it wrong. put the i in the wrong spot |
| 04:14.33 | jblack | jaytee: That's ok. I want the idea out. |
| 04:14.52 | jblack | fb should have a retweet option |
| 04:14.57 | Katty | auto tune <3 |
| 04:15.32 | *** join/#asterisk riddlebox (~james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 04:16.12 | [TK]D-Fender | Katty: Auto-tune? You know... people debate that were they to succeed in making a time machine, would they go back in time to kill Hitler. Me? No.. I'd go back to kill T-Pain. |
| 04:16.56 | smooth_penguin | just wants a hoverboard |
| 04:17.13 | smooth_penguin | http://en.wikipedia.org/wiki/Hoverboard |
| 04:17.19 | [TK]D-Fender | HELLO?!?!?!? MCFLY!?!??!?! |
| 04:17.27 | [TK]D-Fender | pummels smooth_penguin with his cane |
| 04:17.31 | smooth_penguin | haha :> |
| 04:17.42 | smooth_penguin | I love that movie |
| 04:19.05 | smooth_penguin | Im out, good night. |
| 04:19.18 | jaytee | time for me to get some sleep too |
| 04:19.20 | jaytee | nite all |
| 04:19.42 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 04:19.59 | *** part/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 04:25.18 | Katty | naptime! |
| 04:25.20 | Katty | g'lator |
| 04:27.02 | Raden | Katty, is it not past your bedtime ? |
| 04:30.07 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 04:31.05 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 04:35.23 | b14ck | I'm having a weird problem. |
| 04:35.40 | b14ck | MeetMe() seems to be hanging up on calls. I'm using a pure SIP system with dahdi_dummy loaded :x |
| 04:51.25 | *** join/#asterisk Katty (~asteriska@mail.copi-rite.com) |
| 04:52.04 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-lcdsqdocjikobjfk) |
| 04:52.44 | *** join/#asterisk loopy577 (~jlupresto@74-81-241-158.static.sdyl005.digis.net) |
| 04:52.52 | *** join/#asterisk sjobeck (~sjobeck@valdisere.sjobeck.com) |
| 04:56.25 | b14ck | Whenever I create a meetme bridge I get the following CLI warning: "" and the call hangs up after 30ish seconds. I've confirmed that it isn't nat or my provider. Any suggestions? I think maybe DAHDI dummy is messed up somehow :x |
| 04:57.32 | b14ck | *warning: [Feb 9 08:03:38] WARNING[30823]: channel.c:3963 ast_request: No channel type registered for 'DAHDI' |
| 04:57.55 | [TK]D-Fender | b14ck: where do we see chan_dahdi loaded? |
| 04:58.31 | b14ck | ahha! |
| 04:58.39 | b14ck | chan_dahdi.so does not exist in this build :x |
| 04:58.49 | b14ck | Command 'module load chan_dahdi.so' failed. |
| 04:58.50 | b14ck | [Feb 9 08:04:34] WARNING[30827]: loader.c:385 load_dynamic_module: Error loading module 'chan_dahdi.so': /usr/lib/asterisk/modules/chan_dahdi.so: cannot open shared object file: No such file or directory |
| 05:00.00 | [TK]D-Fender | b14ck: SMRT |
| 05:00.06 | b14ck | [TK]D-Fender, thanks :) |
| 05:00.15 | b14ck | *testing it now* |
| 05:12.43 | b14ck | [TK]D-Fender, still having the issue with chan_dahdi loaded, actually :( But the warning is now gone. The symptom is that all meetme calls hang up after a few seconds. |
| 05:12.51 | b14ck | The sip debug / core debug shows that we are sending the BYE. |
| 05:13.46 | b14ck | I also get this right before the hangup: http://pastie.org/private/5426l1fvldrzn2kk7dxgaw |
| 05:14.33 | b14ck | The only other thing I see is the devstate_event message |
| 05:14.39 | b14ck | setting the meetme state to 1 |
| 05:24.31 | [TK]D-Fender | checkout time, later all |
| 05:31.20 | *** join/#asterisk vAd0r (~IceChat7@65.67.210.121) |
| 05:31.27 | *** join/#asterisk KingDavidNYC (~Chris1232@pool-96-224-162-219.nycmny.east.verizon.net) |
| 05:31.28 | vAd0r | what file do i change my new pub ip on |
| 05:31.41 | KingDavidNYC | Hello everybody!! |
| 05:32.09 | *** join/#asterisk AeroCloud (~aero@ip72-222-149-220.ph.ph.cox.net) |
| 05:32.52 | AeroCloud | on a dial option t, and when you transfer to a new extension.. can I force the transfer to hangup? I have tried added hangup in the dialplan and doesnt work |
| 05:33.11 | AeroCloud | the person doing the transfer |
| 05:33.32 | KingDavidNYC | can anybody please tell me how to compile 1.4.23 without dahdi support |
| 05:34.20 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 05:43.51 | Raden | make menu |
| 05:43.55 | Raden | disable in there i believe |
| 05:44.01 | Raden | make menuselect |
| 05:44.03 | Raden | something like that |
| 05:46.26 | KingDavidNYC | thanks, I did, but the option is not there |
| 05:47.17 | *** join/#asterisk fiddur (~fiddur@192.121.104.121) |
| 05:48.02 | Raden | cant u just disable it ? |
| 05:48.27 | KingDavidNYC | dont know how to |
| 05:49.16 | KingDavidNYC | I was doing meet me very well with 1.2, but I recompiled to 1.4 and now meetme does not work |
| 05:50.25 | ruben23 | hi |
| 05:50.27 | ruben23 | is this |
| 05:50.35 | ruben23 | valid exten => _9X.,2,Dial(${TrunkDEL}/${EXTEN:1},,o) |
| 05:58.48 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 05:58.52 | *** join/#asterisk joobie (~joobz@mx01.anric.com.au) |
| 05:59.09 | ChannelZ | I guess that depends on what ${TrunkDEL} is |
| 05:59.11 | joobie | hey guys.. with extensions.conf, how can i do an execif to see if ${VAR} is defined? |
| 05:59.23 | joobie | just want to check if it's defined, not what it's set to |
| 06:00.22 | ChannelZ | joobie: core show function EXISTS |
| 06:00.23 | ChannelZ | perhaps |
| 06:01.13 | joobie | thanks |
| 06:04.19 | ruben23 | can i make shortcut with this extensions..? ---> exten => _91NXXNXXXXXX,2,Dial(${TRUNKX}/${EXTEN:1},,To) |
| 06:04.28 | *** join/#asterisk JAMMAN2110 (~james@unaffiliated/jamman2110) |
| 06:04.39 | ruben23 | instead of using long _91NXXNXXXXXX |
| 06:05.01 | ChannelZ | huh? |
| 06:05.11 | *** join/#asterisk b14ck (~comradeb1@cpe-24-24-136-239.socal.res.rr.com) |
| 06:05.22 | ruben23 | ChannelZ: like making it _X. |
| 06:05.44 | ruben23 | not so familliar but i have seen it on several setups |
| 06:05.48 | ChannelZ | well technically yes if that doesn't interfere with the rest of your dialplan |
| 06:05.58 | ChannelZ | IE everything you dial would match that |
| 06:07.07 | ruben23 | ChannelZ: ill just put like this..? _91X. right..? |
| 06:07.33 | ChannelZ | sure knock yourself out |
| 06:09.14 | joobie | EXISTS($INT_CALLER_ID)|NoOp|w00t |
| 06:09.34 | joobie | something wrong with that? $INT_CALLER_ID is defined in sip.conmf against a user with setvar=INT_CALLER_ID=blah |
| 06:09.43 | ChannelZ | yeah pretty much everything is wrong with that |
| 06:10.12 | ChannelZ | EXISTS is a function, not an application |
| 06:10.49 | ChannelZ | so you still need a 'GotoIf' or some other application which actually does something |
| 06:11.27 | ChannelZ | Also the syntax for a function would be ${EXISTS(args)} |
| 06:12.01 | joobie | exten => 4001,1,ExecIf(EXISTS(${INT_CALLER_ID})|NoOp|w00t) |
| 06:12.03 | ChannelZ | and $INT_CALLER_ID should be ${INT_CALLER_ID} |
| 06:12.04 | joobie | what i do |
| 06:12.41 | joobie | sorry - slight variation from waht i typed |
| 06:12.41 | joobie | :P |
| 06:12.43 | ChannelZ | and the whole thing needs to be evaluated |
| 06:13.00 | joobie | so enclose in $[] ? |
| 06:13.39 | ChannelZ | yeah |
| 06:13.51 | joobie | no good |
| 06:13.53 | ChannelZ | the rest is wrong though |
| 06:13.58 | ChannelZ | core show application ExecIf |
| 06:13.59 | joobie | what's wrong? |
| 06:14.00 | random_mike | whats the best method for disabling debug into /var/log/asterisk/full - for lines like: [Feb 9 16:43:13] VERBOSE[8197] logger.c: Really destroying SIP dialog '1223fc5473e6e96b7188bc755d345a68@219.90.129.208' Method: REGISTER |
| 06:14.12 | ChannelZ | it's ExecIf(evaluation?true:false) |
| 06:14.43 | joobie | but i dont want to do anything on false, so ive left it out |
| 06:14.48 | ChannelZ | random_mike: see /etc/asterisk/logger.conf |
| 06:14.55 | random_mike | thx |
| 06:15.00 | ChannelZ | joobie: that's fine but your syntax is still wrong |
| 06:15.46 | joobie | exten => 4001,1,ExecIf($[EXISTS(${INT_CALLER_ID})]|NoOp|w00t) |
| 06:15.52 | joobie | that's my syntax now |
| 06:16.16 | joobie | im just using | as the seperator - which works for other execif functions |
| 06:16.32 | ChannelZ | still wrong.. RE: ExecIf(evaluation?whattodoiftrue:whattodoiffalse) |
| 06:16.54 | ChannelZ | you still messed up the function |
| 06:17.35 | ChannelZ | ExecIf($[${EXISTS(${INT_CALLER_ID})}]?NoOp:w00t) |
| 06:19.06 | *** join/#asterisk VtlChris (chrish@box1.vitel.net) |
| 06:21.02 | joobie | exten => 4001,1,ExecIf($[${EXISTS(${INT_CALLER_ID})}]|CALLERID(name)=${INT_CALLER_ID}) |
| 06:21.23 | *** join/#asterisk pawz_ (~pawz@ppp118-208-3-188.lns20.bne1.internode.on.net) |
| 06:21.25 | joobie | what about that? it goes through now but says it cant find application (CALLEDID(name)=var) |
| 06:21.46 | joobie | ahh woops dropped the set |
| 06:21.48 | ChannelZ | now you have new problems |
| 06:22.47 | joobie | man |
| 06:22.47 | joobie | wat a drama |
| 06:22.47 | joobie | exten => 4001,1,ExecIf($[${EXISTS(${INT_CALLER_ID})}]|Set(CALLERID(name)=${INT_CALLER_ID})) |
| 06:22.47 | joobie | that's what i have now |
| 06:23.06 | joobie | i jsut want to go home and get pissed |
| 06:23.17 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 06:23.55 | joobie | yey got it |
| 06:23.58 | *** part/#asterisk benngard (~benngard@90-230-92-67-no148.tbcn.telia.com) |
| 06:23.58 | joobie | thanks ChannelZ |
| 06:24.07 | joobie | stupid delimiters |
| 06:29.11 | *** join/#asterisk tris (tristan@camel.ethereal.net) |
| 06:31.42 | vAd0r | ive been able to make calls out my friends box to mine but when i call him i get -- SIP/hispubip-09e3af08 is circuit-busy |
| 06:31.49 | vAd0r | any ideas? |
| 06:34.23 | ChannelZ | Perhaps his circuit is busy. Or his side is misconfigured to handle the call |
| 06:35.12 | ChannelZ | What are you Dial()ing? Does he have an entry in your sip.conf or are you calling 'anonymously'? |
| 06:35.41 | *** join/#asterisk tris (tristan@camel.ethereal.net) |
| 06:35.49 | vAd0r | i think im calling anonymously |
| 06:35.51 | *** join/#asterisk funtoo_nbu (seb@jet.bayhost.net) |
| 06:36.48 | ChannelZ | ok then he probably has an issue |
| 06:37.17 | vAd0r | i see the call in his astrisk console |
| 06:37.23 | vAd0r | it hits his system |
| 06:38.30 | ChannelZ | really well I'm not clairvoyant so I can't see his console to venture a futher guess |
| 06:38.38 | vAd0r | lol |
| 06:38.54 | vAd0r | i get the person you are calling is unavailable please try again |
| 06:39.05 | VtlChris | trixbox? |
| 06:39.43 | vAd0r | his is |
| 06:41.51 | *** join/#asterisk Tim_Toady (~moi@77.49.48.61.dsl.dyn.forthnet.gr) |
| 06:44.00 | *** join/#asterisk xpot (~xpot@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 06:53.26 | *** join/#asterisk joako_ (~joako@opensuse/member/joak0) |
| 06:53.48 | *** join/#asterisk |Cybex| (~John@atwork-21.r-212.178.82.atwork.nl) |
| 06:54.50 | vAd0r | SIP/2.0 403 Forbidden |
| 06:55.38 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
| 06:57.15 | *** join/#asterisk Heretic (~fallen@dsl-246-76-110.telkomadsl.co.za) |
| 06:57.54 | *** join/#asterisk soman (~somnath@stargate.starnet.fi) |
| 06:58.02 | ChannelZ | that's your side saying it or his? |
| 06:58.55 | ChannelZ | I mean it's his side saying it but you're looking at a debug on your side? |
| 06:59.39 | Heretic | lo all |
| 07:00.19 | vAd0r | his side |
| 07:00.25 | vAd0r | his debug |
| 07:03.02 | ChannelZ | well there's a lot of reasons why it might return that, again, not the psychic hotline here without seeing a lot more console and configs |
| 07:03.12 | ChannelZ | Probably it's an auth problem. Why, I don't know |
| 07:04.44 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-hxxhqenfnhjuyaoi) |
| 07:09.04 | *** join/#asterisk benngard (~benngard@213.88.138.230) |
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| 07:16.53 | *** join/#asterisk dudee (~dude@115.42.138.246) |
| 07:18.53 | dudee | Hi , have a question, my DISA lines are not being recorded, i get this, No AMPUSER db entry for {number} . Not recording" |
| 07:21.50 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 07:25.44 | funtoo_nbu | Is it a pia to setup an asterisk answering service? anyone got a good howto? |
| 07:30.03 | JAMMAN2110 | I dont expect much of a response to this, but does anyone know of any problems running Asterisk on a HP / Compaq D330UT? |
| 07:32.06 | ChannelZ | funtoo_nbu: not really but that's sort of a broad question |
| 07:38.29 | funtoo_nbu | yea ima newb |
| 07:46.10 | dudee | Hi , have a question, my DISA lines are not being recorded, i get this, No AMPUSER db entry for {number} . Not recording" |
| 07:48.47 | ChannelZ | is this freepbx? |
| 07:49.07 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 07:52.32 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 07:53.46 | *** join/#asterisk mbp (~mbp@81.161.143.71) |
| 07:54.13 | mbp | What are the best tools for extracting queue statistics / history on calls using the Asterisk AMI? |
| 08:00.19 | *** join/#asterisk c0rnoTa (~c0rnoTa@178.176.146.168) |
| 08:02.43 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-winffjdvgmivtray) |
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| 08:05.38 | *** join/#asterisk c0rnoTa (~c0rnoTa@178.176.146.168) |
| 08:11.14 | *** join/#asterisk mpe (~mpe@gate.ipvision.dk) |
| 08:20.47 | *** join/#asterisk Sky_11 (~jleicher@mail.sysun-technologies.com) |
| 08:21.00 | Sky_11 | Hi ! |
| 08:24.33 | *** join/#asterisk hurdman (~ngeek@ys.antredugeek.fr) |
| 08:25.20 | hurdman | hello folk ! |
| 08:25.43 | *** join/#asterisk Sky_11 (~jleicher@mail.sysun-technologies.com) |
| 08:25.52 | Sky_11 | Hi ! :/ |
| 08:27.00 | hurdman | is there anyone who use ss7 with asterisk and have a good feed-back of chan_ss7 ? |
| 08:29.54 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
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| 08:46.25 | *** join/#asterisk pietro (~pietro@88.149.164.32) |
| 08:46.38 | *** join/#asterisk mbranca (~matteo@host139-217-static.224-95-b.business.telecomitalia.it) |
| 08:47.45 | pietro | hi, currently is possible to use asterisk 1.6 an a hfc isdb BRI card ? |
| 08:48.10 | c0rnoTa | good day, All |
| 08:48.27 | *** join/#asterisk pawz (~pawz@ppp118-208-108-108.lns20.bne4.internode.on.net) |
| 08:52.30 | sun28 | moin \o/ |
| 08:58.41 | Sky_11 | Does anyone ever use mjua ? |
| 09:01.01 | *** join/#asterisk franck (~franck@tikiwiki/franck) |
| 09:02.09 | *** join/#asterisk Da-Geek (~Da-Geek@80.235.230.227) |
| 09:06.14 | *** join/#asterisk |Cybex| (~John@atwork-21.r-212.178.82.atwork.nl) |
| 09:07.08 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 09:07.14 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 09:22.10 | c0rnoTa | pietro: i don't know |
| 09:22.32 | c0rnoTa | Sky_11: what is mjua. never heared about it |
| 09:23.11 | c0rnoTa | hurdman: i can only said, that it works :) |
| 09:34.20 | hurdman | c0rnoTa: ok :) |
| 09:34.52 | hurdman | c0rnoTa: with the standard version of chan_ss7 ? |
| 09:36.49 | c0rnoTa | hurdman: http://www.netfors.com/media/download/chan_ss7-1.3.tar.gz |
| 09:37.26 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 09:37.37 | c0rnoTa | Version 1.3, 2009-12-08 |
| 09:40.20 | Sky_11 | c0rnoTa: It's an example UA based on mjsip stack, I try to write a webphone based on that UA |
| 09:41.54 | hurdman | c0rnoTa: it's the version i use, but in france, we have a stupid frogg's procole calls "spirou" ... /o\ which is a specific implementation of isup-v2 |
| 09:43.08 | c0rnoTa | hurdman: never heared about spirou |
| 09:43.28 | hurdman | c0rnoTa: you're a luky man :) |
| 09:44.53 | c0rnoTa | hurdman: i'm not sure :) |
| 09:46.01 | *** join/#asterisk TimeRider (~steve@94-194-90-178.zone8.bethere.co.uk) |
| 09:51.13 | *** join/#asterisk sgimeno (~chatzilla@226.Red-80-33-64.staticIP.rima-tde.net) |
| 09:53.42 | *** join/#asterisk Faithful (~Faithful@ns.linuxterminal.com) |
| 10:16.38 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 10:16.53 | *** join/#asterisk pentanol (~pentanol@77-35-1-202.pppoe.primorye.net.ru) |
| 10:17.18 | pentanol | hi |
| 10:17.27 | pentanol | amubody alive there? |
| 10:19.22 | hurdman | yes me ^^ |
| 10:20.18 | Sky_11 | Me too ... I think |
| 10:20.42 | Sky_11 | Wait ! ... Yeah, I'm alive |
| 10:21.31 | *** join/#asterisk mrprozac (~mrprozac@62.59.46.85) |
| 10:23.33 | *** join/#asterisk UQlev (~yuriy@nb11-125.static.cytanet.com.cy) |
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| 10:25.33 | *** join/#asterisk bzing2 (~dr105@dhcp-194-66-208-236.canterbury.ac.uk) |
| 10:25.43 | c0rnoTa | yes, me too |
| 10:29.40 | tzafrir_laptop | infobot, tell pentanol about ask |
| 10:29.42 | *** join/#asterisk Buklov (~buklov@213.138.71.254) |
| 10:30.12 | tzafrir_laptop | notes pentanol is probably alive and is in the channel. |
| 10:30.20 | pentanol | right |
| 10:30.46 | pentanol | I guess someone use asterisk with radius? |
| 10:31.05 | tzafrir_laptop | I guess someone does. |
| 10:31.07 | *** join/#asterisk _omer (~omer@119.152.240.49) |
| 10:31.17 | _omer | hello |
| 10:31.20 | pentanol | actually I'm looking for DB schema.... |
| 10:31.29 | pentanol | perhaps someone have this one? |
| 10:31.30 | _omer | Suggestions/Help needed ... http://www.pastebin.org/88770 |
| 10:32.29 | *** join/#asterisk TimeRider (~steve@78.32.26.1) |
| 10:32.45 | c0rnoTa | pentanol: i have used radius only for CDR |
| 10:32.57 | c0rnoTa | pentanol: not for auth |
| 10:33.37 | tzafrir_laptop | pentanol, did you read the dcrradius doc? under doc/tex ? |
| 10:33.45 | _omer | Suggestions/Help needed, related to Realtime Extensions ... http://www.pastebin.org/88770 |
| 10:33.51 | tzafrir_laptop | Which means it is in asterisk.pdf |
| 10:35.48 | *** join/#asterisk ccesario (~ccesario@189-19-6-236.dsl.telesp.net.br) |
| 10:36.39 | pentanol | c0rnoTa cool ;) could you make show create .... \g; and post some where? |
| 10:37.46 | pentanol | tzafrir_laptop hm, which doc? I'm not found any sane doc with DB schema |
| 10:40.13 | c0rnoTa | pentanol: in my case it was only dictionary for radius |
| 10:41.19 | pentanol | c0rnoTa i.e. you use shema from freeradius branch? |
| 10:41.26 | pentanol | schema* |
| 10:42.41 | c0rnoTa | pentanol: yes. it was dictionary.digium |
| 10:43.08 | pentanol | agh, I got it idea |
| 10:43.12 | c0rnoTa | http://www.voip-info.org/wiki/view/Asterisk+Documentation+1.4+radius.txt |
| 10:44.16 | *** join/#asterisk Godfather_ (~Godfather@184.Red-83-58-87.dynamicIP.rima-tde.net) |
| 10:44.20 | Godfather_ | o/ |
| 10:46.30 | c0rnoTa | what means o/ ? :) or /o\ ? |
| 10:49.15 | tzafrir_laptop | hand-waving? |
| 10:50.06 | Faustov | o-slash. a polite version of oh fuck |
| 10:50.50 | tzafrir_laptop | I can't really prove he meant that, though |
| 10:51.36 | _omer | o/ |
| 10:51.47 | _omer | ;) |
| 10:52.27 | _omer | is there any one have a minute for me ? |
| 10:52.31 | _omer | Suggestions/Help needed, related to Realtime Extensions ... http://www.pastebin.org/88770 |
| 10:54.31 | *** join/#asterisk mallchin (~mallchin@mail.dataproservices.co.uk) |
| 10:54.59 | mallchin | hi guys, I'm trying to do a cut but not getting much luck using a comma, can someone assist please? |
| 10:55.41 | mallchin | exten => s,1(loop_start),Set(STORIES_LIST="one,two,three") |
| 10:55.41 | mallchin | exten => s,n,Set(PICK_FILE=${RAND(1,3)}) |
| 10:55.42 | mallchin | exten => s,n,Set(PLAY_FILE=${CUT(STORIES_LIST,\,,${PICK_FILE})}) |
| 10:56.08 | mallchin | PLAY_FILE = "" :-( |
| 10:56.42 | mallchin | Chainsaw: o/ |
| 10:57.14 | *** join/#asterisk TimeRider (~steve@78.32.26.1) |
| 11:00.41 | _omer | am I lost in asterisk world now ? |
| 11:07.32 | kaldemar | mallchin: show a cli output of a call for that. |
| 11:07.43 | *** join/#asterisk Diogo (~info@a85-138-16-15.cpe.netcabo.pt) |
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| 11:13.49 | _trine | anyone know how to get audio when I connect ekiga to my asterisk over a VPN tunnel ;;; everything works until I try it over a tunnel |
| 11:14.21 | *** join/#asterisk zamba (marius@flage.org) |
| 11:14.30 | zamba | my asterisk is giving me syslog messages like this: rc_avpair_new: unknown attribute 1490026597 |
| 11:15.06 | *** join/#asterisk Diogo (~info@a85-138-16-15.cpe.netcabo.pt) |
| 11:15.37 | Chainsaw | Hi there mallchin. |
| 11:17.15 | *** join/#asterisk sgimeno (~chatzilla@226.Red-80-33-64.staticIP.rima-tde.net) |
| 11:42.53 | Acro | default has 0 calls (max unlimited) in 'ringall' strategy (0s holdtime), C:0, A:0, SL:0.0% within 0s |
| 11:42.59 | mallchin | kaldemar: thanks, I decided to opt for '+' as a delimiter instead, would have been good to get comma working though. If I re-visit the issue I'll come back and paste cli output :) |
| 11:42.59 | Acro | Can anyone explain the above values? |
| 11:43.06 | Acro | Chainsaw, A and holdtime in particular. |
| 11:43.17 | Acro | Sorry Chainsaw, didn't mean to highlight. |
| 11:43.22 | Acro | * C, A and ... |
| 11:44.02 | c0rnoTa | С - еÑеÑд ÑÑÐ´Ð´Ñ |
| 11:44.06 | c0rnoTa | C - total calls |
| 11:44.19 | c0rnoTa | A - abandon calls |
| 11:44.30 | c0rnoTa | SL - service level (i think) |
| 11:45.00 | c0rnoTa | hold time - current maximum hold time |
| 11:45.14 | Acro | c0rnoTa, thanks a lot. |
| 11:45.25 | c0rnoTa | Acro: wait |
| 11:45.40 | c0rnoTa | Acro: may be C = COMPLETED calls |
| 11:47.07 | c0rnoTa | Acro: it's only my opinion, you should know it |
| 11:47.27 | Acro | Now I just need some way to persist (or store these numbers) after a reload. QueueMetrics or Asternic or something better? I don't care about the reports, just the data persistance in a DB. |
| 11:47.54 | Acro | c0rnoTa, unsure about A. Our A is 50 % of the C. Seems like a high number of abandoned calls, with a very small queue. |
| 11:49.37 | c0rnoTa | Acro: queue3003 has 1 calls (max 99) in 'ringall' strategy (39s holdtime), W:0, C:250, A:62, SL:0.0% within 0s |
| 11:49.37 | c0rnoTa | Seems to be real abandon counter |
| 11:51.10 | c0rnoTa | and it seems that SL not service lever |
| 11:51.53 | c0rnoTa | it looks like i had lied to you. sorry ^( |
| 11:53.38 | Acro | I think SL is service level. It's just rather useless, unless I figure out where I configure the time frame (0s is too low ;-)) |
| 11:53.47 | *** join/#asterisk Poincare (~jefffnode@amp89.ampersant.be) |
| 11:55.54 | *** join/#asterisk bmg505 (~leon@196-209-79-222-rndf-esr-5.dynamic.isadsl.co.za) |
| 11:56.16 | c0rnoTa | :) |
| 12:11.19 | *** join/#asterisk mbranca (~matteo@host139-217-static.224-95-b.business.telecomitalia.it) |
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| 12:21.37 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:21.38 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 12:32.29 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 12:33.14 | _omer | Suggestions/Help needed, related to Realtime Extensions ... http://www.pastebin.org/88770 |
| 12:36.12 | *** join/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 12:40.28 | *** join/#asterisk came0 (~came0@32.132.160.106) |
| 12:40.38 | *** part/#asterisk came0 (~came0@32.132.160.106) |
| 12:43.02 | *** join/#asterisk smooth_penguin (~smoove@59.95.36.64) |
| 12:44.14 | smooth_penguin | so why was zaptel renamed to dahdi? |
| 12:44.50 | c0rnoTa | smooth_penguin: it was described somewhere on asterisk.org |
| 12:45.07 | smooth_penguin | ok |
| 12:46.26 | leifmadsen | smooth_penguin: quick search of 'dahdi' on blogs.digium.com turns up this link: http://blogs.digium.com/2008/05/19/zaptel-project-being-renamed-to-dahdi/ |
| 12:46.59 | leifmadsen | I just found a dead link though -- so I'm reporting that |
| 12:48.05 | *** join/#asterisk benngard (~benngard@213.88.138.230) |
| 12:48.41 | leifmadsen | probably didn't bother to click the links either |
| 12:53.11 | _omer | why do I have to use "switch = Realtime/@" in each context manually in extensions.conf to make this context realtime ? |
| 12:53.24 | _omer | how to avoid this manual work in extensions.conf ??? |
| 13:01.18 | c0rnoTa | _omer: i have realtime extensions |
| 13:01.29 | c0rnoTa | _omer: but only 1 context |
| 13:01.46 | leifmadsen | I think you use /etc/asterisk/extconfig.conf if you don't want to generate any manual extensions.conf |
| 13:02.10 | _omer | yes. I am using extconfig.conf as well |
| 13:02.25 | *** join/#asterisk garymc (~chatzilla@host81-139-136-16.in-addr.btopenworld.com) |
| 13:02.29 | leifmadsen | then the whole file should be loaded into memory and not need extensions.conf as far as I'm aware |
| 13:02.45 | leifmadsen | I don't use extensions.conf from realtime though because I don't like how it operated. I use func_odbc for dynamic data |
| 13:03.24 | _omer | it is not like that... please check it out |
| 13:03.25 | _omer | http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions |
| 13:04.08 | c0rnoTa | _omer: call from extensions.conf in my case looks like switch => Realtime/ivr_holidays@extensions |
| 13:04.17 | leifmadsen | ya... I think that switch example is an alternative method to loading the whole file from the database |
| 13:04.19 | leifmadsen | you shouldn't have to do that |
| 13:04.35 | leifmadsen | honestly, I have no idea -- I'm beyond my knowledge as I've used realtime lots, but never extensions.conf |
| 13:04.42 | leifmadsen | moves along |
| 13:04.46 | _omer | c0rnoTa: yes. where ivr_holidays is your context ... what about if you want to create another context? |
| 13:05.32 | _omer | have you created ivr |
| 13:05.33 | *** join/#asterisk mazpe (~mazpe@ec2-174-129-37-13.compute-1.amazonaws.com) |
| 13:05.48 | mazpe | whts the cli command to know what version i'm running |
| 13:05.59 | leifmadsen | core show version |
| 13:05.59 | _omer | have you created ivr_holidays in extensions.conf manually and then used switch => Realtime/ivr_holidays@extensions under this context to make it working with DB ? |
| 13:06.07 | mazpe | core! |
| 13:06.09 | mazpe | thanks |
| 13:06.20 | *** join/#asterisk ReDNeQ (~ReDNeQ@70.114.229.58) |
| 13:06.25 | c0rnoTa | _omer: for additional context you should update your context |
| 13:06.46 | _omer | in extensions.conf ? |
| 13:07.15 | *** join/#asterisk smooth_penguin (~smoove@59.95.36.64) |
| 13:07.46 | _omer | please check it out ... http://www.pastebin.org/88770 |
| 13:07.56 | c0rnoTa | one moment |
| 13:08.05 | _omer | ok |
| 13:08.38 | smooth_penguin | leifmadsen, thanks .. had to run there |
| 13:08.39 | smooth_penguin | :\ |
| 13:08.48 | leifmadsen | np |
| 13:09.22 | *** join/#asterisk tw56 (~tw56@host86-179-100-216.range86-179.btcentralplus.com) |
| 13:10.10 | tw56 | Hi People. I remember a while back but i can't remember now! How do i disable the asterisk ringing tones so that the standard ring is all that's heard for the user? |
| 13:12.22 | Tim_Toady | you mean the r option in app dial |
| 13:12.50 | binbash_ | Hmm i have an issue with caller id :-) |
| 13:12.55 | binbash_ | hope anybody can help me out |
| 13:13.06 | binbash_ | i have this sutation -> <asterisk box> <asterisk sip gateway> |
| 13:13.16 | binbash_ | they are directly connected to eachother with a sip trunk |
| 13:13.22 | binbash_ | on the gateway |
| 13:13.27 | binbash_ | i have the following in my dialplan -> |
| 13:13.38 | binbash_ | exten => _+31[1-9].,1,Set(CALLERID(all)=<0851119111>) |
| 13:13.38 | binbash_ | exten => _+31[1-9].,n,Dial(SIP/Intorrent-Sip/${EXTEN}) |
| 13:13.48 | *** join/#asterisk sparch (~daniel@pdpc/supporter/active/sparch) |
| 13:13.51 | *** part/#asterisk sparch (~daniel@pdpc/supporter/active/sparch) |
| 13:13.57 | binbash_ | (the intorrent sip is a sip trunk outside) |
| 13:14.12 | binbash_ | but the caller id is still "unknown" on mobile phones/outside phones i call |
| 13:14.14 | *** join/#asterisk etfonhomey (~etfonhome@74-143-192-74.static.insightbb.com) |
| 13:14.16 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 13:14.16 | binbash_ | any idea's :-)? |
| 13:14.42 | c0rnoTa | _omer: have no idea, either you use AMI with UPDATECONFIG when adding new context in database or store full extensions.conf in DB and do not use file. |
| 13:14.43 | binbash_ | (sorry for the wall of text/spam but wanted to tell the whole story :P) |
| 13:15.29 | _omer | you mean realtime static config or something like that? |
| 13:15.43 | *** join/#asterisk kamh (~kamh@xdsl-1814.wroclaw.dialog.net.pl) |
| 13:15.57 | kamh | hi |
| 13:16.11 | c0rnoTa | _omer: yes. realtime static. but i don't like this way |
| 13:16.23 | _omer | I tried UPDATECONFIG through WEB .. asterisk 1.4 ... getting Response SUCCESS ... but extensions.conf is not updated |
| 13:16.28 | _omer | I dont like either.. |
| 13:16.46 | c0rnoTa | _omer: IMHO, the best way is to use UpdateConfig |
| 13:16.54 | _omer | IMHO ? |
| 13:17.19 | binbash_ | nobody :)? |
| 13:17.59 | _omer | binbash_ : may be your trunk doesn't support CLI .. |
| 13:18.03 | c0rnoTa | _omer: In My Humble Opinion |
| 13:18.15 | _omer | hehe |
| 13:18.52 | _omer | I thought it's some kind of way to interact with AMI ;) |
| 13:19.25 | _omer | I have given all rights to Asterisk Manager......but extensions.conf is still not getting updated....getting response: SUCCESS .. |
| 13:19.31 | benngard | Set(CALLERID(all)=<0851119111>) <- shouldnt it be name <number> and u will always send out the same number with that line |
| 13:19.34 | _omer | any good way to troubleshoot it out? |
| 13:19.56 | ManxPower-work | it should also not start with a 0 |
| 13:20.04 | binbash_ | benngard i know, that's what i want :D |
| 13:20.27 | binbash_ | ManxPower-work why not :)? |
| 13:20.34 | c0rnoTa | _omer: i can give you my AMI commands |
| 13:20.41 | *** join/#asterisk mbranca (~matteo@host139-217-static.224-95-b.business.telecomitalia.it) |
| 13:20.47 | c0rnoTa | _omer: that works fine |
| 13:20.59 | _omer | I have them as well...but let me check yours too |
| 13:21.11 | ManxPower-work | binbash_: because callerid information does not start with 0 |
| 13:21.41 | creativx | 007 |
| 13:21.43 | binbash_ | ManxPower-work oke thnx didn't know that |
| 13:21.49 | binbash_ | never worked with caller id before :) |
| 13:21.56 | binbash_ | so i should just make it 85.... then? |
| 13:22.14 | ManxPower-work | callerid is country code + city/area code + number |
| 13:22.27 | binbash_ | oke so +3185xx |
| 13:22.29 | binbash_ | then :) |
| 13:22.33 | ManxPower-work | no leading + |
| 13:22.48 | ManxPower-work | + means "dial whatever you dial for an international call" |
| 13:22.49 | c0rnoTa | _omer: http://pastebin.ru/310394 |
| 13:22.55 | _omer | c0rnota: here is what I have .. AMI Commands..... http://www.pastebin.org/88836 |
| 13:22.55 | *** part/#asterisk tw56 (~tw56@host86-179-100-216.range86-179.btcentralplus.com) |
| 13:22.58 | _omer | ok ... let me check yours |
| 13:23.03 | [TK]D-Fender | ManxPower-work: .. he's outta NANPA.. you sure you know the rules where he lives? |
| 13:23.05 | binbash_ | oke thanks for the help i'll try it straight away |
| 13:23.10 | ManxPower-work | It is also not part of callerid |
| 13:23.21 | creativx | + 00 |
| 13:23.27 | binbash_ | so this is better -> |
| 13:23.27 | ManxPower-work | [TK]D-Fender: He's not outside the ITU |
| 13:23.28 | binbash_ | exten => _+31[1-9].,1,Set(CALLERID(num)=31851119111) |
| 13:23.28 | binbash_ | exten => _+31[1-9].,n,Dial(SIP/Intorrent-Sip/${EXTEN}) |
| 13:24.12 | benngard | binbash_: i had about the same problem, had to do alot of "exten => s,n,ExecIf($["${CALLERID(num)}"="0317998985"]?Set(CALLERID(all)=Cecilia Benngard <317998985>))" before the actual dial command |
| 13:24.15 | ManxPower-work | binbash_: does your phone have a + key? |
| 13:24.31 | binbash_ | no |
| 13:24.38 | creativx | + is shorthand for 00 |
| 13:24.45 | binbash_ | yeah.. |
| 13:24.49 | ManxPower-work | creative not in Asterisk it isn't. |
| 13:24.53 | c0rnoTa | _omer: do not forget to add new cat |
| 13:25.03 | creativx | ManxPower-work: indeed it is not |
| 13:25.03 | binbash_ | some softphones can dial + here inside tho :) |
| 13:25.09 | creativx | thats why you must use 00 |
| 13:25.10 | c0rnoTa | _omer: only once |
| 13:25.13 | [TK]D-Fender | c0rnoTa: That doesn't look like legit PHP to me... |
| 13:25.19 | ManxPower-work | Actually + means "dial your international access code, which will vary from country to country" |
| 13:25.33 | _omer | c0rnoTa: ok....thanks ... let me try this |
| 13:25.35 | creativx | i thought it was 00 everywhere |
| 13:25.42 | creativx | but who needs standardization |
| 13:25.47 | ManxPower-work | creativx: in the united states it is 011 |
| 13:25.56 | binbash_ | i still get "unknown" on my mobile phone tho |
| 13:25.57 | ManxPower-work | Actually in all of NANP |
| 13:25.58 | *** join/#asterisk oej (~olle@static-213-115-251-100.sme.bredbandsbolaget.se) |
| 13:26.05 | creativx | ManxPower-work: cool, didnt know |
| 13:26.14 | c0rnoTa | [TK]D-Fender: should be legit? |
| 13:26.15 | binbash_ | while asterisk shows Set("SIP/asterisk-00000623", "CALLERID(num)=31851119111") |
| 13:26.37 | ManxPower-work | binbash_: maybe your carrier doesn't allow you to set the callerid? |
| 13:26.46 | binbash_ | just checked, it's all open :) |
| 13:26.59 | binbash_ | (i was thinking the same) |
| 13:27.00 | [TK]D-Fender | c0rnoTa: Looks like extra illegal chars to me. No AMI login. No sense of context. |
| 13:27.14 | binbash_ | but when asterisk shows that he's setting the caller id, it's working ok right? |
| 13:27.23 | creativx | binbash_: ITSP can decide to override your cid |
| 13:27.37 | creativx | our previous one only let us set cid to our own number ranges |
| 13:27.44 | binbash_ | maby a stupid question, but what's a ITSP? |
| 13:27.54 | creativx | ~itsp |
| 13:27.55 | infobot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 13:28.04 | binbash_ | ah what i tought |
| 13:28.09 | binbash_ | no they forward everything i send them |
| 13:28.13 | binbash_ | i checked that |
| 13:28.34 | binbash_ | aslong as it's a number in my range, they forward it |
| 13:29.08 | c0rnoTa | _omer: [TK]D-Fender: http://www.pastebin.org/88839 my be this can be readed better? |
| 13:29.56 | c0rnoTa | [TK]D-Fender: my be illegal cahrs was becouse it's RU domain of pastebin? |
| 13:29.58 | [TK]D-Fender | c0rnoTa: whats the the comment lines in there? |
| 13:30.16 | [TK]D-Fender | c0rnoTa: also IIRC you have to have 2 blank lines to end an AMI command |
| 13:30.19 | c0rnoTa | [TK]D-Fender: only for understanding and separete |
| 13:30.32 | binbash_ | maby if i set the caller id, after the actual dial |
| 13:30.35 | binbash_ | but that sounds strange |
| 13:30.42 | c0rnoTa | [TK]D-Fender: it's only for my usage :) |
| 13:30.48 | krion | hi |
| 13:30.57 | c0rnoTa | [TK]D-Fender: this commands works fine in my web |
| 13:31.07 | [TK]D-Fender | c0rnoTa: ar those lines actually part of the transmission? |
| 13:31.07 | krion | anyone using wireshark with g729 flow ? |
| 13:31.24 | krion | i like to have in and out output for a call to echo |
| 13:31.32 | c0rnoTa | [TK]D-Fender: comments - no, 2 blank lines - yes |
| 13:31.39 | [TK]D-Fender | [08:30]<binbash_>maby if i set the caller id, after the actual dial <- that's like offering food to someone who jsut starved to death |
| 13:31.49 | binbash_ | haha |
| 13:31.54 | krion | monitoring quality is not that easy... |
| 13:31.54 | binbash_ | that's what i tought |
| 13:32.12 | c0rnoTa | [TK]D-Fender: my web face for modifing extensions_outgoing.conf works fine. I'm just trying to help _omer |
| 13:32.29 | *** join/#asterisk albertoandrade (~albertoan@189.114.181.254.dynamic.adsl.gvt.net.br) |
| 13:33.06 | [TK]D-Fender | c0rnoTa: guess ti would help if we saw HIS code, and not yours then... |
| 13:33.23 | *** join/#asterisk e-jones (~jkastner@nat/redhat/x-dhresyfezaepqxbj) |
| 13:33.51 | krion | g729 doesn't help at all in that task :D |
| 13:42.21 | c0rnoTa | [TK]D-Fender: he said (16:24:09) _omer: ok ... let me check yours :) that's why i give it to him :) anyway, he gave his AMI commands to us http://www.pastebin.org/88836 |
| 13:48.37 | *** join/#asterisk hluesea (~hulusikah@88.247.127.66) |
| 13:49.49 | Katty | hi |
| 13:49.55 | Katty | it snowed |
| 13:50.00 | creativx | snow is good |
| 13:50.13 | Katty | all the soggy patches iced over |
| 13:50.53 | creativx | man |
| 13:50.57 | creativx | i am about to strangle mysql |
| 13:51.18 | *** join/#asterisk xEBIx (~ebi@office.corpex-net.de) |
| 13:51.41 | *** join/#asterisk TheDavidFactor (~chatzilla@nc-71-0-16-133.dhcp.embarqhsd.net) |
| 13:52.16 | *** join/#asterisk Warp4 (~Robert@firewall-a.buf.ny.i-evolve.net) |
| 13:52.57 | smooth_penguin | snow! |
| 13:53.10 | beek | sucks! |
| 13:53.20 | Katty | fires up both crittercams |
| 13:53.53 | krion | what's the correct syntax of the show channels ? |
| 13:53.57 | krion | so bad at this;.. |
| 13:54.03 | xEBIx | I've got some problems getting a "hello world" config to run on my asterisk. I am using an Online book http://www.das-asterisk-buch.de/2.1/hello-world-im-cli.html first of all they want to execute console dial 1001 but I get no such command, perhaps someone can figure that out with me |
| 13:54.50 | beek | krion: tab completion works well. If you're using 1.6.x, it's "core show channels" |
| 13:55.26 | beek | Same for 1.4 |
| 13:57.03 | Katty | considers breakfast possibilities |
| 13:57.29 | [TK]D-Fender | xEBIx: Asterisk CLI command syntax changes between versions. |
| 13:58.41 | *** part/#asterisk icyValk77 (~icyValk77@gateway.ash.thebunker.net) |
| 14:00.43 | [TK]D-Fender | xEBIx: I would simply not test it in that way. Go set up a soft-phone properly to test your dialplan |
| 14:01.02 | beek | Morning [TK]D-Fender , Katty |
| 14:01.07 | [TK]D-Fender | beek: 'lo |
| 14:01.15 | krion | oh thanks very much beek |
| 14:02.43 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 14:03.18 | *** join/#asterisk RobH_ (~robh@cpe-173-169-30-118.tampabay.res.rr.com) |
| 14:04.00 | c0rnoTa | it's snowing in Moscow too. |
| 14:04.06 | krion | i try to record in and out flow from my sip_phone to demo-echotest for quality audio testing |
| 14:04.18 | krion | it's not like a walk in a park |
| 14:06.33 | krion | in my /var/spool/asterisk/monitor i see bunch of file *-in.wav and *-out.wav but i don't know how they end up there :D |
| 14:09.49 | [TK]D-Fender | krion: My guess would be "calls being recorded" |
| 14:09.57 | krion | :-) |
| 14:10.40 | krion | i try mixmonitor but it mix in and out into raw file |
| 14:12.04 | [TK]D-Fender | krion: And we don't see what you're doing so can't tell you what's wrong with it |
| 14:13.12 | krion | i just did a mixmonitor <channel> |
| 14:15.43 | *** join/#asterisk andres833 (~andres833@190.144.75.22) |
| 14:16.12 | *** join/#asterisk jaytee (~jforde@unaffiliated/jaytee) |
| 14:16.33 | [TK]D-Fender | krion: that isn't a precise call. |
| 14:19.16 | *** join/#asterisk ZaVoid (~zavoid@75-147-121-177-Philadelphia.hfc.comcastbusiness.net) |
| 14:19.19 | ZaVoid | hi all |
| 14:19.59 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 14:21.38 | p3nguin | beek: If he's using 1.4, it's "core show channels" too. |
| 14:21.39 | ZaVoid | anyone have any luck with SetCallerPres and getting carriers to display Blocked from using it.. ie SetCallerPres(prohib) seems to be making all my carriers show caller ID as "unknown" and not Blocked |
| 14:21.54 | jaytee | morning beek [TK]D-Fender |
| 14:22.10 | p3nguin | beek: But nevermind, you already followed up with that. :) |
| 14:25.05 | ZaVoid | ahh prohib_passed_screen works for blocked.. |
| 14:25.33 | beek | morning jaytee |
| 14:25.37 | beek | p3nguin: :D |
| 14:25.52 | Katty | mmm |
| 14:25.58 | Katty | i have fried potatos cooking |
| 14:26.12 | Katty | with yummy caramelized onions, mushrooms, and celery |
| 14:26.15 | jaytee | hash browns? |
| 14:26.20 | p3nguin | krion: If your exten runs MixMonitor(somefile.wav), then your call would be recorded to one file in wav format. |
| 14:26.34 | Katty | jaytee: maybe? i didn't shred the potatos |
| 14:26.40 | Katty | jaytee: i just chopped them up |
| 14:26.45 | jaytee | close enough |
| 14:26.49 | Katty | nods |
| 14:26.51 | jaytee | hugs Katty |
| 14:26.55 | Katty | hugs jaytee |
| 14:27.02 | jaytee | get any snow |
| 14:27.05 | Katty | yep |
| 14:27.06 | jaytee | ? |
| 14:27.14 | Katty | the roads iced over from all the sogginess |
| 14:27.24 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 14:27.27 | jaytee | we got about 3-4 inches so far. powdery, not heavy like last time |
| 14:30.36 | *** part/#asterisk pentanol (~pentanol@77-35-1-202.pppoe.primorye.net.ru) |
| 14:33.03 | Katty | thinks potatos need to cook faster |
| 14:34.58 | beek | hands Katty a propane torch. |
| 14:35.04 | Katty | woooooooot |
| 14:35.30 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 14:35.51 | krion | p3nguin: oh thanks ! any idea how i can have separate in and out flow in separate wav file ? |
| 14:36.05 | Katty | i'm making a list for the fridge that says how many calories per oz on several of the common food items i cook with. |
| 14:36.27 | Katty | potatos are 22 an oz, onions are 11 per oz, celery is 4 per oz, etc. |
| 14:38.03 | p3nguin | krion: Yes. Use Monitor() and don't mix the recordings after they are finished. |
| 14:39.47 | krion | looks like i don't have monitor in my cli (using asterisk 1.2) |
| 14:40.07 | xEBIx | [TK]D-Fender, but generaly "console dial" should work? I am using 1.6.2.2 so should be covered by the book. Or did it change from 1.6.1 to 1.6.2 so much? |
| 14:40.25 | [TK]D-Fender | xEBIx: what "book"? |
| 14:40.37 | p3nguin | krion: "core show application Monitor" ? |
| 14:41.12 | xEBIx | the online book i mentioned at first http://www.das-asterisk-buch.de/2.1/hello-world-im-cli.html |
| 14:41.20 | krion | yes i got something when i show application Monitor... |
| 14:41.33 | *** join/#asterisk cjk (~cjk@vodsl-10270.vo.lu) |
| 14:41.44 | [TK]D-Fender | xEBIx: debian:/etc/asterisk# asterisk -c Asterisk 1.4.21, Copyright (C) 1999 - 2008 Digium, Inc. and others. |
| 14:41.52 | [TK]D-Fender | xEBIx: Guess you didn't even bother reading that they are using 1.4 |
| 14:42.23 | cjk | hi, i have a little background noise (like an old radio) and some beeps on my PRI, do you have any idea? |
| 14:42.24 | xEBIx | [TK]D-Fender, but look at the Title |
| 14:42.27 | [TK]D-Fender | xEBIx: thats 3 releases ago |
| 14:42.44 | [TK]D-Fender | xEBIx: No, YOU loko in the MIDDLE OF ThAT PAGE |
| 14:42.45 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 14:43.02 | [TK]D-Fender | xEBIx: It clearly shows them conencting to a 1.4.21 release |
| 14:43.09 | Katty | growls grumpily at potatos |
| 14:43.20 | Katty | but the good news is my office is closed |
| 14:43.22 | Katty | cheers |
| 14:43.26 | xEBIx | [TK]D-Fender so did change that way in these relases you mean |
| 14:43.31 | [TK]D-Fender | xEBIx: try "help" at * cli. |
| 14:43.51 | xEBIx | [TK]D-Fender, did it or not? |
| 14:44.00 | [TK]D-Fender | xEBIx: 1.6 is not a specific branch. 1.0, 1.2, 1.4, 1.6.0, 1.6.1, 1.6.2 are all branches |
| 14:44.17 | *** join/#asterisk JT (~j@unaffiliated/jt) |
| 14:44.20 | [TK]D-Fender | [09:43]<[TK]D-Fender>xEBIx: try "help" at * cli. |
| 14:45.00 | ManxPower-work | All major changes should be documented in UPGRADE*.txt |
| 14:45.09 | [TK]D-Fender | xEBIx: Anyway jsut set up a softphone already and use that for testing |
| 14:45.23 | xEBIx | [TK]D-Fender, i know but i also know that the command is also unknown if the config is wrong, so it would be interresting if it realy has changed that way |
| 14:45.43 | ManxPower-work | cjk: You have some other issue. |
| 14:45.45 | Katty | yay breakfast!!! |
| 14:46.16 | p3nguin | I don't know if I can get that excited over frosted shredded wheat. |
| 14:46.18 | ManxPower-work | xEBIx: What is your specific question? |
| 14:46.39 | Katty | p3nguin: if you were having fried tators with caramelized onions, mushrooms, and celery you would. |
| 14:46.42 | cjk | ManxPower-work, not really, according to cat /proc/dahdi/1 the interface lost 7 interrupts but i guess thats not really important because the number stays at that value since days |
| 14:47.00 | ManxPower-work | cjk: that would be under "other issue" |
| 14:47.09 | p3nguin | katty: If someone served that to me, they would be taking it back and bringing me shredded wheat. |
| 14:47.10 | ManxPower-work | PRIs are digital so it is impossible to have static on them. |
| 14:47.12 | xEBIx | Does asterisk change the "console dial" thing from 1.4 to 1.6.2 or is just my config wrong? |
| 14:47.22 | Katty | p3nguin: why? it's very healthy |
| 14:47.34 | p3nguin | katty: As I had mentioned before, I don't do fungus. |
| 14:47.39 | ManxPower-work | xEBIx: "console dial" will not be available if you don't have the alsa-devel libraries and headers installed. |
| 14:47.42 | Katty | p3nguin: ah, that's right |
| 14:47.44 | cjk | ManxPower-work, yep I agree on that, however the voice is not clear and i have no idea |
| 14:47.51 | Katty | p3nguin: well they can just leave the shroomies out ;) |
| 14:48.32 | [TK]D-Fender | xEBIx: Do you have chan_oss or chan_alsa configured and loaded? |
| 14:48.33 | p3nguin | katty: I think I would rather have regular hash browns. |
| 14:48.33 | Katty | p3nguin: this is a very rare treat--usually don't have time to make breakfast in the morning! |
| 14:48.39 | ManxPower-work | xEBIx: Do you have a microphone and speaker connected to your Asterisk box? |
| 14:49.02 | xEBIx | module.conf sais that chan_alsa is loaded |
| 14:49.15 | Katty | p3nguin: making breakfast is usally throwing an english muffin with cheese on it in the oven while i get ready for work, and nukerwaving an egg or ham to put on it... |
| 14:49.26 | xEBIx | yes there is a headphone connected |
| 14:49.31 | krion | p3nguin: monitor is show as registered but can't use it in the asterisk> cli... |
| 14:49.34 | ManxPower-work | xEBIx: try "module load chan_alsa.so" in the CLI. |
| 14:49.36 | xEBIx | with a mic build in |
| 14:49.49 | p3nguin | krion: Why would you be able to use it on the CLI, when it is a dialplan application? |
| 14:49.53 | [TK]D-Fender | xEBIx: No, that may say to TRY to load it. Go prove that its actually loaded. And go look at the docs you were referre to and look at the cLI help |
| 14:49.56 | ManxPower-work | p3nguin: because you can. |
| 14:50.04 | p3nguin | Apparently he can't. |
| 14:50.18 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 14:50.18 | krion | p3nguin: ok, as i used to use mixmonitor i was thinking that it as the same behaviour |
| 14:50.20 | [TK]D-Fender | p3nguin: Dialing from cli has always existed. |
| 14:50.20 | ManxPower-work | p3nguin: that's just because he doesn't have the right libraries installed. |
| 14:50.39 | ManxPower-work | It's stupid to dial from the CLI, but you can do it. |
| 14:50.40 | p3nguin | [tk]d-fender: Really? I thought that was brand new today. |
| 14:50.51 | Katty | Dear Universe, thank you for potatos. Love, Katty |
| 14:50.53 | ManxPower-work | p3nguin: existed at least since Asterisk 0.65 |
| 14:50.57 | [TK]D-Fender | wants a Born-Again CLI |
| 14:51.33 | p3nguin | manxpower-work: When did we jump from using Monitor() to dialing from CLI? |
| 14:51.48 | ManxPower-work | p3nguin: I said nothing about Monitor. |
| 14:52.00 | p3nguin | manxpower-work: You should have, because that's what krion and I were talking about. |
| 14:52.08 | xEBIx | ahh there we go - thats strange alsa_card_init: snd_pcm_open failed: No such file or directory |
| 14:52.23 | xEBIx | No sound card detected -- console channel will be unavailable |
| 14:52.26 | ManxPower-work | xEBIx: there you go. Can you play a sound file OUTSIDE of Asterisk |
| 14:52.55 | xEBIx | yes indeed |
| 14:53.22 | ManxPower-work | xEBIx: Did you have the alsa development libraries and headers installed when you installed Asterisk |
| 14:53.31 | [TK]D-Fender | xEBIx: You do not NEED "console dial" |
| 14:53.36 | krion | p3nguin: i'll try to add and exten then, thanks |
| 14:53.39 | *** join/#asterisk jo8330 (~d04149c9@gateway/web/freenode/x-orbvtnbbbucbazth) |
| 14:53.46 | jo8330 | Morning folks. |
| 14:53.48 | [TK]D-Fender | xEBIx: Now stop wasting time and test with another soft-phone on a system that DOES have a working sound setup |
| 14:54.08 | ManxPower-work | *shrug* I was going to help him waste his time. |
| 14:54.16 | jo8330 | Quick question, is it normal for span to restart by itself? I often see " -- B-channel 0/1 successfully restarted on span 1" for each channel every so often |
| 14:54.26 | jo8330 | i'm wondering if that will cause issues should there be active calls. |
| 14:54.26 | xEBIx | I am using Teamspeak on that system so it has a working sound setup... |
| 14:54.35 | ManxPower-work | jo8330: yes. Did you not read chan_dahdi.conf.sample? |
| 14:54.58 | p3nguin | krion: Add a priority right before the Dial() where you run Monitor(wav:filebase). |
| 14:55.01 | xEBIx | and meanwhile iam using Gentoo not Debian... |
| 14:56.07 | krion | i'm using 'n' instead of priority, and it's right before Echo() |
| 14:56.16 | ManxPower-work | p3nguin: be careful. The format of Record/Monitor/MixMonitor changed between Asterisk versions. You just gave him and example from 1.4.x |
| 14:56.22 | p3nguin | krion: n is a priority in relative terms. |
| 14:56.54 | jo8330 | ManxPower-work: I have several times but didn't notice this. it's related to resetinterval ? |
| 14:57.00 | p3nguin | krion: As long as you have a 1 to start with, n is the next priority. |
| 14:57.06 | ManxPower-work | jo8330: Correct. |
| 14:57.14 | krion | yes that part i get ;) |
| 14:57.34 | p3nguin | krion: And double check the usage of Monitor() since I never used 1.2 before. |
| 14:57.51 | krion | was pretty simple and do what i want :) |
| 14:57.59 | krion | Monitor(wav,filename) |
| 14:58.16 | jo8330 | ManxPower-work: thanks! Is this feature just a safeguard should channel get into crappy state? |
| 14:58.19 | krion | i will now try to pass it to sevana software quality analysis |
| 14:58.46 | krion | p3nguin: my goal is to have automatic quality test in order to monitor audio quality call |
| 14:59.07 | jo8330 | i'm tempted to set it to 'never' |
| 14:59.18 | p3nguin | krion: Will you check each side of the call, or only one side? |
| 14:59.20 | ManxPower-work | krion: you will find that Monitor/MixMonitor will not give you a true idea of the call quality. |
| 14:59.50 | ManxPower-work | It will not expose echo and will not provide information about the far end of the call. |
| 14:59.57 | krion | i'm calling echo in order to have the same file and compare quality |
| 15:00.17 | krion | so i try to monitor both side of the call |
| 15:00.21 | ManxPower-work | You would want to use the QoS variables to determine more information about the RTP of the call. |
| 15:00.25 | krion | ManxPower-work: how would you do it |
| 15:00.26 | leifmadsen | krion: any info on the audio quality test tool? |
| 15:00.41 | ManxPower-work | krion: What you want to do is extremely difficult. |
| 15:00.54 | krion | leifmadsen: of course if you want, but i'm not related to it or try to do publicity about it |
| 15:01.05 | *** join/#asterisk robl^laptop (~robl@m4b5336d0.tmodns.net) |
| 15:01.14 | leifmadsen | krion: nevermind, I found the site |
| 15:01.20 | krion | ManxPower-work: and my customer are very difficult to handle when they call the support for bad quality |
| 15:01.32 | krion | leifmadsen: ok, tell me what you think about it |
| 15:01.44 | leifmadsen | I have no thoughts about it -- I haven't done any research on said subject |
| 15:01.50 | Katty | hmm. now what to do...hmm hmm. |
| 15:01.52 | Katty | nap perhaps |
| 15:01.57 | leifmadsen | Katty: video game! |
| 15:01.58 | krion | it's comme from fi so has the linux kernel come so it should be good software (even it's not FOSS° |
| 15:02.02 | Katty | leifmadsen: bleh |
| 15:02.12 | Katty | leifmadsen: nap sounds better |
| 15:02.23 | ManxPower-work | krion: the general solution to that is to not run your calls over a non-QoS network. If you run calls over the internet you will have occasional call quality issues and you will be able to do nothing about it. |
| 15:02.49 | krion | ManxPower-work: i have qos on the client side router, but still got quality issue |
| 15:03.03 | ManxPower-work | krion: QoS means nothing if it is not end-to-end. |
| 15:03.11 | krion | i'm using cheap thomson st20{22,30} so that doesn't help |
| 15:03.24 | krion | ManxPower-work: what do you mean ? |
| 15:03.47 | ManxPower-work | I mean QoS does not really work unless all routers between the two end points are also configured to support QoS |
| 15:03.47 | krion | i can't have QoS on the "collect" side (where all the atm links goes) |
| 15:03.58 | krion | oh you think ? |
| 15:04.07 | leifmadsen | which over the internet, you don't have the control of the path or the routers |
| 15:04.19 | leifmadsen | they may or may not have it configured, and may or may not respect your "request" |
| 15:04.30 | leifmadsen | only way to solve that is MPLS networks end-to-end |
| 15:04.40 | krion | leifmadsen: i'm directly connected to my peer |
| 15:04.41 | ManxPower-work | or, like BellSouth they strip the QoS info off the packets when the packet enters their network. |
| 15:04.43 | angryuser | <PROTECTED> |
| 15:04.47 | leifmadsen | krion: over the internet... |
| 15:04.53 | krion | using a trunk sip |
| 15:05.00 | leifmadsen | krion: that's not directly connected |
| 15:05.09 | ManxPower-work | krion: you have a direct line between the two end points that does not go over the interhet? |
| 15:05.12 | leifmadsen | you -> internet --> routers you don't control --> other end |
| 15:05.16 | krion | ok i got it, you mean directly connected like a private bgp peer |
| 15:05.29 | ManxPower-work | no, I mean directly connected, like directly connected. |
| 15:05.38 | leifmadsen | ummm... BGP peer is not a good example :) |
| 15:05.42 | leifmadsen | like an MPLS network :) |
| 15:05.49 | ManxPower-work | you <-> other end <-- that is a direct connection |
| 15:05.59 | leifmadsen | or something where you control all points between the two connections |
| 15:06.00 | p3nguin | Directly connected, like a serial crossover cable between your two routers. |
| 15:06.03 | ManxPower-work | you <-> Internet <-> other end <-- NOT direct connection. |
| 15:06.04 | leifmadsen | yes :) |
| 15:06.06 | Katty | nite, i'm goin back to bed |
| 15:06.11 | leifmadsen | Katty: night! |
| 15:06.17 | krion | ok ok got it |
| 15:06.25 | ManxPower-work | hugs his 1ms jitter on his network |
| 15:06.38 | krion | anyway i'm probably sure i don't have qos in every routeur inside my backbone so :) |
| 15:06.58 | ManxPower-work | krion: correct. So QoS will have little impact on voice quality. |
| 15:07.02 | c0rnoTa | why night? |
| 15:07.03 | p3nguin | That was the point manxpower-work and leifmadsen were trying to make. |
| 15:07.10 | ManxPower-work | If QoS was easy everyone would be doing it. |
| 15:07.15 | p3nguin | c0rnota: time for sleep! |
| 15:07.40 | leifmadsen | and if everyone was doing it, then everyone would take highest-priority, which means QoS would still not work |
| 15:07.58 | *** join/#asterisk jetlag (jetlag@pool-70-18-186-205.pskn.east.verizon.net) |
| 15:09.11 | c0rnoTa | p3nguin: O_o 1 hour ago "Katty considers breakfast possibilities" |
| 15:09.17 | krion | i've a client with 100M fiber, my backbone is in GB, so i should not have quality issue right ? |
| 15:09.27 | krion | related to networking (so QoS) i mean |
| 15:09.30 | leifmadsen | moves on from this dead end conversation |
| 15:09.40 | leifmadsen | I'm stuck in a loop! |
| 15:09.40 | ManxPower-work | You might be able to have some "Quality Improvement" when uploading large files if you are very clever with your router. |
| 15:10.02 | ManxPower-work | krion: what is the latency and jitter between the two end points? |
| 15:10.10 | p3nguin | implements STP on leifmadsen |
| 15:10.12 | c0rnoTa | p3nguin: night coming only in Russia. It's 18 o'clock now. You might have 12 or 13 o'clock, i think. |
| 15:10.17 | leifmadsen | p3nguin: phew! |
| 15:10.38 | p3nguin | c0rnota: 09:10 here. |
| 15:10.54 | c0rnoTa | p3nguin: :) too early for night |
| 15:11.01 | [TK]D-Fender | p3nguin: Stone temple Pilots? |
| 15:11.06 | [TK]D-Fender | rocks out |
| 15:11.18 | *** join/#asterisk andres833 (~andres833@190.144.75.22) |
| 15:11.19 | p3nguin | c0rnota: katty's office is closed for the day, so she was going back to bed. |
| 15:11.30 | leifmadsen | [TK]D-Fender: spanning tree protocol :) |
| 15:11.39 | p3nguin | haha |
| 15:12.15 | plundra | What's wrong when *, # etc. isn't getting through to ChanSpy()? For switching channel. |
| 15:12.32 | krion | ManxPower-work: always under 60ms |
| 15:12.35 | leifmadsen | typically a dtmf problem... |
| 15:12.46 | leifmadsen | or perhaps some flag missing to enable the feature you're trying to use |
| 15:12.48 | krion | the jitter i don't know |
| 15:13.08 | *** join/#asterisk UQlev (~yuriy@nb11-125.static.cytanet.com.cy) |
| 15:13.22 | krion | i try to set up ntop in order to have more information, but looks like full voip support is only available with nprobe who's gpl but not freely available... |
| 15:13.23 | ManxPower-work | krion: you have 60 ms of jitter???? |
| 15:13.33 | krion | ManxPower-work: ping latency |
| 15:13.40 | plundra | leifmadsen: Yeah, that's what I think, but DTMF is fine when Authenticate() is used or selecting a extension. |
| 15:14.06 | ManxPower-work | krion: There are three things that impact quality of a call. Bandwidth, packet loss, and jitter. If you have problems with any of those you will suffer call quality issues. |
| 15:14.24 | plundra | And there is no flag to enable switching in ChanSpy() |
| 15:14.43 | krion | ManxPower-work: how do you get the jitter ? |
| 15:15.09 | krion | ifmy question is to dumb you can tell me RTFM or STFW |
| 15:15.20 | ManxPower-work | krion: jitter is the how much packet latency jitters. It will vary over time as the usage of the connection increases and decreases. |
| 15:15.33 | *** join/#asterisk e4voip (~chatzilla@99-27-170-70.lightspeed.cicril.sbcglobal.net) |
| 15:15.34 | ManxPower-work | krion: Real QoS is one of the hardest thing to do in networking |
| 15:15.38 | *** join/#asterisk Akiraa (~Akiraaaa@79.112.13.16) |
| 15:15.39 | *** join/#asterisk Akiraaa (~Akiraaaa@79.112.13.16) |
| 15:15.52 | p3nguin | one akiraa isn't enough? |
| 15:17.05 | p3nguin | Guess someone else thought so. |
| 15:17.25 | krion | <PROTECTED> |
| 15:17.31 | Akiraa | not sure what happened there ^^ I ghosted that nick |
| 15:17.37 | *** join/#asterisk KingDavidNYC (~Chris1232@69.193.218.18) |
| 15:17.41 | *** join/#asterisk Akiraaa (~Akiraaaa@79.112.13.16) |
| 15:17.43 | krion | that's my MOS value when i compare in and out file recorded using Monitor |
| 15:17.47 | KingDavidNYC | Hello Everybody |
| 15:17.49 | *** join/#asterisk hfb (~hfb@pool-96-247-114-78.lsanca.dsl-w.verizon.net) |
| 15:17.52 | p3nguin | akiraaa: You're back! |
| 15:18.04 | Akiraa | wtf |
| 15:18.18 | dlynes_laptop | Akiraa, it's all coming from your ip |
| 15:18.31 | *** join/#asterisk smooth_penguin (~smoove@59.95.36.64) |
| 15:19.05 | KingDavidNYC | can anyone help me figure out why my meetme conference does not work after I recompiled to 1.4 from 1.2? |
| 15:19.11 | *** join/#asterisk Akiraaa (~Akiraaaa@79.112.13.16) |
| 15:21.49 | krion | looks like a correct MOS value for G729a is 3,7 |
| 15:22.08 | krion | so i'm way below the expected value |
| 15:22.19 | *** join/#asterisk klmarkea (~c233da02@gateway/web/freenode/x-xgggzapmfsgihizw) |
| 15:22.48 | *** join/#asterisk basilgohar (~basilgoha@cpe-75-185-126-63.insight.res.rr.com) |
| 15:22.54 | klmarkea | Hello everyone |
| 15:23.29 | klmarkea | I'm encountering problmes during compilation of asterisk-addons-1.6.2.0 under ubuntu server |
| 15:23.44 | KingDavidNYC | My issue has something to do with dahdi supoprt, but I dont see how to disable it when recompile |
| 15:23.50 | basilgohar | Where can I find a good tutorial on setting-up Asterisk for SIP videoconferencing? |
| 15:24.43 | klmarkea | It seems gcc can't find /usr/include/asterisk/buildopts.h |
| 15:24.54 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:25.17 | klmarkea | I tried to configure with the option --with-asterisk=path/to/asterisk/sources |
| 15:25.36 | klmarkea | But it didn't work. Any Idea ? |
| 15:27.46 | *** part/#asterisk benngard (~benngard@213.88.138.230) |
| 15:27.52 | klmarkea | Another point that I don't find normal : after making changes in the menuselect procédure, lots of messages appear : /bin/grep: /usr/include/asterisk/buildopts.h: no such file or directory |
| 15:28.16 | krion | who told me that with echo i would not have a correct answer ? :) |
| 15:28.27 | [TK]D-Fender | KingDavidNYC: Provide actual precise version numbers please |
| 15:29.28 | KingDavidNYC | sure ma, thank you..... 1.4.23 |
| 15:29.36 | KingDavidNYC | man |
| 15:29.37 | [TK]D-Fender | basilgohar: * doesn't do video "conferencing". It can however allow video to pass from 1 client to another in a direct Dial |
| 15:29.41 | [TK]D-Fender | kiBOTH |
| 15:29.47 | [TK]D-Fender | KingDavidNYC: BOTH |
| 15:30.50 | KingDavidNYC | soryy, didn't look at the old one, as I did not install it.... it was something like 1.2.21...but to be honest, I didn't look |
| 15:30.52 | basilgohar | [TK]D-Fender: That's what I meant. |
| 15:31.20 | [TK]D-Fender | basilgohar: "videosupport=yes" , "allow=h263" , "allow=h263p" thats about it |
| 15:31.33 | KingDavidNYC | I dont know anything about dahdi nor do I desire to know |
| 15:31.40 | Naikrovek | how much do polycom video phones cost, anyway |
| 15:31.41 | [TK]D-Fender | KingDavidNYC: Doesnt' matter if you were the opne who installed it. What VERSIONS do you HAVE? |
| 15:32.02 | Naikrovek | eeps |
| 15:32.16 | KingDavidNYC | I compiled 1.4.23, thats the one it is running |
| 15:32.26 | [TK]D-Fender | \kingthats the ASTERISk version... and the other? |
| 15:33.02 | KingDavidNYC | which other?... the previous version?... 1.2.21... I think |
| 15:33.16 | *** join/#asterisk jpeeler (~jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:33.35 | [TK]D-Fender | KingDavidNYC: gYou need Zaptel/dahdi for meetme, so what VERSION are you running? |
| 15:34.27 | Warp4 | KingDavidNYC, I think he is asking what version of Zaptel/dahdi are you running. |
| 15:34.34 | KingDavidNYC | I will tell you in a minute.... but I dont want to use dahdi...it was using good old ztdummy befor, and I am happy with it |
| 15:34.55 | krion | the md5sum of in and out flow has both the same md5sum... |
| 15:35.03 | KingDavidNYC | but how doo I know what was it running, if I recomplied? |
| 15:35.18 | krion | anyone know a "public" echo service ? |
| 15:35.22 | [TK]D-Fender | KingDavidNYC: If you just compiled it, how the hell do you NOT know what you "just installed"? |
| 15:35.47 | klmarkea | Er, any help for my problem please ? |
| 15:36.17 | [TK]D-Fender | krion: I'm sure you have immediate access to about 4 or more. Pick a direction and yell ;) |
| 15:36.30 | KingDavidNYC | tk: you are asking me what version of dahdi I installed? it is the onw that says current on the web site |
| 15:36.51 | [TK]D-Fender | KingDavidNYC: ........ you just installed DAHDI? |
| 15:37.07 | krion | [TK]D-Fender: i don't get it :) |
| 15:37.14 | [TK]D-Fender | krion: WALLS |
| 15:37.20 | krion | oh ok |
| 15:37.28 | [TK]D-Fender | krion: pwned |
| 15:37.33 | krion | ;-) |
| 15:37.53 | KingDavidNYC | yes, first I only compiled 1.4, then, since the meetme was not responding, I compliked dahdi too |
| 15:38.00 | KingDavidNYC | compiled |
| 15:38.16 | KingDavidNYC | but I regret it |
| 15:38.22 | [TK]D-Fender | KingDavidNYC: well if you installed and configured dahdi then you need to RECOMPILE ASTERISK to have support for it |
| 15:38.24 | KingDavidNYC | it gives me all sorts of errors |
| 15:38.34 | *** part/#asterisk ManxPower-work (~EWieling@216.186.151.147) |
| 15:38.46 | KingDavidNYC | so, I prefer to stick to ztdummy, which was working very well |
| 15:39.10 | KingDavidNYC | ....before I recompiled |
| 15:39.16 | [TK]D-Fender | KingDavidNYC: Well you jsut installed DAHDI. MAKE UP YOUR MIND |
| 15:39.17 | *** join/#asterisk dacqueries (~gdecaires@65.48.199.140) |
| 15:39.25 | p3nguin | haha |
| 15:39.42 | p3nguin | zaptel and dahdi conflict, you know. |
| 15:39.44 | dacqueries | hello...anyone here has experience with setting up a call centre? |
| 15:40.06 | [TK]D-Fender | dacqueries: Sorry, could you be a little more vague please? |
| 15:40.19 | dacqueries | I could try but I'd probably lose you |
| 15:40.43 | KingDavidNYC | tk: I am going to recompile asterisk so I get rid of dahdi....do you know how to exclud dahdi support? I am running make maenuselect and it does not give me the choice |
| 15:40.49 | Naikrovek | call center(re) stuff is pretty standard fare with Asterisk. not hard to do |
| 15:41.12 | dacqueries | I'm working on a business plan to set up a call centre and I'm looking for someone that can point me in the direction of some good reading material |
| 15:41.24 | KingDavidNYC | tk: the other question is: can I do meetme in 1.4 without dahdi, only with ztdummy |
| 15:41.25 | [TK]D-Fender | dacqueries: there are multiple kinds fo call-centers... |
| 15:41.31 | [TK]D-Fender | KingDavidNYC: Yes |
| 15:41.39 | Naikrovek | 8 years ago i left Australia after living there for 2 years, and I still spell things like they do in commonwealth countries. colour. fibre. endeavour |
| 15:42.00 | Naikrovek | ~thebook |
| 15:42.00 | infobot | extra, extra, read all about it, thebook is 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://astbook.asteriskdocs.org |
| 15:42.03 | Naikrovek | dacqueries: ^^^^^^^^^^^ |
| 15:42.03 | KingDavidNYC | tk: but only if I disable dahdi support before I recompile? |
| 15:42.12 | [TK]D-Fender | king didn't say that... |
| 15:42.16 | [TK]D-Fender | KingDavidNYC: didn't say that... |
| 15:42.45 | dacqueries | thanks that's great |
| 15:42.48 | Naikrovek | welcome |
| 15:42.56 | KingDavidNYC | tk: can you please ellaborate? |
| 15:43.15 | [TK]D-Fender | KingDavidNYC: How about YOU elaborate. I asked you for versions 5 times already. |
| 15:43.16 | klmarkea | Well, i'l try to reexplain my problem : I'm trying to compile *-addons with cdr mysql support on an ubuntu server. The menuselect options quits with errors : /bin/grep: /usr/include/asterisk/buildopts.h: no such file or directory |
| 15:43.45 | klmarkea | any idea to solve this error ? |
| 15:43.48 | klmarkea | please |
| 15:44.13 | krion | you got /usr/include/asterisk/buildopts.h ? |
| 15:44.29 | KingDavidNYC | tk: the version of what? the version of dahdi?, is 2.2.1 |
| 15:44.43 | basilgohar | [TK]D-Fender: Thanks. But I'm a complete newby to Asterisk. I've only used FreeSWITCH before this. |
| 15:44.54 | basilgohar | I was hoping to find a guide for setting it up. |
| 15:45.20 | [TK]D-Fender | basilgohar: I jsut gave you the only lines you need to add to your peers. there is nothing to "setup" beyond this |
| 15:46.09 | klmarkea | krion : thanks for help. I created an asterisk symlink in /usr/include that is pointing to /usr/src/*-1.6.2.2/include |
| 15:46.22 | basilgohar | [TK]D-Fender: How about creating actual accounts? Does Asterisk act as a registrar, as well? |
| 15:46.30 | [TK]D-Fender | KingDavidNYC: ZAPTEL GOD-FUCKING-DAMMIT! You say you want to use ZTDUMMY. That is ZAPTEL, not FUCKING DAHDI. What fucking version of ***ZAPTEL** do you have installed? |
| 15:46.43 | [TK]D-Fender | reaches for his ClueBat (tm) |
| 15:46.56 | [TK]D-Fender | basilgohar: .... |
| 15:46.57 | [TK]D-Fender | ~book |
| 15:46.58 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 15:46.59 | [TK]D-Fender | ^^^^^^^^^^ |
| 15:47.12 | KingDavidNYC | tk: then say it.... say "whats the zaptel versioin".... ha ha ha.... I dont know it man |
| 15:47.14 | basilgohar | [TK]D-Fender: Thanks. |
| 15:48.21 | [TK]D-Fender | KingDavidNYC: Seriously... any neuro-electric activity going on in there? Yes the &*#ing ZAPTEL VERSION is important! |
| 15:48.22 | tzafrir_laptop | [TK]D-Fender, templer, please |
| 15:48.50 | [TK]D-Fender | tzafrir_laptop: Yes... a templar sword would be fairly effective, but you know I prefer my weaponry to be Japanese :) |
| 15:51.20 | tzafrir_laptop | klmarkea, have you built asterisk yourself? or have you installed asterisk and asterisk-dev from debs? |
| 15:51.32 | *** join/#asterisk Faithful (~Faithful@ns.linuxterminal.com) |
| 15:51.58 | klmarkea | well, I found the problem. Thanks for all |
| 15:52.08 | tzafrir_laptop | klmarkea, and the problem was? |
| 15:52.38 | klmarkea | a problem with my fingers ... bad symlink |
| 15:53.17 | tzafrir_laptop | you have symlinks in your fingers? |
| 15:53.31 | tzafrir_laptop | Or is it merely in your hand? |
| 15:54.13 | klmarkea | :-) Ok, I continue with my compilation. Tnahks |
| 15:54.29 | klmarkea | Oops. Thanks. Still my fingers ... |
| 15:54.30 | krion | exten = 909,n,Dial(sip:*673500@ekiga.net) |
| 15:54.35 | krion | is that correct ? |
| 15:54.50 | [TK]D-Fender | krion: Nope |
| 15:55.16 | krion | [TK]D-Fender: a link to the a "Dial" manpage ? |
| 15:55.17 | [TK]D-Fender | krion: s/://// |
| 15:55.38 | krion | ok, s!:!//! |
| 15:55.45 | KingDavidNYC | TK: I have no idea how to locate nor find what zaptel driver was used.... what I think I better do, is download and reinstall a newer version of a zaptel driver that is compatible with 1.4.23....does that makes sense? |
| 15:55.59 | [TK]D-Fender | KingDavidNYC: ztcfg -vvvv <----------------- |
| 15:56.49 | KingDavidNYC | great... it says: "0 channels configured" |
| 15:57.15 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 15:57.16 | *** join/#asterisk b14ck (~comradeb1@cpe-24-24-136-239.socal.res.rr.com) |
| 15:57.25 | krion | [TK]D-Fender: two / you sure ? |
| 15:57.37 | [TK]D-Fender | krion: ONE. I escaped it |
| 15:57.45 | pietro | I have a BRI on a HFC-S interface with dahdi vzaphfc driver. |
| 15:57.49 | [TK]D-Fender | krion: "core show application dial" |
| 15:58.01 | pietro | i view the span down: |
| 15:58.06 | pietro | *CLI> pri show spans |
| 15:58.06 | pietro | PRI span 1/0: Provisioned, Down, Active |
| 15:58.08 | krion | thanks... |
| 15:58.16 | pietro | why the span is down ? |
| 15:58.24 | krion | is the noob of the day |
| 15:58.45 | *** join/#asterisk ruben23 (~AGENT@122.55.48.243) |
| 15:59.16 | [TK]D-Fender | pietro: got it plugged it? Does your smartjack give you a status light? Is your mode set right for it? Care to share your configs? |
| 15:59.20 | [TK]D-Fender | ~pb |
| 15:59.21 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 15:59.22 | [TK]D-Fender | ^^^^^^^^^^^^^^^ |
| 16:00.19 | pietro | [TK]D-Fender: is plugged, my card don't have a status led, the mode is TE |
| 16:00.19 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 16:01.22 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
| 16:01.36 | krion | 2010-02-09 16:58:57 VERBOSE[7234] logger.c: -- Called *673500@ekiga.net |
| 16:01.36 | krion | 2010-02-09 16:58:58 VERBOSE[7234] logger.c: -- SIP/ekiga.net-08369fc0 is circuit-busy |
| 16:01.37 | [TK]D-Fender | pietro: TE is a BRI term... pastebin your configs for it |
| 16:01.39 | krion | too bad |
| 16:01.44 | pietro | [TK]D-Fender: this is my chan_dahdi.conf http://paste.lisp.org/display/94648 |
| 16:02.14 | [TK]D-Fender | krion: "sip set debug" <- then pastebin your next attempt |
| 16:02.36 | krion | may be i'm not authorize to it |
| 16:02.47 | [TK]D-Fender | pietro: You did a PRI span debug and you have signalling = bri_cpe_ptmp |
| 16:03.03 | [TK]D-Fender | pietro: Are you using PRI or is it BRI? Make up your mind |
| 16:03.42 | [TK]D-Fender | pietro: What model of card? |
| 16:04.37 | pietro | [TK]D-Fender: i use BRI, the card is a Cologne hfc-s, i use with zaphfc |
| 16:05.03 | krion | [TK]D-Fender: i found one that's working :) |
| 16:05.05 | [TK]D-Fender | pietro: Ok, can't help with BRI. |
| 16:05.09 | [TK]D-Fender | Any takers for this one? |
| 16:05.13 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
| 16:05.15 | *** join/#asterisk brezular (~brezular@adsl-d18.84-47-120.t-com.sk) |
| 16:05.37 | Tim_Toady | for me these cards never worked well with zaptel-dahdi, i prepfer using misdn on them |
| 16:05.44 | Tim_Toady | hfc-s that is |
| 16:07.28 | *** join/#asterisk coppice (~chatzilla@59.165.96.88) |
| 16:07.30 | tzafrir_laptop | this should work well with asterisk >= 1.6.0 |
| 16:07.50 | tzafrir_laptop | dahdi driver for that is not in the tree, sadly, but is maintained |
| 16:07.55 | Tim_Toady | its a matter of asterisk version or dadhi version? |
| 16:08.00 | tzafrir_laptop | (and also included in dahdi-extra) |
| 16:08.21 | tzafrir_laptop | Tim_Toady, two separate issues |
| 16:09.09 | pietro | i have asterisk 1.6.2, asterisk see the card, but the span is down. |
| 16:09.25 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 16:09.59 | [TK]D-Fender | KingDavidNYC: .... so? 1/3 later... got a ZAPTEL VERSIOn for us? |
| 16:10.03 | [TK]D-Fender | 1/2H* |
| 16:10.05 | [TK]D-Fender | gah |
| 16:10.45 | c0rnoTa | I can't install Dahdi-Linux 2.2.1 on Gentoo with 'kernel 2.6.31-gentoo-r6' - sethdlc.c gives me error: 'union <anonymous>' has no member named 'ifru_settings' :( |
| 16:13.24 | c0rnoTa | looks like described here issue: http://bugs.gentoo.org/show_bug.cgi?id=270283 |
| 16:13.39 | c0rnoTa | but only with 2.6.31 and 2.2.1 tools |
| 16:17.45 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 16:18.21 | *** join/#asterisk soman (~somnath@dsl-jklbrasgw1-fe19fb00-113.dhcp.inet.fi) |
| 16:18.49 | *** join/#asterisk n3hxs (~HAMming@63.68.135.4) |
| 16:26.54 | mrprozac | https://issues.asterisk.org/view.php?id=15254 |
| 16:27.03 | beek | CU all later. |
| 16:27.04 | mrprozac | @c0rnoTa: https://issues.asterisk.org/view.php?id=15254 |
| 16:27.10 | *** part/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 16:27.43 | mrprozac | @c0rnoTa: https://issues.asterisk.org/view.php?id=13439 |
| 16:28.20 | *** join/#asterisk mrbnet_ (~mrbnet@74-95-100-233-Minnesota.hfc.comcastbusiness.net) |
| 16:29.39 | *** join/#asterisk moy (~moy@74.12.121.212) |
| 16:29.49 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-172.dslextreme.com) |
| 16:30.05 | mrbnet_ | I have complied and installed libpri but asterisk doesn't appear to be loading it. How do I make sure it get loaded? |
| 16:31.33 | c0rnoTa | mrprozac: it's all about 2.2.0 and kernel < 2.6.30. I have never experienced this problem on kernel 2.6.30-r6 and dahdi 2.2.0 on gentoo. |
| 16:34.05 | c0rnoTa | mrprozac: so, the problem didn't appeared because sethdlc is not installed :) |
| 16:41.19 | c0rnoTa | mrprozac: and for my tasks i don't need sethdlc. Problem was not solved but it does not touch us :) |
| 16:41.34 | c0rnoTa | may be i didn't install something |
| 16:41.48 | Qwell | c0rnoTa: yell at Gentoo to stop patching their kernel. |
| 16:41.50 | c0rnoTa | s/install/installed |
| 16:42.01 | Qwell | OR, update their ebuild to do the right thing |
| 16:42.27 | Qwell | They added a useless patch that breaks compiling. Nothing we can do about it. |
| 16:43.17 | c0rnoTa | Qwell: thanks, i'll remember about it! |
| 16:44.00 | c0rnoTa | So, thanks all |
| 16:44.05 | c0rnoTa | and good bye |
| 16:45.03 | *** join/#asterisk garymc (~chatzilla@host81-139-136-16.in-addr.btopenworld.com) |
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| 16:57.58 | pta200 | anyone know if there will be any changes in the res_ais module since the openais project got split into two projects? |
| 16:59.48 | [TK]D-Fender | pta200: Install res_8ball.so and see what it returns |
| 17:01.00 | jaytee | lol |
| 17:02.41 | *** join/#asterisk naif (~5d233753@gateway/web/freenode/x-tegplbbzaffovfjh) |
| 17:02.57 | naif | Hi, anyone that's a Master of ZRTP protocol? Need support for a consultancy of a mid-sized project |
| 17:05.24 | *** join/#asterisk Badrobot- (~badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 17:06.08 | *** part/#asterisk naif (~5d233753@gateway/web/freenode/x-tegplbbzaffovfjh) |
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| 17:07.39 | cjk | hi, i do a sip to dahdi call, but then when I put hold on my mobile phone the local moh starts which i do not want, how can i tell dahdi to ignore the moh signal it gets? |
| 17:07.49 | *** join/#asterisk gurra (~gurra__@unaffiliated/gurra) |
| 17:09.56 | russellb | cjk: I'm not aware of a way to configure that. You'd have to hack the code. |
| 17:12.31 | dlynes_laptop | russellb, setting to an mohclass that just plays silence would work, wouldn't it? |
| 17:12.58 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.157) |
| 17:13.01 | carrar | cell phone is providing the moh |
| 17:13.02 | *** part/#asterisk pta200 (~paolo@goose.specialai.com) |
| 17:13.05 | dlynes_laptop | CunningPike, good morning |
| 17:13.15 | dlynes_laptop | carrar, oh, in that case...nothing to do with asterisk? |
| 17:13.21 | carrar | I think |
| 17:13.22 | cjk | russellb, but like this i can not hear the remote musiconhold, thats not good? |
| 17:14.21 | leifmadsen | russellb: !!! |
| 17:15.09 | *** join/#asterisk corretico (~laguilar@201.201.46.106) |
| 17:19.20 | *** join/#asterisk fish-bulb (~cstewart@nat/digium/x-iascdqolthdcsmvk) |
| 17:20.36 | *** part/#asterisk pietro (~pietro@88.149.164.32) |
| 17:23.27 | CunningPike | Morning, dlynes_laptop! |
| 17:27.04 | krion | thanks everyone, see you tomorrow :) |
| 17:27.23 | krion | i've to write an ivr if you know what i mean ;) |
| 17:28.57 | [TK]D-Fender | krion: We do if you know what you said |
| 17:29.11 | krion | think so :) |
| 17:29.33 | krion | doesn't fully understand yet how's the macro called using realtime but will soon hopefully |
| 17:34.56 | *** join/#asterisk mnick86 (~mnick86@188.195.88.246) |
| 17:35.44 | mnick86 | Hey, I get: "Span number should be a valid span number, not '> 1'" , system.conf: "span => 1,1,0,ccs,ami,..." where is my fault ? |
| 17:36.12 | *** join/#asterisk Victor_Yure (~victor@unaffiliated/victoryure/x-837844) |
| 17:37.08 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 17:41.42 | [TK]D-Fender | krion: mnicremove the > |
| 17:41.48 | [TK]D-Fender | mnicRemove the ">" |
| 17:41.51 | [TK]D-Fender | ashdashgdjhfd |
| 17:47.18 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 17:48.03 | *** join/#asterisk mort_gib (~mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 17:51.12 | *** join/#asterisk nesa-laptop (~nesa@109.121.12.97) |
| 17:51.13 | *** join/#asterisk cuco (~Diego@95.35.44.83) |
| 17:51.33 | ariel_ | Afternoon folks |
| 17:51.52 | nesa-laptop | hi. :) |
| 17:51.53 | *** join/#asterisk rocksfrow (~kyle@74-92-146-97-WashingtonDC.hfc.comcastbusiness.net) |
| 17:52.17 | rocksfrow | in what mysql table could i find queue statistics? |
| 17:52.30 | rocksfrow | and/or could somebody direct me in the right path for enabling a callback function on JOIN of a queue |
| 17:52.36 | rocksfrow | i know i read about it somewhere, but can't find it again |
| 17:53.00 | KingDavidNYC | heeelp... I recompiled to 1.4.23, reinstalled zaptel to 1.4.21... I get "invalid conference number" |
| 17:53.02 | rocksfrow | this way I could for example, execute a script everytime somebody joins the queue |
| 17:53.04 | rocksfrow | ? |
| 17:53.18 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.182.155) |
| 17:53.19 | raden_work | bmoraca_work, ? |
| 17:53.48 | KingDavidNYC | at least now I get to hear the prompt :) |
| 17:54.19 | KingDavidNYC | zaptel is running |
| 17:55.14 | Qwell | rocksfrow: there are manager events that get fired |
| 17:55.52 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 17:56.20 | [TK]D-Fender | KingDavidNYC: stop *. do "modprobe ztdummy", do "ztcfg -vvvv" if that doesn't complain, restart * and retest |
| 17:56.34 | rocksfrow | Qwell...hrm? |
| 17:56.49 | raden_work | anyone ever cluster mysql databases for use with asterisk ? |
| 17:56.49 | Qwell | rocksfrow: what was unclear about what I said? |
| 17:56.55 | rocksfrow | nvm i see what you mean |
| 17:57.08 | [TK]D-Fender | rocksfrow: Who are you talking about. caller's joining, or members? |
| 17:57.18 | rocksfrow | [TK]D-Fender, callers.. |
| 17:57.23 | rocksfrow | when they call and i dump then into the queue |
| 17:57.38 | nesa-laptop | do you guys have any idea if i could expect the web gui to work out of the box on debian after "apt-get install asterisk" if i edited http.conf and manager.conf? because i get not found or access denied errors all the time :S |
| 17:57.41 | rocksfrow | i'd like to put a record in a table |
| 17:57.49 | [TK]D-Fender | rocksfrow: Indeed Qwell has informed you of one. Also you could fire off some action YOURSELF just before calling Queue. This is YOUR dialplan you know... |
| 17:57.52 | rocksfrow | (so i can do an ajax call on the customer service system to show an incoming call) |
| 17:58.05 | [TK]D-Fender | rocksfrow: So call something BEFore dumping them into Queue() |
| 17:58.26 | rocksfrow | yeah that would give me more oncontrol, huh |
| 17:58.42 | rocksfrow | let me ask you, by default is there a mysql table an entry is created in already? |
| 17:58.57 | [TK]D-Fender | rocksfrow: Depends if you set up logging to one |
| 17:59.04 | [TK]D-Fender | rocksfrow: by default, no. |
| 17:59.11 | rocksfrow | default no, ok |
| 17:59.52 | [TK]D-Fender | rocksfrow: * can't magically know you're running MySQL and know connection details, or what DB... or what table... or that the fields are sane, etc |
| 18:00.02 | *** join/#asterisk moy (~moy@74.12.123.196) |
| 18:00.05 | rocksfrow | [TK]D-Fender, can you confirm for me on some vocab, 'member' is the caller correct..and 'agent' is obviously the customer service re |
| 18:00.27 | [TK]D-Fender | rocksfrow: No, "members" are the people who ANSWER callers in a queue. |
| 18:00.44 | rocksfrow | so a member is the same as an agent? |
| 18:00.54 | Qwell | Every time somebody uses the word "agent", a kitten is killed. |
| 18:01.01 | rocksfrow | well |
| 18:01.02 | nesa-laptop | :D |
| 18:01.04 | [TK]D-Fender | rocksfrow: "Agent" is a special term referring to Chan_agent... or one misused by people who should be referring to them as "members" |
| 18:01.05 | rocksfrow | then maybe it shouldn't be in the doc :) |
| 18:01.17 | [TK]D-Fender | rocksfrow: "the doc"? pardon? |
| 18:01.19 | rocksfrow | ah.. |
| 18:01.22 | [TK]D-Fender | rocksfrow: WHICH doc? |
| 18:01.36 | nesa-laptop | i'd like to read some docs tbh :D |
| 18:01.41 | rocksfrow | 's mistake |
| 18:01.46 | Qwell | ~book |
| 18:01.47 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 18:01.49 | [TK]D-Fender | nesa-laptop: Why start now? :p |
| 18:01.50 | Qwell | nesa-laptop: ^ |
| 18:01.52 | rocksfrow | yeah i have the book |
| 18:01.57 | rocksfrow | somewhere... |
| 18:02.06 | rocksfrow | needs to read it all the way through :-p |
| 18:02.13 | rocksfrow | still very new to * |
| 18:02.13 | nesa-laptop | thanks |
| 18:02.31 | nesa-laptop | [TK]D-Fender: i'm not really a guy who uses manuals usually |
| 18:02.40 | [TK]D-Fender | ~osmosis |
| 18:02.41 | infobot | [~osmosis] Osmosis is the act of beating yourself on the head repeatedly with THE BOOK, until some measure of absorption has occured ... or at least until your unconsciousness restores peace to the channel ... |
| 18:02.42 | [TK]D-Fender | ^^^^ |
| 18:02.42 | nesa-laptop | but with this i feel like it'd be a good thing to do |
| 18:02.52 | nesa-laptop | :D |
| 18:03.13 | ariel_ | wow, at least people have the book now to read. |
| 18:03.33 | rocksfrow | [TK]D-Fender, osmosis..nice ha |
| 18:03.48 | paulc | AGI question: On a low powered box, I've seen occasional instances of an error message when launching an AGI script (written in PHP using PHPAGI, if it makes any difference?). safe_wait something something, and not all the channel vars are accessible from my script. Likely cause = slowness of loading PHP? because it works "most of the time" and I can't reliably replicate the problem. (low spec dev box != what I'll use in production) |
| 18:04.09 | KingDavidNYC | TK: nada... I stopped asterisk, did "/etc/init.d/zaptel stop" then start, reinitiated *....nothing..."invalid conference number" |
| 18:04.09 | [TK]D-Fender | paulc: What variables? Where do we see your calls? |
| 18:04.27 | [TK]D-Fender | KingDavidNYC: those aren't the commands I gave you and I don't see your CLI output |
| 18:04.28 | KingDavidNYC | TK: zaptet doesnt complaint |
| 18:04.36 | Qwell | ariel_: "Back in my day, we read the source code, and loved it"? |
| 18:04.56 | nesa-laptop | the wonders of open-source. |
| 18:05.09 | ariel_ | Qwell: yes that was the normal reply in this channel 6 years ago when I started using asterisk or was it more then 6 years... humm time flys |
| 18:06.09 | nesa-laptop | so, since i'm a total newbie (been "using" asterisk for a little more than an hour), where should i ask if i'm using asterisk on debian? :) |
| 18:06.39 | ariel_ | nesa-laptop: what do you want to do? Asterisk runs fine on Debian |
| 18:07.03 | nesa-laptop | ariel_: after running apt-get install asterisk and changing some config files, i can't get webgui to work |
| 18:07.16 | ariel_ | webgui? |
| 18:07.17 | nesa-laptop | i get 404 and access denied |
| 18:07.18 | nesa-laptop | yeah |
| 18:07.21 | nesa-laptop | well |
| 18:07.42 | ariel_ | I have never ever installed asterisk with a apt-get |
| 18:07.42 | nesa-laptop | i was reading some tutorials |
| 18:07.54 | nesa-laptop | and they said there was a way to access the webgui |
| 18:08.13 | ariel_ | you have to set the gui up asterisk has no default gui |
| 18:08.30 | nesa-laptop | well, that points me in the right direction at least |
| 18:08.36 | nesa-laptop | what is the gui called? |
| 18:09.14 | KingDavidNYC | TK: went back, did exactly as you said, and same thing |
| 18:09.15 | ariel_ | well there is freepbx, there is asterisknow, elastix there is asterisk-gui but it's kinda not very active |
| 18:09.36 | [TK]D-Fender | ~wmmfp ? |
| 18:09.39 | ariel_ | nesa-laptop: google |
| 18:09.48 | nesa-laptop | okay, thanks! |
| 18:09.51 | [TK]D-Fender | ~wmmfp |
| 18:09.52 | KingDavidNYC | TK: I will send you cli here, it is 3 lines |
| 18:10.10 | [TK]D-Fender | dangit |
| 18:10.17 | KingDavidNYC | <PROTECTED> |
| 18:10.17 | KingDavidNYC | <PROTECTED> |
| 18:10.17 | KingDavidNYC | <PROTECTED> |
| 18:10.38 | [TK]D-Fender | KingDavidNYC: pastebin <- |
| 18:11.23 | KingDavidNYC | ok, you need more cli? |
| 18:11.42 | [TK]D-Fender | KingDavidNYC: All the shit I told you to do plus CONFIGS |
| 18:11.55 | KingDavidNYC | ok |
| 18:17.03 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 18:18.06 | nesa-laptop | hey, one more quickie question, google wasn't very helpful: anyone ever installed asterisknow from usb? |
| 18:18.15 | nesa-laptop | the target machine doesn't have a cd drive |
| 18:18.29 | KingDavidNYC | TK: mmmmm... I see....I stopped *, then I did /etc/init.d/zaptel stop, then I did modprobe ztdummy, and it now complaints!.... here is the pastebin http://pastebin.com/m59a97fee |
| 18:18.36 | Qwell | nesa-laptop: look for instructions for doing so with CentOS. AsteriskNOW won't be very different from that |
| 18:18.50 | nesa-laptop | thanks |
| 18:22.30 | [TK]D-Fender | KingDavidNYC: Took too long to come up with this... maybe someone else can help you from here. |
| 18:22.44 | [TK]D-Fender | is off drilling for oil... |
| 18:23.15 | nesa-laptop | Qwell, ariel_, thanks for the help and pointers, gtg now, see ya around maybe :) |
| 18:23.18 | nesa-laptop | bye all |
| 18:28.18 | *** join/#asterisk trelane` (~trelane@funtoo/staff/trelane) |
| 18:28.52 | *** join/#asterisk jax55 (~jax55@c-98-254-160-125.hsd1.fl.comcast.net) |
| 18:28.54 | trelane` | what are all the cool people using for doing rating (billing) on asterisk cdr records (for CSV or MySQL) |
| 18:29.03 | Kobaz | postgres |
| 18:29.08 | Qwell | awk |
| 18:29.14 | jaytee | lol |
| 18:29.19 | trelane` | Qwell, thinking more commercial :) |
| 18:29.41 | trelane` | a2billing or it's ilk |
| 18:29.42 | paulc | [TK]D-Fender: delayed reply, sorry - people turned up at my desk.. Specifically the channel name. I have a "registration" AGI script, meaning I know caller X is now online.. I want to register which channel they're on so I can redirect them round the dialplan later. The var ends up being blank. I can have it fail, then have 15 calls with no problems. Using PHPAGI so the intermittent-ness bugs me. |
| 18:29.43 | Qwell | awk that generates html |
| 18:30.11 | jax55 | how do i know T38 is enabled for faxing? is it supposed to show as a codec or something else? |
| 18:30.17 | jaytee | "Once there was a bird called the auk that forgot how to fly and learned how to walk" |
| 18:30.39 | jaytee | jax55, it's a matter of faith |
| 18:31.15 | ariel_ | trelane`: we use radius here for billing |
| 18:31.24 | jax55 | jaytee :) and lets say i beleive really hard, how can i see it? |
| 18:31.35 | trelane` | ariel_, yep I've set up radius before, and there's no way in hell I'm doing it again :) |
| 18:31.52 | jaytee | jax55, I suppose that's between you and the Gods of T.38 |
| 18:32.18 | ariel_ | We have allot of Cisco routers and termination points with them so it is easy for us to have our asterisk system talk the same as our cisco's |
| 18:32.41 | jax55 | jaytee hehehe....the gods told me RTFM, so i did....but now i need to verify that i put the right things in the confs |
| 18:33.32 | ariel_ | jax55: you need to set it up in your dial plan for it to work |
| 18:33.43 | *** join/#asterisk JAMMAN2110 (~james@unaffiliated/jamman2110) |
| 18:33.53 | jaytee | jax55, if you're talking the new fax for asterisk then you're way ahead of me at this point. still running 1.4 with t38_udptl passthrough using Sipura ATA's. |
| 18:34.29 | jax55 | ariel_ so its per device? i can't set it system wide? |
| 18:34.57 | jax55 | jaytee i have the latest trixbox, so i guess the newest asteriks too |
| 18:35.04 | Kobaz | all kinds of bad writes :( |
| 18:35.05 | Kobaz | er |
| 18:35.24 | jax55 | jaytee how are the Sipura ATAs? problems with NAT? plug&play? |
| 18:35.27 | ariel_ | argh you said the words that I stop help with, trixbox |
| 18:35.27 | *** join/#asterisk vexter (~guilherme@unaffiliated/vexter) |
| 18:35.36 | p3nguin | jax55: Are you using 1.6.x? |
| 18:35.39 | *** part/#asterisk vexter (~guilherme@unaffiliated/vexter) |
| 18:36.23 | jaytee | jax55, I've been using them for analog phones and for fax machines here just fine. |
| 18:36.26 | jax55 | Asterisk 1.6.0.10-FONCORE-r40 |
| 18:36.45 | *** join/#asterisk jdnWEST (~jdn@mx1.westparkcom.net) |
| 18:36.47 | jaytee | some of my fax machines still use pots but some are over PRI through Asterisk |
| 18:36.56 | jax55 | nice |
| 18:37.17 | *** join/#asterisk Tech_Travis (~tech_trav@208.179.137.131) |
| 18:37.33 | jax55 | p3nguin Asterisk 1.6.0.10-FONCORE-r40 |
| 18:37.51 | p3nguin | Did you already install FFA? |
| 18:38.01 | jax55 | no, what is that? |
| 18:38.08 | p3nguin | Fax For Asterisk |
| 18:38.17 | jax55 | how can i tell if its there? |
| 18:38.40 | p3nguin | fax show licenses |
| 18:39.05 | jaytee | I'd like to use the new FFA with Asterisk to take some unused DID#'s and route incoming fax calls over PRI to my Asterisk box and then to Exchange UM. I've done it internally from a Sipura ATA but not from outside from a standard fax machine hooked to a POTS line. I need to somehow make asterisk bridge from a non-T.38 fax device calling in and act as a go-between to Exchange UM using T.38. |
| 18:40.09 | p3nguin | If you're going to take fax in and then send it right back out, wouldn't you consider iaxmodem (possibly with hylafax)? |
| 18:40.14 | jax55 | p3nguin that command does not work, but rpm shows: asterisk16-fax-1.6.0.10-5_trixbox |
| 18:40.36 | p3nguin | There's a reason we don't support trixbox, I guess. |
| 18:41.22 | jaytee | p3nguin, don't think iaxmodem would be any help. Exchange Unified Messaging only talks SIP/TCP and supports T.38 but doesn't support IAX |
| 18:41.23 | p3nguin | Look in that rpm and see what modules it provides. I bet it's not FFA at all, but is actually app_fax. |
| 18:41.54 | jax55 | p3nguin ok, i will do that...and if it is app_fax, is that bad? |
| 18:42.08 | p3nguin | app_fax and fax for asterisk are two totally different things. Not necessarily bad things, just different. |
| 18:42.23 | jax55 | gotcha |
| 18:43.06 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 18:43.42 | jax55 | p3nguin is t38 a protocol or a codec? meaning if it was installed should i see it in the list of codecs? |
| 18:44.51 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 18:51.11 | p3nguin | jax55: It's a faxing standard... I don't know if that makes it a protocol or not. I would think it does. |
| 18:51.37 | jax55 | ok, thanks for all the help and direction, i will look into it |
| 18:53.45 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.178.71.dsl.dyn.forthnet.gr) |
| 18:54.24 | *** join/#asterisk staffmember (~singulari@static-71-183-79-24.nycmny.fios.verizon.net) |
| 18:54.44 | staffmember | is there anywhere besides the cisco website where i can get Sip Firmware for cisco 79xx phones ? |
| 18:55.46 | TheDavidFactor | is it possible to reload manager.conf in asterisk 1.4.24.1 without having to restart asterisk? |
| 18:57.53 | *** join/#asterisk franck (~franck@tikiwiki/franck) |
| 18:58.28 | *** join/#asterisk spenguin[work] (~penguin@59.162.86.164) |
| 18:58.57 | *** join/#asterisk mikkel (~mikkel@84-238-113-66.u.parknet.dk) |
| 18:59.31 | meshe | staffmember: http://www.asteriskguru.com/tutorials/cisco_7960_ip_phone_configuration.html |
| 18:59.32 | Qwell | staffmember: nope |
| 19:00.15 | meshe | http://www.cisco.com/pcgi-bin/tablebuild.pl/sip-ip-phone7960 |
| 19:08.35 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 19:09.20 | KingDavidNYC | can somebody please help me figure out why ztdummy is not installing well? |
| 19:09.56 | Qwell | not without seeing what it's doing |
| 19:09.59 | Qwell | ~pastebin |
| 19:09.59 | infobot | [~pastebin] 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 |
| 19:10.16 | KingDavidNYC | Qwell: thanks, what do you need pasted? |
| 19:10.19 | *** join/#asterisk cweagans (~quassel@66-193-105-126.static.twtelecom.net) |
| 19:10.35 | cweagans | is there a guide to installing asterisk from source somewhere? |
| 19:10.41 | Qwell | something that shows the problem |
| 19:10.45 | Qwell | cweagans: ./configure; make; make install |
| 19:11.22 | cweagans | Qwell: for everything? I got asterisk, dahdi-linux, dahdi-tools, and libpri |
| 19:11.41 | p3nguin | There's ordinarily a README in stuff. |
| 19:11.43 | Qwell | pretty much |
| 19:11.44 | rocksfrow | does anybody have any _bad_ to say about asteriskNOW |
| 19:11.54 | rocksfrow | anything* |
| 19:12.16 | p3nguin | rocksfrow: Other than it uses a GUI which destroys the normal method of administration via command line, no. |
| 19:12.21 | KingDavidNYC | let me give background.... my problem is that meetme says "invalid conference number" I assume the ztdummy is not installing well |
| 19:12.25 | ariel_ | but setup libpri first, then dahdi-linux, dahdi-tools then asterisk |
| 19:13.00 | Qwell | KingDavidNYC: I would start by looking at your config |
| 19:13.10 | ariel_ | rocksfrow: it's a good iso with a GUI |
| 19:13.18 | cweagans | actually, I just had a thought. Anybody know if asterisk + dependencies are in the Ubuntu repos? (or any other apt repo?) |
| 19:13.37 | p3nguin | kingdavidnyc: A couple hours ago you were talking about installing dahdi. What's it gonna be, dahdi or zaptel? |
| 19:13.50 | p3nguin | cweagans: It is. |
| 19:13.55 | rocksfrow | p3nguin, destroys the normal method? |
| 19:14.09 | rocksfrow | hrm.. |
| 19:14.19 | p3nguin | rocksfrow: Once you use a GUI on Asterisk, you can no longer edit you config files by hand. |
| 19:14.24 | p3nguin | ~freepbx |
| 19:14.24 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 19:14.32 | rocksfrow | yeah i use freepbx currently |
| 19:14.33 | ariel_ | p3nguin: not true |
| 19:15.00 | rocksfrow | p3nguin, it DESTROYS it? or changes it? |
| 19:15.12 | Qwell | rocksfrow: it completely controls it |
| 19:15.14 | p3nguin | If you already use FreePBX, then you already know you can't edit the configs manually. |
| 19:15.18 | ariel_ | if you learn it correctly you can edit the custom files on freepbx |
| 19:15.19 | Qwell | there is no proper manual configuration |
| 19:15.33 | rocksfrow | p3nguin, i'm very new to this..the current setup was not setup by me, but a previous employee.... |
| 19:15.35 | p3nguin | AsteriskNOW is pretty nice, with all that excluded. |
| 19:15.38 | KingDavidNYC | a couple of hours ago I was asking how to disable dahdi from the compilation, so that it would not conflict with zaptel.... I did exclude it from the compilation and now I am troubleshooting zaptel |
| 19:15.40 | rocksfrow | i'm going to be building my own setup..and am looking at my options |
| 19:15.53 | *** join/#asterisk cuco (~Diego@95.35.77.153) |
| 19:15.54 | p3nguin | rocksfrow: Do you want to run FreePBX? |
| 19:16.02 | ariel_ | KingDavidNYC: why go backwards? |
| 19:16.03 | *** join/#asterisk ChUbB (~IceChat7@cpc2-aztw12-0-0-cust229.aztw.cable.virginmedia.com) |
| 19:16.04 | rocksfrow | p3nguin, ..thats what i'm trying to figure out :-p |
| 19:16.21 | p3nguin | rocksfrow: Do you know how to admin the box via command line, without any GUI at all? |
| 19:16.22 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-145.cablep.bezeqint.net) |
| 19:16.24 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 19:17.04 | rocksfrow | p3nguin, i'm leaning towards it bc it's the only thing i'm atleast at ALL familiar with...but, i have to say it does complicate support |
| 19:17.08 | rocksfrow | especially here in #asterisk |
| 19:17.15 | p3nguin | rocksfrow: Do you want to learn, or do you want to continue on at the level in which you currently operate? |
| 19:17.21 | *** join/#asterisk LemensTS (~customgt@adsl-70-238-130-71.dsl.stlsmo.sbcglobal.net) |
| 19:17.35 | LemensTS | how do you send voicemail to the h extensiion upon hangup in 1.4? |
| 19:17.48 | p3nguin | rocksfrow: If you aren't willing to spend time learning to admin via CLI, AsteriskNOW is a great implementation of Asterisk on CentOS. |
| 19:17.52 | KingDavidNYC | this server was running very well with 1.2 with ztdummy, but I had to upgrade to 1.4.... at a point I thought dahdi was a problem....but now I was able to install 1..4.23 well, plus zaptel 1.4.21 to, what I believe, satisfactorily |
| 19:18.18 | p3nguin | rocksfrow: And you can get Digium support if you need it. |
| 19:18.28 | rocksfrow | p3nguin, are there other options instead of freepbx that offers an optional GUI, but still leaves the CLI in-tact? |
| 19:18.34 | p3nguin | no |
| 19:18.37 | spenguin[work] | what do you guys think of a career in just VoIP with asterisk |
| 19:18.41 | spenguin[work] | is it possible |
| 19:18.52 | spenguin[work] | or can it exist |
| 19:18.53 | Gido-E | if you have customers |
| 19:18.53 | ariel_ | rocksfrow: they all leave the cli like normal |
| 19:19.06 | KingDavidNYC | the problem is that when I run "modprobe ztdummy" the first time after I stop zaptel, I get error "unable to open master device /dev/zap/ctl" |
| 19:19.18 | rocksfrow | ariel_, ...? am i misunderstaning p3nguin ? |
| 19:19.25 | ariel_ | Yes |
| 19:19.34 | ariel_ | CLI has nothing to do with config files |
| 19:19.37 | p3nguin | rocksfrow: You've incorrectly used the term "CLI" so he's giving you grief. |
| 19:19.39 | Gido-E | KingDavidNYC lsmod | grep zt |
| 19:19.54 | rocksfrow | p3nguin, :p |
| 19:19.55 | carrar | spenguin[work], lots of people do |
| 19:19.57 | Gido-E | proppably it removes the ztdummy |
| 19:20.20 | p3nguin | rocksfrow: When being Asterisk-specific, the CLI is Asterisk's console. When talking about sysadmin in general, the CLI is simply your shell prompt. |
| 19:20.27 | spenguin[work] | carrar: hrm |
| 19:20.33 | carrar | key is customers |
| 19:20.34 | rocksfrow | p3nguin, right i understand that, lol.. |
| 19:20.54 | carrar | and a good sales person |
| 19:20.59 | KingDavidNYC | then if I type modprobe ztdummy again, no error |
| 19:21.01 | rocksfrow | so, basically if you've got a freepbx system..all of the config files are wack |
| 19:21.10 | spenguin[work] | carrar: what I meant to ask is - are there job opportunities in just asterisk stuff |
| 19:21.11 | p3nguin | rocksfrow: So FreePBX won't break Asterisk's CLI, but it will make it nearly impossible to admin your asterisk from the system's CLI. |
| 19:21.11 | rocksfrow | hence, no regular asterisk documentation is going to help? |
| 19:21.22 | spenguin[work] | and is it worth specializing in? |
| 19:21.24 | p3nguin | rocksfrow: right |
| 19:21.24 | rocksfrow | p3nguin, well put, thanks. |
| 19:21.25 | KingDavidNYC | here is the output from lsmod | grep zt |
| 19:21.27 | dlynes_laptop | CunningPike, How's Vancouver? |
| 19:21.27 | KingDavidNYC | ztdummy 7876 0 |
| 19:21.27 | KingDavidNYC | zaptel 194308 5 ztdummy |
| 19:21.29 | carrar | spenguin[work], yeah even in here sometimes people will ask |
| 19:21.33 | p3nguin | ~freepbx |
| 19:21.34 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 19:21.36 | p3nguin | rocksfrow: ^^^ |
| 19:21.50 | rocksfrow | thx p3nguin |
| 19:22.48 | p3nguin | rocksfrow: Operations of Asterisk remain the same, but FreePBX completely overtakes your config files, leaving them unable to be handled manually. |
| 19:23.12 | spenguin[work] | carrar: ok |
| 19:24.12 | KingDavidNYC | this simple mmetme conf setup has taken me the whole day :( |
| 19:24.15 | rocksfrow | p3nguin, gotcha. |
| 19:24.38 | ariel_ | p3nguin: that is only 1/2 correct, you have the custom files that can be edited as well as others like the nat files. |
| 19:27.32 | cweagans | When I try to start asterisk with `asterisk -vvvvgc`, i get the error "ERROR[6578]: chan_h323.c:3166 load_module: Unable to create H323 listener" |
| 19:27.33 | cweagans | any idea why? |
| 19:27.55 | *** join/#asterisk athayde (~atha@unaffiliated/athayde) |
| 19:28.04 | ariel_ | cweagans: are you using h323? |
| 19:28.35 | cweagans | er. i dunno? :) |
| 19:28.48 | cweagans | new pbx, new user/admin, new knowledge :) |
| 19:28.55 | cweagans | (or lack thereof) |
| 19:29.40 | pigpen | Hi all, it has been years since I have looked at limiting concurrent iax channels on a trunk. Is there a way in asterisk 1.6 to handle limiting the concurrent number of calls in a trunk? |
| 19:29.57 | pigpen | Seems there is very little info on this. |
| 19:30.07 | *** join/#asterisk mafrac (~manuel@77.211.184.206) |
| 19:30.18 | p3nguin | pigpen: maxcallnumbers, maybe? |
| 19:30.43 | *** part/#asterisk mafrac (~manuel@77.211.184.206) |
| 19:30.59 | Kobaz | pigpen: call-limit= ? |
| 19:31.22 | pigpen | nothing of either of these is mentioned in the 1.6.1 example file. |
| 19:31.40 | *** part/#asterisk LemensTS (~customgt@adsl-70-238-130-71.dsl.stlsmo.sbcglobal.net) |
| 19:31.49 | p3nguin | kobaz: That's for sip. |
| 19:32.10 | Kobaz | hmm |
| 19:32.18 | p3nguin | pigpen: I don't have 1.6.x, so I have to go by what I have for 1.4. |
| 19:32.19 | *** join/#asterisk TuneMan6 (TuneMan6@94-225-140-86.access.telenet.be) |
| 19:32.32 | Kobaz | hah |
| 19:32.38 | Kobaz | you know, yeah that is just in sip |
| 19:32.43 | pigpen | yeah, it was a scarce subject years ago...seems it still is |
| 19:32.46 | Kobaz | i've been using it in iax.conf forever |
| 19:32.51 | pigpen | yeah, me too.... |
| 19:32.52 | TuneMan6 | Hello guys, I just installed AsteriskNow x64 on vmWare workstation, what is the default login/password or the IP to logon to the web GUI? |
| 19:32.55 | Kobaz | i guess it's been silently ignored |
| 19:32.58 | pigpen | just never been sucessful with limiting. |
| 19:33.01 | *** join/#asterisk goshawk (~quassel@93-34-48-63.ip48.fastwebnet.it) |
| 19:33.08 | p3nguin | There's maxcallnumbers and callnumberlimits in iax.conf. |
| 19:33.12 | goshawk | hi |
| 19:33.46 | pigpen | hmm... |
| 19:33.47 | goshawk | is it possible to configure asterisk to act like a PSTN gateway using a common modem interface which comes in bundle with some pcs? |
| 19:33.54 | TuneMan6 | Hello guys, I just installed AsteriskNow x64 on vmWare workstation, what is the default login/password or the IP to logon to the web GUI? |
| 19:33.55 | p3nguin | goshawk: no |
| 19:34.02 | p3nguin | ~modem |
| 19:34.03 | infobot | somebody said modem was (Modulator/Demodulator) A device to turn digital signals to analog ones and back again, so they can be transmitted and translated back to digital at another modem without loss. Used for communication through means of audio, telephone, CB, etc. Random disconnects? S10=255 sure to do the trick! |
| 19:34.06 | Kobaz | yeah, looks like callnumbers is the way to go |
| 19:34.08 | TuneMan6 | i've already tried admin/admin or freepbx / fpbx |
| 19:34.15 | *** join/#asterisk jelly-bean (~mikesmull@75-148-103-190-Utah.hfc.comcastbusiness.net) |
| 19:34.15 | Kobaz | TuneMan6: #asterisknow |
| 19:34.51 | p3nguin | tuneman6: I'll presume you installed the x86_64 version... the quickstart guide says freepbx/fpbx for the login info. |
| 19:34.59 | goshawk | p3nguin: what can i use? |
| 19:35.05 | TuneMan6 | Already tried that, no luck... |
| 19:35.12 | jelly-bean | our users are reporting phone calls cutting out intermittently, calls dropping, and some delays in the phone call getting connected when the phone is picked up |
| 19:35.21 | p3nguin | tuneman6: You did get to the GUI login, though, right? |
| 19:35.22 | Kobaz | jelly-bean: more details |
| 19:35.22 | TuneMan6 | Is there any way to get the IP, so I can see the web GUI? |
| 19:35.25 | Kobaz | ~details |
| 19:35.25 | infobot | If you want help on a topic, you HAVE to say more than "it doesn't work, help!" or else you'll get no help whatsoever. Give as many details as you can or else no one can give any suggestions. |
| 19:35.34 | TuneMan6 | I just get a black scren with login: |
| 19:35.50 | jelly-bean | details like what |
| 19:35.53 | p3nguin | tuneman6: That sounds like the system console, not the web interface. |
| 19:36.04 | Kobaz | jelly-bean: what devices are you using... t1? analog? paste your debug logs, configs, etc |
| 19:36.05 | p3nguin | tuneman6: You can login as root with whatever root password you put on it. |
| 19:36.11 | jelly-bean | 4 t1 pri |
| 19:36.20 | jelly-bean | theres nothing interesting in the logs |
| 19:36.22 | Kobaz | jelly-bean: your signalling is probably wrong, or clocking |
| 19:36.31 | Kobaz | paste your t1 counters |
| 19:36.34 | goshawk | p3nguin: which hardware should i buy then to make a PSTN gateway with a common pc? |
| 19:36.37 | Kobaz | ~pastebin |
| 19:36.37 | infobot | [~pastebin] 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 |
| 19:36.41 | jelly-bean | kobaz: where can i get the t1 counters |
| 19:36.42 | KingDavidNYC | Anybody would know shy I get message "this is not a valid conference number" please |
| 19:36.42 | TuneMan6 | it always says "login incorrect" |
| 19:36.53 | Kobaz | jelly-bean: if you're using a sangoma card, you can use wanpipemon |
| 19:36.54 | p3nguin | goshawk: SPA-3102 |
| 19:36.59 | Kobaz | jelly-bean: other than that, you're out of luck |
| 19:37.24 | TuneMan6 | I've been using 3CX for a year now, and it's always bad in sound quality, I'm hoping to get it resolved using Asterisk |
| 19:37.29 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 19:37.31 | goshawk | p3nguin: it's by cisco, isn't it? |
| 19:37.33 | TuneMan6 | But I don't have any technical background on Linux |
| 19:37.37 | Kobaz | jelly-bean: it could quite possibly be clocking |
| 19:37.43 | Qwell | TuneMan6: quickstart guide covers all of this |
| 19:37.44 | p3nguin | tuneman6: Not much I can tell you if you don't know your root password. |
| 19:38.01 | p3nguin | goshawk: Yes, Cisco Business (Linksys branded). |
| 19:38.09 | Kobaz | TuneMan6: you can ask it #asterisknow |
| 19:38.24 | TuneMan6 | Yeah but it's very quiet there, no responses :p |
| 19:38.33 | Kobaz | they know their system much better than we do |
| 19:38.38 | *** join/#asterisk magic_hat (~geoffdoug@24-148-27-181.grn-bsr1.chi-grn.il.cable.rcn.com) |
| 19:38.49 | TuneMan6 | ok, I got in |
| 19:39.02 | TuneMan6 | what's the command to get the IP, so that I can login to the web GUI with my host? |
| 19:39.07 | p3nguin | tuneman6: Use the 'ifconfig' command. |
| 19:39.07 | Kobaz | ifconfig |
| 19:39.23 | Kobaz | TuneMan6: for linux questions, there's #linux |
| 19:39.45 | TuneMan6 | great! |
| 19:39.47 | TuneMan6 | it works :p |
| 19:40.13 | ThoMe | hello! |
| 19:40.18 | p3nguin | More specifically: "ifconfig eth0|grep "inet addr"|sed 's^ inet addr:^^g'|cut -d " " -f1" |
| 19:40.42 | cweagans | p3nguin: nice =P |
| 19:40.45 | Kobaz | depends if his device is eth0 or not... he may have two network cards... or udev remaping devices |
| 19:41.20 | *** join/#asterisk voipmonk (~shido6@12.130.116.139) |
| 19:41.48 | *** join/#asterisk Tim_Toady (~moi@77.49.48.61.dsl.dyn.forthnet.gr) |
| 19:44.13 | pigpen | I find this interesting, that using the command "iax2 show callnumber usage" it shows that peered IP address max is 2048. |
| 19:44.24 | p3nguin | Does Digium store have huge Asterisk decals to stick on computer cases? |
| 19:44.32 | pigpen | adding maxcallnumbers= to the peer/user in iax.conf does nothing. |
| 19:44.47 | p3nguin | Did you "iax2 reload"? |
| 19:44.52 | pigpen | yes |
| 19:44.54 | *** join/#asterisk moos3 (~rgenthner@216.52.121.66) |
| 19:45.06 | p3nguin | checks |
| 19:45.47 | pigpen | did too. |
| 19:45.51 | moos3 | will this work http://pastie.org/private/cmxrzywdyavjwfockf0urq |
| 19:45.57 | pigpen | yeah, no dice. |
| 19:46.28 | Qwell | moos3: why wouldn't it? |
| 19:46.38 | pigpen | shit, I have to leave. If you find anything, post...I'll be back. |
| 19:46.55 | moos3 | Qwell: I haven't dont anything like that before and just wanted to make sure |
| 19:48.00 | moos3 | I have to remap 80+ extensions, I was trying to figure out away to do some type of look up |
| 19:48.27 | p3nguin | pigpen: Neither maxcallnumbers nor callnumberlimits on the IP address for my peer seem to bring the number down. |
| 19:51.33 | p3nguin | pigpen: 74.54.54.178 2 32 |
| 19:51.37 | p3nguin | pigpen: got it |
| 19:51.58 | p3nguin | <PROTECTED> |
| 19:52.20 | p3nguin | That is by using maxcallnumbers=32 in the peer definition. |
| 19:53.11 | *** part/#asterisk moos3 (~rgenthner@216.52.121.66) |
| 19:54.38 | drfreeze | Hi |
| 19:54.55 | drfreeze | Anyone have a case where a polycom phone only rings on the first ring, then stops ringing? |
| 19:55.21 | drfreeze | I have 2 phone 611 and 612. Both a dialed, but 611 only rings one time. 612 keeps ringing |
| 19:55.28 | cweagans | drfreeze: it sounds like you do... |
| 19:55.32 | Naikrovek | are you using followme |
| 19:55.53 | pigpen | p3nguin, did you test in trunk mode? |
| 19:56.04 | drfreeze | cweagans: :) |
| 19:56.36 | p3nguin | pigpen: Yes, I use trunk mode. |
| 19:56.45 | p3nguin | <PROTECTED> |
| 19:57.12 | drfreeze | it must have a windows subsystem. Restarting the phone fixed the problem |
| 19:57.29 | raden_work | bmoraca_work, ? you around ? |
| 19:57.34 | jelly-bean | the whole system is locking up hardcore |
| 19:57.44 | jelly-bean | calls not getting thru. dropping and cutting and lagging out like crazy |
| 19:59.01 | pigpen | p3nguin, did you do this on the local or remote box (remote being the one with the "real" dial tone) |
| 19:59.27 | pigpen | as I am applying to the remote (ie: the one with the pri's) and it still shows 2048 |
| 19:59.33 | p3nguin | pigpen: I did it on my side of my trunk. I can't control the ITSP's box. |
| 19:59.56 | *** join/#asterisk Micc (~quassel@c-98-225-57-96.hsd1.wa.comcast.net) |
| 20:00.16 | pigpen | hmm...I am trying to limit a customer's number of calls via my service. |
| 20:00.19 | Micc | I keep seeing zombie parking channels in my logs. anyone know why that happens? |
| 20:00.23 | Kobaz | jelly-bean: what t1 card |
| 20:00.26 | pigpen | I don't like the idea they control this. |
| 20:00.35 | Micc | exited non-zero on 'SIPPeer/SIP/dandd4-00011acc<ZOMBIE>' |
| 20:00.36 | Kobaz | jelly-bean: you still haven't given any real details as to your system |
| 20:01.44 | p3nguin | pigpen: Try the callnumberlimits setting. |
| 20:02.05 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 20:02.05 | *** mode/#asterisk [+o file] by ChanServ |
| 20:02.32 | Kobaz | jelly-bean: do you bring you car to a mechanic and tell them your car has problems, but won't let them look under the hood? |
| 20:03.06 | *** part/#asterisk TuneMan6 (TuneMan6@94-225-140-86.access.telenet.be) |
| 20:08.02 | p3nguin | pigpen: If I use maxcallnumbers= in general or use the callnumberlimits setting, then iax2 show callnumber usage shows the new limit, but iax2 show peer voipms shows unlimited. |
| 20:08.29 | cweagans | I'm looking through the tutorial at http://onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html?page=3. I've based my config files on the ones on that page. Is there a softphone that I can test it with? I tried Xlite, but it gave me some asterisk errors about not being able to identify or something |
| 20:08.49 | p3nguin | pigpen: If I use maxcallnumbers= in the peer definition, then both the show peer and show callnumber usage will show the new limit. |
| 20:10.15 | jax55 | ok, so the fax t38 thing is no good for me... |
| 20:10.46 | jax55 | i'll explain the exact situation, and maybe someone has that great solution i am looking for... |
| 20:12.03 | jax55 | i have an FM audio signal (1200Hz) that I want to capture as the voicemail, it is a signal very similar to what a fax "sounds" like...i then take the wav file produced from the voicemail and import it into some other system. |
| 20:12.39 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 20:12.45 | jax55 | right now, the SIP seems to be distorting the audio signal in a way that make the import come out as garbage |
| 20:12.59 | voipmonk | what codec are you using? |
| 20:13.16 | jax55 | i tried the highest quality protocols / codecs, g722, speex, g729 |
| 20:14.10 | jax55 | any ideas? |
| 20:16.03 | p3nguin | For faxing over VoIP, you need to use ulaw (g.711u). |
| 20:16.48 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 20:17.08 | jax55 | p3nguin it is not exactly faxing, but i will do that and test again....is there any specific reason why a codec with less "quality" would be better for this task? |
| 20:17.36 | voipmonk | its going to be best effort |
| 20:17.40 | voipmonk | ulaw |
| 20:17.42 | voipmonk | g711 |
| 20:17.48 | p3nguin | Well, g.711u is not a compressing codec, so I don't know how the quality is lesser. |
| 20:17.55 | voipmonk | if there is any data in the sound, best of luck |
| 20:18.11 | jax55 | yes, there is embedded data in the FM signal |
| 20:18.44 | jax55 | ok, i will do tests right now with the g711, and will let you know, thx |
| 20:20.49 | *** part/#asterisk jelly-bean (~mikesmull@75-148-103-190-Utah.hfc.comcastbusiness.net) |
| 20:25.14 | Qwell | p3nguin: compression != lossy |
| 20:28.30 | *** join/#asterisk TSM2 (~the_softw@87-194-32-212.bethere.co.uk) |
| 20:32.35 | *** join/#asterisk mpe (~mpe@0x4dd624b2.adsl.cybercity.dk) |
| 20:33.39 | jax55 | p3nguin from what i understand, g722 is considered "wideband", 16KHz, which is a higher quality then g711 |
| 20:33.52 | jax55 | g711 / ulaw seems to distort the data as well |
| 20:33.58 | jax55 | any other ideas? |
| 20:35.24 | jax55 | so the exact question would be, how do you transmit an FM audio signal over VOIP without distorting / destroying the data in the signal? |
| 20:38.51 | Chainsaw | jax55: With a very wide-band codec. |
| 20:39.00 | Chainsaw | jax55: Wide enough to accept your non-voice signal. |
| 20:39.00 | KingDavidNYC | Anybody please help me why I am getting "invalid conference number" |
| 20:39.34 | jax55 | Chainsaw i tried g722 and speex....both wideband and kill the data...anything else i can try? |
| 20:39.45 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-148.cablep.bezeqint.net) |
| 20:39.49 | p3nguin | kingdavidnyc: Dynamically create the conference number OR create the conference number in the meetme config file. |
| 20:40.06 | Amorsen | jax55: I don't know any codec wide enough for FM |
| 20:41.10 | jax55 | Amorsen aha, so this is the source you think? the FM signal is 1200Hz, and G722 is supposidly 16000Hz...but is that something else? |
| 20:41.12 | leifmadsen | the point of codecs is that they are designed for the human voice |
| 20:41.30 | leifmadsen | and the frequencies they carry |
| 20:41.46 | leifmadsen | I would presume an FM signal could contain signals outside that range |
| 20:41.55 | leifmadsen | s/signals/frequencies/ |
| 20:42.01 | Amorsen | jax55: I bet the FM signal is a lot more than 1200Hz, or 12000Hz. |
| 20:42.32 | KingDavidNYC | p3guin, all conference configuration is ok, I am 100% sure... it was working well before....the problem is with a faulty timer installation |
| 20:42.52 | Katty | hi |
| 20:42.53 | jax55 | <PROTECTED> |
| 20:43.13 | Amorsen | jax55: If it works over a regular phone it'll work fine with ulaw or Alaw, because that's what regular phones ARE |
| 20:43.21 | Amorsen | Ah, unless it's a leased line? |
| 20:43.25 | KingDavidNYC | p3nguin: I cant figure it out, but has to do with zaptel |
| 20:43.42 | Amorsen | You can still get 2-wire leased lines without filters in some places, and they'll do way more than 8kHz |
| 20:43.43 | p3nguin | kingdavidnyc: As before, load either the ztdummy or the dahdi_dummy module. |
| 20:43.58 | jax55 | Amorsen leased line? this is direct connection between my voip phone and my server, no trunks, nada |
| 20:44.00 | Katty | ahh what a lovely day it's been |
| 20:44.17 | Amorsen | jax55: Yes, but you say it works over a regular phone |
| 20:44.19 | freezey | so i am transfering these 2 extensions between PBX's all other extensions work except 2 the difference is the 2 that are not working with direct extension are not setup on a phone it is supposed to forward to an outside number.. you can dial the full number just not direct extension.. anyway to resolve this? |
| 20:44.26 | KingDavidNYC | I am loading the ztdummy module, but I dont know whats wrong... as far as dahdi, I dont want to go there |
| 20:44.30 | Amorsen | How regular is that phone actually which you used for testing? |
| 20:44.54 | jax55 | Amorsen just a standard bellsouth line |
| 20:45.05 | Amorsen | jax55: Permanent or did you dial the number? |
| 20:45.13 | p3nguin | kingdavidnyc: Did you unload it recompile zaptel, then load the new module? Perhaps there is something wrong with the old module. |
| 20:46.02 | Amorsen | Many radio stations used phone lines for getting audio from the studio to the transmitter, but those were unfiltered pure analog lines |
| 20:46.25 | jax55 | Amorsen not sure what you mean, on the regular phones, if i transmit the signals between 2 bellsouth land lines, where one side is the transmitter and the other is the receiver, then it works fine |
| 20:46.33 | KingDavidNYC | p3nguin: yes, I recompiled both 1.4.23, and then recompiled zaptl 1.4.21... and it all did not give me complaints |
| 20:47.00 | Amorsen | jax55: On a regular phone line where you have to dial a number and wait for the recipient to pick up, like any other phone? |
| 20:47.04 | p3nguin | I should remove dahdi and load zaptel just for the heck of it. |
| 20:47.14 | jax55 | Amorsen correct |
| 20:47.22 | Amorsen | Ok, then ulaw will work |
| 20:47.30 | Amorsen | If it doesn't it can be made to work |
| 20:47.30 | jaytee | does zaptel even work with * 1.4.23? I thought that zaptel only worked with 1.4.22 or earlier and you had to use dahdi with 1.4.23 |
| 20:47.49 | Katty | hugs jaytee |
| 20:47.50 | Amorsen | It'll sound like crap, but the phone line will sound like crap too. |
| 20:47.50 | KingDavidNYC | p3nguin: to the best of my knowledge, I recompiled 1.4.23 exclucing all dahdi references |
| 20:47.53 | jax55 | Amorsen, it doesn't work....please tell me how :) |
| 20:48.01 | p3nguin | jaytee: Oh, I'm using 1.4.29, so you think I can't use zaptel on it? |
| 20:48.20 | jaytee | with that new a version definitely not |
| 20:48.23 | p3nguin | I should still try just for the heck of it. |
| 20:48.35 | p3nguin | See what the result it, ya know? |
| 20:48.46 | p3nguin | s/it/is/ |
| 20:48.57 | Amorsen | jax55: You want to get rid of any jitter you may have on the RTP packets. You want to set jitter buffers to fixed (large values) |
| 20:48.59 | jax55 | Amorsen, none work.....ulaw, g722, speex, nada....unless aterix is tampering with the file when it is saving it to voicemail |
| 20:49.13 | jaytee | russellb, PING! |
| 20:49.30 | Amorsen | And you need to make sure you aren't losing packets |
| 20:49.32 | jaytee | leif, you here? |
| 20:49.53 | jax55 | Amorsen checking confs for jitter |
| 20:50.13 | Amorsen | ATA's or Digium analog cards? |
| 20:50.24 | Amorsen | Or non-Digium, for that matter |
| 20:50.40 | leifmadsen | jaytee: ya |
| 20:50.49 | jax55 | Amorsen using a voip phone |
| 20:51.02 | Amorsen | You're playing it through the handset of the VoIP phone? |
| 20:51.15 | voipmonk | ..... |
| 20:51.16 | jaytee | do you remember which was the last 1.4.x version to support zaptel? I thought it was 1.4.22 or 1.4.21 |
| 20:51.22 | Amorsen | Echo cancellation will kill you then |
| 20:51.32 | thehar | 1.4.21` |
| 20:51.53 | jaytee | thehar, thanks, I was pretty sure it was either .22 or .21 |
| 20:52.09 | thehar | i _think_ .22 kinda worked.. but not so well =) |
| 20:52.24 | jax55 | Amorsen yes, through the handset |
| 20:52.35 | jaytee | p3nguin, so all you can hope to accomplish is to see lots of fun error messages |
| 20:53.14 | Amorsen | jax55: You'll need to turn off echo cancellation in the phone, if that is possible, as well as AGC |
| 20:53.15 | jaytee | thehar, yeah I think I tried .22 at first and dropped back to .21 cuz it was acting "hinky" |
| 20:53.16 | p3nguin | It'll only take a minute to remove one and put back the other, so I'll do it just for fun and reference. |
| 20:53.24 | Amorsen | But basically I'd recommend giving up on that route |
| 20:53.29 | jax55 | Amorsen jbenable=yes && jbforce=yes |
| 20:53.49 | jax55 | Amorsen why give up? this works over handsets on regular phones as well |
| 20:54.03 | leifmadsen | jaytee: they all suppose zaptel as far as I'm aware |
| 20:54.12 | Amorsen | Regular phones don't have echo cancellation circuitry, and most of them don't have AGC either |
| 20:54.14 | leifmadsen | s/suppose/support/ |
| 20:54.41 | leifmadsen | jaytee: it's after a certain 1.4 version that it supports both zaptel and dahdi -- only the 1.6.x versions don't support zaptel |
| 20:54.54 | jaytee | leifmadsen, ah ok. |
| 20:54.56 | Amorsen | You'll have more luck with a regular phone connected to an ATA set up for modem use |
| 20:55.13 | leifmadsen | I keep seeing that a lot -- not sure why people think it doesn't support zaptel in 1.4 after a certain version :) |
| 20:55.21 | leifmadsen | I don't remember that being documented anywhere |
| 20:55.32 | jaytee | probably cuz it's such a pain to get working after 1.4.21 |
| 20:55.36 | jax55 | Amorsen ok, i'll try that then...i have an ATA as well |
| 20:55.41 | Amorsen | Anyway, how do I reject a call in an exten => the same way Asterisk rejects it if there's no matching extension at all? |
| 20:55.54 | leifmadsen | jaytee: pretty sure you just modify asterisk.conf to use zaptel... |
| 20:56.10 | jaytee | i had no probs getting zaptel working fine with .21 but .22 I had an easier time getting dahdi to work than zaptel |
| 20:56.47 | Amorsen | I'd like to make it indistinguishable for peers whether their calls were rejected by non-matching an extension or by my filtering rules |
| 20:58.07 | CunningPike | dlynes_laptop: Sunny and 48 degrees! :-) Enjoying the snow? |
| 20:59.19 | Amorsen | It can't be that difficult... Does Asterisk indicate anything to the peer when the first matching extension is reached? |
| 20:59.33 | Amorsen | (With SIP) |
| 20:59.47 | p3nguin | Hmm. I installed zaptel, but modprobe won't load the modules. |
| 21:00.05 | p3nguin | # modprobe ztdummy |
| 21:00.05 | p3nguin | FATAL: Module ztdummy not found. |
| 21:00.43 | p3nguin | But I have the module installed: # find /lib/modules/ -name ztdummy\* |
| 21:00.43 | p3nguin | /lib/modules/2.6.32-ARCH/misc/ztdummy.ko |
| 21:01.16 | staffmember | anyone have sip firmware for a cisco 7075g ? |
| 21:01.26 | p3nguin | Cisco might. |
| 21:01.32 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
| 21:01.35 | Amorsen | 2.6.32-ARCH sounds like an unusual directory |
| 21:01.42 | Amorsen | Is there anything else in there? |
| 21:01.48 | p3nguin | Why is it unusual? |
| 21:01.52 | jax55 | Amorsen my voip phone has AGC disabled, and Jitter Buffer Minimum Delay = 35 msec and Jitter Buffer Optimization Factor 07 (out of 13) |
| 21:02.13 | jax55 | Amorsen should i change this? |
| 21:02.19 | Amorsen | jax55: But can you turn off echo cancellation? |
| 21:02.33 | p3nguin | /lib/modules/2.6.32-ARCH/ is /lib/modules/`uname -r`/ |
| 21:02.33 | Amorsen | If you can make the jitter buffer fixed size, that helps |
| 21:02.43 | Amorsen | Ok then |
| 21:03.04 | jax55 | Amorsen i can set jitter to 13 fixed, but i cannot find echo cancelation settings |
| 21:03.09 | Amorsen | p3nguin: depmod -a ? Perhaps that isn't necessary anymore |
| 21:03.31 | Amorsen | jax55: I don't think I've seen a VoIP phone with selectable echo cancellation |
| 21:03.39 | p3nguin | amorsen: Good call. Now it loads. |
| 21:03.40 | jax55 | Amorsen, there is something called Silence Compression and it is disabled |
| 21:03.58 | Amorsen | jax55: Good that it's disabled |
| 21:04.08 | Amorsen | But that isn't echo cancellation either |
| 21:04.16 | jax55 | Amorsen, ok, then i will try with this new jitter settings |
| 21:05.07 | *** join/#asterisk grey-monkey (~ericshell@75-148-103-190-Utah.hfc.comcastbusiness.net) |
| 21:05.26 | grey-monkey | Does anyone know what the default T1 call sequence is in Asterisk? A-sequence or D-sequence? |
| 21:05.49 | grey-monkey | I have 4 T1s and the first is set to A-sequence and the other three are set to D-sequence |
| 21:06.27 | grey-monkey | I'm having problems and I'm trying to find Asterisk's default so that I can tell my communication provider which sequence to set my T1s |
| 21:06.37 | Amorsen | grey-monkey: Do you mean the order in which calls are made to the individual lines, or does it mean something else? |
| 21:06.58 | grey-monkey | Yes, example 1-12 channels are inbound and 13-23 are outbound |
| 21:06.59 | Amorsen | grey-monkey: If so, that is something you set in your Dial statement, not in configuration files |
| 21:07.08 | jax55 | Amorsen do you still recomend sticking with G711.ulaw or should i go for the wider bands? |
| 21:07.23 | Amorsen | Ah right grey-monkey, you want groups of channels |
| 21:07.27 | *** join/#asterisk doctorray (~ray@static-71-177-137-76.lsanca.fios.verizon.net) |
| 21:07.53 | Amorsen | jax55: ulaw for sure. The wider bands will mess up anything which isn't voice |
| 21:08.04 | grey-monkey | I'm actually using switchvox right now and I don't see anyway of seeing how to do that |
| 21:08.12 | jax55 | Amorsen you got it, thx, trying now |
| 21:08.29 | Amorsen | jax55: And set relaxdtmf=no and dtmfmode= to something which isn't inband |
| 21:08.45 | p3nguin | I guess I have to recompile asterisk after zaptel is installed, because it certainly does not work by only removing dahdi and installing zaptel. |
| 21:09.03 | thehar | that is in channel admin under system setup > you create a new channel group from there |
| 21:09.03 | jax55 | Amorsen like what? 100Hz ? |
| 21:09.42 | Amorsen | jax55: dtmfmode? Should be RFC2833 or info, depending on what the phone is set to. Just avoid inband for your purposes |
| 21:09.53 | *** join/#asterisk Tim_Toady (~moi@77.49.48.61.dsl.dyn.forthnet.gr) |
| 21:10.08 | Amorsen | And check on the console with lots of -vvvvv that Asterisk isn't receiving dial tones |
| 21:10.27 | jax55 | Amorsen perfect, will do |
| 21:12.36 | *** join/#asterisk ManxPower-work (~EWieling@216.186.151.147) |
| 21:13.00 | *** join/#asterisk jtexter3 (~jtexter3@74-44-78-248.dsl1-field.roch.ny.frontiernet.net) |
| 21:13.34 | jtexter3 | I just want to confirm what I've on google. If I'm getting dahdi_test of ~98.3, that can cause some issues? |
| 21:14.50 | *** join/#asterisk ttyS1 (~julian@173-12-98-213-miami.hfc.comcastbusiness.net) |
| 21:15.21 | jtexter3 | I have some cases where a caller comes in, and the prompt sounds robotic. This seems to only affect TDM calls, as I have SIP traffic hitting the same box without issue |
| 21:16.20 | *** join/#asterisk iq (~chatzilla@unaffiliated/iq) |
| 21:16.21 | iq | Hi |
| 21:17.16 | *** part/#asterisk grey-monkey (~ericshell@75-148-103-190-Utah.hfc.comcastbusiness.net) |
| 21:20.31 | jax55 | Amorsen still very distorted data |
| 21:23.05 | Amorsen | jax55: Try a regular phone and ATA instead. I doubt you'll get anywhere with a VoIP phone |
| 21:23.29 | *** join/#asterisk terracon (~greisky@CPE001cf097a750-CM0012254076d6.cpe.net.cable.rogers.com) |
| 21:23.46 | jax55 | Amorsen ok, i will try |
| 21:24.51 | *** join/#asterisk Diogo (~info@a85-138-16-15.cpe.netcabo.pt) |
| 21:25.35 | *** join/#asterisk mpe (~mpe@0x4dd624b2.adsl.cybercity.dk) |
| 21:27.19 | *** join/#asterisk niekvlessert (~niek@5ED25657.cable.ziggo.nl) |
| 21:27.25 | niekvlessert | hello, i've got a question |
| 21:27.31 | niekvlessert | there's a lot of isdn drivers |
| 21:27.37 | niekvlessert | i don't understand anymore |
| 21:27.55 | niekvlessert | misdn, chan_capi, zaptel(dahdi), which one does what? |
| 21:28.03 | niekvlessert | i wan't to support junghanns cards |
| 21:34.21 | *** join/#asterisk duckz (~duckz@86.107.84.186) |
| 21:34.59 | *** join/#asterisk e-jones (~jkastner@84.242.102.36) |
| 21:37.08 | *** join/#asterisk lanning (~lanning@208.87.235.224) |
| 21:54.43 | raden_work | is there a way to run xterm -fg green -bg black and then connect to ssh -X user@192.168.1.* as one shortcut on my desktop ? |
| 21:55.22 | Chainsaw | raden_work: Sure, through the miracle of bash. |
| 21:55.42 | Chainsaw | raden_work: #!/bin/bash, your two lines, chmod +x and give it a friendly name with .sh in it. |
| 21:56.02 | raden_work | Chainsaw, sweet and just a link to my desktop ? |
| 21:56.14 | Chainsaw | raden_work: Yeah, stick it somewhere convenient and just link to your .sh |
| 21:56.30 | raden_work | like in my bin folder ? |
| 21:56.52 | Chainsaw | raden_work: Hm, wait. xterm. Just xterm -fg green -bg black ssh -X user@192.168.1.* should do then. |
| 21:57.19 | Chainsaw | raden_work: Documentation suggests: xterm -fg green -bg black -e ssh -X user@192.168.1.* |
| 21:57.30 | raden_work | -e ? |
| 21:57.32 | Chainsaw | raden_work: If it whines at all, stick a -- after that -e |
| 21:57.50 | Chainsaw | raden_work: [-e command args ...] <- You want to run ssh as the "command". |
| 21:57.57 | raden_work | gotcha |
| 21:58.13 | pigpen | p3nguin, thanks for the additional info, however it seems that "iax2 show callnumber usage" is not available in 1.6.1, whereas 1.6.1.12 it is. I'll update the older, so I have a level playing field. |
| 21:58.33 | pigpen | p3nguin, nevertheless, I could never get the max call no to say "1" vs. 2048. |
| 21:58.42 | *** join/#asterisk Kobaz (~kobaz@its.kobaz.net) |
| 21:58.44 | pigpen | p3nguin, think it is a special compile bit? |
| 21:59.10 | raden_work | xterm -fg green -bg black -fn 9x15 -sl 9999 -geometry 80x40-0+0 -e ssh -X xxxxxxxx@192.168.1.xxx |
| 21:59.55 | raden_work | sweet :D |
| 22:00.22 | *** join/#asterisk Diogo (~info@a85-138-16-15.cpe.netcabo.pt) |
| 22:00.28 | raden_work | Chainsaw, thanks a ton bro :) |
| 22:00.31 | Chainsaw | Sure thing :) |
| 22:01.38 | p3nguin | raden_work: You'd probably like aterm better than xterm. |
| 22:01.59 | p3nguin | pigpen: Let me see if I can get max call to show 1. |
| 22:02.40 | Chainsaw | is one of those gnome-terminal heathens |
| 22:03.19 | pigpen | p3nguin, yeah, who knows, maybe a min of 2 |
| 22:03.53 | p3nguin | maxcallnumber limit of 1 for 74.54.54.178 has been reached! |
| 22:03.57 | p3nguin | 74.54.54.178 1 1 |
| 22:04.12 | p3nguin | iax2_do_register: Unable to create call for registration |
| 22:05.00 | *** join/#asterisk Diogo (~info@a85-138-16-15.cpe.netcabo.pt) |
| 22:05.08 | pigpen | heh. |
| 22:05.22 | *** part/#asterisk ruben23 (~AGENT@122.55.48.243) |
| 22:06.12 | p3nguin | Seems like 2 or more is desired. |
| 22:06.20 | raden_work | heya p3nguin |
| 22:06.26 | p3nguin | hi |
| 22:06.58 | pigpen | Ok, so I would want the callnumberlimits=2 maxcallnumbers=2 both in the peer only? |
| 22:08.27 | pigpen | I would rather not do anything to the "customer" box, as I would hope that I could "account" for everything on my box, which is providing the service. |
| 22:08.35 | raden_work | Chainsaw, so with bash i could technically launch 4 xterms using that same command 4 times with slight variation ? |
| 22:08.45 | Chainsaw | raden_work: Indeed. |
| 22:08.54 | raden_work | omfg thatd save me like 20 min a day |
| 22:09.17 | raden_work | Chainsaw, i know the security concerns but is there a way to pass the ssh password ? |
| 22:09.25 | Chainsaw | raden_work: Sure, SSH keys. |
| 22:09.29 | raden_work | ? |
| 22:09.46 | Chainsaw | raden_work: You can identify by a public key instead. It's how I log into my kit. |
| 22:10.06 | raden_work | your kit ? |
| 22:10.47 | Chainsaw | raden_work: Yes. I have about 42 servers. |
| 22:11.02 | raden_work | so you know the frusteration :( |
| 22:11.08 | raden_work | only has 8 |
| 22:14.38 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 22:14.43 | Chainsaw | Hi Fender. |
| 22:15.10 | [TK]D-Fender | Chainsaw: Still slightly... |
| 22:15.25 | niekvlessert | raden_work: ssh-keygen -t rsa |
| 22:15.26 | [TK]D-Fender | <-- root canal |
| 22:16.02 | niekvlessert | copy the public key to the host in ~/.ssh/authorized_keys |
| 22:16.03 | niekvlessert | done |
| 22:16.47 | Chainsaw | [TK]D-Fender: Ouch :( |
| 22:17.07 | p3nguin | pigpen: maxcallnumbers= can go in general and apply to all peers. |
| 22:17.59 | p3nguin | ssh-copy-id |
| 22:18.14 | p3nguin | saves time/trouble |
| 22:18.30 | raden_work | nicknick, sweet :) thanks :) |
| 22:18.59 | raden_work | is there a way to copy from my client machine to the host machine via ssh ? from command line ? |
| 22:19.24 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 22:21.17 | *** join/#asterisk shimizu (~shimizu@91.103.58.11) |
| 22:21.30 | niekvlessert | yeah |
| 22:21.33 | p3nguin | raden_work: (1617.58) <p3nguin> ssh-copy-id |
| 22:21.37 | p3nguin | raden_work: (1618.13) <p3nguin> saves time/trouble |
| 22:21.53 | niekvlessert | p3nguin: didn't know that :) |
| 22:21.59 | shimizu | good day |
| 22:22.38 | *** join/#asterisk Deeewayne (~dwayne@75.76.254.162) |
| 22:22.38 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 22:22.43 | shimizu | how can i change voicemail greetings |
| 22:22.46 | shimizu | except using VoicemailMain |
| 22:24.25 | raden_work | p3nguin, is there a way to copy files through a ssh connection ? |
| 22:24.25 | raden_work | or am i outta my mind ;) |
| 22:24.25 | p3nguin | raden_work: Sure. |
| 22:24.25 | shimizu | raden_work: use scp |
| 22:24.36 | p3nguin | raden_work: But if you use ssh-copy-id, you wouldn't need to worry with it. |
| 22:24.55 | shimizu | scp file-to-copy user@host:file-to-write |
| 22:25.08 | pigpen | p3nguin, well, cannot do it there. I have several peers that use more channels. |
| 22:25.14 | raden_work | p3nguin, im doing that but I need it for many other things at times |
| 22:25.19 | pigpen | ie: one is 23 channel, another is 2, another is 4, etc... |
| 22:25.51 | p3nguin | raden_work: You can use sftp, scp, or rsync without too much trouble. |
| 22:26.41 | raden_work | thanks :) |
| 22:27.03 | shimizu | hey someone ? |
| 22:27.33 | p3nguin | shimizu: Record sound files, put them in the appropriate directories, enjoy. |
| 22:28.27 | KingDavidNYC | TK: help me please with this ztdummy sh*t, I cant go home till I fix it |
| 22:29.00 | p3nguin | I tried zaptel/ztdummy with asterisk 1.4.29 and it won't work. The solution is to use dahdi. |
| 22:29.23 | russellb | zaptel has been unmaintained for almost 1.5 years ... |
| 22:29.34 | russellb | you absolutely should never try to do a new install with it, it's a terrible idea |
| 22:29.37 | KingDavidNYC | can I use dahdi as a dummy timer? |
| 22:29.38 | niekvlessert | raden_work: scp file user@host |
| 22:29.46 | russellb | dahdi is zaptel with a new name. |
| 22:30.03 | niekvlessert | raden_work: wait until u understand the full potential from ssh |
| 22:30.04 | russellb | ztdummy became dahdi_dummy |
| 22:30.05 | niekvlessert | will blow your mind :) |
| 22:30.27 | p3nguin | kingdavidnyc: That's the ONLY reason I use dahdi at all -- dahdi_dummy.ko. |
| 22:30.44 | KingDavidNYC | I want to cry... I would have to get acquainted with dahdi |
| 22:31.26 | russellb | it's not a big change |
| 22:31.35 | p3nguin | Install dahdi and dahdi-tools. configure/build/install asterisk. Load the dahdi_dummy module. |
| 22:33.16 | KingDavidNYC | ok guys.... I'll be back.... feel weird about having to that, but ok, you guys are the experts |
| 22:33.38 | *** join/#asterisk jax55 (~jax55@c-98-254-166-182.hsd1.fl.comcast.net) |
| 22:33.54 | jax55 | Amorsen no luck with the ATA either |
| 22:34.34 | KingDavidNYC | p3nguin: you mean you tried zaptel/ztdummy because of my question? |
| 22:34.41 | p3nguin | yes |
| 22:34.58 | KingDavidNYC | p3nguin: wow man... thanks |
| 22:36.25 | p3nguin | Now I didn't do any hacking on source code or anything like that... I just installed zaptel-svn, then build asterisk. MeetMe would not work after installing the recompiled asterisk and loading zaptel/ztdummy modules. |
| 22:37.25 | p3nguin | So I went back to dahdi and the previous asterisk build... MeetMe worked again. |
| 22:38.06 | KingDavidNYC | p3nguin: exactly whats happening to me... but I have other 1.4 servers running ztdummy.... it probably is something related with the specific version |
| 22:38.39 | KingDavidNYC | p3nguin: ok thanks, I am gonna do that |
| 22:38.42 | p3nguin | People were talking about not having luck with zaptel after 1.4.21, if you didn't see that discussion. |
| 22:38.56 | raden_work | jon@Server200:/SERVER100/COMPUTER/ASTERISK> scp asterisk-addons-1.6.0.4.tar.gz tss@192.168.1.38:/home/tss/SOURCE |
| 22:38.56 | raden_work | Password: |
| 22:38.56 | raden_work | scp: /home/tss/SOURCE/asterisk-addons-1.6.0.4.tar.gz: Permission denied |
| 22:39.26 | p3nguin | What are you trying to do? |
| 22:39.33 | raden_work | copy |
| 22:39.41 | KingDavidNYC | p3guin: you know, I am gonna give it one more try compling 1.4.19 |
| 22:39.50 | russellb | oh dear god, why 1.4.19? that's ancient |
| 22:39.52 | *** join/#asterisk chilicuil (~sistemas@unaffiliated/chilicuil) |
| 22:39.58 | raden_work | copy to 192.168.1.38 |
| 22:40.11 | p3nguin | raden_work: ssh 192.168.1.38 wget http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-addons-1.6.0.4.tar.gz |
| 22:40.15 | niekvlessert | does dir SOURCE exist on target? |
| 22:40.25 | p3nguin | or course it didn't. |
| 22:40.38 | raden_work | p3nguin, I know i can do that but im trying to get scp figured out just a small file to toss around :) |
| 22:40.41 | russellb | There have been 1493 updates (fixes) to Asterisk 1.4 since 1.4.19 |
| 22:40.57 | raden_work | drops jaw |
| 22:41.26 | p3nguin | raden_work: scp localfile user@remotehost: |
| 22:41.38 | p3nguin | raden_work: That will put localfile in user's homedir. |
| 22:41.54 | raden_work | ohhhhhhhh |
| 22:42.13 | *** join/#asterisk JerJer (~PhatJ@asterisk/original-h323-guy/JerJer) |
| 22:42.28 | JerJer | is there a way to disable fax detection via dialplan ? |
| 22:42.33 | JerJer | (like per call) |
| 22:42.36 | *** join/#asterisk hluesea (~hulusikah@88.247.127.66) |
| 22:42.36 | raden_work | what if i wanted in in source in my home directory ? |
| 22:43.26 | raden_work | why is it i always find the tutorials online that dont work ? |
| 22:44.08 | niekvlessert | you first create the directory SOURCE on the server you want to copy to |
| 22:44.26 | russellb | JerJer: not that I know of |
| 22:44.40 | JerJer | might be a nice option |
| 22:44.47 | p3nguin | ssh user@remotehost mkdir SOURCE && scp localfile user@remotehost:SOURCE/ |
| 22:44.51 | p3nguin | raden_work: ^ |
| 22:45.15 | raden_work | SOURCE is there |
| 22:45.21 | p3nguin | raden_work: Alternatively you can copy a local directory recursively. |
| 22:45.34 | p3nguin | "man scp" |
| 22:45.38 | p3nguin | It's all in there. |
| 22:45.48 | raden_work | jon@Server200:/SERVER100/COMPUTER/ASTERISK> scp asterisk-addons-1.6.0.4.tar.gz tss@192.168.1.38:SOURCE/ |
| 22:45.48 | raden_work | Password: |
| 22:45.48 | raden_work | scp: SOURCE//asterisk-addons-1.6.0.4.tar.gz: Permission denied |
| 22:46.27 | p3nguin | Maybe you should try a different file and a different destination just to see how things work. |
| 22:46.41 | raden_work | nm SOURCE owned by root jessus |
| 22:46.49 | p3nguin | It's seriously not rocket surgery. |
| 22:46.54 | niekvlessert | here you go :) |
| 22:47.06 | raden_work | p3nguin, i know it was cause SOURCE owned by root |
| 22:47.10 | raden_work | im retarded :( |
| 22:47.24 | niekvlessert | i feel a little better now :) |
| 22:47.34 | niekvlessert | i feel like you when i'm in the dev channel |
| 22:47.39 | p3nguin | hahaha why did you feel bad at all? |
| 22:47.39 | *** join/#asterisk jcarter1 (~bla@c-68-48-2-59.hsd1.md.comcast.net) |
| 22:48.24 | raden_work | p3nguin, thanks all working :) |
| 22:48.53 | p3nguin | Now you need to learn about rsync for copying files. |
| 22:49.03 | raden_work | i have used it a lil |
| 22:49.07 | raden_work | i will look into more :) |
| 22:49.44 | niekvlessert | rsync is a great tool too |
| 22:49.58 | niekvlessert | rsync with hard linking |
| 22:50.01 | niekvlessert | SO beautiful |
| 22:51.00 | niekvlessert | p3nguin: i was asking stuff about changing TON/NPI in libpri, my c knowledge it not that good to say the least... |
| 22:51.20 | p3nguin | Yeah, I'm an admin, not a programmer... so I know what you mean. |
| 22:52.01 | *** join/#asterisk citywok (~chatzilla@vpn.csgopenline.com) |
| 22:53.05 | raden_work | wow the readme for the asterisk addons has nothing telling how to install |
| 22:53.21 | jcarter1 | does anyone have experience with x-lite? |
| 22:53.21 | russellb | raden_work: sounds like an opportunity for your first patch! |
| 22:54.32 | raden_work | russellb, im documenting this whole project so there will be plenty of PDF howtos in about a month on how to setup asterisk from the ground up to a ITSP provider |
| 22:54.36 | p3nguin | Oh, I should make a patch to correct grammar usage in iax.conf.sample. |
| 22:54.46 | raden_work | p3nguin, :) |
| 22:55.42 | raden_work | p3nguin, im actually writing a pdf on howto setup a asterisk server from console from source to getting a simple 2 phone system with SIP provider and simple IVR & voicemail going |
| 22:56.07 | jcarter1 | nice raden |
| 22:56.27 | raden_work | jcarter1, the simple one or the itsp one ? |
| 22:56.40 | jcarter1 | the simple one |
| 22:56.46 | raden_work | good :) |
| 22:56.46 | p3nguin | I should also write a patch for raden_work's source code so he can correctly use "an" in front of works that begin with vowel sounds... but that's for another day. |
| 22:56.55 | p3nguin | words, rather |
| 22:57.01 | jcarter1 | i figured out how to get it running, it even registers with one of the users i created in the sip conf |
| 22:57.02 | raden_work | p3nguin, Thanks :) |
| 22:57.14 | p3nguin | writes a patch for himself to help with typos |
| 22:57.33 | raden_work | p3nguin, i care in business documents. In here my grammar is not so good. |
| 22:57.54 | p3nguin | If you don't know it here, how can you get it right in documentation? |
| 22:58.05 | raden_work | p3nguin, I am making the stupidest command line mistakes today :( |
| 22:58.06 | p3nguin | Also, what is your native language? |
| 22:58.24 | raden_work | p3nguin, English |
| 22:58.28 | p3nguin | oh |
| 22:58.37 | raden_work | p3nguin, is there a problem with my language ? |
| 22:58.46 | jcarter1 | now i want to call either another client or myself and have the audio being streamed through a custom program (that cuts out high and low wavelengths) before it reaches the destination |
| 22:58.50 | raden_work | for some reason when chatting all goes to hell |
| 22:58.52 | p3nguin | I'm suddenly reminded of a King of the Hill episode. |
| 22:59.02 | raden_work | I write business documents they are usually flawless |
| 22:59.15 | raden_work | p3nguin, rock on, what one ? |
| 22:59.17 | jcarter1 | does anyone have experience with streaming the audio of a call to a custom program? |
| 22:59.25 | raden_work | p3nguin, or should i say which one ? |
| 22:59.27 | raden_work | lol |
| 22:59.36 | jcarter1 | i found exec(), but i am not sure if thats what i am looking for |
| 23:00.23 | raden_work | Ok, I should not be allowed to touch a keyboard today |
| 23:00.35 | p3nguin | raden_work: I don't know the episode name, but Hank says, "An F in English? Bobby, you speak English!" |
| 23:00.54 | raden_work | LMAO :) |
| 23:00.59 | raden_work | sweet dude |
| 23:01.11 | raden_work | I can see Hank Hill saying that |
| 23:01.54 | *** join/#asterisk KingDavidNYC (~Chris1232@69.193.218.18) |
| 23:02.52 | Nugget | heh |
| 23:03.03 | dlynes_laptop | CunningPike, snow's not too bad |
| 23:03.15 | niekvlessert | i'm noob also |
| 23:03.17 | dlynes_laptop | CunningPike, the lack of negative cash flow makes it more enjoyable :) |
| 23:03.24 | niekvlessert | how do i force a phone to reregister again |
| 23:03.55 | dlynes_laptop | niekvlessert, reboot it |
| 23:04.06 | niekvlessert | no quicker way? |
| 23:04.15 | dlynes_laptop | niekvlessert, that's not quick enough? |
| 23:04.20 | niekvlessert | no, aastra boot slow :) |
| 23:04.34 | dlynes_laptop | niekvlessert, aastra's still boot faster than others |
| 23:04.45 | dlynes_laptop | niekvlessert, try rebooting a mediatrix box |
| 23:05.03 | niekvlessert | try rebooting windows xp when it's been running for 2 years ;) |
| 23:05.11 | p3nguin | What? |
| 23:05.17 | p3nguin | XP won't run for two years! |
| 23:05.22 | niekvlessert | :) |
| 23:05.39 | dlynes_laptop | p3nguin, that's why it takes so long to reboot it after it's been running for two years |
| 23:05.39 | niekvlessert | i mean with going on/of |
| 23:05.56 | p3nguin | It might idle that long if automatic updates have been disabled, but it won't run that long if someone is using it. |
| 23:05.58 | jcarter1 | can anyop |
| 23:06.06 | jcarter1 | can anyone please help me with my asterisk question? |
| 23:06.16 | p3nguin | ~ask |
| 23:06.17 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 23:06.24 | dlynes_laptop | jcarter1, if you had one, maybe |
| 23:06.35 | jcarter1 | i already asked :) |
| 23:06.43 | p3nguin | scrolls up |
| 23:06.54 | dlynes_laptop | jcarter1, exec()'s not waht you should be looking for |
| 23:06.58 | dlynes_laptop | jcarter1, shoutcast |
| 23:07.03 | jcarter1 | thought so |
| 23:07.05 | dlynes_laptop | jcarter1, music on hold |
| 23:07.07 | jcarter1 | ah - |
| 23:07.09 | jcarter1 | *looking |
| 23:07.20 | dlynes_laptop | jcarter1, and other similar sound servers |
| 23:07.30 | dlynes_laptop | jcarter1, you stream it to music on hold |
| 23:07.32 | hardwire | meh. |
| 23:07.50 | jcarter1 | ok i am looking for it - thnks for the tip |
| 23:08.25 | dlynes_laptop | jcarter1, try the following page on voip-info.org (it'll be a bit dated): http://www.voip-info.org/wiki/index.php?page_id=107 |
| 23:09.20 | jcarter1 | yeah i just looked at it |
| 23:10.35 | raden_work | p3nguin, my windows box been up 45 days miracle :) |
| 23:11.44 | niekvlessert | you guys have looked into REAL free music? |
| 23:12.11 | *** join/#asterisk jax55_ (~jax55@c-98-254-160-125.hsd1.fl.comcast.net) |
| 23:16.53 | raden_work | ok i have installed asterisk addons now what lol |
| 23:17.03 | *** join/#asterisk rossand (~aross@74.198.12.4) |
| 23:17.08 | raden_work | i soo need to make some documentation for this |
| 23:17.39 | p3nguin | Did you remember to install dahdi and dahdi-tools so you can have a timer? |
| 23:21.31 | dlynes_laptop | niekvlessert, as opposed to fake free music? |
| 23:22.10 | niekvlessert | no, when music seems free, and the author says it's free, it's still not free sometimes |
| 23:22.17 | dlynes_laptop | jcarter1, what exactly are you looking for? I thought you were looking for music on hold... |
| 23:22.35 | dlynes_laptop | niekvlessert, ummm...how so? |
| 23:22.48 | niekvlessert | look up creative common license |
| 23:22.54 | Katty | peeks in |
| 23:22.58 | dlynes_laptop | and? |
| 23:23.06 | jcarter1 | i want to write a program that takes a audiostream and manipulates it in a way that it cuts of high and/or low wavelength |
| 23:23.13 | niekvlessert | well then you can read yourselve what's going on :) |
| 23:23.21 | jcarter1 | that audiostream should be the call i do via asterisk |
| 23:23.24 | dlynes_laptop | jcarter1, so basically you want to convert it to mp3 |
| 23:23.31 | Katty | i think i must have walked into a warzone |
| 23:23.44 | jcarter1 | no |
| 23:23.46 | dlynes_laptop | jcarter1, and then create a call to someone, and just play the recording back to them |
| 23:24.01 | jcarter1 | no - i want this to be "on the fly" |
| 23:24.14 | jcarter1 | while i am speaking into the phone i want this to happen |
| 23:24.24 | Katty | well you know what they say |
| 23:24.25 | dlynes_laptop | you need to explain what you need a little bit better, then |
| 23:24.26 | jcarter1 | and be transmitted to the destination |
| 23:24.27 | Katty | you can want in one hand.... |
| 23:24.29 | raden_work | p3nguin, its all SIP do i still need dahdi ? |
| 23:24.47 | p3nguin | raden_work: If you ever want to do conference, yes. |
| 23:25.08 | raden_work | p3nguin, just 3 way or more than 3 ? |
| 23:25.09 | p3nguin | raden_work: MeetMe requires a timer, which dahdi_dummy provides. To get dahdi_dummy, you need dahdi. |
| 23:25.31 | raden_work | p3nguin, but 3 way does not need dahdi correct ? |
| 23:25.34 | dlynes_laptop | jcarter1, You want buddy to talk into the phone, strip off his high and low tones, and then retransmit it to someone on a different phone conversation? |
| 23:25.40 | jcarter1 | hm - ok let me try it this way: i want to cut out background sounds during a call |
| 23:25.43 | jaskew | jcarter: why do you need to trim the highs & lows if you are getting the audio from a phone? There shouldn't be any aliasing. Are you looking for a "sepcial effect" type sound? |
| 23:25.46 | raden_work | p3nguin, this is being used more as just a switch than anything |
| 23:25.59 | jaskew | jcarter: nm - you answered it |
| 23:26.02 | raden_work | p3nguin, any place with more than 4 lines will have a small itx asterisk box at there location |
| 23:26.04 | dlynes_laptop | jcarter1, that's completely different than cutting out high and low frequencies |
| 23:26.17 | raden_work | p3nguin, is there any other reason id need dahdi ? |
| 23:26.26 | dlynes_laptop | jcarter1, background noise isn't necessarily high and low freq's |
| 23:26.29 | jcarter1 | sorry, i maybe said something wrong. What i want to do is cut out background sounds during a conversation |
| 23:27.16 | jaskew | jcarter: that might be hard to do unless the background sound is a specific frequency. Even then... |
| 23:27.26 | niekvlessert | raden_work: no, only for timing, for a few applications |
| 23:27.43 | niekvlessert | did you do make menuselect, you can see what you couldn't compile because of dependencies |
| 23:27.59 | jcarter1 | i can use whatever backgroundsound i want |
| 23:28.08 | jcarter1 | so it is a specific frequency |
| 23:28.11 | *** join/#asterisk ruben23 (~AGENT@122.55.48.243) |
| 23:28.39 | raden_work | niekvlessert, make menu on the addons or asterisk core ? |
| 23:28.46 | p3nguin | raden_work: It's really a trivial thing to install dahdi, so why not include it for good measure? |
| 23:28.47 | raden_work | when i did the core i did |
| 23:28.47 | niekvlessert | core |
| 23:28.53 | niekvlessert | ok |
| 23:29.00 | raden_work | well how do i install or tell if it is instaled ? |
| 23:29.06 | ruben23 | hi there |
| 23:29.12 | raden_work | p3nguin, good point |
| 23:29.31 | [TK]D-Fender | RadenWere you present when you installed it? Did you see yourself do it? Do you have any witnesses? |
| 23:29.51 | p3nguin | raden_work: When I first installed Asterisk, I didn't expect to need a timer... then I decided to do MeetMe for conferencing, and I had to figure out how to get a timer installed. |
| 23:29.59 | jaskew | <PROTECTED> |
| 23:30.10 | raden_work | how do i tell if dahdi is installed ? |
| 23:30.16 | p3nguin | Did you install it? |
| 23:30.21 | raden_work | idk |
| 23:30.33 | raden_work | isnt there a simple command to tell |
| 23:30.38 | jcarter1 | yeah dont ask - i am supposed to do it. and to tell if its working i want to be able to turn the filter off and on |
| 23:30.55 | [TK]D-Fender | raden_work: dahdi_cfg -vvvv |
| 23:31.29 | jaskew | jcarter: I"m looking at sox, but I'm not sure it will handle streaming audio |
| 23:31.32 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 23:32.27 | *** part/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 23:32.53 | raden_work | solves that command not found |
| 23:32.54 | jcarter1 | looks like it needs a file as input |
| 23:33.01 | jcarter1 | jaskew |
| 23:33.33 | jaskew | jcarter: look at ffmpeg or VLC. I think one or both of thise will handle streaming audio and let you apply filters. |
| 23:33.46 | jcarter1 | well - if streaming doesnt work, i will have to record something i say and THEN play it back after it was used by the program |
| 23:33.55 | raden_work | is the onlyway to install dahdi to reinstall asterisk ? |
| 23:34.04 | jaskew | jcarter: then you need to find a way to pipe it in & out of asterisk. I'm sorry but I don;t know how to do that offhand |
| 23:34.25 | [TK]D-Fender | raden_work: No, but the only way to get Asterisk to acknowledge DAHDI is to compile it after |
| 23:35.00 | jcarter1 | hm too bad - one would think that this should be easy to do. I will google some more. Thanks for the help so far |
| 23:35.15 | raden_work | so best way is to make menuselect at beginging ? |
| 23:35.29 | jaskew | jcarter: record/playback might be easier. There's an echo application out there. You could probably build off of that and use SOX in the middie to filter |
| 23:36.55 | jcarter1 | is sox able to filter an audio file? |
| 23:38.15 | jaskew | jacrter: yes |
| 23:39.57 | raden_work | <PROTECTED> |
| 23:40.10 | raden_work | how do i make the 3 xxx's go away |
| 23:40.38 | jaskew | jcarter: do a 'man soxeffect' and there are a number of filtering options (high, low, bandpass) |
| 23:40.47 | raden_work | <PROTECTED> |
| 23:40.47 | raden_work | <PROTECTED> |
| 23:40.47 | raden_work | <PROTECTED> |
| 23:41.34 | jaskew | jcarter: and of course 'removeBackground' |
| 23:41.37 | raden_work | do i need OBDC storage in voicemail options to use mysql to store voicemail ? |
| 23:42.20 | jaskew | jcarter: OK - I'm kidding about that one... but there is a bandreject and lots of others. |
| 23:46.35 | raden_work | [TK]D-Fender, im presuming i must install dandhi package then install asterisk again ? |
| 23:49.32 | *** join/#asterisk moy (~moy@74.12.123.196) |
| 23:49.50 | [TK]D-Fender | Radencorrect |
| 23:50.33 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 23:55.44 | raden_work | thanks [TK]D-Fender |
| 23:56.24 | niekvlessert | how do i call a macro? |
| 23:56.25 | russellb | raden_work: be sure to re-run the asterisk configure script after installing dahdi, that is what determines what you have installed |
| 23:56.28 | niekvlessert | ${CONNECTED_LINE_SEND_CALLER_MACRO} Macro to call before sending a connected line update to the caller |
| 23:57.12 | [TK]D-Fender | niekvlessert: "core show application macro" |
| 23:57.38 | russellb | niekvlessert: in that case, you're just setting a variable with the name of a macro and it is called automatically for you at the right time during a call |
| 23:58.12 | jaskew | >>niekvlessert: how do i call a macro? Try "Here macro macro macro, c'mon macro..." |
| 23:58.27 | niekvlessert | jaskew: already tried that |
| 23:58.39 | jaskew | heh - must be a cat macro |
